AudioFlinger.cpp revision 106e8a42038f9e90d5ff97f8ab6f1a42258bde9e
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 "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "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 us whether it wants double-buffering or N-buffering, 173// so we 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() will remove first entry from mRecordThreads 262 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 263 } 264 while (!mPlaybackThreads.isEmpty()) { 265 // closeOutput() will remove first entry from mPlaybackThreads 266 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 267 } 268 269 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 270 // no mHardwareLock needed, as there are no other references to this 271 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 272 delete mAudioHwDevs.valueAt(i); 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 } else { 295 // check a match for the requested module handle 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 297 if (audioHwDevice != NULL) { 298 return audioHwDevice; 299 } 300 } 301 // then try to find a module supporting the requested device. 302 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 303 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 304 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 305 if ((dev->get_supported_devices(dev) & devices) == devices) 306 return audioHwDevice; 307 } 308 309 return NULL; 310} 311 312void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 313{ 314 const size_t SIZE = 256; 315 char buffer[SIZE]; 316 String8 result; 317 318 result.append("Clients:\n"); 319 for (size_t i = 0; i < mClients.size(); ++i) { 320 sp<Client> client = mClients.valueAt(i).promote(); 321 if (client != 0) { 322 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 323 result.append(buffer); 324 } 325 } 326 327 result.append("Global session refs:\n"); 328 result.append(" session pid count\n"); 329 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 330 AudioSessionRef *r = mAudioSessionRefs[i]; 331 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 332 result.append(buffer); 333 } 334 write(fd, result.string(), result.size()); 335} 336 337 338void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 339{ 340 const size_t SIZE = 256; 341 char buffer[SIZE]; 342 String8 result; 343 hardware_call_state hardwareStatus = mHardwareStatus; 344 345 snprintf(buffer, SIZE, "Hardware status: %d\n" 346 "Standby Time mSec: %u\n", 347 hardwareStatus, 348 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351} 352 353void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 354{ 355 const size_t SIZE = 256; 356 char buffer[SIZE]; 357 String8 result; 358 snprintf(buffer, SIZE, "Permission Denial: " 359 "can't dump AudioFlinger from pid=%d, uid=%d\n", 360 IPCThreadState::self()->getCallingPid(), 361 IPCThreadState::self()->getCallingUid()); 362 result.append(buffer); 363 write(fd, result.string(), result.size()); 364} 365 366static bool tryLock(Mutex& mutex) 367{ 368 bool locked = false; 369 for (int i = 0; i < kDumpLockRetries; ++i) { 370 if (mutex.tryLock() == NO_ERROR) { 371 locked = true; 372 break; 373 } 374 usleep(kDumpLockSleepUs); 375 } 376 return locked; 377} 378 379status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 380{ 381 if (!dumpAllowed()) { 382 dumpPermissionDenial(fd, args); 383 } else { 384 // get state of hardware lock 385 bool hardwareLocked = tryLock(mHardwareLock); 386 if (!hardwareLocked) { 387 String8 result(kHardwareLockedString); 388 write(fd, result.string(), result.size()); 389 } else { 390 mHardwareLock.unlock(); 391 } 392 393 bool locked = tryLock(mLock); 394 395 // failed to lock - AudioFlinger is probably deadlocked 396 if (!locked) { 397 String8 result(kDeadlockedString); 398 write(fd, result.string(), result.size()); 399 } 400 401 dumpClients(fd, args); 402 dumpInternals(fd, args); 403 404 // dump playback threads 405 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 406 mPlaybackThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump record threads 410 for (size_t i = 0; i < mRecordThreads.size(); i++) { 411 mRecordThreads.valueAt(i)->dump(fd, args); 412 } 413 414 // dump all hardware devs 415 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 416 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 417 dev->dump(dev, fd); 418 } 419 if (locked) mLock.unlock(); 420 } 421 return NO_ERROR; 422} 423 424sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 425{ 426 // If pid is already in the mClients wp<> map, then use that entry 427 // (for which promote() is always != 0), otherwise create a new entry and Client. 428 sp<Client> client = mClients.valueFor(pid).promote(); 429 if (client == 0) { 430 client = new Client(this, pid); 431 mClients.add(pid, client); 432 } 433 434 return client; 435} 436 437// IAudioFlinger interface 438 439 440sp<IAudioTrack> AudioFlinger::createTrack( 441 pid_t pid, 442 audio_stream_type_t streamType, 443 uint32_t sampleRate, 444 audio_format_t format, 445 audio_channel_mask_t channelMask, 446 int frameCount, 447 IAudioFlinger::track_flags_t flags, 448 const sp<IMemory>& sharedBuffer, 449 audio_io_handle_t output, 450 pid_t tid, 451 int *sessionId, 452 status_t *status) 453{ 454 sp<PlaybackThread::Track> track; 455 sp<TrackHandle> trackHandle; 456 sp<Client> client; 457 status_t lStatus; 458 int lSessionId; 459 460 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 461 // but if someone uses binder directly they could bypass that and cause us to crash 462 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 463 ALOGE("createTrack() invalid stream type %d", streamType); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 { 469 Mutex::Autolock _l(mLock); 470 PlaybackThread *thread = checkPlaybackThread_l(output); 471 PlaybackThread *effectThread = NULL; 472 if (thread == NULL) { 473 ALOGE("unknown output thread"); 474 lStatus = BAD_VALUE; 475 goto Exit; 476 } 477 478 client = registerPid_l(pid); 479 480 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 481 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 482 // check if an effect chain with the same session ID is present on another 483 // output thread and move it here. 484 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 485 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 486 if (mPlaybackThreads.keyAt(i) != output) { 487 uint32_t sessions = t->hasAudioSession(*sessionId); 488 if (sessions & PlaybackThread::EFFECT_SESSION) { 489 effectThread = t.get(); 490 break; 491 } 492 } 493 } 494 lSessionId = *sessionId; 495 } else { 496 // if no audio session id is provided, create one here 497 lSessionId = nextUniqueId(); 498 if (sessionId != NULL) { 499 *sessionId = lSessionId; 500 } 501 } 502 ALOGV("createTrack() lSessionId: %d", lSessionId); 503 504 track = thread->createTrack_l(client, streamType, sampleRate, format, 505 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 506 507 // move effect chain to this output thread if an effect on same session was waiting 508 // for a track to be created 509 if (lStatus == NO_ERROR && effectThread != NULL) { 510 Mutex::Autolock _dl(thread->mLock); 511 Mutex::Autolock _sl(effectThread->mLock); 512 moveEffectChain_l(lSessionId, effectThread, thread, true); 513 } 514 515 // Look for sync events awaiting for a session to be used. 516 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 517 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 518 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 519 if (lStatus == NO_ERROR) { 520 (void) track->setSyncEvent(mPendingSyncEvents[i]); 521 } else { 522 mPendingSyncEvents[i]->cancel(); 523 } 524 mPendingSyncEvents.removeAt(i); 525 i--; 526 } 527 } 528 } 529 } 530 if (lStatus == NO_ERROR) { 531 trackHandle = new TrackHandle(track); 532 } else { 533 // remove local strong reference to Client before deleting the Track so that the Client 534 // destructor is called by the TrackBase destructor with mLock held 535 client.clear(); 536 track.clear(); 537 } 538 539Exit: 540 if (status != NULL) { 541 *status = lStatus; 542 } 543 return trackHandle; 544} 545 546uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("sampleRate() unknown thread %d", output); 552 return 0; 553 } 554 return thread->sampleRate(); 555} 556 557int AudioFlinger::channelCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("channelCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->channelCount(); 566} 567 568audio_format_t AudioFlinger::format(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("format() unknown thread %d", output); 574 return AUDIO_FORMAT_INVALID; 575 } 576 return thread->format(); 577} 578 579size_t AudioFlinger::frameCount(audio_io_handle_t output) const 580{ 581 Mutex::Autolock _l(mLock); 582 PlaybackThread *thread = checkPlaybackThread_l(output); 583 if (thread == NULL) { 584 ALOGW("frameCount() unknown thread %d", output); 585 return 0; 586 } 587 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 588 // should examine all callers and fix them to handle smaller counts 589 return thread->frameCount(); 590} 591 592uint32_t AudioFlinger::latency(audio_io_handle_t output) const 593{ 594 Mutex::Autolock _l(mLock); 595 PlaybackThread *thread = checkPlaybackThread_l(output); 596 if (thread == NULL) { 597 ALOGW("latency() unknown thread %d", output); 598 return 0; 599 } 600 return thread->latency(); 601} 602 603status_t AudioFlinger::setMasterVolume(float value) 604{ 605 status_t ret = initCheck(); 606 if (ret != NO_ERROR) { 607 return ret; 608 } 609 610 // check calling permissions 611 if (!settingsAllowed()) { 612 return PERMISSION_DENIED; 613 } 614 615 Mutex::Autolock _l(mLock); 616 mMasterVolume = value; 617 618 // Set master volume in the HALs which support it. 619 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 620 AutoMutex lock(mHardwareLock); 621 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 622 623 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 624 if (dev->canSetMasterVolume()) { 625 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 626 } 627 mHardwareStatus = AUDIO_HW_IDLE; 628 } 629 630 // Now set the master volume in each playback thread. Playback threads 631 // assigned to HALs which do not have master volume support will apply 632 // master volume during the mix operation. Threads with HALs which do 633 // support master volume will simply ignore the setting. 634 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 635 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 636 637 return NO_ERROR; 638} 639 640status_t AudioFlinger::setMode(audio_mode_t mode) 641{ 642 status_t ret = initCheck(); 643 if (ret != NO_ERROR) { 644 return ret; 645 } 646 647 // check calling permissions 648 if (!settingsAllowed()) { 649 return PERMISSION_DENIED; 650 } 651 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 652 ALOGW("Illegal value: setMode(%d)", mode); 653 return BAD_VALUE; 654 } 655 656 { // scope for the lock 657 AutoMutex lock(mHardwareLock); 658 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 659 mHardwareStatus = AUDIO_HW_SET_MODE; 660 ret = dev->set_mode(dev, mode); 661 mHardwareStatus = AUDIO_HW_IDLE; 662 } 663 664 if (NO_ERROR == ret) { 665 Mutex::Autolock _l(mLock); 666 mMode = mode; 667 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 668 mPlaybackThreads.valueAt(i)->setMode(mode); 669 } 670 671 return ret; 672} 673 674status_t AudioFlinger::setMicMute(bool state) 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return ret; 679 } 680 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 AutoMutex lock(mHardwareLock); 687 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 688 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 689 ret = dev->set_mic_mute(dev, state); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return ret; 692} 693 694bool AudioFlinger::getMicMute() const 695{ 696 status_t ret = initCheck(); 697 if (ret != NO_ERROR) { 698 return false; 699 } 700 701 bool state = AUDIO_MODE_INVALID; 702 AutoMutex lock(mHardwareLock); 703 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 704 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 705 dev->get_mic_mute(dev, &state); 706 mHardwareStatus = AUDIO_HW_IDLE; 707 return state; 708} 709 710status_t AudioFlinger::setMasterMute(bool muted) 711{ 712 status_t ret = initCheck(); 713 if (ret != NO_ERROR) { 714 return ret; 715 } 716 717 // check calling permissions 718 if (!settingsAllowed()) { 719 return PERMISSION_DENIED; 720 } 721 722 Mutex::Autolock _l(mLock); 723 mMasterMute = muted; 724 725 // Set master mute in the HALs which support it. 726 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 727 AutoMutex lock(mHardwareLock); 728 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 729 730 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 731 if (dev->canSetMasterMute()) { 732 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 733 } 734 mHardwareStatus = AUDIO_HW_IDLE; 735 } 736 737 // Now set the master mute in each playback thread. Playback threads 738 // assigned to HALs which do not have master mute support will apply master 739 // mute during the mix operation. Threads with HALs which do support master 740 // mute will simply ignore the setting. 741 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 742 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 743 744 return NO_ERROR; 745} 746 747float AudioFlinger::masterVolume() const 748{ 749 Mutex::Autolock _l(mLock); 750 return masterVolume_l(); 751} 752 753bool AudioFlinger::masterMute() const 754{ 755 Mutex::Autolock _l(mLock); 756 return masterMute_l(); 757} 758 759float AudioFlinger::masterVolume_l() const 760{ 761 return mMasterVolume; 762} 763 764bool AudioFlinger::masterMute_l() const 765{ 766 return mMasterMute; 767} 768 769status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 770 audio_io_handle_t output) 771{ 772 // check calling permissions 773 if (!settingsAllowed()) { 774 return PERMISSION_DENIED; 775 } 776 777 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 778 ALOGE("setStreamVolume() invalid stream %d", stream); 779 return BAD_VALUE; 780 } 781 782 AutoMutex lock(mLock); 783 PlaybackThread *thread = NULL; 784 if (output) { 785 thread = checkPlaybackThread_l(output); 786 if (thread == NULL) { 787 return BAD_VALUE; 788 } 789 } 790 791 mStreamTypes[stream].volume = value; 792 793 if (thread == NULL) { 794 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 795 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 796 } 797 } else { 798 thread->setStreamVolume(stream, value); 799 } 800 801 return NO_ERROR; 802} 803 804status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 805{ 806 // check calling permissions 807 if (!settingsAllowed()) { 808 return PERMISSION_DENIED; 809 } 810 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 812 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 813 ALOGE("setStreamMute() invalid stream %d", stream); 814 return BAD_VALUE; 815 } 816 817 AutoMutex lock(mLock); 818 mStreamTypes[stream].mute = muted; 819 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 820 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 821 822 return NO_ERROR; 823} 824 825float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 826{ 827 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 828 return 0.0f; 829 } 830 831 AutoMutex lock(mLock); 832 float volume; 833 if (output) { 834 PlaybackThread *thread = checkPlaybackThread_l(output); 835 if (thread == NULL) { 836 return 0.0f; 837 } 838 volume = thread->streamVolume(stream); 839 } else { 840 volume = streamVolume_l(stream); 841 } 842 843 return volume; 844} 845 846bool AudioFlinger::streamMute(audio_stream_type_t stream) const 847{ 848 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 849 return true; 850 } 851 852 AutoMutex lock(mLock); 853 return streamMute_l(stream); 854} 855 856status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 857{ 858 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 859 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 860 // check calling permissions 861 if (!settingsAllowed()) { 862 return PERMISSION_DENIED; 863 } 864 865 // ioHandle == 0 means the parameters are global to the audio hardware interface 866 if (ioHandle == 0) { 867 Mutex::Autolock _l(mLock); 868 status_t final_result = NO_ERROR; 869 { 870 AutoMutex lock(mHardwareLock); 871 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 872 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 873 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 874 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 875 final_result = result ?: final_result; 876 } 877 mHardwareStatus = AUDIO_HW_IDLE; 878 } 879 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 880 AudioParameter param = AudioParameter(keyValuePairs); 881 String8 value; 882 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 883 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 884 if (mBtNrecIsOff != btNrecIsOff) { 885 for (size_t i = 0; i < mRecordThreads.size(); i++) { 886 sp<RecordThread> thread = mRecordThreads.valueAt(i); 887 audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL; 888 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 889 // collect all of the thread's session IDs 890 KeyedVector<int, bool> ids = thread->sessionIds(); 891 // suspend effects associated with those session IDs 892 for (size_t j = 0; j < ids.size(); ++j) { 893 int sessionId = ids.keyAt(j); 894 thread->setEffectSuspended(FX_IID_AEC, 895 suspend, 896 sessionId); 897 thread->setEffectSuspended(FX_IID_NS, 898 suspend, 899 sessionId); 900 } 901 } 902 mBtNrecIsOff = btNrecIsOff; 903 } 904 } 905 String8 screenState; 906 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 907 bool isOff = screenState == "off"; 908 if (isOff != (gScreenState & 1)) { 909 gScreenState = ((gScreenState & ~1) + 2) | isOff; 910 } 911 } 912 return final_result; 913 } 914 915 // hold a strong ref on thread in case closeOutput() or closeInput() is called 916 // and the thread is exited once the lock is released 917 sp<ThreadBase> thread; 918 { 919 Mutex::Autolock _l(mLock); 920 thread = checkPlaybackThread_l(ioHandle); 921 if (thread == 0) { 922 thread = checkRecordThread_l(ioHandle); 923 } else if (thread == primaryPlaybackThread_l()) { 924 // indicate output device change to all input threads for pre processing 925 AudioParameter param = AudioParameter(keyValuePairs); 926 int value; 927 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 928 (value != 0)) { 929 for (size_t i = 0; i < mRecordThreads.size(); i++) { 930 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 931 } 932 } 933 } 934 } 935 if (thread != 0) { 936 return thread->setParameters(keyValuePairs); 937 } 938 return BAD_VALUE; 939} 940 941String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 942{ 943// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 944// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 945 946 Mutex::Autolock _l(mLock); 947 948 if (ioHandle == 0) { 949 String8 out_s8; 950 951 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 952 char *s; 953 { 954 AutoMutex lock(mHardwareLock); 955 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 956 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 957 s = dev->get_parameters(dev, keys.string()); 958 mHardwareStatus = AUDIO_HW_IDLE; 959 } 960 out_s8 += String8(s ? s : ""); 961 free(s); 962 } 963 return out_s8; 964 } 965 966 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 967 if (playbackThread != NULL) { 968 return playbackThread->getParameters(keys); 969 } 970 RecordThread *recordThread = checkRecordThread_l(ioHandle); 971 if (recordThread != NULL) { 972 return recordThread->getParameters(keys); 973 } 974 return String8(""); 975} 976 977size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 978 audio_channel_mask_t channelMask) const 979{ 980 status_t ret = initCheck(); 981 if (ret != NO_ERROR) { 982 return 0; 983 } 984 985 AutoMutex lock(mHardwareLock); 986 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 987 struct audio_config config = { 988 sample_rate: sampleRate, 989 channel_mask: channelMask, 990 format: format, 991 }; 992 