AudioFlinger.cpp revision b603744e96b07b1d5bf745bde593fb2c025cefcf
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 size_t 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 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 476 // and we don't yet support 8.24 or 32-bit PCM 477 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 478 ALOGE("createTrack() invalid format %d", format); 479 lStatus = BAD_VALUE; 480 goto Exit; 481 } 482 483 { 484 Mutex::Autolock _l(mLock); 485 PlaybackThread *thread = checkPlaybackThread_l(output); 486 PlaybackThread *effectThread = NULL; 487 if (thread == NULL) { 488 ALOGE("unknown output thread"); 489 lStatus = BAD_VALUE; 490 goto Exit; 491 } 492 493 client = registerPid_l(pid); 494 495 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 496 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 497 // check if an effect chain with the same session ID is present on another 498 // output thread and move it here. 499 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 500 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 501 if (mPlaybackThreads.keyAt(i) != output) { 502 uint32_t sessions = t->hasAudioSession(*sessionId); 503 if (sessions & PlaybackThread::EFFECT_SESSION) { 504 effectThread = t.get(); 505 break; 506 } 507 } 508 } 509 lSessionId = *sessionId; 510 } else { 511 // if no audio session id is provided, create one here 512 lSessionId = nextUniqueId(); 513 if (sessionId != NULL) { 514 *sessionId = lSessionId; 515 } 516 } 517 ALOGV("createTrack() lSessionId: %d", lSessionId); 518 519 track = thread->createTrack_l(client, streamType, sampleRate, format, 520 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 521 522 // move effect chain to this output thread if an effect on same session was waiting 523 // for a track to be created 524 if (lStatus == NO_ERROR && effectThread != NULL) { 525 Mutex::Autolock _dl(thread->mLock); 526 Mutex::Autolock _sl(effectThread->mLock); 527 moveEffectChain_l(lSessionId, effectThread, thread, true); 528 } 529 530 // Look for sync events awaiting for a session to be used. 531 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 532 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 533 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 534 if (lStatus == NO_ERROR) { 535 (void) track->setSyncEvent(mPendingSyncEvents[i]); 536 } else { 537 mPendingSyncEvents[i]->cancel(); 538 } 539 mPendingSyncEvents.removeAt(i); 540 i--; 541 } 542 } 543 } 544 } 545 if (lStatus == NO_ERROR) { 546 trackHandle = new TrackHandle(track); 547 } else { 548 // remove local strong reference to Client before deleting the Track so that the Client 549 // destructor is called by the TrackBase destructor with mLock held 550 client.clear(); 551 track.clear(); 552 } 553 554Exit: 555 if (status != NULL) { 556 *status = lStatus; 557 } 558 return trackHandle; 559} 560 561uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 562{ 563 Mutex::Autolock _l(mLock); 564 PlaybackThread *thread = checkPlaybackThread_l(output); 565 if (thread == NULL) { 566 ALOGW("sampleRate() unknown thread %d", output); 567 return 0; 568 } 569 return thread->sampleRate(); 570} 571 572int AudioFlinger::channelCount(audio_io_handle_t output) const 573{ 574 Mutex::Autolock _l(mLock); 575 PlaybackThread *thread = checkPlaybackThread_l(output); 576 if (thread == NULL) { 577 ALOGW("channelCount() unknown thread %d", output); 578 return 0; 579 } 580 return thread->channelCount(); 581} 582 583audio_format_t AudioFlinger::format(audio_io_handle_t output) const 584{ 585 Mutex::Autolock _l(mLock); 586 PlaybackThread *thread = checkPlaybackThread_l(output); 587 if (thread == NULL) { 588 ALOGW("format() unknown thread %d", output); 589 return AUDIO_FORMAT_INVALID; 590 } 591 return thread->format(); 592} 593 594size_t AudioFlinger::frameCount(audio_io_handle_t output) const 595{ 596 Mutex::Autolock _l(mLock); 597 PlaybackThread *thread = checkPlaybackThread_l(output); 598 if (thread == NULL) { 599 ALOGW("frameCount() unknown thread %d", output); 600 return 0; 601 } 602 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 603 // should examine all callers and fix them to handle smaller counts 604 return thread->frameCount(); 605} 606 607uint32_t AudioFlinger::latency(audio_io_handle_t output) const 608{ 609 Mutex::Autolock _l(mLock); 610 PlaybackThread *thread = checkPlaybackThread_l(output); 611 if (thread == NULL) { 612 ALOGW("latency() unknown thread %d", output); 613 return 0; 614 } 615 return thread->latency(); 616} 617 618status_t AudioFlinger::setMasterVolume(float value) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 630 Mutex::Autolock _l(mLock); 631 mMasterVolume = value; 632 633 // Set master volume in the HALs which support it. 634 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 635 AutoMutex lock(mHardwareLock); 636 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 637 638 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 639 if (dev->canSetMasterVolume()) { 640 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 641 } 642 mHardwareStatus = AUDIO_HW_IDLE; 643 } 644 645 // Now set the master volume in each playback thread. Playback threads 646 // assigned to HALs which do not have master volume support will apply 647 // master volume during the mix operation. Threads with HALs which do 648 // support master volume will simply ignore the setting. 649 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 650 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 651 652 return NO_ERROR; 653} 654 655status_t AudioFlinger::setMode(audio_mode_t mode) 656{ 657 status_t ret = initCheck(); 658 if (ret != NO_ERROR) { 659 return ret; 660 } 661 662 // check calling permissions 663 if (!settingsAllowed()) { 664 return PERMISSION_DENIED; 665 } 666 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 667 ALOGW("Illegal value: setMode(%d)", mode); 668 return BAD_VALUE; 669 } 670 671 { // scope for the lock 672 AutoMutex lock(mHardwareLock); 673 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 674 mHardwareStatus = AUDIO_HW_SET_MODE; 675 ret = dev->set_mode(dev, mode); 676 mHardwareStatus = AUDIO_HW_IDLE; 677 } 678 679 if (NO_ERROR == ret) { 680 Mutex::Autolock _l(mLock); 681 mMode = mode; 682 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 683 mPlaybackThreads.valueAt(i)->setMode(mode); 684 } 685 686 return ret; 687} 688 689status_t AudioFlinger::setMicMute(bool state) 690{ 691 status_t ret = initCheck(); 692 if (ret != NO_ERROR) { 693 return ret; 694 } 695 696 // check calling permissions 697 if (!settingsAllowed()) { 698 return PERMISSION_DENIED; 699 } 700 701 AutoMutex lock(mHardwareLock); 702 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 703 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 704 ret = dev->set_mic_mute(dev, state); 705 mHardwareStatus = AUDIO_HW_IDLE; 706 return ret; 707} 708 709bool AudioFlinger::getMicMute() const 710{ 711 status_t ret = initCheck(); 712 if (ret != NO_ERROR) { 713 return false; 714 } 715 716 bool state = AUDIO_MODE_INVALID; 717 AutoMutex lock(mHardwareLock); 718 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 719 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 720 dev->get_mic_mute(dev, &state); 721 mHardwareStatus = AUDIO_HW_IDLE; 722 return state; 723} 724 725status_t AudioFlinger::setMasterMute(bool muted) 726{ 727 status_t ret = initCheck(); 728 if (ret != NO_ERROR) { 729 return ret; 730 } 731 732 // check calling permissions 733 if (!settingsAllowed()) { 734 return PERMISSION_DENIED; 735 } 736 737 Mutex::Autolock _l(mLock); 738 mMasterMute = muted; 739 740 // Set master mute in the HALs which support it. 741 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 742 AutoMutex lock(mHardwareLock); 743 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 744 745 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 746 if (dev->canSetMasterMute()) { 747 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 748 } 749 mHardwareStatus = AUDIO_HW_IDLE; 750 } 751 752 // Now set the master mute in each playback thread. Playback threads 753 // assigned to HALs which do not have master mute support will apply master 754 // mute during the mix operation. Threads with HALs which do support master 755 // mute will simply ignore the setting. 756 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 757 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 758 759 return NO_ERROR; 760} 761 762float AudioFlinger::masterVolume() const 763{ 764 Mutex::Autolock _l(mLock); 765 return masterVolume_l(); 766} 767 768bool AudioFlinger::masterMute() const 769{ 770 Mutex::Autolock _l(mLock); 771 return masterMute_l(); 772} 773 774float AudioFlinger::masterVolume_l() const 775{ 776 return mMasterVolume; 777} 778 779bool AudioFlinger::masterMute_l() const 780{ 781 return mMasterMute; 782} 783 784status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 785 audio_io_handle_t output) 786{ 787 // check calling permissions 788 if (!settingsAllowed()) { 789 return PERMISSION_DENIED; 790 } 791 792 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 793 ALOGE("setStreamVolume() invalid stream %d", stream); 794 return BAD_VALUE; 795 } 796 797 AutoMutex lock(mLock); 798 PlaybackThread *thread = NULL; 799 if (output) { 800 thread = checkPlaybackThread_l(output); 801 if (thread == NULL) { 802 return BAD_VALUE; 803 } 804 } 805 806 mStreamTypes[stream].volume = value; 807 808 if (thread == NULL) { 809 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 810 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 811 } 812 } else { 813 thread->setStreamVolume(stream, value); 814 } 815 816 return NO_ERROR; 817} 818 819status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 820{ 821 // check calling permissions 822 if (!settingsAllowed()) { 823 return PERMISSION_DENIED; 824 } 825 826 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 827 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 828 ALOGE("setStreamMute() invalid stream %d", stream); 829 return BAD_VALUE; 830 } 831 832 AutoMutex lock(mLock); 833 mStreamTypes[stream].mute = muted; 834 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 835 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 836 837 return NO_ERROR; 838} 839 840float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 841{ 842 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 843 return 0.0f; 844 } 845 846 AutoMutex lock(mLock); 847 float volume; 848 if (output) { 849 PlaybackThread *thread = checkPlaybackThread_l(output); 850 if (thread == NULL) { 851 return 0.0f; 852 } 853 volume = thread->streamVolume(stream); 854 } else { 855 volume = streamVolume_l(stream); 856 } 857 858 return volume; 859} 860 861bool AudioFlinger::streamMute(audio_stream_type_t stream) const 862{ 863 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 864 return true; 865 } 866 867 AutoMutex lock(mLock); 868 return streamMute_l(stream); 869} 870 871status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 872{ 873 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 874 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 875 // check calling permissions 876 if (!settingsAllowed()) { 877 return PERMISSION_DENIED; 878 } 879 880 // ioHandle == 0 means the parameters are global to the audio hardware interface 881 if (ioHandle == 0) { 882 Mutex::Autolock _l(mLock); 883 status_t final_result = NO_ERROR; 884 { 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 887 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 888 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 889 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 890 final_result = result ?: final_result; 891 } 892 mHardwareStatus = AUDIO_HW_IDLE; 893 } 894 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 895 AudioParameter param = AudioParameter(keyValuePairs); 896 String8 value; 897 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 898 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 899 if (mBtNrecIsOff != btNrecIsOff) { 900 for (size_t i = 0; i < mRecordThreads.size(); i++) { 901 sp<RecordThread> thread = mRecordThreads.valueAt(i); 902 audio_devices_t device = thread->inDevice(); 903 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 904 // collect all of the thread's session IDs 905 KeyedVector<int, bool> ids = thread->sessionIds(); 906 // suspend effects associated with those session IDs 907 for (size_t j = 0; j < ids.size(); ++j) { 908 int sessionId = ids.keyAt(j); 909 thread->setEffectSuspended(FX_IID_AEC, 910 suspend, 911 sessionId); 912 thread->setEffectSuspended(FX_IID_NS, 913 suspend, 914 sessionId); 915 } 916 } 917 mBtNrecIsOff = btNrecIsOff; 918 } 919 } 920 String8 screenState; 921 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 922 bool isOff = screenState == "off"; 923 if (isOff != (gScreenState & 1)) { 924 gScreenState = ((gScreenState & ~1) + 2) | isOff; 925 } 926 } 927 return final_result; 928 } 929 930 // hold a strong ref on thread in case closeOutput() or closeInput() is called 931 // and the thread is exited once the lock is released 932 sp<ThreadBase> thread; 933 { 934 Mutex::Autolock _l(mLock); 935 thread = checkPlaybackThread_l(ioHandle); 936 if (thread == 0) { 937 thread = checkRecordThread_l(ioHandle); 938 } else if (thread == primaryPlaybackThread_l()) { 939 // indicate output device change to all input threads for pre processing 940 AudioParameter param = AudioParameter(keyValuePairs); 941 int value; 942 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 943 (value != 0)) { 944 for (size_t i = 0; i < mRecordThreads.size(); i++) { 945 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 946 } 947 } 948 } 949 } 950 if (thread != 0) { 951 return thread->setParameters(keyValuePairs); 952 } 953 return BAD_VALUE; 954} 955 956String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 957{ 958 ALOGVV("getParameters() io %d, keys %s, tid %d, calling pid %d", 959 ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 960 961 Mutex::Autolock _l(mLock); 962 963 if (ioHandle == 0) { 964 String8 out_s8; 965 966 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 967 char *s; 968 { 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 971 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 972 s = dev->get_parameters(dev, keys.string()); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 } 975 out_s8 += String8(s ? s : ""); 976 free(s); 977 } 978 return out_s8; 979 } 980 981 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 982 if (playbackThread != NULL) { 983 return playbackThread->getParameters(keys); 984 } 985 RecordThread *recordThread = checkRecordThread_l(ioHandle); 986 if (recordThread != NULL) { 987 return recordThread->getParameters(keys); 988 } 989 return String8(""); 990} 991 992size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 993 audio_channel_mask_t channelMask) const 994{ 995 status_t ret = initCheck(); 996 if (ret != NO_ERROR) { 997 return 0; 998 } 999 1000 AutoMutex lock(mHardwareLock); 1001 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1002 struct audio_config config = { 1003 sample_rate: sampleRate, 1004 channel_mask: channelMask, 1005 format: format, 1006 }; 1007 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1008 size_t size = dev->get_input_buffer_size(dev, &config); 1009 mHardwareStatus = AUDIO_HW_IDLE; 1010 return size; 1011} 1012 1013unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1014{ 1015 Mutex::Autolock _l(mLock); 1016 1017 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1018 if (recordThread != NULL) { 1019 return recordThread->getInputFramesLost(); 1020 } 1021 return 0; 1022} 1023 1024status_t AudioFlinger::setVoiceVolume(float value) 1025{ 1026 status_t ret = initCheck(); 1027 if (ret != NO_ERROR) { 1028 return ret; 1029 } 1030 1031 // check calling permissions 1032 if (!settingsAllowed()) { 1033 return PERMISSION_DENIED; 1034 } 1035 1036 AutoMutex lock(mHardwareLock); 1037 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1038 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1039 ret = dev->set_voice_volume(dev, value); 1040 mHardwareStatus = AUDIO_HW_IDLE; 1041 1042 return ret; 1043} 1044 1045status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1046 audio_io_handle_t output) const 1047{ 1048 status_t status; 1049 1050 Mutex::Autolock _l(mLock); 1051 1052 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1053 if (playbackThread != NULL) { 1054 return playbackThread->getRenderPosition(halFrames, dspFrames); 1055 } 1056 1057 return BAD_VALUE; 1058} 1059 1060void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1061{ 1062 1063 Mutex::Autolock _l(mLock); 1064 1065 pid_t pid = IPCThreadState::self()->getCallingPid(); 1066 if (mNotificationClients.indexOfKey(pid) < 0) { 1067 sp<NotificationClient> notificationClient = new NotificationClient(this, 1068 client, 1069 pid); 1070 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1071 1072 mNotificationClients.add(pid, notificationClient); 1073 1074 sp<IBinder> binder = client->asBinder(); 1075 binder->linkToDeath(notificationClient); 1076 1077 // the config change is always sent from playback or record threads to avoid deadlock 1078 // with AudioSystem::gLock 1079 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1080 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1081 } 1082 1083 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1084 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1085 } 1086 } 1087} 1088 1089void AudioFlinger::removeNotificationClient(pid_t pid) 1090{ 1091 Mutex::Autolock _l(mLock); 1092 1093 mNotificationClients.removeItem(pid); 1094 1095 ALOGV("%d died, releasing its sessions", pid); 1096 size_t num = mAudioSessionRefs.size(); 1097 bool removed = false; 1098 for (size_t i = 0; i< num; ) { 1099 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1100 ALOGV(" pid %d @ %d", ref->mPid, i); 1101 if (ref->mPid == pid) { 1102 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1103 mAudioSessionRefs.removeAt(i); 1104 delete ref; 1105 removed = true; 1106 num--; 1107 } else { 1108 i++; 1109 } 1110 } 1111 if (removed) { 1112 purgeStaleEffects_l(); 1113 } 1114} 1115 1116// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1117void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1118{ 1119 size_t size = mNotificationClients.size(); 1120 for (size_t i = 0; i < size; i++) { 1121 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1122 param2); 1123 } 1124} 1125 1126// removeClient_l() must be called with AudioFlinger::mLock held 1127void AudioFlinger::removeClient_l(pid_t pid) 1128{ 1129 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), 1130 IPCThreadState::self()->getCallingPid()); 1131 mClients.removeItem(pid); 1132} 1133 1134// getEffectThread_l() must be called with AudioFlinger::mLock held 1135sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1136{ 1137 sp<PlaybackThread> thread; 1138 1139 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1140 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1141 ALOG_ASSERT(thread == 0); 1142 thread = mPlaybackThreads.valueAt(i); 1143 } 1144 } 1145 1146 return thread; 1147} 1148 1149// ---------------------------------------------------------------------------- 1150 1151AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1152 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1153 : Thread(false /*canCallJava*/), 1154 mType(type), 1155 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1156 // mChannelMask 1157 mChannelCount(0), 1158 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1159 mParamStatus(NO_ERROR), 1160 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1161 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1162 // mName will be set by concrete (non-virtual) subclass 1163 mDeathRecipient(new PMDeathRecipient(this)) 1164{ 1165} 1166 1167AudioFlinger::ThreadBase::~ThreadBase() 1168{ 1169 mParamCond.broadcast(); 1170 // do not lock the mutex in destructor 1171 releaseWakeLock_l(); 1172 if (mPowerManager != 0) { 1173 sp<IBinder> binder = mPowerManager->asBinder(); 1174 binder->unlinkToDeath(mDeathRecipient); 1175 } 1176} 1177 1178void AudioFlinger::ThreadBase::exit() 1179{ 1180 ALOGV("ThreadBase::exit"); 1181 // do any cleanup required for exit to succeed 1182 preExit(); 1183 { 1184 // This lock prevents the following race in thread (uniprocessor for illustration): 1185 // if (!exitPending()) { 1186 // // context switch from here to exit() 1187 // // exit() calls requestExit(), what exitPending() observes 1188 // // exit() calls signal(), which is dropped since no waiters 1189 // // context switch back from exit() to here 1190 // mWaitWorkCV.wait(...); 1191 // // now thread is hung 1192 // } 1193 AutoMutex lock(mLock); 1194 requestExit(); 1195 mWaitWorkCV.broadcast(); 1196 } 1197 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1198 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1199 requestExitAndWait(); 1200} 1201 1202status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1203{ 1204 status_t status; 1205 1206 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1207 Mutex::Autolock _l(mLock); 1208 1209 mNewParameters.add(keyValuePairs); 1210 mWaitWorkCV.signal(); 1211 // wait condition with timeout in case the thread loop has exited 1212 // before the request could be processed 1213 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1214 status = mParamStatus; 1215 mWaitWorkCV.signal(); 1216 } else { 1217 status = TIMED_OUT; 1218 } 1219 return status; 1220} 1221 1222void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 1223{ 1224 Mutex::Autolock _l(mLock); 1225 sendIoConfigEvent_l(event, param); 1226} 1227 1228// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 1229void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 1230{ 1231 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 1232 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 1233 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 1234 param); 1235 mWaitWorkCV.signal(); 1236} 1237 1238// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 1239void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 1240{ 1241 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 1242 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 1243 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 1244 mConfigEvents.size(), pid, tid, prio); 1245 mWaitWorkCV.signal(); 1246} 1247 1248void AudioFlinger::ThreadBase::processConfigEvents() 1249{ 1250 mLock.lock(); 1251 while (!mConfigEvents.isEmpty()) { 1252 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1253 ConfigEvent *event = mConfigEvents[0]; 1254 mConfigEvents.removeAt(0); 1255 // release mLock before locking AudioFlinger mLock: lock order is always 1256 // AudioFlinger then ThreadBase to avoid cross deadlock 1257 mLock.unlock(); 1258 switch(event->type()) { 1259 case CFG_EVENT_PRIO: { 1260 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 1261 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 1262 if (err != 0) { 1263 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 1264 "error %d", 1265 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 1266 } 1267 } break; 1268 case CFG_EVENT_IO: { 1269 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 1270 mAudioFlinger->mLock.lock(); 1271 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 1272 mAudioFlinger->mLock.unlock(); 1273 } break; 1274 default: 1275 ALOGE("processConfigEvents() unknown event type %d", event->type()); 1276 break; 1277 } 1278 delete event; 1279 mLock.lock(); 1280 } 1281 mLock.unlock(); 1282} 1283 1284void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1285{ 1286 const size_t SIZE = 256; 1287 char buffer[SIZE]; 1288 String8 result; 1289 1290 bool locked = tryLock(mLock); 1291 if (!locked) { 1292 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1293 write(fd, buffer, strlen(buffer)); 1294 } 1295 1296 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1297 result.append(buffer); 1298 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1299 result.append(buffer); 1300 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1301 result.append(buffer); 1302 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 1303 result.append(buffer); 1304 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1305 result.append(buffer); 1306 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1307 result.append(buffer); 1308 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1309 result.append(buffer); 1310 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1311 result.append(buffer); 1312 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1313 result.append(buffer); 1314 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1315 result.append(buffer); 1316 1317 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1318 result.append(buffer); 1319 result.append(" Index Command"); 1320 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1321 snprintf(buffer, SIZE, "\n %02d ", i); 1322 result.append(buffer); 1323 result.append(mNewParameters[i]); 1324 } 1325 1326 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1327 result.append(buffer); 1328 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1329 mConfigEvents[i]->dump(buffer, SIZE); 1330 result.append(buffer); 1331 } 1332 result.append("\n"); 1333 1334 write(fd, result.string(), result.size()); 1335 1336 if (locked) { 1337 mLock.unlock(); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1342{ 1343 const size_t SIZE = 256; 1344 char buffer[SIZE]; 1345 String8 result; 1346 1347 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1348 write(fd, buffer, strlen(buffer)); 1349 1350 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1351 sp<EffectChain> chain = mEffectChains[i]; 1352 if (chain != 0) { 1353 chain->dump(fd, args); 1354 } 1355 } 1356} 1357 1358void AudioFlinger::ThreadBase::acquireWakeLock() 1359{ 1360 Mutex::Autolock _l(mLock); 1361 acquireWakeLock_l(); 1362} 1363 1364void AudioFlinger::ThreadBase::acquireWakeLock_l() 1365{ 1366 if (mPowerManager == 0) { 1367 // use checkService() to avoid blocking if power service is not up yet 1368 sp<IBinder> binder = 1369 defaultServiceManager()->checkService(String16("power")); 1370 if (binder == 0) { 1371 ALOGW("Thread %s cannot connect to the power manager service", mName); 1372 } else { 1373 mPowerManager = interface_cast<IPowerManager>(binder); 1374 binder->linkToDeath(mDeathRecipient); 1375 } 1376 } 1377 if (mPowerManager != 0) { 1378 sp<IBinder> binder = new BBinder(); 1379 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1380 binder, 1381 String16(mName)); 1382 if (status == NO_ERROR) { 1383 mWakeLockToken = binder; 1384 } 1385 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1386 } 1387} 1388 1389void AudioFlinger::ThreadBase::releaseWakeLock() 1390{ 1391 Mutex::Autolock _l(mLock); 1392 releaseWakeLock_l(); 1393} 1394 1395void AudioFlinger::ThreadBase::releaseWakeLock_l() 1396{ 1397 if (mWakeLockToken != 0) { 1398 ALOGV("releaseWakeLock_l() %s", mName); 1399 if (mPowerManager != 0) { 1400 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1401 } 1402 mWakeLockToken.clear(); 1403 } 1404} 1405 1406void AudioFlinger::ThreadBase::clearPowerManager() 1407{ 1408 Mutex::Autolock _l(mLock); 1409 releaseWakeLock_l(); 1410 mPowerManager.clear(); 1411} 1412 1413void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1414{ 1415 sp<ThreadBase> thread = mThread.promote(); 1416 if (thread != 0) { 1417 thread->clearPowerManager(); 1418 } 1419 ALOGW("power manager service died !!!"); 1420} 1421 1422void AudioFlinger::ThreadBase::setEffectSuspended( 1423 const effect_uuid_t *type, bool suspend, int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 setEffectSuspended_l(type, suspend, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::setEffectSuspended_l( 1430 const effect_uuid_t *type, bool suspend, int sessionId) 1431{ 1432 sp<EffectChain> chain = getEffectChain_l(sessionId); 1433 if (chain != 0) { 1434 if (type != NULL) { 1435 chain->setEffectSuspended_l(type, suspend); 1436 } else { 1437 chain->setEffectSuspendedAll_l(suspend); 1438 } 1439 } 1440 1441 updateSuspendedSessions_l(type, suspend, sessionId); 1442} 1443 1444void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1445{ 1446 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1447 if (index < 0) { 1448 return; 1449 } 1450 1451 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1452 mSuspendedSessions.valueAt(index); 1453 1454 for (size_t i = 0; i < sessionEffects.size(); i++) { 1455 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1456 for (int j = 0; j < desc->mRefCount; j++) { 1457 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1458 chain->setEffectSuspendedAll_l(true); 1459 } else { 1460 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1461 desc->mType.timeLow); 1462 chain->setEffectSuspended_l(&desc->mType, true); 1463 } 1464 } 1465 } 1466} 1467 1468void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1469 bool suspend, 1470 int sessionId) 1471{ 1472 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1473 1474 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1475 1476 if (suspend) { 1477 if (index >= 0) { 1478 sessionEffects = mSuspendedSessions.valueAt(index); 1479 } else { 1480 mSuspendedSessions.add(sessionId, sessionEffects); 1481 } 1482 } else { 1483 if (index < 0) { 1484 return; 1485 } 1486 sessionEffects = mSuspendedSessions.valueAt(index); 1487 } 1488 1489 1490 int key = EffectChain::kKeyForSuspendAll; 1491 if (type != NULL) { 1492 key = type->timeLow; 1493 } 1494 index = sessionEffects.indexOfKey(key); 1495 1496 sp<SuspendedSessionDesc> desc; 1497 if (suspend) { 1498 if (index >= 0) { 1499 desc = sessionEffects.valueAt(index); 1500 } else { 1501 desc = new SuspendedSessionDesc(); 1502 if (type != NULL) { 1503 desc->mType = *type; 1504 } 1505 sessionEffects.add(key, desc); 1506 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1507 } 1508 desc->mRefCount++; 1509 } else { 1510 if (index < 0) { 1511 return; 1512 } 1513 desc = sessionEffects.valueAt(index); 1514 if (--desc->mRefCount == 0) { 1515 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1516 sessionEffects.removeItemsAt(index); 1517 if (sessionEffects.isEmpty()) { 1518 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1519 sessionId); 1520 mSuspendedSessions.removeItem(sessionId); 1521 } 1522 } 1523 } 1524 if (!sessionEffects.isEmpty()) { 1525 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1526 } 1527} 1528 1529void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1530 bool enabled, 1531 int sessionId) 1532{ 1533 Mutex::Autolock _l(mLock); 1534 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1535} 1536 1537void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1538 bool enabled, 1539 int sessionId) 1540{ 1541 if (mType != RECORD) { 1542 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1543 // another session. This gives the priority to well behaved effect control panels 1544 // and applications not using global effects. 1545 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1546 // global effects 1547 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1548 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1549 } 1550 } 1551 1552 sp<EffectChain> chain = getEffectChain_l(sessionId); 1553 if (chain != 0) { 1554 chain->checkSuspendOnEffectEnabled(effect, enabled); 1555 } 1556} 1557 1558// ---------------------------------------------------------------------------- 1559 1560AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1561 AudioStreamOut* output, 1562 audio_io_handle_t id, 1563 audio_devices_t device, 1564 type_t type) 1565 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1566 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1567 // mStreamTypes[] initialized in constructor body 1568 mOutput(output), 1569 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1570 mMixerStatus(MIXER_IDLE), 1571 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1572 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1573 mScreenState(gScreenState), 1574 // index 0 is reserved for normal mixer's submix 1575 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1576{ 1577 snprintf(mName, kNameLength, "AudioOut_%X", id); 1578 1579 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1580 // it would be safer to explicitly pass initial masterVolume/masterMute as 1581 // parameter. 1582 // 1583 // If the HAL we are using has support for master volume or master mute, 1584 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1585 // and the mute set to false). 1586 mMasterVolume = audioFlinger->masterVolume_l(); 1587 mMasterMute = audioFlinger->masterMute_l(); 1588 if (mOutput && mOutput->audioHwDev) { 1589 if (mOutput->audioHwDev->canSetMasterVolume()) { 1590 mMasterVolume = 1.0; 1591 } 1592 1593 if (mOutput->audioHwDev->canSetMasterMute()) { 1594 mMasterMute = false; 1595 } 1596 } 1597 1598 readOutputParameters(); 1599 1600 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1601 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1602 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1603 stream = (audio_stream_type_t) (stream + 1)) { 1604 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1605 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1606 } 1607 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1608 // because mAudioFlinger doesn't have one to copy from 1609} 1610 1611AudioFlinger::PlaybackThread::~PlaybackThread() 1612{ 1613 delete [] mMixBuffer; 1614} 1615 1616void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1617{ 1618 dumpInternals(fd, args); 1619 dumpTracks(fd, args); 1620 dumpEffectChains(fd, args); 1621} 1622 1623void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1624{ 1625 const size_t SIZE = 256; 1626 char buffer[SIZE]; 1627 String8 result; 1628 1629 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1630 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1631 const stream_type_t *st = &mStreamTypes[i]; 1632 if (i > 0) { 1633 result.appendFormat(", "); 1634 } 1635 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1636 if (st->mute) { 1637 result.append("M"); 1638 } 1639 } 1640 result.append("\n"); 1641 write(fd, result.string(), result.length()); 1642 result.clear(); 1643 1644 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1645 result.append(buffer); 1646 Track::appendDumpHeader(result); 1647 for (size_t i = 0; i < mTracks.size(); ++i) { 1648 sp<Track> track = mTracks[i]; 1649 if (track != 0) { 1650 track->dump(buffer, SIZE); 1651 result.append(buffer); 1652 } 1653 } 1654 1655 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1656 result.append(buffer); 1657 Track::appendDumpHeader(result); 1658 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1659 sp<Track> track = mActiveTracks[i].promote(); 1660 if (track != 0) { 1661 track->dump(buffer, SIZE); 1662 result.append(buffer); 1663 } 1664 } 1665 write(fd, result.string(), result.size()); 1666 1667 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1668 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1669 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1670 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1671} 1672 1673void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1674{ 1675 const size_t SIZE = 256; 1676 char buffer[SIZE]; 1677 String8 result; 1678 1679 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1680 result.append(buffer); 1681 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1682 ns2ms(systemTime() - mLastWriteTime)); 1683 result.append(buffer); 1684 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1685 result.append(buffer); 1686 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1687 result.append(buffer); 1688 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1689 result.append(buffer); 1690 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1691 result.append(buffer); 1692 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1693 result.append(buffer); 1694 write(fd, result.string(), result.size()); 1695 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1696 1697 dumpBase(fd, args); 1698} 1699 1700// Thread virtuals 1701status_t AudioFlinger::PlaybackThread::readyToRun() 1702{ 1703 status_t status = initCheck(); 1704 if (status == NO_ERROR) { 1705 ALOGI("AudioFlinger's thread %p ready to run", this); 1706 } else { 1707 ALOGE("No working audio driver found."); 1708 } 1709 return status; 1710} 1711 1712void AudioFlinger::PlaybackThread::onFirstRef() 1713{ 1714 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1715} 1716 1717// ThreadBase virtuals 1718void AudioFlinger::PlaybackThread::preExit() 1719{ 1720 ALOGV(" preExit()"); 1721 // FIXME this is using hard-coded strings but in the future, this functionality will be 1722 // converted to use audio HAL extensions required to support tunneling 1723 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1724} 1725 1726// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1727sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1728 const sp<AudioFlinger::Client>& client, 1729 audio_stream_type_t streamType, 1730 uint32_t sampleRate, 1731 audio_format_t format, 1732 audio_channel_mask_t channelMask, 1733 size_t frameCount, 1734 const sp<IMemory>& sharedBuffer, 1735 int sessionId, 1736 IAudioFlinger::track_flags_t *flags, 1737 pid_t tid, 1738 status_t *status) 1739{ 1740 sp<Track> track; 1741 status_t lStatus; 1742 1743 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1744 1745 // client expresses a preference for FAST, but we get the final say 1746 if (*flags & IAudioFlinger::TRACK_FAST) { 1747 if ( 1748 // not timed 1749 (!isTimed) && 1750 // either of these use cases: 1751 ( 1752 // use case 1: shared buffer with any frame count 1753 ( 1754 (sharedBuffer != 0) 1755 ) || 1756 // use case 2: callback handler and frame count is default or at least as large as HAL 1757 ( 1758 (tid != -1) && 1759 ((frameCount == 0) || 1760 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1761 ) 1762 ) && 1763 // PCM data 1764 audio_is_linear_pcm(format) && 1765 // mono or stereo 1766 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1767 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1768#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1769 // hardware sample rate 1770 (sampleRate == mSampleRate) && 1771#endif 1772 // normal mixer has an associated fast mixer 1773 hasFastMixer() && 1774 // there are sufficient fast track slots available 1775 (mFastTrackAvailMask != 0) 1776 // FIXME test that MixerThread for this fast track has a capable output HAL 1777 // FIXME add a permission test also? 1778 ) { 1779 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1780 if (frameCount == 0) { 1781 frameCount = mFrameCount * kFastTrackMultiplier; 1782 } 1783 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1784 frameCount, mFrameCount); 1785 } else { 1786 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1787 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1788 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1789 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1790 audio_is_linear_pcm(format), 1791 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1792 *flags &= ~IAudioFlinger::TRACK_FAST; 1793 // For compatibility with AudioTrack calculation, buffer depth is forced 1794 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1795 // This is probably too conservative, but legacy application code may depend on it. 1796 // If you change this calculation, also review the start threshold which is related. 