AudioFlinger.cpp revision 8c32734c1e2dda852011fc46d0caded971464bc2
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include <media/nbaio/AudioStreamOutSink.h> 80#include <media/nbaio/MonoPipe.h> 81#include <media/nbaio/MonoPipeReader.h> 82#include <media/nbaio/Pipe.h> 83#include <media/nbaio/PipeReader.h> 84#include <media/nbaio/SourceAudioBufferProvider.h> 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 169// for the track. The client then sub-divides this into smaller buffers for its use. 170// Currently the client uses double-buffering by default, but doesn't tell us about that. 171// So for now we just assume that client is double-buffered. 172// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 173// N-buffering, so AudioFlinger could allocate the right amount of memory. 174// See the client's minBufCount and mNotificationFramesAct calculations for details. 175static const int kFastTrackMultiplier = 2; 176 177// ---------------------------------------------------------------------------- 178 179#ifdef ADD_BATTERY_DATA 180// To collect the amplifier usage 181static void addBatteryData(uint32_t params) { 182 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 183 if (service == NULL) { 184 // it already logged 185 return; 186 } 187 188 service->addBatteryData(params); 189} 190#endif 191 192static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 193{ 194 const hw_module_t *mod; 195 int rc; 196 197 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 198 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 199 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 200 if (rc) { 201 goto out; 202 } 203 rc = audio_hw_device_open(mod, dev); 204 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 205 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 206 if (rc) { 207 goto out; 208 } 209 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 210 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 211 rc = BAD_VALUE; 212 goto out; 213 } 214 return 0; 215 216out: 217 *dev = NULL; 218 return rc; 219} 220 221// ---------------------------------------------------------------------------- 222 223AudioFlinger::AudioFlinger() 224 : BnAudioFlinger(), 225 mPrimaryHardwareDev(NULL), 226 mHardwareStatus(AUDIO_HW_IDLE), 227 mMasterVolume(1.0f), 228 mMasterMute(false), 229 mNextUniqueId(1), 230 mMode(AUDIO_MODE_INVALID), 231 mBtNrecIsOff(false) 232{ 233} 234 235void AudioFlinger::onFirstRef() 236{ 237 int rc = 0; 238 239 Mutex::Autolock _l(mLock); 240 241 /* TODO: move all this work into an Init() function */ 242 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 243 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 244 uint32_t int_val; 245 if (1 == sscanf(val_str, "%u", &int_val)) { 246 mStandbyTimeInNsecs = milliseconds(int_val); 247 ALOGI("Using %u mSec as standby time.", int_val); 248 } else { 249 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 250 ALOGI("Using default %u mSec as standby time.", 251 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 252 } 253 } 254 255 mMode = AUDIO_MODE_NORMAL; 256} 257 258AudioFlinger::~AudioFlinger() 259{ 260 while (!mRecordThreads.isEmpty()) { 261 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 262 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 263 } 264 while (!mPlaybackThreads.isEmpty()) { 265 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 266 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 267 } 268 269 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 270 // no mHardwareLock needed, as there are no other references to this 271 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 272 delete mAudioHwDevs.valueAt(i); 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367static bool tryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = tryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = tryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 dumpClients(fd, args); 403 dumpInternals(fd, args); 404 405 // dump playback threads 406 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 407 mPlaybackThreads.valueAt(i)->dump(fd, args); 408 } 409 410 // dump record threads 411 for (size_t i = 0; i < mRecordThreads.size(); i++) { 412 mRecordThreads.valueAt(i)->dump(fd, args); 413 } 414 415 // dump all hardware devs 416 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 417 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 418 dev->dump(dev, fd); 419 } 420 421 // dump the serially shared record tee sink 422 if (mRecordTeeSource != 0) { 423 dumpTee(fd, mRecordTeeSource); 424 } 425 426 if (locked) mLock.unlock(); 427 } 428 return NO_ERROR; 429} 430 431sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 432{ 433 // If pid is already in the mClients wp<> map, then use that entry 434 // (for which promote() is always != 0), otherwise create a new entry and Client. 435 sp<Client> client = mClients.valueFor(pid).promote(); 436 if (client == 0) { 437 client = new Client(this, pid); 438 mClients.add(pid, client); 439 } 440 441 return client; 442} 443 444// IAudioFlinger interface 445 446 447sp<IAudioTrack> AudioFlinger::createTrack( 448 pid_t pid, 449 audio_stream_type_t streamType, 450 uint32_t sampleRate, 451 audio_format_t format, 452 audio_channel_mask_t channelMask, 453 int frameCount, 454 IAudioFlinger::track_flags_t flags, 455 const sp<IMemory>& sharedBuffer, 456 audio_io_handle_t output, 457 pid_t tid, 458 int *sessionId, 459 status_t *status) 460{ 461 sp<PlaybackThread::Track> track; 462 sp<TrackHandle> trackHandle; 463 sp<Client> client; 464 status_t lStatus; 465 int lSessionId; 466 467 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 468 // but if someone uses binder directly they could bypass that and cause us to crash 469 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 470 ALOGE("createTrack() invalid stream type %d", streamType); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 { 476 Mutex::Autolock _l(mLock); 477 PlaybackThread *thread = checkPlaybackThread_l(output); 478 PlaybackThread *effectThread = NULL; 479 if (thread == NULL) { 480 ALOGE("unknown output thread"); 481 lStatus = BAD_VALUE; 482 goto Exit; 483 } 484 485 client = registerPid_l(pid); 486 487 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 488 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 489 // check if an effect chain with the same session ID is present on another 490 // output thread and move it here. 491 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 492 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 493 if (mPlaybackThreads.keyAt(i) != output) { 494 uint32_t sessions = t->hasAudioSession(*sessionId); 495 if (sessions & PlaybackThread::EFFECT_SESSION) { 496 effectThread = t.get(); 497 break; 498 } 499 } 500 } 501 lSessionId = *sessionId; 502 } else { 503 // if no audio session id is provided, create one here 504 lSessionId = nextUniqueId(); 505 if (sessionId != NULL) { 506 *sessionId = lSessionId; 507 } 508 } 509 ALOGV("createTrack() lSessionId: %d", lSessionId); 510 511 track = thread->createTrack_l(client, streamType, sampleRate, format, 512 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 513 514 // move effect chain to this output thread if an effect on same session was waiting 515 // for a track to be created 516 if (lStatus == NO_ERROR && effectThread != NULL) { 517 Mutex::Autolock _dl(thread->mLock); 518 Mutex::Autolock _sl(effectThread->mLock); 519 moveEffectChain_l(lSessionId, effectThread, thread, true); 520 } 521 522 // Look for sync events awaiting for a session to be used. 523 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 524 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 525 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 526 if (lStatus == NO_ERROR) { 527 (void) track->setSyncEvent(mPendingSyncEvents[i]); 528 } else { 529 mPendingSyncEvents[i]->cancel(); 530 } 531 mPendingSyncEvents.removeAt(i); 532 i--; 533 } 534 } 535 } 536 } 537 if (lStatus == NO_ERROR) { 538 trackHandle = new TrackHandle(track); 539 } else { 540 // remove local strong reference to Client before deleting the Track so that the Client 541 // destructor is called by the TrackBase destructor with mLock held 542 client.clear(); 543 track.clear(); 544 } 545 546Exit: 547 if (status != NULL) { 548 *status = lStatus; 549 } 550 return trackHandle; 551} 552 553uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 554{ 555 Mutex::Autolock _l(mLock); 556 PlaybackThread *thread = checkPlaybackThread_l(output); 557 if (thread == NULL) { 558 ALOGW("sampleRate() unknown thread %d", output); 559 return 0; 560 } 561 return thread->sampleRate(); 562} 563 564int AudioFlinger::channelCount(audio_io_handle_t output) const 565{ 566 Mutex::Autolock _l(mLock); 567 PlaybackThread *thread = checkPlaybackThread_l(output); 568 if (thread == NULL) { 569 ALOGW("channelCount() unknown thread %d", output); 570 return 0; 571 } 572 return thread->channelCount(); 573} 574 575audio_format_t AudioFlinger::format(audio_io_handle_t output) const 576{ 577 Mutex::Autolock _l(mLock); 578 PlaybackThread *thread = checkPlaybackThread_l(output); 579 if (thread == NULL) { 580 ALOGW("format() unknown thread %d", output); 581 return AUDIO_FORMAT_INVALID; 582 } 583 return thread->format(); 584} 585 586size_t AudioFlinger::frameCount(audio_io_handle_t output) const 587{ 588 Mutex::Autolock _l(mLock); 589 PlaybackThread *thread = checkPlaybackThread_l(output); 590 if (thread == NULL) { 591 ALOGW("frameCount() unknown thread %d", output); 592 return 0; 593 } 594 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 595 // should examine all callers and fix them to handle smaller counts 596 return thread->frameCount(); 597} 598 599uint32_t AudioFlinger::latency(audio_io_handle_t output) const 600{ 601 Mutex::Autolock _l(mLock); 602 PlaybackThread *thread = checkPlaybackThread_l(output); 603 if (thread == NULL) { 604 ALOGW("latency() unknown thread %d", output); 605 return 0; 606 } 607 return thread->latency(); 608} 609 610status_t AudioFlinger::setMasterVolume(float value) 611{ 612 status_t ret = initCheck(); 613 if (ret != NO_ERROR) { 614 return ret; 615 } 616 617 // check calling permissions 618 if (!settingsAllowed()) { 619 return PERMISSION_DENIED; 620 } 621 622 Mutex::Autolock _l(mLock); 623 mMasterVolume = value; 624 625 // Set master volume in the HALs which support it. 626 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 627 AutoMutex lock(mHardwareLock); 628 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 629 630 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 631 if (dev->canSetMasterVolume()) { 632 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 633 } 634 mHardwareStatus = AUDIO_HW_IDLE; 635 } 636 637 // Now set the master volume in each playback thread. Playback threads 638 // assigned to HALs which do not have master volume support will apply 639 // master volume during the mix operation. Threads with HALs which do 640 // support master volume will simply ignore the setting. 641 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 643 644 return NO_ERROR; 645} 646 647status_t AudioFlinger::setMode(audio_mode_t mode) 648{ 649 status_t ret = initCheck(); 650 if (ret != NO_ERROR) { 651 return ret; 652 } 653 654 // check calling permissions 655 if (!settingsAllowed()) { 656 return PERMISSION_DENIED; 657 } 658 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 659 ALOGW("Illegal value: setMode(%d)", mode); 660 return BAD_VALUE; 661 } 662 663 { // scope for the lock 664 AutoMutex lock(mHardwareLock); 665 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 666 mHardwareStatus = AUDIO_HW_SET_MODE; 667 ret = dev->set_mode(dev, mode); 668 mHardwareStatus = AUDIO_HW_IDLE; 669 } 670 671 if (NO_ERROR == ret) { 672 Mutex::Autolock _l(mLock); 673 mMode = mode; 674 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 675 mPlaybackThreads.valueAt(i)->setMode(mode); 676 } 677 678 return ret; 679} 680 681status_t AudioFlinger::setMicMute(bool state) 682{ 683 status_t ret = initCheck(); 684 if (ret != NO_ERROR) { 685 return ret; 686 } 687 688 // check calling permissions 689 if (!settingsAllowed()) { 690 return PERMISSION_DENIED; 691 } 692 693 AutoMutex lock(mHardwareLock); 694 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 695 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 696 ret = dev->set_mic_mute(dev, state); 697 mHardwareStatus = AUDIO_HW_IDLE; 698 return ret; 699} 700 701bool AudioFlinger::getMicMute() const 702{ 703 status_t ret = initCheck(); 704 if (ret != NO_ERROR) { 705 return false; 706 } 707 708 bool state = AUDIO_MODE_INVALID; 709 AutoMutex lock(mHardwareLock); 710 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 711 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 712 dev->get_mic_mute(dev, &state); 713 mHardwareStatus = AUDIO_HW_IDLE; 714 return state; 715} 716 717status_t AudioFlinger::setMasterMute(bool muted) 718{ 719 status_t ret = initCheck(); 720 if (ret != NO_ERROR) { 721 return ret; 722 } 723 724 // check calling permissions 725 if (!settingsAllowed()) { 726 return PERMISSION_DENIED; 727 } 728 729 Mutex::Autolock _l(mLock); 730 mMasterMute = muted; 731 732 // Set master mute in the HALs which support it. 733 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 734 AutoMutex lock(mHardwareLock); 735 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 736 737 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 738 if (dev->canSetMasterMute()) { 739 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 740 } 741 mHardwareStatus = AUDIO_HW_IDLE; 742 } 743 744 // Now set the master mute in each playback thread. Playback threads 745 // assigned to HALs which do not have master mute support will apply master 746 // mute during the mix operation. Threads with HALs which do support master 747 // mute will simply ignore the setting. 748 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 749 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 750 751 return NO_ERROR; 752} 753 754float AudioFlinger::masterVolume() const 755{ 756 Mutex::Autolock _l(mLock); 757 return masterVolume_l(); 758} 759 760bool AudioFlinger::masterMute() const 761{ 762 Mutex::Autolock _l(mLock); 763 return masterMute_l(); 764} 765 766float AudioFlinger::masterVolume_l() const 767{ 768 return mMasterVolume; 769} 770 771bool AudioFlinger::masterMute_l() const 772{ 773 return mMasterMute; 774} 775 776status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 777 audio_io_handle_t output) 778{ 779 // check calling permissions 780 if (!settingsAllowed()) { 781 return PERMISSION_DENIED; 782 } 783 784 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 785 ALOGE("setStreamVolume() invalid stream %d", stream); 786 return BAD_VALUE; 787 } 788 789 AutoMutex lock(mLock); 790 PlaybackThread *thread = NULL; 791 if (output) { 792 thread = checkPlaybackThread_l(output); 793 if (thread == NULL) { 794 return BAD_VALUE; 795 } 796 } 797 798 mStreamTypes[stream].volume = value; 799 800 if (thread == NULL) { 801 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 802 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 803 } 804 } else { 805 thread->setStreamVolume(stream, value); 806 } 807 808 return NO_ERROR; 809} 810 811status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 812{ 813 // check calling permissions 814 if (!settingsAllowed()) { 815 return PERMISSION_DENIED; 816 } 817 818 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 819 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 820 ALOGE("setStreamMute() invalid stream %d", stream); 821 return BAD_VALUE; 822 } 823 824 AutoMutex lock(mLock); 825 mStreamTypes[stream].mute = muted; 826 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 827 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 828 829 return NO_ERROR; 830} 831 832float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 833{ 834 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 835 return 0.0f; 836 } 837 838 AutoMutex lock(mLock); 839 float volume; 840 if (output) { 841 PlaybackThread *thread = checkPlaybackThread_l(output); 842 if (thread == NULL) { 843 return 0.0f; 844 } 845 volume = thread->streamVolume(stream); 846 } else { 847 volume = streamVolume_l(stream); 848 } 849 850 return volume; 851} 852 853bool AudioFlinger::streamMute(audio_stream_type_t stream) const 854{ 855 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 856 return true; 857 } 858 859 AutoMutex lock(mLock); 860 return streamMute_l(stream); 861} 862 863status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 864{ 865 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 866 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 867 // check calling permissions 868 if (!settingsAllowed()) { 869 return PERMISSION_DENIED; 870 } 871 872 // ioHandle == 0 means the parameters are global to the audio hardware interface 873 if (ioHandle == 0) { 874 Mutex::Autolock _l(mLock); 875 status_t final_result = NO_ERROR; 876 { 877 AutoMutex lock(mHardwareLock); 878 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 879 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 880 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 881 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 882 final_result = result ?: final_result; 883 } 884 mHardwareStatus = AUDIO_HW_IDLE; 885 } 886 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 887 AudioParameter param = AudioParameter(keyValuePairs); 888 String8 value; 889 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 890 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 891 if (mBtNrecIsOff != btNrecIsOff) { 892 for (size_t i = 0; i < mRecordThreads.size(); i++) { 893 sp<RecordThread> thread = mRecordThreads.valueAt(i); 894 audio_devices_t device = thread->inDevice(); 895 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 896 // collect all of the thread's session IDs 897 KeyedVector<int, bool> ids = thread->sessionIds(); 898 // suspend effects associated with those session IDs 899 for (size_t j = 0; j < ids.size(); ++j) { 900 int sessionId = ids.keyAt(j); 901 thread->setEffectSuspended(FX_IID_AEC, 902 suspend, 903 sessionId); 904 thread->setEffectSuspended(FX_IID_NS, 905 suspend, 906 sessionId); 907 } 908 } 909 mBtNrecIsOff = btNrecIsOff; 910 } 911 } 912 String8 screenState; 913 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 914 bool isOff = screenState == "off"; 915 if (isOff != (gScreenState & 1)) { 916 gScreenState = ((gScreenState & ~1) + 2) | isOff; 917 } 918 } 919 return final_result; 920 } 921 922 // hold a strong ref on thread in case closeOutput() or closeInput() is called 923 // and the thread is exited once the lock is released 924 sp<ThreadBase> thread; 925 { 926 Mutex::Autolock _l(mLock); 927 thread = checkPlaybackThread_l(ioHandle); 928 if (thread == 0) { 929 thread = checkRecordThread_l(ioHandle); 930 } else if (thread == primaryPlaybackThread_l()) { 931 // indicate output device change to all input threads for pre processing 932 AudioParameter param = AudioParameter(keyValuePairs); 933 int value; 934 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 935 (value != 0)) { 936 for (size_t i = 0; i < mRecordThreads.size(); i++) { 937 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 938 } 939 } 940 } 941 } 942 if (thread != 0) { 943 return thread->setParameters(keyValuePairs); 944 } 945 return BAD_VALUE; 946} 947 948String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 949{ 950 ALOGVV("getParameters() io %d, keys %s, tid %d, calling pid %d", 951 ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 952 953 Mutex::Autolock _l(mLock); 954 955 if (ioHandle == 0) { 956 String8 out_s8; 957 958 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 959 char *s; 960 { 961 AutoMutex lock(mHardwareLock); 962 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 963 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 964 s = dev->get_parameters(dev, keys.string()); 965 mHardwareStatus = AUDIO_HW_IDLE; 966 } 967 out_s8 += String8(s ? s : ""); 968 free(s); 969 } 970 return out_s8; 971 } 972 973 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 974 if (playbackThread != NULL) { 975 return playbackThread->getParameters(keys); 976 } 977 RecordThread *recordThread = checkRecordThread_l(ioHandle); 978 if (recordThread != NULL) { 979 return recordThread->getParameters(keys); 980 } 981 return String8(""); 982} 983 984size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 985 audio_channel_mask_t channelMask) const 986{ 987 status_t ret = initCheck(); 988 if (ret != NO_ERROR) { 989 return 0; 990 } 991 992 AutoMutex lock(mHardwareLock); 993 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 994 struct audio_config config = { 995 sample_rate: sampleRate, 996 channel_mask: channelMask, 997 format: format, 998 }; 999 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1000 size_t size = dev->get_input_buffer_size(dev, &config); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 return size; 1003} 1004 1005unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1006{ 1007 Mutex::Autolock _l(mLock); 1008 1009 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1010 if (recordThread != NULL) { 1011 return recordThread->getInputFramesLost(); 1012 } 1013 return 0; 1014} 1015 1016status_t AudioFlinger::setVoiceVolume(float value) 1017{ 1018 status_t ret = initCheck(); 1019 if (ret != NO_ERROR) { 1020 return ret; 1021 } 1022 1023 // check calling permissions 1024 if (!settingsAllowed()) { 1025 return PERMISSION_DENIED; 1026 } 1027 1028 AutoMutex lock(mHardwareLock); 1029 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1030 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1031 ret = dev->set_voice_volume(dev, value); 1032 mHardwareStatus = AUDIO_HW_IDLE; 1033 1034 return ret; 1035} 1036 1037status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1038 audio_io_handle_t output) const 1039{ 1040 status_t status; 1041 1042 Mutex::Autolock _l(mLock); 1043 1044 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1045 if (playbackThread != NULL) { 1046 return playbackThread->getRenderPosition(halFrames, dspFrames); 1047 } 1048 1049 return BAD_VALUE; 1050} 1051 1052void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1053{ 1054 1055 Mutex::Autolock _l(mLock); 1056 1057 pid_t pid = IPCThreadState::self()->getCallingPid(); 1058 if (mNotificationClients.indexOfKey(pid) < 0) { 1059 sp<NotificationClient> notificationClient = new NotificationClient(this, 1060 client, 1061 pid); 1062 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1063 1064 mNotificationClients.add(pid, notificationClient); 1065 1066 sp<IBinder> binder = client->asBinder(); 1067 binder->linkToDeath(notificationClient); 1068 1069 // the config change is always sent from playback or record threads to avoid deadlock 1070 // with AudioSystem::gLock 1071 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1072 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1073 } 1074 1075 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1076 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1077 } 1078 } 1079} 1080 1081void AudioFlinger::removeNotificationClient(pid_t pid) 1082{ 1083 Mutex::Autolock _l(mLock); 1084 1085 mNotificationClients.removeItem(pid); 1086 1087 ALOGV("%d died, releasing its sessions", pid); 1088 size_t num = mAudioSessionRefs.size(); 1089 bool removed = false; 1090 for (size_t i = 0; i< num; ) { 1091 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1092 ALOGV(" pid %d @ %d", ref->mPid, i); 1093 if (ref->mPid == pid) { 1094 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1095 mAudioSessionRefs.removeAt(i); 1096 delete ref; 1097 removed = true; 1098 num--; 1099 } else { 1100 i++; 1101 } 1102 } 1103 if (removed) { 1104 purgeStaleEffects_l(); 1105 } 1106} 1107 1108// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1109void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1110{ 1111 size_t size = mNotificationClients.size(); 1112 for (size_t i = 0; i < size; i++) { 1113 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1114 param2); 1115 } 1116} 1117 1118// removeClient_l() must be called with AudioFlinger::mLock held 1119void AudioFlinger::removeClient_l(pid_t pid) 1120{ 1121 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1122 mClients.removeItem(pid); 1123} 1124 1125// getEffectThread_l() must be called with AudioFlinger::mLock held 1126sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1127{ 1128 sp<PlaybackThread> thread; 1129 1130 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1131 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1132 ALOG_ASSERT(thread == 0); 1133 thread = mPlaybackThreads.valueAt(i); 1134 } 1135 } 1136 1137 return thread; 1138} 1139 1140// ---------------------------------------------------------------------------- 1141 1142AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1143 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1144 : Thread(false /*canCallJava*/), 1145 mType(type), 1146 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1147 // mChannelMask 1148 mChannelCount(0), 1149 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1150 mParamStatus(NO_ERROR), 1151 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1152 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1153 // mName will be set by concrete (non-virtual) subclass 1154 mDeathRecipient(new PMDeathRecipient(this)) 1155{ 1156} 1157 1158AudioFlinger::ThreadBase::~ThreadBase() 1159{ 1160 mParamCond.broadcast(); 1161 // do not lock the mutex in destructor 1162 releaseWakeLock_l(); 1163 if (mPowerManager != 0) { 1164 sp<IBinder> binder = mPowerManager->asBinder(); 1165 binder->unlinkToDeath(mDeathRecipient); 1166 } 1167} 1168 1169void AudioFlinger::ThreadBase::exit() 1170{ 1171 ALOGV("ThreadBase::exit"); 1172 // do any cleanup required for exit to succeed 1173 preExit(); 1174 { 1175 // This lock prevents the following race in thread (uniprocessor for illustration): 1176 // if (!exitPending()) { 1177 // // context switch from here to exit() 1178 // // exit() calls requestExit(), what exitPending() observes 1179 // // exit() calls signal(), which is dropped since no waiters 1180 // // context switch back from exit() to here 1181 // mWaitWorkCV.wait(...); 1182 // // now thread is hung 1183 // } 1184 AutoMutex lock(mLock); 1185 requestExit(); 1186 mWaitWorkCV.broadcast(); 1187 } 1188 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1189 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1190 requestExitAndWait(); 1191} 1192 1193status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1194{ 1195 status_t status; 1196 1197 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1198 Mutex::Autolock _l(mLock); 1199 1200 mNewParameters.add(keyValuePairs); 1201 mWaitWorkCV.signal(); 1202 // wait condition with timeout in case the thread loop has exited 1203 // before the request could be processed 1204 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1205 status = mParamStatus; 1206 mWaitWorkCV.signal(); 1207 } else { 1208 status = TIMED_OUT; 1209 } 1210 return status; 1211} 1212 1213void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 1214{ 1215 Mutex::Autolock _l(mLock); 1216 sendIoConfigEvent_l(event, param); 1217} 1218 1219// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 1220void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 1221{ 1222 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 1223 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 1224 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1225 mWaitWorkCV.signal(); 1226} 1227 1228// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 1229void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 1230{ 1231 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 1232 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 1233 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 1234 mConfigEvents.size(), pid, tid, prio); 1235 mWaitWorkCV.signal(); 1236} 1237 1238void AudioFlinger::ThreadBase::processConfigEvents() 1239{ 1240 mLock.lock(); 1241 while (!mConfigEvents.isEmpty()) { 1242 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1243 ConfigEvent *event = mConfigEvents[0]; 1244 mConfigEvents.removeAt(0); 1245 // release mLock before locking AudioFlinger mLock: lock order is always 1246 // AudioFlinger then ThreadBase to avoid cross deadlock 1247 mLock.unlock(); 1248 switch(event->type()) { 1249 case CFG_EVENT_PRIO: { 1250 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 1251 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 1252 if (err != 0) { 1253 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1254 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 1255 } 1256 } break; 1257 case CFG_EVENT_IO: { 1258 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 1259 mAudioFlinger->mLock.lock(); 1260 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 1261 mAudioFlinger->mLock.unlock(); 1262 } break; 1263 default: 1264 ALOGE("processConfigEvents() unknown event type %d", event->type()); 1265 break; 1266 } 1267 delete event; 1268 mLock.lock(); 1269 } 1270 mLock.unlock(); 1271} 1272 1273void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1274{ 1275 const size_t SIZE = 256; 1276 char buffer[SIZE]; 1277 String8 result; 1278 1279 bool locked = tryLock(mLock); 1280 if (!locked) { 1281 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1282 write(fd, buffer, strlen(buffer)); 1283 } 1284 1285 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1286 result.append(buffer); 1287 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1288 result.append(buffer); 1289 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1290 result.append(buffer); 1291 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1292 result.append(buffer); 1293 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1294 result.append(buffer); 1295 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1296 result.append(buffer); 1297 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1298 result.append(buffer); 1299 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1300 result.append(buffer); 1301 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1302 result.append(buffer); 1303 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1304 result.append(buffer); 1305 1306 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1307 result.append(buffer); 1308 result.append(" Index Command"); 1309 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1310 snprintf(buffer, SIZE, "\n %02d ", i); 1311 result.append(buffer); 1312 result.append(mNewParameters[i]); 1313 } 1314 1315 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1316 result.append(buffer); 1317 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1318 mConfigEvents[i]->dump(buffer, SIZE); 1319 result.append(buffer); 1320 } 1321 result.append("\n"); 1322 1323 write(fd, result.string(), result.size()); 1324 1325 if (locked) { 1326 mLock.unlock(); 1327 } 1328} 1329 1330void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1331{ 1332 const size_t SIZE = 256; 1333 char buffer[SIZE]; 1334 String8 result; 1335 1336 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1337 write(fd, buffer, strlen(buffer)); 1338 1339 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1340 sp<EffectChain> chain = mEffectChains[i]; 1341 if (chain != 0) { 1342 chain->dump(fd, args); 1343 } 1344 } 1345} 1346 1347void AudioFlinger::ThreadBase::acquireWakeLock() 1348{ 1349 Mutex::Autolock _l(mLock); 1350 acquireWakeLock_l(); 1351} 1352 1353void AudioFlinger::ThreadBase::acquireWakeLock_l() 1354{ 1355 if (mPowerManager == 0) { 1356 // use checkService() to avoid blocking if power service is not up yet 1357 sp<IBinder> binder = 1358 defaultServiceManager()->checkService(String16("power")); 1359 if (binder == 0) { 1360 ALOGW("Thread %s cannot connect to the power manager service", mName); 1361 } else { 1362 mPowerManager = interface_cast<IPowerManager>(binder); 1363 binder->linkToDeath(mDeathRecipient); 1364 } 1365 } 1366 if (mPowerManager != 0) { 1367 sp<IBinder> binder = new BBinder(); 1368 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1369 binder, 1370 String16(mName)); 1371 if (status == NO_ERROR) { 1372 mWakeLockToken = binder; 1373 } 1374 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1375 } 1376} 1377 1378void AudioFlinger::ThreadBase::releaseWakeLock() 1379{ 1380 Mutex::Autolock _l(mLock); 1381 releaseWakeLock_l(); 1382} 1383 1384void AudioFlinger::ThreadBase::releaseWakeLock_l() 1385{ 1386 if (mWakeLockToken != 0) { 1387 ALOGV("releaseWakeLock_l() %s", mName); 1388 if (mPowerManager != 0) { 1389 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1390 } 1391 mWakeLockToken.clear(); 1392 } 1393} 1394 1395void AudioFlinger::ThreadBase::clearPowerManager() 1396{ 1397 Mutex::Autolock _l(mLock); 1398 releaseWakeLock_l(); 1399 mPowerManager.clear(); 1400} 1401 1402void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1403{ 1404 sp<ThreadBase> thread = mThread.promote(); 1405 if (thread != 0) { 1406 thread->clearPowerManager(); 1407 } 1408 ALOGW("power manager service died !!!"); 1409} 1410 1411void AudioFlinger::ThreadBase::setEffectSuspended( 1412 const effect_uuid_t *type, bool suspend, int sessionId) 1413{ 1414 Mutex::Autolock _l(mLock); 1415 setEffectSuspended_l(type, suspend, sessionId); 1416} 1417 1418void AudioFlinger::ThreadBase::setEffectSuspended_l( 1419 const effect_uuid_t *type, bool suspend, int sessionId) 1420{ 1421 sp<EffectChain> chain = getEffectChain_l(sessionId); 1422 if (chain != 0) { 1423 if (type != NULL) { 1424 chain->setEffectSuspended_l(type, suspend); 1425 } else { 1426 chain->setEffectSuspendedAll_l(suspend); 1427 } 1428 } 1429 1430 updateSuspendedSessions_l(type, suspend, sessionId); 1431} 1432 1433void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1434{ 1435 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1436 if (index < 0) { 1437 return; 1438 } 1439 1440 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1441 mSuspendedSessions.valueAt(index); 1442 1443 for (size_t i = 0; i < sessionEffects.size(); i++) { 1444 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1445 for (int j = 0; j < desc->mRefCount; j++) { 1446 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1447 chain->setEffectSuspendedAll_l(true); 1448 } else { 1449 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1450 desc->mType.timeLow); 1451 chain->setEffectSuspended_l(&desc->mType, true); 1452 } 1453 } 1454 } 1455} 1456 1457void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1458 bool suspend, 1459 int sessionId) 1460{ 1461 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1462 1463 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1464 1465 if (suspend) { 1466 if (index >= 0) { 1467 sessionEffects = mSuspendedSessions.valueAt(index); 1468 } else { 1469 mSuspendedSessions.add(sessionId, sessionEffects); 1470 } 1471 } else { 1472 if (index < 0) { 1473 return; 1474 } 1475 sessionEffects = mSuspendedSessions.valueAt(index); 1476 } 1477 1478 1479 int key = EffectChain::kKeyForSuspendAll; 1480 if (type != NULL) { 1481 key = type->timeLow; 1482 } 1483 index = sessionEffects.indexOfKey(key); 1484 1485 sp<SuspendedSessionDesc> desc; 1486 if (suspend) { 1487 if (index >= 0) { 1488 desc = sessionEffects.valueAt(index); 1489 } else { 1490 desc = new SuspendedSessionDesc(); 1491 if (type != NULL) { 1492 desc->mType = *type; 1493 } 1494 sessionEffects.add(key, desc); 1495 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1496 } 1497 desc->mRefCount++; 1498 } else { 1499 if (index < 0) { 1500 return; 1501 } 1502 desc = sessionEffects.valueAt(index); 1503 if (--desc->mRefCount == 0) { 1504 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1505 sessionEffects.removeItemsAt(index); 1506 if (sessionEffects.isEmpty()) { 1507 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1508 sessionId); 1509 mSuspendedSessions.removeItem(sessionId); 1510 } 1511 } 1512 } 1513 if (!sessionEffects.isEmpty()) { 1514 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1515 } 1516} 1517 1518void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1519 bool enabled, 1520 int sessionId) 1521{ 1522 Mutex::Autolock _l(mLock); 1523 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1524} 1525 1526void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1527 bool enabled, 1528 int sessionId) 1529{ 1530 if (mType != RECORD) { 1531 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1532 // another session. This gives the priority to well behaved effect control panels 1533 // and applications not using global effects. 