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 993 size_t size = dev->get_input_buffer_size(dev, &config); 994 mHardwareStatus = AUDIO_HW_IDLE; 995 return size; 996} 997 998unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 999{ 1000 Mutex::Autolock _l(mLock); 1001 1002 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1003 if (recordThread != NULL) { 1004 return recordThread->getInputFramesLost(); 1005 } 1006 return 0; 1007} 1008 1009status_t AudioFlinger::setVoiceVolume(float value) 1010{ 1011 status_t ret = initCheck(); 1012 if (ret != NO_ERROR) { 1013 return ret; 1014 } 1015 1016 // check calling permissions 1017 if (!settingsAllowed()) { 1018 return PERMISSION_DENIED; 1019 } 1020 1021 AutoMutex lock(mHardwareLock); 1022 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1023 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1024 ret = dev->set_voice_volume(dev, value); 1025 mHardwareStatus = AUDIO_HW_IDLE; 1026 1027 return ret; 1028} 1029 1030status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1031 audio_io_handle_t output) const 1032{ 1033 status_t status; 1034 1035 Mutex::Autolock _l(mLock); 1036 1037 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1038 if (playbackThread != NULL) { 1039 return playbackThread->getRenderPosition(halFrames, dspFrames); 1040 } 1041 1042 return BAD_VALUE; 1043} 1044 1045void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1046{ 1047 1048 Mutex::Autolock _l(mLock); 1049 1050 pid_t pid = IPCThreadState::self()->getCallingPid(); 1051 if (mNotificationClients.indexOfKey(pid) < 0) { 1052 sp<NotificationClient> notificationClient = new NotificationClient(this, 1053 client, 1054 pid); 1055 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1056 1057 mNotificationClients.add(pid, notificationClient); 1058 1059 sp<IBinder> binder = client->asBinder(); 1060 binder->linkToDeath(notificationClient); 1061 1062 // the config change is always sent from playback or record threads to avoid deadlock 1063 // with AudioSystem::gLock 1064 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1065 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1066 } 1067 1068 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1069 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1070 } 1071 } 1072} 1073 1074void AudioFlinger::removeNotificationClient(pid_t pid) 1075{ 1076 Mutex::Autolock _l(mLock); 1077 1078 mNotificationClients.removeItem(pid); 1079 1080 ALOGV("%d died, releasing its sessions", pid); 1081 size_t num = mAudioSessionRefs.size(); 1082 bool removed = false; 1083 for (size_t i = 0; i< num; ) { 1084 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1085 ALOGV(" pid %d @ %d", ref->mPid, i); 1086 if (ref->mPid == pid) { 1087 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1088 mAudioSessionRefs.removeAt(i); 1089 delete ref; 1090 removed = true; 1091 num--; 1092 } else { 1093 i++; 1094 } 1095 } 1096 if (removed) { 1097 purgeStaleEffects_l(); 1098 } 1099} 1100 1101// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1102void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1103{ 1104 size_t size = mNotificationClients.size(); 1105 for (size_t i = 0; i < size; i++) { 1106 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1107 param2); 1108 } 1109} 1110 1111// removeClient_l() must be called with AudioFlinger::mLock held 1112void AudioFlinger::removeClient_l(pid_t pid) 1113{ 1114 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1115 mClients.removeItem(pid); 1116} 1117 1118// getEffectThread_l() must be called with AudioFlinger::mLock held 1119sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1120{ 1121 sp<PlaybackThread> thread; 1122 1123 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1124 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1125 ALOG_ASSERT(thread == 0); 1126 thread = mPlaybackThreads.valueAt(i); 1127 } 1128 } 1129 1130 return thread; 1131} 1132 1133// ---------------------------------------------------------------------------- 1134 1135AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1136 audio_devices_t device, type_t type) 1137 : Thread(false), 1138 mType(type), 1139 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1140 // mChannelMask 1141 mChannelCount(0), 1142 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1143 mParamStatus(NO_ERROR), 1144 mStandby(false), mDevice(device), mId(id), 1145 mDeathRecipient(new PMDeathRecipient(this)) 1146{ 1147} 1148 1149AudioFlinger::ThreadBase::~ThreadBase() 1150{ 1151 mParamCond.broadcast(); 1152 // do not lock the mutex in destructor 1153 releaseWakeLock_l(); 1154 if (mPowerManager != 0) { 1155 sp<IBinder> binder = mPowerManager->asBinder(); 1156 binder->unlinkToDeath(mDeathRecipient); 1157 } 1158} 1159 1160void AudioFlinger::ThreadBase::exit() 1161{ 1162 ALOGV("ThreadBase::exit"); 1163 { 1164 // This lock prevents the following race in thread (uniprocessor for illustration): 1165 // if (!exitPending()) { 1166 // // context switch from here to exit() 1167 // // exit() calls requestExit(), what exitPending() observes 1168 // // exit() calls signal(), which is dropped since no waiters 1169 // // context switch back from exit() to here 1170 // mWaitWorkCV.wait(...); 1171 // // now thread is hung 1172 // } 1173 AutoMutex lock(mLock); 1174 requestExit(); 1175 mWaitWorkCV.signal(); 1176 } 1177 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1178 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1179 requestExitAndWait(); 1180} 1181 1182status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1183{ 1184 status_t status; 1185 1186 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1187 Mutex::Autolock _l(mLock); 1188 1189 mNewParameters.add(keyValuePairs); 1190 mWaitWorkCV.signal(); 1191 // wait condition with timeout in case the thread loop has exited 1192 // before the request could be processed 1193 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1194 status = mParamStatus; 1195 mWaitWorkCV.signal(); 1196 } else { 1197 status = TIMED_OUT; 1198 } 1199 return status; 1200} 1201 1202void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1203{ 1204 Mutex::Autolock _l(mLock); 1205 sendConfigEvent_l(event, param); 1206} 1207 1208// sendConfigEvent_l() must be called with ThreadBase::mLock held 1209void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1210{ 1211 ConfigEvent configEvent; 1212 configEvent.mEvent = event; 1213 configEvent.mParam = param; 1214 mConfigEvents.add(configEvent); 1215 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1216 mWaitWorkCV.signal(); 1217} 1218 1219void AudioFlinger::ThreadBase::processConfigEvents() 1220{ 1221 mLock.lock(); 1222 while (!mConfigEvents.isEmpty()) { 1223 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1224 ConfigEvent configEvent = mConfigEvents[0]; 1225 mConfigEvents.removeAt(0); 1226 // release mLock before locking AudioFlinger mLock: lock order is always 1227 // AudioFlinger then ThreadBase to avoid cross deadlock 1228 mLock.unlock(); 1229 mAudioFlinger->mLock.lock(); 1230 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1231 mAudioFlinger->mLock.unlock(); 1232 mLock.lock(); 1233 } 1234 mLock.unlock(); 1235} 1236 1237void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1238{ 1239 const size_t SIZE = 256; 1240 char buffer[SIZE]; 1241 String8 result; 1242 1243 bool locked = tryLock(mLock); 1244 if (!locked) { 1245 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1246 write(fd, buffer, strlen(buffer)); 1247 } 1248 1249 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1250 result.append(buffer); 1251 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1252 result.append(buffer); 1253 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1254 result.append(buffer); 1255 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1256 result.append(buffer); 1257 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1258 result.append(buffer); 1259 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1260 result.append(buffer); 1261 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1262 result.append(buffer); 1263 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1264 result.append(buffer); 1265 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1266 result.append(buffer); 1267 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1268 result.append(buffer); 1269 1270 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1271 result.append(buffer); 1272 result.append(" Index Command"); 1273 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1274 snprintf(buffer, SIZE, "\n %02d ", i); 1275 result.append(buffer); 1276 result.append(mNewParameters[i]); 1277 } 1278 1279 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1280 result.append(buffer); 1281 snprintf(buffer, SIZE, " Index event param\n"); 1282 result.append(buffer); 1283 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1284 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1285 result.append(buffer); 1286 } 1287 result.append("\n"); 1288 1289 write(fd, result.string(), result.size()); 1290 1291 if (locked) { 1292 mLock.unlock(); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1297{ 1298 const size_t SIZE = 256; 1299 char buffer[SIZE]; 1300 String8 result; 1301 1302 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1303 write(fd, buffer, strlen(buffer)); 1304 1305 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1306 sp<EffectChain> chain = mEffectChains[i]; 1307 if (chain != 0) { 1308 chain->dump(fd, args); 1309 } 1310 } 1311} 1312 1313void AudioFlinger::ThreadBase::acquireWakeLock() 1314{ 1315 Mutex::Autolock _l(mLock); 1316 acquireWakeLock_l(); 1317} 1318 1319void AudioFlinger::ThreadBase::acquireWakeLock_l() 1320{ 1321 if (mPowerManager == 0) { 1322 // use checkService() to avoid blocking if power service is not up yet 1323 sp<IBinder> binder = 1324 defaultServiceManager()->checkService(String16("power")); 1325 if (binder == 0) { 1326 ALOGW("Thread %s cannot connect to the power manager service", mName); 1327 } else { 1328 mPowerManager = interface_cast<IPowerManager>(binder); 1329 binder->linkToDeath(mDeathRecipient); 1330 } 1331 } 1332 if (mPowerManager != 0) { 1333 sp<IBinder> binder = new BBinder(); 1334 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1335 binder, 1336 String16(mName)); 1337 if (status == NO_ERROR) { 1338 mWakeLockToken = binder; 1339 } 1340 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1341 } 1342} 1343 1344void AudioFlinger::ThreadBase::releaseWakeLock() 1345{ 1346 Mutex::Autolock _l(mLock); 1347 releaseWakeLock_l(); 1348} 1349 1350void AudioFlinger::ThreadBase::releaseWakeLock_l() 1351{ 1352 if (mWakeLockToken != 0) { 1353 ALOGV("releaseWakeLock_l() %s", mName); 1354 if (mPowerManager != 0) { 1355 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1356 } 1357 mWakeLockToken.clear(); 1358 } 1359} 1360 1361void AudioFlinger::ThreadBase::clearPowerManager() 1362{ 1363 Mutex::Autolock _l(mLock); 1364 releaseWakeLock_l(); 1365 mPowerManager.clear(); 1366} 1367 1368void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1369{ 1370 sp<ThreadBase> thread = mThread.promote(); 1371 if (thread != 0) { 1372 thread->clearPowerManager(); 1373 } 1374 ALOGW("power manager service died !!!"); 1375} 1376 1377void AudioFlinger::ThreadBase::setEffectSuspended( 1378 const effect_uuid_t *type, bool suspend, int sessionId) 1379{ 1380 Mutex::Autolock _l(mLock); 1381 setEffectSuspended_l(type, suspend, sessionId); 1382} 1383 1384void AudioFlinger::ThreadBase::setEffectSuspended_l( 1385 const effect_uuid_t *type, bool suspend, int sessionId) 1386{ 1387 sp<EffectChain> chain = getEffectChain_l(sessionId); 1388 if (chain != 0) { 1389 if (type != NULL) { 1390 chain->setEffectSuspended_l(type, suspend); 1391 } else { 1392 chain->setEffectSuspendedAll_l(suspend); 1393 } 1394 } 1395 1396 updateSuspendedSessions_l(type, suspend, sessionId); 1397} 1398 1399void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1400{ 1401 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1402 if (index < 0) { 1403 return; 1404 } 1405 1406 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1407 mSuspendedSessions.valueAt(index); 1408 1409 for (size_t i = 0; i < sessionEffects.size(); i++) { 1410 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1411 for (int j = 0; j < desc->mRefCount; j++) { 1412 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1413 chain->setEffectSuspendedAll_l(true); 1414 } else { 1415 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1416 desc->mType.timeLow); 1417 chain->setEffectSuspended_l(&desc->mType, true); 1418 } 1419 } 1420 } 1421} 1422 1423void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1424 bool suspend, 1425 int sessionId) 1426{ 1427 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1428 1429 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1430 1431 if (suspend) { 1432 if (index >= 0) { 1433 sessionEffects = mSuspendedSessions.valueAt(index); 1434 } else { 1435 mSuspendedSessions.add(sessionId, sessionEffects); 1436 } 1437 } else { 1438 if (index < 0) { 1439 return; 1440 } 1441 sessionEffects = mSuspendedSessions.valueAt(index); 1442 } 1443 1444 1445 int key = EffectChain::kKeyForSuspendAll; 1446 if (type != NULL) { 1447 key = type->timeLow; 1448 } 1449 index = sessionEffects.indexOfKey(key); 1450 1451 sp<SuspendedSessionDesc> desc; 1452 if (suspend) { 1453 if (index >= 0) { 1454 desc = sessionEffects.valueAt(index); 1455 } else { 1456 desc = new SuspendedSessionDesc(); 1457 if (type != NULL) { 1458 desc->mType = *type; 1459 } 1460 sessionEffects.add(key, desc); 1461 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1462 } 1463 desc->mRefCount++; 1464 } else { 1465 if (index < 0) { 1466 return; 1467 } 1468 desc = sessionEffects.valueAt(index); 1469 if (--desc->mRefCount == 0) { 1470 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1471 sessionEffects.removeItemsAt(index); 1472 if (sessionEffects.isEmpty()) { 1473 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1474 sessionId); 1475 mSuspendedSessions.removeItem(sessionId); 1476 } 1477 } 1478 } 1479 if (!sessionEffects.isEmpty()) { 1480 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1481 } 1482} 1483 1484void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1485 bool enabled, 1486 int sessionId) 1487{ 1488 Mutex::Autolock _l(mLock); 1489 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1490} 1491 1492void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1493 bool enabled, 1494 int sessionId) 1495{ 1496 if (mType != RECORD) { 1497 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1498 // another session. This gives the priority to well behaved effect control panels 1499 // and applications not using global effects. 1500 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1501 // global effects 1502 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1503 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1504 } 1505 } 1506 1507 sp<EffectChain> chain = getEffectChain_l(sessionId); 1508 if (chain != 0) { 1509 chain->checkSuspendOnEffectEnabled(effect, enabled); 1510 } 1511} 1512 1513// ---------------------------------------------------------------------------- 1514 1515AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1516 AudioStreamOut* output, 1517 audio_io_handle_t id, 1518 audio_devices_t device, 1519 type_t type) 1520 : ThreadBase(audioFlinger, id, device, type), 1521 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1522 // mStreamTypes[] initialized in constructor body 1523 mOutput(output), 1524 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1525 mMixerStatus(MIXER_IDLE), 1526 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1527 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1528 mScreenState(gScreenState), 1529 // index 0 is reserved for normal mixer's submix 1530 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1531{ 1532 snprintf(mName, kNameLength, "AudioOut_%X", id); 1533 1534 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1535 // it would be safer to explicitly pass initial masterVolume/masterMute as 1536 // parameter. 1537 // 1538 // If the HAL we are using has support for master volume or master mute, 1539 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1540 // and the mute set to false). 1541 mMasterVolume = audioFlinger->masterVolume_l(); 1542 mMasterMute = audioFlinger->masterMute_l(); 1543 if (mOutput && mOutput->audioHwDev) { 1544 if (mOutput->audioHwDev->canSetMasterVolume()) { 1545 mMasterVolume = 1.0; 1546 } 1547 1548 if (mOutput->audioHwDev->canSetMasterMute()) { 1549 mMasterMute = false; 1550 } 1551 } 1552 1553 readOutputParameters(); 1554 1555 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1556 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1557 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1558 stream = (audio_stream_type_t) (stream + 1)) { 1559 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1560 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1561 } 1562 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1563 // because mAudioFlinger doesn't have one to copy from 1564} 1565 1566AudioFlinger::PlaybackThread::~PlaybackThread() 1567{ 1568 delete [] mMixBuffer; 1569} 1570 1571void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1572{ 1573 dumpInternals(fd, args); 1574 dumpTracks(fd, args); 1575 dumpEffectChains(fd, args); 1576} 1577 1578void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1579{ 1580 const size_t SIZE = 256; 1581 char buffer[SIZE]; 1582 String8 result; 1583 1584 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1585 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1586 const stream_type_t *st = &mStreamTypes[i]; 1587 if (i > 0) { 1588 result.appendFormat(", "); 1589 } 1590 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1591 if (st->mute) { 1592 result.append("M"); 1593 } 1594 } 1595 result.append("\n"); 1596 write(fd, result.string(), result.length()); 1597 result.clear(); 1598 1599 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1600 result.append(buffer); 1601 Track::appendDumpHeader(result); 1602 for (size_t i = 0; i < mTracks.size(); ++i) { 1603 sp<Track> track = mTracks[i]; 1604 if (track != 0) { 1605 track->dump(buffer, SIZE); 1606 result.append(buffer); 1607 } 1608 } 1609 1610 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1611 result.append(buffer); 1612 Track::appendDumpHeader(result); 1613 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1614 sp<Track> track = mActiveTracks[i].promote(); 1615 if (track != 0) { 1616 track->dump(buffer, SIZE); 1617 result.append(buffer); 1618 } 1619 } 1620 write(fd, result.string(), result.size()); 1621 1622 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1623 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1624 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1625 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1626} 1627 1628void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1629{ 1630 const size_t SIZE = 256; 1631 char buffer[SIZE]; 1632 String8 result; 1633 1634 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1635 result.append(buffer); 1636 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1637 result.append(buffer); 1638 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1639 result.append(buffer); 1640 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1641 result.append(buffer); 1642 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1643 result.append(buffer); 1644 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1645 result.append(buffer); 1646 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1647 result.append(buffer); 1648 write(fd, result.string(), result.size()); 1649 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1650 1651 dumpBase(fd, args); 1652} 1653 1654// Thread virtuals 1655status_t AudioFlinger::PlaybackThread::readyToRun() 1656{ 1657 status_t status = initCheck(); 1658 if (status == NO_ERROR) { 1659 ALOGI("AudioFlinger's thread %p ready to run", this); 1660 } else { 1661 ALOGE("No working audio driver found."); 1662 } 1663 return status; 1664} 1665 1666void AudioFlinger::PlaybackThread::onFirstRef() 1667{ 1668 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1669} 1670 1671// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1672sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1673 const sp<AudioFlinger::Client>& client, 1674 audio_stream_type_t streamType, 1675 uint32_t sampleRate, 1676 audio_format_t format, 1677 audio_channel_mask_t channelMask, 1678 int frameCount, 1679 const sp<IMemory>& sharedBuffer, 1680 int sessionId, 1681 IAudioFlinger::track_flags_t flags, 1682 pid_t tid, 1683 status_t *status) 1684{ 1685 sp<Track> track; 1686 status_t lStatus; 1687 1688 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1689 1690 // client expresses a preference for FAST, but we get the final say 1691 if (flags & IAudioFlinger::TRACK_FAST) { 1692 if ( 1693 // not timed 1694 (!isTimed) && 1695 // either of these use cases: 1696 ( 1697 // use case 1: shared buffer with any frame count 1698 ( 1699 (sharedBuffer != 0) 1700 ) || 1701 // use case 2: callback handler and frame count is default or at least as large as HAL 1702 ( 1703 (tid != -1) && 1704 ((frameCount == 0) || 1705 (frameCount >= (int) (mFrameCount * kFastTrackMultiplier))) 1706 ) 1707 ) && 1708 // PCM data 1709 audio_is_linear_pcm(format) && 1710 // mono or stereo 1711 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1712 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1713#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1714 // hardware sample rate 1715 (sampleRate == mSampleRate) && 1716#endif 1717 // normal mixer has an associated fast mixer 1718 hasFastMixer() && 1719 // there are sufficient fast track slots available 1720 (mFastTrackAvailMask != 0) 1721 // FIXME test that MixerThread for this fast track has a capable output HAL 1722 // FIXME add a permission test also? 1723 ) { 1724 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1725 if (frameCount == 0) { 1726 frameCount = mFrameCount * kFastTrackMultiplier; 1727 } 1728 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1729 frameCount, mFrameCount); 1730 } else { 1731 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1732 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1733 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1734 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1735 audio_is_linear_pcm(format), 1736 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1737 flags &= ~IAudioFlinger::TRACK_FAST; 1738 // For compatibility with AudioTrack calculation, buffer depth is forced 1739 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1740 // This is probably too conservative, but legacy application code may depend on it. 1741 // If you change this calculation, also review the start threshold which is related. 