1797 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1798 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1799 if (minBufCount < 2) { 1800 minBufCount = 2; 1801 } 1802 size_t minFrameCount = mNormalFrameCount * minBufCount; 1803 if (frameCount < minFrameCount) { 1804 frameCount = minFrameCount; 1805 } 1806 } 1807 } 1808 1809 if (mType == DIRECT) { 1810 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1811 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1812 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1813 "for output %p with format %d", 1814 sampleRate, format, channelMask, mOutput, mFormat); 1815 lStatus = BAD_VALUE; 1816 goto Exit; 1817 } 1818 } 1819 } else { 1820 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1821 if (sampleRate > mSampleRate*2) { 1822 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1823 lStatus = BAD_VALUE; 1824 goto Exit; 1825 } 1826 } 1827 1828 lStatus = initCheck(); 1829 if (lStatus != NO_ERROR) { 1830 ALOGE("Audio driver not initialized."); 1831 goto Exit; 1832 } 1833 1834 { // scope for mLock 1835 Mutex::Autolock _l(mLock); 1836 1837 // all tracks in same audio session must share the same routing strategy otherwise 1838 // conflicts will happen when tracks are moved from one output to another by audio policy 1839 // manager 1840 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1841 for (size_t i = 0; i < mTracks.size(); ++i) { 1842 sp<Track> t = mTracks[i]; 1843 if (t != 0 && !t->isOutputTrack()) { 1844 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1845 if (sessionId == t->sessionId() && strategy != actual) { 1846 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1847 strategy, actual); 1848 lStatus = BAD_VALUE; 1849 goto Exit; 1850 } 1851 } 1852 } 1853 1854 if (!isTimed) { 1855 track = new Track(this, client, streamType, sampleRate, format, 1856 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1857 } else { 1858 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1859 channelMask, frameCount, sharedBuffer, sessionId); 1860 } 1861 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1862 lStatus = NO_MEMORY; 1863 goto Exit; 1864 } 1865 mTracks.add(track); 1866 1867 sp<EffectChain> chain = getEffectChain_l(sessionId); 1868 if (chain != 0) { 1869 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1870 track->setMainBuffer(chain->inBuffer()); 1871 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1872 chain->incTrackCnt(); 1873 } 1874 1875 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1876 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1877 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1878 // so ask activity manager to do this on our behalf 1879 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1880 } 1881 } 1882 1883 lStatus = NO_ERROR; 1884 1885Exit: 1886 if (status) { 1887 *status = lStatus; 1888 } 1889 return track; 1890} 1891 1892uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1893{ 1894 if (mFastMixer != NULL) { 1895 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1896 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1897 } 1898 return latency; 1899} 1900 1901uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1902{ 1903 return latency; 1904} 1905 1906uint32_t AudioFlinger::PlaybackThread::latency() const 1907{ 1908 Mutex::Autolock _l(mLock); 1909 return latency_l(); 1910} 1911uint32_t AudioFlinger::PlaybackThread::latency_l() const 1912{ 1913 if (initCheck() == NO_ERROR) { 1914 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1915 } else { 1916 return 0; 1917 } 1918} 1919 1920void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1921{ 1922 Mutex::Autolock _l(mLock); 1923 // Don't apply master volume in SW if our HAL can do it for us. 1924 if (mOutput && mOutput->audioHwDev && 1925 mOutput->audioHwDev->canSetMasterVolume()) { 1926 mMasterVolume = 1.0; 1927 } else { 1928 mMasterVolume = value; 1929 } 1930} 1931 1932void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1933{ 1934 Mutex::Autolock _l(mLock); 1935 // Don't apply master mute in SW if our HAL can do it for us. 1936 if (mOutput && mOutput->audioHwDev && 1937 mOutput->audioHwDev->canSetMasterMute()) { 1938 mMasterMute = false; 1939 } else { 1940 mMasterMute = muted; 1941 } 1942} 1943 1944void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1945{ 1946 Mutex::Autolock _l(mLock); 1947 mStreamTypes[stream].volume = value; 1948} 1949 1950void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1951{ 1952 Mutex::Autolock _l(mLock); 1953 mStreamTypes[stream].mute = muted; 1954} 1955 1956float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1957{ 1958 Mutex::Autolock _l(mLock); 1959 return mStreamTypes[stream].volume; 1960} 1961 1962// addTrack_l() must be called with ThreadBase::mLock held 1963status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1964{ 1965 status_t status = ALREADY_EXISTS; 1966 1967 // set retry count for buffer fill 1968 track->mRetryCount = kMaxTrackStartupRetries; 1969 if (mActiveTracks.indexOf(track) < 0) { 1970 // the track is newly added, make sure it fills up all its 1971 // buffers before playing. This is to ensure the client will 1972 // effectively get the latency it requested. 1973 track->mFillingUpStatus = Track::FS_FILLING; 1974 track->mResetDone = false; 1975 track->mPresentationCompleteFrames = 0; 1976 mActiveTracks.add(track); 1977 if (track->mainBuffer() != mMixBuffer) { 1978 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1979 if (chain != 0) { 1980 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1981 track->sessionId()); 1982 chain->incActiveTrackCnt(); 1983 } 1984 } 1985 1986 status = NO_ERROR; 1987 } 1988 1989 ALOGV("mWaitWorkCV.broadcast"); 1990 mWaitWorkCV.broadcast(); 1991 1992 return status; 1993} 1994 1995// destroyTrack_l() must be called with ThreadBase::mLock held 1996void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1997{ 1998 track->mState = TrackBase::TERMINATED; 1999 // active tracks are removed by threadLoop() 2000 if (mActiveTracks.indexOf(track) < 0) { 2001 removeTrack_l(track); 2002 } 2003} 2004 2005void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2006{ 2007 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2008 mTracks.remove(track); 2009 deleteTrackName_l(track->name()); 2010 // redundant as track is about to be destroyed, for dumpsys only 2011 track->mName = -1; 2012 if (track->isFastTrack()) { 2013 int index = track->mFastIndex; 2014 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2015 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2016 mFastTrackAvailMask |= 1 << index; 2017 // redundant as track is about to be destroyed, for dumpsys only 2018 track->mFastIndex = -1; 2019 } 2020 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2021 if (chain != 0) { 2022 chain->decTrackCnt(); 2023 } 2024} 2025 2026String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2027{ 2028 String8 out_s8 = String8(""); 2029 char *s; 2030 2031 Mutex::Autolock _l(mLock); 2032 if (initCheck() != NO_ERROR) { 2033 return out_s8; 2034 } 2035 2036 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2037 out_s8 = String8(s); 2038 free(s); 2039 return out_s8; 2040} 2041 2042// audioConfigChanged_l() must be called with AudioFlinger::mLock held 2043void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 2044 AudioSystem::OutputDescriptor desc; 2045 void *param2 = NULL; 2046 2047 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 2048 param); 2049 2050 switch (event) { 2051 case AudioSystem::OUTPUT_OPENED: 2052 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2053 desc.channels = mChannelMask; 2054 desc.samplingRate = mSampleRate; 2055 desc.format = mFormat; 2056 desc.frameCount = mNormalFrameCount; // FIXME see 2057 // AudioFlinger::frameCount(audio_io_handle_t) 2058 desc.latency = latency(); 2059 param2 = &desc; 2060 break; 2061 2062 case AudioSystem::STREAM_CONFIG_CHANGED: 2063 param2 = ¶m; 2064 case AudioSystem::OUTPUT_CLOSED: 2065 default: 2066 break; 2067 } 2068 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2069} 2070 2071void AudioFlinger::PlaybackThread::readOutputParameters() 2072{ 2073 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2074 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2075 mChannelCount = (uint16_t)popcount(mChannelMask); 2076 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2077 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2078 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2079 if (mFrameCount & 15) { 2080 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2081 mFrameCount); 2082 } 2083 2084 // Calculate size of normal mix buffer relative to the HAL output buffer size 2085 double multiplier = 1.0; 2086 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2087 kUseFastMixer == FastMixer_Dynamic)) { 2088 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2089 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2090 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2091 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2092 maxNormalFrameCount = maxNormalFrameCount & ~15; 2093 if (maxNormalFrameCount < minNormalFrameCount) { 2094 maxNormalFrameCount = minNormalFrameCount; 2095 } 2096 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2097 if (multiplier <= 1.0) { 2098 multiplier = 1.0; 2099 } else if (multiplier <= 2.0) { 2100 if (2 * mFrameCount <= maxNormalFrameCount) { 2101 multiplier = 2.0; 2102 } else { 2103 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2104 } 2105 } else { 2106 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2107 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 2108 // track, but we sometimes have to do this to satisfy the maximum frame count 2109 // constraint) 2110 // FIXME this rounding up should not be done if no HAL SRC 2111 uint32_t truncMult = (uint32_t) multiplier; 2112 if ((truncMult & 1)) { 2113 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2114 ++truncMult; 2115 } 2116 } 2117 multiplier = (double) truncMult; 2118 } 2119 } 2120 mNormalFrameCount = multiplier * mFrameCount; 2121 // round up to nearest 16 frames to satisfy AudioMixer 2122 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2123 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 2124 mNormalFrameCount); 2125 2126 delete[] mMixBuffer; 2127 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2128 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2129 2130 // force reconfiguration of effect chains and engines to take new buffer size and audio 2131 // parameters into account 2132 // Note that mLock is not held when readOutputParameters() is called from the constructor 2133 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2134 // matter. 2135 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2136 Vector< sp<EffectChain> > effectChains = mEffectChains; 2137 for (size_t i = 0; i < effectChains.size(); i ++) { 2138 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2139 } 2140} 2141 2142 2143status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2144{ 2145 if (halFrames == NULL || dspFrames == NULL) { 2146 return BAD_VALUE; 2147 } 2148 Mutex::Autolock _l(mLock); 2149 if (initCheck() != NO_ERROR) { 2150 return INVALID_OPERATION; 2151 } 2152 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2153 2154 if (isSuspended()) { 2155 // return an estimation of rendered frames when the output is suspended 2156 int32_t frames = mBytesWritten - latency_l(); 2157 if (frames < 0) { 2158 frames = 0; 2159 } 2160 *dspFrames = (uint32_t)frames; 2161 return NO_ERROR; 2162 } else { 2163 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2164 } 2165} 2166 2167uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2168{ 2169 Mutex::Autolock _l(mLock); 2170 uint32_t result = 0; 2171 if (getEffectChain_l(sessionId) != 0) { 2172 result = EFFECT_SESSION; 2173 } 2174 2175 for (size_t i = 0; i < mTracks.size(); ++i) { 2176 sp<Track> track = mTracks[i]; 2177 if (sessionId == track->sessionId() && 2178 !(track->mCblk->flags & CBLK_INVALID)) { 2179 result |= TRACK_SESSION; 2180 break; 2181 } 2182 } 2183 2184 return result; 2185} 2186 2187uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2188{ 2189 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2190 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2191 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2192 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2193 } 2194 for (size_t i = 0; i < mTracks.size(); i++) { 2195 sp<Track> track = mTracks[i]; 2196 if (sessionId == track->sessionId() && 2197 !(track->mCblk->flags & CBLK_INVALID)) { 2198 return AudioSystem::getStrategyForStream(track->streamType()); 2199 } 2200 } 2201 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2202} 2203 2204 2205AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2206{ 2207 Mutex::Autolock _l(mLock); 2208 return mOutput; 2209} 2210 2211AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2212{ 2213 Mutex::Autolock _l(mLock); 2214 AudioStreamOut *output = mOutput; 2215 mOutput = NULL; 2216 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2217 // must push a NULL and wait for ack 2218 mOutputSink.clear(); 2219 mPipeSink.clear(); 2220 mNormalSink.clear(); 2221 return output; 2222} 2223 2224// this method must always be called either with ThreadBase mLock held or inside the thread loop 2225audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2226{ 2227 if (mOutput == NULL) { 2228 return NULL; 2229 } 2230 return &mOutput->stream->common; 2231} 2232 2233uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2234{ 2235 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2236} 2237 2238status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2239{ 2240 if (!isValidSyncEvent(event)) { 2241 return BAD_VALUE; 2242 } 2243 2244 Mutex::Autolock _l(mLock); 2245 2246 for (size_t i = 0; i < mTracks.size(); ++i) { 2247 sp<Track> track = mTracks[i]; 2248 if (event->triggerSession() == track->sessionId()) { 2249 (void) track->setSyncEvent(event); 2250 return NO_ERROR; 2251 } 2252 } 2253 2254 return NAME_NOT_FOUND; 2255} 2256 2257bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2258{ 2259 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2260} 2261 2262void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2263 const Vector< sp<Track> >& tracksToRemove) 2264{ 2265 size_t count = tracksToRemove.size(); 2266 if (CC_UNLIKELY(count)) { 2267 for (size_t i = 0 ; i < count ; i++) { 2268 const sp<Track>& track = tracksToRemove.itemAt(i); 2269 if ((track->sharedBuffer() != 0) && 2270 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2271 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2272 } 2273 } 2274 } 2275 2276} 2277 2278// ---------------------------------------------------------------------------- 2279 2280AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2281 audio_io_handle_t id, audio_devices_t device, type_t type) 2282 : PlaybackThread(audioFlinger, output, id, device, type), 2283 // mAudioMixer below 2284 // mFastMixer below 2285 mFastMixerFutex(0) 2286 // mOutputSink below 2287 // mPipeSink below 2288 // mNormalSink below 2289{ 2290 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2291 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2292 "mFrameCount=%d, mNormalFrameCount=%d", 2293 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2294 mNormalFrameCount); 2295 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2296 2297 // FIXME - Current mixer implementation only supports stereo output 2298 if (mChannelCount != FCC_2) { 2299 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2300 } 2301 2302 // create an NBAIO sink for the HAL output stream, and negotiate 2303 mOutputSink = new AudioStreamOutSink(output->stream); 2304 size_t numCounterOffers = 0; 2305 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2306 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2307 ALOG_ASSERT(index == 0); 2308 2309 // initialize fast mixer depending on configuration 2310 bool initFastMixer; 2311 switch (kUseFastMixer) { 2312 case FastMixer_Never: 2313 initFastMixer = false; 2314 break; 2315 case FastMixer_Always: 2316 initFastMixer = true; 2317 break; 2318 case FastMixer_Static: 2319 case FastMixer_Dynamic: 2320 initFastMixer = mFrameCount < mNormalFrameCount; 2321 break; 2322 } 2323 if (initFastMixer) { 2324 2325 // create a MonoPipe to connect our submix to FastMixer 2326 NBAIO_Format format = mOutputSink->format(); 2327 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2328 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2329 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2330 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2331 const NBAIO_Format offers[1] = {format}; 2332 size_t numCounterOffers = 0; 2333 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2334 ALOG_ASSERT(index == 0); 2335 monoPipe->setAvgFrames((mScreenState & 1) ? 2336 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2337 mPipeSink = monoPipe; 2338 2339#ifdef TEE_SINK_FRAMES 2340 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2341 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2342 numCounterOffers = 0; 2343 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2344 ALOG_ASSERT(index == 0); 2345 mTeeSink = teeSink; 2346 PipeReader *teeSource = new PipeReader(*teeSink); 2347 numCounterOffers = 0; 2348 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2349 ALOG_ASSERT(index == 0); 2350 mTeeSource = teeSource; 2351#endif 2352 2353 // create fast mixer and configure it initially with just one fast track for our submix 2354 mFastMixer = new FastMixer(); 2355 FastMixerStateQueue *sq = mFastMixer->sq(); 2356#ifdef STATE_QUEUE_DUMP 2357 sq->setObserverDump(&mStateQueueObserverDump); 2358 sq->setMutatorDump(&mStateQueueMutatorDump); 2359#endif 2360 FastMixerState *state = sq->begin(); 2361 FastTrack *fastTrack = &state->mFastTracks[0]; 2362 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2363 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2364 fastTrack->mVolumeProvider = NULL; 2365 fastTrack->mGeneration++; 2366 state->mFastTracksGen++; 2367 state->mTrackMask = 1; 2368 // fast mixer will use the HAL output sink 2369 state->mOutputSink = mOutputSink.get(); 2370 state->mOutputSinkGen++; 2371 state->mFrameCount = mFrameCount; 2372 state->mCommand = FastMixerState::COLD_IDLE; 2373 // already done in constructor initialization list 2374 //mFastMixerFutex = 0; 2375 state->mColdFutexAddr = &mFastMixerFutex; 2376 state->mColdGen++; 2377 state->mDumpState = &mFastMixerDumpState; 2378 state->mTeeSink = mTeeSink.get(); 2379 sq->end(); 2380 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2381 2382 // start the fast mixer 2383 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2384 pid_t tid = mFastMixer->getTid(); 2385 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2386 if (err != 0) { 2387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2388 kPriorityFastMixer, getpid_cached, tid, err); 2389 } 2390 2391#ifdef AUDIO_WATCHDOG 2392 // create and start the watchdog 2393 mAudioWatchdog = new AudioWatchdog(); 2394 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2395 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2396 tid = mAudioWatchdog->getTid(); 2397 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2398 if (err != 0) { 2399 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2400 kPriorityFastMixer, getpid_cached, tid, err); 2401 } 2402#endif 2403 2404 } else { 2405 mFastMixer = NULL; 2406 } 2407 2408 switch (kUseFastMixer) { 2409 case FastMixer_Never: 2410 case FastMixer_Dynamic: 2411 mNormalSink = mOutputSink; 2412 break; 2413 case FastMixer_Always: 2414 mNormalSink = mPipeSink; 2415 break; 2416 case FastMixer_Static: 2417 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2418 break; 2419 } 2420} 2421 2422AudioFlinger::MixerThread::~MixerThread() 2423{ 2424 if (mFastMixer != NULL) { 2425 FastMixerStateQueue *sq = mFastMixer->sq(); 2426 FastMixerState *state = sq->begin(); 2427 if (state->mCommand == FastMixerState::COLD_IDLE) { 2428 int32_t old = android_atomic_inc(&mFastMixerFutex); 2429 if (old == -1) { 2430 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2431 } 2432 } 2433 state->mCommand = FastMixerState::EXIT; 2434 sq->end(); 2435 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2436 mFastMixer->join(); 2437 // Though the fast mixer thread has exited, it's state queue is still valid. 2438 // We'll use that extract the final state which contains one remaining fast track 2439 // corresponding to our sub-mix. 2440 state = sq->begin(); 2441 ALOG_ASSERT(state->mTrackMask == 1); 2442 FastTrack *fastTrack = &state->mFastTracks[0]; 2443 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2444 delete fastTrack->mBufferProvider; 2445 sq->end(false /*didModify*/); 2446 delete mFastMixer; 2447#ifdef AUDIO_WATCHDOG 2448 if (mAudioWatchdog != 0) { 2449 mAudioWatchdog->requestExit(); 2450 mAudioWatchdog->requestExitAndWait(); 2451 mAudioWatchdog.clear(); 2452 } 2453#endif 2454 } 2455 delete mAudioMixer; 2456} 2457 2458class CpuStats { 2459public: 2460 CpuStats(); 2461 void sample(const String8 &title); 2462#ifdef DEBUG_CPU_USAGE 2463private: 2464 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2465 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2466 2467 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2468 2469 int mCpuNum; // thread's current CPU number 2470 int mCpukHz; // frequency of thread's current CPU in kHz 2471#endif 2472}; 2473 2474CpuStats::CpuStats() 2475#ifdef DEBUG_CPU_USAGE 2476 : mCpuNum(-1), mCpukHz(-1) 2477#endif 2478{ 2479} 2480 2481void CpuStats::sample(const String8 &title) { 2482#ifdef DEBUG_CPU_USAGE 2483 // get current thread's delta CPU time in wall clock ns 2484 double wcNs; 2485 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2486 2487 // record sample for wall clock statistics 2488 if (valid) { 2489 mWcStats.sample(wcNs); 2490 } 2491 2492 // get the current CPU number 2493 int cpuNum = sched_getcpu(); 2494 2495 // get the current CPU frequency in kHz 2496 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2497 2498 // check if either CPU number or frequency changed 2499 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2500 mCpuNum = cpuNum; 2501 mCpukHz = cpukHz; 2502 // ignore sample for purposes of cycles 2503 valid = false; 2504 } 2505 2506 // if no change in CPU number or frequency, then record sample for cycle statistics 2507 if (valid && mCpukHz > 0) { 2508 double cycles = wcNs * cpukHz * 0.000001; 2509 mHzStats.sample(cycles); 2510 } 2511 2512 unsigned n = mWcStats.n(); 2513 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2514 if ((n & 127) == 1) { 2515 long long elapsed = mCpuUsage.elapsed(); 2516 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2517 double perLoop = elapsed / (double) n; 2518 double perLoop100 = perLoop * 0.01; 2519 double perLoop1k = perLoop * 0.001; 2520 double mean = mWcStats.mean(); 2521 double stddev = mWcStats.stddev(); 2522 double minimum = mWcStats.minimum(); 2523 double maximum = mWcStats.maximum(); 2524 double meanCycles = mHzStats.mean(); 2525 double stddevCycles = mHzStats.stddev(); 2526 double minCycles = mHzStats.minimum(); 2527 double maxCycles = mHzStats.maximum(); 2528 mCpuUsage.resetElapsed(); 2529 mWcStats.reset(); 2530 mHzStats.reset(); 2531 ALOGD("CPU usage for %s over past %.1f secs\n" 2532 " (%u mixer loops at %.1f mean ms per loop):\n" 2533 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2534 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2535 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2536 title.string(), 2537 elapsed * .000000001, n, perLoop * .000001, 2538 mean * .001, 2539 stddev * .001, 2540 minimum * .001, 2541 maximum * .001, 2542 mean / perLoop100, 2543 stddev / perLoop100, 2544 minimum / perLoop100, 2545 maximum / perLoop100, 2546 meanCycles / perLoop1k, 2547 stddevCycles / perLoop1k, 2548 minCycles / perLoop1k, 2549 maxCycles / perLoop1k); 2550 2551 } 2552 } 2553#endif 2554}; 2555 2556void AudioFlinger::PlaybackThread::checkSilentMode_l() 2557{ 2558 if (!mMasterMute) { 2559 char value[PROPERTY_VALUE_MAX]; 2560 if (property_get("ro.audio.silent", value, "0") > 0) { 2561 char *endptr; 2562 unsigned long ul = strtoul(value, &endptr, 0); 2563 if (*endptr == '\0' && ul != 0) { 2564 ALOGD("Silence is golden"); 2565 // The setprop command will not allow a property to be changed after 2566 // the first time it is set, so we don't have to worry about un-muting. 2567 setMasterMute_l(true); 2568 } 2569 } 2570 } 2571} 2572 2573bool AudioFlinger::PlaybackThread::threadLoop() 2574{ 2575 Vector< sp<Track> > tracksToRemove; 2576 2577 standbyTime = systemTime(); 2578 2579 // MIXER 2580 nsecs_t lastWarning = 0; 2581 2582 // DUPLICATING 2583 // FIXME could this be made local to while loop? 2584 writeFrames = 0; 2585 2586 cacheParameters_l(); 2587 sleepTime = idleSleepTime; 2588 2589 if (mType == MIXER) { 2590 sleepTimeShift = 0; 2591 } 2592 2593 CpuStats cpuStats; 2594 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2595 2596 acquireWakeLock(); 2597 2598 while (!exitPending()) 2599 { 2600 cpuStats.sample(myName); 2601 2602 Vector< sp<EffectChain> > effectChains; 2603 2604 processConfigEvents(); 2605 2606 { // scope for mLock 2607 2608 Mutex::Autolock _l(mLock); 2609 2610 if (checkForNewParameters_l()) { 2611 cacheParameters_l(); 2612 } 2613 2614 saveOutputTracks(); 2615 2616 // put audio hardware into standby after short delay 2617 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2618 isSuspended())) { 2619 if (!mStandby) { 2620 2621 threadLoop_standby(); 2622 2623 mStandby = true; 2624 } 2625 2626 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2627 // we're about to wait, flush the binder command buffer 2628 IPCThreadState::self()->flushCommands(); 2629 2630 clearOutputTracks(); 2631 2632 if (exitPending()) break; 2633 2634 releaseWakeLock_l(); 2635 // wait until we have something to do... 2636 ALOGV("%s going to sleep", myName.string()); 2637 mWaitWorkCV.wait(mLock); 2638 ALOGV("%s waking up", myName.string()); 2639 acquireWakeLock_l(); 2640 2641 mMixerStatus = MIXER_IDLE; 2642 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2643 mBytesWritten = 0; 2644 2645 checkSilentMode_l(); 2646 2647 standbyTime = systemTime() + standbyDelay; 2648 sleepTime = idleSleepTime; 2649 if (mType == MIXER) { 2650 sleepTimeShift = 0; 2651 } 2652 2653 continue; 2654 } 2655 } 2656 2657 // mMixerStatusIgnoringFastTracks is also updated internally 2658 mMixerStatus = prepareTracks_l(&tracksToRemove); 2659 2660 // prevent any changes in effect chain list and in each effect chain 2661 // during mixing and effect process as the audio buffers could be deleted 2662 // or modified if an effect is created or deleted 2663 lockEffectChains_l(effectChains); 2664 } 2665 2666 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2667 threadLoop_mix(); 2668 } else { 2669 threadLoop_sleepTime(); 2670 } 2671 2672 if (isSuspended()) { 2673 sleepTime = suspendSleepTimeUs(); 2674 mBytesWritten += mixBufferSize; 2675 } 2676 2677 // only process effects if we're going to write 2678 if (sleepTime == 0) { 2679 for (size_t i = 0; i < effectChains.size(); i ++) { 2680 effectChains[i]->process_l(); 2681 } 2682 } 2683 2684 // enable changes in effect chain 2685 unlockEffectChains(effectChains); 2686 2687 // sleepTime == 0 means we must write to audio hardware 2688 if (sleepTime == 0) { 2689 2690 threadLoop_write(); 2691 2692if (mType == MIXER) { 2693 // write blocked detection 2694 nsecs_t now = systemTime(); 2695 nsecs_t delta = now - mLastWriteTime; 2696 if (!mStandby && delta > maxPeriod) { 2697 mNumDelayedWrites++; 2698 if ((now - lastWarning) > kWarningThrottleNs) { 2699#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2700 ScopedTrace st(ATRACE_TAG, "underrun"); 2701#endif 2702 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2703 ns2ms(delta), mNumDelayedWrites, this); 2704 lastWarning = now; 2705 } 2706 } 2707} 2708 2709 mStandby = false; 2710 } else { 2711 usleep(sleepTime); 2712 } 2713 2714 // Finally let go of removed track(s), without the lock held 2715 // since we can't guarantee the destructors won't acquire that 2716 // same lock. This will also mutate and push a new fast mixer state. 2717 threadLoop_removeTracks(tracksToRemove); 2718 tracksToRemove.clear(); 2719 2720 // FIXME I don't understand the need for this here; 2721 // it was in the original code but maybe the 2722 // assignment in saveOutputTracks() makes this unnecessary? 2723 clearOutputTracks(); 2724 2725 // Effect chains will be actually deleted here if they were removed from 2726 // mEffectChains list during mixing or effects processing 2727 effectChains.clear(); 2728 2729 // FIXME Note that the above .clear() is no longer necessary since effectChains 2730 // is now local to this block, but will keep it for now (at least until merge done). 2731 } 2732 2733 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2734 if (mType == MIXER || mType == DIRECT) { 2735 // put output stream into standby mode 2736 if (!mStandby) { 2737 mOutput->stream->common.standby(&mOutput->stream->common); 2738 } 2739 } 2740 2741 releaseWakeLock(); 2742 2743 ALOGV("Thread %p type %d exiting", this, mType); 2744 return false; 2745} 2746 2747void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2748{ 2749 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2750} 2751 2752void AudioFlinger::MixerThread::threadLoop_write() 2753{ 2754 // FIXME we should only do one push per cycle; confirm this is true 2755 // Start the fast mixer if it's not already running 2756 if (mFastMixer != NULL) { 2757 FastMixerStateQueue *sq = mFastMixer->sq(); 2758 FastMixerState *state = sq->begin(); 2759 if (state->mCommand != FastMixerState::MIX_WRITE && 2760 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2761 if (state->mCommand == FastMixerState::COLD_IDLE) { 2762 int32_t old = android_atomic_inc(&mFastMixerFutex); 2763 if (old == -1) { 2764 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2765 } 2766#ifdef AUDIO_WATCHDOG 2767 if (mAudioWatchdog != 0) { 2768 mAudioWatchdog->resume(); 2769 } 2770#endif 2771 } 2772 state->mCommand = FastMixerState::MIX_WRITE; 2773 sq->end(); 2774 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2775 if (kUseFastMixer == FastMixer_Dynamic) { 2776 mNormalSink = mPipeSink; 2777 } 2778 } else { 2779 sq->end(false /*didModify*/); 2780 } 2781 } 2782 PlaybackThread::threadLoop_write(); 2783} 2784 2785// shared by MIXER and DIRECT, overridden by DUPLICATING 2786void AudioFlinger::PlaybackThread::threadLoop_write() 2787{ 2788 // FIXME rewrite to reduce number of system calls 2789 mLastWriteTime = systemTime(); 2790 mInWrite = true; 2791 int bytesWritten; 2792 2793 // If an NBAIO sink is present, use it to write the normal mixer's submix 2794 if (mNormalSink != 0) { 2795#define mBitShift 2 // FIXME 2796 size_t count = mixBufferSize >> mBitShift; 2797#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2798 Tracer::traceBegin(ATRACE_TAG, "write"); 2799#endif 2800 // update the setpoint when gScreenState changes 2801 uint32_t screenState = gScreenState; 2802 if (screenState != mScreenState) { 2803 mScreenState = screenState; 2804 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2805 if (pipe != NULL) { 2806 pipe->setAvgFrames((mScreenState & 1) ? 2807 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2808 } 2809 } 2810 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2811#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2812 Tracer::traceEnd(ATRACE_TAG); 2813#endif 2814 if (framesWritten > 0) { 2815 bytesWritten = framesWritten << mBitShift; 2816 } else { 2817 bytesWritten = framesWritten; 2818 } 2819 // otherwise use the HAL / AudioStreamOut directly 2820 } else { 2821 // Direct output thread. 2822 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2823 } 2824 2825 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2826 mNumWrites++; 2827 mInWrite = false; 2828} 2829 2830void AudioFlinger::MixerThread::threadLoop_standby() 2831{ 2832 // Idle the fast mixer if it's currently running 2833 if (mFastMixer != NULL) { 2834 FastMixerStateQueue *sq = mFastMixer->sq(); 2835 FastMixerState *state = sq->begin(); 2836 if (!(state->mCommand & FastMixerState::IDLE)) { 2837 state->mCommand = FastMixerState::COLD_IDLE; 2838 state->mColdFutexAddr = &mFastMixerFutex; 2839 state->mColdGen++; 2840 mFastMixerFutex = 0; 2841 sq->end(); 2842 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2843 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2844 if (kUseFastMixer == FastMixer_Dynamic) { 2845 mNormalSink = mOutputSink; 2846 } 2847#ifdef AUDIO_WATCHDOG 2848 if (mAudioWatchdog != 0) { 2849 mAudioWatchdog->pause(); 2850 } 2851#endif 2852 } else { 2853 sq->end(false /*didModify*/); 2854 } 2855 } 2856 PlaybackThread::threadLoop_standby(); 2857} 2858 2859// shared by MIXER and DIRECT, overridden by DUPLICATING 2860void AudioFlinger::PlaybackThread::threadLoop_standby() 2861{ 2862 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2863 mOutput->stream->common.standby(&mOutput->stream->common); 2864} 2865 2866void AudioFlinger::MixerThread::threadLoop_mix() 2867{ 2868 // obtain the presentation timestamp of the next output buffer 2869 int64_t pts; 2870 status_t status = INVALID_OPERATION; 2871 2872 if (mNormalSink != 0) { 2873 status = mNormalSink->getNextWriteTimestamp(&pts); 2874 } else { 2875 status = mOutputSink->getNextWriteTimestamp(&pts); 2876 } 2877 2878 if (status != NO_ERROR) { 2879 pts = AudioBufferProvider::kInvalidPTS; 2880 } 2881 2882 // mix buffers... 