1534 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1535 // global effects 1536 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1537 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1538 } 1539 } 1540 1541 sp<EffectChain> chain = getEffectChain_l(sessionId); 1542 if (chain != 0) { 1543 chain->checkSuspendOnEffectEnabled(effect, enabled); 1544 } 1545} 1546 1547// ---------------------------------------------------------------------------- 1548 1549AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1550 AudioStreamOut* output, 1551 audio_io_handle_t id, 1552 audio_devices_t device, 1553 type_t type) 1554 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1555 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1556 // mStreamTypes[] initialized in constructor body 1557 mOutput(output), 1558 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1559 mMixerStatus(MIXER_IDLE), 1560 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1561 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1562 mScreenState(gScreenState), 1563 // index 0 is reserved for normal mixer's submix 1564 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1565{ 1566 snprintf(mName, kNameLength, "AudioOut_%X", id); 1567 1568 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1569 // it would be safer to explicitly pass initial masterVolume/masterMute as 1570 // parameter. 1571 // 1572 // If the HAL we are using has support for master volume or master mute, 1573 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1574 // and the mute set to false). 1575 mMasterVolume = audioFlinger->masterVolume_l(); 1576 mMasterMute = audioFlinger->masterMute_l(); 1577 if (mOutput && mOutput->audioHwDev) { 1578 if (mOutput->audioHwDev->canSetMasterVolume()) { 1579 mMasterVolume = 1.0; 1580 } 1581 1582 if (mOutput->audioHwDev->canSetMasterMute()) { 1583 mMasterMute = false; 1584 } 1585 } 1586 1587 readOutputParameters(); 1588 1589 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1590 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1591 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1592 stream = (audio_stream_type_t) (stream + 1)) { 1593 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1594 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1595 } 1596 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1597 // because mAudioFlinger doesn't have one to copy from 1598} 1599 1600AudioFlinger::PlaybackThread::~PlaybackThread() 1601{ 1602 delete [] mMixBuffer; 1603} 1604 1605void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1606{ 1607 dumpInternals(fd, args); 1608 dumpTracks(fd, args); 1609 dumpEffectChains(fd, args); 1610} 1611 1612void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1613{ 1614 const size_t SIZE = 256; 1615 char buffer[SIZE]; 1616 String8 result; 1617 1618 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1619 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1620 const stream_type_t *st = &mStreamTypes[i]; 1621 if (i > 0) { 1622 result.appendFormat(", "); 1623 } 1624 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1625 if (st->mute) { 1626 result.append("M"); 1627 } 1628 } 1629 result.append("\n"); 1630 write(fd, result.string(), result.length()); 1631 result.clear(); 1632 1633 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1634 result.append(buffer); 1635 Track::appendDumpHeader(result); 1636 for (size_t i = 0; i < mTracks.size(); ++i) { 1637 sp<Track> track = mTracks[i]; 1638 if (track != 0) { 1639 track->dump(buffer, SIZE); 1640 result.append(buffer); 1641 } 1642 } 1643 1644 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1645 result.append(buffer); 1646 Track::appendDumpHeader(result); 1647 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1648 sp<Track> track = mActiveTracks[i].promote(); 1649 if (track != 0) { 1650 track->dump(buffer, SIZE); 1651 result.append(buffer); 1652 } 1653 } 1654 write(fd, result.string(), result.size()); 1655 1656 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1657 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1658 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1659 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1660} 1661 1662void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1663{ 1664 const size_t SIZE = 256; 1665 char buffer[SIZE]; 1666 String8 result; 1667 1668 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1669 result.append(buffer); 1670 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1671 result.append(buffer); 1672 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1673 result.append(buffer); 1674 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1675 result.append(buffer); 1676 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1677 result.append(buffer); 1678 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1679 result.append(buffer); 1680 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1681 result.append(buffer); 1682 write(fd, result.string(), result.size()); 1683 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1684 1685 dumpBase(fd, args); 1686} 1687 1688// Thread virtuals 1689status_t AudioFlinger::PlaybackThread::readyToRun() 1690{ 1691 status_t status = initCheck(); 1692 if (status == NO_ERROR) { 1693 ALOGI("AudioFlinger's thread %p ready to run", this); 1694 } else { 1695 ALOGE("No working audio driver found."); 1696 } 1697 return status; 1698} 1699 1700void AudioFlinger::PlaybackThread::onFirstRef() 1701{ 1702 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1703} 1704 1705// ThreadBase virtuals 1706void AudioFlinger::PlaybackThread::preExit() 1707{ 1708 ALOGV(" preExit()"); 1709 // FIXME this is using hard-coded strings but in the future, this functionality will be 1710 // converted to use audio HAL extensions required to support tunneling 1711 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1712} 1713 1714// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1715sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1716 const sp<AudioFlinger::Client>& client, 1717 audio_stream_type_t streamType, 1718 uint32_t sampleRate, 1719 audio_format_t format, 1720 audio_channel_mask_t channelMask, 1721 int frameCount, 1722 const sp<IMemory>& sharedBuffer, 1723 int sessionId, 1724 IAudioFlinger::track_flags_t flags, 1725 pid_t tid, 1726 status_t *status) 1727{ 1728 sp<Track> track; 1729 status_t lStatus; 1730 1731 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1732 1733 // client expresses a preference for FAST, but we get the final say 1734 if (flags & IAudioFlinger::TRACK_FAST) { 1735 if ( 1736 // not timed 1737 (!isTimed) && 1738 // either of these use cases: 1739 ( 1740 // use case 1: shared buffer with any frame count 1741 ( 1742 (sharedBuffer != 0) 1743 ) || 1744 // use case 2: callback handler and frame count is default or at least as large as HAL 1745 ( 1746 (tid != -1) && 1747 ((frameCount == 0) || 1748 (frameCount >= (int) (mFrameCount * kFastTrackMultiplier))) 1749 ) 1750 ) && 1751 // PCM data 1752 audio_is_linear_pcm(format) && 1753 // mono or stereo 1754 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1755 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1756#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1757 // hardware sample rate 1758 (sampleRate == mSampleRate) && 1759#endif 1760 // normal mixer has an associated fast mixer 1761 hasFastMixer() && 1762 // there are sufficient fast track slots available 1763 (mFastTrackAvailMask != 0) 1764 // FIXME test that MixerThread for this fast track has a capable output HAL 1765 // FIXME add a permission test also? 1766 ) { 1767 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1768 if (frameCount == 0) { 1769 frameCount = mFrameCount * kFastTrackMultiplier; 1770 } 1771 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1772 frameCount, mFrameCount); 1773 } else { 1774 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1775 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1776 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1777 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1778 audio_is_linear_pcm(format), 1779 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1780 flags &= ~IAudioFlinger::TRACK_FAST; 1781 // For compatibility with AudioTrack calculation, buffer depth is forced 1782 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1783 // This is probably too conservative, but legacy application code may depend on it. 1784 // If you change this calculation, also review the start threshold which is related. 1785 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1786 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1787 if (minBufCount < 2) { 1788 minBufCount = 2; 1789 } 1790 int minFrameCount = mNormalFrameCount * minBufCount; 1791 if (frameCount < minFrameCount) { 1792 frameCount = minFrameCount; 1793 } 1794 } 1795 } 1796 1797 if (mType == DIRECT) { 1798 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1799 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1800 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1801 "for output %p with format %d", 1802 sampleRate, format, channelMask, mOutput, mFormat); 1803 lStatus = BAD_VALUE; 1804 goto Exit; 1805 } 1806 } 1807 } else { 1808 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1809 if (sampleRate > mSampleRate*2) { 1810 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1811 lStatus = BAD_VALUE; 1812 goto Exit; 1813 } 1814 } 1815 1816 lStatus = initCheck(); 1817 if (lStatus != NO_ERROR) { 1818 ALOGE("Audio driver not initialized."); 1819 goto Exit; 1820 } 1821 1822 { // scope for mLock 1823 Mutex::Autolock _l(mLock); 1824 1825 // all tracks in same audio session must share the same routing strategy otherwise 1826 // conflicts will happen when tracks are moved from one output to another by audio policy 1827 // manager 1828 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1829 for (size_t i = 0; i < mTracks.size(); ++i) { 1830 sp<Track> t = mTracks[i]; 1831 if (t != 0 && !t->isOutputTrack()) { 1832 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1833 if (sessionId == t->sessionId() && strategy != actual) { 1834 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1835 strategy, actual); 1836 lStatus = BAD_VALUE; 1837 goto Exit; 1838 } 1839 } 1840 } 1841 1842 if (!isTimed) { 1843 track = new Track(this, client, streamType, sampleRate, format, 1844 channelMask, frameCount, sharedBuffer, sessionId, flags); 1845 } else { 1846 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1847 channelMask, frameCount, sharedBuffer, sessionId); 1848 } 1849 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1850 lStatus = NO_MEMORY; 1851 goto Exit; 1852 } 1853 mTracks.add(track); 1854 1855 sp<EffectChain> chain = getEffectChain_l(sessionId); 1856 if (chain != 0) { 1857 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1858 track->setMainBuffer(chain->inBuffer()); 1859 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1860 chain->incTrackCnt(); 1861 } 1862 1863 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1864 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1865 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1866 // so ask activity manager to do this on our behalf 1867 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1868 } 1869 } 1870 1871 lStatus = NO_ERROR; 1872 1873Exit: 1874 if (status) { 1875 *status = lStatus; 1876 } 1877 return track; 1878} 1879 1880uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1881{ 1882 if (mFastMixer != NULL) { 1883 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1884 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1885 } 1886 return latency; 1887} 1888 1889uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1890{ 1891 return latency; 1892} 1893 1894uint32_t AudioFlinger::PlaybackThread::latency() const 1895{ 1896 Mutex::Autolock _l(mLock); 1897 return latency_l(); 1898} 1899uint32_t AudioFlinger::PlaybackThread::latency_l() const 1900{ 1901 if (initCheck() == NO_ERROR) { 1902 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1903 } else { 1904 return 0; 1905 } 1906} 1907 1908void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1909{ 1910 Mutex::Autolock _l(mLock); 1911 // Don't apply master volume in SW if our HAL can do it for us. 1912 if (mOutput && mOutput->audioHwDev && 1913 mOutput->audioHwDev->canSetMasterVolume()) { 1914 mMasterVolume = 1.0; 1915 } else { 1916 mMasterVolume = value; 1917 } 1918} 1919 1920void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1921{ 1922 Mutex::Autolock _l(mLock); 1923 // Don't apply master mute in SW if our HAL can do it for us. 1924 if (mOutput && mOutput->audioHwDev && 1925 mOutput->audioHwDev->canSetMasterMute()) { 1926 mMasterMute = false; 1927 } else { 1928 mMasterMute = muted; 1929 } 1930} 1931 1932void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1933{ 1934 Mutex::Autolock _l(mLock); 1935 mStreamTypes[stream].volume = value; 1936} 1937 1938void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1939{ 1940 Mutex::Autolock _l(mLock); 1941 mStreamTypes[stream].mute = muted; 1942} 1943 1944float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1945{ 1946 Mutex::Autolock _l(mLock); 1947 return mStreamTypes[stream].volume; 1948} 1949 1950// addTrack_l() must be called with ThreadBase::mLock held 1951status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1952{ 1953 status_t status = ALREADY_EXISTS; 1954 1955 // set retry count for buffer fill 1956 track->mRetryCount = kMaxTrackStartupRetries; 1957 if (mActiveTracks.indexOf(track) < 0) { 1958 // the track is newly added, make sure it fills up all its 1959 // buffers before playing. This is to ensure the client will 1960 // effectively get the latency it requested. 1961 track->mFillingUpStatus = Track::FS_FILLING; 1962 track->mResetDone = false; 1963 track->mPresentationCompleteFrames = 0; 1964 mActiveTracks.add(track); 1965 if (track->mainBuffer() != mMixBuffer) { 1966 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1967 if (chain != 0) { 1968 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1969 chain->incActiveTrackCnt(); 1970 } 1971 } 1972 1973 status = NO_ERROR; 1974 } 1975 1976 ALOGV("mWaitWorkCV.broadcast"); 1977 mWaitWorkCV.broadcast(); 1978 1979 return status; 1980} 1981 1982// destroyTrack_l() must be called with ThreadBase::mLock held 1983void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1984{ 1985 track->mState = TrackBase::TERMINATED; 1986 // active tracks are removed by threadLoop() 1987 if (mActiveTracks.indexOf(track) < 0) { 1988 removeTrack_l(track); 1989 } 1990} 1991 1992void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1993{ 1994 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1995 mTracks.remove(track); 1996 deleteTrackName_l(track->name()); 1997 // redundant as track is about to be destroyed, for dumpsys only 1998 track->mName = -1; 1999 if (track->isFastTrack()) { 2000 int index = track->mFastIndex; 2001 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2002 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2003 mFastTrackAvailMask |= 1 << index; 2004 // redundant as track is about to be destroyed, for dumpsys only 2005 track->mFastIndex = -1; 2006 } 2007 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2008 if (chain != 0) { 2009 chain->decTrackCnt(); 2010 } 2011} 2012 2013String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2014{ 2015 String8 out_s8 = String8(""); 2016 char *s; 2017 2018 Mutex::Autolock _l(mLock); 2019 if (initCheck() != NO_ERROR) { 2020 return out_s8; 2021 } 2022 2023 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2024 out_s8 = String8(s); 2025 free(s); 2026 return out_s8; 2027} 2028 2029// audioConfigChanged_l() must be called with AudioFlinger::mLock held 2030void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 2031 AudioSystem::OutputDescriptor desc; 2032 void *param2 = NULL; 2033 2034 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 2035 2036 switch (event) { 2037 case AudioSystem::OUTPUT_OPENED: 2038 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2039 desc.channels = mChannelMask; 2040 desc.samplingRate = mSampleRate; 2041 desc.format = mFormat; 2042 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 2043 desc.latency = latency(); 2044 param2 = &desc; 2045 break; 2046 2047 case AudioSystem::STREAM_CONFIG_CHANGED: 2048 param2 = ¶m; 2049 case AudioSystem::OUTPUT_CLOSED: 2050 default: 2051 break; 2052 } 2053 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2054} 2055 2056void AudioFlinger::PlaybackThread::readOutputParameters() 2057{ 2058 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2059 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2060 mChannelCount = (uint16_t)popcount(mChannelMask); 2061 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2062 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2063 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2064 if (mFrameCount & 15) { 2065 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2066 mFrameCount); 2067 } 2068 2069 // Calculate size of normal mix buffer relative to the HAL output buffer size 2070 double multiplier = 1.0; 2071 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2072 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2073 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2074 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2075 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2076 maxNormalFrameCount = maxNormalFrameCount & ~15; 2077 if (maxNormalFrameCount < minNormalFrameCount) { 2078 maxNormalFrameCount = minNormalFrameCount; 2079 } 2080 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2081 if (multiplier <= 1.0) { 2082 multiplier = 1.0; 2083 } else if (multiplier <= 2.0) { 2084 if (2 * mFrameCount <= maxNormalFrameCount) { 2085 multiplier = 2.0; 2086 } else { 2087 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2088 } 2089 } else { 2090 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2091 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2092 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2093 // FIXME this rounding up should not be done if no HAL SRC 2094 uint32_t truncMult = (uint32_t) multiplier; 2095 if ((truncMult & 1)) { 2096 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2097 ++truncMult; 2098 } 2099 } 2100 multiplier = (double) truncMult; 2101 } 2102 } 2103 mNormalFrameCount = multiplier * mFrameCount; 2104 // round up to nearest 16 frames to satisfy AudioMixer 2105 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2106 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2107 2108 delete[] mMixBuffer; 2109 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2110 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2111 2112 // force reconfiguration of effect chains and engines to take new buffer size and audio 2113 // parameters into account 2114 // Note that mLock is not held when readOutputParameters() is called from the constructor 2115 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2116 // matter. 2117 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2118 Vector< sp<EffectChain> > effectChains = mEffectChains; 2119 for (size_t i = 0; i < effectChains.size(); i ++) { 2120 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2121 } 2122} 2123 2124 2125status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2126{ 2127 if (halFrames == NULL || dspFrames == NULL) { 2128 return BAD_VALUE; 2129 } 2130 Mutex::Autolock _l(mLock); 2131 if (initCheck() != NO_ERROR) { 2132 return INVALID_OPERATION; 2133 } 2134 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2135 2136 if (isSuspended()) { 2137 // return an estimation of rendered frames when the output is suspended 2138 int32_t frames = mBytesWritten - latency_l(); 2139 if (frames < 0) { 2140 frames = 0; 2141 } 2142 *dspFrames = (uint32_t)frames; 2143 return NO_ERROR; 2144 } else { 2145 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2146 } 2147} 2148 2149uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2150{ 2151 Mutex::Autolock _l(mLock); 2152 uint32_t result = 0; 2153 if (getEffectChain_l(sessionId) != 0) { 2154 result = EFFECT_SESSION; 2155 } 2156 2157 for (size_t i = 0; i < mTracks.size(); ++i) { 2158 sp<Track> track = mTracks[i]; 2159 if (sessionId == track->sessionId() && 2160 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2161 result |= TRACK_SESSION; 2162 break; 2163 } 2164 } 2165 2166 return result; 2167} 2168 2169uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2170{ 2171 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2172 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2173 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2174 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2175 } 2176 for (size_t i = 0; i < mTracks.size(); i++) { 2177 sp<Track> track = mTracks[i]; 2178 if (sessionId == track->sessionId() && 2179 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2180 return AudioSystem::getStrategyForStream(track->streamType()); 2181 } 2182 } 2183 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2184} 2185 2186 2187AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2188{ 2189 Mutex::Autolock _l(mLock); 2190 return mOutput; 2191} 2192 2193AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2194{ 2195 Mutex::Autolock _l(mLock); 2196 AudioStreamOut *output = mOutput; 2197 mOutput = NULL; 2198 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2199 // must push a NULL and wait for ack 2200 mOutputSink.clear(); 2201 mPipeSink.clear(); 2202 mNormalSink.clear(); 2203 return output; 2204} 2205 2206// this method must always be called either with ThreadBase mLock held or inside the thread loop 2207audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2208{ 2209 if (mOutput == NULL) { 2210 return NULL; 2211 } 2212 return &mOutput->stream->common; 2213} 2214 2215uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2216{ 2217 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2218} 2219 2220status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2221{ 2222 if (!isValidSyncEvent(event)) { 2223 return BAD_VALUE; 2224 } 2225 2226 Mutex::Autolock _l(mLock); 2227 2228 for (size_t i = 0; i < mTracks.size(); ++i) { 2229 sp<Track> track = mTracks[i]; 2230 if (event->triggerSession() == track->sessionId()) { 2231 (void) track->setSyncEvent(event); 2232 return NO_ERROR; 2233 } 2234 } 2235 2236 return NAME_NOT_FOUND; 2237} 2238 2239bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2240{ 2241 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2242} 2243 2244void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2245{ 2246 size_t count = tracksToRemove.size(); 2247 if (CC_UNLIKELY(count)) { 2248 for (size_t i = 0 ; i < count ; i++) { 2249 const sp<Track>& track = tracksToRemove.itemAt(i); 2250 if ((track->sharedBuffer() != 0) && 2251 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2252 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2253 } 2254 } 2255 } 2256 2257} 2258 2259// ---------------------------------------------------------------------------- 2260 2261AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2262 audio_io_handle_t id, audio_devices_t device, type_t type) 2263 : PlaybackThread(audioFlinger, output, id, device, type), 2264 // mAudioMixer below 2265 // mFastMixer below 2266 mFastMixerFutex(0) 2267 // mOutputSink below 2268 // mPipeSink below 2269 // mNormalSink below 2270{ 2271 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2272 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2273 "mFrameCount=%d, mNormalFrameCount=%d", 2274 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2275 mNormalFrameCount); 2276 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2277 2278 // FIXME - Current mixer implementation only supports stereo output 2279 if (mChannelCount != FCC_2) { 2280 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2281 } 2282 2283 // create an NBAIO sink for the HAL output stream, and negotiate 2284 mOutputSink = new AudioStreamOutSink(output->stream); 2285 size_t numCounterOffers = 0; 2286 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2287 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2288 ALOG_ASSERT(index == 0); 2289 2290 // initialize fast mixer depending on configuration 2291 bool initFastMixer; 2292 switch (kUseFastMixer) { 2293 case FastMixer_Never: 2294 initFastMixer = false; 2295 break; 2296 case FastMixer_Always: 2297 initFastMixer = true; 2298 break; 2299 case FastMixer_Static: 2300 case FastMixer_Dynamic: 2301 initFastMixer = mFrameCount < mNormalFrameCount; 2302 break; 2303 } 2304 if (initFastMixer) { 2305 2306 // create a MonoPipe to connect our submix to FastMixer 2307 NBAIO_Format format = mOutputSink->format(); 2308 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2309 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2310 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2311 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2312 const NBAIO_Format offers[1] = {format}; 2313 size_t numCounterOffers = 0; 2314 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2315 ALOG_ASSERT(index == 0); 2316 monoPipe->setAvgFrames((mScreenState & 1) ? 2317 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2318 mPipeSink = monoPipe; 2319 2320#ifdef TEE_SINK_FRAMES 2321 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2322 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2323 numCounterOffers = 0; 2324 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2325 ALOG_ASSERT(index == 0); 2326 mTeeSink = teeSink; 2327 PipeReader *teeSource = new PipeReader(*teeSink); 2328 numCounterOffers = 0; 2329 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2330 ALOG_ASSERT(index == 0); 2331 mTeeSource = teeSource; 2332#endif 2333 2334 // create fast mixer and configure it initially with just one fast track for our submix 2335 mFastMixer = new FastMixer(); 2336 FastMixerStateQueue *sq = mFastMixer->sq(); 2337#ifdef STATE_QUEUE_DUMP 2338 sq->setObserverDump(&mStateQueueObserverDump); 2339 sq->setMutatorDump(&mStateQueueMutatorDump); 2340#endif 2341 FastMixerState *state = sq->begin(); 2342 FastTrack *fastTrack = &state->mFastTracks[0]; 2343 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2344 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2345 fastTrack->mVolumeProvider = NULL; 2346 fastTrack->mGeneration++; 2347 state->mFastTracksGen++; 2348 state->mTrackMask = 1; 2349 // fast mixer will use the HAL output sink 2350 state->mOutputSink = mOutputSink.get(); 2351 state->mOutputSinkGen++; 2352 state->mFrameCount = mFrameCount; 2353 state->mCommand = FastMixerState::COLD_IDLE; 2354 // already done in constructor initialization list 2355 //mFastMixerFutex = 0; 2356 state->mColdFutexAddr = &mFastMixerFutex; 2357 state->mColdGen++; 2358 state->mDumpState = &mFastMixerDumpState; 2359 state->mTeeSink = mTeeSink.get(); 2360 sq->end(); 2361 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2362 2363 // start the fast mixer 2364 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2365 pid_t tid = mFastMixer->getTid(); 2366 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2367 if (err != 0) { 2368 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2369 kPriorityFastMixer, getpid_cached, tid, err); 2370 } 2371 2372#ifdef AUDIO_WATCHDOG 2373 // create and start the watchdog 2374 mAudioWatchdog = new AudioWatchdog(); 2375 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2376 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2377 tid = mAudioWatchdog->getTid(); 2378 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2379 if (err != 0) { 2380 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2381 kPriorityFastMixer, getpid_cached, tid, err); 2382 } 2383#endif 2384 2385 } else { 2386 mFastMixer = NULL; 2387 } 2388 2389 switch (kUseFastMixer) { 2390 case FastMixer_Never: 2391 case FastMixer_Dynamic: 2392 mNormalSink = mOutputSink; 2393 break; 2394 case FastMixer_Always: 2395 mNormalSink = mPipeSink; 2396 break; 2397 case FastMixer_Static: 2398 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2399 break; 2400 } 2401} 2402 2403AudioFlinger::MixerThread::~MixerThread() 2404{ 2405 if (mFastMixer != NULL) { 2406 FastMixerStateQueue *sq = mFastMixer->sq(); 2407 FastMixerState *state = sq->begin(); 2408 if (state->mCommand == FastMixerState::COLD_IDLE) { 2409 int32_t old = android_atomic_inc(&mFastMixerFutex); 2410 if (old == -1) { 2411 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2412 } 2413 } 2414 state->mCommand = FastMixerState::EXIT; 2415 sq->end(); 2416 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2417 mFastMixer->join(); 2418 // Though the fast mixer thread has exited, it's state queue is still valid. 2419 // We'll use that extract the final state which contains one remaining fast track 2420 // corresponding to our sub-mix. 2421 state = sq->begin(); 2422 ALOG_ASSERT(state->mTrackMask == 1); 2423 FastTrack *fastTrack = &state->mFastTracks[0]; 2424 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2425 delete fastTrack->mBufferProvider; 2426 sq->end(false /*didModify*/); 2427 delete mFastMixer; 2428#ifdef AUDIO_WATCHDOG 2429 if (mAudioWatchdog != 0) { 2430 mAudioWatchdog->requestExit(); 2431 mAudioWatchdog->requestExitAndWait(); 2432 mAudioWatchdog.clear(); 2433 } 2434#endif 2435 } 2436 delete mAudioMixer; 2437} 2438 2439class CpuStats { 2440public: 2441 CpuStats(); 2442 void sample(const String8 &title); 2443#ifdef DEBUG_CPU_USAGE 2444private: 2445 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2446 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2447 2448 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2449 2450 int mCpuNum; // thread's current CPU number 2451 int mCpukHz; // frequency of thread's current CPU in kHz 2452#endif 2453}; 2454 2455CpuStats::CpuStats() 2456#ifdef DEBUG_CPU_USAGE 2457 : mCpuNum(-1), mCpukHz(-1) 2458#endif 2459{ 2460} 2461 2462void CpuStats::sample(const String8 &title) { 2463#ifdef DEBUG_CPU_USAGE 2464 // get current thread's delta CPU time in wall clock ns 2465 double wcNs; 2466 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2467 2468 // record sample for wall clock statistics 2469 if (valid) { 2470 mWcStats.sample(wcNs); 2471 } 2472 2473 // get the current CPU number 2474 int cpuNum = sched_getcpu(); 2475 2476 // get the current CPU frequency in kHz 2477 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2478 2479 // check if either CPU number or frequency changed 2480 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2481 mCpuNum = cpuNum; 2482 mCpukHz = cpukHz; 2483 // ignore sample for purposes of cycles 2484 valid = false; 2485 } 2486 2487 // if no change in CPU number or frequency, then record sample for cycle statistics 2488 if (valid && mCpukHz > 0) { 2489 double cycles = wcNs * cpukHz * 0.000001; 2490 mHzStats.sample(cycles); 2491 } 2492 2493 unsigned n = mWcStats.n(); 2494 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2495 if ((n & 127) == 1) { 2496 long long elapsed = mCpuUsage.elapsed(); 2497 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2498 double perLoop = elapsed / (double) n; 2499 double perLoop100 = perLoop * 0.01; 2500 double perLoop1k = perLoop * 0.001; 2501 double mean = mWcStats.mean(); 2502 double stddev = mWcStats.stddev(); 2503 double minimum = mWcStats.minimum(); 2504 double maximum = mWcStats.maximum(); 2505 double meanCycles = mHzStats.mean(); 2506 double stddevCycles = mHzStats.stddev(); 2507 double minCycles = mHzStats.minimum(); 2508 double maxCycles = mHzStats.maximum(); 2509 mCpuUsage.resetElapsed(); 2510 mWcStats.reset(); 2511 mHzStats.reset(); 2512 ALOGD("CPU usage for %s over past %.1f secs\n" 2513 " (%u mixer loops at %.1f mean ms per loop):\n" 2514 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2515 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2516 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2517 title.string(), 2518 elapsed * .000000001, n, perLoop * .000001, 2519 mean * .001, 2520 stddev * .001, 2521 minimum * .001, 2522 maximum * .001, 2523 mean / perLoop100, 2524 stddev / perLoop100, 2525 minimum / perLoop100, 2526 maximum / perLoop100, 2527 meanCycles / perLoop1k, 2528 stddevCycles / perLoop1k, 2529 minCycles / perLoop1k, 2530 maxCycles / perLoop1k); 2531 2532 } 2533 } 2534#endif 2535}; 2536 2537void AudioFlinger::PlaybackThread::checkSilentMode_l() 2538{ 2539 if (!mMasterMute) { 2540 char value[PROPERTY_VALUE_MAX]; 2541 if (property_get("ro.audio.silent", value, "0") > 0) { 2542 char *endptr; 2543 unsigned long ul = strtoul(value, &endptr, 0); 2544 if (*endptr == '\0' && ul != 0) { 2545 ALOGD("Silence is golden"); 2546 // The setprop command will not allow a property to be changed after 2547 // the first time it is set, so we don't have to worry about un-muting. 2548 setMasterMute_l(true); 2549 } 2550 } 2551 } 2552} 2553 2554bool AudioFlinger::PlaybackThread::threadLoop() 2555{ 2556 Vector< sp<Track> > tracksToRemove; 2557 2558 standbyTime = systemTime(); 2559 2560 // MIXER 2561 nsecs_t lastWarning = 0; 2562 2563 // DUPLICATING 2564 // FIXME could this be made local to while loop? 2565 writeFrames = 0; 2566 2567 cacheParameters_l(); 2568 sleepTime = idleSleepTime; 2569 2570 if (mType == MIXER) { 2571 sleepTimeShift = 0; 2572 } 2573 2574 CpuStats cpuStats; 2575 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2576 2577 acquireWakeLock(); 2578 2579 while (!exitPending()) 2580 { 2581 cpuStats.sample(myName); 2582 2583 Vector< sp<EffectChain> > effectChains; 2584 2585 processConfigEvents(); 2586 2587 { // scope for mLock 2588 2589 Mutex::Autolock _l(mLock); 2590 2591 if (checkForNewParameters_l()) { 2592 cacheParameters_l(); 2593 } 2594 2595 saveOutputTracks(); 2596 2597 // put audio hardware into standby after short delay 2598 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2599 isSuspended())) { 2600 if (!mStandby) { 2601 2602 threadLoop_standby(); 2603 2604 mStandby = true; 2605 } 2606 2607 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2608 // we're about to wait, flush the binder command buffer 2609 IPCThreadState::self()->flushCommands(); 2610 2611 clearOutputTracks(); 2612 2613 if (exitPending()) break; 2614 2615 releaseWakeLock_l(); 2616 // wait until we have something to do... 2617 ALOGV("%s going to sleep", myName.string()); 2618 mWaitWorkCV.wait(mLock); 2619 ALOGV("%s waking up", myName.string()); 2620 acquireWakeLock_l(); 2621 2622 mMixerStatus = MIXER_IDLE; 2623 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2624 mBytesWritten = 0; 2625 2626 checkSilentMode_l(); 2627 2628 standbyTime = systemTime() + standbyDelay; 2629 sleepTime = idleSleepTime; 2630 if (mType == MIXER) { 2631 sleepTimeShift = 0; 2632 } 2633 2634 continue; 2635 } 2636 } 2637 2638 // mMixerStatusIgnoringFastTracks is also updated internally 2639 mMixerStatus = prepareTracks_l(&tracksToRemove); 2640 2641 // prevent any changes in effect chain list and in each effect chain 2642 // during mixing and effect process as the audio buffers could be deleted 2643 // or modified if an effect is created or deleted 2644 lockEffectChains_l(effectChains); 2645 } 2646 2647 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2648 threadLoop_mix(); 2649 } else { 2650 threadLoop_sleepTime(); 2651 } 2652 2653 if (isSuspended()) { 2654 sleepTime = suspendSleepTimeUs(); 2655 mBytesWritten += mixBufferSize; 2656 } 2657 2658 // only process effects if we're going to write 2659 if (sleepTime == 0) { 2660 for (size_t i = 0; i < effectChains.size(); i ++) { 2661 effectChains[i]->process_l(); 2662 } 2663 } 2664 2665 // enable changes in effect chain 2666 unlockEffectChains(effectChains); 2667 2668 // sleepTime == 0 means we must write to audio hardware 2669 if (sleepTime == 0) { 2670 2671 threadLoop_write(); 2672 2673if (mType == MIXER) { 2674 // write blocked detection 2675 nsecs_t now = systemTime(); 2676 nsecs_t delta = now - mLastWriteTime; 2677 if (!mStandby && delta > maxPeriod) { 2678 mNumDelayedWrites++; 2679 if ((now - lastWarning) > kWarningThrottleNs) { 2680#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2681 ScopedTrace st(ATRACE_TAG, "underrun"); 2682#endif 2683 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2684 ns2ms(delta), mNumDelayedWrites, this); 2685 lastWarning = now; 2686 } 2687 } 2688} 2689 2690 mStandby = false; 2691 } else { 2692 usleep(sleepTime); 2693 } 2694 2695 // Finally let go of removed track(s), without the lock held 2696 // since we can't guarantee the destructors won't acquire that 2697 // same lock. This will also mutate and push a new fast mixer state. 2698 threadLoop_removeTracks(tracksToRemove); 2699 tracksToRemove.clear(); 2700 2701 // FIXME I don't understand the need for this here; 2702 // it was in the original code but maybe the 2703 // assignment in saveOutputTracks() makes this unnecessary? 2704 clearOutputTracks(); 2705 2706 // Effect chains will be actually deleted here if they were removed from 2707 // mEffectChains list during mixing or effects processing 2708 effectChains.clear(); 2709 2710 // FIXME Note that the above .clear() is no longer necessary since effectChains 2711 // is now local to this block, but will keep it for now (at least until merge done). 