AudioFlinger.cpp revision f1c04f952916cf70407051c9f824ab84fb2b6e09
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 if (locked) mLock.unlock(); 421 } 422 return NO_ERROR; 423} 424 425sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 426{ 427 // If pid is already in the mClients wp<> map, then use that entry 428 // (for which promote() is always != 0), otherwise create a new entry and Client. 429 sp<Client> client = mClients.valueFor(pid).promote(); 430 if (client == 0) { 431 client = new Client(this, pid); 432 mClients.add(pid, client); 433 } 434 435 return client; 436} 437 438// IAudioFlinger interface 439 440 441sp<IAudioTrack> AudioFlinger::createTrack( 442 pid_t pid, 443 audio_stream_type_t streamType, 444 uint32_t sampleRate, 445 audio_format_t format, 446 audio_channel_mask_t channelMask, 447 int frameCount, 448 IAudioFlinger::track_flags_t flags, 449 const sp<IMemory>& sharedBuffer, 450 audio_io_handle_t output, 451 pid_t tid, 452 int *sessionId, 453 status_t *status) 454{ 455 sp<PlaybackThread::Track> track; 456 sp<TrackHandle> trackHandle; 457 sp<Client> client; 458 status_t lStatus; 459 int lSessionId; 460 461 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 462 // but if someone uses binder directly they could bypass that and cause us to crash 463 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 464 ALOGE("createTrack() invalid stream type %d", streamType); 465 lStatus = BAD_VALUE; 466 goto Exit; 467 } 468 469 { 470 Mutex::Autolock _l(mLock); 471 PlaybackThread *thread = checkPlaybackThread_l(output); 472 PlaybackThread *effectThread = NULL; 473 if (thread == NULL) { 474 ALOGE("unknown output thread"); 475 lStatus = BAD_VALUE; 476 goto Exit; 477 } 478 479 client = registerPid_l(pid); 480 481 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 482 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 483 // check if an effect chain with the same session ID is present on another 484 // output thread and move it here. 485 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 486 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 487 if (mPlaybackThreads.keyAt(i) != output) { 488 uint32_t sessions = t->hasAudioSession(*sessionId); 489 if (sessions & PlaybackThread::EFFECT_SESSION) { 490 effectThread = t.get(); 491 break; 492 } 493 } 494 } 495 lSessionId = *sessionId; 496 } else { 497 // if no audio session id is provided, create one here 498 lSessionId = nextUniqueId(); 499 if (sessionId != NULL) { 500 *sessionId = lSessionId; 501 } 502 } 503 ALOGV("createTrack() lSessionId: %d", lSessionId); 504 505 track = thread->createTrack_l(client, streamType, sampleRate, format, 506 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 507 508 // move effect chain to this output thread if an effect on same session was waiting 509 // for a track to be created 510 if (lStatus == NO_ERROR && effectThread != NULL) { 511 Mutex::Autolock _dl(thread->mLock); 512 Mutex::Autolock _sl(effectThread->mLock); 513 moveEffectChain_l(lSessionId, effectThread, thread, true); 514 } 515 516 // Look for sync events awaiting for a session to be used. 517 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 518 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 519 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 520 if (lStatus == NO_ERROR) { 521 (void) track->setSyncEvent(mPendingSyncEvents[i]); 522 } else { 523 mPendingSyncEvents[i]->cancel(); 524 } 525 mPendingSyncEvents.removeAt(i); 526 i--; 527 } 528 } 529 } 530 } 531 if (lStatus == NO_ERROR) { 532 trackHandle = new TrackHandle(track); 533 } else { 534 // remove local strong reference to Client before deleting the Track so that the Client 535 // destructor is called by the TrackBase destructor with mLock held 536 client.clear(); 537 track.clear(); 538 } 539 540Exit: 541 if (status != NULL) { 542 *status = lStatus; 543 } 544 return trackHandle; 545} 546 547uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("sampleRate() unknown thread %d", output); 553 return 0; 554 } 555 return thread->sampleRate(); 556} 557 558int AudioFlinger::channelCount(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("channelCount() unknown thread %d", output); 564 return 0; 565 } 566 return thread->channelCount(); 567} 568 569audio_format_t AudioFlinger::format(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("format() unknown thread %d", output); 575 return AUDIO_FORMAT_INVALID; 576 } 577 return thread->format(); 578} 579 580size_t AudioFlinger::frameCount(audio_io_handle_t output) const 581{ 582 Mutex::Autolock _l(mLock); 583 PlaybackThread *thread = checkPlaybackThread_l(output); 584 if (thread == NULL) { 585 ALOGW("frameCount() unknown thread %d", output); 586 return 0; 587 } 588 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 589 // should examine all callers and fix them to handle smaller counts 590 return thread->frameCount(); 591} 592 593uint32_t AudioFlinger::latency(audio_io_handle_t output) const 594{ 595 Mutex::Autolock _l(mLock); 596 PlaybackThread *thread = checkPlaybackThread_l(output); 597 if (thread == NULL) { 598 ALOGW("latency() unknown thread %d", output); 599 return 0; 600 } 601 return thread->latency(); 602} 603 604status_t AudioFlinger::setMasterVolume(float value) 605{ 606 status_t ret = initCheck(); 607 if (ret != NO_ERROR) { 608 return ret; 609 } 610 611 // check calling permissions 612 if (!settingsAllowed()) { 613 return PERMISSION_DENIED; 614 } 615 616 Mutex::Autolock _l(mLock); 617 mMasterVolume = value; 618 619 // Set master volume in the HALs which support it. 620 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 621 AutoMutex lock(mHardwareLock); 622 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 623 624 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 625 if (dev->canSetMasterVolume()) { 626 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 627 } 628 mHardwareStatus = AUDIO_HW_IDLE; 629 } 630 631 // Now set the master volume in each playback thread. Playback threads 632 // assigned to HALs which do not have master volume support will apply 633 // master volume during the mix operation. Threads with HALs which do 634 // support master volume will simply ignore the setting. 635 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 636 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 637 638 return NO_ERROR; 639} 640 641status_t AudioFlinger::setMode(audio_mode_t mode) 642{ 643 status_t ret = initCheck(); 644 if (ret != NO_ERROR) { 645 return ret; 646 } 647 648 // check calling permissions 649 if (!settingsAllowed()) { 650 return PERMISSION_DENIED; 651 } 652 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 653 ALOGW("Illegal value: setMode(%d)", mode); 654 return BAD_VALUE; 655 } 656 657 { // scope for the lock 658 AutoMutex lock(mHardwareLock); 659 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 660 mHardwareStatus = AUDIO_HW_SET_MODE; 661 ret = dev->set_mode(dev, mode); 662 mHardwareStatus = AUDIO_HW_IDLE; 663 } 664 665 if (NO_ERROR == ret) { 666 Mutex::Autolock _l(mLock); 667 mMode = mode; 668 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 669 mPlaybackThreads.valueAt(i)->setMode(mode); 670 } 671 672 return ret; 673} 674 675status_t AudioFlinger::setMicMute(bool state) 676{ 677 status_t ret = initCheck(); 678 if (ret != NO_ERROR) { 679 return ret; 680 } 681 682 // check calling permissions 683 if (!settingsAllowed()) { 684 return PERMISSION_DENIED; 685 } 686 687 AutoMutex lock(mHardwareLock); 688 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 689 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 690 ret = dev->set_mic_mute(dev, state); 691 mHardwareStatus = AUDIO_HW_IDLE; 692 return ret; 693} 694 695bool AudioFlinger::getMicMute() const 696{ 697 status_t ret = initCheck(); 698 if (ret != NO_ERROR) { 699 return false; 700 } 701 702 bool state = AUDIO_MODE_INVALID; 703 AutoMutex lock(mHardwareLock); 704 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 705 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 706 dev->get_mic_mute(dev, &state); 707 mHardwareStatus = AUDIO_HW_IDLE; 708 return state; 709} 710 711status_t AudioFlinger::setMasterMute(bool muted) 712{ 713 status_t ret = initCheck(); 714 if (ret != NO_ERROR) { 715 return ret; 716 } 717 718 // check calling permissions 719 if (!settingsAllowed()) { 720 return PERMISSION_DENIED; 721 } 722 723 Mutex::Autolock _l(mLock); 724 mMasterMute = muted; 725 726 // Set master mute in the HALs which support it. 727 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 728 AutoMutex lock(mHardwareLock); 729 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 730 731 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 732 if (dev->canSetMasterMute()) { 733 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 734 } 735 mHardwareStatus = AUDIO_HW_IDLE; 736 } 737 738 // Now set the master mute in each playback thread. Playback threads 739 // assigned to HALs which do not have master mute support will apply master 740 // mute during the mix operation. Threads with HALs which do support master 741 // mute will simply ignore the setting. 742 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 743 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 744 745 return NO_ERROR; 746} 747 748float AudioFlinger::masterVolume() const 749{ 750 Mutex::Autolock _l(mLock); 751 return masterVolume_l(); 752} 753 754bool AudioFlinger::masterMute() const 755{ 756 Mutex::Autolock _l(mLock); 757 return masterMute_l(); 758} 759 760float AudioFlinger::masterVolume_l() const 761{ 762 return mMasterVolume; 763} 764 765bool AudioFlinger::masterMute_l() const 766{ 767 return mMasterMute; 768} 769 770status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 771 audio_io_handle_t output) 772{ 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 778 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 779 ALOGE("setStreamVolume() invalid stream %d", stream); 780 return BAD_VALUE; 781 } 782 783 AutoMutex lock(mLock); 784 PlaybackThread *thread = NULL; 785 if (output) { 786 thread = checkPlaybackThread_l(output); 787 if (thread == NULL) { 788 return BAD_VALUE; 789 } 790 } 791 792 mStreamTypes[stream].volume = value; 793 794 if (thread == NULL) { 795 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 796 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 797 } 798 } else { 799 thread->setStreamVolume(stream, value); 800 } 801 802 return NO_ERROR; 803} 804 805status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 806{ 807 // check calling permissions 808 if (!settingsAllowed()) { 809 return PERMISSION_DENIED; 810 } 811 812 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 813 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 814 ALOGE("setStreamMute() invalid stream %d", stream); 815 return BAD_VALUE; 816 } 817 818 AutoMutex lock(mLock); 819 mStreamTypes[stream].mute = muted; 820 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 821 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 822 823 return NO_ERROR; 824} 825 826float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 827{ 828 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 829 return 0.0f; 830 } 831 832 AutoMutex lock(mLock); 833 float volume; 834 if (output) { 835 PlaybackThread *thread = checkPlaybackThread_l(output); 836 if (thread == NULL) { 837 return 0.0f; 838 } 839 volume = thread->streamVolume(stream); 840 } else { 841 volume = streamVolume_l(stream); 842 } 843 844 return volume; 845} 846 847bool AudioFlinger::streamMute(audio_stream_type_t stream) const 848{ 849 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 850 return true; 851 } 852 853 AutoMutex lock(mLock); 854 return streamMute_l(stream); 855} 856 857status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 858{ 859 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 860 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 861 // check calling permissions 862 if (!settingsAllowed()) { 863 return PERMISSION_DENIED; 864 } 865 866 // ioHandle == 0 means the parameters are global to the audio hardware interface 867 if (ioHandle == 0) { 868 Mutex::Autolock _l(mLock); 869 status_t final_result = NO_ERROR; 870 { 871 AutoMutex lock(mHardwareLock); 872 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 873 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 874 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 875 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 876 final_result = result ?: final_result; 877 } 878 mHardwareStatus = AUDIO_HW_IDLE; 879 } 880 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 881 AudioParameter param = AudioParameter(keyValuePairs); 882 String8 value; 883 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 884 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 885 if (mBtNrecIsOff != btNrecIsOff) { 886 for (size_t i = 0; i < mRecordThreads.size(); i++) { 887 sp<RecordThread> thread = mRecordThreads.valueAt(i); 888 audio_devices_t device = thread->inDevice(); 889 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 890 // collect all of the thread's session IDs 891 KeyedVector<int, bool> ids = thread->sessionIds(); 892 // suspend effects associated with those session IDs 893 for (size_t j = 0; j < ids.size(); ++j) { 894 int sessionId = ids.keyAt(j); 895 thread->setEffectSuspended(FX_IID_AEC, 896 suspend, 897 sessionId); 898 thread->setEffectSuspended(FX_IID_NS, 899 suspend, 900 sessionId); 901 } 902 } 903 mBtNrecIsOff = btNrecIsOff; 904 } 905 } 906 String8 screenState; 907 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 908 bool isOff = screenState == "off"; 909 if (isOff != (gScreenState & 1)) { 910 gScreenState = ((gScreenState & ~1) + 2) | isOff; 911 } 912 } 913 return final_result; 914 } 915 916 // hold a strong ref on thread in case closeOutput() or closeInput() is called 917 // and the thread is exited once the lock is released 918 sp<ThreadBase> thread; 919 { 920 Mutex::Autolock _l(mLock); 921 thread = checkPlaybackThread_l(ioHandle); 922 if (thread == 0) { 923 thread = checkRecordThread_l(ioHandle); 924 } else if (thread == primaryPlaybackThread_l()) { 925 // indicate output device change to all input threads for pre processing 926 AudioParameter param = AudioParameter(keyValuePairs); 927 int value; 928 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 929 (value != 0)) { 930 for (size_t i = 0; i < mRecordThreads.size(); i++) { 931 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 932 } 933 } 934 } 935 } 936 if (thread != 0) { 937 return thread->setParameters(keyValuePairs); 938 } 939 return BAD_VALUE; 940} 941 942String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 943{ 944// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 945// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 946 947 Mutex::Autolock _l(mLock); 948 949 if (ioHandle == 0) { 950 String8 out_s8; 951 952 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 953 char *s; 954 { 955 AutoMutex lock(mHardwareLock); 956 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 957 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 958 s = dev->get_parameters(dev, keys.string()); 959 mHardwareStatus = AUDIO_HW_IDLE; 960 } 961 out_s8 += String8(s ? s : ""); 962 free(s); 963 } 964 return out_s8; 965 } 966 967 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 968 if (playbackThread != NULL) { 969 return playbackThread->getParameters(keys); 970 } 971 RecordThread *recordThread = checkRecordThread_l(ioHandle); 972 if (recordThread != NULL) { 973 return recordThread->getParameters(keys); 974 } 975 return String8(""); 976} 977 978size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 979 audio_channel_mask_t channelMask) const 980{ 981 status_t ret = initCheck(); 982 if (ret != NO_ERROR) { 983 return 0; 984 } 985 986 AutoMutex lock(mHardwareLock); 987 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 988 struct audio_config config = { 989 sample_rate: sampleRate, 990 channel_mask: channelMask, 991 format: format, 992 }; 993 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 994 size_t size = dev->get_input_buffer_size(dev, &config); 995 mHardwareStatus = AUDIO_HW_IDLE; 996 return size; 997} 998 999unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1000{ 1001 Mutex::Autolock _l(mLock); 1002 1003 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1004 if (recordThread != NULL) { 1005 return recordThread->getInputFramesLost(); 1006 } 1007 return 0; 1008} 1009 1010status_t AudioFlinger::setVoiceVolume(float value) 1011{ 1012 status_t ret = initCheck(); 1013 if (ret != NO_ERROR) { 1014 return ret; 1015 } 1016 1017 // check calling permissions 1018 if (!settingsAllowed()) { 1019 return PERMISSION_DENIED; 1020 } 1021 1022 AutoMutex lock(mHardwareLock); 1023 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1024 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1025 ret = dev->set_voice_volume(dev, value); 1026 mHardwareStatus = AUDIO_HW_IDLE; 1027 1028 return ret; 1029} 1030 1031status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1032 audio_io_handle_t output) const 1033{ 1034 status_t status; 1035 1036 Mutex::Autolock _l(mLock); 1037 1038 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1039 if (playbackThread != NULL) { 1040 return playbackThread->getRenderPosition(halFrames, dspFrames); 1041 } 1042 1043 return BAD_VALUE; 1044} 1045 1046void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1047{ 1048 1049 Mutex::Autolock _l(mLock); 1050 1051 pid_t pid = IPCThreadState::self()->getCallingPid(); 1052 if (mNotificationClients.indexOfKey(pid) < 0) { 1053 sp<NotificationClient> notificationClient = new NotificationClient(this, 1054 client, 1055 pid); 1056 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1057 1058 mNotificationClients.add(pid, notificationClient); 1059 1060 sp<IBinder> binder = client->asBinder(); 1061 binder->linkToDeath(notificationClient); 1062 1063 // the config change is always sent from playback or record threads to avoid deadlock 1064 // with AudioSystem::gLock 1065 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1066 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1067 } 1068 1069 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1070 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1071 } 1072 } 1073} 1074 1075void AudioFlinger::removeNotificationClient(pid_t pid) 1076{ 1077 Mutex::Autolock _l(mLock); 1078 1079 mNotificationClients.removeItem(pid); 1080 1081 ALOGV("%d died, releasing its sessions", pid); 1082 size_t num = mAudioSessionRefs.size(); 1083 bool removed = false; 1084 for (size_t i = 0; i< num; ) { 1085 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1086 ALOGV(" pid %d @ %d", ref->mPid, i); 1087 if (ref->mPid == pid) { 1088 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1089 mAudioSessionRefs.removeAt(i); 1090 delete ref; 1091 removed = true; 1092 num--; 1093 } else { 1094 i++; 1095 } 1096 } 1097 if (removed) { 1098 purgeStaleEffects_l(); 1099 } 1100} 1101 1102// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1103void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1104{ 1105 size_t size = mNotificationClients.size(); 1106 for (size_t i = 0; i < size; i++) { 1107 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1108 param2); 1109 } 1110} 1111 1112// removeClient_l() must be called with AudioFlinger::mLock held 1113void AudioFlinger::removeClient_l(pid_t pid) 1114{ 1115 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1116 mClients.removeItem(pid); 1117} 1118 1119// getEffectThread_l() must be called with AudioFlinger::mLock held 1120sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1121{ 1122 sp<PlaybackThread> thread; 1123 1124 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1125 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1126 ALOG_ASSERT(thread == 0); 1127 thread = mPlaybackThreads.valueAt(i); 1128 } 1129 } 1130 1131 return thread; 1132} 1133 1134// ---------------------------------------------------------------------------- 1135 1136AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1137 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1138 : Thread(false /*canCallJava*/), 1139 mType(type), 1140 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1141 // mChannelMask 1142 mChannelCount(0), 1143 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1144 mParamStatus(NO_ERROR), 1145 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1146 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1147 // mName will be set by concrete (non-virtual) subclass 1148 mDeathRecipient(new PMDeathRecipient(this)) 1149{ 1150} 1151 1152AudioFlinger::ThreadBase::~ThreadBase() 1153{ 1154 mParamCond.broadcast(); 1155 // do not lock the mutex in destructor 1156 releaseWakeLock_l(); 1157 if (mPowerManager != 0) { 1158 sp<IBinder> binder = mPowerManager->asBinder(); 1159 binder->unlinkToDeath(mDeathRecipient); 1160 } 1161} 1162 1163void AudioFlinger::ThreadBase::exit() 1164{ 1165 ALOGV("ThreadBase::exit"); 1166 { 1167 // This lock prevents the following race in thread (uniprocessor for illustration): 1168 // if (!exitPending()) { 1169 // // context switch from here to exit() 1170 // // exit() calls requestExit(), what exitPending() observes 1171 // // exit() calls signal(), which is dropped since no waiters 1172 // // context switch back from exit() to here 1173 // mWaitWorkCV.wait(...); 1174 // // now thread is hung 1175 // } 1176 AutoMutex lock(mLock); 1177 requestExit(); 1178 mWaitWorkCV.signal(); 1179 } 1180 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1181 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1182 requestExitAndWait(); 1183} 1184 1185status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1186{ 1187 status_t status; 1188 1189 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1190 Mutex::Autolock _l(mLock); 1191 1192 mNewParameters.add(keyValuePairs); 1193 mWaitWorkCV.signal(); 1194 // wait condition with timeout in case the thread loop has exited 1195 // before the request could be processed 1196 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1197 status = mParamStatus; 1198 mWaitWorkCV.signal(); 1199 } else { 1200 status = TIMED_OUT; 1201 } 1202 return status; 1203} 1204 1205void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1206{ 1207 Mutex::Autolock _l(mLock); 1208 sendConfigEvent_l(event, param); 1209} 1210 1211// sendConfigEvent_l() must be called with ThreadBase::mLock held 1212void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1213{ 1214 ConfigEvent configEvent; 1215 configEvent.mEvent = event; 1216 configEvent.mParam = param; 1217 mConfigEvents.add(configEvent); 1218 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1219 mWaitWorkCV.signal(); 1220} 1221 1222void AudioFlinger::ThreadBase::processConfigEvents() 1223{ 1224 mLock.lock(); 1225 while (!mConfigEvents.isEmpty()) { 1226 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1227 ConfigEvent configEvent = mConfigEvents[0]; 1228 mConfigEvents.removeAt(0); 1229 // release mLock before locking AudioFlinger mLock: lock order is always 1230 // AudioFlinger then ThreadBase to avoid cross deadlock 1231 mLock.unlock(); 1232 mAudioFlinger->mLock.lock(); 1233 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1234 mAudioFlinger->mLock.unlock(); 1235 mLock.lock(); 1236 } 1237 mLock.unlock(); 1238} 1239 1240void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1241{ 1242 const size_t SIZE = 256; 1243 char buffer[SIZE]; 1244 String8 result; 1245 1246 bool locked = tryLock(mLock); 1247 if (!locked) { 1248 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1249 write(fd, buffer, strlen(buffer)); 1250 } 1251 1252 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1253 result.append(buffer); 1254 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1255 result.append(buffer); 1256 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1257 result.append(buffer); 1258 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1259 result.append(buffer); 1260 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1261 result.append(buffer); 1262 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1263 result.append(buffer); 1264 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1265 result.append(buffer); 1266 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1267 result.append(buffer); 1268 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1269 result.append(buffer); 1270 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1271 result.append(buffer); 1272 1273 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1274 result.append(buffer); 1275 result.append(" Index Command"); 1276 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1277 snprintf(buffer, SIZE, "\n %02d ", i); 1278 result.append(buffer); 1279 result.append(mNewParameters[i]); 1280 } 1281 1282 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1283 result.append(buffer); 1284 snprintf(buffer, SIZE, " Index event param\n"); 1285 result.append(buffer); 1286 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1287 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1288 result.append(buffer); 1289 } 1290 result.append("\n"); 1291 1292 write(fd, result.string(), result.size()); 1293 1294 if (locked) { 1295 mLock.unlock(); 1296 } 1297} 1298 1299void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1300{ 1301 const size_t SIZE = 256; 1302 char buffer[SIZE]; 1303 String8 result; 1304 1305 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1306 write(fd, buffer, strlen(buffer)); 1307 1308 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1309 sp<EffectChain> chain = mEffectChains[i]; 1310 if (chain != 0) { 1311 chain->dump(fd, args); 1312 } 1313 } 1314} 1315 1316void AudioFlinger::ThreadBase::acquireWakeLock() 1317{ 1318 Mutex::Autolock _l(mLock); 1319 acquireWakeLock_l(); 1320} 1321 1322void AudioFlinger::ThreadBase::acquireWakeLock_l() 1323{ 1324 if (mPowerManager == 0) { 1325 // use checkService() to avoid blocking if power service is not up yet 1326 sp<IBinder> binder = 1327 defaultServiceManager()->checkService(String16("power")); 1328 if (binder == 0) { 1329 ALOGW("Thread %s cannot connect to the power manager service", mName); 1330 } else { 1331 mPowerManager = interface_cast<IPowerManager>(binder); 1332 binder->linkToDeath(mDeathRecipient); 1333 } 1334 } 1335 if (mPowerManager != 0) { 1336 sp<IBinder> binder = new BBinder(); 1337 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1338 binder, 1339 String16(mName)); 1340 if (status == NO_ERROR) { 1341 mWakeLockToken = binder; 1342 } 1343 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1344 } 1345} 1346 1347void AudioFlinger::ThreadBase::releaseWakeLock() 1348{ 1349 Mutex::Autolock _l(mLock); 1350 releaseWakeLock_l(); 1351} 1352 1353void AudioFlinger::ThreadBase::releaseWakeLock_l() 1354{ 1355 if (mWakeLockToken != 0) { 1356 ALOGV("releaseWakeLock_l() %s", mName); 1357 if (mPowerManager != 0) { 1358 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1359 } 1360 mWakeLockToken.clear(); 1361 } 1362} 1363 1364void AudioFlinger::ThreadBase::clearPowerManager() 1365{ 1366 Mutex::Autolock _l(mLock); 1367 releaseWakeLock_l(); 1368 mPowerManager.clear(); 1369} 1370 1371void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1372{ 1373 sp<ThreadBase> thread = mThread.promote(); 1374 if (thread != 0) { 1375 thread->clearPowerManager(); 1376 } 1377 ALOGW("power manager service died !!!"); 1378} 1379 1380void AudioFlinger::ThreadBase::setEffectSuspended( 1381 const effect_uuid_t *type, bool suspend, int sessionId) 1382{ 1383 Mutex::Autolock _l(mLock); 1384 setEffectSuspended_l(type, suspend, sessionId); 1385} 1386 1387void AudioFlinger::ThreadBase::setEffectSuspended_l( 1388 const effect_uuid_t *type, bool suspend, int sessionId) 1389{ 1390 sp<EffectChain> chain = getEffectChain_l(sessionId); 1391 if (chain != 0) { 1392 if (type != NULL) { 1393 chain->setEffectSuspended_l(type, suspend); 1394 } else { 1395 chain->setEffectSuspendedAll_l(suspend); 1396 } 1397 } 1398 1399 updateSuspendedSessions_l(type, suspend, sessionId); 1400} 1401 1402void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1403{ 1404 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1405 if (index < 0) { 1406 return; 1407 } 1408 1409 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1410 mSuspendedSessions.valueAt(index); 1411 1412 for (size_t i = 0; i < sessionEffects.size(); i++) { 1413 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1414 for (int j = 0; j < desc->mRefCount; j++) { 1415 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1416 chain->setEffectSuspendedAll_l(true); 1417 } else { 1418 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1419 desc->mType.timeLow); 1420 chain->setEffectSuspended_l(&desc->mType, true); 1421 } 1422 } 1423 } 1424} 1425 1426void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1427 bool suspend, 1428 int sessionId) 1429{ 1430 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1431 1432 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1433 1434 if (suspend) { 1435 if (index >= 0) { 1436 sessionEffects = mSuspendedSessions.valueAt(index); 1437 } else { 1438 mSuspendedSessions.add(sessionId, sessionEffects); 1439 } 1440 } else { 1441 if (index < 0) { 1442 return; 1443 } 1444 sessionEffects = mSuspendedSessions.valueAt(index); 1445 } 1446 1447 1448 int key = EffectChain::kKeyForSuspendAll; 1449 if (type != NULL) { 1450 key = type->timeLow; 1451 } 1452 index = sessionEffects.indexOfKey(key); 1453 1454 sp<SuspendedSessionDesc> desc; 1455 if (suspend) { 1456 if (index >= 0) { 1457 desc = sessionEffects.valueAt(index); 1458 } else { 1459 desc = new SuspendedSessionDesc(); 1460 if (type != NULL) { 1461 desc->mType = *type; 1462 } 1463 sessionEffects.add(key, desc); 1464 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1465 } 1466 desc->mRefCount++; 1467 } else { 1468 if (index < 0) { 1469 return; 1470 } 1471 desc = sessionEffects.valueAt(index); 1472 if (--desc->mRefCount == 0) { 1473 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1474 sessionEffects.removeItemsAt(index); 1475 if (sessionEffects.isEmpty()) { 1476 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1477 sessionId); 1478 mSuspendedSessions.removeItem(sessionId); 1479 } 1480 } 1481 } 1482 if (!sessionEffects.isEmpty()) { 1483 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1484 } 1485} 1486 1487void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1488 bool enabled, 1489 int sessionId) 1490{ 1491 Mutex::Autolock _l(mLock); 1492 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1493} 1494 1495void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1496 bool enabled, 1497 int sessionId) 1498{ 1499 if (mType != RECORD) { 1500 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1501 // another session. This gives the priority to well behaved effect control panels 1502 // and applications not using global effects. 1503 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1504 // global effects 1505 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1506 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1507 } 1508 } 1509 1510 sp<EffectChain> chain = getEffectChain_l(sessionId); 1511 if (chain != 0) { 1512 chain->checkSuspendOnEffectEnabled(effect, enabled); 1513 } 1514} 1515 1516// ---------------------------------------------------------------------------- 1517 1518AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1519 AudioStreamOut* output, 1520 audio_io_handle_t id, 1521 audio_devices_t device, 1522 type_t type) 1523 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1524 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1525 // mStreamTypes[] initialized in constructor body 1526 mOutput(output), 1527 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1528 mMixerStatus(MIXER_IDLE), 1529 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1530 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1531 mScreenState(gScreenState), 1532 // index 0 is reserved for normal mixer's submix 1533 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1534{ 1535 snprintf(mName, kNameLength, "AudioOut_%X", id); 1536 1537 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1538 // it would be safer to explicitly pass initial masterVolume/masterMute as 1539 // parameter. 1540 // 1541 // If the HAL we are using has support for master volume or master mute, 1542 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1543 // and the mute set to false). 1544 mMasterVolume = audioFlinger->masterVolume_l(); 1545 mMasterMute = audioFlinger->masterMute_l(); 1546 if (mOutput && mOutput->audioHwDev) { 1547 if (mOutput->audioHwDev->canSetMasterVolume()) { 1548 mMasterVolume = 1.0; 1549 } 1550 1551 if (mOutput->audioHwDev->canSetMasterMute()) { 1552 mMasterMute = false; 1553 } 1554 } 1555 1556 readOutputParameters(); 1557 1558 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1559 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1560 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1561 stream = (audio_stream_type_t) (stream + 1)) { 1562 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1563 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1564 } 1565 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1566 // because mAudioFlinger doesn't have one to copy from 1567} 1568 1569AudioFlinger::PlaybackThread::~PlaybackThread() 1570{ 1571 delete [] mMixBuffer; 1572} 1573 1574void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1575{ 1576 dumpInternals(fd, args); 1577 dumpTracks(fd, args); 1578 dumpEffectChains(fd, args); 1579} 1580 1581void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1582{ 1583 const size_t SIZE = 256; 1584 char buffer[SIZE]; 1585 String8 result; 1586 1587 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1588 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1589 const stream_type_t *st = &mStreamTypes[i]; 1590 if (i > 0) { 1591 result.appendFormat(", "); 1592 } 1593 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1594 if (st->mute) { 1595 result.append("M"); 1596 } 1597 } 1598 result.append("\n"); 1599 write(fd, result.string(), result.length()); 1600 result.clear(); 1601 1602 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1603 result.append(buffer); 1604 Track::appendDumpHeader(result); 1605 for (size_t i = 0; i < mTracks.size(); ++i) { 1606 sp<Track> track = mTracks[i]; 1607 if (track != 0) { 1608 track->dump(buffer, SIZE); 1609 result.append(buffer); 1610 } 1611 } 1612 1613 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1614 result.append(buffer); 1615 Track::appendDumpHeader(result); 1616 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1617 sp<Track> track = mActiveTracks[i].promote(); 1618 if (track != 0) { 1619 track->dump(buffer, SIZE); 1620 result.append(buffer); 1621 } 1622 } 1623 write(fd, result.string(), result.size()); 1624 1625 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1626 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1627 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1628 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1629} 1630 1631void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1632{ 1633 const size_t SIZE = 256; 1634 char buffer[SIZE]; 1635 String8 result; 1636 1637 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1638 result.append(buffer); 1639 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1640 result.append(buffer); 1641 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1642 result.append(buffer); 1643 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1644 result.append(buffer); 1645 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1646 result.append(buffer); 1647 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1648 result.append(buffer); 1649 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1650 result.append(buffer); 1651 write(fd, result.string(), result.size()); 1652 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1653 1654 dumpBase(fd, args); 1655} 1656 1657// Thread virtuals 1658status_t AudioFlinger::PlaybackThread::readyToRun() 1659{ 1660 status_t status = initCheck(); 1661 if (status == NO_ERROR) { 1662 ALOGI("AudioFlinger's thread %p ready to run", this); 1663 } else { 1664 ALOGE("No working audio driver found."); 1665 } 1666 return status; 1667} 1668 1669void AudioFlinger::PlaybackThread::onFirstRef() 1670{ 1671 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1672} 1673 1674// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1675sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1676 const sp<AudioFlinger::Client>& client, 1677 audio_stream_type_t streamType, 1678 uint32_t sampleRate, 1679 audio_format_t format, 1680 audio_channel_mask_t channelMask, 1681 int frameCount, 1682 const sp<IMemory>& sharedBuffer, 1683 int sessionId, 1684 IAudioFlinger::track_flags_t flags, 1685 pid_t tid, 1686 status_t *status) 1687{ 1688 sp<Track> track; 1689 status_t lStatus; 1690 1691 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1692 1693 // client expresses a preference for FAST, but we get the final say 1694 if (flags & IAudioFlinger::TRACK_FAST) { 1695 if ( 1696 // not timed 1697 (!isTimed) && 1698 // either of these use cases: 1699 ( 1700 // use case 1: shared buffer with any frame count 1701 ( 1702 (sharedBuffer != 0) 1703 ) || 1704 // use case 2: callback handler and frame count is default or at least as large as HAL 1705 ( 1706 (tid != -1) && 1707 ((frameCount == 0) || 1708 (frameCount >= (int) (mFrameCount * kFastTrackMultiplier))) 1709 ) 1710 ) && 1711 // PCM data 1712 audio_is_linear_pcm(format) && 1713 // mono or stereo 1714 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1715 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1716#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1717 // hardware sample rate 1718 (sampleRate == mSampleRate) && 1719#endif 1720 // normal mixer has an associated fast mixer 1721 hasFastMixer() && 1722 // there are sufficient fast track slots available 1723 (mFastTrackAvailMask != 0) 1724 // FIXME test that MixerThread for this fast track has a capable output HAL 1725 // FIXME add a permission test also? 1726 ) { 1727 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1728 if (frameCount == 0) { 1729 frameCount = mFrameCount * kFastTrackMultiplier; 1730 } 1731 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1732 frameCount, mFrameCount); 1733 } else { 1734 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1735 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1736 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1737 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1738 audio_is_linear_pcm(format), 1739 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1740 flags &= ~IAudioFlinger::TRACK_FAST; 1741 // For compatibility with AudioTrack calculation, buffer depth is forced 1742 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1743 // This is probably too conservative, but legacy application code may depend on it. 1744 // If you change this calculation, also review the start threshold which is related. 