2883 mAudioMixer->process(pts); 2884 // increase sleep time progressively when application underrun condition clears. 2885 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2886 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2887 // such that we would underrun the audio HAL. 2888 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2889 sleepTimeShift--; 2890 } 2891 sleepTime = 0; 2892 standbyTime = systemTime() + standbyDelay; 2893 //TODO: delay standby when effects have a tail 2894} 2895 2896void AudioFlinger::MixerThread::threadLoop_sleepTime() 2897{ 2898 // If no tracks are ready, sleep once for the duration of an output 2899 // buffer size, then write 0s to the output 2900 if (sleepTime == 0) { 2901 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2902 sleepTime = activeSleepTime >> sleepTimeShift; 2903 if (sleepTime < kMinThreadSleepTimeUs) { 2904 sleepTime = kMinThreadSleepTimeUs; 2905 } 2906 // reduce sleep time in case of consecutive application underruns to avoid 2907 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2908 // duration we would end up writing less data than needed by the audio HAL if 2909 // the condition persists. 2910 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2911 sleepTimeShift++; 2912 } 2913 } else { 2914 sleepTime = idleSleepTime; 2915 } 2916 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2917 memset (mMixBuffer, 0, mixBufferSize); 2918 sleepTime = 0; 2919 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), 2920 "anticipated start"); 2921 } 2922 // TODO add standby time extension fct of effect tail 2923} 2924 2925// prepareTracks_l() must be called with ThreadBase::mLock held 2926AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2927 Vector< sp<Track> > *tracksToRemove) 2928{ 2929 2930 mixer_state mixerStatus = MIXER_IDLE; 2931 // find out which tracks need to be processed 2932 size_t count = mActiveTracks.size(); 2933 size_t mixedTracks = 0; 2934 size_t tracksWithEffect = 0; 2935 // counts only _active_ fast tracks 2936 size_t fastTracks = 0; 2937 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2938 2939 float masterVolume = mMasterVolume; 2940 bool masterMute = mMasterMute; 2941 2942 if (masterMute) { 2943 masterVolume = 0; 2944 } 2945 // Delegate master volume control to effect in output mix effect chain if needed 2946 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2947 if (chain != 0) { 2948 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2949 chain->setVolume_l(&v, &v); 2950 masterVolume = (float)((v + (1 << 23)) >> 24); 2951 chain.clear(); 2952 } 2953 2954 // prepare a new state to push 2955 FastMixerStateQueue *sq = NULL; 2956 FastMixerState *state = NULL; 2957 bool didModify = false; 2958 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2959 if (mFastMixer != NULL) { 2960 sq = mFastMixer->sq(); 2961 state = sq->begin(); 2962 } 2963 2964 for (size_t i=0 ; i<count ; i++) { 2965 sp<Track> t = mActiveTracks[i].promote(); 2966 if (t == 0) continue; 2967 2968 // this const just means the local variable doesn't change 2969 Track* const track = t.get(); 2970 2971 // process fast tracks 2972 if (track->isFastTrack()) { 2973 2974 // It's theoretically possible (though unlikely) for a fast track to be created 2975 // and then removed within the same normal mix cycle. This is not a problem, as 2976 // the track never becomes active so it's fast mixer slot is never touched. 2977 // The converse, of removing an (active) track and then creating a new track 2978 // at the identical fast mixer slot within the same normal mix cycle, 2979 // is impossible because the slot isn't marked available until the end of each cycle. 2980 int j = track->mFastIndex; 2981 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2982 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2983 FastTrack *fastTrack = &state->mFastTracks[j]; 2984 2985 // Determine whether the track is currently in underrun condition, 2986 // and whether it had a recent underrun. 2987 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2988 FastTrackUnderruns underruns = ftDump->mUnderruns; 2989 uint32_t recentFull = (underruns.mBitFields.mFull - 2990 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2991 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2992 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2993 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2994 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2995 uint32_t recentUnderruns = recentPartial + recentEmpty; 2996 track->mObservedUnderruns = underruns; 2997 // don't count underruns that occur while stopping or pausing 2998 // or stopped which can occur when flush() is called while active 2999 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 3000 track->mUnderrunCount += recentUnderruns; 3001 } 3002 3003 // This is similar to the state machine for normal tracks, 3004 // with a few modifications for fast tracks. 3005 bool isActive = true; 3006 switch (track->mState) { 3007 case TrackBase::STOPPING_1: 3008 // track stays active in STOPPING_1 state until first underrun 3009 if (recentUnderruns > 0) { 3010 track->mState = TrackBase::STOPPING_2; 3011 } 3012 break; 3013 case TrackBase::PAUSING: 3014 // ramp down is not yet implemented 3015 track->setPaused(); 3016 break; 3017 case TrackBase::RESUMING: 3018 // ramp up is not yet implemented 3019 track->mState = TrackBase::ACTIVE; 3020 break; 3021 case TrackBase::ACTIVE: 3022 if (recentFull > 0 || recentPartial > 0) { 3023 // track has provided at least some frames recently: reset retry count 3024 track->mRetryCount = kMaxTrackRetries; 3025 } 3026 if (recentUnderruns == 0) { 3027 // no recent underruns: stay active 3028 break; 3029 } 3030 // there has recently been an underrun of some kind 3031 if (track->sharedBuffer() == 0) { 3032 // were any of the recent underruns "empty" (no frames available)? 3033 if (recentEmpty == 0) { 3034 // no, then ignore the partial underruns as they are allowed indefinitely 3035 break; 3036 } 3037 // there has recently been an "empty" underrun: decrement the retry counter 3038 if (--(track->mRetryCount) > 0) { 3039 break; 3040 } 3041 // indicate to client process that the track was disabled because of underrun; 3042 // it will then automatically call start() when data is available 3043 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 3044 // remove from active list, but state remains ACTIVE [confusing but true] 3045 isActive = false; 3046 break; 3047 } 3048 // fall through 3049 case TrackBase::STOPPING_2: 3050 case TrackBase::PAUSED: 3051 case TrackBase::TERMINATED: 3052 case TrackBase::STOPPED: 3053 case TrackBase::FLUSHED: // flush() while active 3054 // Check for presentation complete if track is inactive 3055 // We have consumed all the buffers of this track. 3056 // This would be incomplete if we auto-paused on underrun 3057 { 3058 size_t audioHALFrames = 3059 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3060 size_t framesWritten = 3061 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3062 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3063 // track stays in active list until presentation is complete 3064 break; 3065 } 3066 } 3067 if (track->isStopping_2()) { 3068 track->mState = TrackBase::STOPPED; 3069 } 3070 if (track->isStopped()) { 3071 // Can't reset directly, as fast mixer is still polling this track 3072 // track->reset(); 3073 // So instead mark this track as needing to be reset after push with ack 3074 resetMask |= 1 << i; 3075 } 3076 isActive = false; 3077 break; 3078 case TrackBase::IDLE: 3079 default: 3080 LOG_FATAL("unexpected track state %d", track->mState); 3081 } 3082 3083 if (isActive) { 3084 // was it previously inactive? 3085 if (!(state->mTrackMask & (1 << j))) { 3086 ExtendedAudioBufferProvider *eabp = track; 3087 VolumeProvider *vp = track; 3088 fastTrack->mBufferProvider = eabp; 3089 fastTrack->mVolumeProvider = vp; 3090 fastTrack->mSampleRate = track->mSampleRate; 3091 fastTrack->mChannelMask = track->mChannelMask; 3092 fastTrack->mGeneration++; 3093 state->mTrackMask |= 1 << j; 3094 didModify = true; 3095 // no acknowledgement required for newly active tracks 3096 } 3097 // cache the combined master volume and stream type volume for fast mixer; this 3098 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3099 track->mCachedVolume = track->isMuted() ? 3100 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3101 ++fastTracks; 3102 } else { 3103 // was it previously active? 3104 if (state->mTrackMask & (1 << j)) { 3105 fastTrack->mBufferProvider = NULL; 3106 fastTrack->mGeneration++; 3107 state->mTrackMask &= ~(1 << j); 3108 didModify = true; 3109 // If any fast tracks were removed, we must wait for acknowledgement 3110 // because we're about to decrement the last sp<> on those tracks. 3111 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3112 } else { 3113 LOG_FATAL("fast track %d should have been active", j); 3114 } 3115 tracksToRemove->add(track); 3116 // Avoids a misleading display in dumpsys 3117 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3118 } 3119 continue; 3120 } 3121 3122 { // local variable scope to avoid goto warning 3123 3124 audio_track_cblk_t* cblk = track->cblk(); 3125 3126 // The first time a track is added we wait 3127 // for all its buffers to be filled before processing it 3128 int name = track->name(); 3129 // make sure that we have enough frames to mix one full buffer. 3130 // enforce this condition only once to enable draining the buffer in case the client 3131 // app does not call stop() and relies on underrun to stop: 3132 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3133 // during last round 3134 uint32_t minFrames = 1; 3135 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3136 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3137 if (t->sampleRate() == mSampleRate) { 3138 minFrames = mNormalFrameCount; 3139 } else { 3140 // +1 for rounding and +1 for additional sample needed for interpolation 3141 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3142 // add frames already consumed but not yet released by the resampler 3143 // because cblk->framesReady() will include these frames 3144 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3145 // the minimum track buffer size is normally twice the number of frames necessary 3146 // to fill one buffer and the resampler should not leave more than one buffer worth 3147 // of unreleased frames after each pass, but just in case... 3148 ALOG_ASSERT(minFrames <= cblk->frameCount); 3149 } 3150 } 3151 if ((track->framesReady() >= minFrames) && track->isReady() && 3152 !track->isPaused() && !track->isTerminated()) 3153 { 3154 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 3155 this); 3156 3157 mixedTracks++; 3158 3159 // track->mainBuffer() != mMixBuffer means there is an effect chain 3160 // connected to the track 3161 chain.clear(); 3162 if (track->mainBuffer() != mMixBuffer) { 3163 chain = getEffectChain_l(track->sessionId()); 3164 // Delegate volume control to effect in track effect chain if needed 3165 if (chain != 0) { 3166 tracksWithEffect++; 3167 } else { 3168 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3169 "session %d", 3170 name, track->sessionId()); 3171 } 3172 } 3173 3174 3175 int param = AudioMixer::VOLUME; 3176 if (track->mFillingUpStatus == Track::FS_FILLED) { 3177 // no ramp for the first volume setting 3178 track->mFillingUpStatus = Track::FS_ACTIVE; 3179 if (track->mState == TrackBase::RESUMING) { 3180 track->mState = TrackBase::ACTIVE; 3181 param = AudioMixer::RAMP_VOLUME; 3182 } 3183 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3184 } else if (cblk->server != 0) { 3185 // If the track is stopped before the first frame was mixed, 3186 // do not apply ramp 3187 param = AudioMixer::RAMP_VOLUME; 3188 } 3189 3190 // compute volume for this track 3191 uint32_t vl, vr, va; 3192 if (track->isMuted() || track->isPausing() || 3193 mStreamTypes[track->streamType()].mute) { 3194 vl = vr = va = 0; 3195 if (track->isPausing()) { 3196 track->setPaused(); 3197 } 3198 } else { 3199 3200 // read original volumes with volume control 3201 float typeVolume = mStreamTypes[track->streamType()].volume; 3202 float v = masterVolume * typeVolume; 3203 uint32_t vlr = cblk->getVolumeLR(); 3204 vl = vlr & 0xFFFF; 3205 vr = vlr >> 16; 3206 // track volumes come from shared memory, so can't be trusted and must be clamped 3207 if (vl > MAX_GAIN_INT) { 3208 ALOGV("Track left volume out of range: %04X", vl); 3209 vl = MAX_GAIN_INT; 3210 } 3211 if (vr > MAX_GAIN_INT) { 3212 ALOGV("Track right volume out of range: %04X", vr); 3213 vr = MAX_GAIN_INT; 3214 } 3215 // now apply the master volume and stream type volume 3216 vl = (uint32_t)(v * vl) << 12; 3217 vr = (uint32_t)(v * vr) << 12; 3218 // assuming master volume and stream type volume each go up to 1.0, 3219 // vl and vr are now in 8.24 format 3220 3221 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3222 // send level comes from shared memory and so may be corrupt 3223 if (sendLevel > MAX_GAIN_INT) { 3224 ALOGV("Track send level out of range: %04X", sendLevel); 3225 sendLevel = MAX_GAIN_INT; 3226 } 3227 va = (uint32_t)(v * sendLevel); 3228 } 3229 // Delegate volume control to effect in track effect chain if needed 3230 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3231 // Do not ramp volume if volume is controlled by effect 3232 param = AudioMixer::VOLUME; 3233 track->mHasVolumeController = true; 3234 } else { 3235 // force no volume ramp when volume controller was just disabled or removed 3236 // from effect chain to avoid volume spike 3237 if (track->mHasVolumeController) { 3238 param = AudioMixer::VOLUME; 3239 } 3240 track->mHasVolumeController = false; 3241 } 3242 3243 // Convert volumes from 8.24 to 4.12 format 3244 // This additional clamping is needed in case chain->setVolume_l() overshot 3245 vl = (vl + (1 << 11)) >> 12; 3246 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3247 vr = (vr + (1 << 11)) >> 12; 3248 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3249 3250 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3251 3252 // XXX: these things DON'T need to be done each time 3253 mAudioMixer->setBufferProvider(name, track); 3254 mAudioMixer->enable(name); 3255 3256 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3257 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3258 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3259 mAudioMixer->setParameter( 3260 name, 3261 AudioMixer::TRACK, 3262 AudioMixer::FORMAT, (void *)track->format()); 3263 mAudioMixer->setParameter( 3264 name, 3265 AudioMixer::TRACK, 3266 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3267 mAudioMixer->setParameter( 3268 name, 3269 AudioMixer::RESAMPLE, 3270 AudioMixer::SAMPLE_RATE, 3271 (void *)(cblk->sampleRate)); 3272 mAudioMixer->setParameter( 3273 name, 3274 AudioMixer::TRACK, 3275 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3276 mAudioMixer->setParameter( 3277 name, 3278 AudioMixer::TRACK, 3279 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3280 3281 // reset retry count 3282 track->mRetryCount = kMaxTrackRetries; 3283 3284 // If one track is ready, set the mixer ready if: 3285 // - the mixer was not ready during previous round OR 3286 // - no other track is not ready 3287 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3288 mixerStatus != MIXER_TRACKS_ENABLED) { 3289 mixerStatus = MIXER_TRACKS_READY; 3290 } 3291 } else { 3292 // clear effect chain input buffer if an active track underruns to avoid sending 3293 // previous audio buffer again to effects 3294 chain = getEffectChain_l(track->sessionId()); 3295 if (chain != 0) { 3296 chain->clearInputBuffer(); 3297 } 3298 3299 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 3300 cblk->server, this); 3301 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3302 track->isStopped() || track->isPaused()) { 3303 // We have consumed all the buffers of this track. 3304 // Remove it from the list of active tracks. 3305 // TODO: use actual buffer filling status instead of latency when available from 3306 // audio HAL 3307 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3308 size_t framesWritten = 3309 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3310 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3311 if (track->isStopped()) { 3312 track->reset(); 3313 } 3314 tracksToRemove->add(track); 3315 } 3316 } else { 3317 track->mUnderrunCount++; 3318 // No buffers for this track. Give it a few chances to 3319 // fill a buffer, then remove it from active list. 3320 if (--(track->mRetryCount) <= 0) { 3321 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3322 tracksToRemove->add(track); 3323 // indicate to client process that the track was disabled because of underrun; 3324 // it will then automatically call start() when data is available 3325 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3326 // If one track is not ready, mark the mixer also not ready if: 3327 // - the mixer was ready during previous round OR 3328 // - no other track is ready 3329 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3330 mixerStatus != MIXER_TRACKS_READY) { 3331 mixerStatus = MIXER_TRACKS_ENABLED; 3332 } 3333 } 3334 mAudioMixer->disable(name); 3335 } 3336 3337 } // local variable scope to avoid goto warning 3338track_is_ready: ; 3339 3340 } 3341 3342 // Push the new FastMixer state if necessary 3343 bool pauseAudioWatchdog = false; 3344 if (didModify) { 3345 state->mFastTracksGen++; 3346 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3347 if (kUseFastMixer == FastMixer_Dynamic && 3348 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3349 state->mCommand = FastMixerState::COLD_IDLE; 3350 state->mColdFutexAddr = &mFastMixerFutex; 3351 state->mColdGen++; 3352 mFastMixerFutex = 0; 3353 if (kUseFastMixer == FastMixer_Dynamic) { 3354 mNormalSink = mOutputSink; 3355 } 3356 // If we go into cold idle, need to wait for acknowledgement 3357 // so that fast mixer stops doing I/O. 3358 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3359 pauseAudioWatchdog = true; 3360 } 3361 sq->end(); 3362 } 3363 if (sq != NULL) { 3364 sq->end(didModify); 3365 sq->push(block); 3366 } 3367#ifdef AUDIO_WATCHDOG 3368 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3369 mAudioWatchdog->pause(); 3370 } 3371#endif 3372 3373 // Now perform the deferred reset on fast tracks that have stopped 3374 while (resetMask != 0) { 3375 size_t i = __builtin_ctz(resetMask); 3376 ALOG_ASSERT(i < count); 3377 resetMask &= ~(1 << i); 3378 sp<Track> t = mActiveTracks[i].promote(); 3379 if (t == 0) continue; 3380 Track* track = t.get(); 3381 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3382 track->reset(); 3383 } 3384 3385 // remove all the tracks that need to be... 3386 count = tracksToRemove->size(); 3387 if (CC_UNLIKELY(count)) { 3388 for (size_t i=0 ; i<count ; i++) { 3389 const sp<Track>& track = tracksToRemove->itemAt(i); 3390 mActiveTracks.remove(track); 3391 if (track->mainBuffer() != mMixBuffer) { 3392 chain = getEffectChain_l(track->sessionId()); 3393 if (chain != 0) { 3394 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3395 track->sessionId()); 3396 chain->decActiveTrackCnt(); 3397 } 3398 } 3399 if (track->isTerminated()) { 3400 removeTrack_l(track); 3401 } 3402 } 3403 } 3404 3405 // mix buffer must be cleared if all tracks are connected to an 3406 // effect chain as in this case the mixer will not write to 3407 // mix buffer and track effects will accumulate into it 3408 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3409 (mixedTracks == 0 && fastTracks > 0)) { 3410 // FIXME as a performance optimization, should remember previous zero status 3411 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3412 } 3413 3414 // if any fast tracks, then status is ready 3415 mMixerStatusIgnoringFastTracks = mixerStatus; 3416 if (fastTracks > 0) { 3417 mixerStatus = MIXER_TRACKS_READY; 3418 } 3419 return mixerStatus; 3420} 3421 3422/* 3423The derived values that are cached: 3424 - mixBufferSize from frame count * frame size 3425 - activeSleepTime from activeSleepTimeUs() 3426 - idleSleepTime from idleSleepTimeUs() 3427 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3428 - maxPeriod from frame count and sample rate (MIXER only) 3429 3430The parameters that affect these derived values are: 3431 - frame count 3432 - frame size 3433 - sample rate 3434 - device type: A2DP or not 3435 - device latency 3436 - format: PCM or not 3437 - active sleep time 3438 - idle sleep time 3439*/ 3440 3441void AudioFlinger::PlaybackThread::cacheParameters_l() 3442{ 3443 mixBufferSize = mNormalFrameCount * mFrameSize; 3444 activeSleepTime = activeSleepTimeUs(); 3445 idleSleepTime = idleSleepTimeUs(); 3446} 3447 3448void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3449{ 3450 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3451 this, streamType, mTracks.size()); 3452 Mutex::Autolock _l(mLock); 3453 3454 size_t size = mTracks.size(); 3455 for (size_t i = 0; i < size; i++) { 3456 sp<Track> t = mTracks[i]; 3457 if (t->streamType() == streamType) { 3458 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 3459 t->mCblk->cv.signal(); 3460 } 3461 } 3462} 3463 3464// getTrackName_l() must be called with ThreadBase::mLock held 3465int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3466{ 3467 return mAudioMixer->getTrackName(channelMask, sessionId); 3468} 3469 3470// deleteTrackName_l() must be called with ThreadBase::mLock held 3471void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3472{ 3473 ALOGV("remove track (%d) and delete from mixer", name); 3474 mAudioMixer->deleteTrackName(name); 3475} 3476 3477// checkForNewParameters_l() must be called with ThreadBase::mLock held 3478bool AudioFlinger::MixerThread::checkForNewParameters_l() 3479{ 3480 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3481 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3482 bool reconfig = false; 3483 3484 while (!mNewParameters.isEmpty()) { 3485 3486 if (mFastMixer != NULL) { 3487 FastMixerStateQueue *sq = mFastMixer->sq(); 3488 FastMixerState *state = sq->begin(); 3489 if (!(state->mCommand & FastMixerState::IDLE)) { 3490 previousCommand = state->mCommand; 3491 state->mCommand = FastMixerState::HOT_IDLE; 3492 sq->end(); 3493 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3494 } else { 3495 sq->end(false /*didModify*/); 3496 } 3497 } 3498 3499 status_t status = NO_ERROR; 3500 String8 keyValuePair = mNewParameters[0]; 3501 AudioParameter param = AudioParameter(keyValuePair); 3502 int value; 3503 3504 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3505 reconfig = true; 3506 } 3507 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3508 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3509 status = BAD_VALUE; 3510 } else { 3511 reconfig = true; 3512 } 3513 } 3514 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3515 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3516 status = BAD_VALUE; 3517 } else { 3518 reconfig = true; 3519 } 3520 } 3521 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3522 // do not accept frame count changes if tracks are open as the track buffer 3523 // size depends on frame count and correct behavior would not be guaranteed 3524 // if frame count is changed after track creation 3525 if (!mTracks.isEmpty()) { 3526 status = INVALID_OPERATION; 3527 } else { 3528 reconfig = true; 3529 } 3530 } 3531 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3532#ifdef ADD_BATTERY_DATA 3533 // when changing the audio output device, call addBatteryData to notify 3534 // the change 3535 if (mOutDevice != value) { 3536 uint32_t params = 0; 3537 // check whether speaker is on 3538 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3539 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3540 } 3541 3542 audio_devices_t deviceWithoutSpeaker 3543 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3544 // check if any other device (except speaker) is on 3545 if (value & deviceWithoutSpeaker ) { 3546 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3547 } 3548 3549 if (params != 0) { 3550 addBatteryData(params); 3551 } 3552 } 3553#endif 3554 3555 // forward device change to effects that have requested to be 3556 // aware of attached audio device. 3557 mOutDevice = value; 3558 for (size_t i = 0; i < mEffectChains.size(); i++) { 3559 mEffectChains[i]->setDevice_l(mOutDevice); 3560 } 3561 } 3562 3563 if (status == NO_ERROR) { 3564 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3565 keyValuePair.string()); 3566 if (!mStandby && status == INVALID_OPERATION) { 3567 mOutput->stream->common.standby(&mOutput->stream->common); 3568 mStandby = true; 3569 mBytesWritten = 0; 3570 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3571 keyValuePair.string()); 3572 } 3573 if (status == NO_ERROR && reconfig) { 3574 delete mAudioMixer; 3575 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3576 mAudioMixer = NULL; 3577 readOutputParameters(); 3578 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3579 for (size_t i = 0; i < mTracks.size() ; i++) { 3580 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3581 if (name < 0) break; 3582 mTracks[i]->mName = name; 3583 // limit track sample rate to 2 x new output sample rate 3584 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3585 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3586 } 3587 } 3588 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3589 } 3590 } 3591 3592 mNewParameters.removeAt(0); 3593 3594 mParamStatus = status; 3595 mParamCond.signal(); 3596 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3597 // already timed out waiting for the status and will never signal the condition. 3598 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3599 } 3600 3601 if (!(previousCommand & FastMixerState::IDLE)) { 3602 ALOG_ASSERT(mFastMixer != NULL); 3603 FastMixerStateQueue *sq = mFastMixer->sq(); 3604 FastMixerState *state = sq->begin(); 3605 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3606 state->mCommand = previousCommand; 3607 sq->end(); 3608 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3609 } 3610 3611 return reconfig; 3612} 3613 3614void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3615{ 3616 NBAIO_Source *teeSource = source.get(); 3617 if (teeSource != NULL) { 3618 char teeTime[16]; 3619 struct timeval tv; 3620 gettimeofday(&tv, NULL); 3621 struct tm tm; 3622 localtime_r(&tv.tv_sec, &tm); 3623 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3624 char teePath[64]; 3625 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3626 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3627 if (teeFd >= 0) { 3628 char wavHeader[44]; 3629 memcpy(wavHeader, 3630 "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", 3631 sizeof(wavHeader)); 3632 NBAIO_Format format = teeSource->format(); 3633 unsigned channelCount = Format_channelCount(format); 3634 ALOG_ASSERT(channelCount <= FCC_2); 3635 uint32_t sampleRate = Format_sampleRate(format); 3636 wavHeader[22] = channelCount; // number of channels 3637 wavHeader[24] = sampleRate; // sample rate 3638 wavHeader[25] = sampleRate >> 8; 3639 wavHeader[32] = channelCount * 2; // block alignment 3640 write(teeFd, wavHeader, sizeof(wavHeader)); 3641 size_t total = 0; 3642 bool firstRead = true; 3643 for (;;) { 3644#define TEE_SINK_READ 1024 3645 short buffer[TEE_SINK_READ * FCC_2]; 3646 size_t count = TEE_SINK_READ; 3647 ssize_t actual = teeSource->read(buffer, count, 3648 AudioBufferProvider::kInvalidPTS); 3649 bool wasFirstRead = firstRead; 3650 firstRead = false; 3651 if (actual <= 0) { 3652 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3653 continue; 3654 } 3655 break; 3656 } 3657 ALOG_ASSERT(actual <= (ssize_t)count); 3658 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3659 total += actual; 3660 } 3661 lseek(teeFd, (off_t) 4, SEEK_SET); 3662 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3663 write(teeFd, &temp, sizeof(temp)); 3664 lseek(teeFd, (off_t) 40, SEEK_SET); 3665 temp = total * channelCount * sizeof(short); 3666 write(teeFd, &temp, sizeof(temp)); 3667 close(teeFd); 3668 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3669 } else { 3670 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3671 } 3672 } 3673} 3674 3675void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3676{ 3677 const size_t SIZE = 256; 3678 char buffer[SIZE]; 3679 String8 result; 3680 3681 PlaybackThread::dumpInternals(fd, args); 3682 3683 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3684 result.append(buffer); 3685 write(fd, result.string(), result.size()); 3686 3687 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3688 FastMixerDumpState copy = mFastMixerDumpState; 3689 copy.dump(fd); 3690 3691#ifdef STATE_QUEUE_DUMP 3692 // Similar for state queue 3693 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3694 observerCopy.dump(fd); 3695 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3696 mutatorCopy.dump(fd); 3697#endif 3698 3699 // Write the tee output to a .wav file 3700 dumpTee(fd, mTeeSource, mId); 3701 3702#ifdef AUDIO_WATCHDOG 3703 if (mAudioWatchdog != 0) { 3704 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3705 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3706 wdCopy.dump(fd); 3707 } 3708#endif 3709} 3710 3711uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3712{ 3713 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3714} 3715 3716uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3717{ 3718 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3719} 3720 3721void AudioFlinger::MixerThread::cacheParameters_l() 3722{ 3723 PlaybackThread::cacheParameters_l(); 3724 3725 // FIXME: Relaxed timing because of a certain device that can't meet latency 3726 // Should be reduced to 2x after the vendor fixes the driver issue 3727 // increase threshold again due to low power audio mode. The way this warning 3728 // threshold is calculated and its usefulness should be reconsidered anyway. 3729 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3730} 3731 3732// ---------------------------------------------------------------------------- 3733AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3734 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3735 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3736 // mLeftVolFloat, mRightVolFloat 3737{ 3738} 3739 3740AudioFlinger::DirectOutputThread::~DirectOutputThread() 3741{ 3742} 3743 3744AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3745 Vector< sp<Track> > *tracksToRemove 3746) 3747{ 3748 sp<Track> trackToRemove; 3749 3750 mixer_state mixerStatus = MIXER_IDLE; 3751 3752 // find out which tracks need to be processed 3753 if (mActiveTracks.size() != 0) { 3754 sp<Track> t = mActiveTracks[0].promote(); 3755 // The track died recently 3756 if (t == 0) return MIXER_IDLE; 3757 3758 Track* const track = t.get(); 3759 audio_track_cblk_t* cblk = track->cblk(); 3760 3761 // The first time a track is added we wait 3762 // for all its buffers to be filled before processing it 3763 uint32_t minFrames; 3764 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3765 minFrames = mNormalFrameCount; 3766 } else { 3767 minFrames = 1; 3768 } 3769 if ((track->framesReady() >= minFrames) && track->isReady() && 3770 !track->isPaused() && !track->isTerminated()) 3771 { 3772 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3773 3774 if (track->mFillingUpStatus == Track::FS_FILLED) { 3775 track->mFillingUpStatus = Track::FS_ACTIVE; 3776 mLeftVolFloat = mRightVolFloat = 0; 3777 if (track->mState == TrackBase::RESUMING) { 3778 track->mState = TrackBase::ACTIVE; 3779 } 3780 } 3781 3782 // compute volume for this track 3783 float left, right; 3784 if (track->isMuted() || mMasterMute || track->isPausing() || 3785 mStreamTypes[track->streamType()].mute) { 3786 left = right = 0; 3787 if (track->isPausing()) { 3788 track->setPaused(); 3789 } 3790 } else { 3791 float typeVolume = mStreamTypes[track->streamType()].volume; 3792 float v = mMasterVolume * typeVolume; 3793 uint32_t vlr = cblk->getVolumeLR(); 3794 float v_clamped = v * (vlr & 0xFFFF); 3795 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3796 left = v_clamped/MAX_GAIN; 3797 v_clamped = v * (vlr >> 16); 3798 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3799 right = v_clamped/MAX_GAIN; 3800 } 3801 3802 if (left != mLeftVolFloat || right != mRightVolFloat) { 3803 mLeftVolFloat = left; 3804 mRightVolFloat = right; 3805 3806 // Convert volumes from float to 8.24 3807 uint32_t vl = (uint32_t)(left * (1 << 24)); 3808 uint32_t vr = (uint32_t)(right * (1 << 24)); 3809 3810 // Delegate volume control to effect in track effect chain if needed 3811 // only one effect chain can be present on DirectOutputThread, so if 3812 // there is one, the track is connected to it 3813 if (!mEffectChains.isEmpty()) { 3814 // Do not ramp volume if volume is controlled by effect 3815 mEffectChains[0]->setVolume_l(&vl, &vr); 3816 left = (float)vl / (1 << 24); 3817 right = (float)vr / (1 << 24); 3818 } 3819 mOutput->stream->set_volume(mOutput->stream, left, right); 3820 } 3821 3822 // reset retry count 3823 track->mRetryCount = kMaxTrackRetriesDirect; 3824 mActiveTrack = t; 3825 mixerStatus = MIXER_TRACKS_READY; 3826 } else { 3827 // clear effect chain input buffer if an active track underruns to avoid sending 3828 // previous audio buffer again to effects 3829 if (!mEffectChains.isEmpty()) { 3830 mEffectChains[0]->clearInputBuffer(); 3831 } 3832 3833 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3834 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3835 track->isStopped() || track->isPaused()) { 3836 // We have consumed all the buffers of this track. 3837 // Remove it from the list of active tracks. 3838 // TODO: implement behavior for compressed audio 3839 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3840 size_t framesWritten = 3841 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3842 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3843 if (track->isStopped()) { 3844 track->reset(); 3845 } 3846 trackToRemove = track; 3847 } 3848 } else { 3849 // No buffers for this track. Give it a few chances to 3850 // fill a buffer, then remove it from active list. 3851 if (--(track->mRetryCount) <= 0) { 3852 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3853 trackToRemove = track; 3854 } else { 3855 mixerStatus = MIXER_TRACKS_ENABLED; 3856 } 3857 } 3858 } 3859 } 3860 3861 // FIXME merge this with similar code for removing multiple tracks 3862 // remove all the tracks that need to be... 3863 if (CC_UNLIKELY(trackToRemove != 0)) { 3864 tracksToRemove->add(trackToRemove); 3865 mActiveTracks.remove(trackToRemove); 3866 if (!mEffectChains.isEmpty()) { 3867 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3868 trackToRemove->sessionId()); 3869 mEffectChains[0]->decActiveTrackCnt(); 3870 } 3871 if (trackToRemove->isTerminated()) { 3872 removeTrack_l(trackToRemove); 3873 } 3874 } 3875 3876 return mixerStatus; 3877} 3878 3879void AudioFlinger::DirectOutputThread::threadLoop_mix() 3880{ 3881 AudioBufferProvider::Buffer buffer; 3882 size_t frameCount = mFrameCount; 3883 int8_t *curBuf = (int8_t *)mMixBuffer; 3884 // output audio to hardware 3885 while (frameCount) { 3886 buffer.frameCount = frameCount; 3887 mActiveTrack->getNextBuffer(&buffer); 3888 if (CC_UNLIKELY(buffer.raw == NULL)) { 3889 memset(curBuf, 0, frameCount * mFrameSize); 3890 break; 3891 } 3892 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3893 frameCount -= buffer.frameCount; 3894 curBuf += buffer.frameCount * mFrameSize; 3895 mActiveTrack->releaseBuffer(&buffer); 3896 } 3897 sleepTime = 0; 3898 standbyTime = systemTime() + standbyDelay; 3899 mActiveTrack.clear(); 3900 3901} 3902 3903void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3904{ 3905 if (sleepTime == 0) { 3906 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3907 sleepTime = activeSleepTime; 3908 } else { 3909 sleepTime = idleSleepTime; 3910 } 3911 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3912 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3913 sleepTime = 0; 3914 } 3915} 3916 3917// getTrackName_l() must be called with ThreadBase::mLock held 3918int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3919 int sessionId) 3920{ 3921 return 0; 3922} 3923 3924// deleteTrackName_l() must be called with ThreadBase::mLock held 3925void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3926{ 3927} 3928 3929// checkForNewParameters_l() must be called with ThreadBase::mLock held 3930bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3931{ 3932 bool reconfig = false; 3933 3934 while (!mNewParameters.isEmpty()) { 3935 status_t status = NO_ERROR; 3936 String8 keyValuePair = mNewParameters[0]; 3937 AudioParameter param = AudioParameter(keyValuePair); 3938 int value; 3939 3940 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3941 // do not accept frame count changes if tracks are open as the track buffer 3942 // size depends on frame count and correct behavior would not be garantied 3943 // if frame count is changed after track creation 3944 if (!mTracks.isEmpty()) { 3945 status = INVALID_OPERATION; 3946 } else { 3947 reconfig = true; 3948 } 3949 } 3950 if (status == NO_ERROR) { 3951 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3952 keyValuePair.string()); 3953 if (!mStandby && status == INVALID_OPERATION) { 3954 mOutput->stream->common.standby(&mOutput->stream->common); 3955 mStandby = true; 3956 mBytesWritten = 0; 3957 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3958 keyValuePair.string()); 3959 } 3960 if (status == NO_ERROR && reconfig) { 3961 readOutputParameters(); 3962 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3963 } 3964 } 3965 3966 mNewParameters.removeAt(0); 3967 3968 mParamStatus = status; 3969 mParamCond.signal(); 3970 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3971 // already timed out waiting for the status and will never signal the condition. 