2712 } 2713 2714 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2715 if (mType == MIXER || mType == DIRECT) { 2716 // put output stream into standby mode 2717 if (!mStandby) { 2718 mOutput->stream->common.standby(&mOutput->stream->common); 2719 } 2720 } 2721 2722 releaseWakeLock(); 2723 2724 ALOGV("Thread %p type %d exiting", this, mType); 2725 return false; 2726} 2727 2728void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2729{ 2730 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2731} 2732 2733void AudioFlinger::MixerThread::threadLoop_write() 2734{ 2735 // FIXME we should only do one push per cycle; confirm this is true 2736 // Start the fast mixer if it's not already running 2737 if (mFastMixer != NULL) { 2738 FastMixerStateQueue *sq = mFastMixer->sq(); 2739 FastMixerState *state = sq->begin(); 2740 if (state->mCommand != FastMixerState::MIX_WRITE && 2741 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2742 if (state->mCommand == FastMixerState::COLD_IDLE) { 2743 int32_t old = android_atomic_inc(&mFastMixerFutex); 2744 if (old == -1) { 2745 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2746 } 2747#ifdef AUDIO_WATCHDOG 2748 if (mAudioWatchdog != 0) { 2749 mAudioWatchdog->resume(); 2750 } 2751#endif 2752 } 2753 state->mCommand = FastMixerState::MIX_WRITE; 2754 sq->end(); 2755 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2756 if (kUseFastMixer == FastMixer_Dynamic) { 2757 mNormalSink = mPipeSink; 2758 } 2759 } else { 2760 sq->end(false /*didModify*/); 2761 } 2762 } 2763 PlaybackThread::threadLoop_write(); 2764} 2765 2766// shared by MIXER and DIRECT, overridden by DUPLICATING 2767void AudioFlinger::PlaybackThread::threadLoop_write() 2768{ 2769 // FIXME rewrite to reduce number of system calls 2770 mLastWriteTime = systemTime(); 2771 mInWrite = true; 2772 int bytesWritten; 2773 2774 // If an NBAIO sink is present, use it to write the normal mixer's submix 2775 if (mNormalSink != 0) { 2776#define mBitShift 2 // FIXME 2777 size_t count = mixBufferSize >> mBitShift; 2778#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2779 Tracer::traceBegin(ATRACE_TAG, "write"); 2780#endif 2781 // update the setpoint when gScreenState changes 2782 uint32_t screenState = gScreenState; 2783 if (screenState != mScreenState) { 2784 mScreenState = screenState; 2785 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2786 if (pipe != NULL) { 2787 pipe->setAvgFrames((mScreenState & 1) ? 2788 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2789 } 2790 } 2791 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2792#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2793 Tracer::traceEnd(ATRACE_TAG); 2794#endif 2795 if (framesWritten > 0) { 2796 bytesWritten = framesWritten << mBitShift; 2797 } else { 2798 bytesWritten = framesWritten; 2799 } 2800 // otherwise use the HAL / AudioStreamOut directly 2801 } else { 2802 // Direct output thread. 2803 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2804 } 2805 2806 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2807 mNumWrites++; 2808 mInWrite = false; 2809} 2810 2811void AudioFlinger::MixerThread::threadLoop_standby() 2812{ 2813 // Idle the fast mixer if it's currently running 2814 if (mFastMixer != NULL) { 2815 FastMixerStateQueue *sq = mFastMixer->sq(); 2816 FastMixerState *state = sq->begin(); 2817 if (!(state->mCommand & FastMixerState::IDLE)) { 2818 state->mCommand = FastMixerState::COLD_IDLE; 2819 state->mColdFutexAddr = &mFastMixerFutex; 2820 state->mColdGen++; 2821 mFastMixerFutex = 0; 2822 sq->end(); 2823 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2824 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2825 if (kUseFastMixer == FastMixer_Dynamic) { 2826 mNormalSink = mOutputSink; 2827 } 2828#ifdef AUDIO_WATCHDOG 2829 if (mAudioWatchdog != 0) { 2830 mAudioWatchdog->pause(); 2831 } 2832#endif 2833 } else { 2834 sq->end(false /*didModify*/); 2835 } 2836 } 2837 PlaybackThread::threadLoop_standby(); 2838} 2839 2840// shared by MIXER and DIRECT, overridden by DUPLICATING 2841void AudioFlinger::PlaybackThread::threadLoop_standby() 2842{ 2843 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2844 mOutput->stream->common.standby(&mOutput->stream->common); 2845} 2846 2847void AudioFlinger::MixerThread::threadLoop_mix() 2848{ 2849 // obtain the presentation timestamp of the next output buffer 2850 int64_t pts; 2851 status_t status = INVALID_OPERATION; 2852 2853 if (mNormalSink != 0) { 2854 status = mNormalSink->getNextWriteTimestamp(&pts); 2855 } else { 2856 status = mOutputSink->getNextWriteTimestamp(&pts); 2857 } 2858 2859 if (status != NO_ERROR) { 2860 pts = AudioBufferProvider::kInvalidPTS; 2861 } 2862 2863 // mix buffers... 2864 mAudioMixer->process(pts); 2865 // increase sleep time progressively when application underrun condition clears. 2866 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2867 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2868 // such that we would underrun the audio HAL. 2869 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2870 sleepTimeShift--; 2871 } 2872 sleepTime = 0; 2873 standbyTime = systemTime() + standbyDelay; 2874 //TODO: delay standby when effects have a tail 2875} 2876 2877void AudioFlinger::MixerThread::threadLoop_sleepTime() 2878{ 2879 // If no tracks are ready, sleep once for the duration of an output 2880 // buffer size, then write 0s to the output 2881 if (sleepTime == 0) { 2882 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2883 sleepTime = activeSleepTime >> sleepTimeShift; 2884 if (sleepTime < kMinThreadSleepTimeUs) { 2885 sleepTime = kMinThreadSleepTimeUs; 2886 } 2887 // reduce sleep time in case of consecutive application underruns to avoid 2888 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2889 // duration we would end up writing less data than needed by the audio HAL if 2890 // the condition persists. 2891 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2892 sleepTimeShift++; 2893 } 2894 } else { 2895 sleepTime = idleSleepTime; 2896 } 2897 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2898 memset (mMixBuffer, 0, mixBufferSize); 2899 sleepTime = 0; 2900 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2901 } 2902 // TODO add standby time extension fct of effect tail 2903} 2904 2905// prepareTracks_l() must be called with ThreadBase::mLock held 2906AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2907 Vector< sp<Track> > *tracksToRemove) 2908{ 2909 2910 mixer_state mixerStatus = MIXER_IDLE; 2911 // find out which tracks need to be processed 2912 size_t count = mActiveTracks.size(); 2913 size_t mixedTracks = 0; 2914 size_t tracksWithEffect = 0; 2915 // counts only _active_ fast tracks 2916 size_t fastTracks = 0; 2917 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2918 2919 float masterVolume = mMasterVolume; 2920 bool masterMute = mMasterMute; 2921 2922 if (masterMute) { 2923 masterVolume = 0; 2924 } 2925 // Delegate master volume control to effect in output mix effect chain if needed 2926 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2927 if (chain != 0) { 2928 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2929 chain->setVolume_l(&v, &v); 2930 masterVolume = (float)((v + (1 << 23)) >> 24); 2931 chain.clear(); 2932 } 2933 2934 // prepare a new state to push 2935 FastMixerStateQueue *sq = NULL; 2936 FastMixerState *state = NULL; 2937 bool didModify = false; 2938 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2939 if (mFastMixer != NULL) { 2940 sq = mFastMixer->sq(); 2941 state = sq->begin(); 2942 } 2943 2944 for (size_t i=0 ; i<count ; i++) { 2945 sp<Track> t = mActiveTracks[i].promote(); 2946 if (t == 0) continue; 2947 2948 // this const just means the local variable doesn't change 2949 Track* const track = t.get(); 2950 2951 // process fast tracks 2952 if (track->isFastTrack()) { 2953 2954 // It's theoretically possible (though unlikely) for a fast track to be created 2955 // and then removed within the same normal mix cycle. This is not a problem, as 2956 // the track never becomes active so it's fast mixer slot is never touched. 2957 // The converse, of removing an (active) track and then creating a new track 2958 // at the identical fast mixer slot within the same normal mix cycle, 2959 // is impossible because the slot isn't marked available until the end of each cycle. 2960 int j = track->mFastIndex; 2961 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2962 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2963 FastTrack *fastTrack = &state->mFastTracks[j]; 2964 2965 // Determine whether the track is currently in underrun condition, 2966 // and whether it had a recent underrun. 2967 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2968 FastTrackUnderruns underruns = ftDump->mUnderruns; 2969 uint32_t recentFull = (underruns.mBitFields.mFull - 2970 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2971 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2972 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2973 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2974 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2975 uint32_t recentUnderruns = recentPartial + recentEmpty; 2976 track->mObservedUnderruns = underruns; 2977 // don't count underruns that occur while stopping or pausing 2978 // or stopped which can occur when flush() is called while active 2979 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2980 track->mUnderrunCount += recentUnderruns; 2981 } 2982 2983 // This is similar to the state machine for normal tracks, 2984 // with a few modifications for fast tracks. 2985 bool isActive = true; 2986 switch (track->mState) { 2987 case TrackBase::STOPPING_1: 2988 // track stays active in STOPPING_1 state until first underrun 2989 if (recentUnderruns > 0) { 2990 track->mState = TrackBase::STOPPING_2; 2991 } 2992 break; 2993 case TrackBase::PAUSING: 2994 // ramp down is not yet implemented 2995 track->setPaused(); 2996 break; 2997 case TrackBase::RESUMING: 2998 // ramp up is not yet implemented 2999 track->mState = TrackBase::ACTIVE; 3000 break; 3001 case TrackBase::ACTIVE: 3002 if (recentFull > 0 || recentPartial > 0) { 3003 // track has provided at least some frames recently: reset retry count 3004 track->mRetryCount = kMaxTrackRetries; 3005 } 3006 if (recentUnderruns == 0) { 3007 // no recent underruns: stay active 3008 break; 3009 } 3010 // there has recently been an underrun of some kind 3011 if (track->sharedBuffer() == 0) { 3012 // were any of the recent underruns "empty" (no frames available)? 3013 if (recentEmpty == 0) { 3014 // no, then ignore the partial underruns as they are allowed indefinitely 3015 break; 3016 } 3017 // there has recently been an "empty" underrun: decrement the retry counter 3018 if (--(track->mRetryCount) > 0) { 3019 break; 3020 } 3021 // indicate to client process that the track was disabled because of underrun; 3022 // it will then automatically call start() when data is available 3023 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 3024 // remove from active list, but state remains ACTIVE [confusing but true] 3025 isActive = false; 3026 break; 3027 } 3028 // fall through 3029 case TrackBase::STOPPING_2: 3030 case TrackBase::PAUSED: 3031 case TrackBase::TERMINATED: 3032 case TrackBase::STOPPED: 3033 case TrackBase::FLUSHED: // flush() while active 3034 // Check for presentation complete if track is inactive 3035 // We have consumed all the buffers of this track. 3036 // This would be incomplete if we auto-paused on underrun 3037 { 3038 size_t audioHALFrames = 3039 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3040 size_t framesWritten = 3041 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3042 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3043 // track stays in active list until presentation is complete 3044 break; 3045 } 3046 } 3047 if (track->isStopping_2()) { 3048 track->mState = TrackBase::STOPPED; 3049 } 3050 if (track->isStopped()) { 3051 // Can't reset directly, as fast mixer is still polling this track 3052 // track->reset(); 3053 // So instead mark this track as needing to be reset after push with ack 3054 resetMask |= 1 << i; 3055 } 3056 isActive = false; 3057 break; 3058 case TrackBase::IDLE: 3059 default: 3060 LOG_FATAL("unexpected track state %d", track->mState); 3061 } 3062 3063 if (isActive) { 3064 // was it previously inactive? 3065 if (!(state->mTrackMask & (1 << j))) { 3066 ExtendedAudioBufferProvider *eabp = track; 3067 VolumeProvider *vp = track; 3068 fastTrack->mBufferProvider = eabp; 3069 fastTrack->mVolumeProvider = vp; 3070 fastTrack->mSampleRate = track->mSampleRate; 3071 fastTrack->mChannelMask = track->mChannelMask; 3072 fastTrack->mGeneration++; 3073 state->mTrackMask |= 1 << j; 3074 didModify = true; 3075 // no acknowledgement required for newly active tracks 3076 } 3077 // cache the combined master volume and stream type volume for fast mixer; this 3078 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3079 track->mCachedVolume = track->isMuted() ? 3080 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3081 ++fastTracks; 3082 } else { 3083 // was it previously active? 3084 if (state->mTrackMask & (1 << j)) { 3085 fastTrack->mBufferProvider = NULL; 3086 fastTrack->mGeneration++; 3087 state->mTrackMask &= ~(1 << j); 3088 didModify = true; 3089 // If any fast tracks were removed, we must wait for acknowledgement 3090 // because we're about to decrement the last sp<> on those tracks. 3091 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3092 } else { 3093 LOG_FATAL("fast track %d should have been active", j); 3094 } 3095 tracksToRemove->add(track); 3096 // Avoids a misleading display in dumpsys 3097 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3098 } 3099 continue; 3100 } 3101 3102 { // local variable scope to avoid goto warning 3103 3104 audio_track_cblk_t* cblk = track->cblk(); 3105 3106 // The first time a track is added we wait 3107 // for all its buffers to be filled before processing it 3108 int name = track->name(); 3109 // make sure that we have enough frames to mix one full buffer. 3110 // enforce this condition only once to enable draining the buffer in case the client 3111 // app does not call stop() and relies on underrun to stop: 3112 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3113 // during last round 3114 uint32_t minFrames = 1; 3115 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3116 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3117 if (t->sampleRate() == (int)mSampleRate) { 3118 minFrames = mNormalFrameCount; 3119 } else { 3120 // +1 for rounding and +1 for additional sample needed for interpolation 3121 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3122 // add frames already consumed but not yet released by the resampler 3123 // because cblk->framesReady() will include these frames 3124 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3125 // the minimum track buffer size is normally twice the number of frames necessary 3126 // to fill one buffer and the resampler should not leave more than one buffer worth 3127 // of unreleased frames after each pass, but just in case... 3128 ALOG_ASSERT(minFrames <= cblk->frameCount); 3129 } 3130 } 3131 if ((track->framesReady() >= minFrames) && track->isReady() && 3132 !track->isPaused() && !track->isTerminated()) 3133 { 3134 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3135 3136 mixedTracks++; 3137 3138 // track->mainBuffer() != mMixBuffer means there is an effect chain 3139 // connected to the track 3140 chain.clear(); 3141 if (track->mainBuffer() != mMixBuffer) { 3142 chain = getEffectChain_l(track->sessionId()); 3143 // Delegate volume control to effect in track effect chain if needed 3144 if (chain != 0) { 3145 tracksWithEffect++; 3146 } else { 3147 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3148 name, track->sessionId()); 3149 } 3150 } 3151 3152 3153 int param = AudioMixer::VOLUME; 3154 if (track->mFillingUpStatus == Track::FS_FILLED) { 3155 // no ramp for the first volume setting 3156 track->mFillingUpStatus = Track::FS_ACTIVE; 3157 if (track->mState == TrackBase::RESUMING) { 3158 track->mState = TrackBase::ACTIVE; 3159 param = AudioMixer::RAMP_VOLUME; 3160 } 3161 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3162 } else if (cblk->server != 0) { 3163 // If the track is stopped before the first frame was mixed, 3164 // do not apply ramp 3165 param = AudioMixer::RAMP_VOLUME; 3166 } 3167 3168 // compute volume for this track 3169 uint32_t vl, vr, va; 3170 if (track->isMuted() || track->isPausing() || 3171 mStreamTypes[track->streamType()].mute) { 3172 vl = vr = va = 0; 3173 if (track->isPausing()) { 3174 track->setPaused(); 3175 } 3176 } else { 3177 3178 // read original volumes with volume control 3179 float typeVolume = mStreamTypes[track->streamType()].volume; 3180 float v = masterVolume * typeVolume; 3181 uint32_t vlr = cblk->getVolumeLR(); 3182 vl = vlr & 0xFFFF; 3183 vr = vlr >> 16; 3184 // track volumes come from shared memory, so can't be trusted and must be clamped 3185 if (vl > MAX_GAIN_INT) { 3186 ALOGV("Track left volume out of range: %04X", vl); 3187 vl = MAX_GAIN_INT; 3188 } 3189 if (vr > MAX_GAIN_INT) { 3190 ALOGV("Track right volume out of range: %04X", vr); 3191 vr = MAX_GAIN_INT; 3192 } 3193 // now apply the master volume and stream type volume 3194 vl = (uint32_t)(v * vl) << 12; 3195 vr = (uint32_t)(v * vr) << 12; 3196 // assuming master volume and stream type volume each go up to 1.0, 3197 // vl and vr are now in 8.24 format 3198 3199 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3200 // send level comes from shared memory and so may be corrupt 3201 if (sendLevel > MAX_GAIN_INT) { 3202 ALOGV("Track send level out of range: %04X", sendLevel); 3203 sendLevel = MAX_GAIN_INT; 3204 } 3205 va = (uint32_t)(v * sendLevel); 3206 } 3207 // Delegate volume control to effect in track effect chain if needed 3208 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3209 // Do not ramp volume if volume is controlled by effect 3210 param = AudioMixer::VOLUME; 3211 track->mHasVolumeController = true; 3212 } else { 3213 // force no volume ramp when volume controller was just disabled or removed 3214 // from effect chain to avoid volume spike 3215 if (track->mHasVolumeController) { 3216 param = AudioMixer::VOLUME; 3217 } 3218 track->mHasVolumeController = false; 3219 } 3220 3221 // Convert volumes from 8.24 to 4.12 format 3222 // This additional clamping is needed in case chain->setVolume_l() overshot 3223 vl = (vl + (1 << 11)) >> 12; 3224 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3225 vr = (vr + (1 << 11)) >> 12; 3226 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3227 3228 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3229 3230 // XXX: these things DON'T need to be done each time 3231 mAudioMixer->setBufferProvider(name, track); 3232 mAudioMixer->enable(name); 3233 3234 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3235 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3236 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3237 mAudioMixer->setParameter( 3238 name, 3239 AudioMixer::TRACK, 3240 AudioMixer::FORMAT, (void *)track->format()); 3241 mAudioMixer->setParameter( 3242 name, 3243 AudioMixer::TRACK, 3244 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3245 mAudioMixer->setParameter( 3246 name, 3247 AudioMixer::RESAMPLE, 3248 AudioMixer::SAMPLE_RATE, 3249 (void *)(cblk->sampleRate)); 3250 mAudioMixer->setParameter( 3251 name, 3252 AudioMixer::TRACK, 3253 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3254 mAudioMixer->setParameter( 3255 name, 3256 AudioMixer::TRACK, 3257 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3258 3259 // reset retry count 3260 track->mRetryCount = kMaxTrackRetries; 3261 3262 // If one track is ready, set the mixer ready if: 3263 // - the mixer was not ready during previous round OR 3264 // - no other track is not ready 3265 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3266 mixerStatus != MIXER_TRACKS_ENABLED) { 3267 mixerStatus = MIXER_TRACKS_READY; 3268 } 3269 } else { 3270 // clear effect chain input buffer if an active track underruns to avoid sending 3271 // previous audio buffer again to effects 3272 chain = getEffectChain_l(track->sessionId()); 3273 if (chain != 0) { 3274 chain->clearInputBuffer(); 3275 } 3276 3277 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3278 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3279 track->isStopped() || track->isPaused()) { 3280 // We have consumed all the buffers of this track. 3281 // Remove it from the list of active tracks. 3282 // TODO: use actual buffer filling status instead of latency when available from 3283 // audio HAL 3284 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3285 size_t framesWritten = 3286 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3287 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3288 if (track->isStopped()) { 3289 track->reset(); 3290 } 3291 tracksToRemove->add(track); 3292 } 3293 } else { 3294 track->mUnderrunCount++; 3295 // No buffers for this track. Give it a few chances to 3296 // fill a buffer, then remove it from active list. 3297 if (--(track->mRetryCount) <= 0) { 3298 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3299 tracksToRemove->add(track); 3300 // indicate to client process that the track was disabled because of underrun; 3301 // it will then automatically call start() when data is available 3302 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3303 // If one track is not ready, mark the mixer also not ready if: 3304 // - the mixer was ready during previous round OR 3305 // - no other track is ready 3306 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3307 mixerStatus != MIXER_TRACKS_READY) { 3308 mixerStatus = MIXER_TRACKS_ENABLED; 3309 } 3310 } 3311 mAudioMixer->disable(name); 3312 } 3313 3314 } // local variable scope to avoid goto warning 3315track_is_ready: ; 3316 3317 } 3318 3319 // Push the new FastMixer state if necessary 3320 bool pauseAudioWatchdog = false; 3321 if (didModify) { 3322 state->mFastTracksGen++; 3323 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3324 if (kUseFastMixer == FastMixer_Dynamic && 3325 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3326 state->mCommand = FastMixerState::COLD_IDLE; 3327 state->mColdFutexAddr = &mFastMixerFutex; 3328 state->mColdGen++; 3329 mFastMixerFutex = 0; 3330 if (kUseFastMixer == FastMixer_Dynamic) { 3331 mNormalSink = mOutputSink; 3332 } 3333 // If we go into cold idle, need to wait for acknowledgement 3334 // so that fast mixer stops doing I/O. 3335 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3336 pauseAudioWatchdog = true; 3337 } 3338 sq->end(); 3339 } 3340 if (sq != NULL) { 3341 sq->end(didModify); 3342 sq->push(block); 3343 } 3344#ifdef AUDIO_WATCHDOG 3345 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3346 mAudioWatchdog->pause(); 3347 } 3348#endif 3349 3350 // Now perform the deferred reset on fast tracks that have stopped 3351 while (resetMask != 0) { 3352 size_t i = __builtin_ctz(resetMask); 3353 ALOG_ASSERT(i < count); 3354 resetMask &= ~(1 << i); 3355 sp<Track> t = mActiveTracks[i].promote(); 3356 if (t == 0) continue; 3357 Track* track = t.get(); 3358 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3359 track->reset(); 3360 } 3361 3362 // remove all the tracks that need to be... 3363 count = tracksToRemove->size(); 3364 if (CC_UNLIKELY(count)) { 3365 for (size_t i=0 ; i<count ; i++) { 3366 const sp<Track>& track = tracksToRemove->itemAt(i); 3367 mActiveTracks.remove(track); 3368 if (track->mainBuffer() != mMixBuffer) { 3369 chain = getEffectChain_l(track->sessionId()); 3370 if (chain != 0) { 3371 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3372 chain->decActiveTrackCnt(); 3373 } 3374 } 3375 if (track->isTerminated()) { 3376 removeTrack_l(track); 3377 } 3378 } 3379 } 3380 3381 // mix buffer must be cleared if all tracks are connected to an 3382 // effect chain as in this case the mixer will not write to 3383 // mix buffer and track effects will accumulate into it 3384 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3385 // FIXME as a performance optimization, should remember previous zero status 3386 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3387 } 3388 3389 // if any fast tracks, then status is ready 3390 mMixerStatusIgnoringFastTracks = mixerStatus; 3391 if (fastTracks > 0) { 3392 mixerStatus = MIXER_TRACKS_READY; 3393 } 3394 return mixerStatus; 3395} 3396 3397/* 3398The derived values that are cached: 3399 - mixBufferSize from frame count * frame size 3400 - activeSleepTime from activeSleepTimeUs() 3401 - idleSleepTime from idleSleepTimeUs() 3402 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3403 - maxPeriod from frame count and sample rate (MIXER only) 3404 3405The parameters that affect these derived values are: 3406 - frame count 3407 - frame size 3408 - sample rate 3409 - device type: A2DP or not 3410 - device latency 3411 - format: PCM or not 3412 - active sleep time 3413 - idle sleep time 3414*/ 3415 3416void AudioFlinger::PlaybackThread::cacheParameters_l() 3417{ 3418 mixBufferSize = mNormalFrameCount * mFrameSize; 3419 activeSleepTime = activeSleepTimeUs(); 3420 idleSleepTime = idleSleepTimeUs(); 3421} 3422 3423void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3424{ 3425 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3426 this, streamType, mTracks.size()); 3427 Mutex::Autolock _l(mLock); 3428 3429 size_t size = mTracks.size(); 3430 for (size_t i = 0; i < size; i++) { 3431 sp<Track> t = mTracks[i]; 3432 if (t->streamType() == streamType) { 3433 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3434 t->mCblk->cv.signal(); 3435 } 3436 } 3437} 3438 3439// getTrackName_l() must be called with ThreadBase::mLock held 3440int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3441{ 3442 return mAudioMixer->getTrackName(channelMask, sessionId); 3443} 3444 3445// deleteTrackName_l() must be called with ThreadBase::mLock held 3446void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3447{ 3448 ALOGV("remove track (%d) and delete from mixer", name); 3449 mAudioMixer->deleteTrackName(name); 3450} 3451 3452// checkForNewParameters_l() must be called with ThreadBase::mLock held 3453bool AudioFlinger::MixerThread::checkForNewParameters_l() 3454{ 3455 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3456 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3457 bool reconfig = false; 3458 3459 while (!mNewParameters.isEmpty()) { 3460 3461 if (mFastMixer != NULL) { 3462 FastMixerStateQueue *sq = mFastMixer->sq(); 3463 FastMixerState *state = sq->begin(); 3464 if (!(state->mCommand & FastMixerState::IDLE)) { 3465 previousCommand = state->mCommand; 3466 state->mCommand = FastMixerState::HOT_IDLE; 3467 sq->end(); 3468 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3469 } else { 3470 sq->end(false /*didModify*/); 3471 } 3472 } 3473 3474 status_t status = NO_ERROR; 3475 String8 keyValuePair = mNewParameters[0]; 3476 AudioParameter param = AudioParameter(keyValuePair); 3477 int value; 3478 3479 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3480 reconfig = true; 3481 } 3482 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3483 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3484 status = BAD_VALUE; 3485 } else { 3486 reconfig = true; 3487 } 3488 } 3489 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3490 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3491 status = BAD_VALUE; 3492 } else { 3493 reconfig = true; 3494 } 3495 } 3496 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3497 // do not accept frame count changes if tracks are open as the track buffer 3498 // size depends on frame count and correct behavior would not be guaranteed 3499 // if frame count is changed after track creation 3500 if (!mTracks.isEmpty()) { 3501 status = INVALID_OPERATION; 3502 } else { 3503 reconfig = true; 3504 } 3505 } 3506 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3507#ifdef ADD_BATTERY_DATA 3508 // when changing the audio output device, call addBatteryData to notify 3509 // the change 3510 if (mOutDevice != value) { 3511 uint32_t params = 0; 3512 // check whether speaker is on 3513 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3514 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3515 } 3516 3517 audio_devices_t deviceWithoutSpeaker 3518 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3519 // check if any other device (except speaker) is on 3520 if (value & deviceWithoutSpeaker ) { 3521 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3522 } 3523 3524 if (params != 0) { 3525 addBatteryData(params); 3526 } 3527 } 3528#endif 3529 3530 // forward device change to effects that have requested to be 3531 // aware of attached audio device. 3532 mOutDevice = value; 3533 for (size_t i = 0; i < mEffectChains.size(); i++) { 3534 mEffectChains[i]->setDevice_l(mOutDevice); 3535 } 3536 } 3537 3538 if (status == NO_ERROR) { 3539 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3540 keyValuePair.string()); 3541 if (!mStandby && status == INVALID_OPERATION) { 3542 mOutput->stream->common.standby(&mOutput->stream->common); 3543 mStandby = true; 3544 mBytesWritten = 0; 3545 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3546 keyValuePair.string()); 3547 } 3548 if (status == NO_ERROR && reconfig) { 3549 delete mAudioMixer; 3550 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3551 mAudioMixer = NULL; 3552 readOutputParameters(); 3553 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3554 for (size_t i = 0; i < mTracks.size() ; i++) { 3555 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3556 if (name < 0) break; 3557 mTracks[i]->mName = name; 3558 // limit track sample rate to 2 x new output sample rate 3559 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3560 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3561 } 3562 } 3563 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3564 } 3565 } 3566 3567 mNewParameters.removeAt(0); 3568 3569 mParamStatus = status; 3570 mParamCond.signal(); 3571 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3572 // already timed out waiting for the status and will never signal the condition. 3573 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3574 } 3575 3576 if (!(previousCommand & FastMixerState::IDLE)) { 3577 ALOG_ASSERT(mFastMixer != NULL); 3578 FastMixerStateQueue *sq = mFastMixer->sq(); 3579 FastMixerState *state = sq->begin(); 3580 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3581 state->mCommand = previousCommand; 3582 sq->end(); 3583 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3584 } 3585 3586 return reconfig; 3587} 3588 3589void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3590{ 3591 NBAIO_Source *teeSource = source.get(); 3592 if (teeSource != NULL) { 3593 char teeTime[16]; 3594 struct timeval tv; 3595 gettimeofday(&tv, NULL); 3596 struct tm tm; 3597 localtime_r(&tv.tv_sec, &tm); 3598 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3599 char teePath[64]; 3600 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3601 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3602 if (teeFd >= 0) { 3603 char wavHeader[44]; 3604 memcpy(wavHeader, 3605 "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", 3606 sizeof(wavHeader)); 3607 NBAIO_Format format = teeSource->format(); 3608 unsigned channelCount = Format_channelCount(format); 3609 ALOG_ASSERT(channelCount <= FCC_2); 3610 unsigned sampleRate = Format_sampleRate(format); 3611 wavHeader[22] = channelCount; // number of channels 3612 wavHeader[24] = sampleRate; // sample rate 3613 wavHeader[25] = sampleRate >> 8; 3614 wavHeader[32] = channelCount * 2; // block alignment 3615 write(teeFd, wavHeader, sizeof(wavHeader)); 3616 size_t total = 0; 3617 bool firstRead = true; 3618 for (;;) { 3619#define TEE_SINK_READ 1024 3620 short buffer[TEE_SINK_READ * FCC_2]; 3621 size_t count = TEE_SINK_READ; 3622 ssize_t actual = teeSource->read(buffer, count, 3623 AudioBufferProvider::kInvalidPTS); 3624 bool wasFirstRead = firstRead; 3625 firstRead = false; 3626 if (actual <= 0) { 3627 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3628 continue; 3629 } 3630 break; 3631 } 3632 ALOG_ASSERT(actual <= (ssize_t)count); 3633 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3634 total += actual; 3635 } 3636 lseek(teeFd, (off_t) 4, SEEK_SET); 3637 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3638 write(teeFd, &temp, sizeof(temp)); 3639 lseek(teeFd, (off_t) 40, SEEK_SET); 3640 temp = total * channelCount * sizeof(short); 3641 write(teeFd, &temp, sizeof(temp)); 3642 close(teeFd); 3643 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3644 } else { 3645 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3646 } 3647 } 3648} 3649 3650void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3651{ 3652 const size_t SIZE = 256; 3653 char buffer[SIZE]; 3654 String8 result; 3655 3656 PlaybackThread::dumpInternals(fd, args); 3657 3658 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3659 result.append(buffer); 3660 write(fd, result.string(), result.size()); 3661 3662 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3663 FastMixerDumpState copy = mFastMixerDumpState; 3664 copy.dump(fd); 3665 3666#ifdef STATE_QUEUE_DUMP 3667 // Similar for state queue 3668 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3669 observerCopy.dump(fd); 3670 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3671 mutatorCopy.dump(fd); 3672#endif 3673 3674 // Write the tee output to a .wav file 3675 dumpTee(fd, mTeeSource, mId); 3676 3677#ifdef AUDIO_WATCHDOG 3678 if (mAudioWatchdog != 0) { 3679 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3680 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3681 wdCopy.dump(fd); 3682 } 3683#endif 3684} 3685 3686uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3687{ 3688 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3689} 3690 3691uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3692{ 3693 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3694} 3695 3696void AudioFlinger::MixerThread::cacheParameters_l() 3697{ 3698 PlaybackThread::cacheParameters_l(); 3699 3700 // FIXME: Relaxed timing because of a certain device that can't meet latency 3701 // Should be reduced to 2x after the vendor fixes the driver issue 3702 // increase threshold again due to low power audio mode. The way this warning 3703 // threshold is calculated and its usefulness should be reconsidered anyway. 3704 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3705} 3706 3707// ---------------------------------------------------------------------------- 3708AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3709 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3710 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3711 // mLeftVolFloat, mRightVolFloat 3712{ 3713} 3714 3715AudioFlinger::DirectOutputThread::~DirectOutputThread() 3716{ 3717} 3718 3719AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3720 Vector< sp<Track> > *tracksToRemove 3721) 3722{ 3723 sp<Track> trackToRemove; 3724 3725 mixer_state mixerStatus = MIXER_IDLE; 3726 3727 // find out which tracks need to be processed 3728 if (mActiveTracks.size() != 0) { 3729 sp<Track> t = mActiveTracks[0].promote(); 3730 // The track died recently 3731 if (t == 0) return MIXER_IDLE; 3732 3733 Track* const track = t.get(); 3734 audio_track_cblk_t* cblk = track->cblk(); 3735 3736 // The first time a track is added we wait 3737 // for all its buffers to be filled before processing it 3738 uint32_t minFrames; 3739 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3740 minFrames = mNormalFrameCount; 3741 } else { 3742 minFrames = 1; 3743 } 3744 if ((track->framesReady() >= minFrames) && track->isReady() && 3745 !track->isPaused() && !track->isTerminated()) 3746 { 3747 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3748 3749 if (track->mFillingUpStatus == Track::FS_FILLED) { 3750 track->mFillingUpStatus = Track::FS_ACTIVE; 3751 mLeftVolFloat = mRightVolFloat = 0; 3752 if (track->mState == TrackBase::RESUMING) { 3753 track->mState = TrackBase::ACTIVE; 3754 } 3755 } 3756 3757 // compute volume for this track 3758 float left, right; 3759 if (track->isMuted() || mMasterMute || track->isPausing() || 3760 mStreamTypes[track->streamType()].mute) { 3761 left = right = 0; 3762 if (track->isPausing()) { 3763 track->setPaused(); 3764 } 3765 } else { 3766 float typeVolume = mStreamTypes[track->streamType()].volume; 3767 float v = mMasterVolume * typeVolume; 3768 uint32_t vlr = cblk->getVolumeLR(); 3769 float v_clamped = v * (vlr & 0xFFFF); 3770 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3771 left = v_clamped/MAX_GAIN; 3772 v_clamped = v * (vlr >> 16); 3773 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3774 right = v_clamped/MAX_GAIN; 3775 } 3776 3777 if (left != mLeftVolFloat || right != mRightVolFloat) { 3778 mLeftVolFloat = left; 3779 mRightVolFloat = right; 3780 3781 // Convert volumes from float to 8.24 3782 uint32_t vl = (uint32_t)(left * (1 << 24)); 3783 uint32_t vr = (uint32_t)(right * (1 << 24)); 3784 3785 // Delegate volume control to effect in track effect chain if needed 3786 // only one effect chain can be present on DirectOutputThread, so if 3787 // there is one, the track is connected to it 3788 if (!mEffectChains.isEmpty()) { 3789 // Do not ramp volume if volume is controlled by effect 3790 mEffectChains[0]->setVolume_l(&vl, &vr); 3791 left = (float)vl / (1 << 24); 3792 right = (float)vr / (1 << 24); 3793 } 3794 mOutput->stream->set_volume(mOutput->stream, left, right); 3795 } 3796 3797 // reset retry count 3798 track->mRetryCount = kMaxTrackRetriesDirect; 3799 mActiveTrack = t; 3800 mixerStatus = MIXER_TRACKS_READY; 3801 } else { 3802 // clear effect chain input buffer if an active track underruns to avoid sending 3803 // previous audio buffer again to effects 3804 if (!mEffectChains.isEmpty()) { 3805 mEffectChains[0]->clearInputBuffer(); 3806 } 3807 3808 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3809 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3810 track->isStopped() || track->isPaused()) { 3811 // We have consumed all the buffers of this track. 