1742 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1743 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1744 if (minBufCount < 2) { 1745 minBufCount = 2; 1746 } 1747 int minFrameCount = mNormalFrameCount * minBufCount; 1748 if (frameCount < minFrameCount) { 1749 frameCount = minFrameCount; 1750 } 1751 } 1752 } 1753 1754 if (mType == DIRECT) { 1755 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1756 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1757 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1758 "for output %p with format %d", 1759 sampleRate, format, channelMask, mOutput, mFormat); 1760 lStatus = BAD_VALUE; 1761 goto Exit; 1762 } 1763 } 1764 } else { 1765 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1766 if (sampleRate > mSampleRate*2) { 1767 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1768 lStatus = BAD_VALUE; 1769 goto Exit; 1770 } 1771 } 1772 1773 lStatus = initCheck(); 1774 if (lStatus != NO_ERROR) { 1775 ALOGE("Audio driver not initialized."); 1776 goto Exit; 1777 } 1778 1779 { // scope for mLock 1780 Mutex::Autolock _l(mLock); 1781 1782 // all tracks in same audio session must share the same routing strategy otherwise 1783 // conflicts will happen when tracks are moved from one output to another by audio policy 1784 // manager 1785 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1786 for (size_t i = 0; i < mTracks.size(); ++i) { 1787 sp<Track> t = mTracks[i]; 1788 if (t != 0 && !t->isOutputTrack()) { 1789 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1790 if (sessionId == t->sessionId() && strategy != actual) { 1791 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1792 strategy, actual); 1793 lStatus = BAD_VALUE; 1794 goto Exit; 1795 } 1796 } 1797 } 1798 1799 if (!isTimed) { 1800 track = new Track(this, client, streamType, sampleRate, format, 1801 channelMask, frameCount, sharedBuffer, sessionId, flags); 1802 } else { 1803 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1804 channelMask, frameCount, sharedBuffer, sessionId); 1805 } 1806 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1807 lStatus = NO_MEMORY; 1808 goto Exit; 1809 } 1810 mTracks.add(track); 1811 1812 sp<EffectChain> chain = getEffectChain_l(sessionId); 1813 if (chain != 0) { 1814 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1815 track->setMainBuffer(chain->inBuffer()); 1816 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1817 chain->incTrackCnt(); 1818 } 1819 } 1820 1821 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1822 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1823 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1824 // so ask activity manager to do this on our behalf 1825 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1826 if (err != 0) { 1827 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1828 kPriorityAudioApp, callingPid, tid, err); 1829 } 1830 } 1831 1832 lStatus = NO_ERROR; 1833 1834Exit: 1835 if (status) { 1836 *status = lStatus; 1837 } 1838 return track; 1839} 1840 1841uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1842{ 1843 if (mFastMixer != NULL) { 1844 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1845 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1846 } 1847 return latency; 1848} 1849 1850uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1851{ 1852 return latency; 1853} 1854 1855uint32_t AudioFlinger::PlaybackThread::latency() const 1856{ 1857 Mutex::Autolock _l(mLock); 1858 return latency_l(); 1859} 1860uint32_t AudioFlinger::PlaybackThread::latency_l() const 1861{ 1862 if (initCheck() == NO_ERROR) { 1863 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1864 } else { 1865 return 0; 1866 } 1867} 1868 1869void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1870{ 1871 Mutex::Autolock _l(mLock); 1872 // Don't apply master volume in SW if our HAL can do it for us. 1873 if (mOutput && mOutput->audioHwDev && 1874 mOutput->audioHwDev->canSetMasterVolume()) { 1875 mMasterVolume = 1.0; 1876 } else { 1877 mMasterVolume = value; 1878 } 1879} 1880 1881void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1882{ 1883 Mutex::Autolock _l(mLock); 1884 // Don't apply master mute in SW if our HAL can do it for us. 1885 if (mOutput && mOutput->audioHwDev && 1886 mOutput->audioHwDev->canSetMasterMute()) { 1887 mMasterMute = false; 1888 } else { 1889 mMasterMute = muted; 1890 } 1891} 1892 1893void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1894{ 1895 Mutex::Autolock _l(mLock); 1896 mStreamTypes[stream].volume = value; 1897} 1898 1899void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1900{ 1901 Mutex::Autolock _l(mLock); 1902 mStreamTypes[stream].mute = muted; 1903} 1904 1905float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1906{ 1907 Mutex::Autolock _l(mLock); 1908 return mStreamTypes[stream].volume; 1909} 1910 1911// addTrack_l() must be called with ThreadBase::mLock held 1912status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1913{ 1914 status_t status = ALREADY_EXISTS; 1915 1916 // set retry count for buffer fill 1917 track->mRetryCount = kMaxTrackStartupRetries; 1918 if (mActiveTracks.indexOf(track) < 0) { 1919 // the track is newly added, make sure it fills up all its 1920 // buffers before playing. This is to ensure the client will 1921 // effectively get the latency it requested. 1922 track->mFillingUpStatus = Track::FS_FILLING; 1923 track->mResetDone = false; 1924 track->mPresentationCompleteFrames = 0; 1925 mActiveTracks.add(track); 1926 if (track->mainBuffer() != mMixBuffer) { 1927 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1928 if (chain != 0) { 1929 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1930 chain->incActiveTrackCnt(); 1931 } 1932 } 1933 1934 status = NO_ERROR; 1935 } 1936 1937 ALOGV("mWaitWorkCV.broadcast"); 1938 mWaitWorkCV.broadcast(); 1939 1940 return status; 1941} 1942 1943// destroyTrack_l() must be called with ThreadBase::mLock held 1944void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1945{ 1946 track->mState = TrackBase::TERMINATED; 1947 // active tracks are removed by threadLoop() 1948 if (mActiveTracks.indexOf(track) < 0) { 1949 removeTrack_l(track); 1950 } 1951} 1952 1953void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1954{ 1955 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1956 mTracks.remove(track); 1957 deleteTrackName_l(track->name()); 1958 // redundant as track is about to be destroyed, for dumpsys only 1959 track->mName = -1; 1960 if (track->isFastTrack()) { 1961 int index = track->mFastIndex; 1962 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1963 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1964 mFastTrackAvailMask |= 1 << index; 1965 // redundant as track is about to be destroyed, for dumpsys only 1966 track->mFastIndex = -1; 1967 } 1968 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1969 if (chain != 0) { 1970 chain->decTrackCnt(); 1971 } 1972} 1973 1974String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1975{ 1976 String8 out_s8 = String8(""); 1977 char *s; 1978 1979 Mutex::Autolock _l(mLock); 1980 if (initCheck() != NO_ERROR) { 1981 return out_s8; 1982 } 1983 1984 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1985 out_s8 = String8(s); 1986 free(s); 1987 return out_s8; 1988} 1989 1990// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1991void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1992 AudioSystem::OutputDescriptor desc; 1993 void *param2 = NULL; 1994 1995 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1996 1997 switch (event) { 1998 case AudioSystem::OUTPUT_OPENED: 1999 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2000 desc.channels = mChannelMask; 2001 desc.samplingRate = mSampleRate; 2002 desc.format = mFormat; 2003 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 2004 desc.latency = latency(); 2005 param2 = &desc; 2006 break; 2007 2008 case AudioSystem::STREAM_CONFIG_CHANGED: 2009 param2 = ¶m; 2010 case AudioSystem::OUTPUT_CLOSED: 2011 default: 2012 break; 2013 } 2014 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2015} 2016 2017void AudioFlinger::PlaybackThread::readOutputParameters() 2018{ 2019 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2020 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2021 mChannelCount = (uint16_t)popcount(mChannelMask); 2022 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2023 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2024 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2025 if (mFrameCount & 15) { 2026 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2027 mFrameCount); 2028 } 2029 2030 // Calculate size of normal mix buffer relative to the HAL output buffer size 2031 double multiplier = 1.0; 2032 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2033 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2034 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2035 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2036 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2037 maxNormalFrameCount = maxNormalFrameCount & ~15; 2038 if (maxNormalFrameCount < minNormalFrameCount) { 2039 maxNormalFrameCount = minNormalFrameCount; 2040 } 2041 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2042 if (multiplier <= 1.0) { 2043 multiplier = 1.0; 2044 } else if (multiplier <= 2.0) { 2045 if (2 * mFrameCount <= maxNormalFrameCount) { 2046 multiplier = 2.0; 2047 } else { 2048 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2049 } 2050 } else { 2051 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2052 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2053 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2054 // FIXME this rounding up should not be done if no HAL SRC 2055 uint32_t truncMult = (uint32_t) multiplier; 2056 if ((truncMult & 1)) { 2057 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2058 ++truncMult; 2059 } 2060 } 2061 multiplier = (double) truncMult; 2062 } 2063 } 2064 mNormalFrameCount = multiplier * mFrameCount; 2065 // round up to nearest 16 frames to satisfy AudioMixer 2066 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2067 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2068 2069 delete[] mMixBuffer; 2070 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2071 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2072 2073 // force reconfiguration of effect chains and engines to take new buffer size and audio 2074 // parameters into account 2075 // Note that mLock is not held when readOutputParameters() is called from the constructor 2076 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2077 // matter. 2078 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2079 Vector< sp<EffectChain> > effectChains = mEffectChains; 2080 for (size_t i = 0; i < effectChains.size(); i ++) { 2081 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2082 } 2083} 2084 2085 2086status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2087{ 2088 if (halFrames == NULL || dspFrames == NULL) { 2089 return BAD_VALUE; 2090 } 2091 Mutex::Autolock _l(mLock); 2092 if (initCheck() != NO_ERROR) { 2093 return INVALID_OPERATION; 2094 } 2095 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2096 2097 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2098} 2099 2100uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2101{ 2102 Mutex::Autolock _l(mLock); 2103 uint32_t result = 0; 2104 if (getEffectChain_l(sessionId) != 0) { 2105 result = EFFECT_SESSION; 2106 } 2107 2108 for (size_t i = 0; i < mTracks.size(); ++i) { 2109 sp<Track> track = mTracks[i]; 2110 if (sessionId == track->sessionId() && 2111 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2112 result |= TRACK_SESSION; 2113 break; 2114 } 2115 } 2116 2117 return result; 2118} 2119 2120uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2121{ 2122 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2123 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2124 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2125 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2126 } 2127 for (size_t i = 0; i < mTracks.size(); i++) { 2128 sp<Track> track = mTracks[i]; 2129 if (sessionId == track->sessionId() && 2130 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2131 return AudioSystem::getStrategyForStream(track->streamType()); 2132 } 2133 } 2134 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2135} 2136 2137 2138AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2139{ 2140 Mutex::Autolock _l(mLock); 2141 return mOutput; 2142} 2143 2144AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2145{ 2146 Mutex::Autolock _l(mLock); 2147 AudioStreamOut *output = mOutput; 2148 mOutput = NULL; 2149 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2150 // must push a NULL and wait for ack 2151 mOutputSink.clear(); 2152 mPipeSink.clear(); 2153 mNormalSink.clear(); 2154 return output; 2155} 2156 2157// this method must always be called either with ThreadBase mLock held or inside the thread loop 2158audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2159{ 2160 if (mOutput == NULL) { 2161 return NULL; 2162 } 2163 return &mOutput->stream->common; 2164} 2165 2166uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2167{ 2168 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2169} 2170 2171status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2172{ 2173 if (!isValidSyncEvent(event)) { 2174 return BAD_VALUE; 2175 } 2176 2177 Mutex::Autolock _l(mLock); 2178 2179 for (size_t i = 0; i < mTracks.size(); ++i) { 2180 sp<Track> track = mTracks[i]; 2181 if (event->triggerSession() == track->sessionId()) { 2182 (void) track->setSyncEvent(event); 2183 return NO_ERROR; 2184 } 2185 } 2186 2187 return NAME_NOT_FOUND; 2188} 2189 2190bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2191{ 2192 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2193} 2194 2195void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2196{ 2197 size_t count = tracksToRemove.size(); 2198 if (CC_UNLIKELY(count)) { 2199 for (size_t i = 0 ; i < count ; i++) { 2200 const sp<Track>& track = tracksToRemove.itemAt(i); 2201 if ((track->sharedBuffer() != 0) && 2202 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2203 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2204 } 2205 } 2206 } 2207 2208} 2209 2210// ---------------------------------------------------------------------------- 2211 2212AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2213 audio_io_handle_t id, audio_devices_t device, type_t type) 2214 : PlaybackThread(audioFlinger, output, id, device, type), 2215 // mAudioMixer below 2216 // mFastMixer below 2217 mFastMixerFutex(0) 2218 // mOutputSink below 2219 // mPipeSink below 2220 // mNormalSink below 2221{ 2222 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2223 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2224 "mFrameCount=%d, mNormalFrameCount=%d", 2225 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2226 mNormalFrameCount); 2227 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2228 2229 // FIXME - Current mixer implementation only supports stereo output 2230 if (mChannelCount != FCC_2) { 2231 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2232 } 2233 2234 // create an NBAIO sink for the HAL output stream, and negotiate 2235 mOutputSink = new AudioStreamOutSink(output->stream); 2236 size_t numCounterOffers = 0; 2237 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2238 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2239 ALOG_ASSERT(index == 0); 2240 2241 // initialize fast mixer depending on configuration 2242 bool initFastMixer; 2243 switch (kUseFastMixer) { 2244 case FastMixer_Never: 2245 initFastMixer = false; 2246 break; 2247 case FastMixer_Always: 2248 initFastMixer = true; 2249 break; 2250 case FastMixer_Static: 2251 case FastMixer_Dynamic: 2252 initFastMixer = mFrameCount < mNormalFrameCount; 2253 break; 2254 } 2255 if (initFastMixer) { 2256 2257 // create a MonoPipe to connect our submix to FastMixer 2258 NBAIO_Format format = mOutputSink->format(); 2259 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2260 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2261 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2262 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2263 const NBAIO_Format offers[1] = {format}; 2264 size_t numCounterOffers = 0; 2265 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2266 ALOG_ASSERT(index == 0); 2267 monoPipe->setAvgFrames((mScreenState & 1) ? 