1745 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1746 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1747 if (minBufCount < 2) { 1748 minBufCount = 2; 1749 } 1750 int minFrameCount = mNormalFrameCount * minBufCount; 1751 if (frameCount < minFrameCount) { 1752 frameCount = minFrameCount; 1753 } 1754 } 1755 } 1756 1757 if (mType == DIRECT) { 1758 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1759 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1760 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1761 "for output %p with format %d", 1762 sampleRate, format, channelMask, mOutput, mFormat); 1763 lStatus = BAD_VALUE; 1764 goto Exit; 1765 } 1766 } 1767 } else { 1768 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1769 if (sampleRate > mSampleRate*2) { 1770 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1771 lStatus = BAD_VALUE; 1772 goto Exit; 1773 } 1774 } 1775 1776 lStatus = initCheck(); 1777 if (lStatus != NO_ERROR) { 1778 ALOGE("Audio driver not initialized."); 1779 goto Exit; 1780 } 1781 1782 { // scope for mLock 1783 Mutex::Autolock _l(mLock); 1784 1785 // all tracks in same audio session must share the same routing strategy otherwise 1786 // conflicts will happen when tracks are moved from one output to another by audio policy 1787 // manager 1788 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1789 for (size_t i = 0; i < mTracks.size(); ++i) { 1790 sp<Track> t = mTracks[i]; 1791 if (t != 0 && !t->isOutputTrack()) { 1792 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1793 if (sessionId == t->sessionId() && strategy != actual) { 1794 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1795 strategy, actual); 1796 lStatus = BAD_VALUE; 1797 goto Exit; 1798 } 1799 } 1800 } 1801 1802 if (!isTimed) { 1803 track = new Track(this, client, streamType, sampleRate, format, 1804 channelMask, frameCount, sharedBuffer, sessionId, flags); 1805 } else { 1806 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1807 channelMask, frameCount, sharedBuffer, sessionId); 1808 } 1809 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1810 lStatus = NO_MEMORY; 1811 goto Exit; 1812 } 1813 mTracks.add(track); 1814 1815 sp<EffectChain> chain = getEffectChain_l(sessionId); 1816 if (chain != 0) { 1817 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1818 track->setMainBuffer(chain->inBuffer()); 1819 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1820 chain->incTrackCnt(); 1821 } 1822 } 1823 1824 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1825 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1826 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1827 // so ask activity manager to do this on our behalf 1828 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1829 if (err != 0) { 1830 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1831 kPriorityAudioApp, callingPid, tid, err); 1832 } 1833 } 1834 1835 lStatus = NO_ERROR; 1836 1837Exit: 1838 if (status) { 1839 *status = lStatus; 1840 } 1841 return track; 1842} 1843 1844uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1845{ 1846 if (mFastMixer != NULL) { 1847 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1848 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1849 } 1850 return latency; 1851} 1852 1853uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1854{ 1855 return latency; 1856} 1857 1858uint32_t AudioFlinger::PlaybackThread::latency() const 1859{ 1860 Mutex::Autolock _l(mLock); 1861 return latency_l(); 1862} 1863uint32_t AudioFlinger::PlaybackThread::latency_l() const 1864{ 1865 if (initCheck() == NO_ERROR) { 1866 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1867 } else { 1868 return 0; 1869 } 1870} 1871 1872void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1873{ 1874 Mutex::Autolock _l(mLock); 1875 // Don't apply master volume in SW if our HAL can do it for us. 1876 if (mOutput && mOutput->audioHwDev && 1877 mOutput->audioHwDev->canSetMasterVolume()) { 1878 mMasterVolume = 1.0; 1879 } else { 1880 mMasterVolume = value; 1881 } 1882} 1883 1884void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1885{ 1886 Mutex::Autolock _l(mLock); 1887 // Don't apply master mute in SW if our HAL can do it for us. 1888 if (mOutput && mOutput->audioHwDev && 1889 mOutput->audioHwDev->canSetMasterMute()) { 1890 mMasterMute = false; 1891 } else { 1892 mMasterMute = muted; 1893 } 1894} 1895 1896void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1897{ 1898 Mutex::Autolock _l(mLock); 1899 mStreamTypes[stream].volume = value; 1900} 1901 1902void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1903{ 1904 Mutex::Autolock _l(mLock); 1905 mStreamTypes[stream].mute = muted; 1906} 1907 1908float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1909{ 1910 Mutex::Autolock _l(mLock); 1911 return mStreamTypes[stream].volume; 1912} 1913 1914// addTrack_l() must be called with ThreadBase::mLock held 1915status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1916{ 1917 status_t status = ALREADY_EXISTS; 1918 1919 // set retry count for buffer fill 1920 track->mRetryCount = kMaxTrackStartupRetries; 1921 if (mActiveTracks.indexOf(track) < 0) { 1922 // the track is newly added, make sure it fills up all its 1923 // buffers before playing. This is to ensure the client will 1924 // effectively get the latency it requested. 1925 track->mFillingUpStatus = Track::FS_FILLING; 1926 track->mResetDone = false; 1927 track->mPresentationCompleteFrames = 0; 1928 mActiveTracks.add(track); 1929 if (track->mainBuffer() != mMixBuffer) { 1930 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1931 if (chain != 0) { 1932 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1933 chain->incActiveTrackCnt(); 1934 } 1935 } 1936 1937 status = NO_ERROR; 1938 } 1939 1940 ALOGV("mWaitWorkCV.broadcast"); 1941 mWaitWorkCV.broadcast(); 1942 1943 return status; 1944} 1945 1946// destroyTrack_l() must be called with ThreadBase::mLock held 1947void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1948{ 1949 track->mState = TrackBase::TERMINATED; 1950 // active tracks are removed by threadLoop() 1951 if (mActiveTracks.indexOf(track) < 0) { 1952 removeTrack_l(track); 1953 } 1954} 1955 1956void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1957{ 1958 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1959 mTracks.remove(track); 1960 deleteTrackName_l(track->name()); 1961 // redundant as track is about to be destroyed, for dumpsys only 1962 track->mName = -1; 1963 if (track->isFastTrack()) { 1964 int index = track->mFastIndex; 1965 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1966 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1967 mFastTrackAvailMask |= 1 << index; 1968 // redundant as track is about to be destroyed, for dumpsys only 1969 track->mFastIndex = -1; 1970 } 1971 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1972 if (chain != 0) { 1973 chain->decTrackCnt(); 1974 } 1975} 1976 1977String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1978{ 1979 String8 out_s8 = String8(""); 1980 char *s; 1981 1982 Mutex::Autolock _l(mLock); 1983 if (initCheck() != NO_ERROR) { 1984 return out_s8; 1985 } 1986 1987 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1988 out_s8 = String8(s); 1989 free(s); 1990 return out_s8; 1991} 1992 1993// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1994void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1995 AudioSystem::OutputDescriptor desc; 1996 void *param2 = NULL; 1997 1998 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1999 2000 switch (event) { 2001 case AudioSystem::OUTPUT_OPENED: 2002 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2003 desc.channels = mChannelMask; 2004 desc.samplingRate = mSampleRate; 2005 desc.format = mFormat; 2006 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 2007 desc.latency = latency(); 2008 param2 = &desc; 2009 break; 2010 2011 case AudioSystem::STREAM_CONFIG_CHANGED: 2012 param2 = ¶m; 2013 case AudioSystem::OUTPUT_CLOSED: 2014 default: 2015 break; 2016 } 2017 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2018} 2019 2020void AudioFlinger::PlaybackThread::readOutputParameters() 2021{ 2022 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2023 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2024 mChannelCount = (uint16_t)popcount(mChannelMask); 2025 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2026 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2027 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2028 if (mFrameCount & 15) { 2029 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2030 mFrameCount); 2031 } 2032 2033 // Calculate size of normal mix buffer relative to the HAL output buffer size 2034 double multiplier = 1.0; 2035 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2036 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2037 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2038 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2039 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2040 maxNormalFrameCount = maxNormalFrameCount & ~15; 2041 if (maxNormalFrameCount < minNormalFrameCount) { 2042 maxNormalFrameCount = minNormalFrameCount; 2043 } 2044 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2045 if (multiplier <= 1.0) { 2046 multiplier = 1.0; 2047 } else if (multiplier <= 2.0) { 2048 if (2 * mFrameCount <= maxNormalFrameCount) { 2049 multiplier = 2.0; 2050 } else { 2051 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2052 } 2053 } else { 2054 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2055 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2056 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2057 // FIXME this rounding up should not be done if no HAL SRC 2058 uint32_t truncMult = (uint32_t) multiplier; 2059 if ((truncMult & 1)) { 2060 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2061 ++truncMult; 2062 } 2063 } 2064 multiplier = (double) truncMult; 2065 } 2066 } 2067 mNormalFrameCount = multiplier * mFrameCount; 2068 // round up to nearest 16 frames to satisfy AudioMixer 2069 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2070 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2071 2072 delete[] mMixBuffer; 2073 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2074 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2075 2076 // force reconfiguration of effect chains and engines to take new buffer size and audio 2077 // parameters into account 2078 // Note that mLock is not held when readOutputParameters() is called from the constructor 2079 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2080 // matter. 2081 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2082 Vector< sp<EffectChain> > effectChains = mEffectChains; 2083 for (size_t i = 0; i < effectChains.size(); i ++) { 2084 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2085 } 2086} 2087 2088 2089status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2090{ 2091 if (halFrames == NULL || dspFrames == NULL) { 2092 return BAD_VALUE; 2093 } 2094 Mutex::Autolock _l(mLock); 2095 if (initCheck() != NO_ERROR) { 2096 return INVALID_OPERATION; 2097 } 2098 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2099 2100 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2101} 2102 2103uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2104{ 2105 Mutex::Autolock _l(mLock); 2106 uint32_t result = 0; 2107 if (getEffectChain_l(sessionId) != 0) { 2108 result = EFFECT_SESSION; 2109 } 2110 2111 for (size_t i = 0; i < mTracks.size(); ++i) { 2112 sp<Track> track = mTracks[i]; 2113 if (sessionId == track->sessionId() && 2114 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2115 result |= TRACK_SESSION; 2116 break; 2117 } 2118 } 2119 2120 return result; 2121} 2122 2123uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2124{ 2125 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2126 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2127 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2128 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2129 } 2130 for (size_t i = 0; i < mTracks.size(); i++) { 2131 sp<Track> track = mTracks[i]; 2132 if (sessionId == track->sessionId() && 2133 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2134 return AudioSystem::getStrategyForStream(track->streamType()); 2135 } 2136 } 2137 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2138} 2139 2140 2141AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2142{ 2143 Mutex::Autolock _l(mLock); 2144 return mOutput; 2145} 2146 2147AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2148{ 2149 Mutex::Autolock _l(mLock); 2150 AudioStreamOut *output = mOutput; 2151 mOutput = NULL; 2152 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2153 // must push a NULL and wait for ack 2154 mOutputSink.clear(); 2155 mPipeSink.clear(); 2156 mNormalSink.clear(); 2157 return output; 2158} 2159 2160// this method must always be called either with ThreadBase mLock held or inside the thread loop 2161audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2162{ 2163 if (mOutput == NULL) { 2164 return NULL; 2165 } 2166 return &mOutput->stream->common; 2167} 2168 2169uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2170{ 2171 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2172} 2173 2174status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2175{ 2176 if (!isValidSyncEvent(event)) { 2177 return BAD_VALUE; 2178 } 2179 2180 Mutex::Autolock _l(mLock); 2181 2182 for (size_t i = 0; i < mTracks.size(); ++i) { 2183 sp<Track> track = mTracks[i]; 2184 if (event->triggerSession() == track->sessionId()) { 2185 (void) track->setSyncEvent(event); 2186 return NO_ERROR; 2187 } 2188 } 2189 2190 return NAME_NOT_FOUND; 2191} 2192 2193bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2194{ 2195 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2196} 2197 2198void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2199{ 2200 size_t count = tracksToRemove.size(); 2201 if (CC_UNLIKELY(count)) { 2202 for (size_t i = 0 ; i < count ; i++) { 2203 const sp<Track>& track = tracksToRemove.itemAt(i); 2204 if ((track->sharedBuffer() != 0) && 2205 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2206 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2207 } 2208 } 2209 } 2210 2211} 2212 2213// ---------------------------------------------------------------------------- 2214 2215AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2216 audio_io_handle_t id, audio_devices_t device, type_t type) 2217 : PlaybackThread(audioFlinger, output, id, device, type), 2218 // mAudioMixer below 2219 // mFastMixer below 2220 mFastMixerFutex(0) 2221 // mOutputSink below 2222 // mPipeSink below 2223 // mNormalSink below 2224{ 2225 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2226 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2227 "mFrameCount=%d, mNormalFrameCount=%d", 2228 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2229 mNormalFrameCount); 2230 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2231 2232 // FIXME - Current mixer implementation only supports stereo output 2233 if (mChannelCount != FCC_2) { 2234 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2235 } 2236 2237 // create an NBAIO sink for the HAL output stream, and negotiate 2238 mOutputSink = new AudioStreamOutSink(output->stream); 2239 size_t numCounterOffers = 0; 2240 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2241 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2242 ALOG_ASSERT(index == 0); 2243 2244 // initialize fast mixer depending on configuration 2245 bool initFastMixer; 2246 switch (kUseFastMixer) { 2247 case FastMixer_Never: 2248 initFastMixer = false; 2249 break; 2250 case FastMixer_Always: 2251 initFastMixer = true; 2252 break; 2253 case FastMixer_Static: 2254 case FastMixer_Dynamic: 2255 initFastMixer = mFrameCount < mNormalFrameCount; 2256 break; 2257 } 2258 if (initFastMixer) { 2259 2260 // create a MonoPipe to connect our submix to FastMixer 2261 NBAIO_Format format = mOutputSink->format(); 2262 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2263 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2264 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2265 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2266 const NBAIO_Format offers[1] = {format}; 2267 size_t numCounterOffers = 0; 2268 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2269 ALOG_ASSERT(index == 0); 2270 monoPipe->setAvgFrames((mScreenState & 1) ? 2271 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2272 mPipeSink = monoPipe; 2273 2274#ifdef TEE_SINK_FRAMES 2275 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2276 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2277 numCounterOffers = 0; 2278 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2279 ALOG_ASSERT(index == 0); 2280 mTeeSink = teeSink; 2281 PipeReader *teeSource = new PipeReader(*teeSink); 2282 numCounterOffers = 0; 2283 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2284 ALOG_ASSERT(index == 0); 2285 mTeeSource = teeSource; 2286#endif 2287 2288 // create fast mixer and configure it initially with just one fast track for our submix 2289 mFastMixer = new FastMixer(); 2290 FastMixerStateQueue *sq = mFastMixer->sq(); 2291#ifdef STATE_QUEUE_DUMP 2292 sq->setObserverDump(&mStateQueueObserverDump); 2293 sq->setMutatorDump(&mStateQueueMutatorDump); 2294#endif 2295 FastMixerState *state = sq->begin(); 2296 FastTrack *fastTrack = &state->mFastTracks[0]; 2297 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2298 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2299 fastTrack->mVolumeProvider = NULL; 2300 fastTrack->mGeneration++; 2301 state->mFastTracksGen++; 2302 state->mTrackMask = 1; 2303 // fast mixer will use the HAL output sink 2304 state->mOutputSink = mOutputSink.get(); 2305 state->mOutputSinkGen++; 2306 state->mFrameCount = mFrameCount; 2307 state->mCommand = FastMixerState::COLD_IDLE; 2308 // already done in constructor initialization list 2309 //mFastMixerFutex = 0; 2310 state->mColdFutexAddr = &mFastMixerFutex; 2311 state->mColdGen++; 2312 state->mDumpState = &mFastMixerDumpState; 2313 state->mTeeSink = mTeeSink.get(); 2314 sq->end(); 2315 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2316 2317 // start the fast mixer 2318 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2319 pid_t tid = mFastMixer->getTid(); 2320 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2321 if (err != 0) { 2322 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2323 kPriorityFastMixer, getpid_cached, tid, err); 2324 } 2325 2326#ifdef AUDIO_WATCHDOG 2327 // create and start the watchdog 2328 mAudioWatchdog = new AudioWatchdog(); 2329 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2330 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2331 tid = mAudioWatchdog->getTid(); 2332 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2333 if (err != 0) { 2334 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2335 kPriorityFastMixer, getpid_cached, tid, err); 2336 } 2337#endif 2338 2339 } else { 2340 mFastMixer = NULL; 2341 } 2342 2343 switch (kUseFastMixer) { 2344 case FastMixer_Never: 2345 case FastMixer_Dynamic: 2346 mNormalSink = mOutputSink; 2347 break; 2348 case FastMixer_Always: 2349 mNormalSink = mPipeSink; 2350 break; 2351 case FastMixer_Static: 2352 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2353 break; 2354 } 2355} 2356 2357AudioFlinger::MixerThread::~MixerThread() 2358{ 2359 if (mFastMixer != NULL) { 2360 FastMixerStateQueue *sq = mFastMixer->sq(); 2361 FastMixerState *state = sq->begin(); 2362 if (state->mCommand == FastMixerState::COLD_IDLE) { 2363 int32_t old = android_atomic_inc(&mFastMixerFutex); 2364 if (old == -1) { 2365 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2366 } 2367 } 2368 state->mCommand = FastMixerState::EXIT; 2369 sq->end(); 2370 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2371 mFastMixer->join(); 2372 // Though the fast mixer thread has exited, it's state queue is still valid. 2373 // We'll use that extract the final state which contains one remaining fast track 2374 // corresponding to our sub-mix. 2375 state = sq->begin(); 2376 ALOG_ASSERT(state->mTrackMask == 1); 2377 FastTrack *fastTrack = &state->mFastTracks[0]; 2378 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2379 delete fastTrack->mBufferProvider; 2380 sq->end(false /*didModify*/); 2381 delete mFastMixer; 2382 if (mAudioWatchdog != 0) { 2383 mAudioWatchdog->requestExit(); 2384 mAudioWatchdog->requestExitAndWait(); 2385 mAudioWatchdog.clear(); 2386 } 2387 } 2388 delete mAudioMixer; 2389} 2390 2391class CpuStats { 2392public: 2393 CpuStats(); 2394 void sample(const String8 &title); 2395#ifdef DEBUG_CPU_USAGE 2396private: 2397 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2398 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2399 2400 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2401 2402 int mCpuNum; // thread's current CPU number 2403 int mCpukHz; // frequency of thread's current CPU in kHz 2404#endif 2405}; 2406 2407CpuStats::CpuStats() 2408#ifdef DEBUG_CPU_USAGE 2409 : mCpuNum(-1), mCpukHz(-1) 2410#endif 2411{ 2412} 2413 2414void CpuStats::sample(const String8 &title) { 2415#ifdef DEBUG_CPU_USAGE 2416 // get current thread's delta CPU time in wall clock ns 2417 double wcNs; 2418 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2419 2420 // record sample for wall clock statistics 2421 if (valid) { 2422 mWcStats.sample(wcNs); 2423 } 2424 2425 // get the current CPU number 2426 int cpuNum = sched_getcpu(); 2427 2428 // get the current CPU frequency in kHz 2429 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2430 2431 // check if either CPU number or frequency changed 2432 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2433 mCpuNum = cpuNum; 2434 mCpukHz = cpukHz; 2435 // ignore sample for purposes of cycles 2436 valid = false; 2437 } 2438 2439 // if no change in CPU number or frequency, then record sample for cycle statistics 2440 if (valid && mCpukHz > 0) { 2441 double cycles = wcNs * cpukHz * 0.000001; 2442 mHzStats.sample(cycles); 2443 } 2444 2445 unsigned n = mWcStats.n(); 2446 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2447 if ((n & 127) == 1) { 2448 long long elapsed = mCpuUsage.elapsed(); 2449 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2450 double perLoop = elapsed / (double) n; 2451 double perLoop100 = perLoop * 0.01; 2452 double perLoop1k = perLoop * 0.001; 2453 double mean = mWcStats.mean(); 2454 double stddev = mWcStats.stddev(); 2455 double minimum = mWcStats.minimum(); 2456 double maximum = mWcStats.maximum(); 2457 double meanCycles = mHzStats.mean(); 2458 double stddevCycles = mHzStats.stddev(); 2459 double minCycles = mHzStats.minimum(); 2460 double maxCycles = mHzStats.maximum(); 2461 mCpuUsage.resetElapsed(); 2462 mWcStats.reset(); 2463 mHzStats.reset(); 2464 ALOGD("CPU usage for %s over past %.1f secs\n" 2465 " (%u mixer loops at %.1f mean ms per loop):\n" 2466 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2467 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2468 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2469 title.string(), 2470 elapsed * .000000001, n, perLoop * .000001, 2471 mean * .001, 2472 stddev * .001, 2473 minimum * .001, 2474 maximum * .001, 2475 mean / perLoop100, 2476 stddev / perLoop100, 2477 minimum / perLoop100, 2478 maximum / perLoop100, 2479 meanCycles / perLoop1k, 2480 stddevCycles / perLoop1k, 2481 minCycles / perLoop1k, 2482 maxCycles / perLoop1k); 2483 2484 } 2485 } 2486#endif 2487}; 2488 2489void AudioFlinger::PlaybackThread::checkSilentMode_l() 2490{ 2491 if (!mMasterMute) { 2492 char value[PROPERTY_VALUE_MAX]; 2493 if (property_get("ro.audio.silent", value, "0") > 0) { 2494 char *endptr; 2495 unsigned long ul = strtoul(value, &endptr, 0); 2496 if (*endptr == '\0' && ul != 0) { 2497 ALOGD("Silence is golden"); 2498 // The setprop command will not allow a property to be changed after 2499 // the first time it is set, so we don't have to worry about un-muting. 2500 setMasterMute_l(true); 2501 } 2502 } 2503 } 2504} 2505 2506bool AudioFlinger::PlaybackThread::threadLoop() 2507{ 2508 Vector< sp<Track> > tracksToRemove; 2509 2510 standbyTime = systemTime(); 2511 2512 // MIXER 2513 nsecs_t lastWarning = 0; 2514 2515 // DUPLICATING 2516 // FIXME could this be made local to while loop? 2517 writeFrames = 0; 2518 2519 cacheParameters_l(); 2520 sleepTime = idleSleepTime; 2521 2522 if (mType == MIXER) { 2523 sleepTimeShift = 0; 2524 } 2525 2526 CpuStats cpuStats; 2527 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2528 2529 acquireWakeLock(); 2530 2531 while (!exitPending()) 2532 { 2533 cpuStats.sample(myName); 2534 2535 Vector< sp<EffectChain> > effectChains; 2536 2537 processConfigEvents(); 2538 2539 { // scope for mLock 2540 2541 Mutex::Autolock _l(mLock); 2542 2543 if (checkForNewParameters_l()) { 2544 cacheParameters_l(); 2545 } 2546 2547 saveOutputTracks(); 2548 2549 // put audio hardware into standby after short delay 2550 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2551 isSuspended())) { 2552 if (!mStandby) { 2553 2554 threadLoop_standby(); 2555 2556 mStandby = true; 2557 mBytesWritten = 0; 2558 } 2559 2560 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2561 // we're about to wait, flush the binder command buffer 2562 IPCThreadState::self()->flushCommands(); 2563 2564 clearOutputTracks(); 2565 2566 if (exitPending()) break; 2567 2568 releaseWakeLock_l(); 2569 // wait until we have something to do... 2570 ALOGV("%s going to sleep", myName.string()); 2571 mWaitWorkCV.wait(mLock); 2572 ALOGV("%s waking up", myName.string()); 2573 acquireWakeLock_l(); 2574 2575 mMixerStatus = MIXER_IDLE; 2576 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2577 2578 checkSilentMode_l(); 2579 2580 standbyTime = systemTime() + standbyDelay; 2581 sleepTime = idleSleepTime; 2582 if (mType == MIXER) { 2583 sleepTimeShift = 0; 2584 } 2585 2586 continue; 2587 } 2588 } 2589 2590 // mMixerStatusIgnoringFastTracks is also updated internally 2591 mMixerStatus = prepareTracks_l(&tracksToRemove); 2592 2593 // prevent any changes in effect chain list and in each effect chain 2594 // during mixing and effect process as the audio buffers could be deleted 2595 // or modified if an effect is created or deleted 2596 lockEffectChains_l(effectChains); 2597 } 2598 2599 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2600 threadLoop_mix(); 2601 } else { 2602 threadLoop_sleepTime(); 2603 } 2604 2605 if (isSuspended()) { 2606 sleepTime = suspendSleepTimeUs(); 2607 } 2608 2609 // only process effects if we're going to write 2610 if (sleepTime == 0) { 2611 for (size_t i = 0; i < effectChains.size(); i ++) { 2612 effectChains[i]->process_l(); 2613 } 2614 } 2615 2616 // enable changes in effect chain 2617 unlockEffectChains(effectChains); 2618 2619 // sleepTime == 0 means we must write to audio hardware 2620 if (sleepTime == 0) { 2621 2622 threadLoop_write(); 2623 2624if (mType == MIXER) { 2625 // write blocked detection 2626 nsecs_t now = systemTime(); 2627 nsecs_t delta = now - mLastWriteTime; 2628 if (!mStandby && delta > maxPeriod) { 2629 mNumDelayedWrites++; 2630 if ((now - lastWarning) > kWarningThrottleNs) { 2631#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2632 ScopedTrace st(ATRACE_TAG, "underrun"); 2633#endif 2634 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2635 ns2ms(delta), mNumDelayedWrites, this); 2636 lastWarning = now; 2637 } 2638 } 2639} 2640 2641 mStandby = false; 2642 } else { 2643 usleep(sleepTime); 2644 } 2645 2646 // Finally let go of removed track(s), without the lock held 2647 // since we can't guarantee the destructors won't acquire that 2648 // same lock. This will also mutate and push a new fast mixer state. 2649 threadLoop_removeTracks(tracksToRemove); 2650 tracksToRemove.clear(); 2651 2652 // FIXME I don't understand the need for this here; 2653 // it was in the original code but maybe the 2654 // assignment in saveOutputTracks() makes this unnecessary? 2655 clearOutputTracks(); 2656 2657 // Effect chains will be actually deleted here if they were removed from 2658 // mEffectChains list during mixing or effects processing 2659 effectChains.clear(); 2660 2661 // FIXME Note that the above .clear() is no longer necessary since effectChains 2662 // is now local to this block, but will keep it for now (at least until merge done). 2663 } 2664 2665 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2666 if (mType == MIXER || mType == DIRECT) { 2667 // put output stream into standby mode 2668 if (!mStandby) { 2669 mOutput->stream->common.standby(&mOutput->stream->common); 2670 } 2671 } 2672 2673 releaseWakeLock(); 2674 2675 ALOGV("Thread %p type %d exiting", this, mType); 2676 return false; 2677} 2678 2679void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2680{ 2681 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2682} 2683 2684void AudioFlinger::MixerThread::threadLoop_write() 2685{ 2686 // FIXME we should only do one push per cycle; confirm this is true 2687 // Start the fast mixer if it's not already running 2688 if (mFastMixer != NULL) { 2689 FastMixerStateQueue *sq = mFastMixer->sq(); 2690 FastMixerState *state = sq->begin(); 2691 if (state->mCommand != FastMixerState::MIX_WRITE && 2692 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2693 if (state->mCommand == FastMixerState::COLD_IDLE) { 2694 int32_t old = android_atomic_inc(&mFastMixerFutex); 2695 if (old == -1) { 2696 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2697 } 2698 if (mAudioWatchdog != 0) { 2699 mAudioWatchdog->resume(); 2700 } 2701 } 2702 state->mCommand = FastMixerState::MIX_WRITE; 2703 sq->end(); 2704 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2705 if (kUseFastMixer == FastMixer_Dynamic) { 2706 mNormalSink = mPipeSink; 2707 } 2708 } else { 2709 sq->end(false /*didModify*/); 2710 } 2711 } 2712 PlaybackThread::threadLoop_write(); 2713} 2714 2715// shared by MIXER and DIRECT, overridden by DUPLICATING 2716void AudioFlinger::PlaybackThread::threadLoop_write() 2717{ 2718 // FIXME rewrite to reduce number of system calls 2719 mLastWriteTime = systemTime(); 2720 mInWrite = true; 2721 int bytesWritten; 2722 2723 // If an NBAIO sink is present, use it to write the normal mixer's submix 2724 if (mNormalSink != 0) { 2725#define mBitShift 2 // FIXME 2726 size_t count = mixBufferSize >> mBitShift; 2727#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2728 Tracer::traceBegin(ATRACE_TAG, "write"); 2729#endif 2730 // update the setpoint when gScreenState changes 2731 uint32_t screenState = gScreenState; 2732 if (screenState != mScreenState) { 2733 mScreenState = screenState; 2734 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2735 if (pipe != NULL) { 2736 pipe->setAvgFrames((mScreenState & 1) ? 2737 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2738 } 2739 } 2740 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2741#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2742 Tracer::traceEnd(ATRACE_TAG); 2743#endif 2744 if (framesWritten > 0) { 2745 bytesWritten = framesWritten << mBitShift; 2746 } else { 2747 bytesWritten = framesWritten; 2748 } 2749 // otherwise use the HAL / AudioStreamOut directly 2750 } else { 2751 // Direct output thread. 2752 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2753 } 2754 2755 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2756 mNumWrites++; 2757 mInWrite = false; 2758} 2759 2760void AudioFlinger::MixerThread::threadLoop_standby() 2761{ 2762 // Idle the fast mixer if it's currently running 2763 if (mFastMixer != NULL) { 2764 FastMixerStateQueue *sq = mFastMixer->sq(); 2765 FastMixerState *state = sq->begin(); 2766 if (!(state->mCommand & FastMixerState::IDLE)) { 2767 state->mCommand = FastMixerState::COLD_IDLE; 2768 state->mColdFutexAddr = &mFastMixerFutex; 2769 state->mColdGen++; 2770 mFastMixerFutex = 0; 2771 sq->end(); 2772 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2773 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2774 if (kUseFastMixer == FastMixer_Dynamic) { 2775 mNormalSink = mOutputSink; 2776 } 2777 if (mAudioWatchdog != 0) { 2778 mAudioWatchdog->pause(); 2779 } 2780 } else { 2781 sq->end(false /*didModify*/); 2782 } 2783 } 2784 PlaybackThread::threadLoop_standby(); 2785} 2786 2787// shared by MIXER and DIRECT, overridden by DUPLICATING 2788void AudioFlinger::PlaybackThread::threadLoop_standby() 2789{ 2790 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2791 mOutput->stream->common.standby(&mOutput->stream->common); 2792} 2793 2794void AudioFlinger::MixerThread::threadLoop_mix() 2795{ 2796 // obtain the presentation timestamp of the next output buffer 2797 int64_t pts; 2798 status_t status = INVALID_OPERATION; 2799 2800 if (mNormalSink != 0) { 2801 status = mNormalSink->getNextWriteTimestamp(&pts); 2802 } else { 2803 status = mOutputSink->getNextWriteTimestamp(&pts); 2804 } 2805 2806 if (status != NO_ERROR) { 2807 pts = AudioBufferProvider::kInvalidPTS; 2808 } 2809 2810 // mix buffers... 