3812 // Remove it from the list of active tracks. 3813 // TODO: implement behavior for compressed audio 3814 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3815 size_t framesWritten = 3816 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3817 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3818 if (track->isStopped()) { 3819 track->reset(); 3820 } 3821 trackToRemove = track; 3822 } 3823 } else { 3824 // No buffers for this track. Give it a few chances to 3825 // fill a buffer, then remove it from active list. 3826 if (--(track->mRetryCount) <= 0) { 3827 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3828 trackToRemove = track; 3829 } else { 3830 mixerStatus = MIXER_TRACKS_ENABLED; 3831 } 3832 } 3833 } 3834 } 3835 3836 // FIXME merge this with similar code for removing multiple tracks 3837 // remove all the tracks that need to be... 3838 if (CC_UNLIKELY(trackToRemove != 0)) { 3839 tracksToRemove->add(trackToRemove); 3840 mActiveTracks.remove(trackToRemove); 3841 if (!mEffectChains.isEmpty()) { 3842 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3843 trackToRemove->sessionId()); 3844 mEffectChains[0]->decActiveTrackCnt(); 3845 } 3846 if (trackToRemove->isTerminated()) { 3847 removeTrack_l(trackToRemove); 3848 } 3849 } 3850 3851 return mixerStatus; 3852} 3853 3854void AudioFlinger::DirectOutputThread::threadLoop_mix() 3855{ 3856 AudioBufferProvider::Buffer buffer; 3857 size_t frameCount = mFrameCount; 3858 int8_t *curBuf = (int8_t *)mMixBuffer; 3859 // output audio to hardware 3860 while (frameCount) { 3861 buffer.frameCount = frameCount; 3862 mActiveTrack->getNextBuffer(&buffer); 3863 if (CC_UNLIKELY(buffer.raw == NULL)) { 3864 memset(curBuf, 0, frameCount * mFrameSize); 3865 break; 3866 } 3867 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3868 frameCount -= buffer.frameCount; 3869 curBuf += buffer.frameCount * mFrameSize; 3870 mActiveTrack->releaseBuffer(&buffer); 3871 } 3872 sleepTime = 0; 3873 standbyTime = systemTime() + standbyDelay; 3874 mActiveTrack.clear(); 3875 3876} 3877 3878void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3879{ 3880 if (sleepTime == 0) { 3881 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3882 sleepTime = activeSleepTime; 3883 } else { 3884 sleepTime = idleSleepTime; 3885 } 3886 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3887 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3888 sleepTime = 0; 3889 } 3890} 3891 3892// getTrackName_l() must be called with ThreadBase::mLock held 3893int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3894 int sessionId) 3895{ 3896 return 0; 3897} 3898 3899// deleteTrackName_l() must be called with ThreadBase::mLock held 3900void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3901{ 3902} 3903 3904// checkForNewParameters_l() must be called with ThreadBase::mLock held 3905bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3906{ 3907 bool reconfig = false; 3908 3909 while (!mNewParameters.isEmpty()) { 3910 status_t status = NO_ERROR; 3911 String8 keyValuePair = mNewParameters[0]; 3912 AudioParameter param = AudioParameter(keyValuePair); 3913 int value; 3914 3915 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3916 // do not accept frame count changes if tracks are open as the track buffer 3917 // size depends on frame count and correct behavior would not be garantied 3918 // if frame count is changed after track creation 3919 if (!mTracks.isEmpty()) { 3920 status = INVALID_OPERATION; 3921 } else { 3922 reconfig = true; 3923 } 3924 } 3925 if (status == NO_ERROR) { 3926 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3927 keyValuePair.string()); 3928 if (!mStandby && status == INVALID_OPERATION) { 3929 mOutput->stream->common.standby(&mOutput->stream->common); 3930 mStandby = true; 3931 mBytesWritten = 0; 3932 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3933 keyValuePair.string()); 3934 } 3935 if (status == NO_ERROR && reconfig) { 3936 readOutputParameters(); 3937 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3938 } 3939 } 3940 3941 mNewParameters.removeAt(0); 3942 3943 mParamStatus = status; 3944 mParamCond.signal(); 3945 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3946 // already timed out waiting for the status and will never signal the condition. 3947 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3948 } 3949 return reconfig; 3950} 3951 3952uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3953{ 3954 uint32_t time; 3955 if (audio_is_linear_pcm(mFormat)) { 3956 time = PlaybackThread::activeSleepTimeUs(); 3957 } else { 3958 time = 10000; 3959 } 3960 return time; 3961} 3962 3963uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3964{ 3965 uint32_t time; 3966 if (audio_is_linear_pcm(mFormat)) { 3967 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3968 } else { 3969 time = 10000; 3970 } 3971 return time; 3972} 3973 3974uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3975{ 3976 uint32_t time; 3977 if (audio_is_linear_pcm(mFormat)) { 3978 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3979 } else { 3980 time = 10000; 3981 } 3982 return time; 3983} 3984 3985void AudioFlinger::DirectOutputThread::cacheParameters_l() 3986{ 3987 PlaybackThread::cacheParameters_l(); 3988 3989 // use shorter standby delay as on normal output to release 3990 // hardware resources as soon as possible 3991 standbyDelay = microseconds(activeSleepTime*2); 3992} 3993 3994// ---------------------------------------------------------------------------- 3995 3996AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3997 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3998 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING), 3999 mWaitTimeMs(UINT_MAX) 4000{ 4001 addOutputTrack(mainThread); 4002} 4003 4004AudioFlinger::DuplicatingThread::~DuplicatingThread() 4005{ 4006 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4007 mOutputTracks[i]->destroy(); 4008 } 4009} 4010 4011void AudioFlinger::DuplicatingThread::threadLoop_mix() 4012{ 4013 // mix buffers... 4014 if (outputsReady(outputTracks)) { 4015 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4016 } else { 4017 memset(mMixBuffer, 0, mixBufferSize); 4018 } 4019 sleepTime = 0; 4020 writeFrames = mNormalFrameCount; 4021 standbyTime = systemTime() + standbyDelay; 4022} 4023 4024void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4025{ 4026 if (sleepTime == 0) { 4027 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4028 sleepTime = activeSleepTime; 4029 } else { 4030 sleepTime = idleSleepTime; 4031 } 4032 } else if (mBytesWritten != 0) { 4033 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4034 writeFrames = mNormalFrameCount; 4035 memset(mMixBuffer, 0, mixBufferSize); 4036 } else { 4037 // flush remaining overflow buffers in output tracks 4038 writeFrames = 0; 4039 } 4040 sleepTime = 0; 4041 } 4042} 4043 4044void AudioFlinger::DuplicatingThread::threadLoop_write() 4045{ 4046 for (size_t i = 0; i < outputTracks.size(); i++) { 4047 outputTracks[i]->write(mMixBuffer, writeFrames); 4048 } 4049 mBytesWritten += mixBufferSize; 4050} 4051 4052void AudioFlinger::DuplicatingThread::threadLoop_standby() 4053{ 4054 // DuplicatingThread implements standby by stopping all tracks 4055 for (size_t i = 0; i < outputTracks.size(); i++) { 4056 outputTracks[i]->stop(); 4057 } 4058} 4059 4060void AudioFlinger::DuplicatingThread::saveOutputTracks() 4061{ 4062 outputTracks = mOutputTracks; 4063} 4064 4065void AudioFlinger::DuplicatingThread::clearOutputTracks() 4066{ 4067 outputTracks.clear(); 4068} 4069 4070void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4071{ 4072 Mutex::Autolock _l(mLock); 4073 // FIXME explain this formula 4074 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4075 OutputTrack *outputTrack = new OutputTrack(thread, 4076 this, 4077 mSampleRate, 4078 mFormat, 4079 mChannelMask, 4080 frameCount); 4081 if (outputTrack->cblk() != NULL) { 4082 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4083 mOutputTracks.add(outputTrack); 4084 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4085 updateWaitTime_l(); 4086 } 4087} 4088 4089void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4090{ 4091 Mutex::Autolock _l(mLock); 4092 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4093 if (mOutputTracks[i]->thread() == thread) { 4094 mOutputTracks[i]->destroy(); 4095 mOutputTracks.removeAt(i); 4096 updateWaitTime_l(); 4097 return; 4098 } 4099 } 4100 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4101} 4102 4103// caller must hold mLock 4104void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4105{ 4106 mWaitTimeMs = UINT_MAX; 4107 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4108 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4109 if (strong != 0) { 4110 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4111 if (waitTimeMs < mWaitTimeMs) { 4112 mWaitTimeMs = waitTimeMs; 4113 } 4114 } 4115 } 4116} 4117 4118 4119bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4120{ 4121 for (size_t i = 0; i < outputTracks.size(); i++) { 4122 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4123 if (thread == 0) { 4124 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4125 return false; 4126 } 4127 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4128 // see note at standby() declaration 4129 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4130 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4131 return false; 4132 } 4133 } 4134 return true; 4135} 4136 4137uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4138{ 4139 return (mWaitTimeMs * 1000) / 2; 4140} 4141 4142void AudioFlinger::DuplicatingThread::cacheParameters_l() 4143{ 4144 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4145 updateWaitTime_l(); 4146 4147 MixerThread::cacheParameters_l(); 4148} 4149 4150// ---------------------------------------------------------------------------- 4151 4152// TrackBase constructor must be called with AudioFlinger::mLock held 4153AudioFlinger::ThreadBase::TrackBase::TrackBase( 4154 ThreadBase *thread, 4155 const sp<Client>& client, 4156 uint32_t sampleRate, 4157 audio_format_t format, 4158 audio_channel_mask_t channelMask, 4159 int frameCount, 4160 const sp<IMemory>& sharedBuffer, 4161 int sessionId) 4162 : RefBase(), 4163 mThread(thread), 4164 mClient(client), 4165 mCblk(NULL), 4166 // mBuffer 4167 // mBufferEnd 4168 mFrameCount(0), 4169 mState(IDLE), 4170 mSampleRate(sampleRate), 4171 mFormat(format), 4172 mStepServerFailed(false), 4173 mSessionId(sessionId) 4174 // mChannelCount 4175 // mChannelMask 4176{ 4177 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4178 4179 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4180 size_t size = sizeof(audio_track_cblk_t); 4181 uint8_t channelCount = popcount(channelMask); 4182 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4183 if (sharedBuffer == 0) { 4184 size += bufferSize; 4185 } 4186 4187 if (client != NULL) { 4188 mCblkMemory = client->heap()->allocate(size); 4189 if (mCblkMemory != 0) { 4190 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4191 if (mCblk != NULL) { // construct the shared structure in-place. 4192 new(mCblk) audio_track_cblk_t(); 4193 // clear all buffers 4194 mCblk->frameCount = frameCount; 4195 mCblk->sampleRate = sampleRate; 4196// uncomment the following lines to quickly test 32-bit wraparound 4197// mCblk->user = 0xffff0000; 4198// mCblk->server = 0xffff0000; 4199// mCblk->userBase = 0xffff0000; 4200// mCblk->serverBase = 0xffff0000; 4201 mChannelCount = channelCount; 4202 mChannelMask = channelMask; 4203 if (sharedBuffer == 0) { 4204 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4205 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4206 // Force underrun condition to avoid false underrun callback until first data is 4207 // written to buffer (other flags are cleared) 4208 mCblk->flags = CBLK_UNDERRUN_ON; 4209 } else { 4210 mBuffer = sharedBuffer->pointer(); 4211 } 4212 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4213 } 4214 } else { 4215 ALOGE("not enough memory for AudioTrack size=%u", size); 4216 client->heap()->dump("AudioTrack"); 4217 return; 4218 } 4219 } else { 4220 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4221 // construct the shared structure in-place. 4222 new(mCblk) audio_track_cblk_t(); 4223 // clear all buffers 4224 mCblk->frameCount = frameCount; 4225 mCblk->sampleRate = sampleRate; 4226// uncomment the following lines to quickly test 32-bit wraparound 4227// mCblk->user = 0xffff0000; 4228// mCblk->server = 0xffff0000; 4229// mCblk->userBase = 0xffff0000; 4230// mCblk->serverBase = 0xffff0000; 4231 mChannelCount = channelCount; 4232 mChannelMask = channelMask; 4233 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4234 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4235 // Force underrun condition to avoid false underrun callback until first data is 4236 // written to buffer (other flags are cleared) 4237 mCblk->flags = CBLK_UNDERRUN_ON; 4238 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4239 } 4240} 4241 4242AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4243{ 4244 if (mCblk != NULL) { 4245 if (mClient == 0) { 4246 delete mCblk; 4247 } else { 4248 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4249 } 4250 } 4251 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4252 if (mClient != 0) { 4253 // Client destructor must run with AudioFlinger mutex locked 4254 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4255 // If the client's reference count drops to zero, the associated destructor 4256 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4257 // relying on the automatic clear() at end of scope. 4258 mClient.clear(); 4259 } 4260} 4261 4262// AudioBufferProvider interface 4263// getNextBuffer() = 0; 4264// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4265void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4266{ 4267 buffer->raw = NULL; 4268 mFrameCount = buffer->frameCount; 4269 // FIXME See note at getNextBuffer() 4270 (void) step(); // ignore return value of step() 4271 buffer->frameCount = 0; 4272} 4273 4274bool AudioFlinger::ThreadBase::TrackBase::step() { 4275 bool result; 4276 audio_track_cblk_t* cblk = this->cblk(); 4277 4278 result = cblk->stepServer(mFrameCount); 4279 if (!result) { 4280 ALOGV("stepServer failed acquiring cblk mutex"); 4281 mStepServerFailed = true; 4282 } 4283 return result; 4284} 4285 4286void AudioFlinger::ThreadBase::TrackBase::reset() { 4287 audio_track_cblk_t* cblk = this->cblk(); 4288 4289 cblk->user = 0; 4290 cblk->server = 0; 4291 cblk->userBase = 0; 4292 cblk->serverBase = 0; 4293 mStepServerFailed = false; 4294 ALOGV("TrackBase::reset"); 4295} 4296 4297int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4298 return (int)mCblk->sampleRate; 4299} 4300 4301void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4302 audio_track_cblk_t* cblk = this->cblk(); 4303 size_t frameSize = cblk->frameSize; 4304 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4305 int8_t *bufferEnd = bufferStart + frames * frameSize; 4306 4307 // Check validity of returned pointer in case the track control block would have been corrupted. 4308 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4309 "TrackBase::getBuffer buffer out of range:\n" 4310 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4311 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4312 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4313 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4314 4315 return bufferStart; 4316} 4317 4318status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4319{ 4320 mSyncEvents.add(event); 4321 return NO_ERROR; 4322} 4323 4324// ---------------------------------------------------------------------------- 4325 4326// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4327AudioFlinger::PlaybackThread::Track::Track( 4328 PlaybackThread *thread, 4329 const sp<Client>& client, 4330 audio_stream_type_t streamType, 4331 uint32_t sampleRate, 4332 audio_format_t format, 4333 audio_channel_mask_t channelMask, 4334 int frameCount, 4335 const sp<IMemory>& sharedBuffer, 4336 int sessionId, 4337 IAudioFlinger::track_flags_t flags) 4338 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4339 mMute(false), 4340 mFillingUpStatus(FS_INVALID), 4341 // mRetryCount initialized later when needed 4342 mSharedBuffer(sharedBuffer), 4343 mStreamType(streamType), 4344 mName(-1), // see note below 4345 mMainBuffer(thread->mixBuffer()), 4346 mAuxBuffer(NULL), 4347 mAuxEffectId(0), mHasVolumeController(false), 4348 mPresentationCompleteFrames(0), 4349 mFlags(flags), 4350 mFastIndex(-1), 4351 mUnderrunCount(0), 4352 mCachedVolume(1.0) 4353{ 4354 if (mCblk != NULL) { 4355 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4356 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4357 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4358 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4359 mName = thread->getTrackName_l(channelMask, sessionId); 4360 mCblk->mName = mName; 4361 if (mName < 0) { 4362 ALOGE("no more track names available"); 4363 return; 4364 } 4365 // only allocate a fast track index if we were able to allocate a normal track name 4366 if (flags & IAudioFlinger::TRACK_FAST) { 4367 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4368 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4369 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4370 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4371 // FIXME This is too eager. We allocate a fast track index before the 4372 // fast track becomes active. Since fast tracks are a scarce resource, 4373 // this means we are potentially denying other more important fast tracks from 4374 // being created. It would be better to allocate the index dynamically. 4375 mFastIndex = i; 4376 mCblk->mName = i; 4377 // Read the initial underruns because this field is never cleared by the fast mixer 4378 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4379 thread->mFastTrackAvailMask &= ~(1 << i); 4380 } 4381 } 4382 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4383} 4384 4385AudioFlinger::PlaybackThread::Track::~Track() 4386{ 4387 ALOGV("PlaybackThread::Track destructor"); 4388} 4389 4390void AudioFlinger::PlaybackThread::Track::destroy() 4391{ 4392 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4393 // by removing it from mTracks vector, so there is a risk that this Tracks's 4394 // destructor is called. As the destructor needs to lock mLock, 4395 // we must acquire a strong reference on this Track before locking mLock 4396 // here so that the destructor is called only when exiting this function. 4397 // On the other hand, as long as Track::destroy() is only called by 4398 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4399 // this Track with its member mTrack. 4400 sp<Track> keep(this); 4401 { // scope for mLock 4402 sp<ThreadBase> thread = mThread.promote(); 4403 if (thread != 0) { 4404 if (!isOutputTrack()) { 4405 if (mState == ACTIVE || mState == RESUMING) { 4406 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4407 4408#ifdef ADD_BATTERY_DATA 4409 // to track the speaker usage 4410 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4411#endif 4412 } 4413 AudioSystem::releaseOutput(thread->id()); 4414 } 4415 Mutex::Autolock _l(thread->mLock); 4416 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4417 playbackThread->destroyTrack_l(this); 4418 } 4419 } 4420} 4421 4422/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4423{ 4424 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4425 " Server User Main buf Aux Buf Flags Underruns\n"); 4426} 4427 4428void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4429{ 4430 uint32_t vlr = mCblk->getVolumeLR(); 4431 if (isFastTrack()) { 4432 sprintf(buffer, " F %2d", mFastIndex); 4433 } else { 4434 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4435 } 4436 track_state state = mState; 4437 char stateChar; 4438 switch (state) { 4439 case IDLE: 4440 stateChar = 'I'; 4441 break; 4442 case TERMINATED: 4443 stateChar = 'T'; 4444 break; 4445 case STOPPING_1: 4446 stateChar = 's'; 4447 break; 4448 case STOPPING_2: 4449 stateChar = '5'; 4450 break; 4451 case STOPPED: 4452 stateChar = 'S'; 4453 break; 4454 case RESUMING: 4455 stateChar = 'R'; 4456 break; 4457 case ACTIVE: 4458 stateChar = 'A'; 4459 break; 4460 case PAUSING: 4461 stateChar = 'p'; 4462 break; 4463 case PAUSED: 4464 stateChar = 'P'; 4465 break; 4466 case FLUSHED: 4467 stateChar = 'F'; 4468 break; 4469 default: 4470 stateChar = '?'; 4471 break; 4472 } 4473 char nowInUnderrun; 4474 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4475 case UNDERRUN_FULL: 4476 nowInUnderrun = ' '; 4477 break; 4478 case UNDERRUN_PARTIAL: 4479 nowInUnderrun = '<'; 4480 break; 4481 case UNDERRUN_EMPTY: 4482 nowInUnderrun = '*'; 4483 break; 4484 default: 4485 nowInUnderrun = '?'; 4486 break; 4487 } 4488 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4489 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4490 (mClient == 0) ? getpid_cached : mClient->pid(), 4491 mStreamType, 4492 mFormat, 4493 mChannelMask, 4494 mSessionId, 4495 mFrameCount, 4496 mCblk->frameCount, 4497 stateChar, 4498 mMute, 4499 mFillingUpStatus, 4500 mCblk->sampleRate, 4501 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4502 20.0 * log10((vlr >> 16) / 4096.0), 4503 mCblk->server, 4504 mCblk->user, 4505 (int)mMainBuffer, 4506 (int)mAuxBuffer, 4507 mCblk->flags, 4508 mUnderrunCount, 4509 nowInUnderrun); 4510} 4511 4512// AudioBufferProvider interface 4513status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4514 AudioBufferProvider::Buffer* buffer, int64_t pts) 4515{ 4516 audio_track_cblk_t* cblk = this->cblk(); 4517 uint32_t framesReady; 4518 uint32_t framesReq = buffer->frameCount; 4519 4520 // Check if last stepServer failed, try to step now 4521 if (mStepServerFailed) { 4522 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4523 // Since the fast mixer is higher priority than client callback thread, 4524 // it does not result in priority inversion for client. 4525 // But a non-blocking solution would be preferable to avoid 4526 // fast mixer being unable to tryLock(), and 4527 // to avoid the extra context switches if the client wakes up, 4528 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4529 if (!step()) goto getNextBuffer_exit; 4530 ALOGV("stepServer recovered"); 4531 mStepServerFailed = false; 4532 } 4533 4534 // FIXME Same as above 4535 framesReady = cblk->framesReady(); 4536 4537 if (CC_LIKELY(framesReady)) { 4538 uint32_t s = cblk->server; 4539 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4540 4541 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4542 if (framesReq > framesReady) { 4543 framesReq = framesReady; 4544 } 4545 if (framesReq > bufferEnd - s) { 4546 framesReq = bufferEnd - s; 4547 } 4548 4549 buffer->raw = getBuffer(s, framesReq); 4550 buffer->frameCount = framesReq; 4551 return NO_ERROR; 4552 } 4553 4554getNextBuffer_exit: 4555 buffer->raw = NULL; 4556 buffer->frameCount = 0; 4557 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4558 return NOT_ENOUGH_DATA; 4559} 4560 4561// Note that framesReady() takes a mutex on the control block using tryLock(). 4562// This could result in priority inversion if framesReady() is called by the normal mixer, 4563// as the normal mixer thread runs at lower 4564// priority than the client's callback thread: there is a short window within framesReady() 4565// during which the normal mixer could be preempted, and the client callback would block. 4566// Another problem can occur if framesReady() is called by the fast mixer: 4567// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4568// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4569size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4570 return mCblk->framesReady(); 4571} 4572 4573// Don't call for fast tracks; the framesReady() could result in priority inversion 4574bool AudioFlinger::PlaybackThread::Track::isReady() const { 4575 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4576 4577 if (framesReady() >= mCblk->frameCount || 4578 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4579 mFillingUpStatus = FS_FILLED; 4580 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4581 return true; 4582 } 4583 return false; 4584} 4585 4586status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4587 int triggerSession) 4588{ 4589 status_t status = NO_ERROR; 4590 ALOGV("start(%d), calling pid %d session %d", 4591 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4592 4593 sp<ThreadBase> thread = mThread.promote(); 4594 if (thread != 0) { 4595 Mutex::Autolock _l(thread->mLock); 4596 track_state state = mState; 4597 // here the track could be either new, or restarted 4598 // in both cases "unstop" the track 4599 if (mState == PAUSED) { 4600 mState = TrackBase::RESUMING; 4601 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4602 } else { 4603 mState = TrackBase::ACTIVE; 4604 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4605 } 4606 4607 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4608 thread->mLock.unlock(); 4609 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4610 thread->mLock.lock(); 4611 4612#ifdef ADD_BATTERY_DATA 4613 // to track the speaker usage 4614 if (status == NO_ERROR) { 4615 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4616 } 4617#endif 4618 } 4619 if (status == NO_ERROR) { 4620 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4621 playbackThread->addTrack_l(this); 4622 } else { 4623 mState = state; 4624 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4625 } 4626 } else { 4627 status = BAD_VALUE; 4628 } 4629 return status; 4630} 4631 4632void AudioFlinger::PlaybackThread::Track::stop() 4633{ 4634 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4635 sp<ThreadBase> thread = mThread.promote(); 4636 if (thread != 0) { 4637 Mutex::Autolock _l(thread->mLock); 4638 track_state state = mState; 4639 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4640 // If the track is not active (PAUSED and buffers full), flush buffers 4641 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4642 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4643 reset(); 4644 mState = STOPPED; 4645 } else if (!isFastTrack()) { 4646 mState = STOPPED; 4647 } else { 4648 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4649 // and then to STOPPED and reset() when presentation is complete 4650 mState = STOPPING_1; 4651 } 4652 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4653 } 4654 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4655 thread->mLock.unlock(); 4656 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4657 thread->mLock.lock(); 4658 4659#ifdef ADD_BATTERY_DATA 4660 // to track the speaker usage 4661 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4662#endif 4663 } 4664 } 4665} 4666 4667void AudioFlinger::PlaybackThread::Track::pause() 4668{ 4669 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4670 sp<ThreadBase> thread = mThread.promote(); 4671 if (thread != 0) { 4672 Mutex::Autolock _l(thread->mLock); 4673 if (mState == ACTIVE || mState == RESUMING) { 4674 mState = PAUSING; 4675 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4676 if (!isOutputTrack()) { 4677 thread->mLock.unlock(); 4678 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4679 thread->mLock.lock(); 4680 4681#ifdef ADD_BATTERY_DATA 4682 // to track the speaker usage 4683 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4684#endif 4685 } 4686 } 4687 } 4688} 4689 4690void AudioFlinger::PlaybackThread::Track::flush() 4691{ 4692 ALOGV("flush(%d)", mName); 4693 sp<ThreadBase> thread = mThread.promote(); 4694 if (thread != 0) { 4695 Mutex::Autolock _l(thread->mLock); 4696 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4697 mState != PAUSING) { 4698 return; 4699 } 4700 // No point remaining in PAUSED state after a flush => go to 4701 // FLUSHED state 4702 mState = FLUSHED; 4703 // do not reset the track if it is still in the process of being stopped or paused. 4704 // this will be done by prepareTracks_l() when the track is stopped. 4705 // prepareTracks_l() will see mState == FLUSHED, then 4706 // remove from active track list, reset(), and trigger presentation complete 4707 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4708 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4709 reset(); 4710 } 4711 } 4712} 4713 4714void AudioFlinger::PlaybackThread::Track::reset() 4715{ 4716 // Do not reset twice to avoid discarding data written just after a flush and before 4717 // the audioflinger thread detects the track is stopped. 4718 if (!mResetDone) { 4719 TrackBase::reset(); 4720 // Force underrun condition to avoid false underrun callback until first data is 4721 // written to buffer 4722 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4723 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4724 mFillingUpStatus = FS_FILLING; 4725 mResetDone = true; 4726 if (mState == FLUSHED) { 4727 mState = IDLE; 4728 } 4729 } 4730} 4731 4732void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4733{ 4734 mMute = muted; 4735} 4736 4737status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4738{ 4739 status_t status = DEAD_OBJECT; 4740 sp<ThreadBase> thread = mThread.promote(); 4741 if (thread != 0) { 4742 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4743 sp<AudioFlinger> af = mClient->audioFlinger(); 4744 4745 Mutex::Autolock _l(af->mLock); 4746 4747 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4748 4749 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4750 Mutex::Autolock _dl(playbackThread->mLock); 4751 Mutex::Autolock _sl(srcThread->mLock); 4752 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4753 if (chain == 0) { 4754 return INVALID_OPERATION; 4755 } 4756 4757 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4758 if (effect == 0) { 4759 return INVALID_OPERATION; 4760 } 4761 srcThread->removeEffect_l(effect); 4762 playbackThread->addEffect_l(effect); 4763 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4764 if (effect->state() == EffectModule::ACTIVE || 4765 effect->state() == EffectModule::STOPPING) { 4766 effect->start(); 4767 } 4768 4769 sp<EffectChain> dstChain = effect->chain().promote(); 4770 if (dstChain == 0) { 4771 srcThread->addEffect_l(effect); 4772 return INVALID_OPERATION; 4773 } 4774 AudioSystem::unregisterEffect(effect->id()); 4775 AudioSystem::registerEffect(&effect->desc(), 4776 srcThread->id(), 4777 dstChain->strategy(), 4778 AUDIO_SESSION_OUTPUT_MIX, 4779 effect->id()); 4780 } 4781 status = playbackThread->attachAuxEffect(this, EffectId); 4782 } 4783 return status; 4784} 4785 4786void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4787{ 4788 mAuxEffectId = EffectId; 4789 mAuxBuffer = buffer; 4790} 4791 4792bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4793 size_t audioHalFrames) 4794{ 4795 // a track is considered presented when the total number of frames written to audio HAL 4796 // corresponds to the number of frames written when presentationComplete() is called for the 4797 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4798 if (mPresentationCompleteFrames == 0) { 4799 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4800 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4801 mPresentationCompleteFrames, audioHalFrames); 4802 } 4803 if (framesWritten >= mPresentationCompleteFrames) { 4804 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4805 mSessionId, framesWritten); 4806 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4807 return true; 4808 } 4809 return false; 4810} 4811 4812void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4813{ 4814 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4815 if (mSyncEvents[i]->type() == type) { 4816 mSyncEvents[i]->trigger(); 4817 mSyncEvents.removeAt(i); 4818 i--; 4819 } 4820 } 4821} 4822 4823// implement VolumeBufferProvider interface 4824 4825uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4826{ 4827 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4828 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4829 uint32_t vlr = mCblk->getVolumeLR(); 4830 uint32_t vl = vlr & 0xFFFF; 4831 uint32_t vr = vlr >> 16; 4832 // track volumes come from shared memory, so can't be trusted and must be clamped 4833 if (vl > MAX_GAIN_INT) { 4834 vl = MAX_GAIN_INT; 4835 } 4836 if (vr > MAX_GAIN_INT) { 4837 vr = MAX_GAIN_INT; 4838 } 4839 // now apply the cached master volume and stream type volume; 4840 // this is trusted but lacks any synchronization or barrier so may be stale 4841 float v = mCachedVolume; 4842 vl *= v; 4843 vr *= v; 4844 // re-combine into U4.16 4845 vlr = (vr << 16) | (vl & 0xFFFF); 4846 // FIXME look at mute, pause, and stop flags 4847 return vlr; 4848} 4849 4850status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4851{ 4852 if (mState == TERMINATED || mState == PAUSED || 4853 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4854 (mState == STOPPED)))) { 4855 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4856 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4857 event->cancel(); 4858 return INVALID_OPERATION; 4859 } 4860 (void) TrackBase::setSyncEvent(event); 4861 return NO_ERROR; 4862} 4863 4864// timed audio tracks 4865 4866sp<AudioFlinger::PlaybackThread::TimedTrack> 4867AudioFlinger::PlaybackThread::TimedTrack::create( 4868 PlaybackThread *thread, 4869 const sp<Client>& client, 4870 audio_stream_type_t streamType, 4871 uint32_t sampleRate, 4872 audio_format_t format, 4873 audio_channel_mask_t channelMask, 4874 int frameCount, 4875 const sp<IMemory>& sharedBuffer, 4876 int sessionId) { 4877 if (!client->reserveTimedTrack()) 4878 return 0; 4879 4880 return new TimedTrack( 4881 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4882 sharedBuffer, sessionId); 4883} 4884 4885AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4886 PlaybackThread *thread, 4887 const sp<Client>& client, 4888 audio_stream_type_t streamType, 4889 uint32_t sampleRate, 4890 audio_format_t format, 4891 audio_channel_mask_t channelMask, 4892 int frameCount, 4893 const sp<IMemory>& sharedBuffer, 4894 int sessionId) 4895 : Track(thread, client, streamType, sampleRate, format, channelMask, 4896 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4897 mQueueHeadInFlight(false), 4898 mTrimQueueHeadOnRelease(false), 4899 mFramesPendingInQueue(0), 4900 mTimedSilenceBuffer(NULL), 4901 mTimedSilenceBufferSize(0), 4902 mTimedAudioOutputOnTime(false), 4903 mMediaTimeTransformValid(false) 4904{ 4905 LocalClock lc; 4906 mLocalTimeFreq = lc.getLocalFreq(); 4907 4908 mLocalTimeToSampleTransform.a_zero = 0; 4909 mLocalTimeToSampleTransform.b_zero = 0; 4910 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4911 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4912 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4913 &mLocalTimeToSampleTransform.a_to_b_denom); 4914 4915 mMediaTimeToSampleTransform.a_zero = 0; 4916 mMediaTimeToSampleTransform.b_zero = 0; 4917 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4918 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4919 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4920 &mMediaTimeToSampleTransform.a_to_b_denom); 4921} 4922 4923AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4924 mClient->releaseTimedTrack(); 4925 delete [] mTimedSilenceBuffer; 4926} 4927 4928status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4929 size_t size, sp<IMemory>* buffer) { 4930 4931 Mutex::Autolock _l(mTimedBufferQueueLock); 4932 4933 trimTimedBufferQueue_l(); 4934 4935 // lazily initialize the shared memory heap for timed buffers 4936 if (mTimedMemoryDealer == NULL) { 4937 const int kTimedBufferHeapSize = 512 << 10; 4938 4939 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4940 "AudioFlingerTimed"); 4941 if (mTimedMemoryDealer == NULL) 4942 return NO_MEMORY; 4943 } 4944 4945 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4946 if (newBuffer == NULL) { 4947 newBuffer = mTimedMemoryDealer->allocate(size); 4948 if (newBuffer == NULL) 4949 return NO_MEMORY; 4950 } 4951 4952 *buffer = newBuffer; 4953 return NO_ERROR; 4954} 4955 4956// caller must hold mTimedBufferQueueLock 4957void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4958 int64_t mediaTimeNow; 4959 { 4960 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4961 if (!mMediaTimeTransformValid) 4962 return; 4963 4964 int64_t targetTimeNow; 4965 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4966 ? mCCHelper.getCommonTime(&targetTimeNow) 4967 : mCCHelper.getLocalTime(&targetTimeNow); 4968 4969 if (OK != res) 4970 return; 4971 4972 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4973 &mediaTimeNow)) { 4974 return; 4975 } 4976 } 4977 4978 size_t trimEnd; 4979 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4980 int64_t bufEnd; 4981 4982 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4983 // We have a next buffer. Just use its PTS as the PTS of the frame 4984 // following the last frame in this buffer. If the stream is sparse 4985 // (ie, there are deliberate gaps left in the stream which should be 4986 // filled with silence by the TimedAudioTrack), then this can result 4987 // in one extra buffer being left un-trimmed when it could have 4988 // been. In general, this is not typical, and we would rather 4989 // optimized away the TS calculation below for the more common case 4990 // where PTSes are contiguous. 