3972 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3973 } 3974 return reconfig; 3975} 3976 3977uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3978{ 3979 uint32_t time; 3980 if (audio_is_linear_pcm(mFormat)) { 3981 time = PlaybackThread::activeSleepTimeUs(); 3982 } else { 3983 time = 10000; 3984 } 3985 return time; 3986} 3987 3988uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3989{ 3990 uint32_t time; 3991 if (audio_is_linear_pcm(mFormat)) { 3992 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3993 } else { 3994 time = 10000; 3995 } 3996 return time; 3997} 3998 3999uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4000{ 4001 uint32_t time; 4002 if (audio_is_linear_pcm(mFormat)) { 4003 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4004 } else { 4005 time = 10000; 4006 } 4007 return time; 4008} 4009 4010void AudioFlinger::DirectOutputThread::cacheParameters_l() 4011{ 4012 PlaybackThread::cacheParameters_l(); 4013 4014 // use shorter standby delay as on normal output to release 4015 // hardware resources as soon as possible 4016 standbyDelay = microseconds(activeSleepTime*2); 4017} 4018 4019// ---------------------------------------------------------------------------- 4020 4021AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4022 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4023 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4024 DUPLICATING), 4025 mWaitTimeMs(UINT_MAX) 4026{ 4027 addOutputTrack(mainThread); 4028} 4029 4030AudioFlinger::DuplicatingThread::~DuplicatingThread() 4031{ 4032 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4033 mOutputTracks[i]->destroy(); 4034 } 4035} 4036 4037void AudioFlinger::DuplicatingThread::threadLoop_mix() 4038{ 4039 // mix buffers... 4040 if (outputsReady(outputTracks)) { 4041 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4042 } else { 4043 memset(mMixBuffer, 0, mixBufferSize); 4044 } 4045 sleepTime = 0; 4046 writeFrames = mNormalFrameCount; 4047 standbyTime = systemTime() + standbyDelay; 4048} 4049 4050void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4051{ 4052 if (sleepTime == 0) { 4053 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4054 sleepTime = activeSleepTime; 4055 } else { 4056 sleepTime = idleSleepTime; 4057 } 4058 } else if (mBytesWritten != 0) { 4059 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4060 writeFrames = mNormalFrameCount; 4061 memset(mMixBuffer, 0, mixBufferSize); 4062 } else { 4063 // flush remaining overflow buffers in output tracks 4064 writeFrames = 0; 4065 } 4066 sleepTime = 0; 4067 } 4068} 4069 4070void AudioFlinger::DuplicatingThread::threadLoop_write() 4071{ 4072 for (size_t i = 0; i < outputTracks.size(); i++) { 4073 outputTracks[i]->write(mMixBuffer, writeFrames); 4074 } 4075 mBytesWritten += mixBufferSize; 4076} 4077 4078void AudioFlinger::DuplicatingThread::threadLoop_standby() 4079{ 4080 // DuplicatingThread implements standby by stopping all tracks 4081 for (size_t i = 0; i < outputTracks.size(); i++) { 4082 outputTracks[i]->stop(); 4083 } 4084} 4085 4086void AudioFlinger::DuplicatingThread::saveOutputTracks() 4087{ 4088 outputTracks = mOutputTracks; 4089} 4090 4091void AudioFlinger::DuplicatingThread::clearOutputTracks() 4092{ 4093 outputTracks.clear(); 4094} 4095 4096void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4097{ 4098 Mutex::Autolock _l(mLock); 4099 // FIXME explain this formula 4100 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4101 OutputTrack *outputTrack = new OutputTrack(thread, 4102 this, 4103 mSampleRate, 4104 mFormat, 4105 mChannelMask, 4106 frameCount); 4107 if (outputTrack->cblk() != NULL) { 4108 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4109 mOutputTracks.add(outputTrack); 4110 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4111 updateWaitTime_l(); 4112 } 4113} 4114 4115void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4116{ 4117 Mutex::Autolock _l(mLock); 4118 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4119 if (mOutputTracks[i]->thread() == thread) { 4120 mOutputTracks[i]->destroy(); 4121 mOutputTracks.removeAt(i); 4122 updateWaitTime_l(); 4123 return; 4124 } 4125 } 4126 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4127} 4128 4129// caller must hold mLock 4130void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4131{ 4132 mWaitTimeMs = UINT_MAX; 4133 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4134 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4135 if (strong != 0) { 4136 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4137 if (waitTimeMs < mWaitTimeMs) { 4138 mWaitTimeMs = waitTimeMs; 4139 } 4140 } 4141 } 4142} 4143 4144 4145bool AudioFlinger::DuplicatingThread::outputsReady( 4146 const SortedVector< sp<OutputTrack> > &outputTracks) 4147{ 4148 for (size_t i = 0; i < outputTracks.size(); i++) { 4149 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4150 if (thread == 0) { 4151 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4152 outputTracks[i].get()); 4153 return false; 4154 } 4155 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4156 // see note at standby() declaration 4157 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4158 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4159 thread.get()); 4160 return false; 4161 } 4162 } 4163 return true; 4164} 4165 4166uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4167{ 4168 return (mWaitTimeMs * 1000) / 2; 4169} 4170 4171void AudioFlinger::DuplicatingThread::cacheParameters_l() 4172{ 4173 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4174 updateWaitTime_l(); 4175 4176 MixerThread::cacheParameters_l(); 4177} 4178 4179// ---------------------------------------------------------------------------- 4180 4181// TrackBase constructor must be called with AudioFlinger::mLock held 4182AudioFlinger::ThreadBase::TrackBase::TrackBase( 4183 ThreadBase *thread, 4184 const sp<Client>& client, 4185 uint32_t sampleRate, 4186 audio_format_t format, 4187 audio_channel_mask_t channelMask, 4188 size_t frameCount, 4189 const sp<IMemory>& sharedBuffer, 4190 int sessionId) 4191 : RefBase(), 4192 mThread(thread), 4193 mClient(client), 4194 mCblk(NULL), 4195 // mBuffer 4196 // mBufferEnd 4197 mStepCount(0), 4198 mState(IDLE), 4199 mSampleRate(sampleRate), 4200 mFormat(format), 4201 mChannelMask(channelMask), 4202 mChannelCount(popcount(channelMask)), 4203 mFrameSize(audio_is_linear_pcm(format) ? 4204 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 4205 mFrameCount(frameCount), 4206 mStepServerFailed(false), 4207 mSessionId(sessionId) 4208{ 4209 // client == 0 implies sharedBuffer == 0 4210 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4211 4212 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4213 sharedBuffer->size()); 4214 4215 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4216 size_t size = sizeof(audio_track_cblk_t); 4217 size_t bufferSize = frameCount * mFrameSize; 4218 if (sharedBuffer == 0) { 4219 size += bufferSize; 4220 } 4221 4222 if (client != 0) { 4223 mCblkMemory = client->heap()->allocate(size); 4224 if (mCblkMemory != 0) { 4225 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4226 // can't assume mCblk != NULL 4227 } else { 4228 ALOGE("not enough memory for AudioTrack size=%u", size); 4229 client->heap()->dump("AudioTrack"); 4230 return; 4231 } 4232 } else { 4233 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4234 // assume mCblk != NULL 4235 } 4236 4237 // construct the shared structure in-place. 4238 if (mCblk != NULL) { 4239 new(mCblk) audio_track_cblk_t(); 4240 // clear all buffers 4241 mCblk->frameCount_ = frameCount; 4242 mCblk->sampleRate = sampleRate; 4243// uncomment the following lines to quickly test 32-bit wraparound 4244// mCblk->user = 0xffff0000; 4245// mCblk->server = 0xffff0000; 4246// mCblk->userBase = 0xffff0000; 4247// mCblk->serverBase = 0xffff0000; 4248 if (sharedBuffer == 0) { 4249 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4250 memset(mBuffer, 0, bufferSize); 4251 // Force underrun condition to avoid false underrun callback until first data is 4252 // written to buffer (other flags are cleared) 4253 mCblk->flags = CBLK_UNDERRUN; 4254 } else { 4255 mBuffer = sharedBuffer->pointer(); 4256 } 4257 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4258 } 4259} 4260 4261AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4262{ 4263 if (mCblk != NULL) { 4264 if (mClient == 0) { 4265 delete mCblk; 4266 } else { 4267 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4268 } 4269 } 4270 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4271 if (mClient != 0) { 4272 // Client destructor must run with AudioFlinger mutex locked 4273 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4274 // If the client's reference count drops to zero, the associated destructor 4275 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4276 // relying on the automatic clear() at end of scope. 4277 mClient.clear(); 4278 } 4279} 4280 4281// AudioBufferProvider interface 4282// getNextBuffer() = 0; 4283// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4284void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4285{ 4286 buffer->raw = NULL; 4287 mStepCount = buffer->frameCount; 4288 // FIXME See note at getNextBuffer() 4289 (void) step(); // ignore return value of step() 4290 buffer->frameCount = 0; 4291} 4292 4293bool AudioFlinger::ThreadBase::TrackBase::step() { 4294 bool result; 4295 audio_track_cblk_t* cblk = this->cblk(); 4296 4297 result = cblk->stepServer(mStepCount, mFrameCount, isOut()); 4298 if (!result) { 4299 ALOGV("stepServer failed acquiring cblk mutex"); 4300 mStepServerFailed = true; 4301 } 4302 return result; 4303} 4304 4305void AudioFlinger::ThreadBase::TrackBase::reset() { 4306 audio_track_cblk_t* cblk = this->cblk(); 4307 4308 cblk->user = 0; 4309 cblk->server = 0; 4310 cblk->userBase = 0; 4311 cblk->serverBase = 0; 4312 mStepServerFailed = false; 4313 ALOGV("TrackBase::reset"); 4314} 4315 4316uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4317 return mCblk->sampleRate; 4318} 4319 4320void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4321 audio_track_cblk_t* cblk = this->cblk(); 4322 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; 4323 int8_t *bufferEnd = bufferStart + frames * mFrameSize; 4324 4325 // Check validity of returned pointer in case the track control block would have been corrupted. 4326 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4327 "TrackBase::getBuffer buffer out of range:\n" 4328 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4329 " server %u, serverBase %u, user %u, userBase %u, frameSize %u", 4330 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4331 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); 4332 4333 return bufferStart; 4334} 4335 4336status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4337{ 4338 mSyncEvents.add(event); 4339 return NO_ERROR; 4340} 4341 4342// ---------------------------------------------------------------------------- 4343 4344// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4345AudioFlinger::PlaybackThread::Track::Track( 4346 PlaybackThread *thread, 4347 const sp<Client>& client, 4348 audio_stream_type_t streamType, 4349 uint32_t sampleRate, 4350 audio_format_t format, 4351 audio_channel_mask_t channelMask, 4352 size_t frameCount, 4353 const sp<IMemory>& sharedBuffer, 4354 int sessionId, 4355 IAudioFlinger::track_flags_t flags) 4356 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4357 sessionId), 4358 mMute(false), 4359 mFillingUpStatus(FS_INVALID), 4360 // mRetryCount initialized later when needed 4361 mSharedBuffer(sharedBuffer), 4362 mStreamType(streamType), 4363 mName(-1), // see note below 4364 mMainBuffer(thread->mixBuffer()), 4365 mAuxBuffer(NULL), 4366 mAuxEffectId(0), mHasVolumeController(false), 4367 mPresentationCompleteFrames(0), 4368 mFlags(flags), 4369 mFastIndex(-1), 4370 mUnderrunCount(0), 4371 mCachedVolume(1.0) 4372{ 4373 if (mCblk != NULL) { 4374 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4375 mName = thread->getTrackName_l(channelMask, sessionId); 4376 mCblk->mName = mName; 4377 if (mName < 0) { 4378 ALOGE("no more track names available"); 4379 return; 4380 } 4381 // only allocate a fast track index if we were able to allocate a normal track name 4382 if (flags & IAudioFlinger::TRACK_FAST) { 4383 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4384 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4385 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4386 // FIXME This is too eager. We allocate a fast track index before the 4387 // fast track becomes active. Since fast tracks are a scarce resource, 4388 // this means we are potentially denying other more important fast tracks from 4389 // being created. It would be better to allocate the index dynamically. 4390 mFastIndex = i; 4391 mCblk->mName = i; 4392 // Read the initial underruns because this field is never cleared by the fast mixer 4393 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4394 thread->mFastTrackAvailMask &= ~(1 << i); 4395 } 4396 } 4397 ALOGV("Track constructor name %d, calling pid %d", mName, 4398 IPCThreadState::self()->getCallingPid()); 4399} 4400 4401AudioFlinger::PlaybackThread::Track::~Track() 4402{ 4403 ALOGV("PlaybackThread::Track destructor"); 4404} 4405 4406void AudioFlinger::PlaybackThread::Track::destroy() 4407{ 4408 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4409 // by removing it from mTracks vector, so there is a risk that this Tracks's 4410 // destructor is called. As the destructor needs to lock mLock, 4411 // we must acquire a strong reference on this Track before locking mLock 4412 // here so that the destructor is called only when exiting this function. 4413 // On the other hand, as long as Track::destroy() is only called by 4414 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4415 // this Track with its member mTrack. 4416 sp<Track> keep(this); 4417 { // scope for mLock 4418 sp<ThreadBase> thread = mThread.promote(); 4419 if (thread != 0) { 4420 if (!isOutputTrack()) { 4421 if (mState == ACTIVE || mState == RESUMING) { 4422 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4423 4424#ifdef ADD_BATTERY_DATA 4425 // to track the speaker usage 4426 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4427#endif 4428 } 4429 AudioSystem::releaseOutput(thread->id()); 4430 } 4431 Mutex::Autolock _l(thread->mLock); 4432 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4433 playbackThread->destroyTrack_l(this); 4434 } 4435 } 4436} 4437 4438/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4439{ 4440 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 4441 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4442} 4443 4444void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4445{ 4446 uint32_t vlr = mCblk->getVolumeLR(); 4447 if (isFastTrack()) { 4448 sprintf(buffer, " F %2d", mFastIndex); 4449 } else { 4450 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4451 } 4452 track_state state = mState; 4453 char stateChar; 4454 switch (state) { 4455 case IDLE: 4456 stateChar = 'I'; 4457 break; 4458 case TERMINATED: 4459 stateChar = 'T'; 4460 break; 4461 case STOPPING_1: 4462 stateChar = 's'; 4463 break; 4464 case STOPPING_2: 4465 stateChar = '5'; 4466 break; 4467 case STOPPED: 4468 stateChar = 'S'; 4469 break; 4470 case RESUMING: 4471 stateChar = 'R'; 4472 break; 4473 case ACTIVE: 4474 stateChar = 'A'; 4475 break; 4476 case PAUSING: 4477 stateChar = 'p'; 4478 break; 4479 case PAUSED: 4480 stateChar = 'P'; 4481 break; 4482 case FLUSHED: 4483 stateChar = 'F'; 4484 break; 4485 default: 4486 stateChar = '?'; 4487 break; 4488 } 4489 char nowInUnderrun; 4490 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4491 case UNDERRUN_FULL: 4492 nowInUnderrun = ' '; 4493 break; 4494 case UNDERRUN_PARTIAL: 4495 nowInUnderrun = '<'; 4496 break; 4497 case UNDERRUN_EMPTY: 4498 nowInUnderrun = '*'; 4499 break; 4500 default: 4501 nowInUnderrun = '?'; 4502 break; 4503 } 4504 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4505 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4506 (mClient == 0) ? getpid_cached : mClient->pid(), 4507 mStreamType, 4508 mFormat, 4509 mChannelMask, 4510 mSessionId, 4511 mStepCount, 4512 mFrameCount, 4513 stateChar, 4514 mMute, 4515 mFillingUpStatus, 4516 mCblk->sampleRate, 4517 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4518 20.0 * log10((vlr >> 16) / 4096.0), 4519 mCblk->server, 4520 mCblk->user, 4521 (int)mMainBuffer, 4522 (int)mAuxBuffer, 4523 mCblk->flags, 4524 mUnderrunCount, 4525 nowInUnderrun); 4526} 4527 4528// AudioBufferProvider interface 4529status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4530 AudioBufferProvider::Buffer* buffer, int64_t pts) 4531{ 4532 audio_track_cblk_t* cblk = this->cblk(); 4533 uint32_t framesReady; 4534 uint32_t framesReq = buffer->frameCount; 4535 4536 // Check if last stepServer failed, try to step now 4537 if (mStepServerFailed) { 4538 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4539 // Since the fast mixer is higher priority than client callback thread, 4540 // it does not result in priority inversion for client. 4541 // But a non-blocking solution would be preferable to avoid 4542 // fast mixer being unable to tryLock(), and 4543 // to avoid the extra context switches if the client wakes up, 4544 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4545 if (!step()) goto getNextBuffer_exit; 4546 ALOGV("stepServer recovered"); 4547 mStepServerFailed = false; 4548 } 4549 4550 // FIXME Same as above 4551 framesReady = cblk->framesReadyOut(); 4552 4553 if (CC_LIKELY(framesReady)) { 4554 uint32_t s = cblk->server; 4555 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 4556 4557 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4558 if (framesReq > framesReady) { 4559 framesReq = framesReady; 4560 } 4561 if (framesReq > bufferEnd - s) { 4562 framesReq = bufferEnd - s; 4563 } 4564 4565 buffer->raw = getBuffer(s, framesReq); 4566 buffer->frameCount = framesReq; 4567 return NO_ERROR; 4568 } 4569 4570getNextBuffer_exit: 4571 buffer->raw = NULL; 4572 buffer->frameCount = 0; 4573 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4574 return NOT_ENOUGH_DATA; 4575} 4576 4577// Note that framesReady() takes a mutex on the control block using tryLock(). 4578// This could result in priority inversion if framesReady() is called by the normal mixer, 4579// as the normal mixer thread runs at lower 4580// priority than the client's callback thread: there is a short window within framesReady() 4581// during which the normal mixer could be preempted, and the client callback would block. 4582// Another problem can occur if framesReady() is called by the fast mixer: 4583// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4584// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4585size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4586 return mCblk->framesReadyOut(); 4587} 4588 4589// Don't call for fast tracks; the framesReady() could result in priority inversion 4590bool AudioFlinger::PlaybackThread::Track::isReady() const { 4591 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4592 4593 if (framesReady() >= mFrameCount || 4594 (mCblk->flags & CBLK_FORCEREADY)) { 4595 mFillingUpStatus = FS_FILLED; 4596 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4597 return true; 4598 } 4599 return false; 4600} 4601 4602status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4603 int triggerSession) 4604{ 4605 status_t status = NO_ERROR; 4606 ALOGV("start(%d), calling pid %d session %d", 4607 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4608 4609 sp<ThreadBase> thread = mThread.promote(); 4610 if (thread != 0) { 4611 Mutex::Autolock _l(thread->mLock); 4612 track_state state = mState; 4613 // here the track could be either new, or restarted 4614 // in both cases "unstop" the track 4615 if (mState == PAUSED) { 4616 mState = TrackBase::RESUMING; 4617 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4618 } else { 4619 mState = TrackBase::ACTIVE; 4620 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4621 } 4622 4623 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4624 thread->mLock.unlock(); 4625 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4626 thread->mLock.lock(); 4627 4628#ifdef ADD_BATTERY_DATA 4629 // to track the speaker usage 4630 if (status == NO_ERROR) { 4631 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4632 } 4633#endif 4634 } 4635 if (status == NO_ERROR) { 4636 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4637 playbackThread->addTrack_l(this); 4638 } else { 4639 mState = state; 4640 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4641 } 4642 } else { 4643 status = BAD_VALUE; 4644 } 4645 return status; 4646} 4647 4648void AudioFlinger::PlaybackThread::Track::stop() 4649{ 4650 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4651 sp<ThreadBase> thread = mThread.promote(); 4652 if (thread != 0) { 4653 Mutex::Autolock _l(thread->mLock); 4654 track_state state = mState; 4655 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4656 // If the track is not active (PAUSED and buffers full), flush buffers 4657 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4658 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4659 reset(); 4660 mState = STOPPED; 4661 } else if (!isFastTrack()) { 4662 mState = STOPPED; 4663 } else { 4664 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4665 // and then to STOPPED and reset() when presentation is complete 4666 mState = STOPPING_1; 4667 } 4668 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4669 playbackThread); 4670 } 4671 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4672 thread->mLock.unlock(); 4673 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4674 thread->mLock.lock(); 4675 4676#ifdef ADD_BATTERY_DATA 4677 // to track the speaker usage 4678 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4679#endif 4680 } 4681 } 4682} 4683 4684void AudioFlinger::PlaybackThread::Track::pause() 4685{ 4686 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4687 sp<ThreadBase> thread = mThread.promote(); 4688 if (thread != 0) { 4689 Mutex::Autolock _l(thread->mLock); 4690 if (mState == ACTIVE || mState == RESUMING) { 4691 mState = PAUSING; 4692 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4693 if (!isOutputTrack()) { 4694 thread->mLock.unlock(); 4695 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4696 thread->mLock.lock(); 4697 4698#ifdef ADD_BATTERY_DATA 4699 // to track the speaker usage 4700 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4701#endif 4702 } 4703 } 4704 } 4705} 4706 4707void AudioFlinger::PlaybackThread::Track::flush() 4708{ 4709 ALOGV("flush(%d)", mName); 4710 sp<ThreadBase> thread = mThread.promote(); 4711 if (thread != 0) { 4712 Mutex::Autolock _l(thread->mLock); 4713 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4714 mState != PAUSING && mState != IDLE && mState != FLUSHED) { 4715 return; 4716 } 4717 // No point remaining in PAUSED state after a flush => go to 4718 // FLUSHED state 4719 mState = FLUSHED; 4720 // do not reset the track if it is still in the process of being stopped or paused. 4721 // this will be done by prepareTracks_l() when the track is stopped. 4722 // prepareTracks_l() will see mState == FLUSHED, then 4723 // remove from active track list, reset(), and trigger presentation complete 4724 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4725 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4726 reset(); 4727 } 4728 } 4729} 4730 4731void AudioFlinger::PlaybackThread::Track::reset() 4732{ 4733 // Do not reset twice to avoid discarding data written just after a flush and before 4734 // the audioflinger thread detects the track is stopped. 4735 if (!mResetDone) { 4736 TrackBase::reset(); 4737 // Force underrun condition to avoid false underrun callback until first data is 4738 // written to buffer 4739 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4740 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4741 mFillingUpStatus = FS_FILLING; 4742 mResetDone = true; 4743 if (mState == FLUSHED) { 4744 mState = IDLE; 4745 } 4746 } 4747} 4748 4749void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4750{ 4751 mMute = muted; 4752} 4753 4754status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4755{ 4756 status_t status = DEAD_OBJECT; 4757 sp<ThreadBase> thread = mThread.promote(); 4758 if (thread != 0) { 4759 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4760 sp<AudioFlinger> af = mClient->audioFlinger(); 4761 4762 Mutex::Autolock _l(af->mLock); 4763 4764 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4765 4766 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4767 Mutex::Autolock _dl(playbackThread->mLock); 4768 Mutex::Autolock _sl(srcThread->mLock); 4769 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4770 if (chain == 0) { 4771 return INVALID_OPERATION; 4772 } 4773 4774 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4775 if (effect == 0) { 4776 return INVALID_OPERATION; 4777 } 4778 srcThread->removeEffect_l(effect); 4779 playbackThread->addEffect_l(effect); 4780 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4781 if (effect->state() == EffectModule::ACTIVE || 4782 effect->state() == EffectModule::STOPPING) { 4783 effect->start(); 4784 } 4785 4786 sp<EffectChain> dstChain = effect->chain().promote(); 4787 if (dstChain == 0) { 4788 srcThread->addEffect_l(effect); 4789 return INVALID_OPERATION; 4790 } 4791 AudioSystem::unregisterEffect(effect->id()); 4792 AudioSystem::registerEffect(&effect->desc(), 4793 srcThread->id(), 4794 dstChain->strategy(), 4795 AUDIO_SESSION_OUTPUT_MIX, 4796 effect->id()); 4797 } 4798 status = playbackThread->attachAuxEffect(this, EffectId); 4799 } 4800 return status; 4801} 4802 4803void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4804{ 4805 mAuxEffectId = EffectId; 4806 mAuxBuffer = buffer; 4807} 4808 4809bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4810 size_t audioHalFrames) 4811{ 4812 // a track is considered presented when the total number of frames written to audio HAL 4813 // corresponds to the number of frames written when presentationComplete() is called for the 4814 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4815 if (mPresentationCompleteFrames == 0) { 4816 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4817 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4818 mPresentationCompleteFrames, audioHalFrames); 4819 } 4820 if (framesWritten >= mPresentationCompleteFrames) { 4821 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4822 mSessionId, framesWritten); 4823 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4824 return true; 4825 } 4826 return false; 4827} 4828 4829void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4830{ 4831 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4832 if (mSyncEvents[i]->type() == type) { 4833 mSyncEvents[i]->trigger(); 4834 mSyncEvents.removeAt(i); 4835 i--; 4836 } 4837 } 4838} 4839 4840// implement VolumeBufferProvider interface 4841 4842uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4843{ 4844 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4845 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4846 uint32_t vlr = mCblk->getVolumeLR(); 4847 uint32_t vl = vlr & 0xFFFF; 4848 uint32_t vr = vlr >> 16; 4849 // track volumes come from shared memory, so can't be trusted and must be clamped 4850 if (vl > MAX_GAIN_INT) { 4851 vl = MAX_GAIN_INT; 4852 } 4853 if (vr > MAX_GAIN_INT) { 4854 vr = MAX_GAIN_INT; 4855 } 4856 // now apply the cached master volume and stream type volume; 4857 // this is trusted but lacks any synchronization or barrier so may be stale 4858 float v = mCachedVolume; 4859 vl *= v; 4860 vr *= v; 4861 // re-combine into U4.16 4862 vlr = (vr << 16) | (vl & 0xFFFF); 4863 // FIXME look at mute, pause, and stop flags 4864 return vlr; 4865} 4866 4867status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4868{ 4869 if (mState == TERMINATED || mState == PAUSED || 4870 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4871 (mState == STOPPED)))) { 4872 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4873 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4874 event->cancel(); 4875 return INVALID_OPERATION; 4876 } 4877 (void) TrackBase::setSyncEvent(event); 4878 return NO_ERROR; 4879} 4880 4881bool AudioFlinger::PlaybackThread::Track::isOut() const 4882{ 4883 return true; 4884} 4885 4886// timed audio tracks 4887 4888sp<AudioFlinger::PlaybackThread::TimedTrack> 4889AudioFlinger::PlaybackThread::TimedTrack::create( 4890 PlaybackThread *thread, 4891 const sp<Client>& client, 4892 audio_stream_type_t streamType, 4893 uint32_t sampleRate, 4894 audio_format_t format, 4895 audio_channel_mask_t channelMask, 4896 size_t frameCount, 4897 const sp<IMemory>& sharedBuffer, 4898 int sessionId) { 4899 if (!client->reserveTimedTrack()) 4900 return 0; 4901 4902 return new TimedTrack( 4903 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4904 sharedBuffer, sessionId); 4905} 4906 4907AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4908 PlaybackThread *thread, 4909 const sp<Client>& client, 4910 audio_stream_type_t streamType, 4911 uint32_t sampleRate, 4912 audio_format_t format, 4913 audio_channel_mask_t channelMask, 4914 size_t frameCount, 4915 const sp<IMemory>& sharedBuffer, 4916 int sessionId) 4917 : Track(thread, client, streamType, sampleRate, format, channelMask, 4918 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4919 mQueueHeadInFlight(false), 4920 mTrimQueueHeadOnRelease(false), 4921 mFramesPendingInQueue(0), 4922 mTimedSilenceBuffer(NULL), 4923 mTimedSilenceBufferSize(0), 4924 mTimedAudioOutputOnTime(false), 4925 mMediaTimeTransformValid(false) 4926{ 4927 LocalClock lc; 4928 mLocalTimeFreq = lc.getLocalFreq(); 4929 4930 mLocalTimeToSampleTransform.a_zero = 0; 4931 mLocalTimeToSampleTransform.b_zero = 0; 4932 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4933 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4934 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4935 &mLocalTimeToSampleTransform.a_to_b_denom); 4936 4937 mMediaTimeToSampleTransform.a_zero = 0; 4938 mMediaTimeToSampleTransform.b_zero = 0; 4939 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4940 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4941 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4942 &mMediaTimeToSampleTransform.a_to_b_denom); 4943} 4944 4945AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4946 mClient->releaseTimedTrack(); 4947 delete [] mTimedSilenceBuffer; 4948} 4949 4950status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4951 size_t size, sp<IMemory>* buffer) { 4952 4953 Mutex::Autolock _l(mTimedBufferQueueLock); 4954 4955 trimTimedBufferQueue_l(); 4956 4957 // lazily initialize the shared memory heap for timed buffers 4958 if (mTimedMemoryDealer == NULL) { 4959 const int kTimedBufferHeapSize = 512 << 10; 4960 4961 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4962 "AudioFlingerTimed"); 4963 if (mTimedMemoryDealer == NULL) 4964 return NO_MEMORY; 4965 } 4966 4967 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4968 if (newBuffer == NULL) { 4969 newBuffer = mTimedMemoryDealer->allocate(size); 4970 if (newBuffer == NULL) 4971 return NO_MEMORY; 4972 } 4973 4974 *buffer = newBuffer; 4975 return NO_ERROR; 4976} 4977 4978// caller must hold mTimedBufferQueueLock 4979void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4980 int64_t mediaTimeNow; 4981 { 4982 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4983 if (!mMediaTimeTransformValid) 4984 return; 4985 4986 int64_t targetTimeNow; 4987 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4988 ? mCCHelper.getCommonTime(&targetTimeNow) 4989 : mCCHelper.getLocalTime(&targetTimeNow); 4990 4991 if (OK != res) 4992 return; 4993 4994 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4995 &mediaTimeNow)) { 4996 return; 4997 } 4998 } 4999 5000 size_t trimEnd; 5001 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 5002 int64_t bufEnd; 5003 5004 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 5005 // We have a next buffer. Just use its PTS as the PTS of the frame 5006 // following the last frame in this buffer. If the stream is sparse 5007 // (ie, there are deliberate gaps left in the stream which should be 5008 // filled with silence by the TimedAudioTrack), then this can result 5009 // in one extra buffer being left un-trimmed when it could have 5010 // been. In general, this is not typical, and we would rather 5011 // optimized away the TS calculation below for the more common case 5012 // where PTSes are contiguous. 5013 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5014 } else { 5015 // We have no next buffer. Compute the PTS of the frame following 5016 // the last frame in this buffer by computing the duration of of 5017 // this frame in media time units and adding it to the PTS of the 5018 // buffer. 5019 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5020 / mFrameSize; 5021 5022 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5023 &bufEnd)) { 5024 ALOGE("Failed to convert frame count of %lld to media time" 5025 " duration" " (scale factor %d/%u) in %s", 5026 frameCount, 5027 mMediaTimeToSampleTransform.a_to_b_numer, 5028 mMediaTimeToSampleTransform.a_to_b_denom, 5029 __PRETTY_FUNCTION__); 5030 break; 5031 } 5032 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5033 } 5034 5035 if (bufEnd > mediaTimeNow) 5036 break; 5037 5038 // Is the buffer we want to use in the middle of a mix operation right 5039 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5040 // from the mixer which should be coming back shortly. 5041 if (!trimEnd && mQueueHeadInFlight) { 5042 mTrimQueueHeadOnRelease = true; 5043 } 5044 } 5045 5046 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5047 if (trimStart < trimEnd) { 5048 // Update the bookkeeping for framesReady() 5049 for (size_t i = trimStart; i < trimEnd; ++i) { 5050 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5051 } 5052 5053 // Now actually remove the buffers from the queue. 