2268 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2269 mPipeSink = monoPipe; 2270 2271#ifdef TEE_SINK_FRAMES 2272 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2273 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2274 numCounterOffers = 0; 2275 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2276 ALOG_ASSERT(index == 0); 2277 mTeeSink = teeSink; 2278 PipeReader *teeSource = new PipeReader(*teeSink); 2279 numCounterOffers = 0; 2280 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2281 ALOG_ASSERT(index == 0); 2282 mTeeSource = teeSource; 2283#endif 2284 2285 // create fast mixer and configure it initially with just one fast track for our submix 2286 mFastMixer = new FastMixer(); 2287 FastMixerStateQueue *sq = mFastMixer->sq(); 2288#ifdef STATE_QUEUE_DUMP 2289 sq->setObserverDump(&mStateQueueObserverDump); 2290 sq->setMutatorDump(&mStateQueueMutatorDump); 2291#endif 2292 FastMixerState *state = sq->begin(); 2293 FastTrack *fastTrack = &state->mFastTracks[0]; 2294 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2295 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2296 fastTrack->mVolumeProvider = NULL; 2297 fastTrack->mGeneration++; 2298 state->mFastTracksGen++; 2299 state->mTrackMask = 1; 2300 // fast mixer will use the HAL output sink 2301 state->mOutputSink = mOutputSink.get(); 2302 state->mOutputSinkGen++; 2303 state->mFrameCount = mFrameCount; 2304 state->mCommand = FastMixerState::COLD_IDLE; 2305 // already done in constructor initialization list 2306 //mFastMixerFutex = 0; 2307 state->mColdFutexAddr = &mFastMixerFutex; 2308 state->mColdGen++; 2309 state->mDumpState = &mFastMixerDumpState; 2310 state->mTeeSink = mTeeSink.get(); 2311 sq->end(); 2312 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2313 2314 // start the fast mixer 2315 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2316 pid_t tid = mFastMixer->getTid(); 2317 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2318 if (err != 0) { 2319 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2320 kPriorityFastMixer, getpid_cached, tid, err); 2321 } 2322 2323#ifdef AUDIO_WATCHDOG 2324 // create and start the watchdog 2325 mAudioWatchdog = new AudioWatchdog(); 2326 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2327 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2328 tid = mAudioWatchdog->getTid(); 2329 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2330 if (err != 0) { 2331 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2332 kPriorityFastMixer, getpid_cached, tid, err); 2333 } 2334#endif 2335 2336 } else { 2337 mFastMixer = NULL; 2338 } 2339 2340 switch (kUseFastMixer) { 2341 case FastMixer_Never: 2342 case FastMixer_Dynamic: 2343 mNormalSink = mOutputSink; 2344 break; 2345 case FastMixer_Always: 2346 mNormalSink = mPipeSink; 2347 break; 2348 case FastMixer_Static: 2349 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2350 break; 2351 } 2352} 2353 2354AudioFlinger::MixerThread::~MixerThread() 2355{ 2356 if (mFastMixer != NULL) { 2357 FastMixerStateQueue *sq = mFastMixer->sq(); 2358 FastMixerState *state = sq->begin(); 2359 if (state->mCommand == FastMixerState::COLD_IDLE) { 2360 int32_t old = android_atomic_inc(&mFastMixerFutex); 2361 if (old == -1) { 2362 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2363 } 2364 } 2365 state->mCommand = FastMixerState::EXIT; 2366 sq->end(); 2367 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2368 mFastMixer->join(); 2369 // Though the fast mixer thread has exited, it's state queue is still valid. 2370 // We'll use that extract the final state which contains one remaining fast track 2371 // corresponding to our sub-mix. 2372 state = sq->begin(); 2373 ALOG_ASSERT(state->mTrackMask == 1); 2374 FastTrack *fastTrack = &state->mFastTracks[0]; 2375 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2376 delete fastTrack->mBufferProvider; 2377 sq->end(false /*didModify*/); 2378 delete mFastMixer; 2379 if (mAudioWatchdog != 0) { 2380 mAudioWatchdog->requestExit(); 2381 mAudioWatchdog->requestExitAndWait(); 2382 mAudioWatchdog.clear(); 2383 } 2384 } 2385 delete mAudioMixer; 2386} 2387 2388class CpuStats { 2389public: 2390 CpuStats(); 2391 void sample(const String8 &title); 2392#ifdef DEBUG_CPU_USAGE 2393private: 2394 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2395 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2396 2397 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2398 2399 int mCpuNum; // thread's current CPU number 2400 int mCpukHz; // frequency of thread's current CPU in kHz 2401#endif 2402}; 2403 2404CpuStats::CpuStats() 2405#ifdef DEBUG_CPU_USAGE 2406 : mCpuNum(-1), mCpukHz(-1) 2407#endif 2408{ 2409} 2410 2411void CpuStats::sample(const String8 &title) { 2412#ifdef DEBUG_CPU_USAGE 2413 // get current thread's delta CPU time in wall clock ns 2414 double wcNs; 2415 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2416 2417 // record sample for wall clock statistics 2418 if (valid) { 2419 mWcStats.sample(wcNs); 2420 } 2421 2422 // get the current CPU number 2423 int cpuNum = sched_getcpu(); 2424 2425 // get the current CPU frequency in kHz 2426 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2427 2428 // check if either CPU number or frequency changed 2429 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2430 mCpuNum = cpuNum; 2431 mCpukHz = cpukHz; 2432 // ignore sample for purposes of cycles 2433 valid = false; 2434 } 2435 2436 // if no change in CPU number or frequency, then record sample for cycle statistics 2437 if (valid && mCpukHz > 0) { 2438 double cycles = wcNs * cpukHz * 0.000001; 2439 mHzStats.sample(cycles); 2440 } 2441 2442 unsigned n = mWcStats.n(); 2443 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2444 if ((n & 127) == 1) { 2445 long long elapsed = mCpuUsage.elapsed(); 2446 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2447 double perLoop = elapsed / (double) n; 2448 double perLoop100 = perLoop * 0.01; 2449 double perLoop1k = perLoop * 0.001; 2450 double mean = mWcStats.mean(); 2451 double stddev = mWcStats.stddev(); 2452 double minimum = mWcStats.minimum(); 2453 double maximum = mWcStats.maximum(); 2454 double meanCycles = mHzStats.mean(); 2455 double stddevCycles = mHzStats.stddev(); 2456 double minCycles = mHzStats.minimum(); 2457 double maxCycles = mHzStats.maximum(); 2458 mCpuUsage.resetElapsed(); 2459 mWcStats.reset(); 2460 mHzStats.reset(); 2461 ALOGD("CPU usage for %s over past %.1f secs\n" 2462 " (%u mixer loops at %.1f mean ms per loop):\n" 2463 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2464 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2465 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2466 title.string(), 2467 elapsed * .000000001, n, perLoop * .000001, 2468 mean * .001, 2469 stddev * .001, 2470 minimum * .001, 2471 maximum * .001, 2472 mean / perLoop100, 2473 stddev / perLoop100, 2474 minimum / perLoop100, 2475 maximum / perLoop100, 2476 meanCycles / perLoop1k, 2477 stddevCycles / perLoop1k, 2478 minCycles / perLoop1k, 2479 maxCycles / perLoop1k); 2480 2481 } 2482 } 2483#endif 2484}; 2485 2486void AudioFlinger::PlaybackThread::checkSilentMode_l() 2487{ 2488 if (!mMasterMute) { 2489 char value[PROPERTY_VALUE_MAX]; 2490 if (property_get("ro.audio.silent", value, "0") > 0) { 2491 char *endptr; 2492 unsigned long ul = strtoul(value, &endptr, 0); 2493 if (*endptr == '\0' && ul != 0) { 2494 ALOGD("Silence is golden"); 2495 // The setprop command will not allow a property to be changed after 2496 // the first time it is set, so we don't have to worry about un-muting. 2497 setMasterMute_l(true); 2498 } 2499 } 2500 } 2501} 2502 2503bool AudioFlinger::PlaybackThread::threadLoop() 2504{ 2505 Vector< sp<Track> > tracksToRemove; 2506 2507 standbyTime = systemTime(); 2508 2509 // MIXER 2510 nsecs_t lastWarning = 0; 2511 2512 // DUPLICATING 2513 // FIXME could this be made local to while loop? 2514 writeFrames = 0; 2515 2516 cacheParameters_l(); 2517 sleepTime = idleSleepTime; 2518 2519 if (mType == MIXER) { 2520 sleepTimeShift = 0; 2521 } 2522 2523 CpuStats cpuStats; 2524 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2525 2526 acquireWakeLock(); 2527 2528 while (!exitPending()) 2529 { 2530 cpuStats.sample(myName); 2531 2532 Vector< sp<EffectChain> > effectChains; 2533 2534 processConfigEvents(); 2535 2536 { // scope for mLock 2537 2538 Mutex::Autolock _l(mLock); 2539 2540 if (checkForNewParameters_l()) { 2541 cacheParameters_l(); 2542 } 2543 2544 saveOutputTracks(); 2545 2546 // put audio hardware into standby after short delay 2547 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2548 isSuspended())) { 2549 if (!mStandby) { 2550 2551 threadLoop_standby(); 2552 2553 mStandby = true; 2554 mBytesWritten = 0; 2555 } 2556 2557 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2558 // we're about to wait, flush the binder command buffer 2559 IPCThreadState::self()->flushCommands(); 2560 2561 clearOutputTracks(); 2562 2563 if (exitPending()) break; 2564 2565 releaseWakeLock_l(); 2566 // wait until we have something to do... 2567 ALOGV("%s going to sleep", myName.string()); 2568 mWaitWorkCV.wait(mLock); 2569 ALOGV("%s waking up", myName.string()); 2570 acquireWakeLock_l(); 2571 2572 mMixerStatus = MIXER_IDLE; 2573 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2574 2575 checkSilentMode_l(); 2576 2577 standbyTime = systemTime() + standbyDelay; 2578 sleepTime = idleSleepTime; 2579 if (mType == MIXER) { 2580 sleepTimeShift = 0; 2581 } 2582 2583 continue; 2584 } 2585 } 2586 2587 // mMixerStatusIgnoringFastTracks is also updated internally 2588 mMixerStatus = prepareTracks_l(&tracksToRemove); 2589 2590 // prevent any changes in effect chain list and in each effect chain 2591 // during mixing and effect process as the audio buffers could be deleted 2592 // or modified if an effect is created or deleted 2593 lockEffectChains_l(effectChains); 2594 } 2595 2596 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2597 threadLoop_mix(); 2598 } else { 2599 threadLoop_sleepTime(); 2600 } 2601 2602 if (isSuspended()) { 2603 sleepTime = suspendSleepTimeUs(); 2604 } 2605 2606 // only process effects if we're going to write 2607 if (sleepTime == 0) { 2608 for (size_t i = 0; i < effectChains.size(); i ++) { 2609 effectChains[i]->process_l(); 2610 } 2611 } 2612 2613 // enable changes in effect chain 2614 unlockEffectChains(effectChains); 2615 2616 // sleepTime == 0 means we must write to audio hardware 2617 if (sleepTime == 0) { 2618 2619 threadLoop_write(); 2620 2621if (mType == MIXER) { 2622 // write blocked detection 2623 nsecs_t now = systemTime(); 2624 nsecs_t delta = now - mLastWriteTime; 2625 if (!mStandby && delta > maxPeriod) { 2626 mNumDelayedWrites++; 2627 if ((now - lastWarning) > kWarningThrottleNs) { 2628#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2629 ScopedTrace st(ATRACE_TAG, "underrun"); 2630#endif 2631 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2632 ns2ms(delta), mNumDelayedWrites, this); 2633 lastWarning = now; 2634 } 2635 } 2636} 2637 2638 mStandby = false; 2639 } else { 2640 usleep(sleepTime); 2641 } 2642 2643 // Finally let go of removed track(s), without the lock held 2644 // since we can't guarantee the destructors won't acquire that 2645 // same lock. This will also mutate and push a new fast mixer state. 2646 threadLoop_removeTracks(tracksToRemove); 2647 tracksToRemove.clear(); 2648 2649 // FIXME I don't understand the need for this here; 2650 // it was in the original code but maybe the 2651 // assignment in saveOutputTracks() makes this unnecessary? 2652 clearOutputTracks(); 2653 2654 // Effect chains will be actually deleted here if they were removed from 2655 // mEffectChains list during mixing or effects processing 2656 effectChains.clear(); 2657 2658 // FIXME Note that the above .clear() is no longer necessary since effectChains 2659 // is now local to this block, but will keep it for now (at least until merge done). 2660 } 2661 2662 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2663 if (mType == MIXER || mType == DIRECT) { 2664 // put output stream into standby mode 2665 if (!mStandby) { 2666 mOutput->stream->common.standby(&mOutput->stream->common); 2667 } 2668 } 2669 2670 releaseWakeLock(); 2671 2672 ALOGV("Thread %p type %d exiting", this, mType); 2673 return false; 2674} 2675 2676void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2677{ 2678 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2679} 2680 2681void AudioFlinger::MixerThread::threadLoop_write() 2682{ 2683 // FIXME we should only do one push per cycle; confirm this is true 2684 // Start the fast mixer if it's not already running 2685 if (mFastMixer != NULL) { 2686 FastMixerStateQueue *sq = mFastMixer->sq(); 2687 FastMixerState *state = sq->begin(); 2688 if (state->mCommand != FastMixerState::MIX_WRITE && 2689 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2690 if (state->mCommand == FastMixerState::COLD_IDLE) { 2691 int32_t old = android_atomic_inc(&mFastMixerFutex); 2692 if (old == -1) { 2693 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2694 } 2695 if (mAudioWatchdog != 0) { 2696 mAudioWatchdog->resume(); 2697 } 2698 } 2699 state->mCommand = FastMixerState::MIX_WRITE; 2700 sq->end(); 2701 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2702 if (kUseFastMixer == FastMixer_Dynamic) { 2703 mNormalSink = mPipeSink; 2704 } 2705 } else { 2706 sq->end(false /*didModify*/); 2707 } 2708 } 2709 PlaybackThread::threadLoop_write(); 2710} 2711 2712// shared by MIXER and DIRECT, overridden by DUPLICATING 2713void AudioFlinger::PlaybackThread::threadLoop_write() 2714{ 2715 // FIXME rewrite to reduce number of system calls 2716 mLastWriteTime = systemTime(); 2717 mInWrite = true; 2718 int bytesWritten; 2719 2720 // If an NBAIO sink is present, use it to write the normal mixer's submix 2721 if (mNormalSink != 0) { 2722#define mBitShift 2 // FIXME 2723 size_t count = mixBufferSize >> mBitShift; 2724#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2725 Tracer::traceBegin(ATRACE_TAG, "write"); 2726#endif 2727 // update the setpoint when gScreenState changes 2728 uint32_t screenState = gScreenState; 2729 if (screenState != mScreenState) { 2730 mScreenState = screenState; 2731 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2732 if (pipe != NULL) { 2733 pipe->setAvgFrames((mScreenState & 1) ? 2734 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2735 } 2736 } 2737 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2738#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2739 Tracer::traceEnd(ATRACE_TAG); 2740#endif 2741 if (framesWritten > 0) { 2742 bytesWritten = framesWritten << mBitShift; 2743 } else { 2744 bytesWritten = framesWritten; 2745 } 2746 // otherwise use the HAL / AudioStreamOut directly 2747 } else { 2748 // Direct output thread. 2749 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2750 } 2751 2752 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2753 mNumWrites++; 2754 mInWrite = false; 2755} 2756 2757void AudioFlinger::MixerThread::threadLoop_standby() 2758{ 2759 // Idle the fast mixer if it's currently running 2760 if (mFastMixer != NULL) { 2761 FastMixerStateQueue *sq = mFastMixer->sq(); 2762 FastMixerState *state = sq->begin(); 2763 if (!(state->mCommand & FastMixerState::IDLE)) { 2764 state->mCommand = FastMixerState::COLD_IDLE; 2765 state->mColdFutexAddr = &mFastMixerFutex; 2766 state->mColdGen++; 2767 mFastMixerFutex = 0; 2768 sq->end(); 2769 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2770 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2771 if (kUseFastMixer == FastMixer_Dynamic) { 2772 mNormalSink = mOutputSink; 2773 } 2774 if (mAudioWatchdog != 0) { 2775 mAudioWatchdog->pause(); 2776 } 2777 } else { 2778 sq->end(false /*didModify*/); 2779 } 2780 } 2781 PlaybackThread::threadLoop_standby(); 2782} 2783 2784// shared by MIXER and DIRECT, overridden by DUPLICATING 2785void AudioFlinger::PlaybackThread::threadLoop_standby() 2786{ 2787 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2788 mOutput->stream->common.standby(&mOutput->stream->common); 2789} 2790 2791void AudioFlinger::MixerThread::threadLoop_mix() 2792{ 2793 // obtain the presentation timestamp of the next output buffer 2794 int64_t pts; 2795 status_t status = INVALID_OPERATION; 2796 2797 if (mNormalSink != 0) { 2798 status = mNormalSink->getNextWriteTimestamp(&pts); 2799 } else { 2800 status = mOutputSink->getNextWriteTimestamp(&pts); 2801 } 2802 2803 if (status != NO_ERROR) { 2804 pts = AudioBufferProvider::kInvalidPTS; 2805 } 2806 2807 // mix buffers... 2808 mAudioMixer->process(pts); 2809 // increase sleep time progressively when application underrun condition clears. 2810 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2811 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2812 // such that we would underrun the audio HAL. 2813 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2814 sleepTimeShift--; 2815 } 2816 sleepTime = 0; 2817 standbyTime = systemTime() + standbyDelay; 2818 //TODO: delay standby when effects have a tail 2819} 2820 2821void AudioFlinger::MixerThread::threadLoop_sleepTime() 2822{ 2823 // If no tracks are ready, sleep once for the duration of an output 2824 // buffer size, then write 0s to the output 2825 if (sleepTime == 0) { 2826 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2827 sleepTime = activeSleepTime >> sleepTimeShift; 2828 if (sleepTime < kMinThreadSleepTimeUs) { 2829 sleepTime = kMinThreadSleepTimeUs; 2830 } 2831 // reduce sleep time in case of consecutive application underruns to avoid 2832 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2833 // duration we would end up writing less data than needed by the audio HAL if 2834 // the condition persists. 2835 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2836 sleepTimeShift++; 2837 } 2838 } else { 2839 sleepTime = idleSleepTime; 2840 } 2841 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2842 memset (mMixBuffer, 0, mixBufferSize); 2843 sleepTime = 0; 2844 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2845 } 2846 // TODO add standby time extension fct of effect tail 2847} 2848 2849// prepareTracks_l() must be called with ThreadBase::mLock held 2850AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2851 Vector< sp<Track> > *tracksToRemove) 2852{ 2853 2854 mixer_state mixerStatus = MIXER_IDLE; 2855 // find out which tracks need to be processed 2856 size_t count = mActiveTracks.size(); 2857 size_t mixedTracks = 0; 2858 size_t tracksWithEffect = 0; 2859 // counts only _active_ fast tracks 2860 size_t fastTracks = 0; 2861 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2862 2863 float masterVolume = mMasterVolume; 2864 bool masterMute = mMasterMute; 2865 2866 if (masterMute) { 2867 masterVolume = 0; 2868 } 2869 // Delegate master volume control to effect in output mix effect chain if needed 2870 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2871 if (chain != 0) { 2872 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2873 chain->setVolume_l(&v, &v); 2874 masterVolume = (float)((v + (1 << 23)) >> 24); 2875 chain.clear(); 2876 } 2877 2878 // prepare a new state to push 2879 FastMixerStateQueue *sq = NULL; 2880 FastMixerState *state = NULL; 2881 bool didModify = false; 2882 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2883 if (mFastMixer != NULL) { 2884 sq = mFastMixer->sq(); 2885 state = sq->begin(); 2886 } 2887 2888 for (size_t i=0 ; i<count ; i++) { 2889 sp<Track> t = mActiveTracks[i].promote(); 2890 if (t == 0) continue; 2891 2892 // this const just means the local variable doesn't change 2893 Track* const track = t.get(); 2894 2895 // process fast tracks 2896 if (track->isFastTrack()) { 2897 2898 // It's theoretically possible (though unlikely) for a fast track to be created 2899 // and then removed within the same normal mix cycle. This is not a problem, as 2900 // the track never becomes active so it's fast mixer slot is never touched. 2901 // The converse, of removing an (active) track and then creating a new track 2902 // at the identical fast mixer slot within the same normal mix cycle, 2903 // is impossible because the slot isn't marked available until the end of each cycle. 2904 int j = track->mFastIndex; 2905 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2906 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2907 FastTrack *fastTrack = &state->mFastTracks[j]; 2908 2909 // Determine whether the track is currently in underrun condition, 2910 // and whether it had a recent underrun. 2911 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2912 FastTrackUnderruns underruns = ftDump->mUnderruns; 2913 uint32_t recentFull = (underruns.mBitFields.mFull - 2914 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2915 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2916 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2917 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2918 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2919 uint32_t recentUnderruns = recentPartial + recentEmpty; 2920 track->mObservedUnderruns = underruns; 2921 // don't count underruns that occur while stopping or pausing 2922 // or stopped which can occur when flush() is called while active 2923 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2924 track->mUnderrunCount += recentUnderruns; 2925 } 2926 2927 // This is similar to the state machine for normal tracks, 2928 // with a few modifications for fast tracks. 