4991 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4992 } else { 4993 // We have no next buffer. Compute the PTS of the frame following 4994 // the last frame in this buffer by computing the duration of of 4995 // this frame in media time units and adding it to the PTS of the 4996 // buffer. 4997 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4998 / mCblk->frameSize; 4999 5000 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5001 &bufEnd)) { 5002 ALOGE("Failed to convert frame count of %lld to media time" 5003 " duration" " (scale factor %d/%u) in %s", 5004 frameCount, 5005 mMediaTimeToSampleTransform.a_to_b_numer, 5006 mMediaTimeToSampleTransform.a_to_b_denom, 5007 __PRETTY_FUNCTION__); 5008 break; 5009 } 5010 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5011 } 5012 5013 if (bufEnd > mediaTimeNow) 5014 break; 5015 5016 // Is the buffer we want to use in the middle of a mix operation right 5017 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5018 // from the mixer which should be coming back shortly. 5019 if (!trimEnd && mQueueHeadInFlight) { 5020 mTrimQueueHeadOnRelease = true; 5021 } 5022 } 5023 5024 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5025 if (trimStart < trimEnd) { 5026 // Update the bookkeeping for framesReady() 5027 for (size_t i = trimStart; i < trimEnd; ++i) { 5028 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5029 } 5030 5031 // Now actually remove the buffers from the queue. 5032 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5033 } 5034} 5035 5036void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5037 const char* logTag) { 5038 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5039 "%s called (reason \"%s\"), but timed buffer queue has no" 5040 " elements to trim.", __FUNCTION__, logTag); 5041 5042 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5043 mTimedBufferQueue.removeAt(0); 5044} 5045 5046void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5047 const TimedBuffer& buf, 5048 const char* logTag) { 5049 uint32_t bufBytes = buf.buffer()->size(); 5050 uint32_t consumedAlready = buf.position(); 5051 5052 ALOG_ASSERT(consumedAlready <= bufBytes, 5053 "Bad bookkeeping while updating frames pending. Timed buffer is" 5054 " only %u bytes long, but claims to have consumed %u" 5055 " bytes. (update reason: \"%s\")", 5056 bufBytes, consumedAlready, logTag); 5057 5058 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 5059 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5060 "Bad bookkeeping while updating frames pending. Should have at" 5061 " least %u queued frames, but we think we have only %u. (update" 5062 " reason: \"%s\")", 5063 bufFrames, mFramesPendingInQueue, logTag); 5064 5065 mFramesPendingInQueue -= bufFrames; 5066} 5067 5068status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5069 const sp<IMemory>& buffer, int64_t pts) { 5070 5071 { 5072 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5073 if (!mMediaTimeTransformValid) 5074 return INVALID_OPERATION; 5075 } 5076 5077 Mutex::Autolock _l(mTimedBufferQueueLock); 5078 5079 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5080 mFramesPendingInQueue += bufFrames; 5081 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5082 5083 return NO_ERROR; 5084} 5085 5086status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5087 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5088 5089 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5090 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5091 target); 5092 5093 if (!(target == TimedAudioTrack::LOCAL_TIME || 5094 target == TimedAudioTrack::COMMON_TIME)) { 5095 return BAD_VALUE; 5096 } 5097 5098 Mutex::Autolock lock(mMediaTimeTransformLock); 5099 mMediaTimeTransform = xform; 5100 mMediaTimeTransformTarget = target; 5101 mMediaTimeTransformValid = true; 5102 5103 return NO_ERROR; 5104} 5105 5106#define min(a, b) ((a) < (b) ? (a) : (b)) 5107 5108// implementation of getNextBuffer for tracks whose buffers have timestamps 5109status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5110 AudioBufferProvider::Buffer* buffer, int64_t pts) 5111{ 5112 if (pts == AudioBufferProvider::kInvalidPTS) { 5113 buffer->raw = NULL; 5114 buffer->frameCount = 0; 5115 mTimedAudioOutputOnTime = false; 5116 return INVALID_OPERATION; 5117 } 5118 5119 Mutex::Autolock _l(mTimedBufferQueueLock); 5120 5121 ALOG_ASSERT(!mQueueHeadInFlight, 5122 "getNextBuffer called without releaseBuffer!"); 5123 5124 while (true) { 5125 5126 // if we have no timed buffers, then fail 5127 if (mTimedBufferQueue.isEmpty()) { 5128 buffer->raw = NULL; 5129 buffer->frameCount = 0; 5130 return NOT_ENOUGH_DATA; 5131 } 5132 5133 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5134 5135 // calculate the PTS of the head of the timed buffer queue expressed in 5136 // local time 5137 int64_t headLocalPTS; 5138 { 5139 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5140 5141 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5142 5143 if (mMediaTimeTransform.a_to_b_denom == 0) { 5144 // the transform represents a pause, so yield silence 5145 timedYieldSilence_l(buffer->frameCount, buffer); 5146 return NO_ERROR; 5147 } 5148 5149 int64_t transformedPTS; 5150 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5151 &transformedPTS)) { 5152 // the transform failed. this shouldn't happen, but if it does 5153 // then just drop this buffer 5154 ALOGW("timedGetNextBuffer transform failed"); 5155 buffer->raw = NULL; 5156 buffer->frameCount = 0; 5157 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5158 return NO_ERROR; 5159 } 5160 5161 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5162 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5163 &headLocalPTS)) { 5164 buffer->raw = NULL; 5165 buffer->frameCount = 0; 5166 return INVALID_OPERATION; 5167 } 5168 } else { 5169 headLocalPTS = transformedPTS; 5170 } 5171 } 5172 5173 // adjust the head buffer's PTS to reflect the portion of the head buffer 5174 // that has already been consumed 5175 int64_t effectivePTS = headLocalPTS + 5176 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5177 5178 // Calculate the delta in samples between the head of the input buffer 5179 // queue and the start of the next output buffer that will be written. 5180 // If the transformation fails because of over or underflow, it means 5181 // that the sample's position in the output stream is so far out of 5182 // whack that it should just be dropped. 5183 int64_t sampleDelta; 5184 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5185 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5186 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5187 " mix"); 5188 continue; 5189 } 5190 if (!mLocalTimeToSampleTransform.doForwardTransform( 5191 (effectivePTS - pts) << 32, &sampleDelta)) { 5192 ALOGV("*** too late during sample rate transform: dropped buffer"); 5193 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5194 continue; 5195 } 5196 5197 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5198 " sampleDelta=[%d.%08x]", 5199 head.pts(), head.position(), pts, 5200 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5201 + (sampleDelta >> 32)), 5202 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5203 5204 // if the delta between the ideal placement for the next input sample and 5205 // the current output position is within this threshold, then we will 5206 // concatenate the next input samples to the previous output 5207 const int64_t kSampleContinuityThreshold = 5208 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5209 5210 // if this is the first buffer of audio that we're emitting from this track 5211 // then it should be almost exactly on time. 5212 const int64_t kSampleStartupThreshold = 1LL << 32; 5213 5214 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5215 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5216 // the next input is close enough to being on time, so concatenate it 5217 // with the last output 5218 timedYieldSamples_l(buffer); 5219 5220 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5221 head.position(), buffer->frameCount); 5222 return NO_ERROR; 5223 } 5224 5225 // Looks like our output is not on time. Reset our on timed status. 5226 // Next time we mix samples from our input queue, then should be within 5227 // the StartupThreshold. 5228 mTimedAudioOutputOnTime = false; 5229 if (sampleDelta > 0) { 5230 // the gap between the current output position and the proper start of 5231 // the next input sample is too big, so fill it with silence 5232 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5233 5234 timedYieldSilence_l(framesUntilNextInput, buffer); 5235 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5236 return NO_ERROR; 5237 } else { 5238 // the next input sample is late 5239 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5240 size_t onTimeSamplePosition = 5241 head.position() + lateFrames * mCblk->frameSize; 5242 5243 if (onTimeSamplePosition > head.buffer()->size()) { 5244 // all the remaining samples in the head are too late, so 5245 // drop it and move on 5246 ALOGV("*** too late: dropped buffer"); 5247 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5248 continue; 5249 } else { 5250 // skip over the late samples 5251 head.setPosition(onTimeSamplePosition); 5252 5253 // yield the available samples 5254 timedYieldSamples_l(buffer); 5255 5256 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5257 return NO_ERROR; 5258 } 5259 } 5260 } 5261} 5262 5263// Yield samples from the timed buffer queue head up to the given output 5264// buffer's capacity. 5265// 5266// Caller must hold mTimedBufferQueueLock 5267void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5268 AudioBufferProvider::Buffer* buffer) { 5269 5270 const TimedBuffer& head = mTimedBufferQueue[0]; 5271 5272 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5273 head.position()); 5274 5275 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5276 mCblk->frameSize); 5277 size_t framesRequested = buffer->frameCount; 5278 buffer->frameCount = min(framesLeftInHead, framesRequested); 5279 5280 mQueueHeadInFlight = true; 5281 mTimedAudioOutputOnTime = true; 5282} 5283 5284// Yield samples of silence up to the given output buffer's capacity 5285// 5286// Caller must hold mTimedBufferQueueLock 5287void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5288 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5289 5290 // lazily allocate a buffer filled with silence 5291 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5292 delete [] mTimedSilenceBuffer; 5293 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5294 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5295 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5296 } 5297 5298 buffer->raw = mTimedSilenceBuffer; 5299 size_t framesRequested = buffer->frameCount; 5300 buffer->frameCount = min(numFrames, framesRequested); 5301 5302 mTimedAudioOutputOnTime = false; 5303} 5304 5305// AudioBufferProvider interface 5306void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5307 AudioBufferProvider::Buffer* buffer) { 5308 5309 Mutex::Autolock _l(mTimedBufferQueueLock); 5310 5311 // If the buffer which was just released is part of the buffer at the head 5312 // of the queue, be sure to update the amt of the buffer which has been 5313 // consumed. If the buffer being returned is not part of the head of the 5314 // queue, its either because the buffer is part of the silence buffer, or 5315 // because the head of the timed queue was trimmed after the mixer called 5316 // getNextBuffer but before the mixer called releaseBuffer. 5317 if (buffer->raw == mTimedSilenceBuffer) { 5318 ALOG_ASSERT(!mQueueHeadInFlight, 5319 "Queue head in flight during release of silence buffer!"); 5320 goto done; 5321 } 5322 5323 ALOG_ASSERT(mQueueHeadInFlight, 5324 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5325 " head in flight."); 5326 5327 if (mTimedBufferQueue.size()) { 5328 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5329 5330 void* start = head.buffer()->pointer(); 5331 void* end = reinterpret_cast<void*>( 5332 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5333 + head.buffer()->size()); 5334 5335 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5336 "released buffer not within the head of the timed buffer" 5337 " queue; qHead = [%p, %p], released buffer = %p", 5338 start, end, buffer->raw); 5339 5340 head.setPosition(head.position() + 5341 (buffer->frameCount * mCblk->frameSize)); 5342 mQueueHeadInFlight = false; 5343 5344 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5345 "Bad bookkeeping during releaseBuffer! Should have at" 5346 " least %u queued frames, but we think we have only %u", 5347 buffer->frameCount, mFramesPendingInQueue); 5348 5349 mFramesPendingInQueue -= buffer->frameCount; 5350 5351 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5352 || mTrimQueueHeadOnRelease) { 5353 trimTimedBufferQueueHead_l("releaseBuffer"); 5354 mTrimQueueHeadOnRelease = false; 5355 } 5356 } else { 5357 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5358 " buffers in the timed buffer queue"); 5359 } 5360 5361done: 5362 buffer->raw = 0; 5363 buffer->frameCount = 0; 5364} 5365 5366size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5367 Mutex::Autolock _l(mTimedBufferQueueLock); 5368 return mFramesPendingInQueue; 5369} 5370 5371AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5372 : mPTS(0), mPosition(0) {} 5373 5374AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5375 const sp<IMemory>& buffer, int64_t pts) 5376 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5377 5378// ---------------------------------------------------------------------------- 5379 5380// RecordTrack constructor must be called with AudioFlinger::mLock held 5381AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5382 RecordThread *thread, 5383 const sp<Client>& client, 5384 uint32_t sampleRate, 5385 audio_format_t format, 5386 audio_channel_mask_t channelMask, 5387 int frameCount, 5388 int sessionId) 5389 : TrackBase(thread, client, sampleRate, format, 5390 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5391 mOverflow(false) 5392{ 5393 if (mCblk != NULL) { 5394 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5395 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5396 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5397 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5398 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5399 } else { 5400 mCblk->frameSize = sizeof(int8_t); 5401 } 5402 } 5403} 5404 5405AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5406{ 5407 ALOGV("%s", __func__); 5408} 5409 5410// AudioBufferProvider interface 5411status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5412{ 5413 audio_track_cblk_t* cblk = this->cblk(); 5414 uint32_t framesAvail; 5415 uint32_t framesReq = buffer->frameCount; 5416 5417 // Check if last stepServer failed, try to step now 5418 if (mStepServerFailed) { 5419 if (!step()) goto getNextBuffer_exit; 5420 ALOGV("stepServer recovered"); 5421 mStepServerFailed = false; 5422 } 5423 5424 framesAvail = cblk->framesAvailable_l(); 5425 5426 if (CC_LIKELY(framesAvail)) { 5427 uint32_t s = cblk->server; 5428 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5429 5430 if (framesReq > framesAvail) { 5431 framesReq = framesAvail; 5432 } 5433 if (framesReq > bufferEnd - s) { 5434 framesReq = bufferEnd - s; 5435 } 5436 5437 buffer->raw = getBuffer(s, framesReq); 5438 buffer->frameCount = framesReq; 5439 return NO_ERROR; 5440 } 5441 5442getNextBuffer_exit: 5443 buffer->raw = NULL; 5444 buffer->frameCount = 0; 5445 return NOT_ENOUGH_DATA; 5446} 5447 5448status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5449 int triggerSession) 5450{ 5451 sp<ThreadBase> thread = mThread.promote(); 5452 if (thread != 0) { 5453 RecordThread *recordThread = (RecordThread *)thread.get(); 5454 return recordThread->start(this, event, triggerSession); 5455 } else { 5456 return BAD_VALUE; 5457 } 5458} 5459 5460void AudioFlinger::RecordThread::RecordTrack::stop() 5461{ 5462 sp<ThreadBase> thread = mThread.promote(); 5463 if (thread != 0) { 5464 RecordThread *recordThread = (RecordThread *)thread.get(); 5465 recordThread->mLock.lock(); 5466 bool doStop = recordThread->stop_l(this); 5467 if (doStop) { 5468 TrackBase::reset(); 5469 // Force overrun condition to avoid false overrun callback until first data is 5470 // read from buffer 5471 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5472 } 5473 recordThread->mLock.unlock(); 5474 if (doStop) { 5475 AudioSystem::stopInput(recordThread->id()); 5476 } 5477 } 5478} 5479 5480/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5481{ 5482 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User FrameCount\n"); 5483} 5484 5485void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5486{ 5487 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5488 (mClient == 0) ? getpid_cached : mClient->pid(), 5489 mFormat, 5490 mChannelMask, 5491 mSessionId, 5492 mFrameCount, 5493 mState, 5494 mCblk->sampleRate, 5495 mCblk->server, 5496 mCblk->user, 5497 mCblk->frameCount); 5498} 5499 5500 5501// ---------------------------------------------------------------------------- 5502 5503AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5504 PlaybackThread *playbackThread, 5505 DuplicatingThread *sourceThread, 5506 uint32_t sampleRate, 5507 audio_format_t format, 5508 audio_channel_mask_t channelMask, 5509 int frameCount) 5510 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5511 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5512 mActive(false), mSourceThread(sourceThread) 5513{ 5514 5515 if (mCblk != NULL) { 5516 mCblk->flags |= CBLK_DIRECTION_OUT; 5517 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5518 mOutBuffer.frameCount = 0; 5519 playbackThread->mTracks.add(this); 5520 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5521 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5522 mCblk, mBuffer, mCblk->buffers, 5523 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5524 } else { 5525 ALOGW("Error creating output track on thread %p", playbackThread); 5526 } 5527} 5528 5529AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5530{ 5531 clearBufferQueue(); 5532} 5533 5534status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5535 int triggerSession) 5536{ 5537 status_t status = Track::start(event, triggerSession); 5538 if (status != NO_ERROR) { 5539 return status; 5540 } 5541 5542 mActive = true; 5543 mRetryCount = 127; 5544 return status; 5545} 5546 5547void AudioFlinger::PlaybackThread::OutputTrack::stop() 5548{ 5549 Track::stop(); 5550 clearBufferQueue(); 5551 mOutBuffer.frameCount = 0; 5552 mActive = false; 5553} 5554 5555bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5556{ 5557 Buffer *pInBuffer; 5558 Buffer inBuffer; 5559 uint32_t channelCount = mChannelCount; 5560 bool outputBufferFull = false; 5561 inBuffer.frameCount = frames; 5562 inBuffer.i16 = data; 5563 5564 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5565 5566 if (!mActive && frames != 0) { 5567 start(); 5568 sp<ThreadBase> thread = mThread.promote(); 5569 if (thread != 0) { 5570 MixerThread *mixerThread = (MixerThread *)thread.get(); 5571 if (mCblk->frameCount > frames){ 5572 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5573 uint32_t startFrames = (mCblk->frameCount - frames); 5574 pInBuffer = new Buffer; 5575 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5576 pInBuffer->frameCount = startFrames; 5577 pInBuffer->i16 = pInBuffer->mBuffer; 5578 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5579 mBufferQueue.add(pInBuffer); 5580 } else { 5581 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5582 } 5583 } 5584 } 5585 } 5586 5587 while (waitTimeLeftMs) { 5588 // First write pending buffers, then new data 5589 if (mBufferQueue.size()) { 5590 pInBuffer = mBufferQueue.itemAt(0); 5591 } else { 5592 pInBuffer = &inBuffer; 5593 } 5594 5595 if (pInBuffer->frameCount == 0) { 5596 break; 5597 } 5598 5599 if (mOutBuffer.frameCount == 0) { 5600 mOutBuffer.frameCount = pInBuffer->frameCount; 5601 nsecs_t startTime = systemTime(); 5602 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5603 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5604 outputBufferFull = true; 5605 break; 5606 } 5607 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5608 if (waitTimeLeftMs >= waitTimeMs) { 5609 waitTimeLeftMs -= waitTimeMs; 5610 } else { 5611 waitTimeLeftMs = 0; 5612 } 5613 } 5614 5615 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5616 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5617 mCblk->stepUser(outFrames); 5618 pInBuffer->frameCount -= outFrames; 5619 pInBuffer->i16 += outFrames * channelCount; 5620 mOutBuffer.frameCount -= outFrames; 5621 mOutBuffer.i16 += outFrames * channelCount; 5622 5623 if (pInBuffer->frameCount == 0) { 5624 if (mBufferQueue.size()) { 5625 mBufferQueue.removeAt(0); 5626 delete [] pInBuffer->mBuffer; 5627 delete pInBuffer; 5628 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5629 } else { 5630 break; 5631 } 5632 } 5633 } 5634 5635 // If we could not write all frames, allocate a buffer and queue it for next time. 5636 if (inBuffer.frameCount) { 5637 sp<ThreadBase> thread = mThread.promote(); 5638 if (thread != 0 && !thread->standby()) { 5639 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5640 pInBuffer = new Buffer; 5641 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5642 pInBuffer->frameCount = inBuffer.frameCount; 5643 pInBuffer->i16 = pInBuffer->mBuffer; 5644 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5645 mBufferQueue.add(pInBuffer); 5646 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5647 } else { 5648 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5649 } 5650 } 5651 } 5652 5653 // Calling write() with a 0 length buffer, means that no more data will be written: 5654 // If no more buffers are pending, fill output track buffer to make sure it is started 5655 // by output mixer. 5656 if (frames == 0 && mBufferQueue.size() == 0) { 5657 if (mCblk->user < mCblk->frameCount) { 5658 frames = mCblk->frameCount - mCblk->user; 5659 pInBuffer = new Buffer; 5660 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5661 pInBuffer->frameCount = frames; 5662 pInBuffer->i16 = pInBuffer->mBuffer; 5663 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5664 mBufferQueue.add(pInBuffer); 5665 } else if (mActive) { 5666 stop(); 5667 } 5668 } 5669 5670 return outputBufferFull; 5671} 5672 5673status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5674{ 5675 int active; 5676 status_t result; 5677 audio_track_cblk_t* cblk = mCblk; 5678 uint32_t framesReq = buffer->frameCount; 5679 5680 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5681 buffer->frameCount = 0; 5682 5683 uint32_t framesAvail = cblk->framesAvailable(); 5684 5685 5686 if (framesAvail == 0) { 5687 Mutex::Autolock _l(cblk->lock); 5688 goto start_loop_here; 5689 while (framesAvail == 0) { 5690 active = mActive; 5691 if (CC_UNLIKELY(!active)) { 5692 ALOGV("Not active and NO_MORE_BUFFERS"); 5693 return NO_MORE_BUFFERS; 5694 } 5695 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5696 if (result != NO_ERROR) { 5697 return NO_MORE_BUFFERS; 5698 } 5699 // read the server count again 5700 start_loop_here: 5701 framesAvail = cblk->framesAvailable_l(); 5702 } 5703 } 5704 5705// if (framesAvail < framesReq) { 5706// return NO_MORE_BUFFERS; 5707// } 5708 5709 if (framesReq > framesAvail) { 5710 framesReq = framesAvail; 5711 } 5712 5713 uint32_t u = cblk->user; 5714 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5715 5716 if (framesReq > bufferEnd - u) { 5717 framesReq = bufferEnd - u; 5718 } 5719 5720 buffer->frameCount = framesReq; 5721 buffer->raw = (void *)cblk->buffer(u); 5722 return NO_ERROR; 5723} 5724 5725 5726void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5727{ 5728 size_t size = mBufferQueue.size(); 5729 5730 for (size_t i = 0; i < size; i++) { 5731 Buffer *pBuffer = mBufferQueue.itemAt(i); 5732 delete [] pBuffer->mBuffer; 5733 delete pBuffer; 5734 } 5735 mBufferQueue.clear(); 5736} 5737 5738// ---------------------------------------------------------------------------- 5739 5740AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5741 : RefBase(), 5742 mAudioFlinger(audioFlinger), 5743 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5744 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5745 mPid(pid), 5746 mTimedTrackCount(0) 5747{ 5748 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5749} 5750 5751// Client destructor must be called with AudioFlinger::mLock held 5752AudioFlinger::Client::~Client() 5753{ 5754 mAudioFlinger->removeClient_l(mPid); 5755} 5756 5757sp<MemoryDealer> AudioFlinger::Client::heap() const 5758{ 5759 return mMemoryDealer; 5760} 5761 5762// Reserve one of the limited slots for a timed audio track associated 5763// with this client 5764bool AudioFlinger::Client::reserveTimedTrack() 5765{ 5766 const int kMaxTimedTracksPerClient = 4; 5767 5768 Mutex::Autolock _l(mTimedTrackLock); 5769 5770 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5771 ALOGW("can not create timed track - pid %d has exceeded the limit", 5772 mPid); 5773 return false; 5774 } 5775 5776 mTimedTrackCount++; 5777 return true; 5778} 5779 5780// Release a slot for a timed audio track 5781void AudioFlinger::Client::releaseTimedTrack() 5782{ 5783 Mutex::Autolock _l(mTimedTrackLock); 5784 mTimedTrackCount--; 5785} 5786 5787// ---------------------------------------------------------------------------- 5788 5789AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5790 const sp<IAudioFlingerClient>& client, 5791 pid_t pid) 5792 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5793{ 5794} 5795 5796AudioFlinger::NotificationClient::~NotificationClient() 5797{ 5798} 5799 5800void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5801{ 5802 sp<NotificationClient> keep(this); 5803 mAudioFlinger->removeNotificationClient(mPid); 5804} 5805 5806// ---------------------------------------------------------------------------- 5807 5808AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5809 : BnAudioTrack(), 5810 mTrack(track) 5811{ 5812} 5813 5814AudioFlinger::TrackHandle::~TrackHandle() { 5815 // just stop the track on deletion, associated resources 5816 // will be freed from the main thread once all pending buffers have 5817 // been played. Unless it's not in the active track list, in which 5818 // case we free everything now... 5819 mTrack->destroy(); 5820} 5821 5822sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5823 return mTrack->getCblk(); 5824} 5825 5826status_t AudioFlinger::TrackHandle::start() { 5827 return mTrack->start(); 5828} 5829 5830void AudioFlinger::TrackHandle::stop() { 5831 mTrack->stop(); 5832} 5833 5834void AudioFlinger::TrackHandle::flush() { 5835 mTrack->flush(); 5836} 5837 5838void AudioFlinger::TrackHandle::mute(bool e) { 5839 mTrack->mute(e); 5840} 5841 5842void AudioFlinger::TrackHandle::pause() { 5843 mTrack->pause(); 5844} 5845 5846status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5847{ 5848 return mTrack->attachAuxEffect(EffectId); 5849} 5850 5851status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5852 sp<IMemory>* buffer) { 5853 if (!mTrack->isTimedTrack()) 5854 return INVALID_OPERATION; 5855 5856 PlaybackThread::TimedTrack* tt = 5857 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5858 return tt->allocateTimedBuffer(size, buffer); 5859} 5860 5861status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5862 int64_t pts) { 5863 if (!mTrack->isTimedTrack()) 5864 return INVALID_OPERATION; 5865 5866 PlaybackThread::TimedTrack* tt = 5867 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5868 return tt->queueTimedBuffer(buffer, pts); 5869} 5870 5871status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5872 const LinearTransform& xform, int target) { 5873 5874 if (!mTrack->isTimedTrack()) 5875 return INVALID_OPERATION; 5876 5877 PlaybackThread::TimedTrack* tt = 5878 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5879 return tt->setMediaTimeTransform( 5880 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5881} 5882 5883status_t AudioFlinger::TrackHandle::onTransact( 5884 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5885{ 5886 return BnAudioTrack::onTransact(code, data, reply, flags); 5887} 5888 5889// ---------------------------------------------------------------------------- 5890 5891sp<IAudioRecord> AudioFlinger::openRecord( 5892 pid_t pid, 5893 audio_io_handle_t input, 5894 uint32_t sampleRate, 5895 audio_format_t format, 5896 audio_channel_mask_t channelMask, 5897 int frameCount, 5898 IAudioFlinger::track_flags_t flags, 5899 pid_t tid, 5900 int *sessionId, 5901 status_t *status) 5902{ 5903 sp<RecordThread::RecordTrack> recordTrack; 5904 sp<RecordHandle> recordHandle; 5905 sp<Client> client; 5906 status_t lStatus; 5907 RecordThread *thread; 5908 size_t inFrameCount; 5909 int lSessionId; 5910 5911 // check calling permissions 5912 if (!recordingAllowed()) { 5913 lStatus = PERMISSION_DENIED; 5914 goto Exit; 5915 } 5916 5917 // add client to list 5918 { // scope for mLock 5919 Mutex::Autolock _l(mLock); 5920 thread = checkRecordThread_l(input); 5921 if (thread == NULL) { 5922 lStatus = BAD_VALUE; 5923 goto Exit; 5924 } 5925 5926 client = registerPid_l(pid); 5927 5928 // If no audio session id is provided, create one here 5929 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5930 lSessionId = *sessionId; 5931 } else { 5932 lSessionId = nextUniqueId(); 5933 if (sessionId != NULL) { 5934 *sessionId = lSessionId; 5935 } 5936 } 5937 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5938 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5939 frameCount, lSessionId, flags, tid, &lStatus); 5940 } 5941 if (lStatus != NO_ERROR) { 5942 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5943 // destructor is called by the TrackBase destructor with mLock held 5944 client.clear(); 5945 recordTrack.clear(); 5946 goto Exit; 5947 } 5948 5949 // return to handle to client 5950 recordHandle = new RecordHandle(recordTrack); 5951 lStatus = NO_ERROR; 5952 5953Exit: 5954 if (status) { 5955 *status = lStatus; 5956 } 5957 return recordHandle; 5958} 5959 5960// ---------------------------------------------------------------------------- 5961 5962AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5963 : BnAudioRecord(), 5964 mRecordTrack(recordTrack) 5965{ 5966} 5967 5968AudioFlinger::RecordHandle::~RecordHandle() { 5969 stop_nonvirtual(); 5970 mRecordTrack->destroy(); 5971} 5972 5973sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5974 return mRecordTrack->getCblk(); 5975} 5976 5977status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5978 ALOGV("RecordHandle::start()"); 5979 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5980} 5981 5982void AudioFlinger::RecordHandle::stop() { 5983 stop_nonvirtual(); 5984} 5985 5986void AudioFlinger::RecordHandle::stop_nonvirtual() { 5987 ALOGV("RecordHandle::stop()"); 5988 mRecordTrack->stop(); 5989} 5990 5991status_t AudioFlinger::RecordHandle::onTransact( 5992 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5993{ 5994 return BnAudioRecord::onTransact(code, data, reply, flags); 5995} 5996 5997// ---------------------------------------------------------------------------- 5998 5999AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6000 AudioStreamIn *input, 6001 uint32_t sampleRate, 6002 audio_channel_mask_t channelMask, 6003 audio_io_handle_t id, 6004 audio_devices_t device, 6005 const sp<NBAIO_Sink>& teeSink) : 6006 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6007 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6008 // mRsmpInIndex and mInputBytes set by readInputParameters() 6009 mReqChannelCount(popcount(channelMask)), 6010 mReqSampleRate(sampleRate), 6011 // mBytesRead is only meaningful while active, and so is cleared in start() 6012 // (but might be better to also clear here for dump?) 6013 mTeeSink(teeSink) 6014{ 6015 snprintf(mName, kNameLength, "AudioIn_%X", id); 6016 6017 readInputParameters(); 6018 6019} 6020 6021 6022AudioFlinger::RecordThread::~RecordThread() 6023{ 6024 delete[] mRsmpInBuffer; 6025 delete mResampler; 6026 delete[] mRsmpOutBuffer; 6027} 6028 6029void AudioFlinger::RecordThread::onFirstRef() 6030{ 6031 run(mName, PRIORITY_URGENT_AUDIO); 6032} 6033 6034status_t AudioFlinger::RecordThread::readyToRun() 6035{ 6036 status_t status = initCheck(); 6037 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6038 return status; 6039} 6040 6041bool AudioFlinger::RecordThread::threadLoop() 6042{ 6043 AudioBufferProvider::Buffer buffer; 6044 sp<RecordTrack> activeTrack; 6045 Vector< sp<EffectChain> > effectChains; 6046 6047 nsecs_t lastWarning = 0; 6048 6049 inputStandBy(); 6050 acquireWakeLock(); 6051 6052 // used to verify we've read at least once before evaluating how many bytes were read 6053 bool readOnce = false; 6054 6055 // start recording 6056 while (!exitPending()) { 6057 6058 processConfigEvents(); 6059 6060 { // scope for mLock 6061 Mutex::Autolock _l(mLock); 6062 checkForNewParameters_l(); 6063 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6064 standby(); 6065 6066 if (exitPending()) break; 6067 6068 releaseWakeLock_l(); 6069 ALOGV("RecordThread: loop stopping"); 6070 // go to sleep 6071 mWaitWorkCV.wait(mLock); 6072 ALOGV("RecordThread: loop starting"); 6073 acquireWakeLock_l(); 6074 continue; 6075 } 6076 if (mActiveTrack != 0) { 6077 if (mActiveTrack->mState == TrackBase::PAUSING) { 6078 standby(); 6079 mActiveTrack.clear(); 6080 mStartStopCond.broadcast(); 6081 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6082 if (mReqChannelCount != mActiveTrack->channelCount()) { 6083 mActiveTrack.clear(); 6084 mStartStopCond.broadcast(); 6085 } else if (readOnce) { 6086 // record start succeeds only if first read from audio input 6087 // succeeds 6088 if (mBytesRead >= 0) { 6089 mActiveTrack->mState = TrackBase::ACTIVE; 6090 } else { 6091 mActiveTrack.clear(); 6092 } 6093 mStartStopCond.broadcast(); 6094 } 6095 mStandby = false; 6096 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6097 removeTrack_l(mActiveTrack); 6098 mActiveTrack.clear(); 6099 } 6100 } 6101 lockEffectChains_l(effectChains); 6102 } 6103 6104 if (mActiveTrack != 0) { 6105 if (mActiveTrack->mState != TrackBase::ACTIVE && 6106 mActiveTrack->mState != TrackBase::RESUMING) { 6107 unlockEffectChains(effectChains); 6108 usleep(kRecordThreadSleepUs); 6109 continue; 6110 } 6111 for (size_t i = 0; i < effectChains.size(); i ++) { 6112 effectChains[i]->process_l(); 6113 } 6114 6115 buffer.frameCount = mFrameCount; 6116 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6117 readOnce = true; 6118 size_t framesOut = buffer.frameCount; 6119 if (mResampler == NULL) { 6120 // no resampling 6121 while (framesOut) { 6122 size_t framesIn = mFrameCount - mRsmpInIndex; 6123 if (framesIn) { 6124 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6125 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6126 if (framesIn > framesOut) 6127 framesIn = framesOut; 6128 mRsmpInIndex += framesIn; 6129 framesOut -= framesIn; 6130 if ((int)mChannelCount == mReqChannelCount || 6131 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6132 memcpy(dst, src, framesIn * mFrameSize); 6133 } else { 6134 if (mChannelCount == 1) { 6135 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6136 (int16_t *)src, framesIn); 6137 } else { 6138 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6139 (int16_t *)src, framesIn); 6140 } 6141 } 6142 } 6143 if (framesOut && mFrameCount == mRsmpInIndex) { 6144 void *readInto; 6145 if (framesOut == mFrameCount && 6146 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6147 readInto = buffer.raw; 6148 framesOut = 0; 6149 } else { 6150 readInto = mRsmpInBuffer; 6151 mRsmpInIndex = 0; 6152 } 6153 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6154 if (mBytesRead <= 0) { 6155 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6156 { 6157 ALOGE("Error reading audio input"); 6158 // Force input into standby so that it tries to 6159 // recover at next read attempt 6160 inputStandBy(); 6161 usleep(kRecordThreadSleepUs); 6162 } 6163 mRsmpInIndex = mFrameCount; 6164 framesOut = 0; 6165 buffer.frameCount = 0; 6166 } else if (mTeeSink != 0) { 6167 (void) mTeeSink->write(readInto, 6168 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6169 } 6170 } 6171 } 6172 } else { 6173 // resampling 6174 6175 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6176 // alter output frame count as if we were expecting stereo samples 6177 if (mChannelCount == 1 && mReqChannelCount == 1) { 6178 framesOut >>= 1; 6179 } 6180 mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); 6181 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6182 // are 32 bit aligned which should be always true. 6183 if (mChannelCount == 2 && mReqChannelCount == 1) { 6184 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6185 // the resampler always outputs stereo samples: do post stereo to mono conversion 6186 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6187 framesOut); 6188 } else { 6189 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6190 } 6191 6192 } 6193 if (mFramestoDrop == 0) { 6194 mActiveTrack->releaseBuffer(&buffer); 6195 } else { 6196 if (mFramestoDrop > 0) { 6197 mFramestoDrop -= buffer.frameCount; 6198 if (mFramestoDrop <= 0) { 6199 clearSyncStartEvent(); 6200 } 6201 } else { 6202 mFramestoDrop += buffer.frameCount; 6203 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6204 mSyncStartEvent->isCancelled()) { 6205 ALOGW("Synced record %s, session %d, trigger session %d", 6206 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6207 mActiveTrack->sessionId(), 6208 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6209 clearSyncStartEvent(); 6210 } 6211 } 6212 } 6213 mActiveTrack->clearOverflow(); 6214 } 6215 // client isn't retrieving buffers fast enough 6216 else { 6217 if (!mActiveTrack->setOverflow()) { 6218 nsecs_t now = systemTime(); 6219 if ((now - lastWarning) > kWarningThrottleNs) { 6220 ALOGW("RecordThread: buffer overflow"); 6221 lastWarning = now; 6222 } 6223 } 6224 // Release the processor for a while before asking for a new buffer. 