5054 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5055 } 5056} 5057 5058void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5059 const char* logTag) { 5060 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5061 "%s called (reason \"%s\"), but timed buffer queue has no" 5062 " elements to trim.", __FUNCTION__, logTag); 5063 5064 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5065 mTimedBufferQueue.removeAt(0); 5066} 5067 5068void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5069 const TimedBuffer& buf, 5070 const char* logTag) { 5071 uint32_t bufBytes = buf.buffer()->size(); 5072 uint32_t consumedAlready = buf.position(); 5073 5074 ALOG_ASSERT(consumedAlready <= bufBytes, 5075 "Bad bookkeeping while updating frames pending. Timed buffer is" 5076 " only %u bytes long, but claims to have consumed %u" 5077 " bytes. (update reason: \"%s\")", 5078 bufBytes, consumedAlready, logTag); 5079 5080 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 5081 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5082 "Bad bookkeeping while updating frames pending. Should have at" 5083 " least %u queued frames, but we think we have only %u. (update" 5084 " reason: \"%s\")", 5085 bufFrames, mFramesPendingInQueue, logTag); 5086 5087 mFramesPendingInQueue -= bufFrames; 5088} 5089 5090status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5091 const sp<IMemory>& buffer, int64_t pts) { 5092 5093 { 5094 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5095 if (!mMediaTimeTransformValid) 5096 return INVALID_OPERATION; 5097 } 5098 5099 Mutex::Autolock _l(mTimedBufferQueueLock); 5100 5101 uint32_t bufFrames = buffer->size() / mFrameSize; 5102 mFramesPendingInQueue += bufFrames; 5103 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5104 5105 return NO_ERROR; 5106} 5107 5108status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5109 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5110 5111 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5112 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5113 target); 5114 5115 if (!(target == TimedAudioTrack::LOCAL_TIME || 5116 target == TimedAudioTrack::COMMON_TIME)) { 5117 return BAD_VALUE; 5118 } 5119 5120 Mutex::Autolock lock(mMediaTimeTransformLock); 5121 mMediaTimeTransform = xform; 5122 mMediaTimeTransformTarget = target; 5123 mMediaTimeTransformValid = true; 5124 5125 return NO_ERROR; 5126} 5127 5128#define min(a, b) ((a) < (b) ? (a) : (b)) 5129 5130// implementation of getNextBuffer for tracks whose buffers have timestamps 5131status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5132 AudioBufferProvider::Buffer* buffer, int64_t pts) 5133{ 5134 if (pts == AudioBufferProvider::kInvalidPTS) { 5135 buffer->raw = NULL; 5136 buffer->frameCount = 0; 5137 mTimedAudioOutputOnTime = false; 5138 return INVALID_OPERATION; 5139 } 5140 5141 Mutex::Autolock _l(mTimedBufferQueueLock); 5142 5143 ALOG_ASSERT(!mQueueHeadInFlight, 5144 "getNextBuffer called without releaseBuffer!"); 5145 5146 while (true) { 5147 5148 // if we have no timed buffers, then fail 5149 if (mTimedBufferQueue.isEmpty()) { 5150 buffer->raw = NULL; 5151 buffer->frameCount = 0; 5152 return NOT_ENOUGH_DATA; 5153 } 5154 5155 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5156 5157 // calculate the PTS of the head of the timed buffer queue expressed in 5158 // local time 5159 int64_t headLocalPTS; 5160 { 5161 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5162 5163 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5164 5165 if (mMediaTimeTransform.a_to_b_denom == 0) { 5166 // the transform represents a pause, so yield silence 5167 timedYieldSilence_l(buffer->frameCount, buffer); 5168 return NO_ERROR; 5169 } 5170 5171 int64_t transformedPTS; 5172 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5173 &transformedPTS)) { 5174 // the transform failed. this shouldn't happen, but if it does 5175 // then just drop this buffer 5176 ALOGW("timedGetNextBuffer transform failed"); 5177 buffer->raw = NULL; 5178 buffer->frameCount = 0; 5179 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5180 return NO_ERROR; 5181 } 5182 5183 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5184 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5185 &headLocalPTS)) { 5186 buffer->raw = NULL; 5187 buffer->frameCount = 0; 5188 return INVALID_OPERATION; 5189 } 5190 } else { 5191 headLocalPTS = transformedPTS; 5192 } 5193 } 5194 5195 // adjust the head buffer's PTS to reflect the portion of the head buffer 5196 // that has already been consumed 5197 int64_t effectivePTS = headLocalPTS + 5198 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 5199 5200 // Calculate the delta in samples between the head of the input buffer 5201 // queue and the start of the next output buffer that will be written. 5202 // If the transformation fails because of over or underflow, it means 5203 // that the sample's position in the output stream is so far out of 5204 // whack that it should just be dropped. 5205 int64_t sampleDelta; 5206 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5207 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5208 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5209 " mix"); 5210 continue; 5211 } 5212 if (!mLocalTimeToSampleTransform.doForwardTransform( 5213 (effectivePTS - pts) << 32, &sampleDelta)) { 5214 ALOGV("*** too late during sample rate transform: dropped buffer"); 5215 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5216 continue; 5217 } 5218 5219 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5220 " sampleDelta=[%d.%08x]", 5221 head.pts(), head.position(), pts, 5222 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5223 + (sampleDelta >> 32)), 5224 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5225 5226 // if the delta between the ideal placement for the next input sample and 5227 // the current output position is within this threshold, then we will 5228 // concatenate the next input samples to the previous output 5229 const int64_t kSampleContinuityThreshold = 5230 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5231 5232 // if this is the first buffer of audio that we're emitting from this track 5233 // then it should be almost exactly on time. 5234 const int64_t kSampleStartupThreshold = 1LL << 32; 5235 5236 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5237 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5238 // the next input is close enough to being on time, so concatenate it 5239 // with the last output 5240 timedYieldSamples_l(buffer); 5241 5242 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5243 head.position(), buffer->frameCount); 5244 return NO_ERROR; 5245 } 5246 5247 // Looks like our output is not on time. Reset our on timed status. 5248 // Next time we mix samples from our input queue, then should be within 5249 // the StartupThreshold. 5250 mTimedAudioOutputOnTime = false; 5251 if (sampleDelta > 0) { 5252 // the gap between the current output position and the proper start of 5253 // the next input sample is too big, so fill it with silence 5254 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5255 5256 timedYieldSilence_l(framesUntilNextInput, buffer); 5257 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5258 return NO_ERROR; 5259 } else { 5260 // the next input sample is late 5261 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5262 size_t onTimeSamplePosition = 5263 head.position() + lateFrames * mFrameSize; 5264 5265 if (onTimeSamplePosition > head.buffer()->size()) { 5266 // all the remaining samples in the head are too late, so 5267 // drop it and move on 5268 ALOGV("*** too late: dropped buffer"); 5269 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5270 continue; 5271 } else { 5272 // skip over the late samples 5273 head.setPosition(onTimeSamplePosition); 5274 5275 // yield the available samples 5276 timedYieldSamples_l(buffer); 5277 5278 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5279 return NO_ERROR; 5280 } 5281 } 5282 } 5283} 5284 5285// Yield samples from the timed buffer queue head up to the given output 5286// buffer's capacity. 5287// 5288// Caller must hold mTimedBufferQueueLock 5289void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5290 AudioBufferProvider::Buffer* buffer) { 5291 5292 const TimedBuffer& head = mTimedBufferQueue[0]; 5293 5294 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5295 head.position()); 5296 5297 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5298 mFrameSize); 5299 size_t framesRequested = buffer->frameCount; 5300 buffer->frameCount = min(framesLeftInHead, framesRequested); 5301 5302 mQueueHeadInFlight = true; 5303 mTimedAudioOutputOnTime = true; 5304} 5305 5306// Yield samples of silence up to the given output buffer's capacity 5307// 5308// Caller must hold mTimedBufferQueueLock 5309void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5310 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5311 5312 // lazily allocate a buffer filled with silence 5313 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 5314 delete [] mTimedSilenceBuffer; 5315 mTimedSilenceBufferSize = numFrames * mFrameSize; 5316 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5317 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5318 } 5319 5320 buffer->raw = mTimedSilenceBuffer; 5321 size_t framesRequested = buffer->frameCount; 5322 buffer->frameCount = min(numFrames, framesRequested); 5323 5324 mTimedAudioOutputOnTime = false; 5325} 5326 5327// AudioBufferProvider interface 5328void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5329 AudioBufferProvider::Buffer* buffer) { 5330 5331 Mutex::Autolock _l(mTimedBufferQueueLock); 5332 5333 // If the buffer which was just released is part of the buffer at the head 5334 // of the queue, be sure to update the amt of the buffer which has been 5335 // consumed. If the buffer being returned is not part of the head of the 5336 // queue, its either because the buffer is part of the silence buffer, or 5337 // because the head of the timed queue was trimmed after the mixer called 5338 // getNextBuffer but before the mixer called releaseBuffer. 5339 if (buffer->raw == mTimedSilenceBuffer) { 5340 ALOG_ASSERT(!mQueueHeadInFlight, 5341 "Queue head in flight during release of silence buffer!"); 5342 goto done; 5343 } 5344 5345 ALOG_ASSERT(mQueueHeadInFlight, 5346 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5347 " head in flight."); 5348 5349 if (mTimedBufferQueue.size()) { 5350 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5351 5352 void* start = head.buffer()->pointer(); 5353 void* end = reinterpret_cast<void*>( 5354 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5355 + head.buffer()->size()); 5356 5357 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5358 "released buffer not within the head of the timed buffer" 5359 " queue; qHead = [%p, %p], released buffer = %p", 5360 start, end, buffer->raw); 5361 5362 head.setPosition(head.position() + 5363 (buffer->frameCount * mFrameSize)); 5364 mQueueHeadInFlight = false; 5365 5366 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5367 "Bad bookkeeping during releaseBuffer! Should have at" 5368 " least %u queued frames, but we think we have only %u", 5369 buffer->frameCount, mFramesPendingInQueue); 5370 5371 mFramesPendingInQueue -= buffer->frameCount; 5372 5373 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5374 || mTrimQueueHeadOnRelease) { 5375 trimTimedBufferQueueHead_l("releaseBuffer"); 5376 mTrimQueueHeadOnRelease = false; 5377 } 5378 } else { 5379 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5380 " buffers in the timed buffer queue"); 5381 } 5382 5383done: 5384 buffer->raw = 0; 5385 buffer->frameCount = 0; 5386} 5387 5388size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5389 Mutex::Autolock _l(mTimedBufferQueueLock); 5390 return mFramesPendingInQueue; 5391} 5392 5393AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5394 : mPTS(0), mPosition(0) {} 5395 5396AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5397 const sp<IMemory>& buffer, int64_t pts) 5398 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5399 5400// ---------------------------------------------------------------------------- 5401 5402// RecordTrack constructor must be called with AudioFlinger::mLock held 5403AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5404 RecordThread *thread, 5405 const sp<Client>& client, 5406 uint32_t sampleRate, 5407 audio_format_t format, 5408 audio_channel_mask_t channelMask, 5409 size_t frameCount, 5410 int sessionId) 5411 : TrackBase(thread, client, sampleRate, format, 5412 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5413 mOverflow(false) 5414{ 5415 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5416} 5417 5418AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5419{ 5420 ALOGV("%s", __func__); 5421} 5422 5423// AudioBufferProvider interface 5424status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5425 int64_t pts) 5426{ 5427 audio_track_cblk_t* cblk = this->cblk(); 5428 uint32_t framesAvail; 5429 uint32_t framesReq = buffer->frameCount; 5430 5431 // Check if last stepServer failed, try to step now 5432 if (mStepServerFailed) { 5433 if (!step()) goto getNextBuffer_exit; 5434 ALOGV("stepServer recovered"); 5435 mStepServerFailed = false; 5436 } 5437 5438 // FIXME lock is not actually held, so overrun is possible 5439 framesAvail = cblk->framesAvailableIn_l(mFrameCount); 5440 5441 if (CC_LIKELY(framesAvail)) { 5442 uint32_t s = cblk->server; 5443 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 5444 5445 if (framesReq > framesAvail) { 5446 framesReq = framesAvail; 5447 } 5448 if (framesReq > bufferEnd - s) { 5449 framesReq = bufferEnd - s; 5450 } 5451 5452 buffer->raw = getBuffer(s, framesReq); 5453 buffer->frameCount = framesReq; 5454 return NO_ERROR; 5455 } 5456 5457getNextBuffer_exit: 5458 buffer->raw = NULL; 5459 buffer->frameCount = 0; 5460 return NOT_ENOUGH_DATA; 5461} 5462 5463status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5464 int triggerSession) 5465{ 5466 sp<ThreadBase> thread = mThread.promote(); 5467 if (thread != 0) { 5468 RecordThread *recordThread = (RecordThread *)thread.get(); 5469 return recordThread->start(this, event, triggerSession); 5470 } else { 5471 return BAD_VALUE; 5472 } 5473} 5474 5475void AudioFlinger::RecordThread::RecordTrack::stop() 5476{ 5477 sp<ThreadBase> thread = mThread.promote(); 5478 if (thread != 0) { 5479 RecordThread *recordThread = (RecordThread *)thread.get(); 5480 recordThread->mLock.lock(); 5481 bool doStop = recordThread->stop_l(this); 5482 if (doStop) { 5483 TrackBase::reset(); 5484 // Force overrun condition to avoid false overrun callback until first data is 5485 // read from buffer 5486 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5487 } 5488 recordThread->mLock.unlock(); 5489 if (doStop) { 5490 AudioSystem::stopInput(recordThread->id()); 5491 } 5492 } 5493} 5494 5495/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5496{ 5497 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 5498} 5499 5500void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5501{ 5502 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5503 (mClient == 0) ? getpid_cached : mClient->pid(), 5504 mFormat, 5505 mChannelMask, 5506 mSessionId, 5507 mStepCount, 5508 mState, 5509 mCblk->sampleRate, 5510 mCblk->server, 5511 mCblk->user, 5512 mFrameCount); 5513} 5514 5515bool AudioFlinger::RecordThread::RecordTrack::isOut() const 5516{ 5517 return false; 5518} 5519 5520// ---------------------------------------------------------------------------- 5521 5522AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5523 PlaybackThread *playbackThread, 5524 DuplicatingThread *sourceThread, 5525 uint32_t sampleRate, 5526 audio_format_t format, 5527 audio_channel_mask_t channelMask, 5528 size_t frameCount) 5529 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5530 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5531 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 5532{ 5533 5534 if (mCblk != NULL) { 5535 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5536 mOutBuffer.frameCount = 0; 5537 playbackThread->mTracks.add(this); 5538 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5539 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 5540 mCblk, mBuffer, mCblk->buffers, 5541 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5542 } else { 5543 ALOGW("Error creating output track on thread %p", playbackThread); 5544 } 5545} 5546 5547AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5548{ 5549 clearBufferQueue(); 5550} 5551 5552status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5553 int triggerSession) 5554{ 5555 status_t status = Track::start(event, triggerSession); 5556 if (status != NO_ERROR) { 5557 return status; 5558 } 5559 5560 mActive = true; 5561 mRetryCount = 127; 5562 return status; 5563} 5564 5565void AudioFlinger::PlaybackThread::OutputTrack::stop() 5566{ 5567 Track::stop(); 5568 clearBufferQueue(); 5569 mOutBuffer.frameCount = 0; 5570 mActive = false; 5571} 5572 5573bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5574{ 5575 Buffer *pInBuffer; 5576 Buffer inBuffer; 5577 uint32_t channelCount = mChannelCount; 5578 bool outputBufferFull = false; 5579 inBuffer.frameCount = frames; 5580 inBuffer.i16 = data; 5581 5582 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5583 5584 if (!mActive && frames != 0) { 5585 start(); 5586 sp<ThreadBase> thread = mThread.promote(); 5587 if (thread != 0) { 5588 MixerThread *mixerThread = (MixerThread *)thread.get(); 5589 if (mFrameCount > frames){ 5590 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5591 uint32_t startFrames = (mFrameCount - frames); 5592 pInBuffer = new Buffer; 5593 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5594 pInBuffer->frameCount = startFrames; 5595 pInBuffer->i16 = pInBuffer->mBuffer; 5596 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5597 mBufferQueue.add(pInBuffer); 5598 } else { 5599 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5600 } 5601 } 5602 } 5603 } 5604 5605 while (waitTimeLeftMs) { 5606 // First write pending buffers, then new data 5607 if (mBufferQueue.size()) { 5608 pInBuffer = mBufferQueue.itemAt(0); 5609 } else { 5610 pInBuffer = &inBuffer; 5611 } 5612 5613 if (pInBuffer->frameCount == 0) { 5614 break; 5615 } 5616 5617 if (mOutBuffer.frameCount == 0) { 5618 mOutBuffer.frameCount = pInBuffer->frameCount; 5619 nsecs_t startTime = systemTime(); 5620 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5621 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5622 mThread.unsafe_get()); 5623 outputBufferFull = true; 5624 break; 5625 } 5626 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5627 if (waitTimeLeftMs >= waitTimeMs) { 5628 waitTimeLeftMs -= waitTimeMs; 5629 } else { 5630 waitTimeLeftMs = 0; 5631 } 5632 } 5633 5634 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5635 pInBuffer->frameCount; 5636 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5637 mCblk->stepUserOut(outFrames, mFrameCount); 5638 pInBuffer->frameCount -= outFrames; 5639 pInBuffer->i16 += outFrames * channelCount; 5640 mOutBuffer.frameCount -= outFrames; 5641 mOutBuffer.i16 += outFrames * channelCount; 5642 5643 if (pInBuffer->frameCount == 0) { 5644 if (mBufferQueue.size()) { 5645 mBufferQueue.removeAt(0); 5646 delete [] pInBuffer->mBuffer; 5647 delete pInBuffer; 5648 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5649 mThread.unsafe_get(), mBufferQueue.size()); 5650 } else { 5651 break; 5652 } 5653 } 5654 } 5655 5656 // If we could not write all frames, allocate a buffer and queue it for next time. 5657 if (inBuffer.frameCount) { 5658 sp<ThreadBase> thread = mThread.promote(); 5659 if (thread != 0 && !thread->standby()) { 5660 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5661 pInBuffer = new Buffer; 5662 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5663 pInBuffer->frameCount = inBuffer.frameCount; 5664 pInBuffer->i16 = pInBuffer->mBuffer; 5665 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5666 sizeof(int16_t)); 5667 mBufferQueue.add(pInBuffer); 5668 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5669 mThread.unsafe_get(), mBufferQueue.size()); 5670 } else { 5671 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5672 mThread.unsafe_get(), this); 5673 } 5674 } 5675 } 5676 5677 // Calling write() with a 0 length buffer, means that no more data will be written: 5678 // If no more buffers are pending, fill output track buffer to make sure it is started 5679 // by output mixer. 5680 if (frames == 0 && mBufferQueue.size() == 0) { 5681 if (mCblk->user < mFrameCount) { 5682 frames = mFrameCount - mCblk->user; 5683 pInBuffer = new Buffer; 5684 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5685 pInBuffer->frameCount = frames; 5686 pInBuffer->i16 = pInBuffer->mBuffer; 5687 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5688 mBufferQueue.add(pInBuffer); 5689 } else if (mActive) { 5690 stop(); 5691 } 5692 } 5693 5694 return outputBufferFull; 5695} 5696 5697status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5698 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5699{ 5700 int active; 5701 status_t result; 5702 audio_track_cblk_t* cblk = mCblk; 5703 uint32_t framesReq = buffer->frameCount; 5704 5705 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5706 buffer->frameCount = 0; 5707 5708 uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); 5709 5710 5711 if (framesAvail == 0) { 5712 Mutex::Autolock _l(cblk->lock); 5713 goto start_loop_here; 5714 while (framesAvail == 0) { 5715 active = mActive; 5716 if (CC_UNLIKELY(!active)) { 5717 ALOGV("Not active and NO_MORE_BUFFERS"); 5718 return NO_MORE_BUFFERS; 5719 } 5720 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5721 if (result != NO_ERROR) { 5722 return NO_MORE_BUFFERS; 5723 } 5724 // read the server count again 5725 start_loop_here: 5726 framesAvail = cblk->framesAvailableOut_l(mFrameCount); 5727 } 5728 } 5729 5730// if (framesAvail < framesReq) { 5731// return NO_MORE_BUFFERS; 5732// } 5733 5734 if (framesReq > framesAvail) { 5735 framesReq = framesAvail; 5736 } 5737 5738 uint32_t u = cblk->user; 5739 uint32_t bufferEnd = cblk->userBase + mFrameCount; 5740 5741 if (framesReq > bufferEnd - u) { 5742 framesReq = bufferEnd - u; 5743 } 5744 5745 buffer->frameCount = framesReq; 5746 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 5747 return NO_ERROR; 5748} 5749 5750 5751void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5752{ 5753 size_t size = mBufferQueue.size(); 5754 5755 for (size_t i = 0; i < size; i++) { 5756 Buffer *pBuffer = mBufferQueue.itemAt(i); 5757 delete [] pBuffer->mBuffer; 5758 delete pBuffer; 5759 } 5760 mBufferQueue.clear(); 5761} 5762 5763// ---------------------------------------------------------------------------- 5764 5765AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5766 : RefBase(), 5767 mAudioFlinger(audioFlinger), 5768 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5769 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5770 mPid(pid), 5771 mTimedTrackCount(0) 5772{ 5773 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5774} 5775 5776// Client destructor must be called with AudioFlinger::mLock held 5777AudioFlinger::Client::~Client() 5778{ 5779 mAudioFlinger->removeClient_l(mPid); 5780} 5781 5782sp<MemoryDealer> AudioFlinger::Client::heap() const 5783{ 5784 return mMemoryDealer; 5785} 5786 5787// Reserve one of the limited slots for a timed audio track associated 5788// with this client 5789bool AudioFlinger::Client::reserveTimedTrack() 5790{ 5791 const int kMaxTimedTracksPerClient = 4; 5792 5793 Mutex::Autolock _l(mTimedTrackLock); 5794 5795 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5796 ALOGW("can not create timed track - pid %d has exceeded the limit", 5797 mPid); 5798 return false; 5799 } 5800 5801 mTimedTrackCount++; 5802 return true; 5803} 5804 5805// Release a slot for a timed audio track 5806void AudioFlinger::Client::releaseTimedTrack() 5807{ 5808 Mutex::Autolock _l(mTimedTrackLock); 5809 mTimedTrackCount--; 5810} 5811 5812// ---------------------------------------------------------------------------- 5813 5814AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5815 const sp<IAudioFlingerClient>& client, 5816 pid_t pid) 5817 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5818{ 5819} 5820 5821AudioFlinger::NotificationClient::~NotificationClient() 5822{ 5823} 5824 5825void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5826{ 5827 sp<NotificationClient> keep(this); 5828 mAudioFlinger->removeNotificationClient(mPid); 5829} 5830 5831// ---------------------------------------------------------------------------- 5832 5833AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5834 : BnAudioTrack(), 5835 mTrack(track) 5836{ 5837} 5838 5839AudioFlinger::TrackHandle::~TrackHandle() { 5840 // just stop the track on deletion, associated resources 5841 // will be freed from the main thread once all pending buffers have 5842 // been played. Unless it's not in the active track list, in which 5843 // case we free everything now... 5844 mTrack->destroy(); 5845} 5846 5847sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5848 return mTrack->getCblk(); 5849} 5850 5851status_t AudioFlinger::TrackHandle::start() { 5852 return mTrack->start(); 5853} 5854 5855void AudioFlinger::TrackHandle::stop() { 5856 mTrack->stop(); 5857} 5858 5859void AudioFlinger::TrackHandle::flush() { 5860 mTrack->flush(); 5861} 5862 5863void AudioFlinger::TrackHandle::mute(bool e) { 5864 mTrack->mute(e); 5865} 5866 5867void AudioFlinger::TrackHandle::pause() { 5868 mTrack->pause(); 5869} 5870 5871status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5872{ 5873 return mTrack->attachAuxEffect(EffectId); 5874} 5875 5876status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5877 sp<IMemory>* buffer) { 5878 if (!mTrack->isTimedTrack()) 5879 return INVALID_OPERATION; 5880 5881 PlaybackThread::TimedTrack* tt = 5882 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5883 return tt->allocateTimedBuffer(size, buffer); 5884} 5885 5886status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5887 int64_t pts) { 5888 if (!mTrack->isTimedTrack()) 5889 return INVALID_OPERATION; 5890 5891 PlaybackThread::TimedTrack* tt = 5892 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5893 return tt->queueTimedBuffer(buffer, pts); 5894} 5895 5896status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5897 const LinearTransform& xform, int target) { 5898 5899 if (!mTrack->isTimedTrack()) 5900 return INVALID_OPERATION; 5901 5902 PlaybackThread::TimedTrack* tt = 5903 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5904 return tt->setMediaTimeTransform( 5905 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5906} 5907 5908status_t AudioFlinger::TrackHandle::onTransact( 5909 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5910{ 5911 return BnAudioTrack::onTransact(code, data, reply, flags); 5912} 5913 5914// ---------------------------------------------------------------------------- 5915 5916sp<IAudioRecord> AudioFlinger::openRecord( 5917 pid_t pid, 5918 audio_io_handle_t input, 5919 uint32_t sampleRate, 5920 audio_format_t format, 5921 audio_channel_mask_t channelMask, 5922 size_t frameCount, 5923 IAudioFlinger::track_flags_t flags, 5924 pid_t tid, 5925 int *sessionId, 5926 status_t *status) 5927{ 5928 sp<RecordThread::RecordTrack> recordTrack; 5929 sp<RecordHandle> recordHandle; 5930 sp<Client> client; 5931 status_t lStatus; 5932 RecordThread *thread; 5933 size_t inFrameCount; 5934 int lSessionId; 5935 5936 // check calling permissions 5937 if (!recordingAllowed()) { 5938 lStatus = PERMISSION_DENIED; 5939 goto Exit; 5940 } 5941 5942 // add client to list 5943 { // scope for mLock 5944 Mutex::Autolock _l(mLock); 5945 thread = checkRecordThread_l(input); 5946 if (thread == NULL) { 5947 lStatus = BAD_VALUE; 5948 goto Exit; 5949 } 5950 5951 client = registerPid_l(pid); 5952 5953 // If no audio session id is provided, create one here 5954 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5955 lSessionId = *sessionId; 5956 } else { 5957 lSessionId = nextUniqueId(); 5958 if (sessionId != NULL) { 5959 *sessionId = lSessionId; 5960 } 5961 } 5962 // create new record track. 5963 // The record track uses one track in mHardwareMixerThread by convention. 5964 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5965 frameCount, lSessionId, flags, tid, &lStatus); 5966 } 5967 if (lStatus != NO_ERROR) { 5968 // remove local strong reference to Client before deleting the RecordTrack so that the 5969 // Client destructor is called by the TrackBase destructor with mLock held 5970 client.clear(); 5971 recordTrack.clear(); 5972 goto Exit; 5973 } 5974 5975 // return to handle to client 5976 recordHandle = new RecordHandle(recordTrack); 5977 lStatus = NO_ERROR; 5978 5979Exit: 5980 if (status) { 5981 *status = lStatus; 5982 } 5983 return recordHandle; 5984} 5985 5986// ---------------------------------------------------------------------------- 5987 5988AudioFlinger::RecordHandle::RecordHandle( 5989 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5990 : BnAudioRecord(), 5991 mRecordTrack(recordTrack) 5992{ 5993} 5994 5995AudioFlinger::RecordHandle::~RecordHandle() { 5996 stop_nonvirtual(); 5997 mRecordTrack->destroy(); 5998} 5999 6000sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 6001 return mRecordTrack->getCblk(); 6002} 6003 6004status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6005 int triggerSession) { 6006 ALOGV("RecordHandle::start()"); 6007 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6008} 6009 6010void AudioFlinger::RecordHandle::stop() { 6011 stop_nonvirtual(); 6012} 6013 6014void AudioFlinger::RecordHandle::stop_nonvirtual() { 6015 ALOGV("RecordHandle::stop()"); 6016 mRecordTrack->stop(); 6017} 6018 6019status_t AudioFlinger::RecordHandle::onTransact( 6020 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6021{ 6022 return BnAudioRecord::onTransact(code, data, reply, flags); 6023} 6024 6025// ---------------------------------------------------------------------------- 6026 6027AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6028 AudioStreamIn *input, 6029 uint32_t sampleRate, 6030 audio_channel_mask_t channelMask, 6031 audio_io_handle_t id, 6032 audio_devices_t device, 6033 const sp<NBAIO_Sink>& teeSink) : 6034 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6035 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6036 // mRsmpInIndex and mInputBytes set by readInputParameters() 6037 mReqChannelCount(popcount(channelMask)), 6038 mReqSampleRate(sampleRate), 6039 // mBytesRead is only meaningful while active, and so is cleared in start() 6040 // (but might be better to also clear here for dump?) 6041 mTeeSink(teeSink) 6042{ 6043 snprintf(mName, kNameLength, "AudioIn_%X", id); 6044 6045 readInputParameters(); 6046 6047} 6048 6049 6050AudioFlinger::RecordThread::~RecordThread() 6051{ 6052 delete[] mRsmpInBuffer; 6053 delete mResampler; 6054 delete[] mRsmpOutBuffer; 6055} 6056 6057void AudioFlinger::RecordThread::onFirstRef() 6058{ 6059 run(mName, PRIORITY_URGENT_AUDIO); 6060} 6061 6062status_t AudioFlinger::RecordThread::readyToRun() 6063{ 6064 status_t status = initCheck(); 6065 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6066 return status; 6067} 6068 6069bool AudioFlinger::RecordThread::threadLoop() 6070{ 6071 AudioBufferProvider::Buffer buffer; 6072 sp<RecordTrack> activeTrack; 6073 Vector< sp<EffectChain> > effectChains; 6074 6075 nsecs_t lastWarning = 0; 6076 6077 inputStandBy(); 6078 acquireWakeLock(); 6079 6080 // used to verify we've read at least once before evaluating how many bytes were read 6081 bool readOnce = false; 6082 6083 // start recording 6084 while (!exitPending()) { 6085 6086 processConfigEvents(); 6087 6088 { // scope for mLock 6089 Mutex::Autolock _l(mLock); 6090 checkForNewParameters_l(); 6091 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6092 standby(); 6093 6094 if (exitPending()) break; 6095 6096 releaseWakeLock_l(); 6097 ALOGV("RecordThread: loop stopping"); 6098 // go to sleep 6099 mWaitWorkCV.wait(mLock); 6100 ALOGV("RecordThread: loop starting"); 6101 acquireWakeLock_l(); 6102 continue; 6103 } 6104 if (mActiveTrack != 0) { 6105 if (mActiveTrack->mState == TrackBase::PAUSING) { 6106 standby(); 6107 mActiveTrack.clear(); 6108 mStartStopCond.broadcast(); 6109 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6110 if (mReqChannelCount != mActiveTrack->channelCount()) { 6111 mActiveTrack.clear(); 6112 mStartStopCond.broadcast(); 6113 } else if (readOnce) { 6114 // record start succeeds only if first read from audio input 6115 // succeeds 6116 if (mBytesRead >= 0) { 6117 mActiveTrack->mState = TrackBase::ACTIVE; 6118 } else { 6119 mActiveTrack.clear(); 6120 } 6121 mStartStopCond.broadcast(); 6122 } 6123 mStandby = false; 6124 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6125 removeTrack_l(mActiveTrack); 6126 mActiveTrack.clear(); 6127 } 6128 } 6129 lockEffectChains_l(effectChains); 6130 } 6131 6132 if (mActiveTrack != 0) { 6133 if (mActiveTrack->mState != TrackBase::ACTIVE && 6134 mActiveTrack->mState != TrackBase::RESUMING) { 6135 unlockEffectChains(effectChains); 6136 usleep(kRecordThreadSleepUs); 6137 continue; 6138 } 6139 for (size_t i = 0; i < effectChains.size(); i ++) { 6140 effectChains[i]->process_l(); 6141 } 6142 6143 buffer.frameCount = mFrameCount; 6144 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6145 readOnce = true; 6146 size_t framesOut = buffer.frameCount; 6147 if (mResampler == NULL) { 6148 // no resampling 6149 while (framesOut) { 6150 size_t framesIn = mFrameCount - mRsmpInIndex; 6151 if (framesIn) { 6152 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6153 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6154 mActiveTrack->mFrameSize; 6155 if (framesIn > framesOut) 6156 framesIn = framesOut; 6157 mRsmpInIndex += framesIn; 6158 framesOut -= framesIn; 6159 if ((int)mChannelCount == mReqChannelCount || 6160 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6161 memcpy(dst, src, framesIn * mFrameSize); 6162 } else { 6163 if (mChannelCount == 1) { 6164 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6165 (int16_t *)src, framesIn); 6166 } else { 6167 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6168 (int16_t *)src, framesIn); 6169 } 6170 } 6171 } 6172 if (framesOut && mFrameCount == mRsmpInIndex) { 6173 void *readInto; 6174 if (framesOut == mFrameCount && 6175 ((int)mChannelCount == mReqChannelCount || 6176 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6177 readInto = buffer.raw; 6178 framesOut = 0; 6179 } else { 6180 readInto = mRsmpInBuffer; 6181 mRsmpInIndex = 0; 6182 } 6183 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6184 if (mBytesRead <= 0) { 6185 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6186 { 6187 ALOGE("Error reading audio input"); 6188 // Force input into standby so that it tries to 6189 // recover at next read attempt 6190 inputStandBy(); 6191 usleep(kRecordThreadSleepUs); 6192 } 6193 mRsmpInIndex = mFrameCount; 6194 framesOut = 0; 6195 buffer.frameCount = 0; 6196 } else if (mTeeSink != 0) { 6197 (void) mTeeSink->write(readInto, 6198 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6199 } 6200 } 6201 } 6202 } else { 6203 // resampling 6204 6205 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6206 // alter output frame count as if we were expecting stereo samples 6207 if (mChannelCount == 1 && mReqChannelCount == 1) { 6208 framesOut >>= 1; 6209 } 6210 mResampler->resample(mRsmpOutBuffer, framesOut, 6211 this /* AudioBufferProvider* */); 6212 // ditherAndClamp() works as long as all buffers returned by 6213 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6214 if (mChannelCount == 2 && mReqChannelCount == 1) { 6215 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6216 // the resampler always outputs stereo samples: 6217 // do post stereo to mono conversion 6218 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6219 framesOut); 6220 } else { 6221 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6222 } 6223 6224 } 6225 if (mFramestoDrop == 0) { 6226 mActiveTrack->releaseBuffer(&buffer); 6227 } else { 6228 if (mFramestoDrop > 0) { 6229 mFramestoDrop -= buffer.frameCount; 6230 if (mFramestoDrop <= 0) { 6231 clearSyncStartEvent(); 6232 } 6233 } else { 6234 mFramestoDrop += buffer.frameCount; 6235 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6236 mSyncStartEvent->isCancelled()) { 6237 ALOGW("Synced record %s, session %d, trigger session %d", 6238 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6239 mActiveTrack->sessionId(), 6240 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6241 clearSyncStartEvent(); 6242 } 6243 } 6244 } 6245 mActiveTrack->clearOverflow(); 6246 } 6247 // client isn't retrieving buffers fast enough 6248 else { 6249 if (!mActiveTrack->setOverflow()) { 6250 nsecs_t now = systemTime(); 6251 if ((now - lastWarning) > kWarningThrottleNs) { 6252 ALOGW("RecordThread: buffer overflow"); 6253 lastWarning = now; 6254 } 6255 } 6256 // Release the processor for a while before asking for a new buffer. 