2929 bool isActive = true; 2930 switch (track->mState) { 2931 case TrackBase::STOPPING_1: 2932 // track stays active in STOPPING_1 state until first underrun 2933 if (recentUnderruns > 0) { 2934 track->mState = TrackBase::STOPPING_2; 2935 } 2936 break; 2937 case TrackBase::PAUSING: 2938 // ramp down is not yet implemented 2939 track->setPaused(); 2940 break; 2941 case TrackBase::RESUMING: 2942 // ramp up is not yet implemented 2943 track->mState = TrackBase::ACTIVE; 2944 break; 2945 case TrackBase::ACTIVE: 2946 if (recentFull > 0 || recentPartial > 0) { 2947 // track has provided at least some frames recently: reset retry count 2948 track->mRetryCount = kMaxTrackRetries; 2949 } 2950 if (recentUnderruns == 0) { 2951 // no recent underruns: stay active 2952 break; 2953 } 2954 // there has recently been an underrun of some kind 2955 if (track->sharedBuffer() == 0) { 2956 // were any of the recent underruns "empty" (no frames available)? 2957 if (recentEmpty == 0) { 2958 // no, then ignore the partial underruns as they are allowed indefinitely 2959 break; 2960 } 2961 // there has recently been an "empty" underrun: decrement the retry counter 2962 if (--(track->mRetryCount) > 0) { 2963 break; 2964 } 2965 // indicate to client process that the track was disabled because of underrun; 2966 // it will then automatically call start() when data is available 2967 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2968 // remove from active list, but state remains ACTIVE [confusing but true] 2969 isActive = false; 2970 break; 2971 } 2972 // fall through 2973 case TrackBase::STOPPING_2: 2974 case TrackBase::PAUSED: 2975 case TrackBase::TERMINATED: 2976 case TrackBase::STOPPED: 2977 case TrackBase::FLUSHED: // flush() while active 2978 // Check for presentation complete if track is inactive 2979 // We have consumed all the buffers of this track. 2980 // This would be incomplete if we auto-paused on underrun 2981 { 2982 size_t audioHALFrames = 2983 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2984 size_t framesWritten = 2985 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2986 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2987 // track stays in active list until presentation is complete 2988 break; 2989 } 2990 } 2991 if (track->isStopping_2()) { 2992 track->mState = TrackBase::STOPPED; 2993 } 2994 if (track->isStopped()) { 2995 // Can't reset directly, as fast mixer is still polling this track 2996 // track->reset(); 2997 // So instead mark this track as needing to be reset after push with ack 2998 resetMask |= 1 << i; 2999 } 3000 isActive = false; 3001 break; 3002 case TrackBase::IDLE: 3003 default: 3004 LOG_FATAL("unexpected track state %d", track->mState); 3005 } 3006 3007 if (isActive) { 3008 // was it previously inactive? 3009 if (!(state->mTrackMask & (1 << j))) { 3010 ExtendedAudioBufferProvider *eabp = track; 3011 VolumeProvider *vp = track; 3012 fastTrack->mBufferProvider = eabp; 3013 fastTrack->mVolumeProvider = vp; 3014 fastTrack->mSampleRate = track->mSampleRate; 3015 fastTrack->mChannelMask = track->mChannelMask; 3016 fastTrack->mGeneration++; 3017 state->mTrackMask |= 1 << j; 3018 didModify = true; 3019 // no acknowledgement required for newly active tracks 3020 } 3021 // cache the combined master volume and stream type volume for fast mixer; this 3022 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3023 track->mCachedVolume = track->isMuted() ? 3024 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3025 ++fastTracks; 3026 } else { 3027 // was it previously active? 3028 if (state->mTrackMask & (1 << j)) { 3029 fastTrack->mBufferProvider = NULL; 3030 fastTrack->mGeneration++; 3031 state->mTrackMask &= ~(1 << j); 3032 didModify = true; 3033 // If any fast tracks were removed, we must wait for acknowledgement 3034 // because we're about to decrement the last sp<> on those tracks. 3035 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3036 } else { 3037 LOG_FATAL("fast track %d should have been active", j); 3038 } 3039 tracksToRemove->add(track); 3040 // Avoids a misleading display in dumpsys 3041 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3042 } 3043 continue; 3044 } 3045 3046 { // local variable scope to avoid goto warning 3047 3048 audio_track_cblk_t* cblk = track->cblk(); 3049 3050 // The first time a track is added we wait 3051 // for all its buffers to be filled before processing it 3052 int name = track->name(); 3053 // make sure that we have enough frames to mix one full buffer. 3054 // enforce this condition only once to enable draining the buffer in case the client 3055 // app does not call stop() and relies on underrun to stop: 3056 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3057 // during last round 3058 uint32_t minFrames = 1; 3059 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3060 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3061 if (t->sampleRate() == (int)mSampleRate) { 3062 minFrames = mNormalFrameCount; 3063 } else { 3064 // +1 for rounding and +1 for additional sample needed for interpolation 3065 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3066 // add frames already consumed but not yet released by the resampler 3067 // because cblk->framesReady() will include these frames 3068 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3069 // the minimum track buffer size is normally twice the number of frames necessary 3070 // to fill one buffer and the resampler should not leave more than one buffer worth 3071 // of unreleased frames after each pass, but just in case... 3072 ALOG_ASSERT(minFrames <= cblk->frameCount); 3073 } 3074 } 3075 if ((track->framesReady() >= minFrames) && track->isReady() && 3076 !track->isPaused() && !track->isTerminated()) 3077 { 3078 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3079 3080 mixedTracks++; 3081 3082 // track->mainBuffer() != mMixBuffer means there is an effect chain 3083 // connected to the track 3084 chain.clear(); 3085 if (track->mainBuffer() != mMixBuffer) { 3086 chain = getEffectChain_l(track->sessionId()); 3087 // Delegate volume control to effect in track effect chain if needed 3088 if (chain != 0) { 3089 tracksWithEffect++; 3090 } else { 3091 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3092 name, track->sessionId()); 3093 } 3094 } 3095 3096 3097 int param = AudioMixer::VOLUME; 3098 if (track->mFillingUpStatus == Track::FS_FILLED) { 3099 // no ramp for the first volume setting 3100 track->mFillingUpStatus = Track::FS_ACTIVE; 3101 if (track->mState == TrackBase::RESUMING) { 3102 track->mState = TrackBase::ACTIVE; 3103 param = AudioMixer::RAMP_VOLUME; 3104 } 3105 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3106 } else if (cblk->server != 0) { 3107 // If the track is stopped before the first frame was mixed, 3108 // do not apply ramp 3109 param = AudioMixer::RAMP_VOLUME; 3110 } 3111 3112 // compute volume for this track 3113 uint32_t vl, vr, va; 3114 if (track->isMuted() || track->isPausing() || 3115 mStreamTypes[track->streamType()].mute) { 3116 vl = vr = va = 0; 3117 if (track->isPausing()) { 3118 track->setPaused(); 3119 } 3120 } else { 3121 3122 // read original volumes with volume control 3123 float typeVolume = mStreamTypes[track->streamType()].volume; 3124 float v = masterVolume * typeVolume; 3125 uint32_t vlr = cblk->getVolumeLR(); 3126 vl = vlr & 0xFFFF; 3127 vr = vlr >> 16; 3128 // track volumes come from shared memory, so can't be trusted and must be clamped 3129 if (vl > MAX_GAIN_INT) { 3130 ALOGV("Track left volume out of range: %04X", vl); 3131 vl = MAX_GAIN_INT; 3132 } 3133 if (vr > MAX_GAIN_INT) { 3134 ALOGV("Track right volume out of range: %04X", vr); 3135 vr = MAX_GAIN_INT; 3136 } 3137 // now apply the master volume and stream type volume 3138 vl = (uint32_t)(v * vl) << 12; 3139 vr = (uint32_t)(v * vr) << 12; 3140 // assuming master volume and stream type volume each go up to 1.0, 3141 // vl and vr are now in 8.24 format 3142 3143 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3144 // send level comes from shared memory and so may be corrupt 3145 if (sendLevel > MAX_GAIN_INT) { 3146 ALOGV("Track send level out of range: %04X", sendLevel); 3147 sendLevel = MAX_GAIN_INT; 3148 } 3149 va = (uint32_t)(v * sendLevel); 3150 } 3151 // Delegate volume control to effect in track effect chain if needed 3152 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3153 // Do not ramp volume if volume is controlled by effect 3154 param = AudioMixer::VOLUME; 3155 track->mHasVolumeController = true; 3156 } else { 3157 // force no volume ramp when volume controller was just disabled or removed 3158 // from effect chain to avoid volume spike 3159 if (track->mHasVolumeController) { 3160 param = AudioMixer::VOLUME; 3161 } 3162 track->mHasVolumeController = false; 3163 } 3164 3165 // Convert volumes from 8.24 to 4.12 format 3166 // This additional clamping is needed in case chain->setVolume_l() overshot 3167 vl = (vl + (1 << 11)) >> 12; 3168 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3169 vr = (vr + (1 << 11)) >> 12; 3170 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3171 3172 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3173 3174 // XXX: these things DON'T need to be done each time 3175 mAudioMixer->setBufferProvider(name, track); 3176 mAudioMixer->enable(name); 3177 3178 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3179 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3180 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3181 mAudioMixer->setParameter( 3182 name, 3183 AudioMixer::TRACK, 3184 AudioMixer::FORMAT, (void *)track->format()); 3185 mAudioMixer->setParameter( 3186 name, 3187 AudioMixer::TRACK, 3188 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3189 mAudioMixer->setParameter( 3190 name, 3191 AudioMixer::RESAMPLE, 3192 AudioMixer::SAMPLE_RATE, 3193 (void *)(cblk->sampleRate)); 3194 mAudioMixer->setParameter( 3195 name, 3196 AudioMixer::TRACK, 3197 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3198 mAudioMixer->setParameter( 3199 name, 3200 AudioMixer::TRACK, 3201 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3202 3203 // reset retry count 3204 track->mRetryCount = kMaxTrackRetries; 3205 3206 // If one track is ready, set the mixer ready if: 3207 // - the mixer was not ready during previous round OR 3208 // - no other track is not ready 3209 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3210 mixerStatus != MIXER_TRACKS_ENABLED) { 3211 mixerStatus = MIXER_TRACKS_READY; 3212 } 3213 } else { 3214 // clear effect chain input buffer if an active track underruns to avoid sending 3215 // previous audio buffer again to effects 3216 chain = getEffectChain_l(track->sessionId()); 3217 if (chain != 0) { 3218 chain->clearInputBuffer(); 3219 } 3220 3221 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3222 if ((track->sharedBuffer() != 0) || 3223 track->isStopped() || track->isPaused()) { 3224 // We have consumed all the buffers of this track. 3225 // Remove it from the list of active tracks. 3226 // TODO: use actual buffer filling status instead of latency when available from 3227 // audio HAL 3228 size_t audioHALFrames = 3229 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3230 size_t framesWritten = 3231 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3232 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3233 if (track->isStopped()) { 3234 track->reset(); 3235 } 3236 tracksToRemove->add(track); 3237 } 3238 } else { 3239 track->mUnderrunCount++; 3240 // No buffers for this track. Give it a few chances to 3241 // fill a buffer, then remove it from active list. 3242 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3243 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3244 tracksToRemove->add(track); 3245 // indicate to client process that the track was disabled because of underrun; 3246 // it will then automatically call start() when data is available 3247 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3248 // If one track is not ready, mark the mixer also not ready if: 3249 // - the mixer was ready during previous round OR 3250 // - no other track is ready 3251 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3252 mixerStatus != MIXER_TRACKS_READY) { 3253 mixerStatus = MIXER_TRACKS_ENABLED; 3254 } 3255 } 3256 mAudioMixer->disable(name); 3257 } 3258 3259 } // local variable scope to avoid goto warning 3260track_is_ready: ; 3261 3262 } 3263 3264 // Push the new FastMixer state if necessary 3265 bool pauseAudioWatchdog = false; 3266 if (didModify) { 3267 state->mFastTracksGen++; 3268 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3269 if (kUseFastMixer == FastMixer_Dynamic && 3270 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3271 state->mCommand = FastMixerState::COLD_IDLE; 3272 state->mColdFutexAddr = &mFastMixerFutex; 3273 state->mColdGen++; 3274 mFastMixerFutex = 0; 3275 if (kUseFastMixer == FastMixer_Dynamic) { 3276 mNormalSink = mOutputSink; 3277 } 3278 // If we go into cold idle, need to wait for acknowledgement 3279 // so that fast mixer stops doing I/O. 3280 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3281 pauseAudioWatchdog = true; 3282 } 3283 sq->end(); 3284 } 3285 if (sq != NULL) { 3286 sq->end(didModify); 3287 sq->push(block); 3288 } 3289 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3290 mAudioWatchdog->pause(); 3291 } 3292 3293 // Now perform the deferred reset on fast tracks that have stopped 3294 while (resetMask != 0) { 3295 size_t i = __builtin_ctz(resetMask); 3296 ALOG_ASSERT(i < count); 3297 resetMask &= ~(1 << i); 3298 sp<Track> t = mActiveTracks[i].promote(); 3299 if (t == 0) continue; 3300 Track* track = t.get(); 3301 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3302 track->reset(); 3303 } 3304 3305 // remove all the tracks that need to be... 3306 count = tracksToRemove->size(); 3307 if (CC_UNLIKELY(count)) { 3308 for (size_t i=0 ; i<count ; i++) { 3309 const sp<Track>& track = tracksToRemove->itemAt(i); 3310 mActiveTracks.remove(track); 3311 if (track->mainBuffer() != mMixBuffer) { 3312 chain = getEffectChain_l(track->sessionId()); 3313 if (chain != 0) { 3314 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3315 chain->decActiveTrackCnt(); 3316 } 3317 } 3318 if (track->isTerminated()) { 3319 removeTrack_l(track); 3320 } 3321 } 3322 } 3323 3324 // mix buffer must be cleared if all tracks are connected to an 3325 // effect chain as in this case the mixer will not write to 3326 // mix buffer and track effects will accumulate into it 3327 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3328 // FIXME as a performance optimization, should remember previous zero status 3329 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3330 } 3331 3332 // if any fast tracks, then status is ready 3333 mMixerStatusIgnoringFastTracks = mixerStatus; 3334 if (fastTracks > 0) { 3335 mixerStatus = MIXER_TRACKS_READY; 3336 } 3337 return mixerStatus; 3338} 3339 3340/* 3341The derived values that are cached: 3342 - mixBufferSize from frame count * frame size 3343 - activeSleepTime from activeSleepTimeUs() 3344 - idleSleepTime from idleSleepTimeUs() 3345 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3346 - maxPeriod from frame count and sample rate (MIXER only) 3347 3348The parameters that affect these derived values are: 3349 - frame count 3350 - frame size 3351 - sample rate 3352 - device type: A2DP or not 3353 - device latency 3354 - format: PCM or not 3355 - active sleep time 3356 - idle sleep time 3357*/ 3358 3359void AudioFlinger::PlaybackThread::cacheParameters_l() 3360{ 3361 mixBufferSize = mNormalFrameCount * mFrameSize; 3362 activeSleepTime = activeSleepTimeUs(); 3363 idleSleepTime = idleSleepTimeUs(); 3364} 3365 3366void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3367{ 3368 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3369 this, streamType, mTracks.size()); 3370 Mutex::Autolock _l(mLock); 3371 3372 size_t size = mTracks.size(); 3373 for (size_t i = 0; i < size; i++) { 3374 sp<Track> t = mTracks[i]; 3375 if (t->streamType() == streamType) { 3376 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3377 t->mCblk->cv.signal(); 3378 } 3379 } 3380} 3381 3382// getTrackName_l() must be called with ThreadBase::mLock held 3383int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3384{ 3385 return mAudioMixer->getTrackName(channelMask); 3386} 3387 3388// deleteTrackName_l() must be called with ThreadBase::mLock held 3389void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3390{ 3391 ALOGV("remove track (%d) and delete from mixer", name); 3392 mAudioMixer->deleteTrackName(name); 3393} 3394 3395// checkForNewParameters_l() must be called with ThreadBase::mLock held 3396bool AudioFlinger::MixerThread::checkForNewParameters_l() 3397{ 3398 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3399 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3400 bool reconfig = false; 3401 3402 while (!mNewParameters.isEmpty()) { 3403 3404 if (mFastMixer != NULL) { 3405 FastMixerStateQueue *sq = mFastMixer->sq(); 3406 FastMixerState *state = sq->begin(); 3407 if (!(state->mCommand & FastMixerState::IDLE)) { 3408 previousCommand = state->mCommand; 3409 state->mCommand = FastMixerState::HOT_IDLE; 3410 sq->end(); 3411 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3412 } else { 3413 sq->end(false /*didModify*/); 3414 } 3415 } 3416 3417 status_t status = NO_ERROR; 3418 String8 keyValuePair = mNewParameters[0]; 3419 AudioParameter param = AudioParameter(keyValuePair); 3420 int value; 3421 3422 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3423 reconfig = true; 3424 } 3425 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3426 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3427 status = BAD_VALUE; 3428 } else { 3429 reconfig = true; 3430 } 3431 } 3432 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3433 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3434 status = BAD_VALUE; 3435 } else { 3436 reconfig = true; 3437 } 3438 } 3439 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3440 // do not accept frame count changes if tracks are open as the track buffer 3441 // size depends on frame count and correct behavior would not be guaranteed 3442 // if frame count is changed after track creation 3443 if (!mTracks.isEmpty()) { 3444 status = INVALID_OPERATION; 3445 } else { 3446 reconfig = true; 3447 } 3448 } 3449 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3450#ifdef ADD_BATTERY_DATA 3451 // when changing the audio output device, call addBatteryData to notify 3452 // the change 3453 if (mDevice != value) { 3454 uint32_t params = 0; 3455 // check whether speaker is on 3456 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3457 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3458 } 3459 3460 audio_devices_t deviceWithoutSpeaker 3461 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3462 // check if any other device (except speaker) is on 3463 if (value & deviceWithoutSpeaker ) { 3464 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3465 } 3466 3467 if (params != 0) { 3468 addBatteryData(params); 3469 } 3470 } 3471#endif 3472 3473 // forward device change to effects that have requested to be 3474 // aware of attached audio device. 3475 mDevice = value; 3476 for (size_t i = 0; i < mEffectChains.size(); i++) { 3477 mEffectChains[i]->setDevice_l(mDevice); 3478 } 3479 } 3480 3481 if (status == NO_ERROR) { 3482 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3483 keyValuePair.string()); 3484 if (!mStandby && status == INVALID_OPERATION) { 3485 mOutput->stream->common.standby(&mOutput->stream->common); 3486 mStandby = true; 3487 mBytesWritten = 0; 3488 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3489 keyValuePair.string()); 3490 } 3491 if (status == NO_ERROR && reconfig) { 3492 delete mAudioMixer; 3493 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3494 mAudioMixer = NULL; 3495 readOutputParameters(); 3496 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3497 for (size_t i = 0; i < mTracks.size() ; i++) { 3498 int name = getTrackName_l(mTracks[i]->mChannelMask); 3499 if (name < 0) break; 3500 mTracks[i]->mName = name; 3501 // limit track sample rate to 2 x new output sample rate 3502 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3503 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3504 } 3505 } 3506 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3507 } 3508 } 3509 3510 mNewParameters.removeAt(0); 3511 3512 mParamStatus = status; 3513 mParamCond.signal(); 3514 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3515 // already timed out waiting for the status and will never signal the condition. 