6225 // This will give the application more chance to read from the buffer and 6226 // clear the overflow. 6227 usleep(kRecordThreadSleepUs); 6228 } 6229 } 6230 // enable changes in effect chain 6231 unlockEffectChains(effectChains); 6232 effectChains.clear(); 6233 } 6234 6235 standby(); 6236 6237 { 6238 Mutex::Autolock _l(mLock); 6239 mActiveTrack.clear(); 6240 mStartStopCond.broadcast(); 6241 } 6242 6243 releaseWakeLock(); 6244 6245 ALOGV("RecordThread %p exiting", this); 6246 return false; 6247} 6248 6249void AudioFlinger::RecordThread::standby() 6250{ 6251 if (!mStandby) { 6252 inputStandBy(); 6253 mStandby = true; 6254 } 6255} 6256 6257void AudioFlinger::RecordThread::inputStandBy() 6258{ 6259 mInput->stream->common.standby(&mInput->stream->common); 6260} 6261 6262sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6263 const sp<AudioFlinger::Client>& client, 6264 uint32_t sampleRate, 6265 audio_format_t format, 6266 audio_channel_mask_t channelMask, 6267 int frameCount, 6268 int sessionId, 6269 IAudioFlinger::track_flags_t flags, 6270 pid_t tid, 6271 status_t *status) 6272{ 6273 sp<RecordTrack> track; 6274 status_t lStatus; 6275 6276 lStatus = initCheck(); 6277 if (lStatus != NO_ERROR) { 6278 ALOGE("Audio driver not initialized."); 6279 goto Exit; 6280 } 6281 6282 // FIXME use flags and tid similar to createTrack_l() 6283 6284 { // scope for mLock 6285 Mutex::Autolock _l(mLock); 6286 6287 track = new RecordTrack(this, client, sampleRate, 6288 format, channelMask, frameCount, sessionId); 6289 6290 if (track->getCblk() == 0) { 6291 lStatus = NO_MEMORY; 6292 goto Exit; 6293 } 6294 mTracks.add(track); 6295 6296 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6297 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6298 mAudioFlinger->btNrecIsOff(); 6299 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6300 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6301 } 6302 lStatus = NO_ERROR; 6303 6304Exit: 6305 if (status) { 6306 *status = lStatus; 6307 } 6308 return track; 6309} 6310 6311status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6312 AudioSystem::sync_event_t event, 6313 int triggerSession) 6314{ 6315 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6316 sp<ThreadBase> strongMe = this; 6317 status_t status = NO_ERROR; 6318 6319 if (event == AudioSystem::SYNC_EVENT_NONE) { 6320 clearSyncStartEvent(); 6321 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6322 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6323 triggerSession, 6324 recordTrack->sessionId(), 6325 syncStartEventCallback, 6326 this); 6327 // Sync event can be cancelled by the trigger session if the track is not in a 6328 // compatible state in which case we start record immediately 6329 if (mSyncStartEvent->isCancelled()) { 6330 clearSyncStartEvent(); 6331 } else { 6332 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6333 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6334 } 6335 } 6336 6337 { 6338 AutoMutex lock(mLock); 6339 if (mActiveTrack != 0) { 6340 if (recordTrack != mActiveTrack.get()) { 6341 status = -EBUSY; 6342 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6343 mActiveTrack->mState = TrackBase::ACTIVE; 6344 } 6345 return status; 6346 } 6347 6348 recordTrack->mState = TrackBase::IDLE; 6349 mActiveTrack = recordTrack; 6350 mLock.unlock(); 6351 status_t status = AudioSystem::startInput(mId); 6352 mLock.lock(); 6353 if (status != NO_ERROR) { 6354 mActiveTrack.clear(); 6355 clearSyncStartEvent(); 6356 return status; 6357 } 6358 mRsmpInIndex = mFrameCount; 6359 mBytesRead = 0; 6360 if (mResampler != NULL) { 6361 mResampler->reset(); 6362 } 6363 mActiveTrack->mState = TrackBase::RESUMING; 6364 // signal thread to start 6365 ALOGV("Signal record thread"); 6366 mWaitWorkCV.broadcast(); 6367 // do not wait for mStartStopCond if exiting 6368 if (exitPending()) { 6369 mActiveTrack.clear(); 6370 status = INVALID_OPERATION; 6371 goto startError; 6372 } 6373 mStartStopCond.wait(mLock); 6374 if (mActiveTrack == 0) { 6375 ALOGV("Record failed to start"); 6376 status = BAD_VALUE; 6377 goto startError; 6378 } 6379 ALOGV("Record started OK"); 6380 return status; 6381 } 6382startError: 6383 AudioSystem::stopInput(mId); 6384 clearSyncStartEvent(); 6385 return status; 6386} 6387 6388void AudioFlinger::RecordThread::clearSyncStartEvent() 6389{ 6390 if (mSyncStartEvent != 0) { 6391 mSyncStartEvent->cancel(); 6392 } 6393 mSyncStartEvent.clear(); 6394 mFramestoDrop = 0; 6395} 6396 6397void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6398{ 6399 sp<SyncEvent> strongEvent = event.promote(); 6400 6401 if (strongEvent != 0) { 6402 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6403 me->handleSyncStartEvent(strongEvent); 6404 } 6405} 6406 6407void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6408{ 6409 if (event == mSyncStartEvent) { 6410 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6411 // from audio HAL 6412 mFramestoDrop = mFrameCount * 2; 6413 } 6414} 6415 6416bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6417 ALOGV("RecordThread::stop"); 6418 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6419 return false; 6420 } 6421 recordTrack->mState = TrackBase::PAUSING; 6422 // do not wait for mStartStopCond if exiting 6423 if (exitPending()) { 6424 return true; 6425 } 6426 mStartStopCond.wait(mLock); 6427 // if we have been restarted, recordTrack == mActiveTrack.get() here 6428 if (exitPending() || recordTrack != mActiveTrack.get()) { 6429 ALOGV("Record stopped OK"); 6430 return true; 6431 } 6432 return false; 6433} 6434 6435bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6436{ 6437 return false; 6438} 6439 6440status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6441{ 6442#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6443 if (!isValidSyncEvent(event)) { 6444 return BAD_VALUE; 6445 } 6446 6447 int eventSession = event->triggerSession(); 6448 status_t ret = NAME_NOT_FOUND; 6449 6450 Mutex::Autolock _l(mLock); 6451 6452 for (size_t i = 0; i < mTracks.size(); i++) { 6453 sp<RecordTrack> track = mTracks[i]; 6454 if (eventSession == track->sessionId()) { 6455 (void) track->setSyncEvent(event); 6456 ret = NO_ERROR; 6457 } 6458 } 6459 return ret; 6460#else 6461 return BAD_VALUE; 6462#endif 6463} 6464 6465void AudioFlinger::RecordThread::RecordTrack::destroy() 6466{ 6467 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6468 sp<RecordTrack> keep(this); 6469 { 6470 sp<ThreadBase> thread = mThread.promote(); 6471 if (thread != 0) { 6472 if (mState == ACTIVE || mState == RESUMING) { 6473 AudioSystem::stopInput(thread->id()); 6474 } 6475 AudioSystem::releaseInput(thread->id()); 6476 Mutex::Autolock _l(thread->mLock); 6477 RecordThread *recordThread = (RecordThread *) thread.get(); 6478 recordThread->destroyTrack_l(this); 6479 } 6480 } 6481} 6482 6483// destroyTrack_l() must be called with ThreadBase::mLock held 6484void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6485{ 6486 track->mState = TrackBase::TERMINATED; 6487 // active tracks are removed by threadLoop() 6488 if (mActiveTrack != track) { 6489 removeTrack_l(track); 6490 } 6491} 6492 6493void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6494{ 6495 mTracks.remove(track); 6496 // need anything related to effects here? 6497} 6498 6499void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6500{ 6501 dumpInternals(fd, args); 6502 dumpTracks(fd, args); 6503 dumpEffectChains(fd, args); 6504} 6505 6506void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6507{ 6508 const size_t SIZE = 256; 6509 char buffer[SIZE]; 6510 String8 result; 6511 6512 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6513 result.append(buffer); 6514 6515 if (mActiveTrack != 0) { 6516 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6517 result.append(buffer); 6518 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6519 result.append(buffer); 6520 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6521 result.append(buffer); 6522 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6523 result.append(buffer); 6524 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6525 result.append(buffer); 6526 } else { 6527 result.append("No active record client\n"); 6528 } 6529 6530 write(fd, result.string(), result.size()); 6531 6532 dumpBase(fd, args); 6533} 6534 6535void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6536{ 6537 const size_t SIZE = 256; 6538 char buffer[SIZE]; 6539 String8 result; 6540 6541 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6542 result.append(buffer); 6543 RecordTrack::appendDumpHeader(result); 6544 for (size_t i = 0; i < mTracks.size(); ++i) { 6545 sp<RecordTrack> track = mTracks[i]; 6546 if (track != 0) { 6547 track->dump(buffer, SIZE); 6548 result.append(buffer); 6549 } 6550 } 6551 6552 if (mActiveTrack != 0) { 6553 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6554 result.append(buffer); 6555 RecordTrack::appendDumpHeader(result); 6556 mActiveTrack->dump(buffer, SIZE); 6557 result.append(buffer); 6558 6559 } 6560 write(fd, result.string(), result.size()); 6561} 6562 6563// AudioBufferProvider interface 6564status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6565{ 6566 size_t framesReq = buffer->frameCount; 6567 size_t framesReady = mFrameCount - mRsmpInIndex; 6568 int channelCount; 6569 6570 if (framesReady == 0) { 6571 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6572 if (mBytesRead <= 0) { 6573 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6574 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6575 // Force input into standby so that it tries to 6576 // recover at next read attempt 6577 inputStandBy(); 6578 usleep(kRecordThreadSleepUs); 6579 } 6580 buffer->raw = NULL; 6581 buffer->frameCount = 0; 6582 return NOT_ENOUGH_DATA; 6583 } 6584 mRsmpInIndex = 0; 6585 framesReady = mFrameCount; 6586 } 6587 6588 if (framesReq > framesReady) { 6589 framesReq = framesReady; 6590 } 6591 6592 if (mChannelCount == 1 && mReqChannelCount == 2) { 6593 channelCount = 1; 6594 } else { 6595 channelCount = 2; 6596 } 6597 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6598 buffer->frameCount = framesReq; 6599 return NO_ERROR; 6600} 6601 6602// AudioBufferProvider interface 6603void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6604{ 6605 mRsmpInIndex += buffer->frameCount; 6606 buffer->frameCount = 0; 6607} 6608 6609bool AudioFlinger::RecordThread::checkForNewParameters_l() 6610{ 6611 bool reconfig = false; 6612 6613 while (!mNewParameters.isEmpty()) { 6614 status_t status = NO_ERROR; 6615 String8 keyValuePair = mNewParameters[0]; 6616 AudioParameter param = AudioParameter(keyValuePair); 6617 int value; 6618 audio_format_t reqFormat = mFormat; 6619 int reqSamplingRate = mReqSampleRate; 6620 int reqChannelCount = mReqChannelCount; 6621 6622 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6623 reqSamplingRate = value; 6624 reconfig = true; 6625 } 6626 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6627 reqFormat = (audio_format_t) value; 6628 reconfig = true; 6629 } 6630 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6631 reqChannelCount = popcount(value); 6632 reconfig = true; 6633 } 6634 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6635 // do not accept frame count changes if tracks are open as the track buffer 6636 // size depends on frame count and correct behavior would not be guaranteed 6637 // if frame count is changed after track creation 6638 if (mActiveTrack != 0) { 6639 status = INVALID_OPERATION; 6640 } else { 6641 reconfig = true; 6642 } 6643 } 6644 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6645 // forward device change to effects that have requested to be 6646 // aware of attached audio device. 6647 for (size_t i = 0; i < mEffectChains.size(); i++) { 6648 mEffectChains[i]->setDevice_l(value); 6649 } 6650 6651 // store input device and output device but do not forward output device to audio HAL. 6652 // Note that status is ignored by the caller for output device 6653 // (see AudioFlinger::setParameters() 6654 if (audio_is_output_devices(value)) { 6655 mOutDevice = value; 6656 status = BAD_VALUE; 6657 } else { 6658 mInDevice = value; 6659 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6660 if (mTracks.size() > 0) { 6661 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6662 mAudioFlinger->btNrecIsOff(); 6663 for (size_t i = 0; i < mTracks.size(); i++) { 6664 sp<RecordTrack> track = mTracks[i]; 6665 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6666 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6667 } 6668 } 6669 } 6670 } 6671 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6672 mAudioSource != (audio_source_t)value) { 6673 // forward device change to effects that have requested to be 6674 // aware of attached audio device. 6675 for (size_t i = 0; i < mEffectChains.size(); i++) { 6676 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6677 } 6678 mAudioSource = (audio_source_t)value; 6679 } 6680 if (status == NO_ERROR) { 6681 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6682 if (status == INVALID_OPERATION) { 6683 inputStandBy(); 6684 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6685 keyValuePair.string()); 6686 } 6687 if (reconfig) { 6688 if (status == BAD_VALUE && 6689 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6690 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6691 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6692 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6693 (reqChannelCount <= FCC_2)) { 6694 status = NO_ERROR; 6695 } 6696 if (status == NO_ERROR) { 6697 readInputParameters(); 6698 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6699 } 6700 } 6701 } 6702 6703 mNewParameters.removeAt(0); 6704 6705 mParamStatus = status; 6706 mParamCond.signal(); 6707 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6708 // already timed out waiting for the status and will never signal the condition. 6709 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6710 } 6711 return reconfig; 6712} 6713 6714String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6715{ 6716 char *s; 6717 String8 out_s8 = String8(); 6718 6719 Mutex::Autolock _l(mLock); 6720 if (initCheck() != NO_ERROR) { 6721 return out_s8; 6722 } 6723 6724 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6725 out_s8 = String8(s); 6726 free(s); 6727 return out_s8; 6728} 6729 6730void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6731 AudioSystem::OutputDescriptor desc; 6732 void *param2 = NULL; 6733 6734 switch (event) { 6735 case AudioSystem::INPUT_OPENED: 6736 case AudioSystem::INPUT_CONFIG_CHANGED: 6737 desc.channels = mChannelMask; 6738 desc.samplingRate = mSampleRate; 6739 desc.format = mFormat; 6740 desc.frameCount = mFrameCount; 6741 desc.latency = 0; 6742 param2 = &desc; 6743 break; 6744 6745 case AudioSystem::INPUT_CLOSED: 6746 default: 6747 break; 6748 } 6749 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6750} 6751 6752void AudioFlinger::RecordThread::readInputParameters() 6753{ 6754 delete mRsmpInBuffer; 6755 // mRsmpInBuffer is always assigned a new[] below 6756 delete mRsmpOutBuffer; 6757 mRsmpOutBuffer = NULL; 6758 delete mResampler; 6759 mResampler = NULL; 6760 6761 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6762 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6763 mChannelCount = (uint16_t)popcount(mChannelMask); 6764 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6765 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6766 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6767 mFrameCount = mInputBytes / mFrameSize; 6768 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6769 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6770 6771 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6772 { 6773 int channelCount; 6774 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6775 // stereo to mono post process as the resampler always outputs stereo. 6776 if (mChannelCount == 1 && mReqChannelCount == 2) { 6777 channelCount = 1; 6778 } else { 6779 channelCount = 2; 6780 } 6781 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6782 mResampler->setSampleRate(mSampleRate); 6783 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6784 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6785 6786 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6787 if (mChannelCount == 1 && mReqChannelCount == 1) { 6788 mFrameCount >>= 1; 6789 } 6790 6791 } 6792 mRsmpInIndex = mFrameCount; 6793} 6794 6795unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6796{ 6797 Mutex::Autolock _l(mLock); 6798 if (initCheck() != NO_ERROR) { 6799 return 0; 6800 } 6801 6802 return mInput->stream->get_input_frames_lost(mInput->stream); 6803} 6804 6805uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6806{ 6807 Mutex::Autolock _l(mLock); 6808 uint32_t result = 0; 6809 if (getEffectChain_l(sessionId) != 0) { 6810 result = EFFECT_SESSION; 6811 } 6812 6813 for (size_t i = 0; i < mTracks.size(); ++i) { 6814 if (sessionId == mTracks[i]->sessionId()) { 6815 result |= TRACK_SESSION; 6816 break; 6817 } 6818 } 6819 6820 return result; 6821} 6822 6823KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6824{ 6825 KeyedVector<int, bool> ids; 6826 Mutex::Autolock _l(mLock); 6827 for (size_t j = 0; j < mTracks.size(); ++j) { 6828 sp<RecordThread::RecordTrack> track = mTracks[j]; 6829 int sessionId = track->sessionId(); 6830 if (ids.indexOfKey(sessionId) < 0) { 6831 ids.add(sessionId, true); 6832 } 6833 } 6834 return ids; 6835} 6836 6837AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6838{ 6839 Mutex::Autolock _l(mLock); 6840 AudioStreamIn *input = mInput; 6841 mInput = NULL; 6842 return input; 6843} 6844 6845// this method must always be called either with ThreadBase mLock held or inside the thread loop 6846audio_stream_t* AudioFlinger::RecordThread::stream() const 6847{ 6848 if (mInput == NULL) { 6849 return NULL; 6850 } 6851 return &mInput->stream->common; 6852} 6853 6854 6855// ---------------------------------------------------------------------------- 6856 6857audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6858{ 6859 if (!settingsAllowed()) { 6860 return 0; 6861 } 6862 Mutex::Autolock _l(mLock); 6863 return loadHwModule_l(name); 6864} 6865 6866// loadHwModule_l() must be called with AudioFlinger::mLock held 6867audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6868{ 6869 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6870 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6871 ALOGW("loadHwModule() module %s already loaded", name); 6872 return mAudioHwDevs.keyAt(i); 6873 } 6874 } 6875 6876 audio_hw_device_t *dev; 6877 6878 int rc = load_audio_interface(name, &dev); 6879 if (rc) { 6880 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6881 return 0; 6882 } 6883 6884 mHardwareStatus = AUDIO_HW_INIT; 6885 rc = dev->init_check(dev); 6886 mHardwareStatus = AUDIO_HW_IDLE; 6887 if (rc) { 6888 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6889 return 0; 6890 } 6891 6892 // Check and cache this HAL's level of support for master mute and master 6893 // volume. If this is the first HAL opened, and it supports the get 6894 // methods, use the initial values provided by the HAL as the current 6895 // master mute and volume settings. 6896 6897 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6898 { // scope for auto-lock pattern 6899 AutoMutex lock(mHardwareLock); 6900 6901 if (0 == mAudioHwDevs.size()) { 6902 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6903 if (NULL != dev->get_master_volume) { 6904 float mv; 6905 if (OK == dev->get_master_volume(dev, &mv)) { 6906 mMasterVolume = mv; 6907 } 6908 } 6909 6910 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6911 if (NULL != dev->get_master_mute) { 6912 bool mm; 6913 if (OK == dev->get_master_mute(dev, &mm)) { 6914 mMasterMute = mm; 6915 } 6916 } 6917 } 6918 6919 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6920 if ((NULL != dev->set_master_volume) && 6921 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6922 flags = static_cast<AudioHwDevice::Flags>(flags | 6923 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6924 } 6925 6926 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6927 if ((NULL != dev->set_master_mute) && 6928 (OK == dev->set_master_mute(dev, mMasterMute))) { 6929 flags = static_cast<AudioHwDevice::Flags>(flags | 6930 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6931 } 6932 6933 mHardwareStatus = AUDIO_HW_IDLE; 6934 } 6935 6936 audio_module_handle_t handle = nextUniqueId(); 6937 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6938 6939 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6940 name, dev->common.module->name, dev->common.module->id, handle); 6941 6942 return handle; 6943 6944} 6945 6946// ---------------------------------------------------------------------------- 6947 6948int32_t AudioFlinger::getPrimaryOutputSamplingRate() 6949{ 6950 Mutex::Autolock _l(mLock); 6951 PlaybackThread *thread = primaryPlaybackThread_l(); 6952 return thread != NULL ? thread->sampleRate() : 0; 6953} 6954 6955int32_t AudioFlinger::getPrimaryOutputFrameCount() 6956{ 6957 Mutex::Autolock _l(mLock); 6958 PlaybackThread *thread = primaryPlaybackThread_l(); 6959 return thread != NULL ? thread->frameCountHAL() : 0; 6960} 6961 6962// ---------------------------------------------------------------------------- 6963 6964audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6965 audio_devices_t *pDevices, 6966 uint32_t *pSamplingRate, 6967 audio_format_t *pFormat, 6968 audio_channel_mask_t *pChannelMask, 6969 uint32_t *pLatencyMs, 6970 audio_output_flags_t flags) 6971{ 6972 status_t status; 6973 PlaybackThread *thread = NULL; 6974 struct audio_config config = { 6975 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6976 channel_mask: pChannelMask ? *pChannelMask : 0, 6977 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6978 }; 6979 audio_stream_out_t *outStream = NULL; 6980 AudioHwDevice *outHwDev; 6981 6982 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6983 module, 6984 (pDevices != NULL) ? *pDevices : 0, 6985 config.sample_rate, 6986 config.format, 6987 config.channel_mask, 6988 flags); 6989 6990 if (pDevices == NULL || *pDevices == 0) { 6991 return 0; 6992 } 6993 6994 Mutex::Autolock _l(mLock); 6995 6996 outHwDev = findSuitableHwDev_l(module, *pDevices); 6997 if (outHwDev == NULL) 6998 return 0; 6999 7000 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7001 audio_io_handle_t id = nextUniqueId(); 7002 7003 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7004 7005 status = hwDevHal->open_output_stream(hwDevHal, 7006 id, 7007 *pDevices, 7008 (audio_output_flags_t)flags, 7009 &config, 7010 &outStream); 7011 7012 mHardwareStatus = AUDIO_HW_IDLE; 7013 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 7014 outStream, 7015 config.sample_rate, 7016 config.format, 7017 config.channel_mask, 7018 status); 7019 7020 if (status == NO_ERROR && outStream != NULL) { 7021 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7022 7023 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7024 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7025 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7026 thread = new DirectOutputThread(this, output, id, *pDevices); 7027 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7028 } else { 7029 thread = new MixerThread(this, output, id, *pDevices); 7030 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7031 } 7032 mPlaybackThreads.add(id, thread); 7033 7034 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7035 if (pFormat != NULL) *pFormat = config.format; 7036 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7037 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7038 7039 // notify client processes of the new output creation 7040 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7041 7042 // the first primary output opened designates the primary hw device 7043 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7044 ALOGI("Using module %d has the primary audio interface", module); 7045 mPrimaryHardwareDev = outHwDev; 7046 7047 AutoMutex lock(mHardwareLock); 7048 mHardwareStatus = AUDIO_HW_SET_MODE; 7049 hwDevHal->set_mode(hwDevHal, mMode); 7050 mHardwareStatus = AUDIO_HW_IDLE; 7051 } 7052 return id; 7053 } 7054 7055 return 0; 7056} 7057 7058audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7059 audio_io_handle_t output2) 7060{ 7061 Mutex::Autolock _l(mLock); 7062 MixerThread *thread1 = checkMixerThread_l(output1); 7063 MixerThread *thread2 = checkMixerThread_l(output2); 7064 7065 if (thread1 == NULL || thread2 == NULL) { 7066 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 7067 return 0; 7068 } 7069 7070 audio_io_handle_t id = nextUniqueId(); 7071 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7072 thread->addOutputTrack(thread2); 7073 mPlaybackThreads.add(id, thread); 7074 // notify client processes of the new output creation 7075 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7076 return id; 7077} 7078 7079status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7080{ 7081 return closeOutput_nonvirtual(output); 7082} 7083 7084status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7085{ 7086 // keep strong reference on the playback thread so that 7087 // it is not destroyed while exit() is executed 7088 sp<PlaybackThread> thread; 7089 { 7090 Mutex::Autolock _l(mLock); 7091 thread = checkPlaybackThread_l(output); 7092 if (thread == NULL) { 7093 return BAD_VALUE; 7094 } 7095 7096 ALOGV("closeOutput() %d", output); 7097 7098 if (thread->type() == ThreadBase::MIXER) { 7099 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7100 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7101 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7102 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7103 } 7104 } 7105 } 7106 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7107 mPlaybackThreads.removeItem(output); 7108 } 7109 thread->exit(); 7110 // The thread entity (active unit of execution) is no longer running here, 7111 // but the ThreadBase container still exists. 7112 7113 if (thread->type() != ThreadBase::DUPLICATING) { 7114 AudioStreamOut *out = thread->clearOutput(); 7115 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7116 // from now on thread->mOutput is NULL 7117 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7118 delete out; 7119 } 7120 return NO_ERROR; 7121} 7122 7123status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7124{ 7125 Mutex::Autolock _l(mLock); 7126 PlaybackThread *thread = checkPlaybackThread_l(output); 7127 7128 if (thread == NULL) { 7129 return BAD_VALUE; 7130 } 7131 7132 ALOGV("suspendOutput() %d", output); 7133 thread->suspend(); 7134 7135 return NO_ERROR; 7136} 7137 7138status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7139{ 7140 Mutex::Autolock _l(mLock); 7141 PlaybackThread *thread = checkPlaybackThread_l(output); 7142 7143 if (thread == NULL) { 7144 return BAD_VALUE; 7145 } 7146 7147 ALOGV("restoreOutput() %d", output); 7148 7149 thread->restore(); 7150 7151 return NO_ERROR; 7152} 7153 7154audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7155 audio_devices_t *pDevices, 7156 uint32_t *pSamplingRate, 7157 audio_format_t *pFormat, 7158 audio_channel_mask_t *pChannelMask) 7159{ 7160 status_t status; 7161 RecordThread *thread = NULL; 7162 struct audio_config config = { 7163 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7164 channel_mask: pChannelMask ? *pChannelMask : 0, 7165 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7166 }; 7167 uint32_t reqSamplingRate = config.sample_rate; 7168 audio_format_t reqFormat = config.format; 7169 audio_channel_mask_t reqChannels = config.channel_mask; 7170 audio_stream_in_t *inStream = NULL; 7171 AudioHwDevice *inHwDev; 7172 7173 if (pDevices == NULL || *pDevices == 0) { 7174 return 0; 7175 } 7176 7177 Mutex::Autolock _l(mLock); 7178 7179 inHwDev = findSuitableHwDev_l(module, *pDevices); 7180 if (inHwDev == NULL) 7181 return 0; 7182 7183 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7184 audio_io_handle_t id = nextUniqueId(); 7185 7186 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7187 &inStream); 7188 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7189 inStream, 7190 config.sample_rate, 7191 config.format, 7192 config.channel_mask, 7193 status); 7194 7195 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7196 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7197 // or stereo to mono conversions on 16 bit PCM inputs. 7198 if (status == BAD_VALUE && 7199 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7200 (config.sample_rate <= 2 * reqSamplingRate) && 7201 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7202 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7203 inStream = NULL; 7204 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7205 } 7206 7207 if (status == NO_ERROR && inStream != NULL) { 7208 7209 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7210 // or (re-)create if current Pipe is idle and does not match the new format 7211 sp<NBAIO_Sink> teeSink; 7212#ifdef TEE_SINK_INPUT_FRAMES 7213 enum { 7214 TEE_SINK_NO, // don't copy input 7215 TEE_SINK_NEW, // copy input using a new pipe 7216 TEE_SINK_OLD, // copy input using an existing pipe 7217 } kind; 7218 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7219 popcount(inStream->common.get_channels(&inStream->common))); 7220 if (format == Format_Invalid) { 7221 kind = TEE_SINK_NO; 7222 } else if (mRecordTeeSink == 0) { 7223 kind = TEE_SINK_NEW; 7224 } else if (mRecordTeeSink->getStrongCount() != 1) { 7225 kind = TEE_SINK_NO; 7226 } else if (format == mRecordTeeSink->format()) { 7227 kind = TEE_SINK_OLD; 7228 } else { 7229 kind = TEE_SINK_NEW; 7230 } 7231 switch (kind) { 7232 case TEE_SINK_NEW: { 7233 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7234 size_t numCounterOffers = 0; 7235 const NBAIO_Format offers[1] = {format}; 7236 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7237 ALOG_ASSERT(index == 0); 7238 PipeReader *pipeReader = new PipeReader(*pipe); 7239 numCounterOffers = 0; 7240 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7241 ALOG_ASSERT(index == 0); 7242 mRecordTeeSink = pipe; 7243 mRecordTeeSource = pipeReader; 7244 teeSink = pipe; 7245 } 7246 break; 7247 case TEE_SINK_OLD: 7248 teeSink = mRecordTeeSink; 7249 break; 7250 case TEE_SINK_NO: 7251 default: 7252 break; 7253 } 7254#endif 7255 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7256 7257 // Start record thread 7258 // RecorThread require both input and output device indication to forward to audio 7259 // pre processing modules 7260 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7261 7262 thread = new RecordThread(this, 7263 input, 7264 reqSamplingRate, 7265 reqChannels, 7266 id, 7267 device, teeSink); 7268 mRecordThreads.add(id, thread); 7269 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7270 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7271 if (pFormat != NULL) *pFormat = config.format; 7272 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7273 7274 // notify client processes of the new input creation 7275 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7276 return id; 7277 } 7278 7279 return 0; 7280} 7281 7282status_t AudioFlinger::closeInput(audio_io_handle_t input) 7283{ 7284 return closeInput_nonvirtual(input); 7285} 7286 7287status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7288{ 7289 // keep strong reference on the record thread so that 7290 // it is not destroyed while exit() is executed 7291 sp<RecordThread> thread; 7292 { 7293 Mutex::Autolock _l(mLock); 7294 thread = checkRecordThread_l(input); 7295 if (thread == 0) { 7296 return BAD_VALUE; 7297 } 7298 7299 ALOGV("closeInput() %d", input); 7300 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7301 mRecordThreads.removeItem(input); 7302 } 7303 thread->exit(); 7304 // The thread entity (active unit of execution) is no longer running here, 7305 // but the ThreadBase container still exists. 