2811 mAudioMixer->process(pts); 2812 // increase sleep time progressively when application underrun condition clears. 2813 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2814 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2815 // such that we would underrun the audio HAL. 2816 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2817 sleepTimeShift--; 2818 } 2819 sleepTime = 0; 2820 standbyTime = systemTime() + standbyDelay; 2821 //TODO: delay standby when effects have a tail 2822} 2823 2824void AudioFlinger::MixerThread::threadLoop_sleepTime() 2825{ 2826 // If no tracks are ready, sleep once for the duration of an output 2827 // buffer size, then write 0s to the output 2828 if (sleepTime == 0) { 2829 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2830 sleepTime = activeSleepTime >> sleepTimeShift; 2831 if (sleepTime < kMinThreadSleepTimeUs) { 2832 sleepTime = kMinThreadSleepTimeUs; 2833 } 2834 // reduce sleep time in case of consecutive application underruns to avoid 2835 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2836 // duration we would end up writing less data than needed by the audio HAL if 2837 // the condition persists. 2838 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2839 sleepTimeShift++; 2840 } 2841 } else { 2842 sleepTime = idleSleepTime; 2843 } 2844 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2845 memset (mMixBuffer, 0, mixBufferSize); 2846 sleepTime = 0; 2847 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2848 } 2849 // TODO add standby time extension fct of effect tail 2850} 2851 2852// prepareTracks_l() must be called with ThreadBase::mLock held 2853AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2854 Vector< sp<Track> > *tracksToRemove) 2855{ 2856 2857 mixer_state mixerStatus = MIXER_IDLE; 2858 // find out which tracks need to be processed 2859 size_t count = mActiveTracks.size(); 2860 size_t mixedTracks = 0; 2861 size_t tracksWithEffect = 0; 2862 // counts only _active_ fast tracks 2863 size_t fastTracks = 0; 2864 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2865 2866 float masterVolume = mMasterVolume; 2867 bool masterMute = mMasterMute; 2868 2869 if (masterMute) { 2870 masterVolume = 0; 2871 } 2872 // Delegate master volume control to effect in output mix effect chain if needed 2873 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2874 if (chain != 0) { 2875 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2876 chain->setVolume_l(&v, &v); 2877 masterVolume = (float)((v + (1 << 23)) >> 24); 2878 chain.clear(); 2879 } 2880 2881 // prepare a new state to push 2882 FastMixerStateQueue *sq = NULL; 2883 FastMixerState *state = NULL; 2884 bool didModify = false; 2885 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2886 if (mFastMixer != NULL) { 2887 sq = mFastMixer->sq(); 2888 state = sq->begin(); 2889 } 2890 2891 for (size_t i=0 ; i<count ; i++) { 2892 sp<Track> t = mActiveTracks[i].promote(); 2893 if (t == 0) continue; 2894 2895 // this const just means the local variable doesn't change 2896 Track* const track = t.get(); 2897 2898 // process fast tracks 2899 if (track->isFastTrack()) { 2900 2901 // It's theoretically possible (though unlikely) for a fast track to be created 2902 // and then removed within the same normal mix cycle. This is not a problem, as 2903 // the track never becomes active so it's fast mixer slot is never touched. 2904 // The converse, of removing an (active) track and then creating a new track 2905 // at the identical fast mixer slot within the same normal mix cycle, 2906 // is impossible because the slot isn't marked available until the end of each cycle. 2907 int j = track->mFastIndex; 2908 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2909 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2910 FastTrack *fastTrack = &state->mFastTracks[j]; 2911 2912 // Determine whether the track is currently in underrun condition, 2913 // and whether it had a recent underrun. 2914 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2915 FastTrackUnderruns underruns = ftDump->mUnderruns; 2916 uint32_t recentFull = (underruns.mBitFields.mFull - 2917 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2918 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2919 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2920 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2921 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2922 uint32_t recentUnderruns = recentPartial + recentEmpty; 2923 track->mObservedUnderruns = underruns; 2924 // don't count underruns that occur while stopping or pausing 2925 // or stopped which can occur when flush() is called while active 2926 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2927 track->mUnderrunCount += recentUnderruns; 2928 } 2929 2930 // This is similar to the state machine for normal tracks, 2931 // with a few modifications for fast tracks. 2932 bool isActive = true; 2933 switch (track->mState) { 2934 case TrackBase::STOPPING_1: 2935 // track stays active in STOPPING_1 state until first underrun 2936 if (recentUnderruns > 0) { 2937 track->mState = TrackBase::STOPPING_2; 2938 } 2939 break; 2940 case TrackBase::PAUSING: 2941 // ramp down is not yet implemented 2942 track->setPaused(); 2943 break; 2944 case TrackBase::RESUMING: 2945 // ramp up is not yet implemented 2946 track->mState = TrackBase::ACTIVE; 2947 break; 2948 case TrackBase::ACTIVE: 2949 if (recentFull > 0 || recentPartial > 0) { 2950 // track has provided at least some frames recently: reset retry count 2951 track->mRetryCount = kMaxTrackRetries; 2952 } 2953 if (recentUnderruns == 0) { 2954 // no recent underruns: stay active 2955 break; 2956 } 2957 // there has recently been an underrun of some kind 2958 if (track->sharedBuffer() == 0) { 2959 // were any of the recent underruns "empty" (no frames available)? 2960 if (recentEmpty == 0) { 2961 // no, then ignore the partial underruns as they are allowed indefinitely 2962 break; 2963 } 2964 // there has recently been an "empty" underrun: decrement the retry counter 2965 if (--(track->mRetryCount) > 0) { 2966 break; 2967 } 2968 // indicate to client process that the track was disabled because of underrun; 2969 // it will then automatically call start() when data is available 2970 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2971 // remove from active list, but state remains ACTIVE [confusing but true] 2972 isActive = false; 2973 break; 2974 } 2975 // fall through 2976 case TrackBase::STOPPING_2: 2977 case TrackBase::PAUSED: 2978 case TrackBase::TERMINATED: 2979 case TrackBase::STOPPED: 2980 case TrackBase::FLUSHED: // flush() while active 2981 // Check for presentation complete if track is inactive 2982 // We have consumed all the buffers of this track. 2983 // This would be incomplete if we auto-paused on underrun 2984 { 2985 size_t audioHALFrames = 2986 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2987 size_t framesWritten = 2988 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2989 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2990 // track stays in active list until presentation is complete 2991 break; 2992 } 2993 } 2994 if (track->isStopping_2()) { 2995 track->mState = TrackBase::STOPPED; 2996 } 2997 if (track->isStopped()) { 2998 // Can't reset directly, as fast mixer is still polling this track 2999 // track->reset(); 3000 // So instead mark this track as needing to be reset after push with ack 3001 resetMask |= 1 << i; 3002 } 3003 isActive = false; 3004 break; 3005 case TrackBase::IDLE: 3006 default: 3007 LOG_FATAL("unexpected track state %d", track->mState); 3008 } 3009 3010 if (isActive) { 3011 // was it previously inactive? 3012 if (!(state->mTrackMask & (1 << j))) { 3013 ExtendedAudioBufferProvider *eabp = track; 3014 VolumeProvider *vp = track; 3015 fastTrack->mBufferProvider = eabp; 3016 fastTrack->mVolumeProvider = vp; 3017 fastTrack->mSampleRate = track->mSampleRate; 3018 fastTrack->mChannelMask = track->mChannelMask; 3019 fastTrack->mGeneration++; 3020 state->mTrackMask |= 1 << j; 3021 didModify = true; 3022 // no acknowledgement required for newly active tracks 3023 } 3024 // cache the combined master volume and stream type volume for fast mixer; this 3025 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3026 track->mCachedVolume = track->isMuted() ? 3027 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3028 ++fastTracks; 3029 } else { 3030 // was it previously active? 3031 if (state->mTrackMask & (1 << j)) { 3032 fastTrack->mBufferProvider = NULL; 3033 fastTrack->mGeneration++; 3034 state->mTrackMask &= ~(1 << j); 3035 didModify = true; 3036 // If any fast tracks were removed, we must wait for acknowledgement 3037 // because we're about to decrement the last sp<> on those tracks. 3038 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3039 } else { 3040 LOG_FATAL("fast track %d should have been active", j); 3041 } 3042 tracksToRemove->add(track); 3043 // Avoids a misleading display in dumpsys 3044 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3045 } 3046 continue; 3047 } 3048 3049 { // local variable scope to avoid goto warning 3050 3051 audio_track_cblk_t* cblk = track->cblk(); 3052 3053 // The first time a track is added we wait 3054 // for all its buffers to be filled before processing it 3055 int name = track->name(); 3056 // make sure that we have enough frames to mix one full buffer. 3057 // enforce this condition only once to enable draining the buffer in case the client 3058 // app does not call stop() and relies on underrun to stop: 3059 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3060 // during last round 3061 uint32_t minFrames = 1; 3062 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3063 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3064 if (t->sampleRate() == (int)mSampleRate) { 3065 minFrames = mNormalFrameCount; 3066 } else { 3067 // +1 for rounding and +1 for additional sample needed for interpolation 3068 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3069 // add frames already consumed but not yet released by the resampler 3070 // because cblk->framesReady() will include these frames 3071 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3072 // the minimum track buffer size is normally twice the number of frames necessary 3073 // to fill one buffer and the resampler should not leave more than one buffer worth 3074 // of unreleased frames after each pass, but just in case... 3075 ALOG_ASSERT(minFrames <= cblk->frameCount); 3076 } 3077 } 3078 if ((track->framesReady() >= minFrames) && track->isReady() && 3079 !track->isPaused() && !track->isTerminated()) 3080 { 3081 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3082 3083 mixedTracks++; 3084 3085 // track->mainBuffer() != mMixBuffer means there is an effect chain 3086 // connected to the track 3087 chain.clear(); 3088 if (track->mainBuffer() != mMixBuffer) { 3089 chain = getEffectChain_l(track->sessionId()); 3090 // Delegate volume control to effect in track effect chain if needed 3091 if (chain != 0) { 3092 tracksWithEffect++; 3093 } else { 3094 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3095 name, track->sessionId()); 3096 } 3097 } 3098 3099 3100 int param = AudioMixer::VOLUME; 3101 if (track->mFillingUpStatus == Track::FS_FILLED) { 3102 // no ramp for the first volume setting 3103 track->mFillingUpStatus = Track::FS_ACTIVE; 3104 if (track->mState == TrackBase::RESUMING) { 3105 track->mState = TrackBase::ACTIVE; 3106 param = AudioMixer::RAMP_VOLUME; 3107 } 3108 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3109 } else if (cblk->server != 0) { 3110 // If the track is stopped before the first frame was mixed, 3111 // do not apply ramp 3112 param = AudioMixer::RAMP_VOLUME; 3113 } 3114 3115 // compute volume for this track 3116 uint32_t vl, vr, va; 3117 if (track->isMuted() || track->isPausing() || 3118 mStreamTypes[track->streamType()].mute) { 3119 vl = vr = va = 0; 3120 if (track->isPausing()) { 3121 track->setPaused(); 3122 } 3123 } else { 3124 3125 // read original volumes with volume control 3126 float typeVolume = mStreamTypes[track->streamType()].volume; 3127 float v = masterVolume * typeVolume; 3128 uint32_t vlr = cblk->getVolumeLR(); 3129 vl = vlr & 0xFFFF; 3130 vr = vlr >> 16; 3131 // track volumes come from shared memory, so can't be trusted and must be clamped 3132 if (vl > MAX_GAIN_INT) { 3133 ALOGV("Track left volume out of range: %04X", vl); 3134 vl = MAX_GAIN_INT; 3135 } 3136 if (vr > MAX_GAIN_INT) { 3137 ALOGV("Track right volume out of range: %04X", vr); 3138 vr = MAX_GAIN_INT; 3139 } 3140 // now apply the master volume and stream type volume 3141 vl = (uint32_t)(v * vl) << 12; 3142 vr = (uint32_t)(v * vr) << 12; 3143 // assuming master volume and stream type volume each go up to 1.0, 3144 // vl and vr are now in 8.24 format 3145 3146 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3147 // send level comes from shared memory and so may be corrupt 3148 if (sendLevel > MAX_GAIN_INT) { 3149 ALOGV("Track send level out of range: %04X", sendLevel); 3150 sendLevel = MAX_GAIN_INT; 3151 } 3152 va = (uint32_t)(v * sendLevel); 3153 } 3154 // Delegate volume control to effect in track effect chain if needed 3155 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3156 // Do not ramp volume if volume is controlled by effect 3157 param = AudioMixer::VOLUME; 3158 track->mHasVolumeController = true; 3159 } else { 3160 // force no volume ramp when volume controller was just disabled or removed 3161 // from effect chain to avoid volume spike 3162 if (track->mHasVolumeController) { 3163 param = AudioMixer::VOLUME; 3164 } 3165 track->mHasVolumeController = false; 3166 } 3167 3168 // Convert volumes from 8.24 to 4.12 format 3169 // This additional clamping is needed in case chain->setVolume_l() overshot 3170 vl = (vl + (1 << 11)) >> 12; 3171 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3172 vr = (vr + (1 << 11)) >> 12; 3173 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3174 3175 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3176 3177 // XXX: these things DON'T need to be done each time 3178 mAudioMixer->setBufferProvider(name, track); 3179 mAudioMixer->enable(name); 3180 3181 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3182 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3183 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3184 mAudioMixer->setParameter( 3185 name, 3186 AudioMixer::TRACK, 3187 AudioMixer::FORMAT, (void *)track->format()); 3188 mAudioMixer->setParameter( 3189 name, 3190 AudioMixer::TRACK, 3191 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3192 mAudioMixer->setParameter( 3193 name, 3194 AudioMixer::RESAMPLE, 3195 AudioMixer::SAMPLE_RATE, 3196 (void *)(cblk->sampleRate)); 3197 mAudioMixer->setParameter( 3198 name, 3199 AudioMixer::TRACK, 3200 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3201 mAudioMixer->setParameter( 3202 name, 3203 AudioMixer::TRACK, 3204 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3205 3206 // reset retry count 3207 track->mRetryCount = kMaxTrackRetries; 3208 3209 // If one track is ready, set the mixer ready if: 3210 // - the mixer was not ready during previous round OR 3211 // - no other track is not ready 3212 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3213 mixerStatus != MIXER_TRACKS_ENABLED) { 3214 mixerStatus = MIXER_TRACKS_READY; 3215 } 3216 } else { 3217 // clear effect chain input buffer if an active track underruns to avoid sending 3218 // previous audio buffer again to effects 3219 chain = getEffectChain_l(track->sessionId()); 3220 if (chain != 0) { 3221 chain->clearInputBuffer(); 3222 } 3223 3224 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3225 if ((track->sharedBuffer() != 0) || 3226 track->isStopped() || track->isPaused()) { 3227 // We have consumed all the buffers of this track. 3228 // Remove it from the list of active tracks. 3229 // TODO: use actual buffer filling status instead of latency when available from 3230 // audio HAL 3231 size_t audioHALFrames = 3232 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3233 size_t framesWritten = 3234 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3235 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3236 if (track->isStopped()) { 3237 track->reset(); 3238 } 3239 tracksToRemove->add(track); 3240 } 3241 } else { 3242 track->mUnderrunCount++; 3243 // No buffers for this track. Give it a few chances to 3244 // fill a buffer, then remove it from active list. 3245 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3246 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3247 tracksToRemove->add(track); 3248 // indicate to client process that the track was disabled because of underrun; 3249 // it will then automatically call start() when data is available 3250 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3251 // If one track is not ready, mark the mixer also not ready if: 3252 // - the mixer was ready during previous round OR 3253 // - no other track is ready 3254 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3255 mixerStatus != MIXER_TRACKS_READY) { 3256 mixerStatus = MIXER_TRACKS_ENABLED; 3257 } 3258 } 3259 mAudioMixer->disable(name); 3260 } 3261 3262 } // local variable scope to avoid goto warning 3263track_is_ready: ; 3264 3265 } 3266 3267 // Push the new FastMixer state if necessary 3268 bool pauseAudioWatchdog = false; 3269 if (didModify) { 3270 state->mFastTracksGen++; 3271 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3272 if (kUseFastMixer == FastMixer_Dynamic && 3273 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3274 state->mCommand = FastMixerState::COLD_IDLE; 3275 state->mColdFutexAddr = &mFastMixerFutex; 3276 state->mColdGen++; 3277 mFastMixerFutex = 0; 3278 if (kUseFastMixer == FastMixer_Dynamic) { 3279 mNormalSink = mOutputSink; 3280 } 3281 // If we go into cold idle, need to wait for acknowledgement 3282 // so that fast mixer stops doing I/O. 3283 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3284 pauseAudioWatchdog = true; 3285 } 3286 sq->end(); 3287 } 3288 if (sq != NULL) { 3289 sq->end(didModify); 3290 sq->push(block); 3291 } 3292 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3293 mAudioWatchdog->pause(); 3294 } 3295 3296 // Now perform the deferred reset on fast tracks that have stopped 3297 while (resetMask != 0) { 3298 size_t i = __builtin_ctz(resetMask); 3299 ALOG_ASSERT(i < count); 3300 resetMask &= ~(1 << i); 3301 sp<Track> t = mActiveTracks[i].promote(); 3302 if (t == 0) continue; 3303 Track* track = t.get(); 3304 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3305 track->reset(); 3306 } 3307 3308 // remove all the tracks that need to be... 3309 count = tracksToRemove->size(); 3310 if (CC_UNLIKELY(count)) { 3311 for (size_t i=0 ; i<count ; i++) { 3312 const sp<Track>& track = tracksToRemove->itemAt(i); 3313 mActiveTracks.remove(track); 3314 if (track->mainBuffer() != mMixBuffer) { 3315 chain = getEffectChain_l(track->sessionId()); 3316 if (chain != 0) { 3317 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3318 chain->decActiveTrackCnt(); 3319 } 3320 } 3321 if (track->isTerminated()) { 3322 removeTrack_l(track); 3323 } 3324 } 3325 } 3326 3327 // mix buffer must be cleared if all tracks are connected to an 3328 // effect chain as in this case the mixer will not write to 3329 // mix buffer and track effects will accumulate into it 3330 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3331 // FIXME as a performance optimization, should remember previous zero status 3332 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3333 } 3334 3335 // if any fast tracks, then status is ready 3336 mMixerStatusIgnoringFastTracks = mixerStatus; 3337 if (fastTracks > 0) { 3338 mixerStatus = MIXER_TRACKS_READY; 3339 } 3340 return mixerStatus; 3341} 3342 3343/* 3344The derived values that are cached: 3345 - mixBufferSize from frame count * frame size 3346 - activeSleepTime from activeSleepTimeUs() 3347 - idleSleepTime from idleSleepTimeUs() 3348 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3349 - maxPeriod from frame count and sample rate (MIXER only) 3350 3351The parameters that affect these derived values are: 3352 - frame count 3353 - frame size 3354 - sample rate 3355 - device type: A2DP or not 3356 - device latency 3357 - format: PCM or not 3358 - active sleep time 3359 - idle sleep time 3360*/ 3361 3362void AudioFlinger::PlaybackThread::cacheParameters_l() 3363{ 3364 mixBufferSize = mNormalFrameCount * mFrameSize; 3365 activeSleepTime = activeSleepTimeUs(); 3366 idleSleepTime = idleSleepTimeUs(); 3367} 3368 3369void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3370{ 3371 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3372 this, streamType, mTracks.size()); 3373 Mutex::Autolock _l(mLock); 3374 3375 size_t size = mTracks.size(); 3376 for (size_t i = 0; i < size; i++) { 3377 sp<Track> t = mTracks[i]; 3378 if (t->streamType() == streamType) { 3379 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3380 t->mCblk->cv.signal(); 3381 } 3382 } 3383} 3384 3385// getTrackName_l() must be called with ThreadBase::mLock held 3386int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3387{ 3388 return mAudioMixer->getTrackName(channelMask); 3389} 3390 3391// deleteTrackName_l() must be called with ThreadBase::mLock held 3392void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3393{ 3394 ALOGV("remove track (%d) and delete from mixer", name); 3395 mAudioMixer->deleteTrackName(name); 3396} 3397 3398// checkForNewParameters_l() must be called with ThreadBase::mLock held 3399bool AudioFlinger::MixerThread::checkForNewParameters_l() 3400{ 3401 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3402 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3403 bool reconfig = false; 3404 3405 while (!mNewParameters.isEmpty()) { 3406 3407 if (mFastMixer != NULL) { 3408 FastMixerStateQueue *sq = mFastMixer->sq(); 3409 FastMixerState *state = sq->begin(); 3410 if (!(state->mCommand & FastMixerState::IDLE)) { 3411 previousCommand = state->mCommand; 3412 state->mCommand = FastMixerState::HOT_IDLE; 3413 sq->end(); 3414 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3415 } else { 3416 sq->end(false /*didModify*/); 3417 } 3418 } 3419 3420 status_t status = NO_ERROR; 3421 String8 keyValuePair = mNewParameters[0]; 3422 AudioParameter param = AudioParameter(keyValuePair); 3423 int value; 3424 3425 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3426 reconfig = true; 3427 } 3428 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3429 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3430 status = BAD_VALUE; 3431 } else { 3432 reconfig = true; 3433 } 3434 } 3435 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3436 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3437 status = BAD_VALUE; 3438 } else { 3439 reconfig = true; 3440 } 3441 } 3442 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3443 // do not accept frame count changes if tracks are open as the track buffer 3444 // size depends on frame count and correct behavior would not be guaranteed 3445 // if frame count is changed after track creation 3446 if (!mTracks.isEmpty()) { 3447 status = INVALID_OPERATION; 3448 } else { 3449 reconfig = true; 3450 } 3451 } 3452 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3453#ifdef ADD_BATTERY_DATA 3454 // when changing the audio output device, call addBatteryData to notify 3455 // the change 3456 if (mOutDevice != value) { 3457 uint32_t params = 0; 3458 // check whether speaker is on 3459 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3460 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3461 } 3462 3463 audio_devices_t deviceWithoutSpeaker 3464 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3465 // check if any other device (except speaker) is on 3466 if (value & deviceWithoutSpeaker ) { 3467 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3468 } 3469 3470 if (params != 0) { 3471 addBatteryData(params); 3472 } 3473 } 3474#endif 3475 3476 // forward device change to effects that have requested to be 3477 // aware of attached audio device. 3478 mOutDevice = value; 3479 for (size_t i = 0; i < mEffectChains.size(); i++) { 3480 mEffectChains[i]->setDevice_l(mOutDevice); 3481 } 3482 } 3483 3484 if (status == NO_ERROR) { 3485 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3486 keyValuePair.string()); 3487 if (!mStandby && status == INVALID_OPERATION) { 3488 mOutput->stream->common.standby(&mOutput->stream->common); 3489 mStandby = true; 3490 mBytesWritten = 0; 3491 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3492 keyValuePair.string()); 3493 } 3494 if (status == NO_ERROR && reconfig) { 3495 delete mAudioMixer; 3496 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3497 mAudioMixer = NULL; 3498 readOutputParameters(); 3499 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3500 for (size_t i = 0; i < mTracks.size() ; i++) { 3501 int name = getTrackName_l(mTracks[i]->mChannelMask); 3502 if (name < 0) break; 3503 mTracks[i]->mName = name; 3504 // limit track sample rate to 2 x new output sample rate 3505 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3506 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3507 } 3508 } 3509 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3510 } 3511 } 3512 3513 mNewParameters.removeAt(0); 3514 3515 mParamStatus = status; 3516 mParamCond.signal(); 3517 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3518 // already timed out waiting for the status and will never signal the condition. 3519 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3520 } 3521 3522 if (!(previousCommand & FastMixerState::IDLE)) { 3523 ALOG_ASSERT(mFastMixer != NULL); 3524 FastMixerStateQueue *sq = mFastMixer->sq(); 3525 FastMixerState *state = sq->begin(); 3526 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3527 state->mCommand = previousCommand; 3528 sq->end(); 3529 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3530 } 3531 3532 return reconfig; 3533} 3534 3535void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3536{ 3537 const size_t SIZE = 256; 3538 char buffer[SIZE]; 3539 String8 result; 3540 3541 PlaybackThread::dumpInternals(fd, args); 3542 3543 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3544 result.append(buffer); 3545 write(fd, result.string(), result.size()); 3546 3547 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3548 FastMixerDumpState copy = mFastMixerDumpState; 3549 copy.dump(fd); 3550 3551#ifdef STATE_QUEUE_DUMP 3552 // Similar for state queue 3553 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3554 observerCopy.dump(fd); 3555 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3556 mutatorCopy.dump(fd); 3557#endif 3558 3559 // Write the tee output to a .wav file 3560 NBAIO_Source *teeSource = mTeeSource.get(); 3561 if (teeSource != NULL) { 3562 char teePath[64]; 3563 struct timeval tv; 3564 gettimeofday(&tv, NULL); 3565 struct tm tm; 3566 localtime_r(&tv.tv_sec, &tm); 3567 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3568 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3569 if (teeFd >= 0) { 3570 char wavHeader[44]; 3571 memcpy(wavHeader, 3572 "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", 3573 sizeof(wavHeader)); 3574 NBAIO_Format format = teeSource->format(); 3575 unsigned channelCount = Format_channelCount(format); 3576 ALOG_ASSERT(channelCount <= FCC_2); 3577 unsigned sampleRate = Format_sampleRate(format); 3578 wavHeader[22] = channelCount; // number of channels 3579 wavHeader[24] = sampleRate; // sample rate 3580 wavHeader[25] = sampleRate >> 8; 3581 wavHeader[32] = channelCount * 2; // block alignment 3582 write(teeFd, wavHeader, sizeof(wavHeader)); 3583 size_t total = 0; 3584 bool firstRead = true; 3585 for (;;) { 3586#define TEE_SINK_READ 1024 3587 short buffer[TEE_SINK_READ * FCC_2]; 3588 size_t count = TEE_SINK_READ; 3589 ssize_t actual = teeSource->read(buffer, count, 3590 AudioBufferProvider::kInvalidPTS); 3591 bool wasFirstRead = firstRead; 3592 firstRead = false; 3593 if (actual <= 0) { 3594 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3595 continue; 3596 } 3597 break; 3598 } 3599 ALOG_ASSERT(actual <= (ssize_t)count); 3600 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3601 total += actual; 3602 } 3603 lseek(teeFd, (off_t) 4, SEEK_SET); 3604 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3605 write(teeFd, &temp, sizeof(temp)); 3606 lseek(teeFd, (off_t) 40, SEEK_SET); 3607 temp = total * channelCount * sizeof(short); 3608 write(teeFd, &temp, sizeof(temp)); 3609 close(teeFd); 3610 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3611 } else { 3612 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3613 } 3614 } 3615 3616 if (mAudioWatchdog != 0) { 3617 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3618 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3619 wdCopy.dump(fd); 3620 } 3621} 3622 3623uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3624{ 3625 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3626} 3627 3628uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3629{ 3630 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3631} 3632 3633void AudioFlinger::MixerThread::cacheParameters_l() 3634{ 3635 PlaybackThread::cacheParameters_l(); 3636 3637 // FIXME: Relaxed timing because of a certain device that can't meet latency 3638 // Should be reduced to 2x after the vendor fixes the driver issue 3639 // increase threshold again due to low power audio mode. The way this warning 3640 // threshold is calculated and its usefulness should be reconsidered anyway. 3641 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3642} 3643 3644// ---------------------------------------------------------------------------- 3645AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3646 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3647 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3648 // mLeftVolFloat, mRightVolFloat 3649{ 3650} 3651 3652AudioFlinger::DirectOutputThread::~DirectOutputThread() 3653{ 3654} 3655 3656AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3657 Vector< sp<Track> > *tracksToRemove 3658) 3659{ 3660 sp<Track> trackToRemove; 3661 3662 mixer_state mixerStatus = MIXER_IDLE; 3663 3664 // find out which tracks need to be processed 3665 if (mActiveTracks.size() != 0) { 3666 sp<Track> t = mActiveTracks[0].promote(); 3667 // The track died recently 3668 if (t == 0) return MIXER_IDLE; 3669 3670 Track* const track = t.get(); 3671 audio_track_cblk_t* cblk = track->cblk(); 3672 3673 // The first time a track is added we wait 3674 // for all its buffers to be filled before processing it 3675 uint32_t minFrames; 3676 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3677 minFrames = mNormalFrameCount; 3678 } else { 3679 minFrames = 1; 3680 } 3681 if ((track->framesReady() >= minFrames) && track->isReady() && 3682 !track->isPaused() && !track->isTerminated()) 3683 { 3684 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3685 3686 if (track->mFillingUpStatus == Track::FS_FILLED) { 3687 track->mFillingUpStatus = Track::FS_ACTIVE; 3688 mLeftVolFloat = mRightVolFloat = 0; 3689 if (track->mState == TrackBase::RESUMING) { 3690 track->mState = TrackBase::ACTIVE; 3691 } 3692 } 3693 3694 // compute volume for this track 3695 float left, right; 3696 if (track->isMuted() || mMasterMute || track->isPausing() || 3697 mStreamTypes[track->streamType()].mute) { 3698 left = right = 0; 3699 if (track->isPausing()) { 3700 track->setPaused(); 3701 } 3702 } else { 3703 float typeVolume = mStreamTypes[track->streamType()].volume; 3704 float v = mMasterVolume * typeVolume; 3705 uint32_t vlr = cblk->getVolumeLR(); 3706 float v_clamped = v * (vlr & 0xFFFF); 3707 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3708 left = v_clamped/MAX_GAIN; 3709 v_clamped = v * (vlr >> 16); 3710 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3711 right = v_clamped/MAX_GAIN; 3712 } 3713 3714 if (left != mLeftVolFloat || right != mRightVolFloat) { 3715 mLeftVolFloat = left; 3716 mRightVolFloat = right; 3717 3718 // Convert volumes from float to 8.24 3719 uint32_t vl = (uint32_t)(left * (1 << 24)); 3720 uint32_t vr = (uint32_t)(right * (1 << 24)); 3721 3722 // Delegate volume control to effect in track effect chain if needed 3723 // only one effect chain can be present on DirectOutputThread, so if 3724 // there is one, the track is connected to it 3725 if (!mEffectChains.isEmpty()) { 3726 // Do not ramp volume if volume is controlled by effect 3727 mEffectChains[0]->setVolume_l(&vl, &vr); 3728 left = (float)vl / (1 << 24); 3729 right = (float)vr / (1 << 24); 3730 } 3731 mOutput->stream->set_volume(mOutput->stream, left, right); 3732 } 3733 3734 // reset retry count 3735 track->mRetryCount = kMaxTrackRetriesDirect; 3736 mActiveTrack = t; 3737 mixerStatus = MIXER_TRACKS_READY; 3738 } else { 3739 // clear effect chain input buffer if an active track underruns to avoid sending 3740 // previous audio buffer again to effects 3741 if (!mEffectChains.isEmpty()) { 3742 mEffectChains[0]->clearInputBuffer(); 3743 } 3744 3745 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3746 if ((track->sharedBuffer() != 0) || 3747 track->isStopped() || track->isPaused()) { 3748 // We have consumed all the buffers of this track. 3749 // Remove it from the list of active tracks. 