6257 // This will give the application more chance to read from the buffer and 6258 // clear the overflow. 6259 usleep(kRecordThreadSleepUs); 6260 } 6261 } 6262 // enable changes in effect chain 6263 unlockEffectChains(effectChains); 6264 effectChains.clear(); 6265 } 6266 6267 standby(); 6268 6269 { 6270 Mutex::Autolock _l(mLock); 6271 mActiveTrack.clear(); 6272 mStartStopCond.broadcast(); 6273 } 6274 6275 releaseWakeLock(); 6276 6277 ALOGV("RecordThread %p exiting", this); 6278 return false; 6279} 6280 6281void AudioFlinger::RecordThread::standby() 6282{ 6283 if (!mStandby) { 6284 inputStandBy(); 6285 mStandby = true; 6286 } 6287} 6288 6289void AudioFlinger::RecordThread::inputStandBy() 6290{ 6291 mInput->stream->common.standby(&mInput->stream->common); 6292} 6293 6294sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6295 const sp<AudioFlinger::Client>& client, 6296 uint32_t sampleRate, 6297 audio_format_t format, 6298 audio_channel_mask_t channelMask, 6299 size_t frameCount, 6300 int sessionId, 6301 IAudioFlinger::track_flags_t flags, 6302 pid_t tid, 6303 status_t *status) 6304{ 6305 sp<RecordTrack> track; 6306 status_t lStatus; 6307 6308 lStatus = initCheck(); 6309 if (lStatus != NO_ERROR) { 6310 ALOGE("Audio driver not initialized."); 6311 goto Exit; 6312 } 6313 6314 // FIXME use flags and tid similar to createTrack_l() 6315 6316 { // scope for mLock 6317 Mutex::Autolock _l(mLock); 6318 6319 track = new RecordTrack(this, client, sampleRate, 6320 format, channelMask, frameCount, sessionId); 6321 6322 if (track->getCblk() == 0) { 6323 lStatus = NO_MEMORY; 6324 goto Exit; 6325 } 6326 mTracks.add(track); 6327 6328 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6329 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6330 mAudioFlinger->btNrecIsOff(); 6331 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6332 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6333 } 6334 lStatus = NO_ERROR; 6335 6336Exit: 6337 if (status) { 6338 *status = lStatus; 6339 } 6340 return track; 6341} 6342 6343status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6344 AudioSystem::sync_event_t event, 6345 int triggerSession) 6346{ 6347 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6348 sp<ThreadBase> strongMe = this; 6349 status_t status = NO_ERROR; 6350 6351 if (event == AudioSystem::SYNC_EVENT_NONE) { 6352 clearSyncStartEvent(); 6353 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6354 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6355 triggerSession, 6356 recordTrack->sessionId(), 6357 syncStartEventCallback, 6358 this); 6359 // Sync event can be cancelled by the trigger session if the track is not in a 6360 // compatible state in which case we start record immediately 6361 if (mSyncStartEvent->isCancelled()) { 6362 clearSyncStartEvent(); 6363 } else { 6364 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6365 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6366 } 6367 } 6368 6369 { 6370 AutoMutex lock(mLock); 6371 if (mActiveTrack != 0) { 6372 if (recordTrack != mActiveTrack.get()) { 6373 status = -EBUSY; 6374 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6375 mActiveTrack->mState = TrackBase::ACTIVE; 6376 } 6377 return status; 6378 } 6379 6380 recordTrack->mState = TrackBase::IDLE; 6381 mActiveTrack = recordTrack; 6382 mLock.unlock(); 6383 status_t status = AudioSystem::startInput(mId); 6384 mLock.lock(); 6385 if (status != NO_ERROR) { 6386 mActiveTrack.clear(); 6387 clearSyncStartEvent(); 6388 return status; 6389 } 6390 mRsmpInIndex = mFrameCount; 6391 mBytesRead = 0; 6392 if (mResampler != NULL) { 6393 mResampler->reset(); 6394 } 6395 mActiveTrack->mState = TrackBase::RESUMING; 6396 // signal thread to start 6397 ALOGV("Signal record thread"); 6398 mWaitWorkCV.broadcast(); 6399 // do not wait for mStartStopCond if exiting 6400 if (exitPending()) { 6401 mActiveTrack.clear(); 6402 status = INVALID_OPERATION; 6403 goto startError; 6404 } 6405 mStartStopCond.wait(mLock); 6406 if (mActiveTrack == 0) { 6407 ALOGV("Record failed to start"); 6408 status = BAD_VALUE; 6409 goto startError; 6410 } 6411 ALOGV("Record started OK"); 6412 return status; 6413 } 6414startError: 6415 AudioSystem::stopInput(mId); 6416 clearSyncStartEvent(); 6417 return status; 6418} 6419 6420void AudioFlinger::RecordThread::clearSyncStartEvent() 6421{ 6422 if (mSyncStartEvent != 0) { 6423 mSyncStartEvent->cancel(); 6424 } 6425 mSyncStartEvent.clear(); 6426 mFramestoDrop = 0; 6427} 6428 6429void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6430{ 6431 sp<SyncEvent> strongEvent = event.promote(); 6432 6433 if (strongEvent != 0) { 6434 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6435 me->handleSyncStartEvent(strongEvent); 6436 } 6437} 6438 6439void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6440{ 6441 if (event == mSyncStartEvent) { 6442 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6443 // from audio HAL 6444 mFramestoDrop = mFrameCount * 2; 6445 } 6446} 6447 6448bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6449 ALOGV("RecordThread::stop"); 6450 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6451 return false; 6452 } 6453 recordTrack->mState = TrackBase::PAUSING; 6454 // do not wait for mStartStopCond if exiting 6455 if (exitPending()) { 6456 return true; 6457 } 6458 mStartStopCond.wait(mLock); 6459 // if we have been restarted, recordTrack == mActiveTrack.get() here 6460 if (exitPending() || recordTrack != mActiveTrack.get()) { 6461 ALOGV("Record stopped OK"); 6462 return true; 6463 } 6464 return false; 6465} 6466 6467bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6468{ 6469 return false; 6470} 6471 6472status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6473{ 6474#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6475 if (!isValidSyncEvent(event)) { 6476 return BAD_VALUE; 6477 } 6478 6479 int eventSession = event->triggerSession(); 6480 status_t ret = NAME_NOT_FOUND; 6481 6482 Mutex::Autolock _l(mLock); 6483 6484 for (size_t i = 0; i < mTracks.size(); i++) { 6485 sp<RecordTrack> track = mTracks[i]; 6486 if (eventSession == track->sessionId()) { 6487 (void) track->setSyncEvent(event); 6488 ret = NO_ERROR; 6489 } 6490 } 6491 return ret; 6492#else 6493 return BAD_VALUE; 6494#endif 6495} 6496 6497void AudioFlinger::RecordThread::RecordTrack::destroy() 6498{ 6499 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6500 sp<RecordTrack> keep(this); 6501 { 6502 sp<ThreadBase> thread = mThread.promote(); 6503 if (thread != 0) { 6504 if (mState == ACTIVE || mState == RESUMING) { 6505 AudioSystem::stopInput(thread->id()); 6506 } 6507 AudioSystem::releaseInput(thread->id()); 6508 Mutex::Autolock _l(thread->mLock); 6509 RecordThread *recordThread = (RecordThread *) thread.get(); 6510 recordThread->destroyTrack_l(this); 6511 } 6512 } 6513} 6514 6515// destroyTrack_l() must be called with ThreadBase::mLock held 6516void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6517{ 6518 track->mState = TrackBase::TERMINATED; 6519 // active tracks are removed by threadLoop() 6520 if (mActiveTrack != track) { 6521 removeTrack_l(track); 6522 } 6523} 6524 6525void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6526{ 6527 mTracks.remove(track); 6528 // need anything related to effects here? 6529} 6530 6531void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6532{ 6533 dumpInternals(fd, args); 6534 dumpTracks(fd, args); 6535 dumpEffectChains(fd, args); 6536} 6537 6538void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6539{ 6540 const size_t SIZE = 256; 6541 char buffer[SIZE]; 6542 String8 result; 6543 6544 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6545 result.append(buffer); 6546 6547 if (mActiveTrack != 0) { 6548 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6549 result.append(buffer); 6550 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6551 result.append(buffer); 6552 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6553 result.append(buffer); 6554 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6555 result.append(buffer); 6556 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 6557 result.append(buffer); 6558 } else { 6559 result.append("No active record client\n"); 6560 } 6561 6562 write(fd, result.string(), result.size()); 6563 6564 dumpBase(fd, args); 6565} 6566 6567void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6568{ 6569 const size_t SIZE = 256; 6570 char buffer[SIZE]; 6571 String8 result; 6572 6573 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6574 result.append(buffer); 6575 RecordTrack::appendDumpHeader(result); 6576 for (size_t i = 0; i < mTracks.size(); ++i) { 6577 sp<RecordTrack> track = mTracks[i]; 6578 if (track != 0) { 6579 track->dump(buffer, SIZE); 6580 result.append(buffer); 6581 } 6582 } 6583 6584 if (mActiveTrack != 0) { 6585 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6586 result.append(buffer); 6587 RecordTrack::appendDumpHeader(result); 6588 mActiveTrack->dump(buffer, SIZE); 6589 result.append(buffer); 6590 6591 } 6592 write(fd, result.string(), result.size()); 6593} 6594 6595// AudioBufferProvider interface 6596status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6597{ 6598 size_t framesReq = buffer->frameCount; 6599 size_t framesReady = mFrameCount - mRsmpInIndex; 6600 int channelCount; 6601 6602 if (framesReady == 0) { 6603 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6604 if (mBytesRead <= 0) { 6605 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6606 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6607 // Force input into standby so that it tries to 6608 // recover at next read attempt 6609 inputStandBy(); 6610 usleep(kRecordThreadSleepUs); 6611 } 6612 buffer->raw = NULL; 6613 buffer->frameCount = 0; 6614 return NOT_ENOUGH_DATA; 6615 } 6616 mRsmpInIndex = 0; 6617 framesReady = mFrameCount; 6618 } 6619 6620 if (framesReq > framesReady) { 6621 framesReq = framesReady; 6622 } 6623 6624 if (mChannelCount == 1 && mReqChannelCount == 2) { 6625 channelCount = 1; 6626 } else { 6627 channelCount = 2; 6628 } 6629 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6630 buffer->frameCount = framesReq; 6631 return NO_ERROR; 6632} 6633 6634// AudioBufferProvider interface 6635void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6636{ 6637 mRsmpInIndex += buffer->frameCount; 6638 buffer->frameCount = 0; 6639} 6640 6641bool AudioFlinger::RecordThread::checkForNewParameters_l() 6642{ 6643 bool reconfig = false; 6644 6645 while (!mNewParameters.isEmpty()) { 6646 status_t status = NO_ERROR; 6647 String8 keyValuePair = mNewParameters[0]; 6648 AudioParameter param = AudioParameter(keyValuePair); 6649 int value; 6650 audio_format_t reqFormat = mFormat; 6651 uint32_t reqSamplingRate = mReqSampleRate; 6652 int reqChannelCount = mReqChannelCount; 6653 6654 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6655 reqSamplingRate = value; 6656 reconfig = true; 6657 } 6658 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6659 reqFormat = (audio_format_t) value; 6660 reconfig = true; 6661 } 6662 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6663 reqChannelCount = popcount(value); 6664 reconfig = true; 6665 } 6666 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6667 // do not accept frame count changes if tracks are open as the track buffer 6668 // size depends on frame count and correct behavior would not be guaranteed 6669 // if frame count is changed after track creation 6670 if (mActiveTrack != 0) { 6671 status = INVALID_OPERATION; 6672 } else { 6673 reconfig = true; 6674 } 6675 } 6676 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6677 // forward device change to effects that have requested to be 6678 // aware of attached audio device. 6679 for (size_t i = 0; i < mEffectChains.size(); i++) { 6680 mEffectChains[i]->setDevice_l(value); 6681 } 6682 6683 // store input device and output device but do not forward output device to audio HAL. 6684 // Note that status is ignored by the caller for output device 6685 // (see AudioFlinger::setParameters() 6686 if (audio_is_output_devices(value)) { 6687 mOutDevice = value; 6688 status = BAD_VALUE; 6689 } else { 6690 mInDevice = value; 6691 // disable AEC and NS if the device is a BT SCO headset supporting those 6692 // pre processings 6693 if (mTracks.size() > 0) { 6694 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6695 mAudioFlinger->btNrecIsOff(); 6696 for (size_t i = 0; i < mTracks.size(); i++) { 6697 sp<RecordTrack> track = mTracks[i]; 6698 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6699 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6700 } 6701 } 6702 } 6703 } 6704 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6705 mAudioSource != (audio_source_t)value) { 6706 // forward device change to effects that have requested to be 6707 // aware of attached audio device. 6708 for (size_t i = 0; i < mEffectChains.size(); i++) { 6709 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6710 } 6711 mAudioSource = (audio_source_t)value; 6712 } 6713 if (status == NO_ERROR) { 6714 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6715 keyValuePair.string()); 6716 if (status == INVALID_OPERATION) { 6717 inputStandBy(); 6718 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6719 keyValuePair.string()); 6720 } 6721 if (reconfig) { 6722 if (status == BAD_VALUE && 6723 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6724 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6725 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6726 <= (2 * reqSamplingRate)) && 6727 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6728 <= FCC_2 && 6729 (reqChannelCount <= FCC_2)) { 6730 status = NO_ERROR; 6731 } 6732 if (status == NO_ERROR) { 6733 readInputParameters(); 6734 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6735 } 6736 } 6737 } 6738 6739 mNewParameters.removeAt(0); 6740 6741 mParamStatus = status; 6742 mParamCond.signal(); 6743 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6744 // already timed out waiting for the status and will never signal the condition. 6745 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6746 } 6747 return reconfig; 6748} 6749 6750String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6751{ 6752 char *s; 6753 String8 out_s8 = String8(); 6754 6755 Mutex::Autolock _l(mLock); 6756 if (initCheck() != NO_ERROR) { 6757 return out_s8; 6758 } 6759 6760 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6761 out_s8 = String8(s); 6762 free(s); 6763 return out_s8; 6764} 6765 6766void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6767 AudioSystem::OutputDescriptor desc; 6768 void *param2 = NULL; 6769 6770 switch (event) { 6771 case AudioSystem::INPUT_OPENED: 6772 case AudioSystem::INPUT_CONFIG_CHANGED: 6773 desc.channels = mChannelMask; 6774 desc.samplingRate = mSampleRate; 6775 desc.format = mFormat; 6776 desc.frameCount = mFrameCount; 6777 desc.latency = 0; 6778 param2 = &desc; 6779 break; 6780 6781 case AudioSystem::INPUT_CLOSED: 6782 default: 6783 break; 6784 } 6785 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6786} 6787 6788void AudioFlinger::RecordThread::readInputParameters() 6789{ 6790 delete mRsmpInBuffer; 6791 // mRsmpInBuffer is always assigned a new[] below 6792 delete mRsmpOutBuffer; 6793 mRsmpOutBuffer = NULL; 6794 delete mResampler; 6795 mResampler = NULL; 6796 6797 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6798 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6799 mChannelCount = (uint16_t)popcount(mChannelMask); 6800 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6801 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6802 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6803 mFrameCount = mInputBytes / mFrameSize; 6804 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6805 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6806 6807 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6808 { 6809 int channelCount; 6810 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6811 // stereo to mono post process as the resampler always outputs stereo. 6812 if (mChannelCount == 1 && mReqChannelCount == 2) { 6813 channelCount = 1; 6814 } else { 6815 channelCount = 2; 6816 } 6817 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6818 mResampler->setSampleRate(mSampleRate); 6819 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6820 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6821 6822 // optmization: if mono to mono, alter input frame count as if we were inputing 6823 // stereo samples 6824 if (mChannelCount == 1 && mReqChannelCount == 1) { 6825 mFrameCount >>= 1; 6826 } 6827 6828 } 6829 mRsmpInIndex = mFrameCount; 6830} 6831 6832unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6833{ 6834 Mutex::Autolock _l(mLock); 6835 if (initCheck() != NO_ERROR) { 6836 return 0; 6837 } 6838 6839 return mInput->stream->get_input_frames_lost(mInput->stream); 6840} 6841 6842uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6843{ 6844 Mutex::Autolock _l(mLock); 6845 uint32_t result = 0; 6846 if (getEffectChain_l(sessionId) != 0) { 6847 result = EFFECT_SESSION; 6848 } 6849 6850 for (size_t i = 0; i < mTracks.size(); ++i) { 6851 if (sessionId == mTracks[i]->sessionId()) { 6852 result |= TRACK_SESSION; 6853 break; 6854 } 6855 } 6856 6857 return result; 6858} 6859 6860KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6861{ 6862 KeyedVector<int, bool> ids; 6863 Mutex::Autolock _l(mLock); 6864 for (size_t j = 0; j < mTracks.size(); ++j) { 6865 sp<RecordThread::RecordTrack> track = mTracks[j]; 6866 int sessionId = track->sessionId(); 6867 if (ids.indexOfKey(sessionId) < 0) { 6868 ids.add(sessionId, true); 6869 } 6870 } 6871 return ids; 6872} 6873 6874AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6875{ 6876 Mutex::Autolock _l(mLock); 6877 AudioStreamIn *input = mInput; 6878 mInput = NULL; 6879 return input; 6880} 6881 6882// this method must always be called either with ThreadBase mLock held or inside the thread loop 6883audio_stream_t* AudioFlinger::RecordThread::stream() const 6884{ 6885 if (mInput == NULL) { 6886 return NULL; 6887 } 6888 return &mInput->stream->common; 6889} 6890 6891 6892// ---------------------------------------------------------------------------- 6893 6894audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6895{ 6896 if (!settingsAllowed()) { 6897 return 0; 6898 } 6899 Mutex::Autolock _l(mLock); 6900 return loadHwModule_l(name); 6901} 6902 6903// loadHwModule_l() must be called with AudioFlinger::mLock held 6904audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6905{ 6906 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6907 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6908 ALOGW("loadHwModule() module %s already loaded", name); 6909 return mAudioHwDevs.keyAt(i); 6910 } 6911 } 6912 6913 audio_hw_device_t *dev; 6914 6915 int rc = load_audio_interface(name, &dev); 6916 if (rc) { 6917 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6918 return 0; 6919 } 6920 6921 mHardwareStatus = AUDIO_HW_INIT; 6922 rc = dev->init_check(dev); 6923 mHardwareStatus = AUDIO_HW_IDLE; 6924 if (rc) { 6925 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6926 return 0; 6927 } 6928 6929 // Check and cache this HAL's level of support for master mute and master 6930 // volume. If this is the first HAL opened, and it supports the get 6931 // methods, use the initial values provided by the HAL as the current 6932 // master mute and volume settings. 6933 6934 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6935 { // scope for auto-lock pattern 6936 AutoMutex lock(mHardwareLock); 6937 6938 if (0 == mAudioHwDevs.size()) { 6939 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6940 if (NULL != dev->get_master_volume) { 6941 float mv; 6942 if (OK == dev->get_master_volume(dev, &mv)) { 6943 mMasterVolume = mv; 6944 } 6945 } 6946 6947 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6948 if (NULL != dev->get_master_mute) { 6949 bool mm; 6950 if (OK == dev->get_master_mute(dev, &mm)) { 6951 mMasterMute = mm; 6952 } 6953 } 6954 } 6955 6956 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6957 if ((NULL != dev->set_master_volume) && 6958 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6959 flags = static_cast<AudioHwDevice::Flags>(flags | 6960 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6961 } 6962 6963 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6964 if ((NULL != dev->set_master_mute) && 6965 (OK == dev->set_master_mute(dev, mMasterMute))) { 6966 flags = static_cast<AudioHwDevice::Flags>(flags | 6967 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6968 } 6969 6970 mHardwareStatus = AUDIO_HW_IDLE; 6971 } 6972 6973 audio_module_handle_t handle = nextUniqueId(); 6974 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6975 6976 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6977 name, dev->common.module->name, dev->common.module->id, handle); 6978 6979 return handle; 6980 6981} 6982 6983// ---------------------------------------------------------------------------- 6984 6985uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 6986{ 6987 Mutex::Autolock _l(mLock); 6988 PlaybackThread *thread = primaryPlaybackThread_l(); 6989 return thread != NULL ? thread->sampleRate() : 0; 6990} 6991 6992size_t AudioFlinger::getPrimaryOutputFrameCount() 6993{ 6994 Mutex::Autolock _l(mLock); 6995 PlaybackThread *thread = primaryPlaybackThread_l(); 6996 return thread != NULL ? thread->frameCountHAL() : 0; 6997} 6998 6999// ---------------------------------------------------------------------------- 7000 7001audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 7002 audio_devices_t *pDevices, 7003 uint32_t *pSamplingRate, 7004 audio_format_t *pFormat, 7005 audio_channel_mask_t *pChannelMask, 7006 uint32_t *pLatencyMs, 7007 audio_output_flags_t flags) 7008{ 7009 status_t status; 7010 PlaybackThread *thread = NULL; 7011 struct audio_config config = { 7012 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7013 channel_mask: pChannelMask ? *pChannelMask : 0, 7014 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7015 }; 7016 audio_stream_out_t *outStream = NULL; 7017 AudioHwDevice *outHwDev; 7018 7019 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7020 module, 7021 (pDevices != NULL) ? *pDevices : 0, 7022 config.sample_rate, 7023 config.format, 7024 config.channel_mask, 7025 flags); 7026 7027 if (pDevices == NULL || *pDevices == 0) { 7028 return 0; 7029 } 7030 7031 Mutex::Autolock _l(mLock); 7032 7033 outHwDev = findSuitableHwDev_l(module, *pDevices); 7034 if (outHwDev == NULL) 7035 return 0; 7036 7037 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7038 audio_io_handle_t id = nextUniqueId(); 7039 7040 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7041 7042 status = hwDevHal->open_output_stream(hwDevHal, 7043 id, 7044 *pDevices, 7045 (audio_output_flags_t)flags, 7046 &config, 7047 &outStream); 7048 7049 mHardwareStatus = AUDIO_HW_IDLE; 7050 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7051 "Channels %x, status %d", 7052 outStream, 7053 config.sample_rate, 7054 config.format, 7055 config.channel_mask, 7056 status); 7057 7058 if (status == NO_ERROR && outStream != NULL) { 7059 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7060 7061 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7062 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7063 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7064 thread = new DirectOutputThread(this, output, id, *pDevices); 7065 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7066 } else { 7067 thread = new MixerThread(this, output, id, *pDevices); 7068 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7069 } 7070 mPlaybackThreads.add(id, thread); 7071 7072 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7073 if (pFormat != NULL) *pFormat = config.format; 7074 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7075 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7076 7077 // notify client processes of the new output creation 7078 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7079 7080 // the first primary output opened designates the primary hw device 7081 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7082 ALOGI("Using module %d has the primary audio interface", module); 7083 mPrimaryHardwareDev = outHwDev; 7084 7085 AutoMutex lock(mHardwareLock); 7086 mHardwareStatus = AUDIO_HW_SET_MODE; 7087 hwDevHal->set_mode(hwDevHal, mMode); 7088 mHardwareStatus = AUDIO_HW_IDLE; 7089 } 7090 return id; 7091 } 7092 7093 return 0; 7094} 7095 7096audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7097 audio_io_handle_t output2) 7098{ 7099 Mutex::Autolock _l(mLock); 7100 MixerThread *thread1 = checkMixerThread_l(output1); 7101 MixerThread *thread2 = checkMixerThread_l(output2); 7102 7103 if (thread1 == NULL || thread2 == NULL) { 7104 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7105 output2); 7106 return 0; 7107 } 7108 7109 audio_io_handle_t id = nextUniqueId(); 7110 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7111 thread->addOutputTrack(thread2); 7112 mPlaybackThreads.add(id, thread); 7113 // notify client processes of the new output creation 7114 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7115 return id; 7116} 7117 7118status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7119{ 7120 return closeOutput_nonvirtual(output); 7121} 7122 7123status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7124{ 7125 // keep strong reference on the playback thread so that 7126 // it is not destroyed while exit() is executed 7127 sp<PlaybackThread> thread; 7128 { 7129 Mutex::Autolock _l(mLock); 7130 thread = checkPlaybackThread_l(output); 7131 if (thread == NULL) { 7132 return BAD_VALUE; 7133 } 7134 7135 ALOGV("closeOutput() %d", output); 7136 7137 if (thread->type() == ThreadBase::MIXER) { 7138 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7139 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7140 DuplicatingThread *dupThread = 7141 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7142 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7143 } 7144 } 7145 } 7146 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7147 mPlaybackThreads.removeItem(output); 7148 } 7149 thread->exit(); 7150 // The thread entity (active unit of execution) is no longer running here, 7151 // but the ThreadBase container still exists. 7152 7153 if (thread->type() != ThreadBase::DUPLICATING) { 7154 AudioStreamOut *out = thread->clearOutput(); 7155 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7156 // from now on thread->mOutput is NULL 7157 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7158 delete out; 7159 } 7160 return NO_ERROR; 7161} 7162 7163status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7164{ 7165 Mutex::Autolock _l(mLock); 7166 PlaybackThread *thread = checkPlaybackThread_l(output); 7167 7168 if (thread == NULL) { 7169 return BAD_VALUE; 7170 } 7171 7172 ALOGV("suspendOutput() %d", output); 7173 thread->suspend(); 7174 7175 return NO_ERROR; 7176} 7177 7178status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7179{ 7180 Mutex::Autolock _l(mLock); 7181 PlaybackThread *thread = checkPlaybackThread_l(output); 7182 7183 if (thread == NULL) { 7184 return BAD_VALUE; 7185 } 7186 7187 ALOGV("restoreOutput() %d", output); 7188 7189 thread->restore(); 7190 7191 return NO_ERROR; 7192} 7193 7194audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7195 audio_devices_t *pDevices, 7196 uint32_t *pSamplingRate, 7197 audio_format_t *pFormat, 7198 audio_channel_mask_t *pChannelMask) 7199{ 7200 status_t status; 7201 RecordThread *thread = NULL; 7202 struct audio_config config = { 7203 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7204 channel_mask: pChannelMask ? *pChannelMask : 0, 7205 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7206 }; 7207 uint32_t reqSamplingRate = config.sample_rate; 7208 audio_format_t reqFormat = config.format; 7209 audio_channel_mask_t reqChannels = config.channel_mask; 7210 audio_stream_in_t *inStream = NULL; 7211 AudioHwDevice *inHwDev; 7212 7213 if (pDevices == NULL || *pDevices == 0) { 7214 return 0; 7215 } 7216 7217 Mutex::Autolock _l(mLock); 7218 7219 inHwDev = findSuitableHwDev_l(module, *pDevices); 7220 if (inHwDev == NULL) 7221 return 0; 7222 7223 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7224 audio_io_handle_t id = nextUniqueId(); 7225 7226 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7227 &inStream); 7228 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7229 "status %d", 7230 inStream, 7231 config.sample_rate, 7232 config.format, 7233 config.channel_mask, 7234 status); 7235 7236 // If the input could not be opened with the requested parameters and we can handle the 7237 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7238 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7239 if (status == BAD_VALUE && 7240 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7241 (config.sample_rate <= 2 * reqSamplingRate) && 7242 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7243 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7244 inStream = NULL; 7245 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7246 } 7247 7248 if (status == NO_ERROR && inStream != NULL) { 7249 7250 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7251 // or (re-)create if current Pipe is idle and does not match the new format 7252 sp<NBAIO_Sink> teeSink; 7253#ifdef TEE_SINK_INPUT_FRAMES 7254 enum { 7255 TEE_SINK_NO, // don't copy input 7256 TEE_SINK_NEW, // copy input using a new pipe 7257 TEE_SINK_OLD, // copy input using an existing pipe 7258 } kind; 7259 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7260 popcount(inStream->common.get_channels(&inStream->common))); 7261 if (format == Format_Invalid) { 7262 kind = TEE_SINK_NO; 7263 } else if (mRecordTeeSink == 0) { 7264 kind = TEE_SINK_NEW; 7265 } else if (mRecordTeeSink->getStrongCount() != 1) { 7266 kind = TEE_SINK_NO; 7267 } else if (format == mRecordTeeSink->format()) { 7268 kind = TEE_SINK_OLD; 7269 } else { 7270 kind = TEE_SINK_NEW; 7271 } 7272 switch (kind) { 7273 case TEE_SINK_NEW: { 7274 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7275 size_t numCounterOffers = 0; 7276 const NBAIO_Format offers[1] = {format}; 7277 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7278 ALOG_ASSERT(index == 0); 7279 PipeReader *pipeReader = new PipeReader(*pipe); 7280 numCounterOffers = 0; 7281 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7282 ALOG_ASSERT(index == 0); 7283 mRecordTeeSink = pipe; 7284 mRecordTeeSource = pipeReader; 7285 teeSink = pipe; 7286 } 7287 break; 7288 case TEE_SINK_OLD: 7289 teeSink = mRecordTeeSink; 7290 break; 7291 case TEE_SINK_NO: 7292 default: 7293 break; 7294 } 7295#endif 7296 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7297 7298 // Start record thread 7299 // RecorThread require both input and output device indication to forward to audio 7300 // pre processing modules 7301 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7302 7303 thread = new RecordThread(this, 7304 input, 7305 reqSamplingRate, 7306 reqChannels, 7307 id, 7308 device, teeSink); 7309 mRecordThreads.add(id, thread); 7310 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7311 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7312 if (pFormat != NULL) *pFormat = config.format; 7313 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7314 7315 // notify client processes of the new input creation 7316 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7317 return id; 7318 } 7319 7320 return 0; 7321} 7322 7323status_t AudioFlinger::closeInput(audio_io_handle_t input) 7324{ 7325 return closeInput_nonvirtual(input); 7326} 7327 7328status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7329{ 7330 // keep strong reference on the record thread so that 7331 // it is not destroyed while exit() is executed 7332 sp<RecordThread> thread; 7333 { 7334 Mutex::Autolock _l(mLock); 7335 thread = checkRecordThread_l(input); 7336 if (thread == 0) { 7337 return BAD_VALUE; 7338 } 7339 7340 ALOGV("closeInput() %d", input); 7341 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7342 mRecordThreads.removeItem(input); 7343 } 7344 thread->exit(); 7345 // The thread entity (active unit of execution) is no longer running here, 7346 // but the ThreadBase container still exists. 