7306 7307 AudioStreamIn *in = thread->clearInput(); 7308 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7309 // from now on thread->mInput is NULL 7310 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7311 delete in; 7312 7313 return NO_ERROR; 7314} 7315 7316status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7317{ 7318 Mutex::Autolock _l(mLock); 7319 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7320 7321 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7322 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7323 thread->invalidateTracks(stream); 7324 } 7325 7326 return NO_ERROR; 7327} 7328 7329 7330int AudioFlinger::newAudioSessionId() 7331{ 7332 return nextUniqueId(); 7333} 7334 7335void AudioFlinger::acquireAudioSessionId(int audioSession) 7336{ 7337 Mutex::Autolock _l(mLock); 7338 pid_t caller = IPCThreadState::self()->getCallingPid(); 7339 ALOGV("acquiring %d from %d", audioSession, caller); 7340 size_t num = mAudioSessionRefs.size(); 7341 for (size_t i = 0; i< num; i++) { 7342 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7343 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7344 ref->mCnt++; 7345 ALOGV(" incremented refcount to %d", ref->mCnt); 7346 return; 7347 } 7348 } 7349 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7350 ALOGV(" added new entry for %d", audioSession); 7351} 7352 7353void AudioFlinger::releaseAudioSessionId(int audioSession) 7354{ 7355 Mutex::Autolock _l(mLock); 7356 pid_t caller = IPCThreadState::self()->getCallingPid(); 7357 ALOGV("releasing %d from %d", audioSession, caller); 7358 size_t num = mAudioSessionRefs.size(); 7359 for (size_t i = 0; i< num; i++) { 7360 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7361 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7362 ref->mCnt--; 7363 ALOGV(" decremented refcount to %d", ref->mCnt); 7364 if (ref->mCnt == 0) { 7365 mAudioSessionRefs.removeAt(i); 7366 delete ref; 7367 purgeStaleEffects_l(); 7368 } 7369 return; 7370 } 7371 } 7372 ALOGW("session id %d not found for pid %d", audioSession, caller); 7373} 7374 7375void AudioFlinger::purgeStaleEffects_l() { 7376 7377 ALOGV("purging stale effects"); 7378 7379 Vector< sp<EffectChain> > chains; 7380 7381 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7382 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7383 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7384 sp<EffectChain> ec = t->mEffectChains[j]; 7385 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7386 chains.push(ec); 7387 } 7388 } 7389 } 7390 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7391 sp<RecordThread> t = mRecordThreads.valueAt(i); 7392 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7393 sp<EffectChain> ec = t->mEffectChains[j]; 7394 chains.push(ec); 7395 } 7396 } 7397 7398 for (size_t i = 0; i < chains.size(); i++) { 7399 sp<EffectChain> ec = chains[i]; 7400 int sessionid = ec->sessionId(); 7401 sp<ThreadBase> t = ec->mThread.promote(); 7402 if (t == 0) { 7403 continue; 7404 } 7405 size_t numsessionrefs = mAudioSessionRefs.size(); 7406 bool found = false; 7407 for (size_t k = 0; k < numsessionrefs; k++) { 7408 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7409 if (ref->mSessionid == sessionid) { 7410 ALOGV(" session %d still exists for %d with %d refs", 7411 sessionid, ref->mPid, ref->mCnt); 7412 found = true; 7413 break; 7414 } 7415 } 7416 if (!found) { 7417 Mutex::Autolock _l (t->mLock); 7418 // remove all effects from the chain 7419 while (ec->mEffects.size()) { 7420 sp<EffectModule> effect = ec->mEffects[0]; 7421 effect->unPin(); 7422 t->removeEffect_l(effect); 7423 if (effect->purgeHandles()) { 7424 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7425 } 7426 AudioSystem::unregisterEffect(effect->id()); 7427 } 7428 } 7429 } 7430 return; 7431} 7432 7433// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7434AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7435{ 7436 return mPlaybackThreads.valueFor(output).get(); 7437} 7438 7439// checkMixerThread_l() must be called with AudioFlinger::mLock held 7440AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7441{ 7442 PlaybackThread *thread = checkPlaybackThread_l(output); 7443 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7444} 7445 7446// checkRecordThread_l() must be called with AudioFlinger::mLock held 7447AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7448{ 7449 return mRecordThreads.valueFor(input).get(); 7450} 7451 7452uint32_t AudioFlinger::nextUniqueId() 7453{ 7454 return android_atomic_inc(&mNextUniqueId); 7455} 7456 7457AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7458{ 7459 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7460 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7461 AudioStreamOut *output = thread->getOutput(); 7462 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7463 return thread; 7464 } 7465 } 7466 return NULL; 7467} 7468 7469audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7470{ 7471 PlaybackThread *thread = primaryPlaybackThread_l(); 7472 7473 if (thread == NULL) { 7474 return 0; 7475 } 7476 7477 return thread->outDevice(); 7478} 7479 7480sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7481 int triggerSession, 7482 int listenerSession, 7483 sync_event_callback_t callBack, 7484 void *cookie) 7485{ 7486 Mutex::Autolock _l(mLock); 7487 7488 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7489 status_t playStatus = NAME_NOT_FOUND; 7490 status_t recStatus = NAME_NOT_FOUND; 7491 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7492 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7493 if (playStatus == NO_ERROR) { 7494 return event; 7495 } 7496 } 7497 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7498 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7499 if (recStatus == NO_ERROR) { 7500 return event; 7501 } 7502 } 7503 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7504 mPendingSyncEvents.add(event); 7505 } else { 7506 ALOGV("createSyncEvent() invalid event %d", event->type()); 7507 event.clear(); 7508 } 7509 return event; 7510} 7511 7512// ---------------------------------------------------------------------------- 7513// Effect management 7514// ---------------------------------------------------------------------------- 7515 7516 7517status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7518{ 7519 Mutex::Autolock _l(mLock); 7520 return EffectQueryNumberEffects(numEffects); 7521} 7522 7523status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7524{ 7525 Mutex::Autolock _l(mLock); 7526 return EffectQueryEffect(index, descriptor); 7527} 7528 7529status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7530 effect_descriptor_t *descriptor) const 7531{ 7532 Mutex::Autolock _l(mLock); 7533 return EffectGetDescriptor(pUuid, descriptor); 7534} 7535 7536 7537sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7538 effect_descriptor_t *pDesc, 7539 const sp<IEffectClient>& effectClient, 7540 int32_t priority, 7541 audio_io_handle_t io, 7542 int sessionId, 7543 status_t *status, 7544 int *id, 7545 int *enabled) 7546{ 7547 status_t lStatus = NO_ERROR; 7548 sp<EffectHandle> handle; 7549 effect_descriptor_t desc; 7550 7551 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7552 pid, effectClient.get(), priority, sessionId, io); 7553 7554 if (pDesc == NULL) { 7555 lStatus = BAD_VALUE; 7556 goto Exit; 7557 } 7558 7559 // check audio settings permission for global effects 7560 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7561 lStatus = PERMISSION_DENIED; 7562 goto Exit; 7563 } 7564 7565 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7566 // that can only be created by audio policy manager (running in same process) 7567 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7568 lStatus = PERMISSION_DENIED; 7569 goto Exit; 7570 } 7571 7572 if (io == 0) { 7573 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7574 // output must be specified by AudioPolicyManager when using session 7575 // AUDIO_SESSION_OUTPUT_STAGE 7576 lStatus = BAD_VALUE; 7577 goto Exit; 7578 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7579 // if the output returned by getOutputForEffect() is removed before we lock the 7580 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7581 // and we will exit safely 7582 io = AudioSystem::getOutputForEffect(&desc); 7583 } 7584 } 7585 7586 { 7587 Mutex::Autolock _l(mLock); 7588 7589 7590 if (!EffectIsNullUuid(&pDesc->uuid)) { 7591 // if uuid is specified, request effect descriptor 7592 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7593 if (lStatus < 0) { 7594 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7595 goto Exit; 7596 } 7597 } else { 7598 // if uuid is not specified, look for an available implementation 7599 // of the required type in effect factory 7600 if (EffectIsNullUuid(&pDesc->type)) { 7601 ALOGW("createEffect() no effect type"); 7602 lStatus = BAD_VALUE; 7603 goto Exit; 7604 } 7605 uint32_t numEffects = 0; 7606 effect_descriptor_t d; 7607 d.flags = 0; // prevent compiler warning 7608 bool found = false; 7609 7610 lStatus = EffectQueryNumberEffects(&numEffects); 7611 if (lStatus < 0) { 7612 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7613 goto Exit; 7614 } 7615 for (uint32_t i = 0; i < numEffects; i++) { 7616 lStatus = EffectQueryEffect(i, &desc); 7617 if (lStatus < 0) { 7618 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7619 continue; 7620 } 7621 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7622 // If matching type found save effect descriptor. If the session is 7623 // 0 and the effect is not auxiliary, continue enumeration in case 7624 // an auxiliary version of this effect type is available 7625 found = true; 7626 d = desc; 7627 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7628 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7629 break; 7630 } 7631 } 7632 } 7633 if (!found) { 7634 lStatus = BAD_VALUE; 7635 ALOGW("createEffect() effect not found"); 7636 goto Exit; 7637 } 7638 // For same effect type, chose auxiliary version over insert version if 7639 // connect to output mix (Compliance to OpenSL ES) 7640 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7641 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7642 desc = d; 7643 } 7644 } 7645 7646 // Do not allow auxiliary effects on a session different from 0 (output mix) 7647 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7648 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7649 lStatus = INVALID_OPERATION; 7650 goto Exit; 7651 } 7652 7653 // check recording permission for visualizer 7654 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7655 !recordingAllowed()) { 7656 lStatus = PERMISSION_DENIED; 7657 goto Exit; 7658 } 7659 7660 // return effect descriptor 7661 *pDesc = desc; 7662 7663 // If output is not specified try to find a matching audio session ID in one of the 7664 // output threads. 7665 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7666 // because of code checking output when entering the function. 7667 // Note: io is never 0 when creating an effect on an input 7668 if (io == 0) { 7669 // look for the thread where the specified audio session is present 7670 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7671 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7672 io = mPlaybackThreads.keyAt(i); 7673 break; 7674 } 7675 } 7676 if (io == 0) { 7677 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7678 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7679 io = mRecordThreads.keyAt(i); 7680 break; 7681 } 7682 } 7683 } 7684 // If no output thread contains the requested session ID, default to 7685 // first output. The effect chain will be moved to the correct output 7686 // thread when a track with the same session ID is created 7687 if (io == 0 && mPlaybackThreads.size()) { 7688 io = mPlaybackThreads.keyAt(0); 7689 } 7690 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7691 } 7692 ThreadBase *thread = checkRecordThread_l(io); 7693 if (thread == NULL) { 7694 thread = checkPlaybackThread_l(io); 7695 if (thread == NULL) { 7696 ALOGE("createEffect() unknown output thread"); 7697 lStatus = BAD_VALUE; 7698 goto Exit; 7699 } 7700 } 7701 7702 sp<Client> client = registerPid_l(pid); 7703 7704 // create effect on selected output thread 7705 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7706 &desc, enabled, &lStatus); 7707 if (handle != 0 && id != NULL) { 7708 *id = handle->id(); 7709 } 7710 } 7711 7712Exit: 7713 if (status != NULL) { 7714 *status = lStatus; 7715 } 7716 return handle; 7717} 7718 7719status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7720 audio_io_handle_t dstOutput) 7721{ 7722 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7723 sessionId, srcOutput, dstOutput); 7724 Mutex::Autolock _l(mLock); 7725 if (srcOutput == dstOutput) { 7726 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7727 return NO_ERROR; 7728 } 7729 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7730 if (srcThread == NULL) { 7731 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7732 return BAD_VALUE; 7733 } 7734 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7735 if (dstThread == NULL) { 7736 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7737 return BAD_VALUE; 7738 } 7739 7740 Mutex::Autolock _dl(dstThread->mLock); 7741 Mutex::Autolock _sl(srcThread->mLock); 7742 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7743 7744 return NO_ERROR; 7745} 7746 7747// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7748status_t AudioFlinger::moveEffectChain_l(int sessionId, 7749 AudioFlinger::PlaybackThread *srcThread, 7750 AudioFlinger::PlaybackThread *dstThread, 7751 bool reRegister) 7752{ 7753 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7754 sessionId, srcThread, dstThread); 7755 7756 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7757 if (chain == 0) { 7758 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7759 sessionId, srcThread); 7760 return INVALID_OPERATION; 7761 } 7762 7763 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7764 // so that a new chain is created with correct parameters when first effect is added. This is 7765 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7766 // removed. 7767 srcThread->removeEffectChain_l(chain); 7768 7769 // transfer all effects one by one so that new effect chain is created on new thread with 7770 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7771 audio_io_handle_t dstOutput = dstThread->id(); 7772 sp<EffectChain> dstChain; 7773 uint32_t strategy = 0; // prevent compiler warning 7774 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7775 while (effect != 0) { 7776 srcThread->removeEffect_l(effect); 7777 dstThread->addEffect_l(effect); 7778 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7779 if (effect->state() == EffectModule::ACTIVE || 7780 effect->state() == EffectModule::STOPPING) { 7781 effect->start(); 7782 } 7783 // if the move request is not received from audio policy manager, the effect must be 7784 // re-registered with the new strategy and output 7785 if (dstChain == 0) { 7786 dstChain = effect->chain().promote(); 7787 if (dstChain == 0) { 7788 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7789 srcThread->addEffect_l(effect); 7790 return NO_INIT; 7791 } 7792 strategy = dstChain->strategy(); 7793 } 7794 if (reRegister) { 7795 AudioSystem::unregisterEffect(effect->id()); 7796 AudioSystem::registerEffect(&effect->desc(), 7797 dstOutput, 7798 strategy, 7799 sessionId, 7800 effect->id()); 7801 } 7802 effect = chain->getEffectFromId_l(0); 7803 } 7804 7805 return NO_ERROR; 7806} 7807 7808 7809// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7810sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7811 const sp<AudioFlinger::Client>& client, 7812 const sp<IEffectClient>& effectClient, 7813 int32_t priority, 7814 int sessionId, 7815 effect_descriptor_t *desc, 7816 int *enabled, 7817 status_t *status 7818 ) 7819{ 7820 sp<EffectModule> effect; 7821 sp<EffectHandle> handle; 7822 status_t lStatus; 7823 sp<EffectChain> chain; 7824 bool chainCreated = false; 7825 bool effectCreated = false; 7826 bool effectRegistered = false; 7827 7828 lStatus = initCheck(); 7829 if (lStatus != NO_ERROR) { 7830 ALOGW("createEffect_l() Audio driver not initialized."); 7831 goto Exit; 7832 } 7833 7834 // Do not allow effects with session ID 0 on direct output or duplicating threads 7835 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7836 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7837 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7838 desc->name, sessionId); 7839 lStatus = BAD_VALUE; 7840 goto Exit; 7841 } 7842 // Only Pre processor effects are allowed on input threads and only on input threads 7843 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7844 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7845 desc->name, desc->flags, mType); 7846 lStatus = BAD_VALUE; 7847 goto Exit; 7848 } 7849 7850 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7851 7852 { // scope for mLock 7853 Mutex::Autolock _l(mLock); 7854 7855 // check for existing effect chain with the requested audio session 7856 chain = getEffectChain_l(sessionId); 7857 if (chain == 0) { 7858 // create a new chain for this session 7859 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7860 chain = new EffectChain(this, sessionId); 7861 addEffectChain_l(chain); 7862 chain->setStrategy(getStrategyForSession_l(sessionId)); 7863 chainCreated = true; 7864 } else { 7865 effect = chain->getEffectFromDesc_l(desc); 7866 } 7867 7868 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7869 7870 if (effect == 0) { 7871 int id = mAudioFlinger->nextUniqueId(); 7872 // Check CPU and memory usage 7873 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7874 if (lStatus != NO_ERROR) { 7875 goto Exit; 7876 } 7877 effectRegistered = true; 7878 // create a new effect module if none present in the chain 7879 effect = new EffectModule(this, chain, desc, id, sessionId); 7880 lStatus = effect->status(); 7881 if (lStatus != NO_ERROR) { 7882 goto Exit; 7883 } 7884 lStatus = chain->addEffect_l(effect); 7885 if (lStatus != NO_ERROR) { 7886 goto Exit; 7887 } 7888 effectCreated = true; 7889 7890 effect->setDevice(mOutDevice); 7891 effect->setDevice(mInDevice); 7892 effect->setMode(mAudioFlinger->getMode()); 7893 effect->setAudioSource(mAudioSource); 7894 } 7895 // create effect handle and connect it to effect module 7896 handle = new EffectHandle(effect, client, effectClient, priority); 7897 lStatus = effect->addHandle(handle.get()); 7898 if (enabled != NULL) { 7899 *enabled = (int)effect->isEnabled(); 7900 } 7901 } 7902 7903Exit: 7904 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7905 Mutex::Autolock _l(mLock); 7906 if (effectCreated) { 7907 chain->removeEffect_l(effect); 7908 } 7909 if (effectRegistered) { 7910 AudioSystem::unregisterEffect(effect->id()); 7911 } 7912 if (chainCreated) { 7913 removeEffectChain_l(chain); 7914 } 7915 handle.clear(); 7916 } 7917 7918 if (status != NULL) { 7919 *status = lStatus; 7920 } 7921 return handle; 7922} 7923 7924sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7925{ 7926 Mutex::Autolock _l(mLock); 7927 return getEffect_l(sessionId, effectId); 7928} 7929 7930sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7931{ 7932 sp<EffectChain> chain = getEffectChain_l(sessionId); 7933 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7934} 7935 7936// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7937// PlaybackThread::mLock held 7938status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7939{ 7940 // check for existing effect chain with the requested audio session 7941 int sessionId = effect->sessionId(); 7942 sp<EffectChain> chain = getEffectChain_l(sessionId); 7943 bool chainCreated = false; 7944 7945 if (chain == 0) { 7946 // create a new chain for this session 7947 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7948 chain = new EffectChain(this, sessionId); 7949 addEffectChain_l(chain); 7950 chain->setStrategy(getStrategyForSession_l(sessionId)); 7951 chainCreated = true; 7952 } 7953 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7954 7955 if (chain->getEffectFromId_l(effect->id()) != 0) { 7956 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7957 this, effect->desc().name, chain.get()); 7958 return BAD_VALUE; 7959 } 7960 7961 status_t status = chain->addEffect_l(effect); 7962 if (status != NO_ERROR) { 7963 if (chainCreated) { 7964 removeEffectChain_l(chain); 7965 } 7966 return status; 7967 } 7968 7969 effect->setDevice(mOutDevice); 7970 effect->setDevice(mInDevice); 7971 effect->setMode(mAudioFlinger->getMode()); 7972 effect->setAudioSource(mAudioSource); 7973 return NO_ERROR; 7974} 7975 7976void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7977 7978 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7979 effect_descriptor_t desc = effect->desc(); 7980 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7981 detachAuxEffect_l(effect->id()); 7982 } 7983 7984 sp<EffectChain> chain = effect->chain().promote(); 7985 if (chain != 0) { 7986 // remove effect chain if removing last effect 7987 if (chain->removeEffect_l(effect) == 0) { 7988 removeEffectChain_l(chain); 7989 } 7990 } else { 7991 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7992 } 7993} 7994 7995void AudioFlinger::ThreadBase::lockEffectChains_l( 7996 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7997{ 7998 effectChains = mEffectChains; 7999 for (size_t i = 0; i < mEffectChains.size(); i++) { 8000 mEffectChains[i]->lock(); 8001 } 8002} 8003 8004void AudioFlinger::ThreadBase::unlockEffectChains( 8005 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8006{ 8007 for (size_t i = 0; i < effectChains.size(); i++) { 8008 effectChains[i]->unlock(); 8009 } 8010} 8011 8012sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8013{ 8014 Mutex::Autolock _l(mLock); 8015 return getEffectChain_l(sessionId); 8016} 8017 8018sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8019{ 8020 size_t size = mEffectChains.size(); 8021 for (size_t i = 0; i < size; i++) { 8022 if (mEffectChains[i]->sessionId() == sessionId) { 8023 return mEffectChains[i]; 8024 } 8025 } 8026 return 0; 8027} 8028 8029void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8030{ 8031 Mutex::Autolock _l(mLock); 8032 size_t size = mEffectChains.size(); 8033 for (size_t i = 0; i < size; i++) { 8034 mEffectChains[i]->setMode_l(mode); 8035 } 8036} 8037 8038void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8039 EffectHandle *handle, 8040 bool unpinIfLast) { 8041 8042 Mutex::Autolock _l(mLock); 8043 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8044 // delete the effect module if removing last handle on it 8045 if (effect->removeHandle(handle) == 0) { 8046 if (!effect->isPinned() || unpinIfLast) { 8047 removeEffect_l(effect); 8048 AudioSystem::unregisterEffect(effect->id()); 8049 } 8050 } 8051} 8052 8053status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8054{ 8055 int session = chain->sessionId(); 8056 int16_t *buffer = mMixBuffer; 8057 bool ownsBuffer = false; 8058 8059 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8060 if (session > 0) { 8061 // Only one effect chain can be present in direct output thread and it uses 8062 // the mix buffer as input 8063 if (mType != DIRECT) { 8064 size_t numSamples = mNormalFrameCount * mChannelCount; 8065 buffer = new int16_t[numSamples]; 8066 memset(buffer, 0, numSamples * sizeof(int16_t)); 8067 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8068 ownsBuffer = true; 8069 } 8070 8071 // Attach all tracks with same session ID to this chain. 8072 for (size_t i = 0; i < mTracks.size(); ++i) { 8073 sp<Track> track = mTracks[i]; 8074 if (session == track->sessionId()) { 8075 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 8076 track->setMainBuffer(buffer); 8077 chain->incTrackCnt(); 8078 } 8079 } 8080 8081 // indicate all active tracks in the chain 8082 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8083 sp<Track> track = mActiveTracks[i].promote(); 8084 if (track == 0) continue; 8085 if (session == track->sessionId()) { 8086 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8087 chain->incActiveTrackCnt(); 8088 } 8089 } 8090 } 8091 8092 chain->setInBuffer(buffer, ownsBuffer); 8093 chain->setOutBuffer(mMixBuffer); 8094 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8095 // chains list in order to be processed last as it contains output stage effects 8096 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8097 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8098 // after track specific effects and before output stage 8099 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8100 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8101 // Effect chain for other sessions are inserted at beginning of effect 8102 // chains list to be processed before output mix effects. Relative order between other 8103 // sessions is not important 8104 size_t size = mEffectChains.size(); 8105 size_t i = 0; 8106 for (i = 0; i < size; i++) { 8107 if (mEffectChains[i]->sessionId() < session) break; 8108 } 8109 mEffectChains.insertAt(chain, i); 8110 checkSuspendOnAddEffectChain_l(chain); 8111 8112 return NO_ERROR; 8113} 8114 8115size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8116{ 8117 int session = chain->sessionId(); 8118 8119 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8120 8121 for (size_t i = 0; i < mEffectChains.size(); i++) { 8122 if (chain == mEffectChains[i]) { 8123 mEffectChains.removeAt(i); 8124 // detach all active tracks from the chain 8125 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8126 sp<Track> track = mActiveTracks[i].promote(); 8127 if (track == 0) continue; 8128 if (session == track->sessionId()) { 8129 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8130 chain.get(), session); 8131 chain->decActiveTrackCnt(); 8132 } 8133 } 8134 8135 // detach all tracks with same session ID from this chain 8136 for (size_t i = 0; i < mTracks.size(); ++i) { 8137 sp<Track> track = mTracks[i]; 8138 if (session == track->sessionId()) { 8139 track->setMainBuffer(mMixBuffer); 8140 chain->decTrackCnt(); 8141 } 8142 } 8143 break; 8144 } 8145 } 8146 return mEffectChains.size(); 8147} 8148 8149status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8150 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8151{ 8152 Mutex::Autolock _l(mLock); 8153 return attachAuxEffect_l(track, EffectId); 8154} 8155 8156status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8157 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8158{ 8159 status_t status = NO_ERROR; 8160 8161 if (EffectId == 0) { 8162 track->setAuxBuffer(0, NULL); 8163 } else { 8164 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8165 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8166 if (effect != 0) { 8167 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8168 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8169 } else { 8170 status = INVALID_OPERATION; 8171 } 8172 } else { 8173 status = BAD_VALUE; 8174 } 8175 } 8176 return status; 8177} 8178 8179void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8180{ 8181 for (size_t i = 0; i < mTracks.size(); ++i) { 8182 sp<Track> track = mTracks[i]; 8183 if (track->auxEffectId() == effectId) { 8184 attachAuxEffect_l(track, 0); 8185 } 8186 } 8187} 8188 8189status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8190{ 8191 // only one chain per input thread 8192 if (mEffectChains.size() != 0) { 8193 return INVALID_OPERATION; 8194 } 8195 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8196 8197 chain->setInBuffer(NULL); 8198 chain->setOutBuffer(NULL); 8199 8200 checkSuspendOnAddEffectChain_l(chain); 8201 8202 mEffectChains.add(chain); 8203 8204 return NO_ERROR; 8205} 8206 8207size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8208{ 8209 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8210 ALOGW_IF(mEffectChains.size() != 1, 8211 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8212 chain.get(), mEffectChains.size(), this); 8213 if (mEffectChains.size() == 1) { 8214 mEffectChains.removeAt(0); 8215 } 8216 return 0; 8217} 8218 8219// ---------------------------------------------------------------------------- 8220// EffectModule implementation 8221// ---------------------------------------------------------------------------- 8222 8223#undef LOG_TAG 8224#define LOG_TAG "AudioFlinger::EffectModule" 8225 8226AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8227 const wp<AudioFlinger::EffectChain>& chain, 8228 effect_descriptor_t *desc, 8229 int id, 8230 int sessionId) 8231 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8232 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8233 mDescriptor(*desc), 8234 // mConfig is set by configure() and not used before then 8235 mEffectInterface(NULL), 8236 mStatus(NO_INIT), mState(IDLE), 8237 // mMaxDisableWaitCnt is set by configure() and not used before then 8238 // mDisableWaitCnt is set by process() and updateState() and not used before then 8239 mSuspended(false) 8240{ 8241 ALOGV("Constructor %p", this); 8242 int lStatus; 8243 8244 // create effect engine from effect factory 8245 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8246 8247 if (mStatus != NO_ERROR) { 8248 return; 8249 } 8250 lStatus = init(); 8251 if (lStatus < 0) { 8252 mStatus = lStatus; 8253 goto Error; 8254 } 8255 8256 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8257 return; 8258Error: 8259 EffectRelease(mEffectInterface); 8260 mEffectInterface = NULL; 8261 ALOGV("Constructor Error %d", mStatus); 8262} 8263 8264AudioFlinger::EffectModule::~EffectModule() 8265{ 8266 ALOGV("Destructor %p", this); 8267 if (mEffectInterface != NULL) { 8268 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8269 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8270 sp<ThreadBase> thread = mThread.promote(); 8271 if (thread != 0) { 8272 audio_stream_t *stream = thread->stream(); 8273 if (stream != NULL) { 8274 stream->remove_audio_effect(stream, mEffectInterface); 8275 } 8276 } 8277 } 8278 // release effect engine 8279 EffectRelease(mEffectInterface); 8280 } 8281} 8282 8283status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8284{ 8285 status_t status; 8286 8287 Mutex::Autolock _l(mLock); 8288 int priority = handle->priority(); 8289 size_t size = mHandles.size(); 8290 EffectHandle *controlHandle = NULL; 8291 size_t i; 8292 for (i = 0; i < size; i++) { 8293 EffectHandle *h = mHandles[i]; 8294 if (h == NULL || h->destroyed_l()) continue; 8295 // first non destroyed handle is considered in control 8296 if (controlHandle == NULL) 8297 controlHandle = h; 8298 if (h->priority() <= priority) break; 8299 } 8300 // if inserted in first place, move effect control from previous owner to this handle 8301 if (i == 0) { 8302 bool enabled = false; 8303 if (controlHandle != NULL) { 8304 enabled = controlHandle->enabled(); 8305 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8306 } 8307 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8308 status = NO_ERROR; 8309 } else { 8310 status = ALREADY_EXISTS; 8311 } 8312 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8313 mHandles.insertAt(handle, i); 8314 return status; 8315} 8316 8317size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8318{ 8319 Mutex::Autolock _l(mLock); 8320 size_t size = mHandles.size(); 8321 size_t i; 8322 for (i = 0; i < size; i++) { 8323 if (mHandles[i] == handle) break; 8324 } 8325 if (i == size) { 8326 return size; 8327 } 8328 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8329 8330 mHandles.removeAt(i); 8331 // if removed from first place, move effect control from this handle to next in line 8332 if (i == 0) { 8333 EffectHandle *h = controlHandle_l(); 8334 if (h != NULL) { 8335 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8336 } 8337 } 8338 8339 // Prevent calls to process() and other functions on effect interface from now on. 8340 // The effect engine will be released by the destructor when the last strong reference on 8341 // this object is released which can happen after next process is called. 8342 if (mHandles.size() == 0 && !mPinned) { 8343 mState = DESTROYED; 8344 } 8345 8346 return mHandles.size(); 8347} 8348 8349// must be called with EffectModule::mLock held 8350AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8351{ 8352 // the first valid handle in the list has control over the module 8353 for (size_t i = 0; i < mHandles.size(); i++) { 8354 EffectHandle *h = mHandles[i]; 8355 if (h != NULL && !h->destroyed_l()) { 8356 return h; 8357 } 8358 } 8359 8360 return NULL; 8361} 8362 8363size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8364{ 8365 ALOGV("disconnect() %p handle %p", this, handle); 8366 // keep a strong reference on this EffectModule to avoid calling the 8367 // destructor before we exit 8368 sp<EffectModule> keep(this); 8369 { 8370 sp<ThreadBase> thread = mThread.promote(); 8371 if (thread != 0) { 8372 thread->disconnectEffect(keep, handle, unpinIfLast); 8373 } 8374 } 8375 return mHandles.size(); 8376} 8377 8378void AudioFlinger::EffectModule::updateState() { 8379 Mutex::Autolock _l(mLock); 8380 8381 switch (mState) { 8382 case RESTART: 8383 reset_l(); 8384 // FALL THROUGH 8385 8386 case STARTING: 8387 // clear auxiliary effect input buffer for next accumulation 8388 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8389 memset(mConfig.inputCfg.buffer.raw, 8390 0, 8391 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8392 } 8393 start_l(); 8394 mState = ACTIVE; 8395 break; 8396 case STOPPING: 8397 stop_l(); 8398 mDisableWaitCnt = mMaxDisableWaitCnt; 8399 mState = STOPPED; 8400 break; 8401 case STOPPED: 8402 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8403 // turn off sequence. 