3516 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3517 } 3518 3519 if (!(previousCommand & FastMixerState::IDLE)) { 3520 ALOG_ASSERT(mFastMixer != NULL); 3521 FastMixerStateQueue *sq = mFastMixer->sq(); 3522 FastMixerState *state = sq->begin(); 3523 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3524 state->mCommand = previousCommand; 3525 sq->end(); 3526 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3527 } 3528 3529 return reconfig; 3530} 3531 3532void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3533{ 3534 const size_t SIZE = 256; 3535 char buffer[SIZE]; 3536 String8 result; 3537 3538 PlaybackThread::dumpInternals(fd, args); 3539 3540 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3541 result.append(buffer); 3542 write(fd, result.string(), result.size()); 3543 3544 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3545 FastMixerDumpState copy = mFastMixerDumpState; 3546 copy.dump(fd); 3547 3548#ifdef STATE_QUEUE_DUMP 3549 // Similar for state queue 3550 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3551 observerCopy.dump(fd); 3552 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3553 mutatorCopy.dump(fd); 3554#endif 3555 3556 // Write the tee output to a .wav file 3557 NBAIO_Source *teeSource = mTeeSource.get(); 3558 if (teeSource != NULL) { 3559 char teePath[64]; 3560 struct timeval tv; 3561 gettimeofday(&tv, NULL); 3562 struct tm tm; 3563 localtime_r(&tv.tv_sec, &tm); 3564 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3565 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3566 if (teeFd >= 0) { 3567 char wavHeader[44]; 3568 memcpy(wavHeader, 3569 "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", 3570 sizeof(wavHeader)); 3571 NBAIO_Format format = teeSource->format(); 3572 unsigned channelCount = Format_channelCount(format); 3573 ALOG_ASSERT(channelCount <= FCC_2); 3574 unsigned sampleRate = Format_sampleRate(format); 3575 wavHeader[22] = channelCount; // number of channels 3576 wavHeader[24] = sampleRate; // sample rate 3577 wavHeader[25] = sampleRate >> 8; 3578 wavHeader[32] = channelCount * 2; // block alignment 3579 write(teeFd, wavHeader, sizeof(wavHeader)); 3580 size_t total = 0; 3581 bool firstRead = true; 3582 for (;;) { 3583#define TEE_SINK_READ 1024 3584 short buffer[TEE_SINK_READ * FCC_2]; 3585 size_t count = TEE_SINK_READ; 3586 ssize_t actual = teeSource->read(buffer, count, 3587 AudioBufferProvider::kInvalidPTS); 3588 bool wasFirstRead = firstRead; 3589 firstRead = false; 3590 if (actual <= 0) { 3591 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3592 continue; 3593 } 3594 break; 3595 } 3596 ALOG_ASSERT(actual <= (ssize_t)count); 3597 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3598 total += actual; 3599 } 3600 lseek(teeFd, (off_t) 4, SEEK_SET); 3601 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3602 write(teeFd, &temp, sizeof(temp)); 3603 lseek(teeFd, (off_t) 40, SEEK_SET); 3604 temp = total * channelCount * sizeof(short); 3605 write(teeFd, &temp, sizeof(temp)); 3606 close(teeFd); 3607 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3608 } else { 3609 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3610 } 3611 } 3612 3613 if (mAudioWatchdog != 0) { 3614 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3615 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3616 wdCopy.dump(fd); 3617 } 3618} 3619 3620uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3621{ 3622 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3623} 3624 3625uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3626{ 3627 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3628} 3629 3630void AudioFlinger::MixerThread::cacheParameters_l() 3631{ 3632 PlaybackThread::cacheParameters_l(); 3633 3634 // FIXME: Relaxed timing because of a certain device that can't meet latency 3635 // Should be reduced to 2x after the vendor fixes the driver issue 3636 // increase threshold again due to low power audio mode. The way this warning 3637 // threshold is calculated and its usefulness should be reconsidered anyway. 3638 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3639} 3640 3641// ---------------------------------------------------------------------------- 3642AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3643 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3644 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3645 // mLeftVolFloat, mRightVolFloat 3646{ 3647} 3648 3649AudioFlinger::DirectOutputThread::~DirectOutputThread() 3650{ 3651} 3652 3653AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3654 Vector< sp<Track> > *tracksToRemove 3655) 3656{ 3657 sp<Track> trackToRemove; 3658 3659 mixer_state mixerStatus = MIXER_IDLE; 3660 3661 // find out which tracks need to be processed 3662 if (mActiveTracks.size() != 0) { 3663 sp<Track> t = mActiveTracks[0].promote(); 3664 // The track died recently 3665 if (t == 0) return MIXER_IDLE; 3666 3667 Track* const track = t.get(); 3668 audio_track_cblk_t* cblk = track->cblk(); 3669 3670 // The first time a track is added we wait 3671 // for all its buffers to be filled before processing it 3672 uint32_t minFrames; 3673 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3674 minFrames = mNormalFrameCount; 3675 } else { 3676 minFrames = 1; 3677 } 3678 if ((track->framesReady() >= minFrames) && track->isReady() && 3679 !track->isPaused() && !track->isTerminated()) 3680 { 3681 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3682 3683 if (track->mFillingUpStatus == Track::FS_FILLED) { 3684 track->mFillingUpStatus = Track::FS_ACTIVE; 3685 mLeftVolFloat = mRightVolFloat = 0; 3686 if (track->mState == TrackBase::RESUMING) { 3687 track->mState = TrackBase::ACTIVE; 3688 } 3689 } 3690 3691 // compute volume for this track 3692 float left, right; 3693 if (track->isMuted() || mMasterMute || track->isPausing() || 3694 mStreamTypes[track->streamType()].mute) { 3695 left = right = 0; 3696 if (track->isPausing()) { 3697 track->setPaused(); 3698 } 3699 } else { 3700 float typeVolume = mStreamTypes[track->streamType()].volume; 3701 float v = mMasterVolume * typeVolume; 3702 uint32_t vlr = cblk->getVolumeLR(); 3703 float v_clamped = v * (vlr & 0xFFFF); 3704 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3705 left = v_clamped/MAX_GAIN; 3706 v_clamped = v * (vlr >> 16); 3707 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3708 right = v_clamped/MAX_GAIN; 3709 } 3710 3711 if (left != mLeftVolFloat || right != mRightVolFloat) { 3712 mLeftVolFloat = left; 3713 mRightVolFloat = right; 3714 3715 // Convert volumes from float to 8.24 3716 uint32_t vl = (uint32_t)(left * (1 << 24)); 3717 uint32_t vr = (uint32_t)(right * (1 << 24)); 3718 3719 // Delegate volume control to effect in track effect chain if needed 3720 // only one effect chain can be present on DirectOutputThread, so if 3721 // there is one, the track is connected to it 3722 if (!mEffectChains.isEmpty()) { 3723 // Do not ramp volume if volume is controlled by effect 3724 mEffectChains[0]->setVolume_l(&vl, &vr); 3725 left = (float)vl / (1 << 24); 3726 right = (float)vr / (1 << 24); 3727 } 3728 mOutput->stream->set_volume(mOutput->stream, left, right); 3729 } 3730 3731 // reset retry count 3732 track->mRetryCount = kMaxTrackRetriesDirect; 3733 mActiveTrack = t; 3734 mixerStatus = MIXER_TRACKS_READY; 3735 } else { 3736 // clear effect chain input buffer if an active track underruns to avoid sending 3737 // previous audio buffer again to effects 3738 if (!mEffectChains.isEmpty()) { 3739 mEffectChains[0]->clearInputBuffer(); 3740 } 3741 3742 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3743 if ((track->sharedBuffer() != 0) || 3744 track->isStopped() || track->isPaused()) { 3745 // We have consumed all the buffers of this track. 3746 // Remove it from the list of active tracks. 3747 // TODO: implement behavior for compressed audio 3748 size_t audioHALFrames = 3749 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3750 size_t framesWritten = 3751 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3752 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3753 if (track->isStopped()) { 3754 track->reset(); 3755 } 3756 trackToRemove = track; 3757 } 3758 } else { 3759 // No buffers for this track. Give it a few chances to 3760 // fill a buffer, then remove it from active list. 3761 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3762 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3763 trackToRemove = track; 3764 } else { 3765 mixerStatus = MIXER_TRACKS_ENABLED; 3766 } 3767 } 3768 } 3769 } 3770 3771 // FIXME merge this with similar code for removing multiple tracks 3772 // remove all the tracks that need to be... 3773 if (CC_UNLIKELY(trackToRemove != 0)) { 3774 tracksToRemove->add(trackToRemove); 3775 mActiveTracks.remove(trackToRemove); 3776 if (!mEffectChains.isEmpty()) { 3777 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3778 trackToRemove->sessionId()); 3779 mEffectChains[0]->decActiveTrackCnt(); 3780 } 3781 if (trackToRemove->isTerminated()) { 3782 removeTrack_l(trackToRemove); 3783 } 3784 } 3785 3786 return mixerStatus; 3787} 3788 3789void AudioFlinger::DirectOutputThread::threadLoop_mix() 3790{ 3791 AudioBufferProvider::Buffer buffer; 3792 size_t frameCount = mFrameCount; 3793 int8_t *curBuf = (int8_t *)mMixBuffer; 3794 // output audio to hardware 3795 while (frameCount) { 3796 buffer.frameCount = frameCount; 3797 mActiveTrack->getNextBuffer(&buffer); 3798 if (CC_UNLIKELY(buffer.raw == NULL)) { 3799 memset(curBuf, 0, frameCount * mFrameSize); 3800 break; 3801 } 3802 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3803 frameCount -= buffer.frameCount; 3804 curBuf += buffer.frameCount * mFrameSize; 3805 mActiveTrack->releaseBuffer(&buffer); 3806 } 3807 sleepTime = 0; 3808 standbyTime = systemTime() + standbyDelay; 3809 mActiveTrack.clear(); 3810 3811} 3812 3813void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3814{ 3815 if (sleepTime == 0) { 3816 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3817 sleepTime = activeSleepTime; 3818 } else { 3819 sleepTime = idleSleepTime; 3820 } 3821 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3822 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3823 sleepTime = 0; 3824 } 3825} 3826 3827// getTrackName_l() must be called with ThreadBase::mLock held 3828int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3829{ 3830 return 0; 3831} 3832 3833// deleteTrackName_l() must be called with ThreadBase::mLock held 3834void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3835{ 3836} 3837 3838// checkForNewParameters_l() must be called with ThreadBase::mLock held 3839bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3840{ 3841 bool reconfig = false; 3842 3843 while (!mNewParameters.isEmpty()) { 3844 status_t status = NO_ERROR; 3845 String8 keyValuePair = mNewParameters[0]; 3846 AudioParameter param = AudioParameter(keyValuePair); 3847 int value; 3848 3849 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3850 // do not accept frame count changes if tracks are open as the track buffer 3851 // size depends on frame count and correct behavior would not be garantied 3852 // if frame count is changed after track creation 3853 if (!mTracks.isEmpty()) { 3854 status = INVALID_OPERATION; 3855 } else { 3856 reconfig = true; 3857 } 3858 } 3859 if (status == NO_ERROR) { 3860 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3861 keyValuePair.string()); 3862 if (!mStandby && status == INVALID_OPERATION) { 3863 mOutput->stream->common.standby(&mOutput->stream->common); 3864 mStandby = true; 3865 mBytesWritten = 0; 3866 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3867 keyValuePair.string()); 3868 } 3869 if (status == NO_ERROR && reconfig) { 3870 readOutputParameters(); 3871 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3872 } 3873 } 3874 3875 mNewParameters.removeAt(0); 3876 3877 mParamStatus = status; 3878 mParamCond.signal(); 3879 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3880 // already timed out waiting for the status and will never signal the condition. 3881 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3882 } 3883 return reconfig; 3884} 3885 3886uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3887{ 3888 uint32_t time; 3889 if (audio_is_linear_pcm(mFormat)) { 3890 time = PlaybackThread::activeSleepTimeUs(); 3891 } else { 3892 time = 10000; 3893 } 3894 return time; 3895} 3896 3897uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3898{ 3899 uint32_t time; 3900 if (audio_is_linear_pcm(mFormat)) { 3901 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3902 } else { 3903 time = 10000; 3904 } 3905 return time; 3906} 3907 3908uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3909{ 3910 uint32_t time; 3911 if (audio_is_linear_pcm(mFormat)) { 3912 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3913 } else { 3914 time = 10000; 3915 } 3916 return time; 3917} 3918 3919void AudioFlinger::DirectOutputThread::cacheParameters_l() 3920{ 3921 PlaybackThread::cacheParameters_l(); 3922 3923 // use shorter standby delay as on normal output to release 3924 // hardware resources as soon as possible 3925 standbyDelay = microseconds(activeSleepTime*2); 3926} 3927 3928// ---------------------------------------------------------------------------- 3929 3930AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3931 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3932 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3933 mWaitTimeMs(UINT_MAX) 3934{ 3935 addOutputTrack(mainThread); 3936} 3937 3938AudioFlinger::DuplicatingThread::~DuplicatingThread() 3939{ 3940 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3941 mOutputTracks[i]->destroy(); 3942 } 3943} 3944 3945void AudioFlinger::DuplicatingThread::threadLoop_mix() 3946{ 3947 // mix buffers... 3948 if (outputsReady(outputTracks)) { 3949 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3950 } else { 3951 memset(mMixBuffer, 0, mixBufferSize); 3952 } 3953 sleepTime = 0; 3954 writeFrames = mNormalFrameCount; 3955 standbyTime = systemTime() + standbyDelay; 3956} 3957 3958void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3959{ 3960 if (sleepTime == 0) { 3961 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3962 sleepTime = activeSleepTime; 3963 } else { 3964 sleepTime = idleSleepTime; 3965 } 3966 } else if (mBytesWritten != 0) { 3967 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3968 writeFrames = mNormalFrameCount; 3969 memset(mMixBuffer, 0, mixBufferSize); 3970 } else { 3971 // flush remaining overflow buffers in output tracks 3972 writeFrames = 0; 3973 } 3974 sleepTime = 0; 3975 } 3976} 3977 3978void AudioFlinger::DuplicatingThread::threadLoop_write() 3979{ 3980 for (size_t i = 0; i < outputTracks.size(); i++) { 3981 outputTracks[i]->write(mMixBuffer, writeFrames); 3982 } 3983 mBytesWritten += mixBufferSize; 3984} 3985 3986void AudioFlinger::DuplicatingThread::threadLoop_standby() 3987{ 3988 // DuplicatingThread implements standby by stopping all tracks 3989 for (size_t i = 0; i < outputTracks.size(); i++) { 3990 outputTracks[i]->stop(); 3991 } 3992} 3993 3994void AudioFlinger::DuplicatingThread::saveOutputTracks() 3995{ 3996 outputTracks = mOutputTracks; 3997} 3998 3999void AudioFlinger::DuplicatingThread::clearOutputTracks() 4000{ 4001 outputTracks.clear(); 4002} 4003 4004void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4005{ 4006 Mutex::Autolock _l(mLock); 4007 // FIXME explain this formula 4008 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4009 OutputTrack *outputTrack = new OutputTrack(thread, 4010 this, 4011 mSampleRate, 4012 mFormat, 4013 mChannelMask, 4014 frameCount); 4015 if (outputTrack->cblk() != NULL) { 4016 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4017 mOutputTracks.add(outputTrack); 4018 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4019 updateWaitTime_l(); 4020 } 4021} 4022 4023void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4024{ 4025 Mutex::Autolock _l(mLock); 4026 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4027 if (mOutputTracks[i]->thread() == thread) { 4028 mOutputTracks[i]->destroy(); 4029 mOutputTracks.removeAt(i); 4030 updateWaitTime_l(); 4031 return; 4032 } 4033 } 4034 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4035} 4036 4037// caller must hold mLock 4038void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4039{ 4040 mWaitTimeMs = UINT_MAX; 4041 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4042 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4043 if (strong != 0) { 4044 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4045 if (waitTimeMs < mWaitTimeMs) { 4046 mWaitTimeMs = waitTimeMs; 4047 } 4048 } 4049 } 4050} 4051 4052 4053bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4054{ 4055 for (size_t i = 0; i < outputTracks.size(); i++) { 4056 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4057 if (thread == 0) { 4058 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4059 return false; 4060 } 4061 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4062 // see note at standby() declaration 4063 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4064 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4065 return false; 4066 } 4067 } 4068 return true; 4069} 4070 4071uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4072{ 4073 return (mWaitTimeMs * 1000) / 2; 4074} 4075 4076void AudioFlinger::DuplicatingThread::cacheParameters_l() 4077{ 4078 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4079 updateWaitTime_l(); 4080 4081 MixerThread::cacheParameters_l(); 4082} 4083 4084// ---------------------------------------------------------------------------- 4085 4086// TrackBase constructor must be called with AudioFlinger::mLock held 4087AudioFlinger::ThreadBase::TrackBase::TrackBase( 4088 ThreadBase *thread, 4089 const sp<Client>& client, 4090 uint32_t sampleRate, 4091 audio_format_t format, 4092 audio_channel_mask_t channelMask, 4093 int frameCount, 4094 const sp<IMemory>& sharedBuffer, 4095 int sessionId) 4096 : RefBase(), 4097 mThread(thread), 4098 mClient(client), 4099 mCblk(NULL), 4100 // mBuffer 4101 // mBufferEnd 4102 mFrameCount(0), 4103 mState(IDLE), 4104 mSampleRate(sampleRate), 4105 mFormat(format), 4106 mStepServerFailed(false), 4107 mSessionId(sessionId) 4108 // mChannelCount 4109 // mChannelMask 4110{ 4111 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4112 4113 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4114 size_t size = sizeof(audio_track_cblk_t); 4115 uint8_t channelCount = popcount(channelMask); 4116 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4117 if (sharedBuffer == 0) { 4118 size += bufferSize; 4119 } 4120 4121 if (client != NULL) { 4122 mCblkMemory = client->heap()->allocate(size); 4123 if (mCblkMemory != 0) { 4124 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4125 if (mCblk != NULL) { // construct the shared structure in-place. 4126 new(mCblk) audio_track_cblk_t(); 4127 // clear all buffers 4128 mCblk->frameCount = frameCount; 4129 mCblk->sampleRate = sampleRate; 4130// uncomment the following lines to quickly test 32-bit wraparound 4131// mCblk->user = 0xffff0000; 4132// mCblk->server = 0xffff0000; 4133// mCblk->userBase = 0xffff0000; 4134// mCblk->serverBase = 0xffff0000; 4135 mChannelCount = channelCount; 4136 mChannelMask = channelMask; 4137 if (sharedBuffer == 0) { 4138 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4139 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4140 // Force underrun condition to avoid false underrun callback until first data is 4141 // written to buffer (other flags are cleared) 4142 mCblk->flags = CBLK_UNDERRUN_ON; 4143 } else { 4144 mBuffer = sharedBuffer->pointer(); 4145 } 4146 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4147 } 4148 } else { 4149 ALOGE("not enough memory for AudioTrack size=%u", size); 4150 client->heap()->dump("AudioTrack"); 4151 return; 4152 } 4153 } else { 4154 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4155 // construct the shared structure in-place. 