3750 // TODO: implement behavior for compressed audio 3751 size_t audioHALFrames = 3752 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3753 size_t framesWritten = 3754 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3755 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3756 if (track->isStopped()) { 3757 track->reset(); 3758 } 3759 trackToRemove = track; 3760 } 3761 } else { 3762 // No buffers for this track. Give it a few chances to 3763 // fill a buffer, then remove it from active list. 3764 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3765 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3766 trackToRemove = track; 3767 } else { 3768 mixerStatus = MIXER_TRACKS_ENABLED; 3769 } 3770 } 3771 } 3772 } 3773 3774 // FIXME merge this with similar code for removing multiple tracks 3775 // remove all the tracks that need to be... 3776 if (CC_UNLIKELY(trackToRemove != 0)) { 3777 tracksToRemove->add(trackToRemove); 3778 mActiveTracks.remove(trackToRemove); 3779 if (!mEffectChains.isEmpty()) { 3780 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3781 trackToRemove->sessionId()); 3782 mEffectChains[0]->decActiveTrackCnt(); 3783 } 3784 if (trackToRemove->isTerminated()) { 3785 removeTrack_l(trackToRemove); 3786 } 3787 } 3788 3789 return mixerStatus; 3790} 3791 3792void AudioFlinger::DirectOutputThread::threadLoop_mix() 3793{ 3794 AudioBufferProvider::Buffer buffer; 3795 size_t frameCount = mFrameCount; 3796 int8_t *curBuf = (int8_t *)mMixBuffer; 3797 // output audio to hardware 3798 while (frameCount) { 3799 buffer.frameCount = frameCount; 3800 mActiveTrack->getNextBuffer(&buffer); 3801 if (CC_UNLIKELY(buffer.raw == NULL)) { 3802 memset(curBuf, 0, frameCount * mFrameSize); 3803 break; 3804 } 3805 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3806 frameCount -= buffer.frameCount; 3807 curBuf += buffer.frameCount * mFrameSize; 3808 mActiveTrack->releaseBuffer(&buffer); 3809 } 3810 sleepTime = 0; 3811 standbyTime = systemTime() + standbyDelay; 3812 mActiveTrack.clear(); 3813 3814} 3815 3816void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3817{ 3818 if (sleepTime == 0) { 3819 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3820 sleepTime = activeSleepTime; 3821 } else { 3822 sleepTime = idleSleepTime; 3823 } 3824 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3825 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3826 sleepTime = 0; 3827 } 3828} 3829 3830// getTrackName_l() must be called with ThreadBase::mLock held 3831int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3832{ 3833 return 0; 3834} 3835 3836// deleteTrackName_l() must be called with ThreadBase::mLock held 3837void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3838{ 3839} 3840 3841// checkForNewParameters_l() must be called with ThreadBase::mLock held 3842bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3843{ 3844 bool reconfig = false; 3845 3846 while (!mNewParameters.isEmpty()) { 3847 status_t status = NO_ERROR; 3848 String8 keyValuePair = mNewParameters[0]; 3849 AudioParameter param = AudioParameter(keyValuePair); 3850 int value; 3851 3852 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3853 // do not accept frame count changes if tracks are open as the track buffer 3854 // size depends on frame count and correct behavior would not be garantied 3855 // if frame count is changed after track creation 3856 if (!mTracks.isEmpty()) { 3857 status = INVALID_OPERATION; 3858 } else { 3859 reconfig = true; 3860 } 3861 } 3862 if (status == NO_ERROR) { 3863 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3864 keyValuePair.string()); 3865 if (!mStandby && status == INVALID_OPERATION) { 3866 mOutput->stream->common.standby(&mOutput->stream->common); 3867 mStandby = true; 3868 mBytesWritten = 0; 3869 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3870 keyValuePair.string()); 3871 } 3872 if (status == NO_ERROR && reconfig) { 3873 readOutputParameters(); 3874 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3875 } 3876 } 3877 3878 mNewParameters.removeAt(0); 3879 3880 mParamStatus = status; 3881 mParamCond.signal(); 3882 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3883 // already timed out waiting for the status and will never signal the condition. 3884 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3885 } 3886 return reconfig; 3887} 3888 3889uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3890{ 3891 uint32_t time; 3892 if (audio_is_linear_pcm(mFormat)) { 3893 time = PlaybackThread::activeSleepTimeUs(); 3894 } else { 3895 time = 10000; 3896 } 3897 return time; 3898} 3899 3900uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3901{ 3902 uint32_t time; 3903 if (audio_is_linear_pcm(mFormat)) { 3904 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3905 } else { 3906 time = 10000; 3907 } 3908 return time; 3909} 3910 3911uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3912{ 3913 uint32_t time; 3914 if (audio_is_linear_pcm(mFormat)) { 3915 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3916 } else { 3917 time = 10000; 3918 } 3919 return time; 3920} 3921 3922void AudioFlinger::DirectOutputThread::cacheParameters_l() 3923{ 3924 PlaybackThread::cacheParameters_l(); 3925 3926 // use shorter standby delay as on normal output to release 3927 // hardware resources as soon as possible 3928 standbyDelay = microseconds(activeSleepTime*2); 3929} 3930 3931// ---------------------------------------------------------------------------- 3932 3933AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3934 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3935 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING), 3936 mWaitTimeMs(UINT_MAX) 3937{ 3938 addOutputTrack(mainThread); 3939} 3940 3941AudioFlinger::DuplicatingThread::~DuplicatingThread() 3942{ 3943 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3944 mOutputTracks[i]->destroy(); 3945 } 3946} 3947 3948void AudioFlinger::DuplicatingThread::threadLoop_mix() 3949{ 3950 // mix buffers... 3951 if (outputsReady(outputTracks)) { 3952 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3953 } else { 3954 memset(mMixBuffer, 0, mixBufferSize); 3955 } 3956 sleepTime = 0; 3957 writeFrames = mNormalFrameCount; 3958 standbyTime = systemTime() + standbyDelay; 3959} 3960 3961void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3962{ 3963 if (sleepTime == 0) { 3964 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3965 sleepTime = activeSleepTime; 3966 } else { 3967 sleepTime = idleSleepTime; 3968 } 3969 } else if (mBytesWritten != 0) { 3970 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3971 writeFrames = mNormalFrameCount; 3972 memset(mMixBuffer, 0, mixBufferSize); 3973 } else { 3974 // flush remaining overflow buffers in output tracks 3975 writeFrames = 0; 3976 } 3977 sleepTime = 0; 3978 } 3979} 3980 3981void AudioFlinger::DuplicatingThread::threadLoop_write() 3982{ 3983 for (size_t i = 0; i < outputTracks.size(); i++) { 3984 outputTracks[i]->write(mMixBuffer, writeFrames); 3985 } 3986 mBytesWritten += mixBufferSize; 3987} 3988 3989void AudioFlinger::DuplicatingThread::threadLoop_standby() 3990{ 3991 // DuplicatingThread implements standby by stopping all tracks 3992 for (size_t i = 0; i < outputTracks.size(); i++) { 3993 outputTracks[i]->stop(); 3994 } 3995} 3996 3997void AudioFlinger::DuplicatingThread::saveOutputTracks() 3998{ 3999 outputTracks = mOutputTracks; 4000} 4001 4002void AudioFlinger::DuplicatingThread::clearOutputTracks() 4003{ 4004 outputTracks.clear(); 4005} 4006 4007void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4008{ 4009 Mutex::Autolock _l(mLock); 4010 // FIXME explain this formula 4011 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4012 OutputTrack *outputTrack = new OutputTrack(thread, 4013 this, 4014 mSampleRate, 4015 mFormat, 4016 mChannelMask, 4017 frameCount); 4018 if (outputTrack->cblk() != NULL) { 4019 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4020 mOutputTracks.add(outputTrack); 4021 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4022 updateWaitTime_l(); 4023 } 4024} 4025 4026void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4027{ 4028 Mutex::Autolock _l(mLock); 4029 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4030 if (mOutputTracks[i]->thread() == thread) { 4031 mOutputTracks[i]->destroy(); 4032 mOutputTracks.removeAt(i); 4033 updateWaitTime_l(); 4034 return; 4035 } 4036 } 4037 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4038} 4039 4040// caller must hold mLock 4041void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4042{ 4043 mWaitTimeMs = UINT_MAX; 4044 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4045 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4046 if (strong != 0) { 4047 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4048 if (waitTimeMs < mWaitTimeMs) { 4049 mWaitTimeMs = waitTimeMs; 4050 } 4051 } 4052 } 4053} 4054 4055 4056bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4057{ 4058 for (size_t i = 0; i < outputTracks.size(); i++) { 4059 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4060 if (thread == 0) { 4061 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4062 return false; 4063 } 4064 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4065 // see note at standby() declaration 4066 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4067 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4068 return false; 4069 } 4070 } 4071 return true; 4072} 4073 4074uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4075{ 4076 return (mWaitTimeMs * 1000) / 2; 4077} 4078 4079void AudioFlinger::DuplicatingThread::cacheParameters_l() 4080{ 4081 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4082 updateWaitTime_l(); 4083 4084 MixerThread::cacheParameters_l(); 4085} 4086 4087// ---------------------------------------------------------------------------- 4088 4089// TrackBase constructor must be called with AudioFlinger::mLock held 4090AudioFlinger::ThreadBase::TrackBase::TrackBase( 4091 ThreadBase *thread, 4092 const sp<Client>& client, 4093 uint32_t sampleRate, 4094 audio_format_t format, 4095 audio_channel_mask_t channelMask, 4096 int frameCount, 4097 const sp<IMemory>& sharedBuffer, 4098 int sessionId) 4099 : RefBase(), 4100 mThread(thread), 4101 mClient(client), 4102 mCblk(NULL), 4103 // mBuffer 4104 // mBufferEnd 4105 mFrameCount(0), 4106 mState(IDLE), 4107 mSampleRate(sampleRate), 4108 mFormat(format), 4109 mStepServerFailed(false), 4110 mSessionId(sessionId) 4111 // mChannelCount 4112 // mChannelMask 4113{ 4114 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4115 4116 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4117 size_t size = sizeof(audio_track_cblk_t); 4118 uint8_t channelCount = popcount(channelMask); 4119 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4120 if (sharedBuffer == 0) { 4121 size += bufferSize; 4122 } 4123 4124 if (client != NULL) { 4125 mCblkMemory = client->heap()->allocate(size); 4126 if (mCblkMemory != 0) { 4127 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4128 if (mCblk != NULL) { // construct the shared structure in-place. 4129 new(mCblk) audio_track_cblk_t(); 4130 // clear all buffers 4131 mCblk->frameCount = frameCount; 4132 mCblk->sampleRate = sampleRate; 4133// uncomment the following lines to quickly test 32-bit wraparound 4134// mCblk->user = 0xffff0000; 4135// mCblk->server = 0xffff0000; 4136// mCblk->userBase = 0xffff0000; 4137// mCblk->serverBase = 0xffff0000; 4138 mChannelCount = channelCount; 4139 mChannelMask = channelMask; 4140 if (sharedBuffer == 0) { 4141 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4142 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4143 // Force underrun condition to avoid false underrun callback until first data is 4144 // written to buffer (other flags are cleared) 4145 mCblk->flags = CBLK_UNDERRUN_ON; 4146 } else { 4147 mBuffer = sharedBuffer->pointer(); 4148 } 4149 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4150 } 4151 } else { 4152 ALOGE("not enough memory for AudioTrack size=%u", size); 4153 client->heap()->dump("AudioTrack"); 4154 return; 4155 } 4156 } else { 4157 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4158 // construct the shared structure in-place. 4159 new(mCblk) audio_track_cblk_t(); 4160 // clear all buffers 4161 mCblk->frameCount = frameCount; 4162 mCblk->sampleRate = sampleRate; 4163// uncomment the following lines to quickly test 32-bit wraparound 4164// mCblk->user = 0xffff0000; 4165// mCblk->server = 0xffff0000; 4166// mCblk->userBase = 0xffff0000; 4167// mCblk->serverBase = 0xffff0000; 4168 mChannelCount = channelCount; 4169 mChannelMask = channelMask; 4170 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4171 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4172 // Force underrun condition to avoid false underrun callback until first data is 4173 // written to buffer (other flags are cleared) 4174 mCblk->flags = CBLK_UNDERRUN_ON; 4175 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4176 } 4177} 4178 4179AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4180{ 4181 if (mCblk != NULL) { 4182 if (mClient == 0) { 4183 delete mCblk; 4184 } else { 4185 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4186 } 4187 } 4188 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4189 if (mClient != 0) { 4190 // Client destructor must run with AudioFlinger mutex locked 4191 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4192 // If the client's reference count drops to zero, the associated destructor 4193 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4194 // relying on the automatic clear() at end of scope. 4195 mClient.clear(); 4196 } 4197} 4198 4199// AudioBufferProvider interface 4200// getNextBuffer() = 0; 4201// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4202void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4203{ 4204 buffer->raw = NULL; 4205 mFrameCount = buffer->frameCount; 4206 // FIXME See note at getNextBuffer() 4207 (void) step(); // ignore return value of step() 4208 buffer->frameCount = 0; 4209} 4210 4211bool AudioFlinger::ThreadBase::TrackBase::step() { 4212 bool result; 4213 audio_track_cblk_t* cblk = this->cblk(); 4214 4215 result = cblk->stepServer(mFrameCount); 4216 if (!result) { 4217 ALOGV("stepServer failed acquiring cblk mutex"); 4218 mStepServerFailed = true; 4219 } 4220 return result; 4221} 4222 4223void AudioFlinger::ThreadBase::TrackBase::reset() { 4224 audio_track_cblk_t* cblk = this->cblk(); 4225 4226 cblk->user = 0; 4227 cblk->server = 0; 4228 cblk->userBase = 0; 4229 cblk->serverBase = 0; 4230 mStepServerFailed = false; 4231 ALOGV("TrackBase::reset"); 4232} 4233 4234int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4235 return (int)mCblk->sampleRate; 4236} 4237 4238void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4239 audio_track_cblk_t* cblk = this->cblk(); 4240 size_t frameSize = cblk->frameSize; 4241 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4242 int8_t *bufferEnd = bufferStart + frames * frameSize; 4243 4244 // Check validity of returned pointer in case the track control block would have been corrupted. 4245 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4246 "TrackBase::getBuffer buffer out of range:\n" 4247 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4248 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4249 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4250 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4251 4252 return bufferStart; 4253} 4254 4255status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4256{ 4257 mSyncEvents.add(event); 4258 return NO_ERROR; 4259} 4260 4261// ---------------------------------------------------------------------------- 4262 4263// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4264AudioFlinger::PlaybackThread::Track::Track( 4265 PlaybackThread *thread, 4266 const sp<Client>& client, 4267 audio_stream_type_t streamType, 4268 uint32_t sampleRate, 4269 audio_format_t format, 4270 audio_channel_mask_t channelMask, 4271 int frameCount, 4272 const sp<IMemory>& sharedBuffer, 4273 int sessionId, 4274 IAudioFlinger::track_flags_t flags) 4275 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4276 mMute(false), 4277 mFillingUpStatus(FS_INVALID), 4278 // mRetryCount initialized later when needed 4279 mSharedBuffer(sharedBuffer), 4280 mStreamType(streamType), 4281 mName(-1), // see note below 4282 mMainBuffer(thread->mixBuffer()), 4283 mAuxBuffer(NULL), 4284 mAuxEffectId(0), mHasVolumeController(false), 4285 mPresentationCompleteFrames(0), 4286 mFlags(flags), 4287 mFastIndex(-1), 4288 mUnderrunCount(0), 4289 mCachedVolume(1.0) 4290{ 4291 if (mCblk != NULL) { 4292 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4293 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4294 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4295 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4296 mName = thread->getTrackName_l(channelMask); 4297 mCblk->mName = mName; 4298 if (mName < 0) { 4299 ALOGE("no more track names available"); 4300 return; 4301 } 4302 // only allocate a fast track index if we were able to allocate a normal track name 4303 if (flags & IAudioFlinger::TRACK_FAST) { 4304 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4305 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4306 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4307 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4308 // FIXME This is too eager. We allocate a fast track index before the 4309 // fast track becomes active. Since fast tracks are a scarce resource, 4310 // this means we are potentially denying other more important fast tracks from 4311 // being created. It would be better to allocate the index dynamically. 4312 mFastIndex = i; 4313 mCblk->mName = i; 4314 // Read the initial underruns because this field is never cleared by the fast mixer 4315 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4316 thread->mFastTrackAvailMask &= ~(1 << i); 4317 } 4318 } 4319 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4320} 4321 4322AudioFlinger::PlaybackThread::Track::~Track() 4323{ 4324 ALOGV("PlaybackThread::Track destructor"); 4325} 4326 4327void AudioFlinger::PlaybackThread::Track::destroy() 4328{ 4329 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4330 // by removing it from mTracks vector, so there is a risk that this Tracks's 4331 // destructor is called. As the destructor needs to lock mLock, 4332 // we must acquire a strong reference on this Track before locking mLock 4333 // here so that the destructor is called only when exiting this function. 4334 // On the other hand, as long as Track::destroy() is only called by 4335 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4336 // this Track with its member mTrack. 4337 sp<Track> keep(this); 4338 { // scope for mLock 4339 sp<ThreadBase> thread = mThread.promote(); 4340 if (thread != 0) { 4341 if (!isOutputTrack()) { 4342 if (mState == ACTIVE || mState == RESUMING) { 4343 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4344 4345#ifdef ADD_BATTERY_DATA 4346 // to track the speaker usage 4347 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4348#endif 4349 } 4350 AudioSystem::releaseOutput(thread->id()); 4351 } 4352 Mutex::Autolock _l(thread->mLock); 4353 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4354 playbackThread->destroyTrack_l(this); 4355 } 4356 } 4357} 4358 4359/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4360{ 4361 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4362 " Server User Main buf Aux Buf Flags Underruns\n"); 4363} 4364 4365void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4366{ 4367 uint32_t vlr = mCblk->getVolumeLR(); 4368 if (isFastTrack()) { 4369 sprintf(buffer, " F %2d", mFastIndex); 4370 } else { 4371 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4372 } 4373 track_state state = mState; 4374 char stateChar; 4375 switch (state) { 4376 case IDLE: 4377 stateChar = 'I'; 4378 break; 4379 case TERMINATED: 4380 stateChar = 'T'; 4381 break; 4382 case STOPPING_1: 4383 stateChar = 's'; 4384 break; 4385 case STOPPING_2: 4386 stateChar = '5'; 4387 break; 4388 case STOPPED: 4389 stateChar = 'S'; 4390 break; 4391 case RESUMING: 4392 stateChar = 'R'; 4393 break; 4394 case ACTIVE: 4395 stateChar = 'A'; 4396 break; 4397 case PAUSING: 4398 stateChar = 'p'; 4399 break; 4400 case PAUSED: 4401 stateChar = 'P'; 4402 break; 4403 case FLUSHED: 4404 stateChar = 'F'; 4405 break; 4406 default: 4407 stateChar = '?'; 4408 break; 4409 } 4410 char nowInUnderrun; 4411 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4412 case UNDERRUN_FULL: 4413 nowInUnderrun = ' '; 4414 break; 4415 case UNDERRUN_PARTIAL: 4416 nowInUnderrun = '<'; 4417 break; 4418 case UNDERRUN_EMPTY: 4419 nowInUnderrun = '*'; 4420 break; 4421 default: 4422 nowInUnderrun = '?'; 4423 break; 4424 } 4425 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4426 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4427 (mClient == 0) ? getpid_cached : mClient->pid(), 4428 mStreamType, 4429 mFormat, 4430 mChannelMask, 4431 mSessionId, 4432 mFrameCount, 4433 mCblk->frameCount, 4434 stateChar, 4435 mMute, 4436 mFillingUpStatus, 4437 mCblk->sampleRate, 4438 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4439 20.0 * log10((vlr >> 16) / 4096.0), 4440 mCblk->server, 4441 mCblk->user, 4442 (int)mMainBuffer, 4443 (int)mAuxBuffer, 4444 mCblk->flags, 4445 mUnderrunCount, 4446 nowInUnderrun); 4447} 4448 4449// AudioBufferProvider interface 4450status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4451 AudioBufferProvider::Buffer* buffer, int64_t pts) 4452{ 4453 audio_track_cblk_t* cblk = this->cblk(); 4454 uint32_t framesReady; 4455 uint32_t framesReq = buffer->frameCount; 4456 4457 // Check if last stepServer failed, try to step now 4458 if (mStepServerFailed) { 4459 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4460 // Since the fast mixer is higher priority than client callback thread, 4461 // it does not result in priority inversion for client. 4462 // But a non-blocking solution would be preferable to avoid 4463 // fast mixer being unable to tryLock(), and 4464 // to avoid the extra context switches if the client wakes up, 4465 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4466 if (!step()) goto getNextBuffer_exit; 4467 ALOGV("stepServer recovered"); 4468 mStepServerFailed = false; 4469 } 4470 4471 // FIXME Same as above 4472 framesReady = cblk->framesReady(); 4473 4474 if (CC_LIKELY(framesReady)) { 4475 uint32_t s = cblk->server; 4476 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4477 4478 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4479 if (framesReq > framesReady) { 4480 framesReq = framesReady; 4481 } 4482 if (framesReq > bufferEnd - s) { 4483 framesReq = bufferEnd - s; 4484 } 4485 4486 buffer->raw = getBuffer(s, framesReq); 4487 buffer->frameCount = framesReq; 4488 return NO_ERROR; 4489 } 4490 4491getNextBuffer_exit: 4492 buffer->raw = NULL; 4493 buffer->frameCount = 0; 4494 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4495 return NOT_ENOUGH_DATA; 4496} 4497 4498// Note that framesReady() takes a mutex on the control block using tryLock(). 4499// This could result in priority inversion if framesReady() is called by the normal mixer, 4500// as the normal mixer thread runs at lower 4501// priority than the client's callback thread: there is a short window within framesReady() 4502// during which the normal mixer could be preempted, and the client callback would block. 4503// Another problem can occur if framesReady() is called by the fast mixer: 4504// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4505// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4506size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4507 return mCblk->framesReady(); 4508} 4509 4510// Don't call for fast tracks; the framesReady() could result in priority inversion 4511bool AudioFlinger::PlaybackThread::Track::isReady() const { 4512 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4513 4514 if (framesReady() >= mCblk->frameCount || 4515 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4516 mFillingUpStatus = FS_FILLED; 4517 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4518 return true; 4519 } 4520 return false; 4521} 4522 4523status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4524 int triggerSession) 4525{ 4526 status_t status = NO_ERROR; 4527 ALOGV("start(%d), calling pid %d session %d", 4528 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4529 4530 sp<ThreadBase> thread = mThread.promote(); 4531 if (thread != 0) { 4532 Mutex::Autolock _l(thread->mLock); 4533 track_state state = mState; 4534 // here the track could be either new, or restarted 4535 // in both cases "unstop" the track 4536 if (mState == PAUSED) { 4537 mState = TrackBase::RESUMING; 4538 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4539 } else { 4540 mState = TrackBase::ACTIVE; 4541 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4542 } 4543 4544 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4545 thread->mLock.unlock(); 4546 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4547 thread->mLock.lock(); 4548 4549#ifdef ADD_BATTERY_DATA 4550 // to track the speaker usage 4551 if (status == NO_ERROR) { 4552 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4553 } 4554#endif 4555 } 4556 if (status == NO_ERROR) { 4557 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4558 playbackThread->addTrack_l(this); 4559 } else { 4560 mState = state; 4561 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4562 } 4563 } else { 4564 status = BAD_VALUE; 4565 } 4566 return status; 4567} 4568 4569void AudioFlinger::PlaybackThread::Track::stop() 4570{ 4571 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4572 sp<ThreadBase> thread = mThread.promote(); 4573 if (thread != 0) { 4574 Mutex::Autolock _l(thread->mLock); 4575 track_state state = mState; 4576 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4577 // If the track is not active (PAUSED and buffers full), flush buffers 4578 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4579 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4580 reset(); 4581 mState = STOPPED; 4582 } else if (!isFastTrack()) { 4583 mState = STOPPED; 4584 } else { 4585 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4586 // and then to STOPPED and reset() when presentation is complete 4587 mState = STOPPING_1; 4588 } 4589 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4590 } 4591 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4592 thread->mLock.unlock(); 4593 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4594 thread->mLock.lock(); 4595 4596#ifdef ADD_BATTERY_DATA 4597 // to track the speaker usage 4598 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4599#endif 4600 } 4601 } 4602} 4603 4604void AudioFlinger::PlaybackThread::Track::pause() 4605{ 4606 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4607 sp<ThreadBase> thread = mThread.promote(); 4608 if (thread != 0) { 4609 Mutex::Autolock _l(thread->mLock); 4610 if (mState == ACTIVE || mState == RESUMING) { 4611 mState = PAUSING; 4612 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4613 if (!isOutputTrack()) { 4614 thread->mLock.unlock(); 4615 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4616 thread->mLock.lock(); 4617 4618#ifdef ADD_BATTERY_DATA 4619 // to track the speaker usage 4620 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4621#endif 4622 } 4623 } 4624 } 4625} 4626 4627void AudioFlinger::PlaybackThread::Track::flush() 4628{ 4629 ALOGV("flush(%d)", mName); 4630 sp<ThreadBase> thread = mThread.promote(); 4631 if (thread != 0) { 4632 Mutex::Autolock _l(thread->mLock); 4633 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4634 mState != PAUSING) { 4635 return; 4636 } 4637 // No point remaining in PAUSED state after a flush => go to 4638 // FLUSHED state 4639 mState = FLUSHED; 4640 // do not reset the track if it is still in the process of being stopped or paused. 4641 // this will be done by prepareTracks_l() when the track is stopped. 4642 // prepareTracks_l() will see mState == FLUSHED, then 4643 // remove from active track list, reset(), and trigger presentation complete 4644 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4645 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4646 reset(); 4647 } 4648 } 4649} 4650 4651void AudioFlinger::PlaybackThread::Track::reset() 4652{ 4653 // Do not reset twice to avoid discarding data written just after a flush and before 4654 // the audioflinger thread detects the track is stopped. 4655 if (!mResetDone) { 4656 TrackBase::reset(); 4657 // Force underrun condition to avoid false underrun callback until first data is 4658 // written to buffer 4659 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4660 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4661 mFillingUpStatus = FS_FILLING; 4662 mResetDone = true; 4663 if (mState == FLUSHED) { 4664 mState = IDLE; 4665 } 4666 } 4667} 4668 4669void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4670{ 4671 mMute = muted; 4672} 4673 4674status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4675{ 4676 status_t status = DEAD_OBJECT; 4677 sp<ThreadBase> thread = mThread.promote(); 4678 if (thread != 0) { 4679 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4680 sp<AudioFlinger> af = mClient->audioFlinger(); 4681 4682 Mutex::Autolock _l(af->mLock); 4683 4684 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4685 4686 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4687 Mutex::Autolock _dl(playbackThread->mLock); 4688 Mutex::Autolock _sl(srcThread->mLock); 4689 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4690 if (chain == 0) { 4691 return INVALID_OPERATION; 4692 } 4693 4694 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4695 if (effect == 0) { 4696 return INVALID_OPERATION; 4697 } 4698 srcThread->removeEffect_l(effect); 4699 playbackThread->addEffect_l(effect); 4700 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4701 if (effect->state() == EffectModule::ACTIVE || 4702 effect->state() == EffectModule::STOPPING) { 4703 effect->start(); 4704 } 4705 4706 sp<EffectChain> dstChain = effect->chain().promote(); 4707 if (dstChain == 0) { 4708 srcThread->addEffect_l(effect); 4709 return INVALID_OPERATION; 4710 } 4711 AudioSystem::unregisterEffect(effect->id()); 4712 AudioSystem::registerEffect(&effect->desc(), 4713 srcThread->id(), 4714 dstChain->strategy(), 4715 AUDIO_SESSION_OUTPUT_MIX, 4716 effect->id()); 4717 } 4718 status = playbackThread->attachAuxEffect(this, EffectId); 4719 } 4720 return status; 4721} 4722 4723void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4724{ 4725 mAuxEffectId = EffectId; 4726 mAuxBuffer = buffer; 4727} 4728 4729bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4730 size_t audioHalFrames) 4731{ 4732 // a track is considered presented when the total number of frames written to audio HAL 4733 // corresponds to the number of frames written when presentationComplete() is called for the 4734 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4735 if (mPresentationCompleteFrames == 0) { 4736 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4737 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4738 mPresentationCompleteFrames, audioHalFrames); 4739 } 4740 if (framesWritten >= mPresentationCompleteFrames) { 4741 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4742 mSessionId, framesWritten); 4743 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4744 return true; 4745 } 4746 return false; 4747} 4748 4749void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4750{ 4751 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4752 if (mSyncEvents[i]->type() == type) { 4753 mSyncEvents[i]->trigger(); 4754 mSyncEvents.removeAt(i); 4755 i--; 4756 } 4757 } 4758} 4759 4760// implement VolumeBufferProvider interface 4761 4762uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4763{ 4764 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4765 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4766 uint32_t vlr = mCblk->getVolumeLR(); 4767 uint32_t vl = vlr & 0xFFFF; 4768 uint32_t vr = vlr >> 16; 4769 // track volumes come from shared memory, so can't be trusted and must be clamped 4770 if (vl > MAX_GAIN_INT) { 4771 vl = MAX_GAIN_INT; 4772 } 4773 if (vr > MAX_GAIN_INT) { 4774 vr = MAX_GAIN_INT; 4775 } 4776 // now apply the cached master volume and stream type volume; 4777 // this is trusted but lacks any synchronization or barrier so may be stale 4778 float v = mCachedVolume; 4779 vl *= v; 4780 vr *= v; 4781 // re-combine into U4.16 4782 vlr = (vr << 16) | (vl & 0xFFFF); 4783 // FIXME look at mute, pause, and stop flags 4784 return vlr; 4785} 4786 4787status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4788{ 4789 if (mState == TERMINATED || mState == PAUSED || 4790 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4791 (mState == STOPPED)))) { 4792 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4793 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4794 event->cancel(); 4795 return INVALID_OPERATION; 4796 } 4797 (void) TrackBase::setSyncEvent(event); 4798 return NO_ERROR; 4799} 4800 4801// timed audio tracks 4802 4803sp<AudioFlinger::PlaybackThread::TimedTrack> 4804AudioFlinger::PlaybackThread::TimedTrack::create( 4805 PlaybackThread *thread, 4806 const sp<Client>& client, 4807 audio_stream_type_t streamType, 4808 uint32_t sampleRate, 4809 audio_format_t format, 4810 audio_channel_mask_t channelMask, 4811 int frameCount, 4812 const sp<IMemory>& sharedBuffer, 4813 int sessionId) { 4814 if (!client->reserveTimedTrack()) 4815 return 0; 4816 4817 return new TimedTrack( 4818 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4819 sharedBuffer, sessionId); 4820} 4821 4822AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4823 PlaybackThread *thread, 4824 const sp<Client>& client, 4825 audio_stream_type_t streamType, 4826 uint32_t sampleRate, 4827 audio_format_t format, 4828 audio_channel_mask_t channelMask, 4829 int frameCount, 4830 const sp<IMemory>& sharedBuffer, 4831 int sessionId) 4832 : Track(thread, client, streamType, sampleRate, format, channelMask, 4833 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4834 mQueueHeadInFlight(false), 4835 mTrimQueueHeadOnRelease(false), 4836 mFramesPendingInQueue(0), 4837 mTimedSilenceBuffer(NULL), 4838 mTimedSilenceBufferSize(0), 4839 mTimedAudioOutputOnTime(false), 4840 mMediaTimeTransformValid(false) 4841{ 4842 LocalClock lc; 4843 mLocalTimeFreq = lc.getLocalFreq(); 4844 4845 mLocalTimeToSampleTransform.a_zero = 0; 4846 mLocalTimeToSampleTransform.b_zero = 0; 4847 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4848 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4849 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4850 &mLocalTimeToSampleTransform.a_to_b_denom); 4851 4852 mMediaTimeToSampleTransform.a_zero = 0; 4853 mMediaTimeToSampleTransform.b_zero = 0; 4854 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4855 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4856 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4857 &mMediaTimeToSampleTransform.a_to_b_denom); 4858} 4859 4860AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4861 mClient->releaseTimedTrack(); 4862 delete [] mTimedSilenceBuffer; 4863} 4864 4865status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4866 size_t size, sp<IMemory>* buffer) { 4867 4868 Mutex::Autolock _l(mTimedBufferQueueLock); 4869 4870 trimTimedBufferQueue_l(); 4871 4872 // lazily initialize the shared memory heap for timed buffers 4873 if (mTimedMemoryDealer == NULL) { 4874 const int kTimedBufferHeapSize = 512 << 10; 4875 4876 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4877 "AudioFlingerTimed"); 4878 if (mTimedMemoryDealer == NULL) 4879 return NO_MEMORY; 4880 } 4881 4882 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4883 if (newBuffer == NULL) { 4884 newBuffer = mTimedMemoryDealer->allocate(size); 4885 if (newBuffer == NULL) 4886 return NO_MEMORY; 4887 } 4888 4889 *buffer = newBuffer; 4890 return NO_ERROR; 4891} 4892 4893// caller must hold mTimedBufferQueueLock 4894void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4895 int64_t mediaTimeNow; 4896 { 4897 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4898 if (!mMediaTimeTransformValid) 4899 return; 4900 4901 int64_t targetTimeNow; 4902 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4903 ? mCCHelper.getCommonTime(&targetTimeNow) 4904 : mCCHelper.getLocalTime(&targetTimeNow); 4905 4906 if (OK != res) 4907 return; 4908 4909 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4910 &mediaTimeNow)) { 4911 return; 4912 } 4913 } 4914 4915 size_t trimEnd; 4916 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4917 int64_t bufEnd; 4918 4919 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4920 // We have a next buffer. Just use its PTS as the PTS of the frame 4921 // following the last frame in this buffer. If the stream is sparse 4922 // (ie, there are deliberate gaps left in the stream which should be 4923 // filled with silence by the TimedAudioTrack), then this can result 4924 // in one extra buffer being left un-trimmed when it could have 4925 // been. In general, this is not typical, and we would rather 4926 // optimized away the TS calculation below for the more common case 4927 // where PTSes are contiguous. 4928 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4929 } else { 4930 // We have no next buffer. Compute the PTS of the frame following 4931 // the last frame in this buffer by computing the duration of of 4932 // this frame in media time units and adding it to the PTS of the 4933 // buffer. 