8404 if (--mDisableWaitCnt == 0) { 8405 reset_l(); 8406 mState = IDLE; 8407 } 8408 break; 8409 default: //IDLE , ACTIVE, DESTROYED 8410 break; 8411 } 8412} 8413 8414void AudioFlinger::EffectModule::process() 8415{ 8416 Mutex::Autolock _l(mLock); 8417 8418 if (mState == DESTROYED || mEffectInterface == NULL || 8419 mConfig.inputCfg.buffer.raw == NULL || 8420 mConfig.outputCfg.buffer.raw == NULL) { 8421 return; 8422 } 8423 8424 if (isProcessEnabled()) { 8425 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8426 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8427 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8428 mConfig.inputCfg.buffer.s32, 8429 mConfig.inputCfg.buffer.frameCount/2); 8430 } 8431 8432 // do the actual processing in the effect engine 8433 int ret = (*mEffectInterface)->process(mEffectInterface, 8434 &mConfig.inputCfg.buffer, 8435 &mConfig.outputCfg.buffer); 8436 8437 // force transition to IDLE state when engine is ready 8438 if (mState == STOPPED && ret == -ENODATA) { 8439 mDisableWaitCnt = 1; 8440 } 8441 8442 // clear auxiliary effect input buffer for next accumulation 8443 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8444 memset(mConfig.inputCfg.buffer.raw, 0, 8445 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8446 } 8447 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8448 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8449 // If an insert effect is idle and input buffer is different from output buffer, 8450 // accumulate input onto output 8451 sp<EffectChain> chain = mChain.promote(); 8452 if (chain != 0 && chain->activeTrackCnt() != 0) { 8453 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8454 int16_t *in = mConfig.inputCfg.buffer.s16; 8455 int16_t *out = mConfig.outputCfg.buffer.s16; 8456 for (size_t i = 0; i < frameCnt; i++) { 8457 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8458 } 8459 } 8460 } 8461} 8462 8463void AudioFlinger::EffectModule::reset_l() 8464{ 8465 if (mEffectInterface == NULL) { 8466 return; 8467 } 8468 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8469} 8470 8471status_t AudioFlinger::EffectModule::configure() 8472{ 8473 if (mEffectInterface == NULL) { 8474 return NO_INIT; 8475 } 8476 8477 sp<ThreadBase> thread = mThread.promote(); 8478 if (thread == 0) { 8479 return DEAD_OBJECT; 8480 } 8481 8482 // TODO: handle configuration of effects replacing track process 8483 audio_channel_mask_t channelMask = thread->channelMask(); 8484 8485 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8486 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8487 } else { 8488 mConfig.inputCfg.channels = channelMask; 8489 } 8490 mConfig.outputCfg.channels = channelMask; 8491 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8492 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8493 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8494 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8495 mConfig.inputCfg.bufferProvider.cookie = NULL; 8496 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8497 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8498 mConfig.outputCfg.bufferProvider.cookie = NULL; 8499 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8500 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8501 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8502 // Insert effect: 8503 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8504 // always overwrites output buffer: input buffer == output buffer 8505 // - in other sessions: 8506 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8507 // other effect: overwrites output buffer: input buffer == output buffer 8508 // Auxiliary effect: 8509 // accumulates in output buffer: input buffer != output buffer 8510 // Therefore: accumulate <=> input buffer != output buffer 8511 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8512 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8513 } else { 8514 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8515 } 8516 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8517 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8518 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8519 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8520 8521 ALOGV("configure() %p thread %p buffer %p framecount %d", 8522 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8523 8524 status_t cmdStatus; 8525 uint32_t size = sizeof(int); 8526 status_t status = (*mEffectInterface)->command(mEffectInterface, 8527 EFFECT_CMD_SET_CONFIG, 8528 sizeof(effect_config_t), 8529 &mConfig, 8530 &size, 8531 &cmdStatus); 8532 if (status == 0) { 8533 status = cmdStatus; 8534 } 8535 8536 if (status == 0 && 8537 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8538 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8539 effect_param_t *p = (effect_param_t *)buf32; 8540 8541 p->psize = sizeof(uint32_t); 8542 p->vsize = sizeof(uint32_t); 8543 size = sizeof(int); 8544 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8545 8546 uint32_t latency = 0; 8547 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8548 if (pbt != NULL) { 8549 latency = pbt->latency_l(); 8550 } 8551 8552 *((int32_t *)p->data + 1)= latency; 8553 (*mEffectInterface)->command(mEffectInterface, 8554 EFFECT_CMD_SET_PARAM, 8555 sizeof(effect_param_t) + 8, 8556 &buf32, 8557 &size, 8558 &cmdStatus); 8559 } 8560 8561 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8562 (1000 * mConfig.outputCfg.buffer.frameCount); 8563 8564 return status; 8565} 8566 8567status_t AudioFlinger::EffectModule::init() 8568{ 8569 Mutex::Autolock _l(mLock); 8570 if (mEffectInterface == NULL) { 8571 return NO_INIT; 8572 } 8573 status_t cmdStatus; 8574 uint32_t size = sizeof(status_t); 8575 status_t status = (*mEffectInterface)->command(mEffectInterface, 8576 EFFECT_CMD_INIT, 8577 0, 8578 NULL, 8579 &size, 8580 &cmdStatus); 8581 if (status == 0) { 8582 status = cmdStatus; 8583 } 8584 return status; 8585} 8586 8587status_t AudioFlinger::EffectModule::start() 8588{ 8589 Mutex::Autolock _l(mLock); 8590 return start_l(); 8591} 8592 8593status_t AudioFlinger::EffectModule::start_l() 8594{ 8595 if (mEffectInterface == NULL) { 8596 return NO_INIT; 8597 } 8598 status_t cmdStatus; 8599 uint32_t size = sizeof(status_t); 8600 status_t status = (*mEffectInterface)->command(mEffectInterface, 8601 EFFECT_CMD_ENABLE, 8602 0, 8603 NULL, 8604 &size, 8605 &cmdStatus); 8606 if (status == 0) { 8607 status = cmdStatus; 8608 } 8609 if (status == 0 && 8610 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8611 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8612 sp<ThreadBase> thread = mThread.promote(); 8613 if (thread != 0) { 8614 audio_stream_t *stream = thread->stream(); 8615 if (stream != NULL) { 8616 stream->add_audio_effect(stream, mEffectInterface); 8617 } 8618 } 8619 } 8620 return status; 8621} 8622 8623status_t AudioFlinger::EffectModule::stop() 8624{ 8625 Mutex::Autolock _l(mLock); 8626 return stop_l(); 8627} 8628 8629status_t AudioFlinger::EffectModule::stop_l() 8630{ 8631 if (mEffectInterface == NULL) { 8632 return NO_INIT; 8633 } 8634 status_t cmdStatus; 8635 uint32_t size = sizeof(status_t); 8636 status_t status = (*mEffectInterface)->command(mEffectInterface, 8637 EFFECT_CMD_DISABLE, 8638 0, 8639 NULL, 8640 &size, 8641 &cmdStatus); 8642 if (status == 0) { 8643 status = cmdStatus; 8644 } 8645 if (status == 0 && 8646 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8647 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8648 sp<ThreadBase> thread = mThread.promote(); 8649 if (thread != 0) { 8650 audio_stream_t *stream = thread->stream(); 8651 if (stream != NULL) { 8652 stream->remove_audio_effect(stream, mEffectInterface); 8653 } 8654 } 8655 } 8656 return status; 8657} 8658 8659status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8660 uint32_t cmdSize, 8661 void *pCmdData, 8662 uint32_t *replySize, 8663 void *pReplyData) 8664{ 8665 Mutex::Autolock _l(mLock); 8666 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8667 8668 if (mState == DESTROYED || mEffectInterface == NULL) { 8669 return NO_INIT; 8670 } 8671 status_t status = (*mEffectInterface)->command(mEffectInterface, 8672 cmdCode, 8673 cmdSize, 8674 pCmdData, 8675 replySize, 8676 pReplyData); 8677 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8678 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8679 for (size_t i = 1; i < mHandles.size(); i++) { 8680 EffectHandle *h = mHandles[i]; 8681 if (h != NULL && !h->destroyed_l()) { 8682 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8683 } 8684 } 8685 } 8686 return status; 8687} 8688 8689status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8690{ 8691 Mutex::Autolock _l(mLock); 8692 return setEnabled_l(enabled); 8693} 8694 8695// must be called with EffectModule::mLock held 8696status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8697{ 8698 8699 ALOGV("setEnabled %p enabled %d", this, enabled); 8700 8701 if (enabled != isEnabled()) { 8702 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8703 if (enabled && status != NO_ERROR) { 8704 return status; 8705 } 8706 8707 switch (mState) { 8708 // going from disabled to enabled 8709 case IDLE: 8710 mState = STARTING; 8711 break; 8712 case STOPPED: 8713 mState = RESTART; 8714 break; 8715 case STOPPING: 8716 mState = ACTIVE; 8717 break; 8718 8719 // going from enabled to disabled 8720 case RESTART: 8721 mState = STOPPED; 8722 break; 8723 case STARTING: 8724 mState = IDLE; 8725 break; 8726 case ACTIVE: 8727 mState = STOPPING; 8728 break; 8729 case DESTROYED: 8730 return NO_ERROR; // simply ignore as we are being destroyed 8731 } 8732 for (size_t i = 1; i < mHandles.size(); i++) { 8733 EffectHandle *h = mHandles[i]; 8734 if (h != NULL && !h->destroyed_l()) { 8735 h->setEnabled(enabled); 8736 } 8737 } 8738 } 8739 return NO_ERROR; 8740} 8741 8742bool AudioFlinger::EffectModule::isEnabled() const 8743{ 8744 switch (mState) { 8745 case RESTART: 8746 case STARTING: 8747 case ACTIVE: 8748 return true; 8749 case IDLE: 8750 case STOPPING: 8751 case STOPPED: 8752 case DESTROYED: 8753 default: 8754 return false; 8755 } 8756} 8757 8758bool AudioFlinger::EffectModule::isProcessEnabled() const 8759{ 8760 switch (mState) { 8761 case RESTART: 8762 case ACTIVE: 8763 case STOPPING: 8764 case STOPPED: 8765 return true; 8766 case IDLE: 8767 case STARTING: 8768 case DESTROYED: 8769 default: 8770 return false; 8771 } 8772} 8773 8774status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8775{ 8776 Mutex::Autolock _l(mLock); 8777 status_t status = NO_ERROR; 8778 8779 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8780 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8781 if (isProcessEnabled() && 8782 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8783 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8784 status_t cmdStatus; 8785 uint32_t volume[2]; 8786 uint32_t *pVolume = NULL; 8787 uint32_t size = sizeof(volume); 8788 volume[0] = *left; 8789 volume[1] = *right; 8790 if (controller) { 8791 pVolume = volume; 8792 } 8793 status = (*mEffectInterface)->command(mEffectInterface, 8794 EFFECT_CMD_SET_VOLUME, 8795 size, 8796 volume, 8797 &size, 8798 pVolume); 8799 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8800 *left = volume[0]; 8801 *right = volume[1]; 8802 } 8803 } 8804 return status; 8805} 8806 8807status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8808{ 8809 if (device == AUDIO_DEVICE_NONE) { 8810 return NO_ERROR; 8811 } 8812 8813 Mutex::Autolock _l(mLock); 8814 status_t status = NO_ERROR; 8815 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8816 status_t cmdStatus; 8817 uint32_t size = sizeof(status_t); 8818 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8819 EFFECT_CMD_SET_INPUT_DEVICE; 8820 status = (*mEffectInterface)->command(mEffectInterface, 8821 cmd, 8822 sizeof(uint32_t), 8823 &device, 8824 &size, 8825 &cmdStatus); 8826 } 8827 return status; 8828} 8829 8830status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8831{ 8832 Mutex::Autolock _l(mLock); 8833 status_t status = NO_ERROR; 8834 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8835 status_t cmdStatus; 8836 uint32_t size = sizeof(status_t); 8837 status = (*mEffectInterface)->command(mEffectInterface, 8838 EFFECT_CMD_SET_AUDIO_MODE, 8839 sizeof(audio_mode_t), 8840 &mode, 8841 &size, 8842 &cmdStatus); 8843 if (status == NO_ERROR) { 8844 status = cmdStatus; 8845 } 8846 } 8847 return status; 8848} 8849 8850status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8851{ 8852 Mutex::Autolock _l(mLock); 8853 status_t status = NO_ERROR; 8854 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8855 uint32_t size = 0; 8856 status = (*mEffectInterface)->command(mEffectInterface, 8857 EFFECT_CMD_SET_AUDIO_SOURCE, 8858 sizeof(audio_source_t), 8859 &source, 8860 &size, 8861 NULL); 8862 } 8863 return status; 8864} 8865 8866void AudioFlinger::EffectModule::setSuspended(bool suspended) 8867{ 8868 Mutex::Autolock _l(mLock); 8869 mSuspended = suspended; 8870} 8871 8872bool AudioFlinger::EffectModule::suspended() const 8873{ 8874 Mutex::Autolock _l(mLock); 8875 return mSuspended; 8876} 8877 8878bool AudioFlinger::EffectModule::purgeHandles() 8879{ 8880 bool enabled = false; 8881 Mutex::Autolock _l(mLock); 8882 for (size_t i = 0; i < mHandles.size(); i++) { 8883 EffectHandle *handle = mHandles[i]; 8884 if (handle != NULL && !handle->destroyed_l()) { 8885 handle->effect().clear(); 8886 if (handle->hasControl()) { 8887 enabled = handle->enabled(); 8888 } 8889 } 8890 } 8891 return enabled; 8892} 8893 8894void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8895{ 8896 const size_t SIZE = 256; 8897 char buffer[SIZE]; 8898 String8 result; 8899 8900 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8901 result.append(buffer); 8902 8903 bool locked = tryLock(mLock); 8904 // failed to lock - AudioFlinger is probably deadlocked 8905 if (!locked) { 8906 result.append("\t\tCould not lock Fx mutex:\n"); 8907 } 8908 8909 result.append("\t\tSession Status State Engine:\n"); 8910 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8911 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8912 result.append(buffer); 8913 8914 result.append("\t\tDescriptor:\n"); 8915 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8916 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8917 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8918 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8919 result.append(buffer); 8920 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8921 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8922 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8923 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8924 result.append(buffer); 8925 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8926 mDescriptor.apiVersion, 8927 mDescriptor.flags); 8928 result.append(buffer); 8929 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8930 mDescriptor.name); 8931 result.append(buffer); 8932 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8933 mDescriptor.implementor); 8934 result.append(buffer); 8935 8936 result.append("\t\t- Input configuration:\n"); 8937 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8938 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8939 (uint32_t)mConfig.inputCfg.buffer.raw, 8940 mConfig.inputCfg.buffer.frameCount, 8941 mConfig.inputCfg.samplingRate, 8942 mConfig.inputCfg.channels, 8943 mConfig.inputCfg.format); 8944 result.append(buffer); 8945 8946 result.append("\t\t- Output configuration:\n"); 8947 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8948 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8949 (uint32_t)mConfig.outputCfg.buffer.raw, 8950 mConfig.outputCfg.buffer.frameCount, 8951 mConfig.outputCfg.samplingRate, 8952 mConfig.outputCfg.channels, 8953 mConfig.outputCfg.format); 8954 result.append(buffer); 8955 8956 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8957 result.append(buffer); 8958 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8959 for (size_t i = 0; i < mHandles.size(); ++i) { 8960 EffectHandle *handle = mHandles[i]; 8961 if (handle != NULL && !handle->destroyed_l()) { 8962 handle->dump(buffer, SIZE); 8963 result.append(buffer); 8964 } 8965 } 8966 8967 result.append("\n"); 8968 8969 write(fd, result.string(), result.length()); 8970 8971 if (locked) { 8972 mLock.unlock(); 8973 } 8974} 8975 8976// ---------------------------------------------------------------------------- 8977// EffectHandle implementation 8978// ---------------------------------------------------------------------------- 8979 8980#undef LOG_TAG 8981#define LOG_TAG "AudioFlinger::EffectHandle" 8982 8983AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8984 const sp<AudioFlinger::Client>& client, 8985 const sp<IEffectClient>& effectClient, 8986 int32_t priority) 8987 : BnEffect(), 8988 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8989 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8990{ 8991 ALOGV("constructor %p", this); 8992 8993 if (client == 0) { 8994 return; 8995 } 8996 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8997 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8998 if (mCblkMemory != 0) { 8999 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9000 9001 if (mCblk != NULL) { 9002 new(mCblk) effect_param_cblk_t(); 9003 mBuffer = (uint8_t *)mCblk + bufOffset; 9004 } 9005 } else { 9006 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 9007 return; 9008 } 9009} 9010 9011AudioFlinger::EffectHandle::~EffectHandle() 9012{ 9013 ALOGV("Destructor %p", this); 9014 9015 if (mEffect == 0) { 9016 mDestroyed = true; 9017 return; 9018 } 9019 mEffect->lock(); 9020 mDestroyed = true; 9021 mEffect->unlock(); 9022 disconnect(false); 9023} 9024 9025status_t AudioFlinger::EffectHandle::enable() 9026{ 9027 ALOGV("enable %p", this); 9028 if (!mHasControl) return INVALID_OPERATION; 9029 if (mEffect == 0) return DEAD_OBJECT; 9030 9031 if (mEnabled) { 9032 return NO_ERROR; 9033 } 9034 9035 mEnabled = true; 9036 9037 sp<ThreadBase> thread = mEffect->thread().promote(); 9038 if (thread != 0) { 9039 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9040 } 9041 9042 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9043 if (mEffect->suspended()) { 9044 return NO_ERROR; 9045 } 9046 9047 status_t status = mEffect->setEnabled(true); 9048 if (status != NO_ERROR) { 9049 if (thread != 0) { 9050 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9051 } 9052 mEnabled = false; 9053 } 9054 return status; 9055} 9056 9057status_t AudioFlinger::EffectHandle::disable() 9058{ 9059 ALOGV("disable %p", this); 9060 if (!mHasControl) return INVALID_OPERATION; 9061 if (mEffect == 0) return DEAD_OBJECT; 9062 9063 if (!mEnabled) { 9064 return NO_ERROR; 9065 } 9066 mEnabled = false; 9067 9068 if (mEffect->suspended()) { 9069 return NO_ERROR; 9070 } 9071 9072 status_t status = mEffect->setEnabled(false); 9073 9074 sp<ThreadBase> thread = mEffect->thread().promote(); 9075 if (thread != 0) { 9076 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9077 } 9078 9079 return status; 9080} 9081 9082void AudioFlinger::EffectHandle::disconnect() 9083{ 9084 disconnect(true); 9085} 9086 9087void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9088{ 9089 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9090 if (mEffect == 0) { 9091 return; 9092 } 9093 // restore suspended effects if the disconnected handle was enabled and the last one. 9094 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9095 sp<ThreadBase> thread = mEffect->thread().promote(); 9096 if (thread != 0) { 9097 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9098 } 9099 } 9100 9101 // release sp on module => module destructor can be called now 9102 mEffect.clear(); 9103 if (mClient != 0) { 9104 if (mCblk != NULL) { 9105 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9106 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9107 } 9108 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9109 // Client destructor must run with AudioFlinger mutex locked 9110 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9111 mClient.clear(); 9112 } 9113} 9114 9115status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9116 uint32_t cmdSize, 9117 void *pCmdData, 9118 uint32_t *replySize, 9119 void *pReplyData) 9120{ 9121 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9122 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9123 9124 // only get parameter command is permitted for applications not controlling the effect 9125 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9126 return INVALID_OPERATION; 9127 } 9128 if (mEffect == 0) return DEAD_OBJECT; 9129 if (mClient == 0) return INVALID_OPERATION; 9130 9131 // handle commands that are not forwarded transparently to effect engine 9132 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9133 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 9134 // no risk to block the whole media server process or mixer threads is we are stuck here 9135 Mutex::Autolock _l(mCblk->lock); 9136 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9137 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9138 mCblk->serverIndex = 0; 9139 mCblk->clientIndex = 0; 9140 return BAD_VALUE; 9141 } 9142 status_t status = NO_ERROR; 9143 while (mCblk->serverIndex < mCblk->clientIndex) { 9144 int reply; 9145 uint32_t rsize = sizeof(int); 9146 int *p = (int *)(mBuffer + mCblk->serverIndex); 9147 int size = *p++; 9148 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9149 ALOGW("command(): invalid parameter block size"); 9150 break; 9151 } 9152 effect_param_t *param = (effect_param_t *)p; 9153 if (param->psize == 0 || param->vsize == 0) { 9154 ALOGW("command(): null parameter or value size"); 9155 mCblk->serverIndex += size; 9156 continue; 9157 } 9158 uint32_t psize = sizeof(effect_param_t) + 9159 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9160 param->vsize; 9161 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9162 psize, 9163 p, 9164 &rsize, 9165 &reply); 9166 // stop at first error encountered 9167 if (ret != NO_ERROR) { 9168 status = ret; 9169 *(int *)pReplyData = reply; 9170 break; 9171 } else if (reply != NO_ERROR) { 9172 *(int *)pReplyData = reply; 9173 break; 9174 } 9175 mCblk->serverIndex += size; 9176 } 9177 mCblk->serverIndex = 0; 9178 mCblk->clientIndex = 0; 9179 return status; 9180 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9181 *(int *)pReplyData = NO_ERROR; 9182 return enable(); 9183 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9184 *(int *)pReplyData = NO_ERROR; 9185 return disable(); 9186 } 9187 9188 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9189} 9190 9191void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9192{ 9193 ALOGV("setControl %p control %d", this, hasControl); 9194 9195 mHasControl = hasControl; 9196 mEnabled = enabled; 9197 9198 if (signal && mEffectClient != 0) { 9199 mEffectClient->controlStatusChanged(hasControl); 9200 } 9201} 9202 9203void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9204 uint32_t cmdSize, 9205 void *pCmdData, 9206 uint32_t replySize, 9207 void *pReplyData) 9208{ 9209 if (mEffectClient != 0) { 9210 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9211 } 9212} 9213 9214 9215 9216void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9217{ 9218 if (mEffectClient != 0) { 9219 mEffectClient->enableStatusChanged(enabled); 9220 } 9221} 9222 9223status_t AudioFlinger::EffectHandle::onTransact( 9224 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9225{ 9226 return BnEffect::onTransact(code, data, reply, flags); 9227} 9228 9229 9230void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9231{ 9232 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9233 9234 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9235 (mClient == 0) ? getpid_cached : mClient->pid(), 9236 mPriority, 9237 mHasControl, 9238 !locked, 9239 mCblk ? mCblk->clientIndex : 0, 9240 mCblk ? mCblk->serverIndex : 0 9241 ); 9242 9243 if (locked) { 9244 mCblk->lock.unlock(); 9245 } 9246} 9247 9248#undef LOG_TAG 9249#define LOG_TAG "AudioFlinger::EffectChain" 9250 9251AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9252 int sessionId) 9253 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9254 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9255 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9256{ 9257 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9258 if (thread == NULL) { 9259 return; 9260 } 9261 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9262 thread->frameCount(); 9263} 9264 9265AudioFlinger::EffectChain::~EffectChain() 9266{ 9267 if (mOwnInBuffer) { 9268 delete mInBuffer; 9269 } 9270 9271} 9272 9273// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9274sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9275{ 9276 size_t size = mEffects.size(); 9277 9278 for (size_t i = 0; i < size; i++) { 9279 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9280 return mEffects[i]; 9281 } 9282 } 9283 return 0; 9284} 9285 9286// getEffectFromId_l() must be called with ThreadBase::mLock held 9287sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9288{ 9289 size_t size = mEffects.size(); 9290 9291 for (size_t i = 0; i < size; i++) { 9292 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9293 if (id == 0 || mEffects[i]->id() == id) { 9294 return mEffects[i]; 9295 } 9296 } 9297 return 0; 9298} 9299 9300// getEffectFromType_l() must be called with ThreadBase::mLock held 9301sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9302 const effect_uuid_t *type) 9303{ 9304 size_t size = mEffects.size(); 9305 9306 for (size_t i = 0; i < size; i++) { 9307 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9308 return mEffects[i]; 9309 } 9310 } 9311 return 0; 9312} 9313 9314void AudioFlinger::EffectChain::clearInputBuffer() 9315{ 9316 Mutex::Autolock _l(mLock); 9317 sp<ThreadBase> thread = mThread.promote(); 9318 if (thread == 0) { 9319 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9320 return; 9321 } 9322 clearInputBuffer_l(thread); 9323} 9324 9325// Must be called with EffectChain::mLock locked 9326void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9327{ 9328 size_t numSamples = thread->frameCount() * thread->channelCount(); 9329 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9330 9331} 9332 9333// Must be called with EffectChain::mLock locked 9334void AudioFlinger::EffectChain::process_l() 9335{ 9336 sp<ThreadBase> thread = mThread.promote(); 9337 if (thread == 0) { 9338 ALOGW("process_l(): cannot promote mixer thread"); 9339 return; 9340 } 9341 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9342 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9343 // always process effects unless no more tracks are on the session and the effect tail 9344 // has been rendered 9345 bool doProcess = true; 9346 if (!isGlobalSession) { 9347 bool tracksOnSession = (trackCnt() != 0); 9348 9349 if (!tracksOnSession && mTailBufferCount == 0) { 9350 doProcess = false; 9351 } 9352 9353 if (activeTrackCnt() == 0) { 9354 // if no track is active and the effect tail has not been rendered, 9355 // the input buffer must be cleared here as the mixer process will not do it 9356 if (tracksOnSession || mTailBufferCount > 0) { 9357 clearInputBuffer_l(thread); 9358 if (mTailBufferCount > 0) { 9359 mTailBufferCount--; 9360 } 9361 } 9362 } 9363 } 9364 9365 size_t size = mEffects.size(); 9366 if (doProcess) { 9367 for (size_t i = 0; i < size; i++) { 9368 mEffects[i]->process(); 9369 } 9370 } 9371 for (size_t i = 0; i < size; i++) { 9372 mEffects[i]->updateState(); 9373 } 9374} 9375 9376// addEffect_l() must be called with PlaybackThread::mLock held 9377status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9378{ 9379 effect_descriptor_t desc = effect->desc(); 9380 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9381 9382 Mutex::Autolock _l(mLock); 9383 effect->setChain(this); 9384 sp<ThreadBase> thread = mThread.promote(); 9385 if (thread == 0) { 9386 return NO_INIT; 9387 } 9388 effect->setThread(thread); 9389 9390 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9391 // Auxiliary effects are inserted at the beginning of mEffects vector as 9392 // they are processed first and accumulated in chain input buffer 9393 mEffects.insertAt(effect, 0); 9394 9395 // the input buffer for auxiliary effect contains mono samples in 9396 // 32 bit format. This is to avoid saturation in AudoMixer 9397 // accumulation stage. Saturation is done in EffectModule::process() before 9398 // calling the process in effect engine 9399 size_t numSamples = thread->frameCount(); 9400 int32_t *buffer = new int32_t[numSamples]; 9401 memset(buffer, 0, numSamples * sizeof(int32_t)); 9402 effect->setInBuffer((int16_t *)buffer); 9403 // auxiliary effects output samples to chain input buffer for further processing 9404 // by insert effects 9405 effect->setOutBuffer(mInBuffer); 9406 } else { 9407 // Insert effects are inserted at the end of mEffects vector as they are processed 9408 // after track and auxiliary effects. 9409 // Insert effect order as a function of indicated preference: 9410 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9411 // another effect is present 9412 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9413 // last effect claiming first position 9414 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9415 // first effect claiming last position 9416 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9417 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9418 // already present 9419 9420 size_t size = mEffects.size(); 9421 size_t idx_insert = size; 9422 ssize_t idx_insert_first = -1; 9423 ssize_t idx_insert_last = -1; 9424 9425 for (size_t i = 0; i < size; i++) { 9426 effect_descriptor_t d = mEffects[i]->desc(); 9427 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9428 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9429 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9430 // check invalid effect chaining combinations 9431 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9432 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9433 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9434 return INVALID_OPERATION; 9435 } 9436 // remember position of first insert effect and by default 9437 // select this as insert position for new effect 9438 if (idx_insert == size) { 9439 idx_insert = i; 9440 } 9441 // remember position of last insert effect claiming 9442 // first position 9443 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9444 idx_insert_first = i; 9445 } 9446 // remember position of first insert effect claiming 9447 // last position 9448 if (iPref == EFFECT_FLAG_INSERT_LAST && 9449 idx_insert_last == -1) { 9450 idx_insert_last = i; 9451 } 9452 } 9453 } 9454 9455 // modify idx_insert from first position if needed 9456 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9457 if (idx_insert_last != -1) { 9458 idx_insert = idx_insert_last; 9459 } else { 9460 idx_insert = size; 9461 } 9462 } else { 9463 if (idx_insert_first != -1) { 9464 idx_insert = idx_insert_first + 1; 9465 } 9466 } 9467 9468 // always read samples from chain input buffer 9469 effect->setInBuffer(mInBuffer); 9470 9471 // if last effect in the chain, output samples to chain 9472 // output buffer, otherwise to chain input buffer 9473 if (idx_insert == size) { 9474 if (idx_insert != 0) { 9475 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9476 mEffects[idx_insert-1]->configure(); 9477 } 9478 effect->setOutBuffer(mOutBuffer); 9479 } else { 9480 effect->setOutBuffer(mInBuffer); 9481 } 9482 mEffects.insertAt(effect, idx_insert); 9483 9484 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9485 } 9486 effect->configure(); 9487 return NO_ERROR; 9488} 9489 9490// removeEffect_l() must be called with PlaybackThread::mLock held 9491size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9492{ 9493 Mutex::Autolock _l(mLock); 9494 size_t size = mEffects.size(); 9495 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9496 9497 for (size_t i = 0; i < size; i++) { 9498 if (effect == mEffects[i]) { 9499 // calling stop here will remove pre-processing effect from the audio HAL. 9500 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9501 // the middle of a read from audio HAL 9502 if (mEffects[i]->state() == EffectModule::ACTIVE || 9503 mEffects[i]->state() == EffectModule::STOPPING) { 9504 mEffects[i]->stop(); 9505 } 9506 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9507 delete[] effect->inBuffer(); 9508 } else { 9509 if (i == size - 1 && i != 0) { 9510 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9511 mEffects[i - 1]->configure(); 9512 } 9513 } 9514 mEffects.removeAt(i); 9515 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9516 break; 9517 } 9518 } 9519 9520 return mEffects.size(); 9521} 9522 9523// setDevice_l() must be called with PlaybackThread::mLock held 9524void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9525{ 9526 size_t size = mEffects.size(); 9527 for (size_t i = 0; i < size; i++) { 9528 mEffects[i]->setDevice(device); 9529 } 9530} 9531 9532// setMode_l() must be called with PlaybackThread::mLock held 9533void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9534{ 9535 size_t size = mEffects.size(); 9536 for (size_t i = 0; i < size; i++) { 9537 mEffects[i]->setMode(mode); 9538 } 9539} 9540 9541// setAudioSource_l() must be called with PlaybackThread::mLock held 9542void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9543{ 9544 size_t size = mEffects.size(); 9545 for (size_t i = 0; i < size; i++) { 9546 mEffects[i]->setAudioSource(source); 9547 } 9548} 9549 9550// setVolume_l() must be called with PlaybackThread::mLock held 9551bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9552{ 9553 uint32_t newLeft = *left; 9554 uint32_t newRight = *right; 9555 bool hasControl = false; 9556 int ctrlIdx = -1; 9557 size_t size = mEffects.size(); 9558 9559 // first update volume controller 9560 for (size_t i = size; i > 0; i--) { 9561 if (mEffects[i - 1]->isProcessEnabled() && 9562 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9563 ctrlIdx = i - 1; 9564 hasControl = true; 9565 break; 9566 } 9567 } 9568 9569 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9570 if (hasControl) { 9571 *left = mNewLeftVolume; 9572 *right = mNewRightVolume; 9573 } 9574 return hasControl; 9575 } 9576 9577 mVolumeCtrlIdx = ctrlIdx; 9578 mLeftVolume = newLeft; 9579 mRightVolume = newRight; 9580 9581 // second get volume update from volume controller 9582 if (ctrlIdx >= 0) { 9583 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9584 mNewLeftVolume = newLeft; 9585 mNewRightVolume = newRight; 9586 } 9587 // then indicate volume to all other effects in chain. 9588 // Pass altered volume to effects before volume controller 9589 // and requested volume to effects after controller 9590 uint32_t lVol = newLeft; 9591 uint32_t rVol = newRight; 9592 9593 for (size_t i = 0; i < size; i++) { 9594 if ((int)i == ctrlIdx) continue; 9595 // this also works for ctrlIdx == -1 when there is no volume controller 9596 if ((int)i > ctrlIdx) { 9597 lVol = *left; 9598 rVol = *right; 9599 } 9600 mEffects[i]->setVolume(&lVol, &rVol, false); 9601 } 9602 *left = newLeft; 9603 *right = newRight; 9604 9605 return hasControl; 9606} 9607 9608void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9609{ 9610 const size_t SIZE = 256; 9611 char buffer[SIZE]; 9612 String8 result; 9613 9614 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9615 result.append(buffer); 9616 9617 bool locked = tryLock(mLock); 9618 // failed to lock - AudioFlinger is probably deadlocked 9619 if (!locked) { 9620 result.append("\tCould not lock mutex:\n"); 9621 } 9622 9623 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9624 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9625 mEffects.size(), 9626 (uint32_t)mInBuffer, 9627 (uint32_t)mOutBuffer, 9628 mActiveTrackCnt); 9629 result.append(buffer); 9630 write(fd, result.string(), result.size()); 9631 9632 for (size_t i = 0; i < mEffects.size(); ++i) { 9633 sp<EffectModule> effect = mEffects[i]; 9634 if (effect != 0) { 9635 effect->dump(fd, args); 9636 } 9637 } 9638 9639 if (locked) { 9640 mLock.unlock(); 9641 } 9642} 9643 9644// must be called with ThreadBase::mLock held 9645void AudioFlinger::EffectChain::setEffectSuspended_l( 9646 const effect_uuid_t *type, bool suspend) 9647{ 9648 sp<SuspendedEffectDesc> desc; 9649 // use effect type UUID timelow as key as there is no real risk of identical 9650 // timeLow fields among effect type UUIDs. 9651 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9652 if (suspend) { 9653 if (index >= 0) { 9654 desc = mSuspendedEffects.valueAt(index); 9655 } else { 9656 desc = new SuspendedEffectDesc(); 9657 desc->mType = *type; 9658 mSuspendedEffects.add(type->timeLow, desc); 9659 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9660 } 9661 if (desc->mRefCount++ == 0) { 9662 sp<EffectModule> effect = getEffectIfEnabled(type); 9663 if (effect != 0) { 9664 desc->mEffect = effect; 9665 effect->setSuspended(true); 9666 effect->setEnabled(false); 9667 } 9668 } 9669 } else { 9670 if (index < 0) { 9671 return; 9672 } 9673 desc = mSuspendedEffects.valueAt(index); 9674 if (desc->mRefCount <= 0) { 9675 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9676 desc->mRefCount = 1; 9677 } 9678 if (--desc->mRefCount == 0) { 9679 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9680 if (desc->mEffect != 0) { 9681 sp<EffectModule> effect = desc->mEffect.promote(); 9682 if (effect != 0) { 9683 effect->setSuspended(false); 9684 effect->lock(); 9685 EffectHandle *handle = effect->controlHandle_l(); 9686 if (handle != NULL && !handle->destroyed_l()) { 9687 effect->setEnabled_l(handle->enabled()); 9688 } 9689 effect->unlock(); 9690 } 9691 desc->mEffect.clear(); 9692 } 9693 mSuspendedEffects.removeItemsAt(index); 9694 } 9695 } 9696} 9697 9698// must be called with ThreadBase::mLock held 9699void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9700{ 9701 sp<SuspendedEffectDesc> desc; 9702 9703 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9704 if (suspend) { 9705 if (index >= 0) { 9706 desc = mSuspendedEffects.valueAt(index); 9707 } else { 9708 desc = new SuspendedEffectDesc(); 9709 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9710 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9711 } 9712 if (desc->mRefCount++ == 0) { 9713 Vector< sp<EffectModule> > effects; 9714 getSuspendEligibleEffects(effects); 9715 for (size_t i = 0; i < effects.size(); i++) { 9716 setEffectSuspended_l(&effects[i]->desc().type, true); 9717 } 9718 } 9719 } else { 9720 if (index < 0) { 9721 return; 9722 } 9723 desc = mSuspendedEffects.valueAt(index); 9724 if (desc->mRefCount <= 0) { 9725 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9726 desc->mRefCount = 1; 9727 } 9728 if (--desc->mRefCount == 0) { 9729 Vector<const effect_uuid_t *> types; 9730 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9731 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9732 continue; 9733 } 9734 types.add(&mSuspendedEffects.valueAt(i)->mType); 9735 } 9736 for (size_t i = 0; i < types.size(); i++) { 9737 setEffectSuspended_l(types[i], false); 9738 } 9739 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9740 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9741 } 9742 } 9743} 9744 9745 9746// The volume effect is used for automated tests only 9747#ifndef OPENSL_ES_H_ 9748static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9749 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9750const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9751#endif //OPENSL_ES_H_ 9752 9753bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9754{ 9755 // auxiliary effects and visualizer are never suspended on output mix 9756 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9757 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9758 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9759 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9760 return false; 9761 } 9762 return true; 9763} 9764 9765void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9766{ 9767 effects.clear(); 9768 for (size_t i = 0; i < mEffects.size(); i++) { 9769 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9770 effects.add(mEffects[i]); 9771 } 9772 } 9773} 9774 9775sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9776 const effect_uuid_t *type) 9777{ 9778 sp<EffectModule> effect = getEffectFromType_l(type); 9779 return effect != 0 && effect->isEnabled() ? effect : 0; 9780} 9781 9782void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9783 bool enabled) 9784{ 9785 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9786 if (enabled) { 9787 if (index < 0) { 9788 // if the effect is not suspend check if all effects are suspended 9789 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9790 if (index < 0) { 9791 return; 9792 } 9793 if (!isEffectEligibleForSuspend(effect->desc())) { 9794 return; 9795 } 9796 setEffectSuspended_l(&effect->desc().type, enabled); 9797 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9798 if (index < 0) { 9799 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9800 return; 9801 } 9802 } 9803 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9804 effect->desc().type.timeLow); 9805 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9806 // if effect is requested to suspended but was not yet enabled, supend it now. 9807 if (desc->mEffect == 0) { 9808 desc->mEffect = effect; 9809 effect->setEnabled(false); 9810 effect->setSuspended(true); 9811 } 9812 } else { 9813 if (index < 0) { 9814 return; 9815 } 9816 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9817 effect->desc().type.timeLow); 9818 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9819 desc->mEffect.clear(); 9820 effect->setSuspended(false); 9821 } 9822} 9823 9824#undef LOG_TAG 9825#define LOG_TAG "AudioFlinger" 9826 9827// ---------------------------------------------------------------------------- 9828 9829status_t AudioFlinger::onTransact( 9830 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9831{ 9832 return BnAudioFlinger::onTransact(code, data, reply, flags); 9833} 9834 9835}; // namespace android 9836