4156 new(mCblk) audio_track_cblk_t(); 4157 // clear all buffers 4158 mCblk->frameCount = frameCount; 4159 mCblk->sampleRate = sampleRate; 4160// uncomment the following lines to quickly test 32-bit wraparound 4161// mCblk->user = 0xffff0000; 4162// mCblk->server = 0xffff0000; 4163// mCblk->userBase = 0xffff0000; 4164// mCblk->serverBase = 0xffff0000; 4165 mChannelCount = channelCount; 4166 mChannelMask = channelMask; 4167 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4168 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4169 // Force underrun condition to avoid false underrun callback until first data is 4170 // written to buffer (other flags are cleared) 4171 mCblk->flags = CBLK_UNDERRUN_ON; 4172 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4173 } 4174} 4175 4176AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4177{ 4178 if (mCblk != NULL) { 4179 if (mClient == 0) { 4180 delete mCblk; 4181 } else { 4182 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4183 } 4184 } 4185 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4186 if (mClient != 0) { 4187 // Client destructor must run with AudioFlinger mutex locked 4188 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4189 // If the client's reference count drops to zero, the associated destructor 4190 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4191 // relying on the automatic clear() at end of scope. 4192 mClient.clear(); 4193 } 4194} 4195 4196// AudioBufferProvider interface 4197// getNextBuffer() = 0; 4198// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4199void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4200{ 4201 buffer->raw = NULL; 4202 mFrameCount = buffer->frameCount; 4203 // FIXME See note at getNextBuffer() 4204 (void) step(); // ignore return value of step() 4205 buffer->frameCount = 0; 4206} 4207 4208bool AudioFlinger::ThreadBase::TrackBase::step() { 4209 bool result; 4210 audio_track_cblk_t* cblk = this->cblk(); 4211 4212 result = cblk->stepServer(mFrameCount); 4213 if (!result) { 4214 ALOGV("stepServer failed acquiring cblk mutex"); 4215 mStepServerFailed = true; 4216 } 4217 return result; 4218} 4219 4220void AudioFlinger::ThreadBase::TrackBase::reset() { 4221 audio_track_cblk_t* cblk = this->cblk(); 4222 4223 cblk->user = 0; 4224 cblk->server = 0; 4225 cblk->userBase = 0; 4226 cblk->serverBase = 0; 4227 mStepServerFailed = false; 4228 ALOGV("TrackBase::reset"); 4229} 4230 4231int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4232 return (int)mCblk->sampleRate; 4233} 4234 4235void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4236 audio_track_cblk_t* cblk = this->cblk(); 4237 size_t frameSize = cblk->frameSize; 4238 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4239 int8_t *bufferEnd = bufferStart + frames * frameSize; 4240 4241 // Check validity of returned pointer in case the track control block would have been corrupted. 4242 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4243 "TrackBase::getBuffer buffer out of range:\n" 4244 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4245 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4246 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4247 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4248 4249 return bufferStart; 4250} 4251 4252status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4253{ 4254 mSyncEvents.add(event); 4255 return NO_ERROR; 4256} 4257 4258// ---------------------------------------------------------------------------- 4259 4260// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4261AudioFlinger::PlaybackThread::Track::Track( 4262 PlaybackThread *thread, 4263 const sp<Client>& client, 4264 audio_stream_type_t streamType, 4265 uint32_t sampleRate, 4266 audio_format_t format, 4267 audio_channel_mask_t channelMask, 4268 int frameCount, 4269 const sp<IMemory>& sharedBuffer, 4270 int sessionId, 4271 IAudioFlinger::track_flags_t flags) 4272 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4273 mMute(false), 4274 mFillingUpStatus(FS_INVALID), 4275 // mRetryCount initialized later when needed 4276 mSharedBuffer(sharedBuffer), 4277 mStreamType(streamType), 4278 mName(-1), // see note below 4279 mMainBuffer(thread->mixBuffer()), 4280 mAuxBuffer(NULL), 4281 mAuxEffectId(0), mHasVolumeController(false), 4282 mPresentationCompleteFrames(0), 4283 mFlags(flags), 4284 mFastIndex(-1), 4285 mUnderrunCount(0), 4286 mCachedVolume(1.0) 4287{ 4288 if (mCblk != NULL) { 4289 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4290 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4291 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4292 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4293 mName = thread->getTrackName_l(channelMask); 4294 mCblk->mName = mName; 4295 if (mName < 0) { 4296 ALOGE("no more track names available"); 4297 return; 4298 } 4299 // only allocate a fast track index if we were able to allocate a normal track name 4300 if (flags & IAudioFlinger::TRACK_FAST) { 4301 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4302 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4303 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4304 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4305 // FIXME This is too eager. We allocate a fast track index before the 4306 // fast track becomes active. Since fast tracks are a scarce resource, 4307 // this means we are potentially denying other more important fast tracks from 4308 // being created. It would be better to allocate the index dynamically. 4309 mFastIndex = i; 4310 mCblk->mName = i; 4311 // Read the initial underruns because this field is never cleared by the fast mixer 4312 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4313 thread->mFastTrackAvailMask &= ~(1 << i); 4314 } 4315 } 4316 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4317} 4318 4319AudioFlinger::PlaybackThread::Track::~Track() 4320{ 4321 ALOGV("PlaybackThread::Track destructor"); 4322} 4323 4324void AudioFlinger::PlaybackThread::Track::destroy() 4325{ 4326 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4327 // by removing it from mTracks vector, so there is a risk that this Tracks's 4328 // destructor is called. As the destructor needs to lock mLock, 4329 // we must acquire a strong reference on this Track before locking mLock 4330 // here so that the destructor is called only when exiting this function. 4331 // On the other hand, as long as Track::destroy() is only called by 4332 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4333 // this Track with its member mTrack. 4334 sp<Track> keep(this); 4335 { // scope for mLock 4336 sp<ThreadBase> thread = mThread.promote(); 4337 if (thread != 0) { 4338 if (!isOutputTrack()) { 4339 if (mState == ACTIVE || mState == RESUMING) { 4340 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4341 4342#ifdef ADD_BATTERY_DATA 4343 // to track the speaker usage 4344 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4345#endif 4346 } 4347 AudioSystem::releaseOutput(thread->id()); 4348 } 4349 Mutex::Autolock _l(thread->mLock); 4350 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4351 playbackThread->destroyTrack_l(this); 4352 } 4353 } 4354} 4355 4356/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4357{ 4358 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4359 " Server User Main buf Aux Buf Flags Underruns\n"); 4360} 4361 4362void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4363{ 4364 uint32_t vlr = mCblk->getVolumeLR(); 4365 if (isFastTrack()) { 4366 sprintf(buffer, " F %2d", mFastIndex); 4367 } else { 4368 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4369 } 4370 track_state state = mState; 4371 char stateChar; 4372 switch (state) { 4373 case IDLE: 4374 stateChar = 'I'; 4375 break; 4376 case TERMINATED: 4377 stateChar = 'T'; 4378 break; 4379 case STOPPING_1: 4380 stateChar = 's'; 4381 break; 4382 case STOPPING_2: 4383 stateChar = '5'; 4384 break; 4385 case STOPPED: 4386 stateChar = 'S'; 4387 break; 4388 case RESUMING: 4389 stateChar = 'R'; 4390 break; 4391 case ACTIVE: 4392 stateChar = 'A'; 4393 break; 4394 case PAUSING: 4395 stateChar = 'p'; 4396 break; 4397 case PAUSED: 4398 stateChar = 'P'; 4399 break; 4400 case FLUSHED: 4401 stateChar = 'F'; 4402 break; 4403 default: 4404 stateChar = '?'; 4405 break; 4406 } 4407 char nowInUnderrun; 4408 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4409 case UNDERRUN_FULL: 4410 nowInUnderrun = ' '; 4411 break; 4412 case UNDERRUN_PARTIAL: 4413 nowInUnderrun = '<'; 4414 break; 4415 case UNDERRUN_EMPTY: 4416 nowInUnderrun = '*'; 4417 break; 4418 default: 4419 nowInUnderrun = '?'; 4420 break; 4421 } 4422 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4423 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4424 (mClient == 0) ? getpid_cached : mClient->pid(), 4425 mStreamType, 4426 mFormat, 4427 mChannelMask, 4428 mSessionId, 4429 mFrameCount, 4430 mCblk->frameCount, 4431 stateChar, 4432 mMute, 4433 mFillingUpStatus, 4434 mCblk->sampleRate, 4435 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4436 20.0 * log10((vlr >> 16) / 4096.0), 4437 mCblk->server, 4438 mCblk->user, 4439 (int)mMainBuffer, 4440 (int)mAuxBuffer, 4441 mCblk->flags, 4442 mUnderrunCount, 4443 nowInUnderrun); 4444} 4445 4446// AudioBufferProvider interface 4447status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4448 AudioBufferProvider::Buffer* buffer, int64_t pts) 4449{ 4450 audio_track_cblk_t* cblk = this->cblk(); 4451 uint32_t framesReady; 4452 uint32_t framesReq = buffer->frameCount; 4453 4454 // Check if last stepServer failed, try to step now 4455 if (mStepServerFailed) { 4456 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4457 // Since the fast mixer is higher priority than client callback thread, 4458 // it does not result in priority inversion for client. 4459 // But a non-blocking solution would be preferable to avoid 4460 // fast mixer being unable to tryLock(), and 4461 // to avoid the extra context switches if the client wakes up, 4462 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4463 if (!step()) goto getNextBuffer_exit; 4464 ALOGV("stepServer recovered"); 4465 mStepServerFailed = false; 4466 } 4467 4468 // FIXME Same as above 4469 framesReady = cblk->framesReady(); 4470 4471 if (CC_LIKELY(framesReady)) { 4472 uint32_t s = cblk->server; 4473 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4474 4475 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4476 if (framesReq > framesReady) { 4477 framesReq = framesReady; 4478 } 4479 if (framesReq > bufferEnd - s) { 4480 framesReq = bufferEnd - s; 4481 } 4482 4483 buffer->raw = getBuffer(s, framesReq); 4484 buffer->frameCount = framesReq; 4485 return NO_ERROR; 4486 } 4487 4488getNextBuffer_exit: 4489 buffer->raw = NULL; 4490 buffer->frameCount = 0; 4491 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4492 return NOT_ENOUGH_DATA; 4493} 4494 4495// Note that framesReady() takes a mutex on the control block using tryLock(). 4496// This could result in priority inversion if framesReady() is called by the normal mixer, 4497// as the normal mixer thread runs at lower 4498// priority than the client's callback thread: there is a short window within framesReady() 4499// during which the normal mixer could be preempted, and the client callback would block. 4500// Another problem can occur if framesReady() is called by the fast mixer: 4501// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4502// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4503size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4504 return mCblk->framesReady(); 4505} 4506 4507// Don't call for fast tracks; the framesReady() could result in priority inversion 4508bool AudioFlinger::PlaybackThread::Track::isReady() const { 4509 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4510 4511 if (framesReady() >= mCblk->frameCount || 4512 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4513 mFillingUpStatus = FS_FILLED; 4514 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4515 return true; 4516 } 4517 return false; 4518} 4519 4520status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4521 int triggerSession) 4522{ 4523 status_t status = NO_ERROR; 4524 ALOGV("start(%d), calling pid %d session %d", 4525 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4526 4527 sp<ThreadBase> thread = mThread.promote(); 4528 if (thread != 0) { 4529 Mutex::Autolock _l(thread->mLock); 4530 track_state state = mState; 4531 // here the track could be either new, or restarted 4532 // in both cases "unstop" the track 4533 if (mState == PAUSED) { 4534 mState = TrackBase::RESUMING; 4535 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4536 } else { 4537 mState = TrackBase::ACTIVE; 4538 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4539 } 4540 4541 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4542 thread->mLock.unlock(); 4543 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4544 thread->mLock.lock(); 4545 4546#ifdef ADD_BATTERY_DATA 4547 // to track the speaker usage 4548 if (status == NO_ERROR) { 4549 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4550 } 4551#endif 4552 } 4553 if (status == NO_ERROR) { 4554 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4555 playbackThread->addTrack_l(this); 4556 } else { 4557 mState = state; 4558 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4559 } 4560 } else { 4561 status = BAD_VALUE; 4562 } 4563 return status; 4564} 4565 4566void AudioFlinger::PlaybackThread::Track::stop() 4567{ 4568 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4569 sp<ThreadBase> thread = mThread.promote(); 4570 if (thread != 0) { 4571 Mutex::Autolock _l(thread->mLock); 4572 track_state state = mState; 4573 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4574 // If the track is not active (PAUSED and buffers full), flush buffers 4575 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4576 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4577 reset(); 4578 mState = STOPPED; 4579 } else if (!isFastTrack()) { 4580 mState = STOPPED; 4581 } else { 4582 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4583 // and then to STOPPED and reset() when presentation is complete 4584 mState = STOPPING_1; 4585 } 4586 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4587 } 4588 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4589 thread->mLock.unlock(); 4590 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4591 thread->mLock.lock(); 4592 4593#ifdef ADD_BATTERY_DATA 4594 // to track the speaker usage 4595 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4596#endif 4597 } 4598 } 4599} 4600 4601void AudioFlinger::PlaybackThread::Track::pause() 4602{ 4603 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4604 sp<ThreadBase> thread = mThread.promote(); 4605 if (thread != 0) { 4606 Mutex::Autolock _l(thread->mLock); 4607 if (mState == ACTIVE || mState == RESUMING) { 4608 mState = PAUSING; 4609 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4610 if (!isOutputTrack()) { 4611 thread->mLock.unlock(); 4612 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4613 thread->mLock.lock(); 4614 4615#ifdef ADD_BATTERY_DATA 4616 // to track the speaker usage 4617 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4618#endif 4619 } 4620 } 4621 } 4622} 4623 4624void AudioFlinger::PlaybackThread::Track::flush() 4625{ 4626 ALOGV("flush(%d)", mName); 4627 sp<ThreadBase> thread = mThread.promote(); 4628 if (thread != 0) { 4629 Mutex::Autolock _l(thread->mLock); 4630 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4631 mState != PAUSING) { 4632 return; 4633 } 4634 // No point remaining in PAUSED state after a flush => go to 4635 // FLUSHED state 4636 mState = FLUSHED; 4637 // do not reset the track if it is still in the process of being stopped or paused. 4638 // this will be done by prepareTracks_l() when the track is stopped. 4639 // prepareTracks_l() will see mState == FLUSHED, then 4640 // remove from active track list, reset(), and trigger presentation complete 4641 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4642 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4643 reset(); 4644 } 4645 } 4646} 4647 4648void AudioFlinger::PlaybackThread::Track::reset() 4649{ 4650 // Do not reset twice to avoid discarding data written just after a flush and before 4651 // the audioflinger thread detects the track is stopped. 4652 if (!mResetDone) { 4653 TrackBase::reset(); 4654 // Force underrun condition to avoid false underrun callback until first data is 4655 // written to buffer 4656 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4657 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4658 mFillingUpStatus = FS_FILLING; 4659 mResetDone = true; 4660 if (mState == FLUSHED) { 4661 mState = IDLE; 4662 } 4663 } 4664} 4665 4666void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4667{ 4668 mMute = muted; 4669} 4670 4671status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4672{ 4673 status_t status = DEAD_OBJECT; 4674 sp<ThreadBase> thread = mThread.promote(); 4675 if (thread != 0) { 4676 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4677 sp<AudioFlinger> af = mClient->audioFlinger(); 4678 4679 Mutex::Autolock _l(af->mLock); 4680 4681 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4682 4683 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4684 Mutex::Autolock _dl(playbackThread->mLock); 4685 Mutex::Autolock _sl(srcThread->mLock); 4686 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4687 if (chain == 0) { 4688 return INVALID_OPERATION; 4689 } 4690 4691 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4692 if (effect == 0) { 4693 return INVALID_OPERATION; 4694 } 4695 srcThread->removeEffect_l(effect); 4696 playbackThread->addEffect_l(effect); 4697 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4698 if (effect->state() == EffectModule::ACTIVE || 4699 effect->state() == EffectModule::STOPPING) { 4700 effect->start(); 4701 } 4702 4703 sp<EffectChain> dstChain = effect->chain().promote(); 4704 if (dstChain == 0) { 4705 srcThread->addEffect_l(effect); 4706 return INVALID_OPERATION; 4707 } 4708 AudioSystem::unregisterEffect(effect->id()); 4709 AudioSystem::registerEffect(&effect->desc(), 4710 srcThread->id(), 4711 dstChain->strategy(), 4712 AUDIO_SESSION_OUTPUT_MIX, 4713 effect->id()); 4714 } 4715 status = playbackThread->attachAuxEffect(this, EffectId); 4716 } 4717 return status; 4718} 4719 4720void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4721{ 4722 mAuxEffectId = EffectId; 4723 mAuxBuffer = buffer; 4724} 4725 4726bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4727 size_t audioHalFrames) 4728{ 4729 // a track is considered presented when the total number of frames written to audio HAL 4730 // corresponds to the number of frames written when presentationComplete() is called for the 4731 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4732 if (mPresentationCompleteFrames == 0) { 4733 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4734 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4735 mPresentationCompleteFrames, audioHalFrames); 4736 } 4737 if (framesWritten >= mPresentationCompleteFrames) { 4738 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4739 mSessionId, framesWritten); 4740 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4741 return true; 4742 } 4743 return false; 4744} 4745 4746void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4747{ 4748 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4749 if (mSyncEvents[i]->type() == type) { 4750 mSyncEvents[i]->trigger(); 4751 mSyncEvents.removeAt(i); 4752 i--; 4753 } 4754 } 4755} 4756 4757// implement VolumeBufferProvider interface 4758 4759uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4760{ 4761 