7347 7348 AudioStreamIn *in = thread->clearInput(); 7349 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7350 // from now on thread->mInput is NULL 7351 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7352 delete in; 7353 7354 return NO_ERROR; 7355} 7356 7357status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7358{ 7359 Mutex::Autolock _l(mLock); 7360 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7361 7362 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7363 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7364 thread->invalidateTracks(stream); 7365 } 7366 7367 return NO_ERROR; 7368} 7369 7370 7371int AudioFlinger::newAudioSessionId() 7372{ 7373 return nextUniqueId(); 7374} 7375 7376void AudioFlinger::acquireAudioSessionId(int audioSession) 7377{ 7378 Mutex::Autolock _l(mLock); 7379 pid_t caller = IPCThreadState::self()->getCallingPid(); 7380 ALOGV("acquiring %d from %d", audioSession, caller); 7381 size_t num = mAudioSessionRefs.size(); 7382 for (size_t i = 0; i< num; i++) { 7383 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7384 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7385 ref->mCnt++; 7386 ALOGV(" incremented refcount to %d", ref->mCnt); 7387 return; 7388 } 7389 } 7390 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7391 ALOGV(" added new entry for %d", audioSession); 7392} 7393 7394void AudioFlinger::releaseAudioSessionId(int audioSession) 7395{ 7396 Mutex::Autolock _l(mLock); 7397 pid_t caller = IPCThreadState::self()->getCallingPid(); 7398 ALOGV("releasing %d from %d", audioSession, caller); 7399 size_t num = mAudioSessionRefs.size(); 7400 for (size_t i = 0; i< num; i++) { 7401 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7402 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7403 ref->mCnt--; 7404 ALOGV(" decremented refcount to %d", ref->mCnt); 7405 if (ref->mCnt == 0) { 7406 mAudioSessionRefs.removeAt(i); 7407 delete ref; 7408 purgeStaleEffects_l(); 7409 } 7410 return; 7411 } 7412 } 7413 ALOGW("session id %d not found for pid %d", audioSession, caller); 7414} 7415 7416void AudioFlinger::purgeStaleEffects_l() { 7417 7418 ALOGV("purging stale effects"); 7419 7420 Vector< sp<EffectChain> > chains; 7421 7422 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7423 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7424 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7425 sp<EffectChain> ec = t->mEffectChains[j]; 7426 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7427 chains.push(ec); 7428 } 7429 } 7430 } 7431 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7432 sp<RecordThread> t = mRecordThreads.valueAt(i); 7433 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7434 sp<EffectChain> ec = t->mEffectChains[j]; 7435 chains.push(ec); 7436 } 7437 } 7438 7439 for (size_t i = 0; i < chains.size(); i++) { 7440 sp<EffectChain> ec = chains[i]; 7441 int sessionid = ec->sessionId(); 7442 sp<ThreadBase> t = ec->mThread.promote(); 7443 if (t == 0) { 7444 continue; 7445 } 7446 size_t numsessionrefs = mAudioSessionRefs.size(); 7447 bool found = false; 7448 for (size_t k = 0; k < numsessionrefs; k++) { 7449 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7450 if (ref->mSessionid == sessionid) { 7451 ALOGV(" session %d still exists for %d with %d refs", 7452 sessionid, ref->mPid, ref->mCnt); 7453 found = true; 7454 break; 7455 } 7456 } 7457 if (!found) { 7458 Mutex::Autolock _l (t->mLock); 7459 // remove all effects from the chain 7460 while (ec->mEffects.size()) { 7461 sp<EffectModule> effect = ec->mEffects[0]; 7462 effect->unPin(); 7463 t->removeEffect_l(effect); 7464 if (effect->purgeHandles()) { 7465 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7466 } 7467 AudioSystem::unregisterEffect(effect->id()); 7468 } 7469 } 7470 } 7471 return; 7472} 7473 7474// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7475AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7476{ 7477 return mPlaybackThreads.valueFor(output).get(); 7478} 7479 7480// checkMixerThread_l() must be called with AudioFlinger::mLock held 7481AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7482{ 7483 PlaybackThread *thread = checkPlaybackThread_l(output); 7484 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7485} 7486 7487// checkRecordThread_l() must be called with AudioFlinger::mLock held 7488AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7489{ 7490 return mRecordThreads.valueFor(input).get(); 7491} 7492 7493uint32_t AudioFlinger::nextUniqueId() 7494{ 7495 return android_atomic_inc(&mNextUniqueId); 7496} 7497 7498AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7499{ 7500 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7501 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7502 AudioStreamOut *output = thread->getOutput(); 7503 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7504 return thread; 7505 } 7506 } 7507 return NULL; 7508} 7509 7510audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7511{ 7512 PlaybackThread *thread = primaryPlaybackThread_l(); 7513 7514 if (thread == NULL) { 7515 return 0; 7516 } 7517 7518 return thread->outDevice(); 7519} 7520 7521sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7522 int triggerSession, 7523 int listenerSession, 7524 sync_event_callback_t callBack, 7525 void *cookie) 7526{ 7527 Mutex::Autolock _l(mLock); 7528 7529 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7530 status_t playStatus = NAME_NOT_FOUND; 7531 status_t recStatus = NAME_NOT_FOUND; 7532 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7533 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7534 if (playStatus == NO_ERROR) { 7535 return event; 7536 } 7537 } 7538 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7539 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7540 if (recStatus == NO_ERROR) { 7541 return event; 7542 } 7543 } 7544 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7545 mPendingSyncEvents.add(event); 7546 } else { 7547 ALOGV("createSyncEvent() invalid event %d", event->type()); 7548 event.clear(); 7549 } 7550 return event; 7551} 7552 7553// ---------------------------------------------------------------------------- 7554// Effect management 7555// ---------------------------------------------------------------------------- 7556 7557 7558status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7559{ 7560 Mutex::Autolock _l(mLock); 7561 return EffectQueryNumberEffects(numEffects); 7562} 7563 7564status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7565{ 7566 Mutex::Autolock _l(mLock); 7567 return EffectQueryEffect(index, descriptor); 7568} 7569 7570status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7571 effect_descriptor_t *descriptor) const 7572{ 7573 Mutex::Autolock _l(mLock); 7574 return EffectGetDescriptor(pUuid, descriptor); 7575} 7576 7577 7578sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7579 effect_descriptor_t *pDesc, 7580 const sp<IEffectClient>& effectClient, 7581 int32_t priority, 7582 audio_io_handle_t io, 7583 int sessionId, 7584 status_t *status, 7585 int *id, 7586 int *enabled) 7587{ 7588 status_t lStatus = NO_ERROR; 7589 sp<EffectHandle> handle; 7590 effect_descriptor_t desc; 7591 7592 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7593 pid, effectClient.get(), priority, sessionId, io); 7594 7595 if (pDesc == NULL) { 7596 lStatus = BAD_VALUE; 7597 goto Exit; 7598 } 7599 7600 // check audio settings permission for global effects 7601 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7602 lStatus = PERMISSION_DENIED; 7603 goto Exit; 7604 } 7605 7606 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7607 // that can only be created by audio policy manager (running in same process) 7608 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7609 lStatus = PERMISSION_DENIED; 7610 goto Exit; 7611 } 7612 7613 if (io == 0) { 7614 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7615 // output must be specified by AudioPolicyManager when using session 7616 // AUDIO_SESSION_OUTPUT_STAGE 7617 lStatus = BAD_VALUE; 7618 goto Exit; 7619 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7620 // if the output returned by getOutputForEffect() is removed before we lock the 7621 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7622 // and we will exit safely 7623 io = AudioSystem::getOutputForEffect(&desc); 7624 } 7625 } 7626 7627 { 7628 Mutex::Autolock _l(mLock); 7629 7630 7631 if (!EffectIsNullUuid(&pDesc->uuid)) { 7632 // if uuid is specified, request effect descriptor 7633 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7634 if (lStatus < 0) { 7635 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7636 goto Exit; 7637 } 7638 } else { 7639 // if uuid is not specified, look for an available implementation 7640 // of the required type in effect factory 7641 if (EffectIsNullUuid(&pDesc->type)) { 7642 ALOGW("createEffect() no effect type"); 7643 lStatus = BAD_VALUE; 7644 goto Exit; 7645 } 7646 uint32_t numEffects = 0; 7647 effect_descriptor_t d; 7648 d.flags = 0; // prevent compiler warning 7649 bool found = false; 7650 7651 lStatus = EffectQueryNumberEffects(&numEffects); 7652 if (lStatus < 0) { 7653 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7654 goto Exit; 7655 } 7656 for (uint32_t i = 0; i < numEffects; i++) { 7657 lStatus = EffectQueryEffect(i, &desc); 7658 if (lStatus < 0) { 7659 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7660 continue; 7661 } 7662 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7663 // If matching type found save effect descriptor. If the session is 7664 // 0 and the effect is not auxiliary, continue enumeration in case 7665 // an auxiliary version of this effect type is available 7666 found = true; 7667 d = desc; 7668 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7669 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7670 break; 7671 } 7672 } 7673 } 7674 if (!found) { 7675 lStatus = BAD_VALUE; 7676 ALOGW("createEffect() effect not found"); 7677 goto Exit; 7678 } 7679 // For same effect type, chose auxiliary version over insert version if 7680 // connect to output mix (Compliance to OpenSL ES) 7681 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7682 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7683 desc = d; 7684 } 7685 } 7686 7687 // Do not allow auxiliary effects on a session different from 0 (output mix) 7688 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7689 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7690 lStatus = INVALID_OPERATION; 7691 goto Exit; 7692 } 7693 7694 // check recording permission for visualizer 7695 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7696 !recordingAllowed()) { 7697 lStatus = PERMISSION_DENIED; 7698 goto Exit; 7699 } 7700 7701 // return effect descriptor 7702 *pDesc = desc; 7703 7704 // If output is not specified try to find a matching audio session ID in one of the 7705 // output threads. 7706 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7707 // because of code checking output when entering the function. 7708 // Note: io is never 0 when creating an effect on an input 7709 if (io == 0) { 7710 // look for the thread where the specified audio session is present 7711 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7712 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7713 io = mPlaybackThreads.keyAt(i); 7714 break; 7715 } 7716 } 7717 if (io == 0) { 7718 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7719 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7720 io = mRecordThreads.keyAt(i); 7721 break; 7722 } 7723 } 7724 } 7725 // If no output thread contains the requested session ID, default to 7726 // first output. The effect chain will be moved to the correct output 7727 // thread when a track with the same session ID is created 7728 if (io == 0 && mPlaybackThreads.size()) { 7729 io = mPlaybackThreads.keyAt(0); 7730 } 7731 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7732 } 7733 ThreadBase *thread = checkRecordThread_l(io); 7734 if (thread == NULL) { 7735 thread = checkPlaybackThread_l(io); 7736 if (thread == NULL) { 7737 ALOGE("createEffect() unknown output thread"); 7738 lStatus = BAD_VALUE; 7739 goto Exit; 7740 } 7741 } 7742 7743 sp<Client> client = registerPid_l(pid); 7744 7745 // create effect on selected output thread 7746 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7747 &desc, enabled, &lStatus); 7748 if (handle != 0 && id != NULL) { 7749 *id = handle->id(); 7750 } 7751 } 7752 7753Exit: 7754 if (status != NULL) { 7755 *status = lStatus; 7756 } 7757 return handle; 7758} 7759 7760status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7761 audio_io_handle_t dstOutput) 7762{ 7763 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7764 sessionId, srcOutput, dstOutput); 7765 Mutex::Autolock _l(mLock); 7766 if (srcOutput == dstOutput) { 7767 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7768 return NO_ERROR; 7769 } 7770 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7771 if (srcThread == NULL) { 7772 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7773 return BAD_VALUE; 7774 } 7775 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7776 if (dstThread == NULL) { 7777 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7778 return BAD_VALUE; 7779 } 7780 7781 Mutex::Autolock _dl(dstThread->mLock); 7782 Mutex::Autolock _sl(srcThread->mLock); 7783 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7784 7785 return NO_ERROR; 7786} 7787 7788// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7789status_t AudioFlinger::moveEffectChain_l(int sessionId, 7790 AudioFlinger::PlaybackThread *srcThread, 7791 AudioFlinger::PlaybackThread *dstThread, 7792 bool reRegister) 7793{ 7794 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7795 sessionId, srcThread, dstThread); 7796 7797 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7798 if (chain == 0) { 7799 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7800 sessionId, srcThread); 7801 return INVALID_OPERATION; 7802 } 7803 7804 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7805 // so that a new chain is created with correct parameters when first effect is added. This is 7806 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7807 // removed. 7808 srcThread->removeEffectChain_l(chain); 7809 7810 // transfer all effects one by one so that new effect chain is created on new thread with 7811 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7812 audio_io_handle_t dstOutput = dstThread->id(); 7813 sp<EffectChain> dstChain; 7814 uint32_t strategy = 0; // prevent compiler warning 7815 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7816 while (effect != 0) { 7817 srcThread->removeEffect_l(effect); 7818 dstThread->addEffect_l(effect); 7819 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7820 if (effect->state() == EffectModule::ACTIVE || 7821 effect->state() == EffectModule::STOPPING) { 7822 effect->start(); 7823 } 7824 // if the move request is not received from audio policy manager, the effect must be 7825 // re-registered with the new strategy and output 7826 if (dstChain == 0) { 7827 dstChain = effect->chain().promote(); 7828 if (dstChain == 0) { 7829 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7830 srcThread->addEffect_l(effect); 7831 return NO_INIT; 7832 } 7833 strategy = dstChain->strategy(); 7834 } 7835 if (reRegister) { 7836 AudioSystem::unregisterEffect(effect->id()); 7837 AudioSystem::registerEffect(&effect->desc(), 7838 dstOutput, 7839 strategy, 7840 sessionId, 7841 effect->id()); 7842 } 7843 effect = chain->getEffectFromId_l(0); 7844 } 7845 7846 return NO_ERROR; 7847} 7848 7849 7850// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7851sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7852 const sp<AudioFlinger::Client>& client, 7853 const sp<IEffectClient>& effectClient, 7854 int32_t priority, 7855 int sessionId, 7856 effect_descriptor_t *desc, 7857 int *enabled, 7858 status_t *status 7859 ) 7860{ 7861 sp<EffectModule> effect; 7862 sp<EffectHandle> handle; 7863 status_t lStatus; 7864 sp<EffectChain> chain; 7865 bool chainCreated = false; 7866 bool effectCreated = false; 7867 bool effectRegistered = false; 7868 7869 lStatus = initCheck(); 7870 if (lStatus != NO_ERROR) { 7871 ALOGW("createEffect_l() Audio driver not initialized."); 7872 goto Exit; 7873 } 7874 7875 // Do not allow effects with session ID 0 on direct output or duplicating threads 7876 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7877 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7878 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7879 desc->name, sessionId); 7880 lStatus = BAD_VALUE; 7881 goto Exit; 7882 } 7883 // Only Pre processor effects are allowed on input threads and only on input threads 7884 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7885 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7886 desc->name, desc->flags, mType); 7887 lStatus = BAD_VALUE; 7888 goto Exit; 7889 } 7890 7891 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7892 7893 { // scope for mLock 7894 Mutex::Autolock _l(mLock); 7895 7896 // check for existing effect chain with the requested audio session 7897 chain = getEffectChain_l(sessionId); 7898 if (chain == 0) { 7899 // create a new chain for this session 7900 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7901 chain = new EffectChain(this, sessionId); 7902 addEffectChain_l(chain); 7903 chain->setStrategy(getStrategyForSession_l(sessionId)); 7904 chainCreated = true; 7905 } else { 7906 effect = chain->getEffectFromDesc_l(desc); 7907 } 7908 7909 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7910 7911 if (effect == 0) { 7912 int id = mAudioFlinger->nextUniqueId(); 7913 // Check CPU and memory usage 7914 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7915 if (lStatus != NO_ERROR) { 7916 goto Exit; 7917 } 7918 effectRegistered = true; 7919 // create a new effect module if none present in the chain 7920 effect = new EffectModule(this, chain, desc, id, sessionId); 7921 lStatus = effect->status(); 7922 if (lStatus != NO_ERROR) { 7923 goto Exit; 7924 } 7925 lStatus = chain->addEffect_l(effect); 7926 if (lStatus != NO_ERROR) { 7927 goto Exit; 7928 } 7929 effectCreated = true; 7930 7931 effect->setDevice(mOutDevice); 7932 effect->setDevice(mInDevice); 7933 effect->setMode(mAudioFlinger->getMode()); 7934 effect->setAudioSource(mAudioSource); 7935 } 7936 // create effect handle and connect it to effect module 7937 handle = new EffectHandle(effect, client, effectClient, priority); 7938 lStatus = effect->addHandle(handle.get()); 7939 if (enabled != NULL) { 7940 *enabled = (int)effect->isEnabled(); 7941 } 7942 } 7943 7944Exit: 7945 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7946 Mutex::Autolock _l(mLock); 7947 if (effectCreated) { 7948 chain->removeEffect_l(effect); 7949 } 7950 if (effectRegistered) { 7951 AudioSystem::unregisterEffect(effect->id()); 7952 } 7953 if (chainCreated) { 7954 removeEffectChain_l(chain); 7955 } 7956 handle.clear(); 7957 } 7958 7959 if (status != NULL) { 7960 *status = lStatus; 7961 } 7962 return handle; 7963} 7964 7965sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7966{ 7967 Mutex::Autolock _l(mLock); 7968 return getEffect_l(sessionId, effectId); 7969} 7970 7971sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7972{ 7973 sp<EffectChain> chain = getEffectChain_l(sessionId); 7974 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7975} 7976 7977// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7978// PlaybackThread::mLock held 7979status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7980{ 7981 // check for existing effect chain with the requested audio session 7982 int sessionId = effect->sessionId(); 7983 sp<EffectChain> chain = getEffectChain_l(sessionId); 7984 bool chainCreated = false; 7985 7986 if (chain == 0) { 7987 // create a new chain for this session 7988 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7989 chain = new EffectChain(this, sessionId); 7990 addEffectChain_l(chain); 7991 chain->setStrategy(getStrategyForSession_l(sessionId)); 7992 chainCreated = true; 7993 } 7994 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7995 7996 if (chain->getEffectFromId_l(effect->id()) != 0) { 7997 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7998 this, effect->desc().name, chain.get()); 7999 return BAD_VALUE; 8000 } 8001 8002 status_t status = chain->addEffect_l(effect); 8003 if (status != NO_ERROR) { 8004 if (chainCreated) { 8005 removeEffectChain_l(chain); 8006 } 8007 return status; 8008 } 8009 8010 effect->setDevice(mOutDevice); 8011 effect->setDevice(mInDevice); 8012 effect->setMode(mAudioFlinger->getMode()); 8013 effect->setAudioSource(mAudioSource); 8014 return NO_ERROR; 8015} 8016 8017void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8018 8019 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8020 effect_descriptor_t desc = effect->desc(); 8021 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8022 detachAuxEffect_l(effect->id()); 8023 } 8024 8025 sp<EffectChain> chain = effect->chain().promote(); 8026 if (chain != 0) { 8027 // remove effect chain if removing last effect 8028 if (chain->removeEffect_l(effect) == 0) { 8029 removeEffectChain_l(chain); 8030 } 8031 } else { 8032 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8033 } 8034} 8035 8036void AudioFlinger::ThreadBase::lockEffectChains_l( 8037 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8038{ 8039 effectChains = mEffectChains; 8040 for (size_t i = 0; i < mEffectChains.size(); i++) { 8041 mEffectChains[i]->lock(); 8042 } 8043} 8044 8045void AudioFlinger::ThreadBase::unlockEffectChains( 8046 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8047{ 8048 for (size_t i = 0; i < effectChains.size(); i++) { 8049 effectChains[i]->unlock(); 8050 } 8051} 8052 8053sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8054{ 8055 Mutex::Autolock _l(mLock); 8056 return getEffectChain_l(sessionId); 8057} 8058 8059sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8060{ 8061 size_t size = mEffectChains.size(); 8062 for (size_t i = 0; i < size; i++) { 8063 if (mEffectChains[i]->sessionId() == sessionId) { 8064 return mEffectChains[i]; 8065 } 8066 } 8067 return 0; 8068} 8069 8070void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8071{ 8072 Mutex::Autolock _l(mLock); 8073 size_t size = mEffectChains.size(); 8074 for (size_t i = 0; i < size; i++) { 8075 mEffectChains[i]->setMode_l(mode); 8076 } 8077} 8078 8079void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8080 EffectHandle *handle, 8081 bool unpinIfLast) { 8082 8083 Mutex::Autolock _l(mLock); 8084 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8085 // delete the effect module if removing last handle on it 8086 if (effect->removeHandle(handle) == 0) { 8087 if (!effect->isPinned() || unpinIfLast) { 8088 removeEffect_l(effect); 8089 AudioSystem::unregisterEffect(effect->id()); 8090 } 8091 } 8092} 8093 8094status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8095{ 8096 int session = chain->sessionId(); 8097 int16_t *buffer = mMixBuffer; 8098 bool ownsBuffer = false; 8099 8100 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8101 if (session > 0) { 8102 // Only one effect chain can be present in direct output thread and it uses 8103 // the mix buffer as input 8104 if (mType != DIRECT) { 8105 size_t numSamples = mNormalFrameCount * mChannelCount; 8106 buffer = new int16_t[numSamples]; 8107 memset(buffer, 0, numSamples * sizeof(int16_t)); 8108 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8109 ownsBuffer = true; 8110 } 8111 8112 // Attach all tracks with same session ID to this chain. 8113 for (size_t i = 0; i < mTracks.size(); ++i) { 8114 sp<Track> track = mTracks[i]; 8115 if (session == track->sessionId()) { 8116 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8117 buffer); 8118 track->setMainBuffer(buffer); 8119 chain->incTrackCnt(); 8120 } 8121 } 8122 8123 // indicate all active tracks in the chain 8124 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8125 sp<Track> track = mActiveTracks[i].promote(); 8126 if (track == 0) continue; 8127 if (session == track->sessionId()) { 8128 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8129 chain->incActiveTrackCnt(); 8130 } 8131 } 8132 } 8133 8134 chain->setInBuffer(buffer, ownsBuffer); 8135 chain->setOutBuffer(mMixBuffer); 8136 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8137 // chains list in order to be processed last as it contains output stage effects 8138 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8139 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8140 // after track specific effects and before output stage 8141 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8142 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8143 // Effect chain for other sessions are inserted at beginning of effect 8144 // chains list to be processed before output mix effects. Relative order between other 8145 // sessions is not important 8146 size_t size = mEffectChains.size(); 8147 size_t i = 0; 8148 for (i = 0; i < size; i++) { 8149 if (mEffectChains[i]->sessionId() < session) break; 8150 } 8151 mEffectChains.insertAt(chain, i); 8152 checkSuspendOnAddEffectChain_l(chain); 8153 8154 return NO_ERROR; 8155} 8156 8157size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8158{ 8159 int session = chain->sessionId(); 8160 8161 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8162 8163 for (size_t i = 0; i < mEffectChains.size(); i++) { 8164 if (chain == mEffectChains[i]) { 8165 mEffectChains.removeAt(i); 8166 // detach all active tracks from the chain 8167 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8168 sp<Track> track = mActiveTracks[i].promote(); 8169 if (track == 0) continue; 8170 if (session == track->sessionId()) { 8171 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8172 chain.get(), session); 8173 chain->decActiveTrackCnt(); 8174 } 8175 } 8176 8177 // detach all tracks with same session ID from this chain 8178 for (size_t i = 0; i < mTracks.size(); ++i) { 8179 sp<Track> track = mTracks[i]; 8180 if (session == track->sessionId()) { 8181 track->setMainBuffer(mMixBuffer); 8182 chain->decTrackCnt(); 8183 } 8184 } 8185 break; 8186 } 8187 } 8188 return mEffectChains.size(); 8189} 8190 8191status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8192 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8193{ 8194 Mutex::Autolock _l(mLock); 8195 return attachAuxEffect_l(track, EffectId); 8196} 8197 8198status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8199 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8200{ 8201 status_t status = NO_ERROR; 8202 8203 if (EffectId == 0) { 8204 track->setAuxBuffer(0, NULL); 8205 } else { 8206 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8207 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8208 if (effect != 0) { 8209 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8210 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8211 } else { 8212 status = INVALID_OPERATION; 8213 } 8214 } else { 8215 status = BAD_VALUE; 8216 } 8217 } 8218 return status; 8219} 8220 8221void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8222{ 8223 for (size_t i = 0; i < mTracks.size(); ++i) { 8224 sp<Track> track = mTracks[i]; 8225 if (track->auxEffectId() == effectId) { 8226 attachAuxEffect_l(track, 0); 8227 } 8228 } 8229} 8230 8231status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8232{ 8233 // only one chain per input thread 8234 if (mEffectChains.size() != 0) { 8235 return INVALID_OPERATION; 8236 } 8237 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8238 8239 chain->setInBuffer(NULL); 8240 chain->setOutBuffer(NULL); 8241 8242 checkSuspendOnAddEffectChain_l(chain); 8243 8244 mEffectChains.add(chain); 8245 8246 return NO_ERROR; 8247} 8248 8249size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8250{ 8251 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8252 ALOGW_IF(mEffectChains.size() != 1, 8253 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8254 chain.get(), mEffectChains.size(), this); 8255 if (mEffectChains.size() == 1) { 8256 mEffectChains.removeAt(0); 8257 } 8258 return 0; 8259} 8260 8261// ---------------------------------------------------------------------------- 8262// EffectModule implementation 8263// ---------------------------------------------------------------------------- 8264 8265#undef LOG_TAG 8266#define LOG_TAG "AudioFlinger::EffectModule" 8267 8268AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8269 const wp<AudioFlinger::EffectChain>& chain, 8270 effect_descriptor_t *desc, 8271 int id, 8272 int sessionId) 8273 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8274 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8275 mDescriptor(*desc), 8276 // mConfig is set by configure() and not used before then 8277 mEffectInterface(NULL), 8278 mStatus(NO_INIT), mState(IDLE), 8279 // mMaxDisableWaitCnt is set by configure() and not used before then 8280 // mDisableWaitCnt is set by process() and updateState() and not used before then 8281 mSuspended(false) 8282{ 8283 ALOGV("Constructor %p", this); 8284 int lStatus; 8285 8286 // create effect engine from effect factory 8287 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8288 8289 if (mStatus != NO_ERROR) { 8290 return; 8291 } 8292 lStatus = init(); 8293 if (lStatus < 0) { 8294 mStatus = lStatus; 8295 goto Error; 8296 } 8297 8298 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8299 return; 8300Error: 8301 EffectRelease(mEffectInterface); 8302 mEffectInterface = NULL; 8303 ALOGV("Constructor Error %d", mStatus); 8304} 8305 8306AudioFlinger::EffectModule::~EffectModule() 8307{ 8308 ALOGV("Destructor %p", this); 8309 if (mEffectInterface != NULL) { 8310 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8311 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8312 sp<ThreadBase> thread = mThread.promote(); 8313 if (thread != 0) { 8314 audio_stream_t *stream = thread->stream(); 8315 if (stream != NULL) { 8316 stream->remove_audio_effect(stream, mEffectInterface); 8317 } 8318 } 8319 } 8320 // release effect engine 8321 EffectRelease(mEffectInterface); 8322 } 8323} 8324 8325status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8326{ 8327 status_t status; 8328 8329 Mutex::Autolock _l(mLock); 8330 int priority = handle->priority(); 8331 size_t size = mHandles.size(); 8332 EffectHandle *controlHandle = NULL; 8333 size_t i; 8334 for (i = 0; i < size; i++) { 8335 EffectHandle *h = mHandles[i]; 8336 if (h == NULL || h->destroyed_l()) continue; 8337 // first non destroyed handle is considered in control 8338 if (controlHandle == NULL) 8339 controlHandle = h; 8340 if (h->priority() <= priority) break; 8341 } 8342 // if inserted in first place, move effect control from previous owner to this handle 8343 if (i == 0) { 8344 bool enabled = false; 8345 if (controlHandle != NULL) { 8346 enabled = controlHandle->enabled(); 8347 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8348 } 8349 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8350 status = NO_ERROR; 8351 } else { 8352 status = ALREADY_EXISTS; 8353 } 8354 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8355 mHandles.insertAt(handle, i); 8356 return status; 8357} 8358 8359size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8360{ 8361 Mutex::Autolock _l(mLock); 8362 size_t size = mHandles.size(); 8363 size_t i; 8364 for (i = 0; i < size; i++) { 8365 if (mHandles[i] == handle) break; 8366 } 8367 if (i == size) { 8368 return size; 8369 } 8370 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8371 8372 mHandles.removeAt(i); 8373 // if removed from first place, move effect control from this handle to next in line 8374 if (i == 0) { 8375 EffectHandle *h = controlHandle_l(); 8376 if (h != NULL) { 8377 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8378 } 8379 } 8380 8381 // Prevent calls to process() and other functions on effect interface from now on. 8382 // The effect engine will be released by the destructor when the last strong reference on 8383 // this object is released which can happen after next process is called. 8384 if (mHandles.size() == 0 && !mPinned) { 8385 mState = DESTROYED; 8386 } 8387 8388 return mHandles.size(); 8389} 8390 8391// must be called with EffectModule::mLock held 8392AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8393{ 8394 // the first valid handle in the list has control over the module 8395 for (size_t i = 0; i < mHandles.size(); i++) { 8396 EffectHandle *h = mHandles[i]; 8397 if (h != NULL && !h->destroyed_l()) { 8398 return h; 8399 } 8400 } 8401 8402 return NULL; 8403} 8404 8405size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8406{ 8407 ALOGV("disconnect() %p handle %p", this, handle); 8408 // keep a strong reference on this EffectModule to avoid calling the 8409 // destructor before we exit 8410 sp<EffectModule> keep(this); 8411 { 8412 sp<ThreadBase> thread = mThread.promote(); 8413 if (thread != 0) { 8414 thread->disconnectEffect(keep, handle, unpinIfLast); 8415 } 8416 } 8417 return mHandles.size(); 8418} 8419 8420void AudioFlinger::EffectModule::updateState() { 8421 Mutex::Autolock _l(mLock); 8422 8423 switch (mState) { 8424 case RESTART: 8425 reset_l(); 8426 // FALL THROUGH 8427 8428 case STARTING: 8429 // clear auxiliary effect input buffer for next accumulation 8430 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8431 memset(mConfig.inputCfg.buffer.raw, 8432 0, 8433 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8434 } 8435 start_l(); 8436 mState = ACTIVE; 8437 break; 8438 case STOPPING: 8439 stop_l(); 8440 mDisableWaitCnt = mMaxDisableWaitCnt; 8441 mState = STOPPED; 8442 break; 8443 case STOPPED: 8444 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8445 // turn off sequence. 8446 if (--mDisableWaitCnt == 0) { 8447 reset_l(); 8448 mState = IDLE; 8449 } 8450 break; 8451 default: //IDLE , ACTIVE, DESTROYED 8452 break; 8453 } 8454} 8455 8456void AudioFlinger::EffectModule::process() 8457{ 8458 Mutex::Autolock _l(mLock); 8459 8460 if (mState == DESTROYED || mEffectInterface == NULL || 8461 mConfig.inputCfg.buffer.raw == NULL || 8462 mConfig.outputCfg.buffer.raw == NULL) { 8463 return; 8464 } 8465 8466 if (isProcessEnabled()) { 8467 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8468 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8469 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8470 mConfig.inputCfg.buffer.s32, 8471 mConfig.inputCfg.buffer.frameCount/2); 8472 } 8473 8474 // do the actual processing in the effect engine 8475 int ret = (*mEffectInterface)->process(mEffectInterface, 8476 &mConfig.inputCfg.buffer, 8477 &mConfig.outputCfg.buffer); 8478 8479 // force transition to IDLE state when engine is ready 8480 if (mState == STOPPED && ret == -ENODATA) { 8481 mDisableWaitCnt = 1; 8482 } 8483 8484 // clear auxiliary effect input buffer for next accumulation 8485 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8486 memset(mConfig.inputCfg.buffer.raw, 0, 8487 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8488 } 8489 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8490 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8491 // If an insert effect is idle and input buffer is different from output buffer, 8492 // accumulate input onto output 8493 sp<EffectChain> chain = mChain.promote(); 8494 if (chain != 0 && chain->activeTrackCnt() != 0) { 8495 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8496 int16_t *in = mConfig.inputCfg.buffer.s16; 8497 int16_t *out = mConfig.outputCfg.buffer.s16; 8498 for (size_t i = 0; i < frameCnt; i++) { 8499 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8500 } 8501 } 8502 } 8503} 8504 8505void AudioFlinger::EffectModule::reset_l() 8506{ 8507 if (mEffectInterface == NULL) { 8508 return; 8509 } 8510 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8511} 8512 8513status_t AudioFlinger::EffectModule::configure() 8514{ 8515 if (mEffectInterface == NULL) { 8516 return NO_INIT; 8517 } 8518 8519 sp<ThreadBase> thread = mThread.promote(); 8520 if (thread == 0) { 8521 return DEAD_OBJECT; 8522 } 8523 8524 // TODO: handle configuration of effects replacing track process 8525 audio_channel_mask_t channelMask = thread->channelMask(); 8526 8527 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8528 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8529 } else { 8530 mConfig.inputCfg.channels = channelMask; 8531 } 8532 mConfig.outputCfg.channels = channelMask; 8533 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8534 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8535 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8536 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8537 mConfig.inputCfg.bufferProvider.cookie = NULL; 8538 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8539 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8540 mConfig.outputCfg.bufferProvider.cookie = NULL; 8541 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8542 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8543 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8544 // Insert effect: 8545 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8546 // always overwrites output buffer: input buffer == output buffer 8547 // - in other sessions: 8548 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8549 // other effect: overwrites output buffer: input buffer == output buffer 8550 // Auxiliary effect: 8551 // accumulates in output buffer: input buffer != output buffer 8552 // Therefore: accumulate <=> input buffer != output buffer 8553 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8554 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8555 } else { 8556 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8557 } 8558 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8559 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8560 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8561 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8562 8563 ALOGV("configure() %p thread %p buffer %p framecount %d", 8564 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8565 8566 status_t cmdStatus; 8567 uint32_t size = sizeof(int); 8568 status_t status = (*mEffectInterface)->command(mEffectInterface, 8569 EFFECT_CMD_SET_CONFIG, 8570 sizeof(effect_config_t), 8571 &mConfig, 8572 &size, 8573 &cmdStatus); 8574 if (status == 0) { 8575 status = cmdStatus; 8576 } 8577 8578 if (status == 0 && 8579 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8580 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8581 effect_param_t *p = (effect_param_t *)buf32; 8582 8583 p->psize = sizeof(uint32_t); 8584 p->vsize = sizeof(uint32_t); 8585 size = sizeof(int); 8586 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8587 8588 uint32_t latency = 0; 8589 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8590 if (pbt != NULL) { 8591 latency = pbt->latency_l(); 8592 } 8593 8594 *((int32_t *)p->data + 1)= latency; 8595 (*mEffectInterface)->command(mEffectInterface, 8596 EFFECT_CMD_SET_PARAM, 8597 sizeof(effect_param_t) + 8, 8598 &buf32, 8599 &size, 8600 &cmdStatus); 8601 } 8602 8603 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8604 (1000 * mConfig.outputCfg.buffer.frameCount); 8605 8606 return status; 8607} 8608 8609status_t AudioFlinger::EffectModule::init() 8610{ 8611 Mutex::Autolock _l(mLock); 8612 if (mEffectInterface == NULL) { 8613 return NO_INIT; 8614 } 8615 status_t cmdStatus; 8616 uint32_t size = sizeof(status_t); 8617 status_t status = (*mEffectInterface)->command(mEffectInterface, 8618 EFFECT_CMD_INIT, 8619 0, 8620 NULL, 8621 &size, 8622 &cmdStatus); 8623 if (status == 0) { 8624 status = cmdStatus; 8625 } 8626 return status; 8627} 8628 8629status_t AudioFlinger::EffectModule::start() 8630{ 8631 Mutex::Autolock _l(mLock); 8632 return start_l(); 8633} 8634 8635status_t AudioFlinger::EffectModule::start_l() 8636{ 8637 if (mEffectInterface == NULL) { 8638 return NO_INIT; 8639 } 8640 status_t cmdStatus; 8641 uint32_t