4934 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4935 / mCblk->frameSize; 4936 4937 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4938 &bufEnd)) { 4939 ALOGE("Failed to convert frame count of %lld to media time" 4940 " duration" " (scale factor %d/%u) in %s", 4941 frameCount, 4942 mMediaTimeToSampleTransform.a_to_b_numer, 4943 mMediaTimeToSampleTransform.a_to_b_denom, 4944 __PRETTY_FUNCTION__); 4945 break; 4946 } 4947 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4948 } 4949 4950 if (bufEnd > mediaTimeNow) 4951 break; 4952 4953 // Is the buffer we want to use in the middle of a mix operation right 4954 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4955 // from the mixer which should be coming back shortly. 4956 if (!trimEnd && mQueueHeadInFlight) { 4957 mTrimQueueHeadOnRelease = true; 4958 } 4959 } 4960 4961 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4962 if (trimStart < trimEnd) { 4963 // Update the bookkeeping for framesReady() 4964 for (size_t i = trimStart; i < trimEnd; ++i) { 4965 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4966 } 4967 4968 // Now actually remove the buffers from the queue. 4969 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4970 } 4971} 4972 4973void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4974 const char* logTag) { 4975 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4976 "%s called (reason \"%s\"), but timed buffer queue has no" 4977 " elements to trim.", __FUNCTION__, logTag); 4978 4979 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4980 mTimedBufferQueue.removeAt(0); 4981} 4982 4983void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4984 const TimedBuffer& buf, 4985 const char* logTag) { 4986 uint32_t bufBytes = buf.buffer()->size(); 4987 uint32_t consumedAlready = buf.position(); 4988 4989 ALOG_ASSERT(consumedAlready <= bufBytes, 4990 "Bad bookkeeping while updating frames pending. Timed buffer is" 4991 " only %u bytes long, but claims to have consumed %u" 4992 " bytes. (update reason: \"%s\")", 4993 bufBytes, consumedAlready, logTag); 4994 4995 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4996 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4997 "Bad bookkeeping while updating frames pending. Should have at" 4998 " least %u queued frames, but we think we have only %u. (update" 4999 " reason: \"%s\")", 5000 bufFrames, mFramesPendingInQueue, logTag); 5001 5002 mFramesPendingInQueue -= bufFrames; 5003} 5004 5005status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5006 const sp<IMemory>& buffer, int64_t pts) { 5007 5008 { 5009 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5010 if (!mMediaTimeTransformValid) 5011 return INVALID_OPERATION; 5012 } 5013 5014 Mutex::Autolock _l(mTimedBufferQueueLock); 5015 5016 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5017 mFramesPendingInQueue += bufFrames; 5018 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5019 5020 return NO_ERROR; 5021} 5022 5023status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5024 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5025 5026 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5027 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5028 target); 5029 5030 if (!(target == TimedAudioTrack::LOCAL_TIME || 5031 target == TimedAudioTrack::COMMON_TIME)) { 5032 return BAD_VALUE; 5033 } 5034 5035 Mutex::Autolock lock(mMediaTimeTransformLock); 5036 mMediaTimeTransform = xform; 5037 mMediaTimeTransformTarget = target; 5038 mMediaTimeTransformValid = true; 5039 5040 return NO_ERROR; 5041} 5042 5043#define min(a, b) ((a) < (b) ? (a) : (b)) 5044 5045// implementation of getNextBuffer for tracks whose buffers have timestamps 5046status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5047 AudioBufferProvider::Buffer* buffer, int64_t pts) 5048{ 5049 if (pts == AudioBufferProvider::kInvalidPTS) { 5050 buffer->raw = NULL; 5051 buffer->frameCount = 0; 5052 mTimedAudioOutputOnTime = false; 5053 return INVALID_OPERATION; 5054 } 5055 5056 Mutex::Autolock _l(mTimedBufferQueueLock); 5057 5058 ALOG_ASSERT(!mQueueHeadInFlight, 5059 "getNextBuffer called without releaseBuffer!"); 5060 5061 while (true) { 5062 5063 // if we have no timed buffers, then fail 5064 if (mTimedBufferQueue.isEmpty()) { 5065 buffer->raw = NULL; 5066 buffer->frameCount = 0; 5067 return NOT_ENOUGH_DATA; 5068 } 5069 5070 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5071 5072 // calculate the PTS of the head of the timed buffer queue expressed in 5073 // local time 5074 int64_t headLocalPTS; 5075 { 5076 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5077 5078 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5079 5080 if (mMediaTimeTransform.a_to_b_denom == 0) { 5081 // the transform represents a pause, so yield silence 5082 timedYieldSilence_l(buffer->frameCount, buffer); 5083 return NO_ERROR; 5084 } 5085 5086 int64_t transformedPTS; 5087 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5088 &transformedPTS)) { 5089 // the transform failed. this shouldn't happen, but if it does 5090 // then just drop this buffer 5091 ALOGW("timedGetNextBuffer transform failed"); 5092 buffer->raw = NULL; 5093 buffer->frameCount = 0; 5094 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5095 return NO_ERROR; 5096 } 5097 5098 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5099 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5100 &headLocalPTS)) { 5101 buffer->raw = NULL; 5102 buffer->frameCount = 0; 5103 return INVALID_OPERATION; 5104 } 5105 } else { 5106 headLocalPTS = transformedPTS; 5107 } 5108 } 5109 5110 // adjust the head buffer's PTS to reflect the portion of the head buffer 5111 // that has already been consumed 5112 int64_t effectivePTS = headLocalPTS + 5113 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5114 5115 // Calculate the delta in samples between the head of the input buffer 5116 // queue and the start of the next output buffer that will be written. 5117 // If the transformation fails because of over or underflow, it means 5118 // that the sample's position in the output stream is so far out of 5119 // whack that it should just be dropped. 5120 int64_t sampleDelta; 5121 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5122 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5123 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5124 " mix"); 5125 continue; 5126 } 5127 if (!mLocalTimeToSampleTransform.doForwardTransform( 5128 (effectivePTS - pts) << 32, &sampleDelta)) { 5129 ALOGV("*** too late during sample rate transform: dropped buffer"); 5130 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5131 continue; 5132 } 5133 5134 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5135 " sampleDelta=[%d.%08x]", 5136 head.pts(), head.position(), pts, 5137 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5138 + (sampleDelta >> 32)), 5139 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5140 5141 // if the delta between the ideal placement for the next input sample and 5142 // the current output position is within this threshold, then we will 5143 // concatenate the next input samples to the previous output 5144 const int64_t kSampleContinuityThreshold = 5145 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5146 5147 // if this is the first buffer of audio that we're emitting from this track 5148 // then it should be almost exactly on time. 5149 const int64_t kSampleStartupThreshold = 1LL << 32; 5150 5151 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5152 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5153 // the next input is close enough to being on time, so concatenate it 5154 // with the last output 5155 timedYieldSamples_l(buffer); 5156 5157 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5158 head.position(), buffer->frameCount); 5159 return NO_ERROR; 5160 } 5161 5162 // Looks like our output is not on time. Reset our on timed status. 5163 // Next time we mix samples from our input queue, then should be within 5164 // the StartupThreshold. 5165 mTimedAudioOutputOnTime = false; 5166 if (sampleDelta > 0) { 5167 // the gap between the current output position and the proper start of 5168 // the next input sample is too big, so fill it with silence 5169 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5170 5171 timedYieldSilence_l(framesUntilNextInput, buffer); 5172 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5173 return NO_ERROR; 5174 } else { 5175 // the next input sample is late 5176 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5177 size_t onTimeSamplePosition = 5178 head.position() + lateFrames * mCblk->frameSize; 5179 5180 if (onTimeSamplePosition > head.buffer()->size()) { 5181 // all the remaining samples in the head are too late, so 5182 // drop it and move on 5183 ALOGV("*** too late: dropped buffer"); 5184 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5185 continue; 5186 } else { 5187 // skip over the late samples 5188 head.setPosition(onTimeSamplePosition); 5189 5190 // yield the available samples 5191 timedYieldSamples_l(buffer); 5192 5193 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5194 return NO_ERROR; 5195 } 5196 } 5197 } 5198} 5199 5200// Yield samples from the timed buffer queue head up to the given output 5201// buffer's capacity. 5202// 5203// Caller must hold mTimedBufferQueueLock 5204void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5205 AudioBufferProvider::Buffer* buffer) { 5206 5207 const TimedBuffer& head = mTimedBufferQueue[0]; 5208 5209 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5210 head.position()); 5211 5212 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5213 mCblk->frameSize); 5214 size_t framesRequested = buffer->frameCount; 5215 buffer->frameCount = min(framesLeftInHead, framesRequested); 5216 5217 mQueueHeadInFlight = true; 5218 mTimedAudioOutputOnTime = true; 5219} 5220 5221// Yield samples of silence up to the given output buffer's capacity 5222// 5223// Caller must hold mTimedBufferQueueLock 5224void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5225 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5226 5227 // lazily allocate a buffer filled with silence 5228 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5229 delete [] mTimedSilenceBuffer; 5230 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5231 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5232 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5233 } 5234 5235 buffer->raw = mTimedSilenceBuffer; 5236 size_t framesRequested = buffer->frameCount; 5237 buffer->frameCount = min(numFrames, framesRequested); 5238 5239 mTimedAudioOutputOnTime = false; 5240} 5241 5242// AudioBufferProvider interface 5243void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5244 AudioBufferProvider::Buffer* buffer) { 5245 5246 Mutex::Autolock _l(mTimedBufferQueueLock); 5247 5248 // If the buffer which was just released is part of the buffer at the head 5249 // of the queue, be sure to update the amt of the buffer which has been 5250 // consumed. If the buffer being returned is not part of the head of the 5251 // queue, its either because the buffer is part of the silence buffer, or 5252 // because the head of the timed queue was trimmed after the mixer called 5253 // getNextBuffer but before the mixer called releaseBuffer. 5254 if (buffer->raw == mTimedSilenceBuffer) { 5255 ALOG_ASSERT(!mQueueHeadInFlight, 5256 "Queue head in flight during release of silence buffer!"); 5257 goto done; 5258 } 5259 5260 ALOG_ASSERT(mQueueHeadInFlight, 5261 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5262 " head in flight."); 5263 5264 if (mTimedBufferQueue.size()) { 5265 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5266 5267 void* start = head.buffer()->pointer(); 5268 void* end = reinterpret_cast<void*>( 5269 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5270 + head.buffer()->size()); 5271 5272 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5273 "released buffer not within the head of the timed buffer" 5274 " queue; qHead = [%p, %p], released buffer = %p", 5275 start, end, buffer->raw); 5276 5277 head.setPosition(head.position() + 5278 (buffer->frameCount * mCblk->frameSize)); 5279 mQueueHeadInFlight = false; 5280 5281 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5282 "Bad bookkeeping during releaseBuffer! Should have at" 5283 " least %u queued frames, but we think we have only %u", 5284 buffer->frameCount, mFramesPendingInQueue); 5285 5286 mFramesPendingInQueue -= buffer->frameCount; 5287 5288 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5289 || mTrimQueueHeadOnRelease) { 5290 trimTimedBufferQueueHead_l("releaseBuffer"); 5291 mTrimQueueHeadOnRelease = false; 5292 } 5293 } else { 5294 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5295 " buffers in the timed buffer queue"); 5296 } 5297 5298done: 5299 buffer->raw = 0; 5300 buffer->frameCount = 0; 5301} 5302 5303size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5304 Mutex::Autolock _l(mTimedBufferQueueLock); 5305 return mFramesPendingInQueue; 5306} 5307 5308AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5309 : mPTS(0), mPosition(0) {} 5310 5311AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5312 const sp<IMemory>& buffer, int64_t pts) 5313 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5314 5315// ---------------------------------------------------------------------------- 5316 5317// RecordTrack constructor must be called with AudioFlinger::mLock held 5318AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5319 RecordThread *thread, 5320 const sp<Client>& client, 5321 uint32_t sampleRate, 5322 audio_format_t format, 5323 audio_channel_mask_t channelMask, 5324 int frameCount, 5325 int sessionId) 5326 : TrackBase(thread, client, sampleRate, format, 5327 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5328 mOverflow(false) 5329{ 5330 if (mCblk != NULL) { 5331 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5332 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5333 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5334 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5335 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5336 } else { 5337 mCblk->frameSize = sizeof(int8_t); 5338 } 5339 } 5340} 5341 5342AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5343{ 5344 ALOGV("%s", __func__); 5345} 5346 5347// AudioBufferProvider interface 5348status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5349{ 5350 audio_track_cblk_t* cblk = this->cblk(); 5351 uint32_t framesAvail; 5352 uint32_t framesReq = buffer->frameCount; 5353 5354 // Check if last stepServer failed, try to step now 5355 if (mStepServerFailed) { 5356 if (!step()) goto getNextBuffer_exit; 5357 ALOGV("stepServer recovered"); 5358 mStepServerFailed = false; 5359 } 5360 5361 framesAvail = cblk->framesAvailable_l(); 5362 5363 if (CC_LIKELY(framesAvail)) { 5364 uint32_t s = cblk->server; 5365 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5366 5367 if (framesReq > framesAvail) { 5368 framesReq = framesAvail; 5369 } 5370 if (framesReq > bufferEnd - s) { 5371 framesReq = bufferEnd - s; 5372 } 5373 5374 buffer->raw = getBuffer(s, framesReq); 5375 buffer->frameCount = framesReq; 5376 return NO_ERROR; 5377 } 5378 5379getNextBuffer_exit: 5380 buffer->raw = NULL; 5381 buffer->frameCount = 0; 5382 return NOT_ENOUGH_DATA; 5383} 5384 5385status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5386 int triggerSession) 5387{ 5388 sp<ThreadBase> thread = mThread.promote(); 5389 if (thread != 0) { 5390 RecordThread *recordThread = (RecordThread *)thread.get(); 5391 return recordThread->start(this, event, triggerSession); 5392 } else { 5393 return BAD_VALUE; 5394 } 5395} 5396 5397void AudioFlinger::RecordThread::RecordTrack::stop() 5398{ 5399 sp<ThreadBase> thread = mThread.promote(); 5400 if (thread != 0) { 5401 RecordThread *recordThread = (RecordThread *)thread.get(); 5402 recordThread->mLock.lock(); 5403 bool doStop = recordThread->stop_l(this); 5404 if (doStop) { 5405 TrackBase::reset(); 5406 // Force overrun condition to avoid false overrun callback until first data is 5407 // read from buffer 5408 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5409 } 5410 recordThread->mLock.unlock(); 5411 if (doStop) { 5412 AudioSystem::stopInput(recordThread->id()); 5413 } 5414 } 5415} 5416 5417/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5418{ 5419 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5420} 5421 5422void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5423{ 5424 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5425 (mClient == 0) ? getpid_cached : mClient->pid(), 5426 mFormat, 5427 mChannelMask, 5428 mSessionId, 5429 mFrameCount, 5430 mState, 5431 mCblk->sampleRate, 5432 mCblk->server, 5433 mCblk->user); 5434} 5435 5436 5437// ---------------------------------------------------------------------------- 5438 5439AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5440 PlaybackThread *playbackThread, 5441 DuplicatingThread *sourceThread, 5442 uint32_t sampleRate, 5443 audio_format_t format, 5444 audio_channel_mask_t channelMask, 5445 int frameCount) 5446 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5447 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5448 mActive(false), mSourceThread(sourceThread) 5449{ 5450 5451 if (mCblk != NULL) { 5452 mCblk->flags |= CBLK_DIRECTION_OUT; 5453 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5454 mOutBuffer.frameCount = 0; 5455 playbackThread->mTracks.add(this); 5456 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5457 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5458 mCblk, mBuffer, mCblk->buffers, 5459 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5460 } else { 5461 ALOGW("Error creating output track on thread %p", playbackThread); 5462 } 5463} 5464 5465AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5466{ 5467 clearBufferQueue(); 5468} 5469 5470status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5471 int triggerSession) 5472{ 5473 status_t status = Track::start(event, triggerSession); 5474 if (status != NO_ERROR) { 5475 return status; 5476 } 5477 5478 mActive = true; 5479 mRetryCount = 127; 5480 return status; 5481} 5482 5483void AudioFlinger::PlaybackThread::OutputTrack::stop() 5484{ 5485 Track::stop(); 5486 clearBufferQueue(); 5487 mOutBuffer.frameCount = 0; 5488 mActive = false; 5489} 5490 5491bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5492{ 5493 Buffer *pInBuffer; 5494 Buffer inBuffer; 5495 uint32_t channelCount = mChannelCount; 5496 bool outputBufferFull = false; 5497 inBuffer.frameCount = frames; 5498 inBuffer.i16 = data; 5499 5500 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5501 5502 if (!mActive && frames != 0) { 5503 start(); 5504 sp<ThreadBase> thread = mThread.promote(); 5505 if (thread != 0) { 5506 MixerThread *mixerThread = (MixerThread *)thread.get(); 5507 if (mCblk->frameCount > frames){ 5508 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5509 uint32_t startFrames = (mCblk->frameCount - frames); 5510 pInBuffer = new Buffer; 5511 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5512 pInBuffer->frameCount = startFrames; 5513 pInBuffer->i16 = pInBuffer->mBuffer; 5514 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5515 mBufferQueue.add(pInBuffer); 5516 } else { 5517 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5518 } 5519 } 5520 } 5521 } 5522 5523 while (waitTimeLeftMs) { 5524 // First write pending buffers, then new data 5525 if (mBufferQueue.size()) { 5526 pInBuffer = mBufferQueue.itemAt(0); 5527 } else { 5528 pInBuffer = &inBuffer; 5529 } 5530 5531 if (pInBuffer->frameCount == 0) { 5532 break; 5533 } 5534 5535 if (mOutBuffer.frameCount == 0) { 5536 mOutBuffer.frameCount = pInBuffer->frameCount; 5537 nsecs_t startTime = systemTime(); 5538 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5539 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5540 outputBufferFull = true; 5541 break; 5542 } 5543 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5544 if (waitTimeLeftMs >= waitTimeMs) { 5545 waitTimeLeftMs -= waitTimeMs; 5546 } else { 5547 waitTimeLeftMs = 0; 5548 } 5549 } 5550 5551 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5552 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5553 mCblk->stepUser(outFrames); 5554 pInBuffer->frameCount -= outFrames; 5555 pInBuffer->i16 += outFrames * channelCount; 5556 mOutBuffer.frameCount -= outFrames; 5557 mOutBuffer.i16 += outFrames * channelCount; 5558 5559 if (pInBuffer->frameCount == 0) { 5560 if (mBufferQueue.size()) { 5561 mBufferQueue.removeAt(0); 5562 delete [] pInBuffer->mBuffer; 5563 delete pInBuffer; 5564 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5565 } else { 5566 break; 5567 } 5568 } 5569 } 5570 5571 // If we could not write all frames, allocate a buffer and queue it for next time. 5572 if (inBuffer.frameCount) { 5573 sp<ThreadBase> thread = mThread.promote(); 5574 if (thread != 0 && !thread->standby()) { 5575 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5576 pInBuffer = new Buffer; 5577 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5578 pInBuffer->frameCount = inBuffer.frameCount; 5579 pInBuffer->i16 = pInBuffer->mBuffer; 5580 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5581 mBufferQueue.add(pInBuffer); 5582 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5583 } else { 5584 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5585 } 5586 } 5587 } 5588 5589 // Calling write() with a 0 length buffer, means that no more data will be written: 5590 // If no more buffers are pending, fill output track buffer to make sure it is started 5591 // by output mixer. 5592 if (frames == 0 && mBufferQueue.size() == 0) { 5593 if (mCblk->user < mCblk->frameCount) { 5594 frames = mCblk->frameCount - mCblk->user; 5595 pInBuffer = new Buffer; 5596 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5597 pInBuffer->frameCount = frames; 5598 pInBuffer->i16 = pInBuffer->mBuffer; 5599 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5600 mBufferQueue.add(pInBuffer); 5601 } else if (mActive) { 5602 stop(); 5603 } 5604 } 5605 5606 return outputBufferFull; 5607} 5608 5609status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5610{ 5611 int active; 5612 status_t result; 5613 audio_track_cblk_t* cblk = mCblk; 5614 uint32_t framesReq = buffer->frameCount; 5615 5616// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5617 buffer->frameCount = 0; 5618 5619 uint32_t framesAvail = cblk->framesAvailable(); 5620 5621 5622 if (framesAvail == 0) { 5623 Mutex::Autolock _l(cblk->lock); 5624 goto start_loop_here; 5625 while (framesAvail == 0) { 5626 active = mActive; 5627 if (CC_UNLIKELY(!active)) { 5628 ALOGV("Not active and NO_MORE_BUFFERS"); 5629 return NO_MORE_BUFFERS; 5630 } 5631 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5632 if (result != NO_ERROR) { 5633 return NO_MORE_BUFFERS; 5634 } 5635 // read the server count again 5636 start_loop_here: 5637 framesAvail = cblk->framesAvailable_l(); 5638 } 5639 } 5640 5641// if (framesAvail < framesReq) { 5642// return NO_MORE_BUFFERS; 5643// } 5644 5645 if (framesReq > framesAvail) { 5646 framesReq = framesAvail; 5647 } 5648 5649 uint32_t u = cblk->user; 5650 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5651 5652 if (framesReq > bufferEnd - u) { 5653 framesReq = bufferEnd - u; 5654 } 5655 5656 buffer->frameCount = framesReq; 5657 buffer->raw = (void *)cblk->buffer(u); 5658 return NO_ERROR; 5659} 5660 5661 5662void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5663{ 5664 size_t size = mBufferQueue.size(); 5665 5666 for (size_t i = 0; i < size; i++) { 5667 Buffer *pBuffer = mBufferQueue.itemAt(i); 5668 delete [] pBuffer->mBuffer; 5669 delete pBuffer; 5670 } 5671 mBufferQueue.clear(); 5672} 5673 5674// ---------------------------------------------------------------------------- 5675 5676AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5677 : RefBase(), 5678 mAudioFlinger(audioFlinger), 5679 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5680 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5681 mPid(pid), 5682 mTimedTrackCount(0) 5683{ 5684 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5685} 5686 5687// Client destructor must be called with AudioFlinger::mLock held 5688AudioFlinger::Client::~Client() 5689{ 5690 mAudioFlinger->removeClient_l(mPid); 5691} 5692 5693sp<MemoryDealer> AudioFlinger::Client::heap() const 5694{ 5695 return mMemoryDealer; 5696} 5697 5698// Reserve one of the limited slots for a timed audio track associated 5699// with this client 5700bool AudioFlinger::Client::reserveTimedTrack() 5701{ 5702 const int kMaxTimedTracksPerClient = 4; 5703 5704 Mutex::Autolock _l(mTimedTrackLock); 5705 5706 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5707 ALOGW("can not create timed track - pid %d has exceeded the limit", 5708 mPid); 5709 return false; 5710 } 5711 5712 mTimedTrackCount++; 5713 return true; 5714} 5715 5716// Release a slot for a timed audio track 5717void AudioFlinger::Client::releaseTimedTrack() 5718{ 5719 Mutex::Autolock _l(mTimedTrackLock); 5720 mTimedTrackCount--; 5721} 5722 5723// ---------------------------------------------------------------------------- 5724 5725AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5726 const sp<IAudioFlingerClient>& client, 5727 pid_t pid) 5728 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5729{ 5730} 5731 5732AudioFlinger::NotificationClient::~NotificationClient() 5733{ 5734} 5735 5736void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5737{ 5738 sp<NotificationClient> keep(this); 5739 mAudioFlinger->removeNotificationClient(mPid); 5740} 5741 5742// ---------------------------------------------------------------------------- 5743 5744AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5745 : BnAudioTrack(), 5746 mTrack(track) 5747{ 5748} 5749 5750AudioFlinger::TrackHandle::~TrackHandle() { 5751 // just stop the track on deletion, associated resources 5752 // will be freed from the main thread once all pending buffers have 5753 // been played. Unless it's not in the active track list, in which 5754 // case we free everything now... 5755 mTrack->destroy(); 5756} 5757 5758sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5759 return mTrack->getCblk(); 5760} 5761 5762status_t AudioFlinger::TrackHandle::start() { 5763 return mTrack->start(); 5764} 5765 5766void AudioFlinger::TrackHandle::stop() { 5767 mTrack->stop(); 5768} 5769 5770void AudioFlinger::TrackHandle::flush() { 5771 mTrack->flush(); 5772} 5773 5774void AudioFlinger::TrackHandle::mute(bool e) { 5775 mTrack->mute(e); 5776} 5777 5778void AudioFlinger::TrackHandle::pause() { 5779 mTrack->pause(); 5780} 5781 5782status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5783{ 5784 return mTrack->attachAuxEffect(EffectId); 5785} 5786 5787status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5788 sp<IMemory>* buffer) { 5789 if (!mTrack->isTimedTrack()) 5790 return INVALID_OPERATION; 5791 5792 PlaybackThread::TimedTrack* tt = 5793 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5794 return tt->allocateTimedBuffer(size, buffer); 5795} 5796 5797status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5798 int64_t pts) { 5799 if (!mTrack->isTimedTrack()) 5800 return INVALID_OPERATION; 5801 5802 PlaybackThread::TimedTrack* tt = 5803 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5804 return tt->queueTimedBuffer(buffer, pts); 5805} 5806 5807status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5808 const LinearTransform& xform, int target) { 5809 5810 if (!mTrack->isTimedTrack()) 5811 return INVALID_OPERATION; 5812 5813 PlaybackThread::TimedTrack* tt = 5814 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5815 return tt->setMediaTimeTransform( 5816 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5817} 5818 5819status_t AudioFlinger::TrackHandle::onTransact( 5820 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5821{ 5822 return BnAudioTrack::onTransact(code, data, reply, flags); 5823} 5824 5825// ---------------------------------------------------------------------------- 5826 5827sp<IAudioRecord> AudioFlinger::openRecord( 5828 pid_t pid, 5829 audio_io_handle_t input, 5830 uint32_t sampleRate, 5831 audio_format_t format, 5832 audio_channel_mask_t channelMask, 5833 int frameCount, 5834 IAudioFlinger::track_flags_t flags, 5835 pid_t tid, 5836 int *sessionId, 5837 status_t *status) 5838{ 5839 sp<RecordThread::RecordTrack> recordTrack; 5840 sp<RecordHandle> recordHandle; 5841 sp<Client> client; 5842 status_t lStatus; 5843 RecordThread *thread; 5844 size_t inFrameCount; 5845 int lSessionId; 5846 5847 // check calling permissions 5848 if (!recordingAllowed()) { 5849 lStatus = PERMISSION_DENIED; 5850 goto Exit; 5851 } 5852 5853 // add client to list 5854 { // scope for mLock 5855 Mutex::Autolock _l(mLock); 5856 thread = checkRecordThread_l(input); 5857 if (thread == NULL) { 5858 lStatus = BAD_VALUE; 5859 goto Exit; 5860 } 5861 5862 client = registerPid_l(pid); 5863 5864 // If no audio session id is provided, create one here 5865 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5866 lSessionId = *sessionId; 5867 } else { 5868 lSessionId = nextUniqueId(); 5869 if (sessionId != NULL) { 5870 *sessionId = lSessionId; 5871 } 5872 } 5873 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5874 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5875 frameCount, lSessionId, flags, tid, &lStatus); 5876 } 5877 if (lStatus != NO_ERROR) { 5878 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5879 // destructor is called by the TrackBase destructor with mLock held 5880 client.clear(); 5881 recordTrack.clear(); 5882 goto Exit; 5883 } 5884 5885 // return to handle to client 5886 recordHandle = new RecordHandle(recordTrack); 5887 lStatus = NO_ERROR; 5888 5889Exit: 5890 if (status) { 5891 *status = lStatus; 5892 } 5893 return recordHandle; 5894} 5895 5896// ---------------------------------------------------------------------------- 5897 5898AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5899 : BnAudioRecord(), 5900 mRecordTrack(recordTrack) 5901{ 5902} 5903 5904AudioFlinger::RecordHandle::~RecordHandle() { 5905 stop_nonvirtual(); 5906 mRecordTrack->destroy(); 5907} 5908 5909sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5910 return mRecordTrack->getCblk(); 5911} 5912 5913status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5914 ALOGV("RecordHandle::start()"); 5915 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5916} 5917 5918void AudioFlinger::RecordHandle::stop() { 5919 stop_nonvirtual(); 5920} 5921 5922void AudioFlinger::RecordHandle::stop_nonvirtual() { 5923 ALOGV("RecordHandle::stop()"); 5924 mRecordTrack->stop(); 5925} 5926 5927status_t AudioFlinger::RecordHandle::onTransact( 5928 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5929{ 5930 return BnAudioRecord::onTransact(code, data, reply, flags); 5931} 5932 5933// ---------------------------------------------------------------------------- 5934 5935AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5936 AudioStreamIn *input, 5937 uint32_t sampleRate, 5938 audio_channel_mask_t channelMask, 5939 audio_io_handle_t id, 5940 audio_devices_t device) : 5941 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 5942 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5943 // mRsmpInIndex and mInputBytes set by readInputParameters() 5944 mReqChannelCount(popcount(channelMask)), 5945 mReqSampleRate(sampleRate) 5946 // mBytesRead is only meaningful while active, and so is cleared in start() 5947 // (but might be better to also clear here for dump?) 5948{ 5949 snprintf(mName, kNameLength, "AudioIn_%X", id); 5950 5951 readInputParameters(); 5952} 5953 5954 5955AudioFlinger::RecordThread::~RecordThread() 5956{ 5957 delete[] mRsmpInBuffer; 5958 delete mResampler; 5959 delete[] mRsmpOutBuffer; 5960} 5961 5962void AudioFlinger::RecordThread::onFirstRef() 5963{ 5964 run(mName, PRIORITY_URGENT_AUDIO); 5965} 5966 5967status_t AudioFlinger::RecordThread::readyToRun() 5968{ 5969 status_t status = initCheck(); 5970 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5971 return status; 5972} 5973 5974bool AudioFlinger::RecordThread::threadLoop() 5975{ 5976 AudioBufferProvider::Buffer buffer; 5977 sp<RecordTrack> activeTrack; 5978 Vector< sp<EffectChain> > effectChains; 5979 5980 nsecs_t lastWarning = 0; 5981 5982 inputStandBy(); 5983 acquireWakeLock(); 5984 5985 // start recording 5986 while (!exitPending()) { 5987 5988 processConfigEvents(); 5989 5990 { // scope for mLock 5991 Mutex::Autolock _l(mLock); 5992 checkForNewParameters_l(); 5993 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5994 standby(); 5995 5996 if (exitPending()) break; 5997 5998 releaseWakeLock_l(); 5999 ALOGV("RecordThread: loop stopping"); 6000 // go to sleep 6001 mWaitWorkCV.wait(mLock); 6002 ALOGV("RecordThread: loop starting"); 6003 acquireWakeLock_l(); 6004 continue; 6005 } 6006 if (mActiveTrack != 0) { 6007 if (mActiveTrack->mState == TrackBase::PAUSING) { 6008 standby(); 6009 mActiveTrack.clear(); 6010 mStartStopCond.broadcast(); 6011 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6012 if (mReqChannelCount != mActiveTrack->channelCount()) { 6013 mActiveTrack.clear(); 6014 mStartStopCond.broadcast(); 6015 } else if (mBytesRead != 0) { 6016 // record start succeeds only if first read from audio input 6017 // succeeds 6018 if (mBytesRead > 0) { 6019 mActiveTrack->mState = TrackBase::ACTIVE; 6020 } else { 6021 mActiveTrack.clear(); 6022 } 6023 mStartStopCond.broadcast(); 6024 } 6025 mStandby = false; 6026 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6027 removeTrack_l(mActiveTrack); 6028 mActiveTrack.clear(); 6029 } 6030 } 6031 lockEffectChains_l(effectChains); 6032 } 6033 6034 if (mActiveTrack != 0) { 6035 if (mActiveTrack->mState != TrackBase::ACTIVE && 6036 mActiveTrack->mState != TrackBase::RESUMING) { 6037 unlockEffectChains(effectChains); 6038 usleep(kRecordThreadSleepUs); 6039 continue; 6040 } 6041 for (size_t i = 0; i < effectChains.size(); i ++) { 6042 effectChains[i]->process_l(); 6043 } 6044 6045 buffer.frameCount = mFrameCount; 6046 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6047 size_t framesOut = buffer.frameCount; 6048 if (mResampler == NULL) { 6049 // no resampling 6050 while (framesOut) { 6051 size_t framesIn = mFrameCount - mRsmpInIndex; 6052 if (framesIn) { 6053 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6054 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6055 if (framesIn > framesOut) 6056 framesIn = framesOut; 6057 mRsmpInIndex += framesIn; 6058 framesOut -= framesIn; 6059 if ((int)mChannelCount == mReqChannelCount || 6060 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6061 memcpy(dst, src, framesIn * mFrameSize); 6062 } else { 6063 if (mChannelCount == 1) { 6064 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6065 (int16_t *)src, framesIn); 6066 } else { 6067 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6068 (int16_t *)src, framesIn); 6069 } 6070 } 6071 } 6072 if (framesOut && mFrameCount == mRsmpInIndex) { 6073 if (framesOut == mFrameCount && 6074 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6075 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6076 framesOut = 0; 6077 } else { 6078 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6079 mRsmpInIndex = 0; 6080 } 6081 if (mBytesRead < 0) { 6082 ALOGE("Error reading audio input"); 6083 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6084 // Force input into standby so that it tries to 6085 // recover at next read attempt 6086 inputStandBy(); 6087 usleep(kRecordThreadSleepUs); 6088 } 6089 mRsmpInIndex = mFrameCount; 6090 framesOut = 0; 6091 buffer.frameCount = 0; 6092 } 6093 } 6094 } 6095 } else { 6096 // resampling 6097 6098 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6099 // alter output frame count as if we were expecting stereo samples 6100 if (mChannelCount == 1 && mReqChannelCount == 1) { 6101 framesOut >>= 1; 6102 } 6103 mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); 6104 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6105 // are 32 bit aligned which should be always true. 6106 if (mChannelCount == 2 && mReqChannelCount == 1) { 6107 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6108 // the resampler always outputs stereo samples: do post stereo to mono conversion 6109 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6110 framesOut); 6111 } else { 6112 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6113 } 6114 6115 } 6116 if (mFramestoDrop == 0) { 6117 mActiveTrack->releaseBuffer(&buffer); 6118 } else { 6119 if (mFramestoDrop > 0) { 6120 mFramestoDrop -= buffer.frameCount; 6121 if (mFramestoDrop <= 0) { 6122 clearSyncStartEvent(); 6123 } 6124 } else { 6125 mFramestoDrop += buffer.frameCount; 6126 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6127 mSyncStartEvent->isCancelled()) { 6128 ALOGW("Synced record %s, session %d, trigger session %d", 6129 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6130 mActiveTrack->sessionId(), 6131 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6132 clearSyncStartEvent(); 6133 } 6134 } 6135 } 6136 mActiveTrack->clearOverflow(); 6137 } 6138 // client isn't retrieving buffers fast enough 6139 else { 6140 if (!mActiveTrack->setOverflow()) { 6141 nsecs_t now = systemTime(); 6142 if ((now - lastWarning) > kWarningThrottleNs) { 6143 ALOGW("RecordThread: buffer overflow"); 6144 lastWarning = now; 6145 } 6146 } 6147 // Release the processor for a while before asking for a new buffer. 