size = sizeof(status_t); 8642 status_t status = (*mEffectInterface)->command(mEffectInterface, 8643 EFFECT_CMD_ENABLE, 8644 0, 8645 NULL, 8646 &size, 8647 &cmdStatus); 8648 if (status == 0) { 8649 status = cmdStatus; 8650 } 8651 if (status == 0 && 8652 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8653 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8654 sp<ThreadBase> thread = mThread.promote(); 8655 if (thread != 0) { 8656 audio_stream_t *stream = thread->stream(); 8657 if (stream != NULL) { 8658 stream->add_audio_effect(stream, mEffectInterface); 8659 } 8660 } 8661 } 8662 return status; 8663} 8664 8665status_t AudioFlinger::EffectModule::stop() 8666{ 8667 Mutex::Autolock _l(mLock); 8668 return stop_l(); 8669} 8670 8671status_t AudioFlinger::EffectModule::stop_l() 8672{ 8673 if (mEffectInterface == NULL) { 8674 return NO_INIT; 8675 } 8676 status_t cmdStatus; 8677 uint32_t size = sizeof(status_t); 8678 status_t status = (*mEffectInterface)->command(mEffectInterface, 8679 EFFECT_CMD_DISABLE, 8680 0, 8681 NULL, 8682 &size, 8683 &cmdStatus); 8684 if (status == 0) { 8685 status = cmdStatus; 8686 } 8687 if (status == 0 && 8688 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8689 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8690 sp<ThreadBase> thread = mThread.promote(); 8691 if (thread != 0) { 8692 audio_stream_t *stream = thread->stream(); 8693 if (stream != NULL) { 8694 stream->remove_audio_effect(stream, mEffectInterface); 8695 } 8696 } 8697 } 8698 return status; 8699} 8700 8701status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8702 uint32_t cmdSize, 8703 void *pCmdData, 8704 uint32_t *replySize, 8705 void *pReplyData) 8706{ 8707 Mutex::Autolock _l(mLock); 8708 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8709 8710 if (mState == DESTROYED || mEffectInterface == NULL) { 8711 return NO_INIT; 8712 } 8713 status_t status = (*mEffectInterface)->command(mEffectInterface, 8714 cmdCode, 8715 cmdSize, 8716 pCmdData, 8717 replySize, 8718 pReplyData); 8719 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8720 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8721 for (size_t i = 1; i < mHandles.size(); i++) { 8722 EffectHandle *h = mHandles[i]; 8723 if (h != NULL && !h->destroyed_l()) { 8724 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8725 } 8726 } 8727 } 8728 return status; 8729} 8730 8731status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8732{ 8733 Mutex::Autolock _l(mLock); 8734 return setEnabled_l(enabled); 8735} 8736 8737// must be called with EffectModule::mLock held 8738status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8739{ 8740 8741 ALOGV("setEnabled %p enabled %d", this, enabled); 8742 8743 if (enabled != isEnabled()) { 8744 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8745 if (enabled && status != NO_ERROR) { 8746 return status; 8747 } 8748 8749 switch (mState) { 8750 // going from disabled to enabled 8751 case IDLE: 8752 mState = STARTING; 8753 break; 8754 case STOPPED: 8755 mState = RESTART; 8756 break; 8757 case STOPPING: 8758 mState = ACTIVE; 8759 break; 8760 8761 // going from enabled to disabled 8762 case RESTART: 8763 mState = STOPPED; 8764 break; 8765 case STARTING: 8766 mState = IDLE; 8767 break; 8768 case ACTIVE: 8769 mState = STOPPING; 8770 break; 8771 case DESTROYED: 8772 return NO_ERROR; // simply ignore as we are being destroyed 8773 } 8774 for (size_t i = 1; i < mHandles.size(); i++) { 8775 EffectHandle *h = mHandles[i]; 8776 if (h != NULL && !h->destroyed_l()) { 8777 h->setEnabled(enabled); 8778 } 8779 } 8780 } 8781 return NO_ERROR; 8782} 8783 8784bool AudioFlinger::EffectModule::isEnabled() const 8785{ 8786 switch (mState) { 8787 case RESTART: 8788 case STARTING: 8789 case ACTIVE: 8790 return true; 8791 case IDLE: 8792 case STOPPING: 8793 case STOPPED: 8794 case DESTROYED: 8795 default: 8796 return false; 8797 } 8798} 8799 8800bool AudioFlinger::EffectModule::isProcessEnabled() const 8801{ 8802 switch (mState) { 8803 case RESTART: 8804 case ACTIVE: 8805 case STOPPING: 8806 case STOPPED: 8807 return true; 8808 case IDLE: 8809 case STARTING: 8810 case DESTROYED: 8811 default: 8812 return false; 8813 } 8814} 8815 8816status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8817{ 8818 Mutex::Autolock _l(mLock); 8819 status_t status = NO_ERROR; 8820 8821 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8822 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8823 if (isProcessEnabled() && 8824 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8825 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8826 status_t cmdStatus; 8827 uint32_t volume[2]; 8828 uint32_t *pVolume = NULL; 8829 uint32_t size = sizeof(volume); 8830 volume[0] = *left; 8831 volume[1] = *right; 8832 if (controller) { 8833 pVolume = volume; 8834 } 8835 status = (*mEffectInterface)->command(mEffectInterface, 8836 EFFECT_CMD_SET_VOLUME, 8837 size, 8838 volume, 8839 &size, 8840 pVolume); 8841 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8842 *left = volume[0]; 8843 *right = volume[1]; 8844 } 8845 } 8846 return status; 8847} 8848 8849status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8850{ 8851 if (device == AUDIO_DEVICE_NONE) { 8852 return NO_ERROR; 8853 } 8854 8855 Mutex::Autolock _l(mLock); 8856 status_t status = NO_ERROR; 8857 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8858 status_t cmdStatus; 8859 uint32_t size = sizeof(status_t); 8860 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8861 EFFECT_CMD_SET_INPUT_DEVICE; 8862 status = (*mEffectInterface)->command(mEffectInterface, 8863 cmd, 8864 sizeof(uint32_t), 8865 &device, 8866 &size, 8867 &cmdStatus); 8868 } 8869 return status; 8870} 8871 8872status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8873{ 8874 Mutex::Autolock _l(mLock); 8875 status_t status = NO_ERROR; 8876 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8877 status_t cmdStatus; 8878 uint32_t size = sizeof(status_t); 8879 status = (*mEffectInterface)->command(mEffectInterface, 8880 EFFECT_CMD_SET_AUDIO_MODE, 8881 sizeof(audio_mode_t), 8882 &mode, 8883 &size, 8884 &cmdStatus); 8885 if (status == NO_ERROR) { 8886 status = cmdStatus; 8887 } 8888 } 8889 return status; 8890} 8891 8892status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8893{ 8894 Mutex::Autolock _l(mLock); 8895 status_t status = NO_ERROR; 8896 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8897 uint32_t size = 0; 8898 status = (*mEffectInterface)->command(mEffectInterface, 8899 EFFECT_CMD_SET_AUDIO_SOURCE, 8900 sizeof(audio_source_t), 8901 &source, 8902 &size, 8903 NULL); 8904 } 8905 return status; 8906} 8907 8908void AudioFlinger::EffectModule::setSuspended(bool suspended) 8909{ 8910 Mutex::Autolock _l(mLock); 8911 mSuspended = suspended; 8912} 8913 8914bool AudioFlinger::EffectModule::suspended() const 8915{ 8916 Mutex::Autolock _l(mLock); 8917 return mSuspended; 8918} 8919 8920bool AudioFlinger::EffectModule::purgeHandles() 8921{ 8922 bool enabled = false; 8923 Mutex::Autolock _l(mLock); 8924 for (size_t i = 0; i < mHandles.size(); i++) { 8925 EffectHandle *handle = mHandles[i]; 8926 if (handle != NULL && !handle->destroyed_l()) { 8927 handle->effect().clear(); 8928 if (handle->hasControl()) { 8929 enabled = handle->enabled(); 8930 } 8931 } 8932 } 8933 return enabled; 8934} 8935 8936void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8937{ 8938 const size_t SIZE = 256; 8939 char buffer[SIZE]; 8940 String8 result; 8941 8942 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8943 result.append(buffer); 8944 8945 bool locked = tryLock(mLock); 8946 // failed to lock - AudioFlinger is probably deadlocked 8947 if (!locked) { 8948 result.append("\t\tCould not lock Fx mutex:\n"); 8949 } 8950 8951 result.append("\t\tSession Status State Engine:\n"); 8952 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8953 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8954 result.append(buffer); 8955 8956 result.append("\t\tDescriptor:\n"); 8957 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8958 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8959 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 8960 mDescriptor.uuid.node[2], 8961 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8962 result.append(buffer); 8963 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8964 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 8965 mDescriptor.type.timeHiAndVersion, 8966 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 8967 mDescriptor.type.node[2], 8968 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8969 result.append(buffer); 8970 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8971 mDescriptor.apiVersion, 8972 mDescriptor.flags); 8973 result.append(buffer); 8974 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8975 mDescriptor.name); 8976 result.append(buffer); 8977 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8978 mDescriptor.implementor); 8979 result.append(buffer); 8980 8981 result.append("\t\t- Input configuration:\n"); 8982 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8983 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8984 (uint32_t)mConfig.inputCfg.buffer.raw, 8985 mConfig.inputCfg.buffer.frameCount, 8986 mConfig.inputCfg.samplingRate, 8987 mConfig.inputCfg.channels, 8988 mConfig.inputCfg.format); 8989 result.append(buffer); 8990 8991 result.append("\t\t- Output configuration:\n"); 8992 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8993 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8994 (uint32_t)mConfig.outputCfg.buffer.raw, 8995 mConfig.outputCfg.buffer.frameCount, 8996 mConfig.outputCfg.samplingRate, 8997 mConfig.outputCfg.channels, 8998 mConfig.outputCfg.format); 8999 result.append(buffer); 9000 9001 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 9002 result.append(buffer); 9003 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 9004 for (size_t i = 0; i < mHandles.size(); ++i) { 9005 EffectHandle *handle = mHandles[i]; 9006 if (handle != NULL && !handle->destroyed_l()) { 9007 handle->dump(buffer, SIZE); 9008 result.append(buffer); 9009 } 9010 } 9011 9012 result.append("\n"); 9013 9014 write(fd, result.string(), result.length()); 9015 9016 if (locked) { 9017 mLock.unlock(); 9018 } 9019} 9020 9021// ---------------------------------------------------------------------------- 9022// EffectHandle implementation 9023// ---------------------------------------------------------------------------- 9024 9025#undef LOG_TAG 9026#define LOG_TAG "AudioFlinger::EffectHandle" 9027 9028AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9029 const sp<AudioFlinger::Client>& client, 9030 const sp<IEffectClient>& effectClient, 9031 int32_t priority) 9032 : BnEffect(), 9033 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9034 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9035{ 9036 ALOGV("constructor %p", this); 9037 9038 if (client == 0) { 9039 return; 9040 } 9041 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9042 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9043 if (mCblkMemory != 0) { 9044 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9045 9046 if (mCblk != NULL) { 9047 new(mCblk) effect_param_cblk_t(); 9048 mBuffer = (uint8_t *)mCblk + bufOffset; 9049 } 9050 } else { 9051 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9052 sizeof(effect_param_cblk_t)); 9053 return; 9054 } 9055} 9056 9057AudioFlinger::EffectHandle::~EffectHandle() 9058{ 9059 ALOGV("Destructor %p", this); 9060 9061 if (mEffect == 0) { 9062 mDestroyed = true; 9063 return; 9064 } 9065 mEffect->lock(); 9066 mDestroyed = true; 9067 mEffect->unlock(); 9068 disconnect(false); 9069} 9070 9071status_t AudioFlinger::EffectHandle::enable() 9072{ 9073 ALOGV("enable %p", this); 9074 if (!mHasControl) return INVALID_OPERATION; 9075 if (mEffect == 0) return DEAD_OBJECT; 9076 9077 if (mEnabled) { 9078 return NO_ERROR; 9079 } 9080 9081 mEnabled = true; 9082 9083 sp<ThreadBase> thread = mEffect->thread().promote(); 9084 if (thread != 0) { 9085 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9086 } 9087 9088 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9089 if (mEffect->suspended()) { 9090 return NO_ERROR; 9091 } 9092 9093 status_t status = mEffect->setEnabled(true); 9094 if (status != NO_ERROR) { 9095 if (thread != 0) { 9096 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9097 } 9098 mEnabled = false; 9099 } 9100 return status; 9101} 9102 9103status_t AudioFlinger::EffectHandle::disable() 9104{ 9105 ALOGV("disable %p", this); 9106 if (!mHasControl) return INVALID_OPERATION; 9107 if (mEffect == 0) return DEAD_OBJECT; 9108 9109 if (!mEnabled) { 9110 return NO_ERROR; 9111 } 9112 mEnabled = false; 9113 9114 if (mEffect->suspended()) { 9115 return NO_ERROR; 9116 } 9117 9118 status_t status = mEffect->setEnabled(false); 9119 9120 sp<ThreadBase> thread = mEffect->thread().promote(); 9121 if (thread != 0) { 9122 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9123 } 9124 9125 return status; 9126} 9127 9128void AudioFlinger::EffectHandle::disconnect() 9129{ 9130 disconnect(true); 9131} 9132 9133void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9134{ 9135 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9136 if (mEffect == 0) { 9137 return; 9138 } 9139 // restore suspended effects if the disconnected handle was enabled and the last one. 9140 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9141 sp<ThreadBase> thread = mEffect->thread().promote(); 9142 if (thread != 0) { 9143 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9144 } 9145 } 9146 9147 // release sp on module => module destructor can be called now 9148 mEffect.clear(); 9149 if (mClient != 0) { 9150 if (mCblk != NULL) { 9151 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9152 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9153 } 9154 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9155 // Client destructor must run with AudioFlinger mutex locked 9156 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9157 mClient.clear(); 9158 } 9159} 9160 9161status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9162 uint32_t cmdSize, 9163 void *pCmdData, 9164 uint32_t *replySize, 9165 void *pReplyData) 9166{ 9167 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9168 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9169 9170 // only get parameter command is permitted for applications not controlling the effect 9171 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9172 return INVALID_OPERATION; 9173 } 9174 if (mEffect == 0) return DEAD_OBJECT; 9175 if (mClient == 0) return INVALID_OPERATION; 9176 9177 // handle commands that are not forwarded transparently to effect engine 9178 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9179 // No need to trylock() here as this function is executed in the binder thread serving a 9180 // particular client process: no risk to block the whole media server process or mixer 9181 // threads if we are stuck here 9182 Mutex::Autolock _l(mCblk->lock); 9183 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9184 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9185 mCblk->serverIndex = 0; 9186 mCblk->clientIndex = 0; 9187 return BAD_VALUE; 9188 } 9189 status_t status = NO_ERROR; 9190 while (mCblk->serverIndex < mCblk->clientIndex) { 9191 int reply; 9192 uint32_t rsize = sizeof(int); 9193 int *p = (int *)(mBuffer + mCblk->serverIndex); 9194 int size = *p++; 9195 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9196 ALOGW("command(): invalid parameter block size"); 9197 break; 9198 } 9199 effect_param_t *param = (effect_param_t *)p; 9200 if (param->psize == 0 || param->vsize == 0) { 9201 ALOGW("command(): null parameter or value size"); 9202 mCblk->serverIndex += size; 9203 continue; 9204 } 9205 uint32_t psize = sizeof(effect_param_t) + 9206 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9207 param->vsize; 9208 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9209 psize, 9210 p, 9211 &rsize, 9212 &reply); 9213 // stop at first error encountered 9214 if (ret != NO_ERROR) { 9215 status = ret; 9216 *(int *)pReplyData = reply; 9217 break; 9218 } else if (reply != NO_ERROR) { 9219 *(int *)pReplyData = reply; 9220 break; 9221 } 9222 mCblk->serverIndex += size; 9223 } 9224 mCblk->serverIndex = 0; 9225 mCblk->clientIndex = 0; 9226 return status; 9227 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9228 *(int *)pReplyData = NO_ERROR; 9229 return enable(); 9230 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9231 *(int *)pReplyData = NO_ERROR; 9232 return disable(); 9233 } 9234 9235 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9236} 9237 9238void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9239{ 9240 ALOGV("setControl %p control %d", this, hasControl); 9241 9242 mHasControl = hasControl; 9243 mEnabled = enabled; 9244 9245 if (signal && mEffectClient != 0) { 9246 mEffectClient->controlStatusChanged(hasControl); 9247 } 9248} 9249 9250void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9251 uint32_t cmdSize, 9252 void *pCmdData, 9253 uint32_t replySize, 9254 void *pReplyData) 9255{ 9256 if (mEffectClient != 0) { 9257 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9258 } 9259} 9260 9261 9262 9263void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9264{ 9265 if (mEffectClient != 0) { 9266 mEffectClient->enableStatusChanged(enabled); 9267 } 9268} 9269 9270status_t AudioFlinger::EffectHandle::onTransact( 9271 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9272{ 9273 return BnEffect::onTransact(code, data, reply, flags); 9274} 9275 9276 9277void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9278{ 9279 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9280 9281 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9282 (mClient == 0) ? getpid_cached : mClient->pid(), 9283 mPriority, 9284 mHasControl, 9285 !locked, 9286 mCblk ? mCblk->clientIndex : 0, 9287 mCblk ? mCblk->serverIndex : 0 9288 ); 9289 9290 if (locked) { 9291 mCblk->lock.unlock(); 9292 } 9293} 9294 9295#undef LOG_TAG 9296#define LOG_TAG "AudioFlinger::EffectChain" 9297 9298AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9299 int sessionId) 9300 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9301 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9302 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9303{ 9304 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9305 if (thread == NULL) { 9306 return; 9307 } 9308 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9309 thread->frameCount(); 9310} 9311 9312AudioFlinger::EffectChain::~EffectChain() 9313{ 9314 if (mOwnInBuffer) { 9315 delete mInBuffer; 9316 } 9317 9318} 9319 9320// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9321sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9322 effect_descriptor_t *descriptor) 9323{ 9324 size_t size = mEffects.size(); 9325 9326 for (size_t i = 0; i < size; i++) { 9327 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9328 return mEffects[i]; 9329 } 9330 } 9331 return 0; 9332} 9333 9334// getEffectFromId_l() must be called with ThreadBase::mLock held 9335sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9336{ 9337 size_t size = mEffects.size(); 9338 9339 for (size_t i = 0; i < size; i++) { 9340 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9341 if (id == 0 || mEffects[i]->id() == id) { 9342 return mEffects[i]; 9343 } 9344 } 9345 return 0; 9346} 9347 9348// getEffectFromType_l() must be called with ThreadBase::mLock held 9349sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9350 const effect_uuid_t *type) 9351{ 9352 size_t size = mEffects.size(); 9353 9354 for (size_t i = 0; i < size; i++) { 9355 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9356 return mEffects[i]; 9357 } 9358 } 9359 return 0; 9360} 9361 9362void AudioFlinger::EffectChain::clearInputBuffer() 9363{ 9364 Mutex::Autolock _l(mLock); 9365 sp<ThreadBase> thread = mThread.promote(); 9366 if (thread == 0) { 9367 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9368 return; 9369 } 9370 clearInputBuffer_l(thread); 9371} 9372 9373// Must be called with EffectChain::mLock locked 9374void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9375{ 9376 size_t numSamples = thread->frameCount() * thread->channelCount(); 9377 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9378 9379} 9380 9381// Must be called with EffectChain::mLock locked 9382void AudioFlinger::EffectChain::process_l() 9383{ 9384 sp<ThreadBase> thread = mThread.promote(); 9385 if (thread == 0) { 9386 ALOGW("process_l(): cannot promote mixer thread"); 9387 return; 9388 } 9389 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9390 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9391 // always process effects unless no more tracks are on the session and the effect tail 9392 // has been rendered 9393 bool doProcess = true; 9394 if (!isGlobalSession) { 9395 bool tracksOnSession = (trackCnt() != 0); 9396 9397 if (!tracksOnSession && mTailBufferCount == 0) { 9398 doProcess = false; 9399 } 9400 9401 if (activeTrackCnt() == 0) { 9402 // if no track is active and the effect tail has not been rendered, 9403 // the input buffer must be cleared here as the mixer process will not do it 9404 if (tracksOnSession || mTailBufferCount > 0) { 9405 clearInputBuffer_l(thread); 9406 if (mTailBufferCount > 0) { 9407 mTailBufferCount--; 9408 } 9409 } 9410 } 9411 } 9412 9413 size_t size = mEffects.size(); 9414 if (doProcess) { 9415 for (size_t i = 0; i < size; i++) { 9416 mEffects[i]->process(); 9417 } 9418 } 9419 for (size_t i = 0; i < size; i++) { 9420 mEffects[i]->updateState(); 9421 } 9422} 9423 9424// addEffect_l() must be called with PlaybackThread::mLock held 9425status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9426{ 9427 effect_descriptor_t desc = effect->desc(); 9428 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9429 9430 Mutex::Autolock _l(mLock); 9431 effect->setChain(this); 9432 sp<ThreadBase> thread = mThread.promote(); 9433 if (thread == 0) { 9434 return NO_INIT; 9435 } 9436 effect->setThread(thread); 9437 9438 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9439 // Auxiliary effects are inserted at the beginning of mEffects vector as 9440 // they are processed first and accumulated in chain input buffer 9441 mEffects.insertAt(effect, 0); 9442 9443 // the input buffer for auxiliary effect contains mono samples in 9444 // 32 bit format. This is to avoid saturation in AudoMixer 9445 // accumulation stage. Saturation is done in EffectModule::process() before 9446 // calling the process in effect engine 9447 size_t numSamples = thread->frameCount(); 9448 int32_t *buffer = new int32_t[numSamples]; 9449 memset(buffer, 0, numSamples * sizeof(int32_t)); 9450 effect->setInBuffer((int16_t *)buffer); 9451 // auxiliary effects output samples to chain input buffer for further processing 9452 // by insert effects 9453 effect->setOutBuffer(mInBuffer); 9454 } else { 9455 // Insert effects are inserted at the end of mEffects vector as they are processed 9456 // after track and auxiliary effects. 9457 // Insert effect order as a function of indicated preference: 9458 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9459 // another effect is present 9460 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9461 // last effect claiming first position 9462 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9463 // first effect claiming last position 9464 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9465 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9466 // already present 9467 9468 size_t size = mEffects.size(); 9469 size_t idx_insert = size; 9470 ssize_t idx_insert_first = -1; 9471 ssize_t idx_insert_last = -1; 9472 9473 for (size_t i = 0; i < size; i++) { 9474 effect_descriptor_t d = mEffects[i]->desc(); 9475 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9476 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9477 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9478 // check invalid effect chaining combinations 9479 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9480 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9481 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9482 desc.name, d.name); 9483 return INVALID_OPERATION; 9484 } 9485 // remember position of first insert effect and by default 9486 // select this as insert position for new effect 9487 if (idx_insert == size) { 9488 idx_insert = i; 9489 } 9490 // remember position of last insert effect claiming 9491 // first position 9492 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9493 idx_insert_first = i; 9494 } 9495 // remember position of first insert effect claiming 9496 // last position 9497 if (iPref == EFFECT_FLAG_INSERT_LAST && 9498 idx_insert_last == -1) { 9499 idx_insert_last = i; 9500 } 9501 } 9502 } 9503 9504 // modify idx_insert from first position if needed 9505 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9506 if (idx_insert_last != -1) { 9507 idx_insert = idx_insert_last; 9508 } else { 9509 idx_insert = size; 9510 } 9511 } else { 9512 if (idx_insert_first != -1) { 9513 idx_insert = idx_insert_first + 1; 9514 } 9515 } 9516 9517 // always read samples from chain input buffer 9518 effect->setInBuffer(mInBuffer); 9519 9520 // if last effect in the chain, output samples to chain 9521 // output buffer, otherwise to chain input buffer 9522 if (idx_insert == size) { 9523 if (idx_insert != 0) { 9524 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9525 mEffects[idx_insert-1]->configure(); 9526 } 9527 effect->setOutBuffer(mOutBuffer); 9528 } else { 9529 effect->setOutBuffer(mInBuffer); 9530 } 9531 mEffects.insertAt(effect, idx_insert); 9532 9533 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9534 idx_insert); 9535 } 9536 effect->configure(); 9537 return NO_ERROR; 9538} 9539 9540// removeEffect_l() must be called with PlaybackThread::mLock held 9541size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9542{ 9543 Mutex::Autolock _l(mLock); 9544 size_t size = mEffects.size(); 9545 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9546 9547 for (size_t i = 0; i < size; i++) { 9548 if (effect == mEffects[i]) { 9549 // calling stop here will remove pre-processing effect from the audio HAL. 9550 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9551 // the middle of a read from audio HAL 9552 if (mEffects[i]->state() == EffectModule::ACTIVE || 9553 mEffects[i]->state() == EffectModule::STOPPING) { 9554 mEffects[i]->stop(); 9555 } 9556 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9557 delete[] effect->inBuffer(); 9558 } else { 9559 if (i == size - 1 && i != 0) { 9560 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9561 mEffects[i - 1]->configure(); 9562 } 9563 } 9564 mEffects.removeAt(i); 9565 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9566 this, i); 9567 break; 9568 } 9569 } 9570 9571 return mEffects.size(); 9572} 9573 9574// setDevice_l() must be called with PlaybackThread::mLock held 9575void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9576{ 9577 size_t size = mEffects.size(); 9578 for (size_t i = 0; i < size; i++) { 9579 mEffects[i]->setDevice(device); 9580 } 9581} 9582 9583// setMode_l() must be called with PlaybackThread::mLock held 9584void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9585{ 9586 size_t size = mEffects.size(); 9587 for (size_t i = 0; i < size; i++) { 9588 mEffects[i]->setMode(mode); 9589 } 9590} 9591 9592// setAudioSource_l() must be called with PlaybackThread::mLock held 9593void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9594{ 9595 size_t size = mEffects.size(); 9596 for (size_t i = 0; i < size; i++) { 9597 mEffects[i]->setAudioSource(source); 9598 } 9599} 9600 9601// setVolume_l() must be called with PlaybackThread::mLock held 9602bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9603{ 9604 uint32_t newLeft = *left; 9605 uint32_t newRight = *right; 9606 bool hasControl = false; 9607 int ctrlIdx = -1; 9608 size_t size = mEffects.size(); 9609 9610 // first update volume controller 9611 for (size_t i = size; i > 0; i--) { 9612 if (mEffects[i - 1]->isProcessEnabled() && 9613 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9614 ctrlIdx = i - 1; 9615 hasControl = true; 9616 break; 9617 } 9618 } 9619 9620 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9621 if (hasControl) { 9622 *left = mNewLeftVolume; 9623 *right = mNewRightVolume; 9624 } 9625 return hasControl; 9626 } 9627 9628 mVolumeCtrlIdx = ctrlIdx; 9629 mLeftVolume = newLeft; 9630 mRightVolume = newRight; 9631 9632 // second get volume update from volume controller 9633 if (ctrlIdx >= 0) { 9634 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9635 mNewLeftVolume = newLeft; 9636 mNewRightVolume = newRight; 9637 } 9638 // then indicate volume to all other effects in chain. 9639 // Pass altered volume to effects before volume controller 9640 // and requested volume to effects after controller 9641 uint32_t lVol = newLeft; 9642 uint32_t rVol = newRight; 9643 9644 for (size_t i = 0; i < size; i++) { 9645 if ((int)i == ctrlIdx) continue; 9646 // this also works for ctrlIdx == -1 when there is no volume controller 9647 if ((int)i > ctrlIdx) { 9648 lVol = *left; 9649 rVol = *right; 9650 } 9651 mEffects[i]->setVolume(&lVol, &rVol, false); 9652 } 9653 *left = newLeft; 9654 *right = newRight; 9655 9656 return hasControl; 9657} 9658 9659void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9660{ 9661 const size_t SIZE = 256; 9662 char buffer[SIZE]; 9663 String8 result; 9664 9665 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9666 result.append(buffer); 9667 9668 bool locked = tryLock(mLock); 9669 // failed to lock - AudioFlinger is probably deadlocked 9670 if (!locked) { 9671 result.append("\tCould not lock mutex:\n"); 9672 } 9673 9674 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9675 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9676 mEffects.size(), 9677 (uint32_t)mInBuffer, 9678 (uint32_t)mOutBuffer, 9679 mActiveTrackCnt); 9680 result.append(buffer); 9681 write(fd, result.string(), result.size()); 9682 9683 for (size_t i = 0; i < mEffects.size(); ++i) { 9684 sp<EffectModule> effect = mEffects[i]; 9685 if (effect != 0) { 9686 effect->dump(fd, args); 9687 } 9688 } 9689 9690 if (locked) { 9691 mLock.unlock(); 9692 } 9693} 9694 9695// must be called with ThreadBase::mLock held 9696void AudioFlinger::EffectChain::setEffectSuspended_l( 9697 const effect_uuid_t *type, bool suspend) 9698{ 9699 sp<SuspendedEffectDesc> desc; 9700 // use effect type UUID timelow as key as there is no real risk of identical 9701 // timeLow fields among effect type UUIDs. 9702 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9703 if (suspend) { 9704 if (index >= 0) { 9705 desc = mSuspendedEffects.valueAt(index); 9706 } else { 9707 desc = new SuspendedEffectDesc(); 9708 desc->mType = *type; 9709 mSuspendedEffects.add(type->timeLow, desc); 9710 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9711 } 9712 if (desc->mRefCount++ == 0) { 9713 sp<EffectModule> effect = getEffectIfEnabled(type); 9714 if (effect != 0) { 9715 desc->mEffect = effect; 9716 effect->setSuspended(true); 9717 effect->setEnabled(false); 9718 } 9719 } 9720 } else { 9721 if (index < 0) { 9722 return; 9723 } 9724 desc = mSuspendedEffects.valueAt(index); 9725 if (desc->mRefCount <= 0) { 9726 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9727 desc->mRefCount = 1; 9728 } 9729 if (--desc->mRefCount == 0) { 9730 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9731 if (desc->mEffect != 0) { 9732 sp<EffectModule> effect = desc->mEffect.promote(); 9733 if (effect != 0) { 9734 effect->setSuspended(false); 9735 effect->lock(); 9736 EffectHandle *handle = effect->controlHandle_l(); 9737 if (handle != NULL && !handle->destroyed_l()) { 9738 effect->setEnabled_l(handle->enabled()); 9739 } 9740 effect->unlock(); 9741 } 9742 desc->mEffect.clear(); 9743 } 9744 mSuspendedEffects.removeItemsAt(index); 9745 } 9746 } 9747} 9748 9749// must be called with ThreadBase::mLock held 9750void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9751{ 9752 sp<SuspendedEffectDesc> desc; 9753 9754 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9755 if (suspend) { 9756 if (index >= 0) { 9757 desc = mSuspendedEffects.valueAt(index); 9758 } else { 9759 desc = new SuspendedEffectDesc(); 9760 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9761 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9762 } 9763 if (desc->mRefCount++ == 0) { 9764 Vector< sp<EffectModule> > effects; 9765 getSuspendEligibleEffects(effects); 9766 for (size_t i = 0; i < effects.size(); i++) { 9767 setEffectSuspended_l(&effects[i]->desc().type, true); 9768 } 9769 } 9770 } else { 9771 if (index < 0) { 9772 return; 9773 } 9774 desc = mSuspendedEffects.valueAt(index); 9775 if (desc->mRefCount <= 0) { 9776 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9777 desc->mRefCount = 1; 9778 } 9779 if (--desc->mRefCount == 0) { 9780 Vector<const effect_uuid_t *> types; 9781 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9782 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9783 continue; 9784 } 9785 types.add(&mSuspendedEffects.valueAt(i)->mType); 9786 } 9787 for (size_t i = 0; i < types.size(); i++) { 9788 setEffectSuspended_l(types[i], false); 9789 } 9790 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9791 mSuspendedEffects.keyAt(index)); 9792 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9793 } 9794 } 9795} 9796 9797 9798// The volume effect is used for automated tests only 9799#ifndef OPENSL_ES_H_ 9800static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9801 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9802const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9803#endif //OPENSL_ES_H_ 9804 9805bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9806{ 9807 // auxiliary effects and visualizer are never suspended on output mix 9808 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9809 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9810 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9811 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9812 return false; 9813 } 9814 return true; 9815} 9816 9817void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9818 Vector< sp<AudioFlinger::EffectModule> > &effects) 9819{ 9820 effects.clear(); 9821 for (size_t i = 0; i < mEffects.size(); i++) { 9822 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9823 effects.add(mEffects[i]); 9824 } 9825 } 9826} 9827 9828sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9829 const effect_uuid_t *type) 9830{ 9831 sp<EffectModule> effect = getEffectFromType_l(type); 9832 return effect != 0 && effect->isEnabled() ? effect : 0; 9833} 9834 9835void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9836 bool enabled) 9837{ 9838 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9839 if (enabled) { 9840 if (index < 0) { 9841 // if the effect is not suspend check if all effects are suspended 9842 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9843 if (index < 0) { 9844 return; 9845 } 9846 if (!isEffectEligibleForSuspend(effect->desc())) { 9847 return; 9848 } 9849 setEffectSuspended_l(&effect->desc().type, enabled); 9850 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9851 if (index < 0) { 9852 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9853 return; 9854 } 9855 } 9856 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9857 effect->desc().type.timeLow); 9858 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9859 // if effect is requested to suspended but was not yet enabled, supend it now. 9860 if (desc->mEffect == 0) { 9861 desc->mEffect = effect; 9862 effect->setEnabled(false); 9863 effect->setSuspended(true); 9864 } 9865 } else { 9866 if (index < 0) { 9867 return; 9868 } 9869 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9870 effect->desc().type.timeLow); 9871 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9872 desc->mEffect.clear(); 9873 effect->setSuspended(false); 9874 } 9875} 9876 9877#undef LOG_TAG 9878#define LOG_TAG "AudioFlinger" 9879 9880// ---------------------------------------------------------------------------- 9881 9882status_t AudioFlinger::onTransact( 9883 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9884{ 9885 return BnAudioFlinger::onTransact(code, data, reply, flags); 9886} 9887 9888}; // namespace android 9889