6148 // This will give the application more chance to read from the buffer and 6149 // clear the overflow. 6150 usleep(kRecordThreadSleepUs); 6151 } 6152 } 6153 // enable changes in effect chain 6154 unlockEffectChains(effectChains); 6155 effectChains.clear(); 6156 } 6157 6158 standby(); 6159 6160 { 6161 Mutex::Autolock _l(mLock); 6162 mActiveTrack.clear(); 6163 mStartStopCond.broadcast(); 6164 } 6165 6166 releaseWakeLock(); 6167 6168 ALOGV("RecordThread %p exiting", this); 6169 return false; 6170} 6171 6172void AudioFlinger::RecordThread::standby() 6173{ 6174 if (!mStandby) { 6175 inputStandBy(); 6176 mStandby = true; 6177 } 6178} 6179 6180void AudioFlinger::RecordThread::inputStandBy() 6181{ 6182 mInput->stream->common.standby(&mInput->stream->common); 6183} 6184 6185sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6186 const sp<AudioFlinger::Client>& client, 6187 uint32_t sampleRate, 6188 audio_format_t format, 6189 audio_channel_mask_t channelMask, 6190 int frameCount, 6191 int sessionId, 6192 IAudioFlinger::track_flags_t flags, 6193 pid_t tid, 6194 status_t *status) 6195{ 6196 sp<RecordTrack> track; 6197 status_t lStatus; 6198 6199 lStatus = initCheck(); 6200 if (lStatus != NO_ERROR) { 6201 ALOGE("Audio driver not initialized."); 6202 goto Exit; 6203 } 6204 6205 // FIXME use flags and tid similar to createTrack_l() 6206 6207 { // scope for mLock 6208 Mutex::Autolock _l(mLock); 6209 6210 track = new RecordTrack(this, client, sampleRate, 6211 format, channelMask, frameCount, sessionId); 6212 6213 if (track->getCblk() == 0) { 6214 lStatus = NO_MEMORY; 6215 goto Exit; 6216 } 6217 mTracks.add(track); 6218 6219 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6220 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6221 mAudioFlinger->btNrecIsOff(); 6222 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6223 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6224 } 6225 lStatus = NO_ERROR; 6226 6227Exit: 6228 if (status) { 6229 *status = lStatus; 6230 } 6231 return track; 6232} 6233 6234status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6235 AudioSystem::sync_event_t event, 6236 int triggerSession) 6237{ 6238 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6239 sp<ThreadBase> strongMe = this; 6240 status_t status = NO_ERROR; 6241 6242 if (event == AudioSystem::SYNC_EVENT_NONE) { 6243 clearSyncStartEvent(); 6244 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6245 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6246 triggerSession, 6247 recordTrack->sessionId(), 6248 syncStartEventCallback, 6249 this); 6250 // Sync event can be cancelled by the trigger session if the track is not in a 6251 // compatible state in which case we start record immediately 6252 if (mSyncStartEvent->isCancelled()) { 6253 clearSyncStartEvent(); 6254 } else { 6255 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6256 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6257 } 6258 } 6259 6260 { 6261 AutoMutex lock(mLock); 6262 if (mActiveTrack != 0) { 6263 if (recordTrack != mActiveTrack.get()) { 6264 status = -EBUSY; 6265 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6266 mActiveTrack->mState = TrackBase::ACTIVE; 6267 } 6268 return status; 6269 } 6270 6271 recordTrack->mState = TrackBase::IDLE; 6272 mActiveTrack = recordTrack; 6273 mLock.unlock(); 6274 status_t status = AudioSystem::startInput(mId); 6275 mLock.lock(); 6276 if (status != NO_ERROR) { 6277 mActiveTrack.clear(); 6278 clearSyncStartEvent(); 6279 return status; 6280 } 6281 mRsmpInIndex = mFrameCount; 6282 mBytesRead = 0; 6283 if (mResampler != NULL) { 6284 mResampler->reset(); 6285 } 6286 mActiveTrack->mState = TrackBase::RESUMING; 6287 // signal thread to start 6288 ALOGV("Signal record thread"); 6289 mWaitWorkCV.signal(); 6290 // do not wait for mStartStopCond if exiting 6291 if (exitPending()) { 6292 mActiveTrack.clear(); 6293 status = INVALID_OPERATION; 6294 goto startError; 6295 } 6296 mStartStopCond.wait(mLock); 6297 if (mActiveTrack == 0) { 6298 ALOGV("Record failed to start"); 6299 status = BAD_VALUE; 6300 goto startError; 6301 } 6302 ALOGV("Record started OK"); 6303 return status; 6304 } 6305startError: 6306 AudioSystem::stopInput(mId); 6307 clearSyncStartEvent(); 6308 return status; 6309} 6310 6311void AudioFlinger::RecordThread::clearSyncStartEvent() 6312{ 6313 if (mSyncStartEvent != 0) { 6314 mSyncStartEvent->cancel(); 6315 } 6316 mSyncStartEvent.clear(); 6317 mFramestoDrop = 0; 6318} 6319 6320void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6321{ 6322 sp<SyncEvent> strongEvent = event.promote(); 6323 6324 if (strongEvent != 0) { 6325 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6326 me->handleSyncStartEvent(strongEvent); 6327 } 6328} 6329 6330void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6331{ 6332 if (event == mSyncStartEvent) { 6333 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6334 // from audio HAL 6335 mFramestoDrop = mFrameCount * 2; 6336 } 6337} 6338 6339bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6340 ALOGV("RecordThread::stop"); 6341 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6342 return false; 6343 } 6344 recordTrack->mState = TrackBase::PAUSING; 6345 // do not wait for mStartStopCond if exiting 6346 if (exitPending()) { 6347 return true; 6348 } 6349 mStartStopCond.wait(mLock); 6350 // if we have been restarted, recordTrack == mActiveTrack.get() here 6351 if (exitPending() || recordTrack != mActiveTrack.get()) { 6352 ALOGV("Record stopped OK"); 6353 return true; 6354 } 6355 return false; 6356} 6357 6358bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6359{ 6360 return false; 6361} 6362 6363status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6364{ 6365#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6366 if (!isValidSyncEvent(event)) { 6367 return BAD_VALUE; 6368 } 6369 6370 int eventSession = event->triggerSession(); 6371 status_t ret = NAME_NOT_FOUND; 6372 6373 Mutex::Autolock _l(mLock); 6374 6375 for (size_t i = 0; i < mTracks.size(); i++) { 6376 sp<RecordTrack> track = mTracks[i]; 6377 if (eventSession == track->sessionId()) { 6378 (void) track->setSyncEvent(event); 6379 ret = NO_ERROR; 6380 } 6381 } 6382 return ret; 6383#else 6384 return BAD_VALUE; 6385#endif 6386} 6387 6388void AudioFlinger::RecordThread::RecordTrack::destroy() 6389{ 6390 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6391 sp<RecordTrack> keep(this); 6392 { 6393 sp<ThreadBase> thread = mThread.promote(); 6394 if (thread != 0) { 6395 if (mState == ACTIVE || mState == RESUMING) { 6396 AudioSystem::stopInput(thread->id()); 6397 } 6398 AudioSystem::releaseInput(thread->id()); 6399 Mutex::Autolock _l(thread->mLock); 6400 RecordThread *recordThread = (RecordThread *) thread.get(); 6401 recordThread->destroyTrack_l(this); 6402 } 6403 } 6404} 6405 6406// destroyTrack_l() must be called with ThreadBase::mLock held 6407void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6408{ 6409 track->mState = TrackBase::TERMINATED; 6410 // active tracks are removed by threadLoop() 6411 if (mActiveTrack != track) { 6412 removeTrack_l(track); 6413 } 6414} 6415 6416void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6417{ 6418 mTracks.remove(track); 6419 // need anything related to effects here? 6420} 6421 6422void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6423{ 6424 dumpInternals(fd, args); 6425 dumpTracks(fd, args); 6426 dumpEffectChains(fd, args); 6427} 6428 6429void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6430{ 6431 const size_t SIZE = 256; 6432 char buffer[SIZE]; 6433 String8 result; 6434 6435 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6436 result.append(buffer); 6437 6438 if (mActiveTrack != 0) { 6439 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6440 result.append(buffer); 6441 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6442 result.append(buffer); 6443 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6444 result.append(buffer); 6445 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6446 result.append(buffer); 6447 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6448 result.append(buffer); 6449 } else { 6450 result.append("No active record client\n"); 6451 } 6452 6453 write(fd, result.string(), result.size()); 6454 6455 dumpBase(fd, args); 6456} 6457 6458void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6459{ 6460 const size_t SIZE = 256; 6461 char buffer[SIZE]; 6462 String8 result; 6463 6464 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6465 result.append(buffer); 6466 RecordTrack::appendDumpHeader(result); 6467 for (size_t i = 0; i < mTracks.size(); ++i) { 6468 sp<RecordTrack> track = mTracks[i]; 6469 if (track != 0) { 6470 track->dump(buffer, SIZE); 6471 result.append(buffer); 6472 } 6473 } 6474 6475 if (mActiveTrack != 0) { 6476 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6477 result.append(buffer); 6478 RecordTrack::appendDumpHeader(result); 6479 mActiveTrack->dump(buffer, SIZE); 6480 result.append(buffer); 6481 6482 } 6483 write(fd, result.string(), result.size()); 6484} 6485 6486// AudioBufferProvider interface 6487status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6488{ 6489 size_t framesReq = buffer->frameCount; 6490 size_t framesReady = mFrameCount - mRsmpInIndex; 6491 int channelCount; 6492 6493 if (framesReady == 0) { 6494 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6495 if (mBytesRead < 0) { 6496 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6497 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6498 // Force input into standby so that it tries to 6499 // recover at next read attempt 6500 inputStandBy(); 6501 usleep(kRecordThreadSleepUs); 6502 } 6503 buffer->raw = NULL; 6504 buffer->frameCount = 0; 6505 return NOT_ENOUGH_DATA; 6506 } 6507 mRsmpInIndex = 0; 6508 framesReady = mFrameCount; 6509 } 6510 6511 if (framesReq > framesReady) { 6512 framesReq = framesReady; 6513 } 6514 6515 if (mChannelCount == 1 && mReqChannelCount == 2) { 6516 channelCount = 1; 6517 } else { 6518 channelCount = 2; 6519 } 6520 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6521 buffer->frameCount = framesReq; 6522 return NO_ERROR; 6523} 6524 6525// AudioBufferProvider interface 6526void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6527{ 6528 mRsmpInIndex += buffer->frameCount; 6529 buffer->frameCount = 0; 6530} 6531 6532bool AudioFlinger::RecordThread::checkForNewParameters_l() 6533{ 6534 bool reconfig = false; 6535 6536 while (!mNewParameters.isEmpty()) { 6537 status_t status = NO_ERROR; 6538 String8 keyValuePair = mNewParameters[0]; 6539 AudioParameter param = AudioParameter(keyValuePair); 6540 int value; 6541 audio_format_t reqFormat = mFormat; 6542 int reqSamplingRate = mReqSampleRate; 6543 int reqChannelCount = mReqChannelCount; 6544 6545 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6546 reqSamplingRate = value; 6547 reconfig = true; 6548 } 6549 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6550 reqFormat = (audio_format_t) value; 6551 reconfig = true; 6552 } 6553 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6554 reqChannelCount = popcount(value); 6555 reconfig = true; 6556 } 6557 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6558 // do not accept frame count changes if tracks are open as the track buffer 6559 // size depends on frame count and correct behavior would not be guaranteed 6560 // if frame count is changed after track creation 6561 if (mActiveTrack != 0) { 6562 status = INVALID_OPERATION; 6563 } else { 6564 reconfig = true; 6565 } 6566 } 6567 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6568 // forward device change to effects that have requested to be 6569 // aware of attached audio device. 6570 for (size_t i = 0; i < mEffectChains.size(); i++) { 6571 mEffectChains[i]->setDevice_l(value); 6572 } 6573 6574 // store input device and output device but do not forward output device to audio HAL. 6575 // Note that status is ignored by the caller for output device 6576 // (see AudioFlinger::setParameters() 6577 if (audio_is_output_devices(value)) { 6578 mOutDevice = value; 6579 status = BAD_VALUE; 6580 } else { 6581 mInDevice = value; 6582 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6583 if (mTracks.size() > 0) { 6584 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6585 mAudioFlinger->btNrecIsOff(); 6586 for (size_t i = 0; i < mTracks.size(); i++) { 6587 sp<RecordTrack> track = mTracks[i]; 6588 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6589 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6590 } 6591 } 6592 } 6593 } 6594 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6595 mAudioSource != (audio_source_t)value) { 6596 // forward device change to effects that have requested to be 6597 // aware of attached audio device. 6598 for (size_t i = 0; i < mEffectChains.size(); i++) { 6599 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6600 } 6601 mAudioSource = (audio_source_t)value; 6602 } 6603 if (status == NO_ERROR) { 6604 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6605 if (status == INVALID_OPERATION) { 6606 inputStandBy(); 6607 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6608 keyValuePair.string()); 6609 } 6610 if (reconfig) { 6611 if (status == BAD_VALUE && 6612 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6613 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6614 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6615 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6616 (reqChannelCount <= FCC_2)) { 6617 status = NO_ERROR; 6618 } 6619 if (status == NO_ERROR) { 6620 readInputParameters(); 6621 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6622 } 6623 } 6624 } 6625 6626 mNewParameters.removeAt(0); 6627 6628 mParamStatus = status; 6629 mParamCond.signal(); 6630 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6631 // already timed out waiting for the status and will never signal the condition. 6632 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6633 } 6634 return reconfig; 6635} 6636 6637String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6638{ 6639 char *s; 6640 String8 out_s8 = String8(); 6641 6642 Mutex::Autolock _l(mLock); 6643 if (initCheck() != NO_ERROR) { 6644 return out_s8; 6645 } 6646 6647 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6648 out_s8 = String8(s); 6649 free(s); 6650 return out_s8; 6651} 6652 6653void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6654 AudioSystem::OutputDescriptor desc; 6655 void *param2 = NULL; 6656 6657 switch (event) { 6658 case AudioSystem::INPUT_OPENED: 6659 case AudioSystem::INPUT_CONFIG_CHANGED: 6660 desc.channels = mChannelMask; 6661 desc.samplingRate = mSampleRate; 6662 desc.format = mFormat; 6663 desc.frameCount = mFrameCount; 6664 desc.latency = 0; 6665 param2 = &desc; 6666 break; 6667 6668 case AudioSystem::INPUT_CLOSED: 6669 default: 6670 break; 6671 } 6672 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6673} 6674 6675void AudioFlinger::RecordThread::readInputParameters() 6676{ 6677 delete mRsmpInBuffer; 6678 // mRsmpInBuffer is always assigned a new[] below 6679 delete mRsmpOutBuffer; 6680 mRsmpOutBuffer = NULL; 6681 delete mResampler; 6682 mResampler = NULL; 6683 6684 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6685 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6686 mChannelCount = (uint16_t)popcount(mChannelMask); 6687 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6688 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6689 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6690 mFrameCount = mInputBytes / mFrameSize; 6691 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6692 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6693 6694 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6695 { 6696 int channelCount; 6697 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6698 // stereo to mono post process as the resampler always outputs stereo. 6699 if (mChannelCount == 1 && mReqChannelCount == 2) { 6700 channelCount = 1; 6701 } else { 6702 channelCount = 2; 6703 } 6704 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6705 mResampler->setSampleRate(mSampleRate); 6706 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6707 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6708 6709 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6710 if (mChannelCount == 1 && mReqChannelCount == 1) { 6711 mFrameCount >>= 1; 6712 } 6713 6714 } 6715 mRsmpInIndex = mFrameCount; 6716} 6717 6718unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6719{ 6720 Mutex::Autolock _l(mLock); 6721 if (initCheck() != NO_ERROR) { 6722 return 0; 6723 } 6724 6725 return mInput->stream->get_input_frames_lost(mInput->stream); 6726} 6727 6728uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6729{ 6730 Mutex::Autolock _l(mLock); 6731 uint32_t result = 0; 6732 if (getEffectChain_l(sessionId) != 0) { 6733 result = EFFECT_SESSION; 6734 } 6735 6736 for (size_t i = 0; i < mTracks.size(); ++i) { 6737 if (sessionId == mTracks[i]->sessionId()) { 6738 result |= TRACK_SESSION; 6739 break; 6740 } 6741 } 6742 6743 return result; 6744} 6745 6746KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6747{ 6748 KeyedVector<int, bool> ids; 6749 Mutex::Autolock _l(mLock); 6750 for (size_t j = 0; j < mTracks.size(); ++j) { 6751 sp<RecordThread::RecordTrack> track = mTracks[j]; 6752 int sessionId = track->sessionId(); 6753 if (ids.indexOfKey(sessionId) < 0) { 6754 ids.add(sessionId, true); 6755 } 6756 } 6757 return ids; 6758} 6759 6760AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6761{ 6762 Mutex::Autolock _l(mLock); 6763 AudioStreamIn *input = mInput; 6764 mInput = NULL; 6765 return input; 6766} 6767 6768// this method must always be called either with ThreadBase mLock held or inside the thread loop 6769audio_stream_t* AudioFlinger::RecordThread::stream() const 6770{ 6771 if (mInput == NULL) { 6772 return NULL; 6773 } 6774 return &mInput->stream->common; 6775} 6776 6777 6778// ---------------------------------------------------------------------------- 6779 6780audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6781{ 6782 if (!settingsAllowed()) { 6783 return 0; 6784 } 6785 Mutex::Autolock _l(mLock); 6786 return loadHwModule_l(name); 6787} 6788 6789// loadHwModule_l() must be called with AudioFlinger::mLock held 6790audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6791{ 6792 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6793 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6794 ALOGW("loadHwModule() module %s already loaded", name); 6795 return mAudioHwDevs.keyAt(i); 6796 } 6797 } 6798 6799 audio_hw_device_t *dev; 6800 6801 int rc = load_audio_interface(name, &dev); 6802 if (rc) { 6803 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6804 return 0; 6805 } 6806 6807 mHardwareStatus = AUDIO_HW_INIT; 6808 rc = dev->init_check(dev); 6809 mHardwareStatus = AUDIO_HW_IDLE; 6810 if (rc) { 6811 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6812 return 0; 6813 } 6814 6815 // Check and cache this HAL's level of support for master mute and master 6816 // volume. If this is the first HAL opened, and it supports the get 6817 // methods, use the initial values provided by the HAL as the current 6818 // master mute and volume settings. 6819 6820 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6821 { // scope for auto-lock pattern 6822 AutoMutex lock(mHardwareLock); 6823 6824 if (0 == mAudioHwDevs.size()) { 6825 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6826 if (NULL != dev->get_master_volume) { 6827 float mv; 6828 if (OK == dev->get_master_volume(dev, &mv)) { 6829 mMasterVolume = mv; 6830 } 6831 } 6832 6833 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6834 if (NULL != dev->get_master_mute) { 6835 bool mm; 6836 if (OK == dev->get_master_mute(dev, &mm)) { 6837 mMasterMute = mm; 6838 } 6839 } 6840 } 6841 6842 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6843 if ((NULL != dev->set_master_volume) && 6844 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6845 flags = static_cast<AudioHwDevice::Flags>(flags | 6846 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6847 } 6848 6849 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6850 if ((NULL != dev->set_master_mute) && 6851 (OK == dev->set_master_mute(dev, mMasterMute))) { 6852 flags = static_cast<AudioHwDevice::Flags>(flags | 6853 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6854 } 6855 6856 mHardwareStatus = AUDIO_HW_IDLE; 6857 } 6858 6859 audio_module_handle_t handle = nextUniqueId(); 6860 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6861 6862 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6863 name, dev->common.module->name, dev->common.module->id, handle); 6864 6865 return handle; 6866 6867} 6868 6869audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6870 audio_devices_t *pDevices, 6871 uint32_t *pSamplingRate, 6872 audio_format_t *pFormat, 6873 audio_channel_mask_t *pChannelMask, 6874 uint32_t *pLatencyMs, 6875 audio_output_flags_t flags) 6876{ 6877 status_t status; 6878 PlaybackThread *thread = NULL; 6879 struct audio_config config = { 6880 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6881 channel_mask: pChannelMask ? *pChannelMask : 0, 6882 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6883 }; 6884 audio_stream_out_t *outStream = NULL; 6885 AudioHwDevice *outHwDev; 6886 6887 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6888 module, 6889 (pDevices != NULL) ? *pDevices : 0, 6890 config.sample_rate, 6891 config.format, 6892 config.channel_mask, 6893 flags); 6894 6895 if (pDevices == NULL || *pDevices == 0) { 6896 return 0; 6897 } 6898 6899 Mutex::Autolock _l(mLock); 6900 6901 outHwDev = findSuitableHwDev_l(module, *pDevices); 6902 if (outHwDev == NULL) 6903 return 0; 6904 6905 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 6906 audio_io_handle_t id = nextUniqueId(); 6907 6908 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6909 6910 status = hwDevHal->open_output_stream(hwDevHal, 6911 id, 6912 *pDevices, 6913 (audio_output_flags_t)flags, 6914 &config, 6915 &outStream); 6916 6917 mHardwareStatus = AUDIO_HW_IDLE; 6918 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6919 outStream, 6920 config.sample_rate, 6921 config.format, 6922 config.channel_mask, 6923 status); 6924 6925 if (status == NO_ERROR && outStream != NULL) { 6926 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6927 6928 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6929 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6930 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6931 thread = new DirectOutputThread(this, output, id, *pDevices); 6932 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6933 } else { 6934 thread = new MixerThread(this, output, id, *pDevices); 6935 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6936 } 6937 mPlaybackThreads.add(id, thread); 6938 6939 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6940 if (pFormat != NULL) *pFormat = config.format; 6941 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6942 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6943 6944 // notify client processes of the new output creation 6945 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6946 6947 // the first primary output opened designates the primary hw device 6948 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6949 ALOGI("Using module %d has the primary audio interface", module); 6950 mPrimaryHardwareDev = outHwDev; 6951 6952 AutoMutex lock(mHardwareLock); 6953 mHardwareStatus = AUDIO_HW_SET_MODE; 6954 hwDevHal->set_mode(hwDevHal, mMode); 6955 mHardwareStatus = AUDIO_HW_IDLE; 6956 } 6957 return id; 6958 } 6959 6960 return 0; 6961} 6962 6963audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6964 audio_io_handle_t output2) 6965{ 6966 Mutex::Autolock _l(mLock); 6967 MixerThread *thread1 = checkMixerThread_l(output1); 6968 MixerThread *thread2 = checkMixerThread_l(output2); 6969 6970 if (thread1 == NULL || thread2 == NULL) { 6971 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6972 return 0; 6973 } 6974 6975 audio_io_handle_t id = nextUniqueId(); 6976 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6977 thread->addOutputTrack(thread2); 6978 mPlaybackThreads.add(id, thread); 6979 // notify client processes of the new output creation 6980 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6981 return id; 6982} 6983 6984status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6985{ 6986 return closeOutput_nonvirtual(output); 6987} 6988 6989status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6990{ 6991 // keep strong reference on the playback thread so that 6992 // it is not destroyed while exit() is executed 6993 sp<PlaybackThread> thread; 6994 { 6995 Mutex::Autolock _l(mLock); 6996 thread = checkPlaybackThread_l(output); 6997 if (thread == NULL) { 6998 return BAD_VALUE; 6999 } 7000 7001 ALOGV("closeOutput() %d", output); 7002 7003 if (thread->type() == ThreadBase::MIXER) { 7004 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7005 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7006 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7007 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7008 } 7009 } 7010 } 7011 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7012 mPlaybackThreads.removeItem(output); 7013 } 7014 thread->exit(); 7015 // The thread entity (active unit of execution) is no longer running here, 7016 // but the ThreadBase container still exists. 7017 7018 if (thread->type() != ThreadBase::DUPLICATING) { 7019 AudioStreamOut *out = thread->clearOutput(); 7020 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7021 // from now on thread->mOutput is NULL 7022 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7023 delete out; 7024 } 7025 return NO_ERROR; 7026} 7027 7028status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7029{ 7030 Mutex::Autolock _l(mLock); 7031 PlaybackThread *thread = checkPlaybackThread_l(output); 7032 7033 if (thread == NULL) { 7034 return BAD_VALUE; 7035 } 7036 7037 ALOGV("suspendOutput() %d", output); 7038 thread->suspend(); 7039 7040 return NO_ERROR; 7041} 7042 7043status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7044{ 7045 Mutex::Autolock _l(mLock); 7046 PlaybackThread *thread = checkPlaybackThread_l(output); 7047 7048 if (thread == NULL) { 7049 return BAD_VALUE; 7050 } 7051 7052 ALOGV("restoreOutput() %d", output); 7053 7054 thread->restore(); 7055 7056 return NO_ERROR; 7057} 7058 7059audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7060 audio_devices_t *pDevices, 7061 uint32_t *pSamplingRate, 7062 audio_format_t *pFormat, 7063 audio_channel_mask_t *pChannelMask) 7064{ 7065 status_t status; 7066 RecordThread *thread = NULL; 7067 struct audio_config config = { 7068 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7069 channel_mask: pChannelMask ? *pChannelMask : 0, 7070 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7071 }; 7072 uint32_t reqSamplingRate = config.sample_rate; 7073 audio_format_t reqFormat = config.format; 7074 audio_channel_mask_t reqChannels = config.channel_mask; 7075 audio_stream_in_t *inStream = NULL; 7076 AudioHwDevice *inHwDev; 7077 7078 if (pDevices == NULL || *pDevices == 0) { 7079 return 0; 7080 } 7081 7082 Mutex::Autolock _l(mLock); 7083 7084 inHwDev = findSuitableHwDev_l(module, *pDevices); 7085 if (inHwDev == NULL) 7086 return 0; 7087 7088 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7089 audio_io_handle_t id = nextUniqueId(); 7090 7091 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7092 &inStream); 7093 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7094 inStream, 7095 config.sample_rate, 7096 config.format, 7097 config.channel_mask, 7098 status); 7099 7100 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7101 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7102 // or stereo to mono conversions on 16 bit PCM inputs. 7103 if (status == BAD_VALUE && 7104 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7105 (config.sample_rate <= 2 * reqSamplingRate) && 7106 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7107 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7108 inStream = NULL; 7109 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7110 } 7111 7112 if (status == NO_ERROR && inStream != NULL) { 7113 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7114 7115 // Start record thread 7116 // RecorThread require both input and output device indication to forward to audio 7117 // pre processing modules 7118 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7119 thread = new RecordThread(this, 7120 input, 7121 reqSamplingRate, 7122 reqChannels, 7123 id, 7124 device); 7125 mRecordThreads.add(id, thread); 7126 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7127 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7128 if (pFormat != NULL) *pFormat = config.format; 7129 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7130 7131 // notify client processes of the new input creation 7132 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7133 return id; 7134 } 7135 7136 return 0; 7137} 7138 7139status_t AudioFlinger::closeInput(audio_io_handle_t input) 7140{ 7141 return closeInput_nonvirtual(input); 7142} 7143 7144status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7145{ 7146 // keep strong reference on the record thread so that 7147 // it is not destroyed while exit() is executed 7148 sp<RecordThread> thread; 7149 { 7150 Mutex::Autolock _l(mLock); 7151 thread = checkRecordThread_l(input); 7152 if (thread == 0) { 7153 return BAD_VALUE; 7154 } 7155 7156 ALOGV("closeInput() %d", input); 7157 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7158 mRecordThreads.removeItem(input); 7159 } 7160 thread->exit(); 7161 // The thread entity (active unit of execution) is no longer running here, 7162 // but the ThreadBase container still exists. 7163 7164 AudioStreamIn *in = thread->clearInput(); 7165 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7166 // from now on thread->mInput is NULL 7167 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7168 delete in; 7169 7170 return NO_ERROR; 7171} 7172 7173status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7174{ 7175 Mutex::Autolock _l(mLock); 7176 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7177 7178 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7179 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7180 thread->invalidateTracks(stream); 7181 } 7182 7183 return NO_ERROR; 7184} 7185 7186 7187int AudioFlinger::newAudioSessionId() 7188{ 7189 return nextUniqueId(); 7190} 7191 7192void AudioFlinger::acquireAudioSessionId(int audioSession) 7193{ 7194 Mutex::Autolock _l(mLock); 7195 pid_t caller = IPCThreadState::self()->getCallingPid(); 7196 ALOGV("acquiring %d from %d", audioSession, caller); 7197 size_t num = mAudioSessionRefs.size(); 7198 for (size_t i = 0; i< num; i++) { 7199 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7200 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7201 ref->mCnt++; 7202 ALOGV(" incremented refcount to %d", ref->mCnt); 7203 return; 7204 } 7205 } 7206 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7207 ALOGV(" added new entry for %d", audioSession); 7208} 7209 7210void AudioFlinger::releaseAudioSessionId(int audioSession) 7211{ 7212 Mutex::Autolock _l(mLock); 7213 pid_t caller = IPCThreadState::self()->getCallingPid(); 7214 ALOGV("releasing %d from %d", audioSession, caller); 7215 size_t num = mAudioSessionRefs.size(); 7216 for (size_t i = 0; i< num; i++) { 7217 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7218 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7219 ref->mCnt--; 7220 ALOGV(" decremented refcount to %d", ref->mCnt); 7221 if (ref->mCnt == 0) { 7222 mAudioSessionRefs.removeAt(i); 7223 delete ref; 7224 purgeStaleEffects_l(); 7225 } 7226 return; 7227 } 7228 } 7229 ALOGW("session id %d not found for pid %d", audioSession, caller); 7230} 7231 7232void AudioFlinger::purgeStaleEffects_l() { 7233 7234 ALOGV("purging stale effects"); 7235 7236 Vector< sp<EffectChain> > chains; 7237 7238 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7239 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7240 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7241 sp<EffectChain> ec = t->mEffectChains[j]; 7242 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7243 chains.push(ec); 7244 } 7245 } 7246 } 7247 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7248 sp<RecordThread> t = mRecordThreads.valueAt(i); 7249 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7250 sp<EffectChain> ec = t->mEffectChains[j]; 7251 chains.push(ec); 7252 } 7253 } 7254 7255 for (size_t i = 0; i < chains.size(); i++) { 7256 sp<EffectChain> ec = chains[i]; 7257 int sessionid = ec->sessionId(); 7258 sp<ThreadBase> t = ec->mThread.promote(); 7259 if (t == 0) { 7260 continue; 7261 } 7262 size_t numsessionrefs = mAudioSessionRefs.size(); 7263 bool found = false; 7264 for (size_t k = 0; k < numsessionrefs; k++) { 7265 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7266 if (ref->mSessionid == sessionid) { 7267 ALOGV(" session %d still exists for %d with %d refs", 7268 sessionid, ref->mPid, ref->mCnt); 7269 found = true; 7270 break; 7271 } 7272 } 7273 if (!found) { 7274 Mutex::Autolock _l (t->mLock); 7275 // remove all effects from the chain 7276 while (ec->mEffects.size()) { 7277 sp<EffectModule> effect = ec->mEffects[0]; 7278 effect->unPin(); 7279 t->removeEffect_l(effect); 7280 if (effect->purgeHandles()) { 7281 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7282 } 7283 AudioSystem::unregisterEffect(effect->id()); 7284 } 7285 } 7286 } 7287 return; 7288} 7289 7290// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7291AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7292{ 7293 return mPlaybackThreads.valueFor(output).get(); 7294} 7295 7296// checkMixerThread_l() must be called with AudioFlinger::mLock held 7297AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7298{ 7299 PlaybackThread *thread = checkPlaybackThread_l(output); 7300 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7301} 7302 7303// checkRecordThread_l() must be called with AudioFlinger::mLock held 7304AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7305{ 7306 return mRecordThreads.valueFor(input).get(); 7307} 7308 7309uint32_t AudioFlinger::nextUniqueId() 7310{ 7311 return android_atomic_inc(&mNextUniqueId); 7312} 7313 7314AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7315{ 7316 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7317 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7318 AudioStreamOut *output = thread->getOutput(); 7319 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7320 return thread; 7321 } 7322 } 7323 return NULL; 7324} 7325 7326audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7327{ 7328 PlaybackThread *thread = primaryPlaybackThread_l(); 7329 7330 if (thread == NULL) { 7331 return 0; 7332 } 7333 7334 return thread->outDevice(); 7335} 7336 7337sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7338 int triggerSession, 7339 int listenerSession, 7340 sync_event_callback_t callBack, 7341 void *cookie) 7342{ 7343 Mutex::Autolock _l(mLock); 7344 7345 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7346 status_t playStatus = NAME_NOT_FOUND; 7347 status_t recStatus = NAME_NOT_FOUND; 7348 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7349 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7350 if (playStatus == NO_ERROR) { 7351 return event; 7352 } 7353 } 7354 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7355 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7356 if (recStatus == NO_ERROR) { 7357 return event; 7358 } 7359 } 7360 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7361 mPendingSyncEvents.add(event); 7362 } else { 7363 ALOGV("createSyncEvent() invalid event %d", event->type()); 7364 event.clear(); 7365 } 7366 return event; 7367} 7368 7369// ---------------------------------------------------------------------------- 7370// Effect management 7371// ---------------------------------------------------------------------------- 7372 7373 7374status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7375{ 7376 Mutex::Autolock _l(mLock); 7377 return EffectQueryNumberEffects(numEffects); 7378} 7379 7380status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7381{ 7382 Mutex::Autolock _l(mLock); 7383 return EffectQueryEffect(index, descriptor); 7384} 7385 7386status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7387 effect_descriptor_t *descriptor) const 7388{ 7389 Mutex::Autolock _l(mLock); 7390 return EffectGetDescriptor(pUuid, descriptor); 7391} 7392 7393 7394sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7395 effect_descriptor_t *pDesc, 7396 const sp<IEffectClient>& effectClient, 7397 int32_t priority, 7398 audio_io_handle_t io, 7399 int sessionId, 7400 status_t *status, 7401 int *id, 7402 int *enabled) 7403{ 7404 status_t lStatus = NO_ERROR; 7405 sp<EffectHandle> handle; 7406 effect_descriptor_t desc; 7407 7408 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7409 pid, effectClient.get(), priority, sessionId, io); 7410 7411 if (pDesc == NULL) { 7412 lStatus = BAD_VALUE; 7413 goto Exit; 7414 } 7415 7416 // check audio settings permission for global effects 7417 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7418 lStatus = PERMISSION_DENIED; 7419 goto Exit; 7420 } 7421 7422 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7423 // that can only be created by audio policy manager (running in same process) 7424 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7425 lStatus = PERMISSION_DENIED; 7426 goto Exit; 7427 } 7428 7429 if (io == 0) { 7430 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7431 // output must be specified by AudioPolicyManager when using session 7432 // AUDIO_SESSION_OUTPUT_STAGE 7433 lStatus = BAD_VALUE; 7434 goto Exit; 7435 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7436 // if the output returned by getOutputForEffect() is removed before we lock the 7437 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7438 // and we will exit safely 7439 io = AudioSystem::getOutputForEffect(&desc); 7440 } 7441 } 7442 7443 { 7444 Mutex::Autolock _l(mLock); 7445 7446 7447 if (!EffectIsNullUuid(&pDesc->uuid)) { 7448 // if uuid is specified, request effect descriptor 7449 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7450 if (lStatus < 0) { 7451 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7452 goto Exit; 7453 } 7454 } else { 7455 // if uuid is not specified, look for an available implementation 7456 // of the required type in effect factory 7457 if (EffectIsNullUuid(&pDesc->type)) { 7458 ALOGW("createEffect() no effect type"); 7459 lStatus = BAD_VALUE; 7460 goto Exit; 7461 } 7462 uint32_t numEffects = 0; 7463 effect_descriptor_t d; 7464 d.flags = 0; // prevent compiler warning 7465 bool found = false; 7466 7467 lStatus = EffectQueryNumberEffects(&numEffects); 7468 if (lStatus < 0) { 7469 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7470 goto Exit; 7471 } 7472 for (uint32_t i = 0; i < numEffects; i++) { 7473 lStatus = EffectQueryEffect(i, &desc); 7474 if (lStatus < 0) { 7475 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7476 continue; 7477 } 7478 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7479 // If matching type found save effect descriptor. If the session is 7480 // 0 and the effect is not auxiliary, continue enumeration in case 7481 // an auxiliary version of this effect type is available 7482 found = true; 7483 d = desc; 7484 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7485 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7486 break; 7487 } 7488 } 7489 } 7490 if (!found) { 7491 lStatus = BAD_VALUE; 7492 ALOGW("createEffect() effect not found"); 7493 goto Exit; 7494 } 7495 // For same effect type, chose auxiliary version over insert version if 7496 // connect to output mix (Compliance to OpenSL ES) 7497 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7498 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7499 desc = d; 7500 } 7501 } 7502 7503 // Do not allow auxiliary effects on a session different from 0 (output mix) 7504 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7505 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7506 lStatus = INVALID_OPERATION; 7507 goto Exit; 7508 } 7509 7510 // check recording permission for visualizer 7511 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7512 !recordingAllowed()) { 7513 lStatus = PERMISSION_DENIED; 7514 goto Exit; 7515 } 7516 7517 // return effect descriptor 7518 *pDesc = desc; 7519 7520 // If output is not specified try to find a matching audio session ID in one of the 7521 // output threads. 7522 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7523 // because of code checking output when entering the function. 7524 // Note: io is never 0 when creating an effect on an input 7525 if (io == 0) { 7526 // look for the thread where the specified audio session is present 7527 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7528 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7529 io = mPlaybackThreads.keyAt(i); 7530 break; 7531 } 7532 } 7533 if (io == 0) { 7534 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7535 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7536 io = mRecordThreads.keyAt(i); 7537 break; 7538 } 7539 } 7540 } 7541 // If no output thread contains the requested session ID, default to 7542 // first output. The effect chain will be moved to the correct output 7543 // thread when a track with the same session ID is created 7544 if (io == 0 && mPlaybackThreads.size()) { 7545 io = mPlaybackThreads.keyAt(0); 7546 } 7547 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7548 } 7549 ThreadBase *thread = checkRecordThread_l(io); 7550 if (thread == NULL) { 7551 thread = checkPlaybackThread_l(io); 7552 if (thread == NULL) { 7553 ALOGE("createEffect() unknown output thread"); 7554 lStatus = BAD_VALUE; 7555 goto Exit; 7556 } 7557 } 7558 7559 sp<Client> client = registerPid_l(pid); 7560 7561 // create effect on selected output thread 7562 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7563 &desc, enabled, &lStatus); 7564 if (handle != 0 && id != NULL) { 7565 *id = handle->id(); 7566 } 7567 } 7568 7569Exit: 7570 if (status != NULL) { 7571 *status = lStatus; 7572 } 7573 return handle; 7574} 7575 7576status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7577 audio_io_handle_t dstOutput) 7578{ 7579 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7580 sessionId, srcOutput, dstOutput); 7581 Mutex::Autolock _l(mLock); 7582 if (srcOutput == dstOutput) { 7583 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7584 return NO_ERROR; 7585 } 7586 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7587 if (srcThread == NULL) { 7588 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7589 return BAD_VALUE; 7590 } 7591 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7592 if (dstThread == NULL) { 7593 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7594 return BAD_VALUE; 7595 } 7596 7597 Mutex::Autolock _dl(dstThread->mLock); 7598 Mutex::Autolock _sl(srcThread->mLock); 7599 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7600 7601 return NO_ERROR; 7602} 7603 7604// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7605status_t AudioFlinger::moveEffectChain_l(int sessionId, 7606 AudioFlinger::PlaybackThread *srcThread, 7607 AudioFlinger::PlaybackThread *dstThread, 7608 bool reRegister) 7609{ 7610 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7611 sessionId, srcThread, dstThread); 7612 7613 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7614 if (chain == 0) { 7615 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7616 sessionId, srcThread); 7617 return INVALID_OPERATION; 7618 } 7619 7620 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7621 // so that a new chain is created with correct parameters when first effect is added. This is 7622 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7623 // removed. 7624 srcThread->removeEffectChain_l(chain); 7625 7626 // transfer all effects one by one so that new effect chain is created on new thread with 7627 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7628 audio_io_handle_t dstOutput = dstThread->id(); 7629 sp<EffectChain> dstChain; 7630 uint32_t strategy = 0; // prevent compiler warning 7631 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7632 while (effect != 0) { 7633 srcThread->removeEffect_l(effect); 7634 dstThread->addEffect_l(effect); 7635 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7636 if (effect->state() == EffectModule::ACTIVE || 7637 effect->state() == EffectModule::STOPPING) { 7638 effect->start(); 7639 } 7640 // if the move request is not received from audio policy manager, the effect must be 7641 // re-registered with the new strategy and output 7642 if (dstChain == 0) { 7643 dstChain = effect->chain().promote(); 7644 if (dstChain == 0) { 7645 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7646 srcThread->addEffect_l(effect); 7647 return NO_INIT; 7648 } 7649 strategy = dstChain->strategy(); 7650 } 7651 if (reRegister) { 7652 AudioSystem::unregisterEffect(effect->id()); 7653 AudioSystem::registerEffect(&effect->desc(), 7654 dstOutput, 7655 strategy, 7656 sessionId, 7657 effect->id()); 7658 } 7659 effect = chain->getEffectFromId_l(0); 7660 } 7661 7662 return NO_ERROR; 7663} 7664 7665 7666// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7667sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7668 const sp<AudioFlinger::Client>& client, 7669 const sp<IEffectClient>& effectClient, 7670 int32_t priority, 7671 int sessionId, 7672 effect_descriptor_t *desc, 7673 int *enabled, 7674 status_t *status 7675 ) 7676{ 7677 sp<EffectModule> effect; 7678 sp<EffectHandle> handle; 7679 status_t lStatus; 7680 sp<EffectChain> chain; 7681 bool chainCreated = false; 7682 bool effectCreated = false; 7683 bool effectRegistered = false; 7684 7685 lStatus = initCheck(); 7686 if (lStatus != NO_ERROR) { 7687 ALOGW("createEffect_l() Audio driver not initialized."); 7688 goto Exit; 7689 } 7690 7691 // Do not allow effects with session ID 0 on direct output or duplicating threads 7692 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7693 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7694 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7695 desc->name, sessionId); 7696 lStatus = BAD_VALUE; 7697 goto Exit; 7698 } 7699 // Only Pre processor effects are allowed on input threads and only on input threads 7700 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7701 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7702 desc->name, desc->flags, mType); 7703 lStatus = BAD_VALUE; 7704 goto Exit; 7705 } 7706 7707 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7708 7709 { // scope for mLock 7710 Mutex::Autolock _l(mLock); 7711 7712 // check for existing effect chain with the requested audio session 7713 chain = getEffectChain_l(sessionId); 7714 if (chain == 0) { 7715 // create a new chain for this session 7716 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7717 chain = new EffectChain(this, sessionId); 7718 addEffectChain_l(chain); 7719 chain->setStrategy(getStrategyForSession_l(sessionId)); 7720 chainCreated = true; 7721 } else { 7722 effect = chain->getEffectFromDesc_l(desc); 7723 } 7724 7725 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7726 7727 if (effect == 0) { 7728 int id = mAudioFlinger->nextUniqueId(); 7729 // Check CPU and memory usage 7730 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7731 if (lStatus != NO_ERROR) { 7732 goto Exit; 7733 } 7734 effectRegistered = true; 7735 // create a new effect module if none present in the chain 7736 effect = new EffectModule(this, chain, desc, id, sessionId); 7737 lStatus = effect->status(); 7738 if (lStatus != NO_ERROR) { 7739 goto Exit; 7740 } 7741 lStatus = chain->addEffect_l(effect); 7742 if (lStatus != NO_ERROR) { 7743 goto Exit; 7744 } 7745 effectCreated = true; 7746 7747 effect->setDevice(mOutDevice); 7748 effect->setDevice(mInDevice); 7749 effect->setMode(mAudioFlinger->getMode()); 7750 effect->setAudioSource(mAudioSource); 7751 } 7752 // create effect handle and connect it to effect module 7753 handle = new EffectHandle(effect, client, effectClient, priority); 7754 lStatus = effect->addHandle(handle.get()); 7755 if (enabled != NULL) { 7756 *enabled = (int)effect->isEnabled(); 7757 } 7758 } 7759 7760Exit: 7761 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7762 Mutex::Autolock _l(mLock); 7763 if (effectCreated) { 7764 chain->removeEffect_l(effect); 7765 } 7766 if (effectRegistered) { 7767 AudioSystem::unregisterEffect(effect->id()); 7768 } 7769 if (chainCreated) { 7770 removeEffectChain_l(chain); 7771 } 7772 handle.clear(); 7773 } 7774 7775 if (status != NULL) { 7776 *status = lStatus; 7777 } 7778 return handle; 7779} 7780 7781sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7782{ 7783 Mutex::Autolock _l(mLock); 7784 return getEffect_l(sessionId, effectId); 7785} 7786 7787sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7788{ 7789 sp<EffectChain> chain = getEffectChain_l(sessionId); 7790 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7791} 7792 7793// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7794// PlaybackThread::mLock held 7795status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7796{ 7797 // check for existing effect chain with the requested audio session 7798 int sessionId = effect->sessionId(); 7799 sp<EffectChain> chain = getEffectChain_l(sessionId); 7800 bool chainCreated = false; 7801 7802 if (chain == 0) { 7803 // create a new chain for this session 7804 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7805 chain = new EffectChain(this, sessionId); 7806 addEffectChain_l(chain); 7807 chain->setStrategy(getStrategyForSession_l(sessionId)); 7808 chainCreated = true; 7809 } 7810 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7811 7812 if (chain->getEffectFromId_l(effect->id()) != 0) { 7813 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7814 this, effect->desc().name, chain.get()); 7815 return BAD_VALUE; 7816 } 7817 7818 status_t status = chain->addEffect_l(effect); 7819 if (status != NO_ERROR) { 7820 if (chainCreated) { 7821 removeEffectChain_l(chain); 7822 } 7823 return status; 7824 } 7825 7826 effect->setDevice(mOutDevice); 7827 effect->setDevice(mInDevice); 7828 effect->setMode(mAudioFlinger->getMode()); 7829 effect->setAudioSource(mAudioSource); 7830 return NO_ERROR; 7831} 7832 7833void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7834 7835 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7836 effect_descriptor_t desc = effect->desc(); 7837 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7838 detachAuxEffect_l(effect->id()); 7839 } 7840 7841 sp<EffectChain> chain = effect->chain().promote(); 7842 if (chain != 0) { 7843 // remove effect chain if removing last effect 7844 if (chain->removeEffect_l(effect) == 0) { 7845 removeEffectChain_l(chain); 7846 } 7847 } else { 7848 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7849 } 7850} 7851 7852void AudioFlinger::ThreadBase::lockEffectChains_l( 7853 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7854{ 7855 effectChains = mEffectChains; 7856 for (size_t i = 0; i < mEffectChains.size(); i++) { 7857 mEffectChains[i]->lock(); 7858 } 7859} 7860 7861void AudioFlinger::ThreadBase::unlockEffectChains( 7862 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7863{ 7864 for (size_t i = 0; i < effectChains.size(); i++) { 7865 effectChains[i]->unlock(); 7866 } 7867} 7868 7869sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7870{ 7871 Mutex::Autolock _l(mLock); 7872 return getEffectChain_l(sessionId); 7873} 7874 7875sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 7876{ 7877 size_t size = mEffectChains.size(); 7878 for (size_t i = 0; i < size; i++) { 7879 if (mEffectChains[i]->sessionId() == sessionId) { 7880 return mEffectChains[i]; 7881 } 7882 } 7883 return 0; 7884} 7885 7886void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7887{ 7888 Mutex::Autolock _l(mLock); 7889 size_t size = mEffectChains.size(); 7890 for (size_t i = 0; i < size; i++) { 7891 mEffectChains[i]->setMode_l(mode); 7892 } 7893} 7894 7895void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7896 EffectHandle *handle, 7897 bool unpinIfLast) { 7898 7899 Mutex::Autolock _l(mLock); 7900 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7901 // delete the effect module if removing last handle on it 7902 if (effect->removeHandle(handle) == 0) { 7903 if (!effect->isPinned() || unpinIfLast) { 7904 removeEffect_l(effect); 7905 AudioSystem::unregisterEffect(effect->id()); 7906 } 7907 } 7908} 7909 7910status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7911{ 7912 int session = chain->sessionId(); 7913 int16_t *buffer = mMixBuffer; 7914 bool ownsBuffer = false; 7915 7916 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7917 if (session > 0) { 7918 // Only one effect chain can be present in direct output thread and it uses 7919 // the mix buffer as input 7920 if (mType != DIRECT) { 7921 size_t numSamples = mNormalFrameCount * mChannelCount; 7922 buffer = new int16_t[numSamples]; 7923 memset(buffer, 0, numSamples * sizeof(int16_t)); 7924 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7925 ownsBuffer = true; 7926 } 7927 7928 // Attach all tracks with same session ID to this chain. 7929 for (size_t i = 0; i < mTracks.size(); ++i) { 7930 sp<Track> track = mTracks[i]; 7931 if (session == track->sessionId()) { 7932 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7933 track->setMainBuffer(buffer); 7934 chain->incTrackCnt(); 7935 } 7936 } 7937 7938 // indicate all active tracks in the chain 7939 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7940 sp<Track> track = mActiveTracks[i].promote(); 7941 if (track == 0) continue; 7942 if (session == track->sessionId()) { 7943 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7944 chain->incActiveTrackCnt(); 7945 } 7946 } 7947 } 7948 7949 chain->setInBuffer(buffer, ownsBuffer); 7950 chain->setOutBuffer(mMixBuffer); 7951 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7952 // chains list in order to be processed last as it contains output stage effects 7953 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7954 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7955 // after track specific effects and before output stage 7956 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7957 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7958 // Effect chain for other sessions are inserted at beginning of effect 7959 // chains list to be processed before output mix effects. Relative order between other 7960 // sessions is not important 7961 size_t size = mEffectChains.size(); 7962 size_t i = 0; 7963 for (i = 0; i < size; i++) { 7964 if (mEffectChains[i]->sessionId() < session) break; 7965 } 7966 mEffectChains.insertAt(chain, i); 7967 checkSuspendOnAddEffectChain_l(chain); 7968 7969 return NO_ERROR; 7970} 7971 7972size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7973{ 7974 int session = chain->sessionId(); 7975 7976 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7977 7978 for (size_t i = 0; i < mEffectChains.size(); i++) { 7979 if (chain == mEffectChains[i]) { 7980 mEffectChains.removeAt(i); 7981 // detach all active tracks from the chain 7982 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7983 sp<Track> track = mActiveTracks[i].promote(); 7984 if (track == 0) continue; 7985 if (session == track->sessionId()) { 7986 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7987 chain.get(), session); 7988 chain->decActiveTrackCnt(); 7989 } 7990 } 7991 7992 // detach all tracks with same session ID from this chain 7993 for (size_t i = 0; i < mTracks.size(); ++i) { 7994 sp<Track> track = mTracks[i]; 7995 if (session == track->sessionId()) { 7996 track->setMainBuffer(mMixBuffer); 7997 chain->decTrackCnt(); 7998 } 7999 } 8000 break; 8001 } 8002 } 8003 return mEffectChains.size(); 8004} 8005 8006status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8007 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8008{ 8009 Mutex::Autolock _l(mLock); 8010 return attachAuxEffect_l(track, EffectId); 8011} 8012 8013status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8014 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8015{ 8016 status_t status = NO_ERROR; 8017 8018 if (EffectId == 0) { 8019 track->setAuxBuffer(0, NULL); 8020 } else { 8021 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8022 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8023 if (effect != 0) { 8024 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8025 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8026 } else { 8027 status = INVALID_OPERATION; 8028 } 8029 } else { 8030 status = BAD_VALUE; 8031 } 8032 } 8033 return status; 8034} 8035 8036void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8037{ 8038 for (size_t i = 0; i < mTracks.size(); ++i) { 8039 sp<Track> track = mTracks[i]; 8040 if (track->auxEffectId() == effectId) { 8041 attachAuxEffect_l(track, 0); 8042 } 8043 } 8044} 8045 8046status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8047{ 8048 // only one chain per input thread 8049 if (mEffectChains.size() != 0) { 8050 return INVALID_OPERATION; 8051 } 8052 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8053 8054 chain->setInBuffer(NULL); 8055 chain->setOutBuffer(NULL); 8056 8057 checkSuspendOnAddEffectChain_l(chain); 8058 8059 mEffectChains.add(chain); 8060 8061 return NO_ERROR; 8062} 8063 8064size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8065{ 8066 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8067 ALOGW_IF(mEffectChains.size() != 1, 8068 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8069 chain.get(), mEffectChains.size(), this); 8070 if (mEffectChains.size() == 1) { 8071 mEffectChains.removeAt(0); 8072 } 8073 return 0; 8074} 8075 8076// ---------------------------------------------------------------------------- 8077// EffectModule implementation 8078// ---------------------------------------------------------------------------- 8079 8080#undef LOG_TAG 8081#define LOG_TAG "AudioFlinger::EffectModule" 8082 8083AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8084 const wp<AudioFlinger::EffectChain>& chain, 8085 effect_descriptor_t *desc, 8086 int id, 8087 int sessionId) 8088 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8089 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8090 mDescriptor(*desc), 8091 // mConfig is set by configure() and not used before then 8092 mEffectInterface(NULL), 8093 mStatus(NO_INIT), mState(IDLE), 8094 // mMaxDisableWaitCnt is set by configure() and not used before then 8095 // mDisableWaitCnt is set by process() and updateState() and not used before then 8096 mSuspended(false) 8097{ 8098 ALOGV("Constructor %p", this); 8099 int lStatus; 8100 8101 // create effect engine from effect factory 8102 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8103 8104 if (mStatus != NO_ERROR) { 8105 return; 8106 } 8107 lStatus = init(); 8108 if (lStatus < 0) { 8109 mStatus = lStatus; 8110 goto Error; 8111 } 8112 8113 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8114 return; 8115Error: 8116 EffectRelease(mEffectInterface); 8117 mEffectInterface = NULL; 8118 ALOGV("Constructor Error %d", mStatus); 8119} 8120 8121AudioFlinger::EffectModule::~EffectModule() 8122{ 8123 ALOGV("Destructor %p", this); 8124 if (mEffectInterface != NULL) { 8125 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8126 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8127 sp<ThreadBase> thread = mThread.promote(); 8128 if (thread != 0) { 8129 audio_stream_t *stream = thread->stream(); 8130 if (stream != NULL) { 8131 stream->remove_audio_effect(stream, mEffectInterface); 8132 } 8133 } 8134 } 8135 // release effect engine 8136 EffectRelease(mEffectInterface); 8137 } 8138} 8139 8140status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8141{ 8142 status_t status; 8143 8144 Mutex::Autolock _l(mLock); 8145 int priority = handle->priority(); 8146 size_t size = mHandles.size(); 8147 EffectHandle *controlHandle = NULL; 8148 size_t i; 8149 for (i = 0; i < size; i++) { 8150 EffectHandle *h = mHandles[i]; 8151 if (h == NULL || h->destroyed_l()) continue; 8152 // first non destroyed handle is considered in control 8153 if (controlHandle == NULL) 8154 controlHandle = h; 8155 if (h->priority() <= priority) break; 8156 } 8157 // if inserted in first place, move effect control from previous owner to this handle 8158 if (i == 0) { 8159 bool enabled = false; 8160 if (controlHandle != NULL) { 8161 enabled = controlHandle->enabled(); 8162 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8163 } 8164 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8165 status = NO_ERROR; 8166 } else { 8167 status = ALREADY_EXISTS; 8168 } 8169 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8170 mHandles.insertAt(handle, i); 8171 return status; 8172} 8173 8174size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8175{ 8176 Mutex::Autolock _l(mLock); 8177 size_t size = mHandles.size(); 8178 size_t i; 8179 for (i = 0; i < size; i++) { 8180 if (mHandles[i] == handle) break; 8181 } 8182 if (i == size) { 8183 return size; 8184 } 8185 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8186 8187 mHandles.removeAt(i); 8188 // if removed from first place, move effect control from this handle to next in line 8189 if (i == 0) { 8190 EffectHandle *h = controlHandle_l(); 8191 if (h != NULL) { 8192 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8193 } 8194 } 8195 8196 // Prevent calls to process() and other functions on effect interface from now on. 8197 // The effect engine will be released by the destructor when the last strong reference on 8198 // this object is released which can happen after next process is called. 8199 if (mHandles.size() == 0 && !mPinned) { 8200 mState = DESTROYED; 8201 } 8202 8203 return mHandles.size(); 8204} 8205 8206// must be called with EffectModule::mLock held 8207AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8208{ 8209 // the first valid handle in the list has control over the module 8210 for (size_t i = 0; i < mHandles.size(); i++) { 8211 EffectHandle *h = mHandles[i]; 8212 if (h != NULL && !h->destroyed_l()) { 8213 return h; 8214 } 8215 } 8216 8217 return NULL; 8218} 8219 8220size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8221{ 8222 ALOGV("disconnect() %p handle %p", this, handle); 8223 // keep a strong reference on this EffectModule to avoid calling the 8224 // destructor before we exit 8225 sp<EffectModule> keep(this); 8226 { 8227 sp<ThreadBase> thread = mThread.promote(); 8228 if (thread != 0) { 8229 thread->disconnectEffect(keep, handle, unpinIfLast); 8230 } 8231 } 8232 return mHandles.size(); 8233} 8234 8235void AudioFlinger::EffectModule::updateState() { 8236 Mutex::Autolock _l(mLock); 8237 8238 switch (mState) { 8239 case RESTART: 8240 reset_l(); 8241 // FALL THROUGH 8242 8243 case STARTING: 8244 // clear auxiliary effect input buffer for next accumulation 8245 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8246 memset(mConfig.inputCfg.buffer.raw, 8247 0, 8248 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8249 } 8250 start_l(); 8251 mState = ACTIVE; 8252 break; 8253 case STOPPING: 8254 stop_l(); 8255 mDisableWaitCnt = mMaxDisableWaitCnt; 8256 mState = STOPPED; 8257 break; 8258 case STOPPED: 8259 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8260 // turn off sequence. 8261 if (--mDisableWaitCnt == 0) { 8262 reset_l(); 8263 mState = IDLE; 8264 } 8265 break; 8266 default: //IDLE , ACTIVE, DESTROYED 8267 break; 8268 } 8269} 8270 8271void AudioFlinger::EffectModule::process() 8272{ 8273 Mutex::Autolock _l(mLock); 8274 8275 if (mState == DESTROYED || mEffectInterface == NULL || 8276 mConfig.inputCfg.buffer.raw == NULL || 8277 mConfig.outputCfg.buffer.raw == NULL) { 8278 return; 8279 } 8280 8281 if (isProcessEnabled()) { 8282 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8283 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8284 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8285 mConfig.inputCfg.buffer.s32, 8286 mConfig.inputCfg.buffer.frameCount/2); 8287 } 8288 8289 // do the actual processing in the effect engine 8290 int ret = (*mEffectInterface)->process(mEffectInterface, 8291 &mConfig.inputCfg.buffer, 8292 &mConfig.outputCfg.buffer); 8293 8294 // force transition to IDLE state when engine is ready 8295 if (mState == STOPPED && ret == -ENODATA) { 8296 mDisableWaitCnt = 1; 8297 } 8298 8299 // clear auxiliary effect input buffer for next accumulation 8300 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8301 memset(mConfig.inputCfg.buffer.raw, 0, 8302 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8303 } 8304 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8305 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8306 // If an insert effect is idle and input buffer is different from output buffer, 8307 // accumulate input onto output 8308 sp<EffectChain> chain = mChain.promote(); 8309 if (chain != 0 && chain->activeTrackCnt() != 0) { 8310 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8311 int16_t *in = mConfig.inputCfg.buffer.s16; 8312 int16_t *out = mConfig.outputCfg.buffer.s16; 8313 for (size_t i = 0; i < frameCnt; i++) { 8314 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8315 } 8316 } 8317 } 8318} 8319 8320void AudioFlinger::EffectModule::reset_l() 8321{ 8322 if (mEffectInterface == NULL) { 8323 return; 8324 } 8325 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8326} 8327 8328status_t AudioFlinger::EffectModule::configure() 8329{ 8330 if (mEffectInterface == NULL) { 8331 return NO_INIT; 8332 } 8333 8334 sp<ThreadBase> thread = mThread.promote(); 8335 if (thread == 0) { 8336 return DEAD_OBJECT; 8337 } 8338 8339 // TODO: handle configuration of effects replacing track process 8340 audio_channel_mask_t channelMask = thread->channelMask(); 8341 8342 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8343 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8344 } else { 8345 mConfig.inputCfg.channels = channelMask; 8346 } 8347 mConfig.outputCfg.channels = channelMask; 8348 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8349 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8350 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8351 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8352 mConfig.inputCfg.bufferProvider.cookie = NULL; 8353 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8354 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8355 mConfig.outputCfg.bufferProvider.cookie = NULL; 8356 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8357 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8358 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8359 // Insert effect: 8360 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8361 // always overwrites output buffer: input buffer == output buffer 8362 // - in other sessions: 8363 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8364 // other effect: overwrites output buffer: input buffer == output buffer 8365 // Auxiliary effect: 8366 // accumulates in output buffer: input buffer != output buffer 8367 // Therefore: accumulate <=> input buffer != output buffer 8368 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8369 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8370 } else { 8371 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8372 } 8373 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8374 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8375 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8376 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8377 8378 ALOGV("configure() %p thread %p buffer %p framecount %d", 8379 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8380 8381 status_t cmdStatus; 8382 uint32_t size = sizeof(int); 8383 status_t status = (*mEffectInterface)->command(mEffectInterface, 8384 EFFECT_CMD_SET_CONFIG, 8385 sizeof(effect_config_t), 8386 &mConfig, 8387 &size, 8388 &cmdStatus); 8389 if (status == 0) { 8390 status = cmdStatus; 8391 } 8392 8393 if (status == 0 && 8394 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8395 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8396 effect_param_t *p = (effect_param_t *)buf32; 8397 8398 p->psize = sizeof(uint32_t); 8399 p->vsize = sizeof(uint32_t); 8400 size = sizeof(int); 8401 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8402 8403 uint32_t latency = 0; 8404 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8405 if (pbt != NULL) { 8406 latency = pbt->latency_l(); 8407 } 8408 8409 *((int32_t *)p->data + 1)= latency; 8410 (*mEffectInterface)->command(mEffectInterface, 8411 EFFECT_CMD_SET_PARAM, 8412 sizeof(effect_param_t) + 8, 8413 &buf32, 8414 &size, 8415 &cmdStatus); 8416 } 8417 8418 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8419 (1000 * mConfig.outputCfg.buffer.frameCount); 8420 8421 return status; 8422} 8423 8424status_t AudioFlinger::EffectModule::init() 8425{ 8426 Mutex::Autolock _l(mLock); 8427 if (mEffectInterface == NULL) { 8428 return NO_INIT; 8429 } 8430 status_t cmdStatus; 8431 uint32_t size = sizeof(status_t); 8432 status_t status = (*mEffectInterface)->command(mEffectInterface, 8433 EFFECT_CMD_INIT, 8434 0, 8435 NULL, 8436 &size, 8437 &cmdStatus); 8438 if (status == 0) { 8439 status = cmdStatus; 8440 } 8441 return status; 8442} 8443 8444status_t AudioFlinger::EffectModule::start() 8445{ 8446 Mutex::Autolock _l(mLock); 8447 return start_l(); 8448} 8449 8450status_t AudioFlinger::EffectModule::start_l() 8451{ 8452 if (mEffectInterface == NULL) { 8453 return NO_INIT; 8454 } 8455 status_t cmdStatus; 8456 uint32_t size = sizeof(status_t); 8457 status_t status = (*mEffectInterface)->command(mEffectInterface, 8458 EFFECT_CMD_ENABLE, 8459 0, 8460 NULL, 8461 &size, 8462 &cmdStatus); 8463 if (status == 0) { 8464 status = cmdStatus; 8465 } 8466 if (status == 0 && 8467 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8468 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8469 sp<ThreadBase> thread = mThread.promote(); 8470 if (thread != 0) { 8471 audio_stream_t *stream = thread->stream(); 8472 if (stream != NULL) { 8473 stream->add_audio_effect(stream, mEffectInterface); 8474 } 8475 } 8476 } 8477 return status; 8478} 8479 8480status_t AudioFlinger::EffectModule::stop() 8481{ 8482 Mutex::Autolock _l(mLock); 8483 return stop_l(); 8484} 8485 8486status_t AudioFlinger::EffectModule::stop_l() 8487{ 8488 if (mEffectInterface == NULL) { 8489 return NO_INIT; 8490 } 8491 status_t cmdStatus; 8492 uint32_t size = sizeof(status_t); 8493 status_t status = (*mEffectInterface)->command(mEffectInterface, 8494 EFFECT_CMD_DISABLE, 8495 0, 8496 NULL, 8497 &size, 8498 &cmdStatus); 8499 if (status == 0) { 8500 status = cmdStatus; 8501 } 8502 if (status == 0 && 8503 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8504 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8505 sp<ThreadBase> thread = mThread.promote(); 8506 if (thread != 0) { 8507 audio_stream_t *stream = thread->stream(); 8508 if (stream != NULL) { 8509 stream->remove_audio_effect(stream, mEffectInterface); 8510 } 8511 } 8512 } 8513 return status; 8514} 8515 8516status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8517 uint32_t cmdSize, 8518 void *pCmdData, 8519 uint32_t *replySize, 8520 void *pReplyData) 8521{ 8522 Mutex::Autolock _l(mLock); 8523// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8524 8525 if (mState == DESTROYED || mEffectInterface == NULL) { 8526 return NO_INIT; 8527 } 8528 status_t status = (*mEffectInterface)->command(mEffectInterface, 8529 cmdCode, 8530 cmdSize, 8531 pCmdData, 8532 replySize, 8533 pReplyData); 8534 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8535 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8536 for (size_t i = 1; i < mHandles.size(); i++) { 8537 EffectHandle *h = mHandles[i]; 8538 if (h != NULL && !h->destroyed_l()) { 8539 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8540 } 8541 } 8542 } 8543 return status; 8544} 8545 8546status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8547{ 8548 Mutex::Autolock _l(mLock); 8549 return setEnabled_l(enabled); 8550} 8551 8552// must be called with EffectModule::mLock held 8553status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8554{ 8555 8556 ALOGV("setEnabled %p enabled %d", this, enabled); 8557 8558 if (enabled != isEnabled()) { 8559 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8560 if (enabled && status != NO_ERROR) { 8561 return status; 8562 } 8563 8564 switch (mState) { 8565 // going from disabled to enabled 8566 case IDLE: 8567 mState = STARTING; 8568 break; 8569 case STOPPED: 8570 mState = RESTART; 8571 break; 8572 case STOPPING: 8573 mState = ACTIVE; 8574 break; 8575 8576 // going from enabled to disabled 8577 case RESTART: 8578 mState = STOPPED; 8579 break; 8580 case STARTING: 8581 mState = IDLE; 8582 break; 8583 case ACTIVE: 8584 mState = STOPPING; 8585 break; 8586 case DESTROYED: 8587 return NO_ERROR; // simply ignore as we are being destroyed 8588 } 8589 for (size_t i = 1; i < mHandles.size(); i++) { 8590 EffectHandle *h = mHandles[i]; 8591 if (h != NULL && !h->destroyed_l()) { 8592 h->setEnabled(enabled); 8593 } 8594 } 8595 } 8596 return NO_ERROR; 8597} 8598 8599bool AudioFlinger::EffectModule::isEnabled() const 8600{ 8601 switch (mState) { 8602 case RESTART: 8603 case STARTING: 8604 case ACTIVE: 8605 return true; 8606 case IDLE: 8607 case STOPPING: 8608 case STOPPED: 8609 case DESTROYED: 8610 default: 8611 return false; 8612 } 8613} 8614 8615bool AudioFlinger::EffectModule::isProcessEnabled() const 8616{ 8617 switch (mState) { 8618 case RESTART: 8619 case ACTIVE: 8620 case STOPPING: 8621 case STOPPED: 8622 return true; 8623 case IDLE: 8624 case STARTING: 8625 case DESTROYED: 8626 default: 8627 return false; 8628 } 8629} 8630 8631status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8632{ 8633 Mutex::Autolock _l(mLock); 8634 status_t status = NO_ERROR; 8635 8636 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8637 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8638 if (isProcessEnabled() && 8639 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8640 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8641 status_t cmdStatus; 8642 uint32_t volume[2]; 8643 uint32_t *pVolume = NULL; 8644 uint32_t size = sizeof(volume); 8645 volume[0] = *left; 8646 volume[1] = *right; 8647 if (controller) { 8648 pVolume = volume; 8649 } 8650 status = (*mEffectInterface)->command(mEffectInterface, 8651 EFFECT_CMD_SET_VOLUME, 8652 size, 8653 volume, 8654 &size, 8655 pVolume); 8656 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8657 *left = volume[0]; 8658 *right = volume[1]; 8659 } 8660 } 8661 return status; 8662} 8663 8664status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8665{ 8666 if (device == AUDIO_DEVICE_NONE) { 8667 return NO_ERROR; 8668 } 8669 8670 Mutex::Autolock _l(mLock); 8671 status_t status = NO_ERROR; 8672 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8673 status_t cmdStatus; 8674 uint32_t size = sizeof(status_t); 8675 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8676 EFFECT_CMD_SET_INPUT_DEVICE; 8677 status = (*mEffectInterface)->command(mEffectInterface, 8678 cmd, 8679 sizeof(uint32_t), 8680 &device, 8681 &size, 8682 &cmdStatus); 8683 } 8684 return status; 8685} 8686 8687status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8688{ 8689 Mutex::Autolock _l(mLock); 8690 status_t status = NO_ERROR; 8691 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8692 status_t cmdStatus; 8693 uint32_t size = sizeof(status_t); 8694 status = (*mEffectInterface)->command(mEffectInterface, 8695 EFFECT_CMD_SET_AUDIO_MODE, 8696 sizeof(audio_mode_t), 8697 &mode, 8698 &size, 8699 &cmdStatus); 8700 if (status == NO_ERROR) { 8701 status = cmdStatus; 8702 } 8703 } 8704 return status; 8705} 8706 8707status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8708{ 8709 Mutex::Autolock _l(mLock); 8710 status_t status = NO_ERROR; 8711 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8712 uint32_t size = 0; 8713 status = (*mEffectInterface)->command(mEffectInterface, 8714 EFFECT_CMD_SET_AUDIO_SOURCE, 8715 sizeof(audio_source_t), 8716 &source, 8717 &size, 8718 NULL); 8719 } 8720 return status; 8721} 8722 8723void AudioFlinger::EffectModule::setSuspended(bool suspended) 8724{ 8725 Mutex::Autolock _l(mLock); 8726 mSuspended = suspended; 8727} 8728 8729bool AudioFlinger::EffectModule::suspended() const 8730{ 8731 Mutex::Autolock _l(mLock); 8732 return mSuspended; 8733} 8734 8735bool AudioFlinger::EffectModule::purgeHandles() 8736{ 8737 bool enabled = false; 8738 Mutex::Autolock _l(mLock); 8739 for (size_t i = 0; i < mHandles.size(); i++) { 8740 EffectHandle *handle = mHandles[i]; 8741 if (handle != NULL && !handle->destroyed_l()) { 8742 handle->effect().clear(); 8743 if (handle->hasControl()) { 8744 enabled = handle->enabled(); 8745 } 8746 } 8747 } 8748 return enabled; 8749} 8750 8751void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8752{ 8753 const size_t SIZE = 256; 8754 char buffer[SIZE]; 8755 String8 result; 8756 8757 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8758 result.append(buffer); 8759 8760 bool locked = tryLock(mLock); 8761 // failed to lock - AudioFlinger is probably deadlocked 8762 if (!locked) { 8763 result.append("\t\tCould not lock Fx mutex:\n"); 8764 } 8765 8766 result.append("\t\tSession Status State Engine:\n"); 8767 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8768 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8769 result.append(buffer); 8770 8771 result.append("\t\tDescriptor:\n"); 8772 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8773 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8774 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8775 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8776 result.append(buffer); 8777 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8778 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8779 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8780 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8781 result.append(buffer); 8782 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8783 mDescriptor.apiVersion, 8784 mDescriptor.flags); 8785 result.append(buffer); 8786 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8787 mDescriptor.name); 8788 result.append(buffer); 8789 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8790 mDescriptor.implementor); 8791 result.append(buffer); 8792 8793 result.append("\t\t- Input configuration:\n"); 8794 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8795 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8796 (uint32_t)mConfig.inputCfg.buffer.raw, 8797 mConfig.inputCfg.buffer.frameCount, 8798 mConfig.inputCfg.samplingRate, 8799 mConfig.inputCfg.channels, 8800 mConfig.inputCfg.format); 8801 result.append(buffer); 8802 8803 result.append("\t\t- Output configuration:\n"); 8804 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8805 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8806 (uint32_t)mConfig.outputCfg.buffer.raw, 8807 mConfig.outputCfg.buffer.frameCount, 8808 mConfig.outputCfg.samplingRate, 8809 mConfig.outputCfg.channels, 8810 mConfig.outputCfg.format); 8811 result.append(buffer); 8812 8813 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8814 result.append(buffer); 8815 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8816 for (size_t i = 0; i < mHandles.size(); ++i) { 8817 EffectHandle *handle = mHandles[i]; 8818 if (handle != NULL && !handle->destroyed_l()) { 8819 handle->dump(buffer, SIZE); 8820 result.append(buffer); 8821 } 8822 } 8823 8824 result.append("\n"); 8825 8826 write(fd, result.string(), result.length()); 8827 8828 if (locked) { 8829 mLock.unlock(); 8830 } 8831} 8832 8833// ---------------------------------------------------------------------------- 8834// EffectHandle implementation 8835// ---------------------------------------------------------------------------- 8836 8837#undef LOG_TAG 8838#define LOG_TAG "AudioFlinger::EffectHandle" 8839 8840AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8841 const sp<AudioFlinger::Client>& client, 8842 const sp<IEffectClient>& effectClient, 8843 int32_t priority) 8844 : BnEffect(), 8845 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8846 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8847{ 8848 ALOGV("constructor %p", this); 8849 8850 if (client == 0) { 8851 return; 8852 } 8853 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8854 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8855 if (mCblkMemory != 0) { 8856 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8857 8858 if (mCblk != NULL) { 8859 new(mCblk) effect_param_cblk_t(); 8860 mBuffer = (uint8_t *)mCblk + bufOffset; 8861 } 8862 } else { 8863 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8864 return; 8865 } 8866} 8867 8868AudioFlinger::EffectHandle::~EffectHandle() 8869{ 8870 ALOGV("Destructor %p", this); 8871 8872 if (mEffect == 0) { 8873 mDestroyed = true; 8874 return; 8875 } 8876 mEffect->lock(); 8877 mDestroyed = true; 8878 mEffect->unlock(); 8879 disconnect(false); 8880} 8881 8882status_t AudioFlinger::EffectHandle::enable() 8883{ 8884 ALOGV("enable %p", this); 8885 if (!mHasControl) return INVALID_OPERATION; 8886 if (mEffect == 0) return DEAD_OBJECT; 8887 8888 if (mEnabled) { 8889 return NO_ERROR; 8890 } 8891 8892 mEnabled = true; 8893 8894 sp<ThreadBase> thread = mEffect->thread().promote(); 8895 if (thread != 0) { 8896 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8897 } 8898 8899 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8900 if (mEffect->suspended()) { 8901 return NO_ERROR; 8902 } 8903 8904 status_t status = mEffect->setEnabled(true); 8905 if (status != NO_ERROR) { 8906 if (thread != 0) { 8907 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8908 } 8909 mEnabled = false; 8910 } 8911 return status; 8912} 8913 8914status_t AudioFlinger::EffectHandle::disable() 8915{ 8916 ALOGV("disable %p", this); 8917 if (!mHasControl) return INVALID_OPERATION; 8918 if (mEffect == 0) return DEAD_OBJECT; 8919 8920 if (!mEnabled) { 8921 return NO_ERROR; 8922 } 8923 mEnabled = false; 8924 8925 if (mEffect->suspended()) { 8926 return NO_ERROR; 8927 } 8928 8929 status_t status = mEffect->setEnabled(false); 8930 8931 sp<ThreadBase> thread = mEffect->thread().promote(); 8932 if (thread != 0) { 8933 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8934 } 8935 8936 return status; 8937} 8938 8939void AudioFlinger::EffectHandle::disconnect() 8940{ 8941 disconnect(true); 8942} 8943 8944void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8945{ 8946 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8947 if (mEffect == 0) { 8948 return; 8949 } 8950 // restore suspended effects if the disconnected handle was enabled and the last one. 8951 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8952 sp<ThreadBase> thread = mEffect->thread().promote(); 8953 if (thread != 0) { 8954 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8955 } 8956 } 8957 8958 // release sp on module => module destructor can be called now 8959 mEffect.clear(); 8960 if (mClient != 0) { 8961 if (mCblk != NULL) { 8962 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8963 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8964 } 8965 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8966 // Client destructor must run with AudioFlinger mutex locked 8967 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8968 mClient.clear(); 8969 } 8970} 8971 8972status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8973 uint32_t cmdSize, 8974 void *pCmdData, 8975 uint32_t *replySize, 8976 void *pReplyData) 8977{ 8978// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8979// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8980 8981 // only get parameter command is permitted for applications not controlling the effect 8982 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8983 return INVALID_OPERATION; 8984 } 8985 if (mEffect == 0) return DEAD_OBJECT; 8986 if (mClient == 0) return INVALID_OPERATION; 8987 8988 // handle commands that are not forwarded transparently to effect engine 8989 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8990 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8991 // no risk to block the whole media server process or mixer threads is we are stuck here 8992 Mutex::Autolock _l(mCblk->lock); 8993 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8994 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8995 mCblk->serverIndex = 0; 8996 mCblk->clientIndex = 0; 8997 return BAD_VALUE; 8998 } 8999 status_t status = NO_ERROR; 9000 while (mCblk->serverIndex < mCblk->clientIndex) { 9001 int reply; 9002 uint32_t rsize = sizeof(int); 9003 int *p = (int *)(mBuffer + mCblk->serverIndex); 9004 int size = *p++; 9005 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9006 ALOGW("command(): invalid parameter block size"); 9007 break; 9008 } 9009 effect_param_t *param = (effect_param_t *)p; 9010 if (param->psize == 0 || param->vsize == 0) { 9011 ALOGW("command(): null parameter or value size"); 9012 mCblk->serverIndex += size; 9013 continue; 9014 } 9015 uint32_t psize = sizeof(effect_param_t) + 9016 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9017 param->vsize; 9018 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9019 psize, 9020 p, 9021 &rsize, 9022 &reply); 9023 // stop at first error encountered 9024 if (ret != NO_ERROR) { 9025 status = ret; 9026 *(int *)pReplyData = reply; 9027 break; 9028 } else if (reply != NO_ERROR) { 9029 *(int *)pReplyData = reply; 9030 break; 9031 } 9032 mCblk->serverIndex += size; 9033 } 9034 mCblk->serverIndex = 0; 9035 mCblk->clientIndex = 0; 9036 return status; 9037 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9038 *(int *)pReplyData = NO_ERROR; 9039 return enable(); 9040 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9041 *(int *)pReplyData = NO_ERROR; 9042 return disable(); 9043 } 9044 9045 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9046} 9047 9048void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9049{ 9050 ALOGV("setControl %p control %d", this, hasControl); 9051 9052 mHasControl = hasControl; 9053 mEnabled = enabled; 9054 9055 if (signal && mEffectClient != 0) { 9056 mEffectClient->controlStatusChanged(hasControl); 9057 } 9058} 9059 9060void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9061 uint32_t cmdSize, 9062 void *pCmdData, 9063 uint32_t replySize, 9064 void *pReplyData) 9065{ 9066 if (mEffectClient != 0) { 9067 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9068 } 9069} 9070 9071 9072 9073void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9074{ 9075 if (mEffectClient != 0) { 9076 mEffectClient->enableStatusChanged(enabled); 9077 } 9078} 9079 9080status_t AudioFlinger::EffectHandle::onTransact( 9081 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9082{ 9083 return BnEffect::onTransact(code, data, reply, flags); 9084} 9085 9086 9087void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9088{ 9089 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9090 9091 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9092 (mClient == 0) ? getpid_cached : mClient->pid(), 9093 mPriority, 9094 mHasControl, 9095 !locked, 9096 mCblk ? mCblk->clientIndex : 0, 9097 mCblk ? mCblk->serverIndex : 0 9098 ); 9099 9100 if (locked) { 9101 mCblk->lock.unlock(); 9102 } 9103} 9104 9105#undef LOG_TAG 9106#define LOG_TAG "AudioFlinger::EffectChain" 9107 9108AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9109 int sessionId) 9110 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9111 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9112 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9113{ 9114 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9115 if (thread == NULL) { 9116 return; 9117 } 9118 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9119 thread->frameCount(); 9120} 9121 9122AudioFlinger::EffectChain::~EffectChain() 9123{ 9124 if (mOwnInBuffer) { 9125 delete mInBuffer; 9126 } 9127 9128} 9129 9130// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9131sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9132{ 9133 size_t size = mEffects.size(); 9134 9135 for (size_t i = 0; i < size; i++) { 9136 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9137 return mEffects[i]; 9138 } 9139 } 9140 return 0; 9141} 9142 9143// getEffectFromId_l() must be called with ThreadBase::mLock held 9144sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9145{ 9146 size_t size = mEffects.size(); 9147 9148 for (size_t i = 0; i < size; i++) { 9149 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9150 if (id == 0 || mEffects[i]->id() == id) { 9151 return mEffects[i]; 9152 } 9153 } 9154 return 0; 9155} 9156 9157// getEffectFromType_l() must be called with ThreadBase::mLock held 9158sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9159 const effect_uuid_t *type) 9160{ 9161 size_t size = mEffects.size(); 9162 9163 for (size_t i = 0; i < size; i++) { 9164 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9165 return mEffects[i]; 9166 } 9167 } 9168 return 0; 9169} 9170 9171void AudioFlinger::EffectChain::clearInputBuffer() 9172{ 9173 Mutex::Autolock _l(mLock); 9174 sp<ThreadBase> thread = mThread.promote(); 9175 if (thread == 0) { 9176 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9177 return; 9178 } 9179 clearInputBuffer_l(thread); 9180} 9181 9182// Must be called with EffectChain::mLock locked 9183void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9184{ 9185 size_t numSamples = thread->frameCount() * thread->channelCount(); 9186 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9187 9188} 9189 9190// Must be called with EffectChain::mLock locked 9191void AudioFlinger::EffectChain::process_l() 9192{ 9193 sp<ThreadBase> thread = mThread.promote(); 9194 if (thread == 0) { 9195 ALOGW("process_l(): cannot promote mixer thread"); 9196 return; 9197 } 9198 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9199 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9200 // always process effects unless no more tracks are on the session and the effect tail 9201 // has been rendered 9202 bool doProcess = true; 9203 if (!isGlobalSession) { 9204 bool tracksOnSession = (trackCnt() != 0); 9205 9206 if (!tracksOnSession && mTailBufferCount == 0) { 9207 doProcess = false; 9208 } 9209 9210 if (activeTrackCnt() == 0) { 9211 // if no track is active and the effect tail has not been rendered, 9212 // the input buffer must be cleared here as the mixer process will not do it 9213 if (tracksOnSession || mTailBufferCount > 0) { 9214 clearInputBuffer_l(thread); 9215 if (mTailBufferCount > 0) { 9216 mTailBufferCount--; 9217 } 9218 } 9219 } 9220 } 9221 9222 size_t size = mEffects.size(); 9223 if (doProcess) { 9224 for (size_t i = 0; i < size; i++) { 9225 mEffects[i]->process(); 9226 } 9227 } 9228 for (size_t i = 0; i < size; i++) { 9229 mEffects[i]->updateState(); 9230 } 9231} 9232 9233// addEffect_l() must be called with PlaybackThread::mLock held 9234status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9235{ 9236 effect_descriptor_t desc = effect->desc(); 9237 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9238 9239 Mutex::Autolock _l(mLock); 9240 effect->setChain(this); 9241 sp<ThreadBase> thread = mThread.promote(); 9242 if (thread == 0) { 9243 return NO_INIT; 9244 } 9245 effect->setThread(thread); 9246 9247 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9248 // Auxiliary effects are inserted at the beginning of mEffects vector as 9249 // they are processed first and accumulated in chain input buffer 9250 mEffects.insertAt(effect, 0); 9251 9252 // the input buffer for auxiliary effect contains mono samples in 9253 // 32 bit format. This is to avoid saturation in AudoMixer 9254 // accumulation stage. Saturation is done in EffectModule::process() before 9255 // calling the process in effect engine 9256 size_t numSamples = thread->frameCount(); 9257 int32_t *buffer = new int32_t[numSamples]; 9258 memset(buffer, 0, numSamples * sizeof(int32_t)); 9259 effect->setInBuffer((int16_t *)buffer); 9260 // auxiliary effects output samples to chain input buffer for further processing 9261 // by insert effects 9262 effect->setOutBuffer(mInBuffer); 9263 } else { 9264 // Insert effects are inserted at the end of mEffects vector as they are processed 9265 // after track and auxiliary effects. 9266 // Insert effect order as a function of indicated preference: 9267 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9268 // another effect is present 9269 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9270 // last effect claiming first position 9271 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9272 // first effect claiming last position 9273 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9274 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9275 // already present 9276 9277 size_t size = mEffects.size(); 9278 size_t idx_insert = size; 9279 ssize_t idx_insert_first = -1; 9280 ssize_t idx_insert_last = -1; 9281 9282 for (size_t i = 0; i < size; i++) { 9283 effect_descriptor_t d = mEffects[i]->desc(); 9284 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9285 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9286 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9287 // check invalid effect chaining combinations 9288 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9289 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9290 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9291 return INVALID_OPERATION; 9292 } 9293 // remember position of first insert effect and by default 9294 // select this as insert position for new effect 9295 if (idx_insert == size) { 9296 idx_insert = i; 9297 } 9298 // remember position of last insert effect claiming 9299 // first position 9300 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9301 idx_insert_first = i; 9302 } 9303 // remember position of first insert effect claiming 9304 // last position 9305 if (iPref == EFFECT_FLAG_INSERT_LAST && 9306 idx_insert_last == -1) { 9307 idx_insert_last = i; 9308 } 9309 } 9310 } 9311 9312 // modify idx_insert from first position if needed 9313 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9314 if (idx_insert_last != -1) { 9315 idx_insert = idx_insert_last; 9316 } else { 9317 idx_insert = size; 9318 } 9319 } else { 9320 if (idx_insert_first != -1) { 9321 idx_insert = idx_insert_first + 1; 9322 } 9323 } 9324 9325 // always read samples from chain input buffer 9326 effect->setInBuffer(mInBuffer); 9327 9328 // if last effect in the chain, output samples to chain 9329 // output buffer, otherwise to chain input buffer 9330 if (idx_insert == size) { 9331 if (idx_insert != 0) { 9332 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9333 mEffects[idx_insert-1]->configure(); 9334 } 9335 effect->setOutBuffer(mOutBuffer); 9336 } else { 9337 effect->setOutBuffer(mInBuffer); 9338 } 9339 mEffects.insertAt(effect, idx_insert); 9340 9341 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9342 } 9343 effect->configure(); 9344 return NO_ERROR; 9345} 9346 9347// removeEffect_l() must be called with PlaybackThread::mLock held 9348size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9349{ 9350 Mutex::Autolock _l(mLock); 9351 size_t size = mEffects.size(); 9352 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9353 9354 for (size_t i = 0; i < size; i++) { 9355 if (effect == mEffects[i]) { 9356 // calling stop here will remove pre-processing effect from the audio HAL. 9357 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9358 // the middle of a read from audio HAL 9359 if (mEffects[i]->state() == EffectModule::ACTIVE || 9360 mEffects[i]->state() == EffectModule::STOPPING) { 9361 mEffects[i]->stop(); 9362 } 9363 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9364 delete[] effect->inBuffer(); 9365 } else { 9366 if (i == size - 1 && i != 0) { 9367 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9368 mEffects[i - 1]->configure(); 9369 } 9370 } 9371 mEffects.removeAt(i); 9372 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9373 break; 9374 } 9375 } 9376 9377 return mEffects.size(); 9378} 9379 9380// setDevice_l() must be called with PlaybackThread::mLock held 9381void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9382{ 9383 size_t size = mEffects.size(); 9384 for (size_t i = 0; i < size; i++) { 9385 mEffects[i]->setDevice(device); 9386 } 9387} 9388 9389// setMode_l() must be called with PlaybackThread::mLock held 9390void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9391{ 9392 size_t size = mEffects.size(); 9393 for (size_t i = 0; i < size; i++) { 9394 mEffects[i]->setMode(mode); 9395 } 9396} 9397 9398// setAudioSource_l() must be called with PlaybackThread::mLock held 9399void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9400{ 9401 size_t size = mEffects.size(); 9402 for (size_t i = 0; i < size; i++) { 9403 mEffects[i]->setAudioSource(source); 9404 } 9405} 9406 9407// setVolume_l() must be called with PlaybackThread::mLock held 9408bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9409{ 9410 uint32_t newLeft = *left; 9411 uint32_t newRight = *right; 9412 bool hasControl = false; 9413 int ctrlIdx = -1; 9414 size_t size = mEffects.size(); 9415 9416 // first update volume controller 9417 for (size_t i = size; i > 0; i--) { 9418 if (mEffects[i - 1]->isProcessEnabled() && 9419 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9420 ctrlIdx = i - 1; 9421 hasControl = true; 9422 break; 9423 } 9424 } 9425 9426 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9427 if (hasControl) { 9428 *left = mNewLeftVolume; 9429 *right = mNewRightVolume; 9430 } 9431 return hasControl; 9432 } 9433 9434 mVolumeCtrlIdx = ctrlIdx; 9435 mLeftVolume = newLeft; 9436 mRightVolume = newRight; 9437 9438 // second get volume update from volume controller 9439 if (ctrlIdx >= 0) { 9440 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9441 mNewLeftVolume = newLeft; 9442 mNewRightVolume = newRight; 9443 } 9444 // then indicate volume to all other effects in chain. 9445 // Pass altered volume to effects before volume controller 9446 // and requested volume to effects after controller 9447 uint32_t lVol = newLeft; 9448 uint32_t rVol = newRight; 9449 9450 for (size_t i = 0; i < size; i++) { 9451 if ((int)i == ctrlIdx) continue; 9452 // this also works for ctrlIdx == -1 when there is no volume controller 9453 if ((int)i > ctrlIdx) { 9454 lVol = *left; 9455 rVol = *right; 9456 } 9457 mEffects[i]->setVolume(&lVol, &rVol, false); 9458 } 9459 *left = newLeft; 9460 *right = newRight; 9461 9462 return hasControl; 9463} 9464 9465void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9466{ 9467 const size_t SIZE = 256; 9468 char buffer[SIZE]; 9469 String8 result; 9470 9471 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9472 result.append(buffer); 9473 9474 bool locked = tryLock(mLock); 9475 // failed to lock - AudioFlinger is probably deadlocked 9476 if (!locked) { 9477 result.append("\tCould not lock mutex:\n"); 9478 } 9479 9480 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9481 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9482 mEffects.size(), 9483 (uint32_t)mInBuffer, 9484 (uint32_t)mOutBuffer, 9485 mActiveTrackCnt); 9486 result.append(buffer); 9487 write(fd, result.string(), result.size()); 9488 9489 for (size_t i = 0; i < mEffects.size(); ++i) { 9490 sp<EffectModule> effect = mEffects[i]; 9491 if (effect != 0) { 9492 effect->dump(fd, args); 9493 } 9494 } 9495 9496 if (locked) { 9497 mLock.unlock(); 9498 } 9499} 9500 9501// must be called with ThreadBase::mLock held 9502void AudioFlinger::EffectChain::setEffectSuspended_l( 9503 const effect_uuid_t *type, bool suspend) 9504{ 9505 sp<SuspendedEffectDesc> desc; 9506 // use effect type UUID timelow as key as there is no real risk of identical 9507 // timeLow fields among effect type UUIDs. 9508 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9509 if (suspend) { 9510 if (index >= 0) { 9511 desc = mSuspendedEffects.valueAt(index); 9512 } else { 9513 desc = new SuspendedEffectDesc(); 9514 desc->mType = *type; 9515 mSuspendedEffects.add(type->timeLow, desc); 9516 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9517 } 9518 if (desc->mRefCount++ == 0) { 9519 sp<EffectModule> effect = getEffectIfEnabled(type); 9520 if (effect != 0) { 9521 desc->mEffect = effect; 9522 effect->setSuspended(true); 9523 effect->setEnabled(false); 9524 } 9525 } 9526 } else { 9527 if (index < 0) { 9528 return; 9529 } 9530 desc = mSuspendedEffects.valueAt(index); 9531 if (desc->mRefCount <= 0) { 9532 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9533 desc->mRefCount = 1; 9534 } 9535 if (--desc->mRefCount == 0) { 9536 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9537 if (desc->mEffect != 0) { 9538 sp<EffectModule> effect = desc->mEffect.promote(); 9539 if (effect != 0) { 9540 effect->setSuspended(false); 9541 effect->lock(); 9542 EffectHandle *handle = effect->controlHandle_l(); 9543 if (handle != NULL && !handle->destroyed_l()) { 9544 effect->setEnabled_l(handle->enabled()); 9545 } 9546 effect->unlock(); 9547 } 9548 desc->mEffect.clear(); 9549 } 9550 mSuspendedEffects.removeItemsAt(index); 9551 } 9552 } 9553} 9554 9555// must be called with ThreadBase::mLock held 9556void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9557{ 9558 sp<SuspendedEffectDesc> desc; 9559 9560 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9561 if (suspend) { 9562 if (index >= 0) { 9563 desc = mSuspendedEffects.valueAt(index); 9564 } else { 9565 desc = new SuspendedEffectDesc(); 9566 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9567 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9568 } 9569 if (desc->mRefCount++ == 0) { 9570 Vector< sp<EffectModule> > effects; 9571 getSuspendEligibleEffects(effects); 9572 for (size_t i = 0; i < effects.size(); i++) { 9573 setEffectSuspended_l(&effects[i]->desc().type, true); 9574 } 9575 } 9576 } else { 9577 if (index < 0) { 9578 return; 9579 } 9580 desc = mSuspendedEffects.valueAt(index); 9581 if (desc->mRefCount <= 0) { 9582 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9583 desc->mRefCount = 1; 9584 } 9585 if (--desc->mRefCount == 0) { 9586 Vector<const effect_uuid_t *> types; 9587 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9588 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9589 continue; 9590 } 9591 types.add(&mSuspendedEffects.valueAt(i)->mType); 9592 } 9593 for (size_t i = 0; i < types.size(); i++) { 9594 setEffectSuspended_l(types[i], false); 9595 } 9596 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9597 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9598 } 9599 } 9600} 9601 9602 9603// The volume effect is used for automated tests only 9604#ifndef OPENSL_ES_H_ 9605static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9606 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9607const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9608#endif //OPENSL_ES_H_ 9609 9610bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9611{ 9612 // auxiliary effects and visualizer are never suspended on output mix 9613 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9614 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9615 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9616 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9617 return false; 9618 } 9619 return true; 9620} 9621 9622void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9623{ 9624 effects.clear(); 9625 for (size_t i = 0; i < mEffects.size(); i++) { 9626 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9627 effects.add(mEffects[i]); 9628 } 9629 } 9630} 9631 9632sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9633 const effect_uuid_t *type) 9634{ 9635 sp<EffectModule> effect = getEffectFromType_l(type); 9636 return effect != 0 && effect->isEnabled() ? effect : 0; 9637} 9638 9639void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9640 bool enabled) 9641{ 9642 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9643 if (enabled) { 9644 if (index < 0) { 9645 // if the effect is not suspend check if all effects are suspended 9646 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9647 if (index < 0) { 9648 return; 9649 } 9650 if (!isEffectEligibleForSuspend(effect->desc())) { 9651 return; 9652 } 9653 setEffectSuspended_l(&effect->desc().type, enabled); 9654 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9655 if (index < 0) { 9656 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9657 return; 9658 } 9659 } 9660 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9661 effect->desc().type.timeLow); 9662 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9663 // if effect is requested to suspended but was not yet enabled, supend it now. 9664 if (desc->mEffect == 0) { 9665 desc->mEffect = effect; 9666 effect->setEnabled(false); 9667 effect->setSuspended(true); 9668 } 9669 } else { 9670 if (index < 0) { 9671 return; 9672 } 9673 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9674 effect->desc().type.timeLow); 9675 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9676 desc->mEffect.clear(); 9677 effect->setSuspended(false); 9678 } 9679} 9680 9681#undef LOG_TAG 9682#define LOG_TAG "AudioFlinger" 9683 9684// ---------------------------------------------------------------------------- 9685 9686status_t AudioFlinger::onTransact( 9687 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9688{ 9689 return BnAudioFlinger::onTransact(code, data, reply, flags); 9690} 9691 9692}; // namespace android 9693