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4762 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4763 uint32_t vlr = mCblk->getVolumeLR(); 4764 uint32_t vl = vlr & 0xFFFF; 4765 uint32_t vr = vlr >> 16; 4766 // track volumes come from shared memory, so can't be trusted and must be clamped 4767 if (vl > MAX_GAIN_INT) { 4768 vl = MAX_GAIN_INT; 4769 } 4770 if (vr > MAX_GAIN_INT) { 4771 vr = MAX_GAIN_INT; 4772 } 4773 // now apply the cached master volume and stream type volume; 4774 // this is trusted but lacks any synchronization or barrier so may be stale 4775 float v = mCachedVolume; 4776 vl *= v; 4777 vr *= v; 4778 // re-combine into U4.16 4779 vlr = (vr << 16) | (vl & 0xFFFF); 4780 // FIXME look at mute, pause, and stop flags 4781 return vlr; 4782} 4783 4784status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4785{ 4786 if (mState == TERMINATED || mState == PAUSED || 4787 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4788 (mState == STOPPED)))) { 4789 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4790 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4791 event->cancel(); 4792 return INVALID_OPERATION; 4793 } 4794 (void) TrackBase::setSyncEvent(event); 4795 return NO_ERROR; 4796} 4797 4798// timed audio tracks 4799 4800sp<AudioFlinger::PlaybackThread::TimedTrack> 4801AudioFlinger::PlaybackThread::TimedTrack::create( 4802 PlaybackThread *thread, 4803 const sp<Client>& client, 4804 audio_stream_type_t streamType, 4805 uint32_t sampleRate, 4806 audio_format_t format, 4807 audio_channel_mask_t channelMask, 4808 int frameCount, 4809 const sp<IMemory>& sharedBuffer, 4810 int sessionId) { 4811 if (!client->reserveTimedTrack()) 4812 return 0; 4813 4814 return new TimedTrack( 4815 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4816 sharedBuffer, sessionId); 4817} 4818 4819AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4820 PlaybackThread *thread, 4821 const sp<Client>& client, 4822 audio_stream_type_t streamType, 4823 uint32_t sampleRate, 4824 audio_format_t format, 4825 audio_channel_mask_t channelMask, 4826 int frameCount, 4827 const sp<IMemory>& sharedBuffer, 4828 int sessionId) 4829 : Track(thread, client, streamType, sampleRate, format, channelMask, 4830 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4831 mQueueHeadInFlight(false), 4832 mTrimQueueHeadOnRelease(false), 4833 mFramesPendingInQueue(0), 4834 mTimedSilenceBuffer(NULL), 4835 mTimedSilenceBufferSize(0), 4836 mTimedAudioOutputOnTime(false), 4837 mMediaTimeTransformValid(false) 4838{ 4839 LocalClock lc; 4840 mLocalTimeFreq = lc.getLocalFreq(); 4841 4842 mLocalTimeToSampleTransform.a_zero = 0; 4843 mLocalTimeToSampleTransform.b_zero = 0; 4844 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4845 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4846 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4847 &mLocalTimeToSampleTransform.a_to_b_denom); 4848 4849 mMediaTimeToSampleTransform.a_zero = 0; 4850 mMediaTimeToSampleTransform.b_zero = 0; 4851 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4852 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4853 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4854 &mMediaTimeToSampleTransform.a_to_b_denom); 4855} 4856 4857AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4858 mClient->releaseTimedTrack(); 4859 delete [] mTimedSilenceBuffer; 4860} 4861 4862status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4863 size_t size, sp<IMemory>* buffer) { 4864 4865 Mutex::Autolock _l(mTimedBufferQueueLock); 4866 4867 trimTimedBufferQueue_l(); 4868 4869 // lazily initialize the shared memory heap for timed buffers 4870 if (mTimedMemoryDealer == NULL) { 4871 const int kTimedBufferHeapSize = 512 << 10; 4872 4873 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4874 "AudioFlingerTimed"); 4875 if (mTimedMemoryDealer == NULL) 4876 return NO_MEMORY; 4877 } 4878 4879 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4880 if (newBuffer == NULL) { 4881 newBuffer = mTimedMemoryDealer->allocate(size); 4882 if (newBuffer == NULL) 4883 return NO_MEMORY; 4884 } 4885 4886 *buffer = newBuffer; 4887 return NO_ERROR; 4888} 4889 4890// caller must hold mTimedBufferQueueLock 4891void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4892 int64_t mediaTimeNow; 4893 { 4894 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4895 if (!mMediaTimeTransformValid) 4896 return; 4897 4898 int64_t targetTimeNow; 4899 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4900 ? mCCHelper.getCommonTime(&targetTimeNow) 4901 : mCCHelper.getLocalTime(&targetTimeNow); 4902 4903 if (OK != res) 4904 return; 4905 4906 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4907 &mediaTimeNow)) { 4908 return; 4909 } 4910 } 4911 4912 size_t trimEnd; 4913 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4914 int64_t bufEnd; 4915 4916 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4917 // We have a next buffer. Just use its PTS as the PTS of the frame 4918 // following the last frame in this buffer. If the stream is sparse 4919 // (ie, there are deliberate gaps left in the stream which should be 4920 // filled with silence by the TimedAudioTrack), then this can result 4921 // in one extra buffer being left un-trimmed when it could have 4922 // been. In general, this is not typical, and we would rather 4923 // optimized away the TS calculation below for the more common case 4924 // where PTSes are contiguous. 4925 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4926 } else { 4927 // We have no next buffer. Compute the PTS of the frame following 4928 // the last frame in this buffer by computing the duration of of 4929 // this frame in media time units and adding it to the PTS of the 4930 // buffer. 4931 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4932 / mCblk->frameSize; 4933 4934 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4935 &bufEnd)) { 4936 ALOGE("Failed to convert frame count of %lld to media time" 4937 " duration" " (scale factor %d/%u) in %s", 4938 frameCount, 4939 mMediaTimeToSampleTransform.a_to_b_numer, 4940 mMediaTimeToSampleTransform.a_to_b_denom, 4941 __PRETTY_FUNCTION__); 4942 break; 4943 } 4944 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4945 } 4946 4947 if (bufEnd > mediaTimeNow) 4948 break; 4949 4950 // Is the buffer we want to use in the middle of a mix operation right 4951 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4952 // from the mixer which should be coming back shortly. 4953 if (!trimEnd && mQueueHeadInFlight) { 4954 mTrimQueueHeadOnRelease = true; 4955 } 4956 } 4957 4958 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4959 if (trimStart < trimEnd) { 4960 // Update the bookkeeping for framesReady() 4961 for (size_t i = trimStart; i < trimEnd; ++i) { 4962 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4963 } 4964 4965 // Now actually remove the buffers from the queue. 4966 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4967 } 4968} 4969 4970void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4971 const char* logTag) { 4972 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4973 "%s called (reason \"%s\"), but timed buffer queue has no" 4974 " elements to trim.", __FUNCTION__, logTag); 4975 4976 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4977 mTimedBufferQueue.removeAt(0); 4978} 4979 4980void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4981 const TimedBuffer& buf, 4982 const char* logTag) { 4983 uint32_t bufBytes = buf.buffer()->size(); 4984 uint32_t consumedAlready = buf.position(); 4985 4986 ALOG_ASSERT(consumedAlready <= bufBytes, 4987 "Bad bookkeeping while updating frames pending. Timed buffer is" 4988 " only %u bytes long, but claims to have consumed %u" 4989 " bytes. (update reason: \"%s\")", 4990 bufBytes, consumedAlready, logTag); 4991 4992 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4993 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4994 "Bad bookkeeping while updating frames pending. Should have at" 4995 " least %u queued frames, but we think we have only %u. (update" 4996 " reason: \"%s\")", 4997 bufFrames, mFramesPendingInQueue, logTag); 4998 4999 mFramesPendingInQueue -= bufFrames; 5000} 5001 5002status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5003 const sp<IMemory>& buffer, int64_t pts) { 5004 5005 { 5006 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5007 if (!mMediaTimeTransformValid) 5008 return INVALID_OPERATION; 5009 } 5010 5011 Mutex::Autolock _l(mTimedBufferQueueLock); 5012 5013 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5014 mFramesPendingInQueue += bufFrames; 5015 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5016 5017 return NO_ERROR; 5018} 5019 5020status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5021 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5022 5023 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5024 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5025 target); 5026 5027 if (!(target == TimedAudioTrack::LOCAL_TIME || 5028 target == TimedAudioTrack::COMMON_TIME)) { 5029 return BAD_VALUE; 5030 } 5031 5032 Mutex::Autolock lock(mMediaTimeTransformLock); 5033 mMediaTimeTransform = xform; 5034 mMediaTimeTransformTarget = target; 5035 mMediaTimeTransformValid = true; 5036 5037 return NO_ERROR; 5038} 5039 5040#define min(a, b) ((a) < (b) ? (a) : (b)) 5041 5042// implementation of getNextBuffer for tracks whose buffers have timestamps 5043status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5044 AudioBufferProvider::Buffer* buffer, int64_t pts) 5045{ 5046 if (pts == AudioBufferProvider::kInvalidPTS) { 5047 buffer->raw = NULL; 5048 buffer->frameCount = 0; 5049 mTimedAudioOutputOnTime = false; 5050 return INVALID_OPERATION; 5051 } 5052 5053 Mutex::Autolock _l(mTimedBufferQueueLock); 5054 5055 ALOG_ASSERT(!mQueueHeadInFlight, 5056 "getNextBuffer called without releaseBuffer!"); 5057 5058 while (true) { 5059 5060 // if we have no timed buffers, then fail 5061 if (mTimedBufferQueue.isEmpty()) { 5062 buffer->raw = NULL; 5063 buffer->frameCount = 0; 5064 return NOT_ENOUGH_DATA; 5065 } 5066 5067 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5068 5069 // calculate the PTS of the head of the timed buffer queue expressed in 5070 // local time 5071 int64_t headLocalPTS; 5072 { 5073 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5074 5075 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5076 5077 if (mMediaTimeTransform.a_to_b_denom == 0) { 5078 // the transform represents a pause, so yield silence 5079 timedYieldSilence_l(buffer->frameCount, buffer); 5080 return NO_ERROR; 5081 } 5082 5083 int64_t transformedPTS; 5084 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5085 &transformedPTS)) { 5086 // the transform failed. this shouldn't happen, but if it does 5087 // then just drop this buffer 5088 ALOGW("timedGetNextBuffer transform failed"); 5089 buffer->raw = NULL; 5090 buffer->frameCount = 0; 5091 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5092 return NO_ERROR; 5093 } 5094 5095 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5096 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5097 &headLocalPTS)) { 5098 buffer->raw = NULL; 5099 buffer->frameCount = 0; 5100 return INVALID_OPERATION; 5101 } 5102 } else { 5103 headLocalPTS = transformedPTS; 5104 } 5105 } 5106 5107 // adjust the head buffer's PTS to reflect the portion of the head buffer 5108 // that has already been consumed 5109 int64_t effectivePTS = headLocalPTS + 5110 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5111 5112 // Calculate the delta in samples between the head of the input buffer 5113 // queue and the start of the next output buffer that will be written. 5114 // If the transformation fails because of over or underflow, it means 5115 // that the sample's position in the output stream is so far out of 5116 // whack that it should just be dropped. 5117 int64_t sampleDelta; 5118 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5119 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5120 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5121 " mix"); 5122 continue; 5123 } 5124 if (!mLocalTimeToSampleTransform.doForwardTransform( 5125 (effectivePTS - pts) << 32, &sampleDelta)) { 5126 ALOGV("*** too late during sample rate transform: dropped buffer"); 5127 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5128 continue; 5129 } 5130 5131 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5132 " sampleDelta=[%d.%08x]", 5133 head.pts(), head.position(), pts, 5134 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5135 + (sampleDelta >> 32)), 5136 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5137 5138 // if the delta between the ideal placement for the next input sample and 5139 // the current output position is within this threshold, then we will 5140 // concatenate the next input samples to the previous output 5141 const int64_t kSampleContinuityThreshold = 5142 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5143 5144 // if this is the first buffer of audio that we're emitting from this track 5145 // then it should be almost exactly on time. 5146 const int64_t kSampleStartupThreshold = 1LL << 32; 5147 5148 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5149 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5150 // the next input is close enough to being on time, so concatenate it 5151 // with the last output 5152 timedYieldSamples_l(buffer); 5153 5154 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5155 head.position(), buffer->frameCount); 5156 return NO_ERROR; 5157 } 5158 5159 // Looks like our output is not on time. Reset our on timed status. 5160 // Next time we mix samples from our input queue, then should be within 5161 // the StartupThreshold. 5162 mTimedAudioOutputOnTime = false; 5163 if (sampleDelta > 0) { 5164 // the gap between the current output position and the proper start of 5165 // the next input sample is too big, so fill it with silence 5166 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5167 5168 timedYieldSilence_l(framesUntilNextInput, buffer); 5169 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5170 return NO_ERROR; 5171 } else { 5172 // the next input sample is late 5173 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5174 size_t onTimeSamplePosition = 5175 head.position() + lateFrames * mCblk->frameSize; 5176 5177 if (onTimeSamplePosition > head.buffer()->size()) { 5178 // all the remaining samples in the head are too late, so 5179 // drop it and move on 5180 ALOGV("*** too late: dropped buffer"); 5181 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5182 continue; 5183 } else { 5184 // skip over the late samples 5185 head.setPosition(onTimeSamplePosition); 5186 5187 // yield the available samples 5188 timedYieldSamples_l(buffer); 5189 5190 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5191 return NO_ERROR; 5192 } 5193 } 5194 } 5195} 5196 5197// Yield samples from the timed buffer queue head up to the given output 5198// buffer's capacity. 5199// 5200// Caller must hold mTimedBufferQueueLock 5201void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5202 AudioBufferProvider::Buffer* buffer) { 5203 5204 const TimedBuffer& head = mTimedBufferQueue[0]; 5205 5206 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5207 head.position()); 5208 5209 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5210 mCblk->frameSize); 5211 size_t framesRequested = buffer->frameCount; 5212 buffer->frameCount = min(framesLeftInHead, framesRequested); 5213 5214 mQueueHeadInFlight = true; 5215 mTimedAudioOutputOnTime = true; 5216} 5217 5218// Yield samples of silence up to the given output buffer's capacity 5219// 5220// Caller must hold mTimedBufferQueueLock 5221void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5222 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5223 5224 // lazily allocate a buffer filled with silence 5225 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5226 delete [] mTimedSilenceBuffer; 5227 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5228 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5229 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5230 } 5231 5232 buffer->raw = mTimedSilenceBuffer; 5233 size_t framesRequested = buffer->frameCount; 5234 buffer->frameCount = min(numFrames, framesRequested); 5235 5236 mTimedAudioOutputOnTime = false; 5237} 5238 5239// AudioBufferProvider interface 5240void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5241 AudioBufferProvider::Buffer* buffer) { 5242 5243 Mutex::Autolock _l(mTimedBufferQueueLock); 5244 5245 // If the buffer which was just released is part of the buffer at the head 5246 // of the queue, be sure to update the amt of the buffer which has been 5247 // consumed. If the buffer being returned is not part of the head of the 5248 // queue, its either because the buffer is part of the silence buffer, or 5249 // because the head of the timed queue was trimmed after the mixer called 5250 // getNextBuffer but before the mixer called releaseBuffer. 5251 if (buffer->raw == mTimedSilenceBuffer) { 5252 ALOG_ASSERT(!mQueueHeadInFlight, 5253 "Queue head in flight during release of silence buffer!"); 5254 goto done; 5255 } 5256 5257 ALOG_ASSERT(mQueueHeadInFlight, 5258 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5259 " head in flight."); 5260 5261 if (mTimedBufferQueue.size()) { 5262 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5263 5264 void* start = head.buffer()->pointer(); 5265 void* end = reinterpret_cast<void*>( 5266 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5267 + head.buffer()->size()); 5268 5269 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5270 "released buffer not within the head of the timed buffer" 5271 " queue; qHead = [%p, %p], released buffer = %p", 5272 start, end, buffer->raw); 5273 5274 head.setPosition(head.position() + 5275 (buffer->frameCount * mCblk->frameSize)); 5276 mQueueHeadInFlight = false; 5277 5278 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5279 "Bad bookkeeping during releaseBuffer! Should have at" 5280 " least %u queued frames, but we think we have only %u", 5281 buffer->frameCount, mFramesPendingInQueue); 5282 5283 mFramesPendingInQueue -= buffer->frameCount; 5284 5285 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5286 || mTrimQueueHeadOnRelease) { 5287 trimTimedBufferQueueHead_l("releaseBuffer"); 5288 mTrimQueueHeadOnRelease = false; 5289 } 5290 } else { 5291 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5292 " buffers in the timed buffer queue"); 5293 } 5294 5295done: 5296 buffer->raw = 0; 5297 buffer->frameCount = 0; 5298} 5299 5300size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5301 Mutex::Autolock _l(mTimedBufferQueueLock); 5302 return mFramesPendingInQueue; 5303} 5304 5305AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5306 : mPTS(0), mPosition(0) {} 5307 5308AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5309 const sp<IMemory>& buffer, int64_t pts) 5310 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5311 5312// ---------------------------------------------------------------------------- 5313 5314// RecordTrack constructor must be called with AudioFlinger::mLock held 5315AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5316 RecordThread *thread, 5317 const sp<Client>& client, 5318 uint32_t sampleRate, 5319 audio_format_t format, 5320 audio_channel_mask_t channelMask, 5321 int frameCount, 5322 int sessionId) 5323 : TrackBase(thread, client, sampleRate, format, 5324 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5325 mOverflow(false) 5326{ 5327 if (mCblk != NULL) { 5328 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5329 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5330 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5331 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5332 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5333 } else { 5334 mCblk->frameSize = sizeof(int8_t); 5335 } 5336 } 5337} 5338 5339AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5340{ 5341 ALOGV("%s", __func__); 5342} 5343 5344// AudioBufferProvider interface 5345status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5346{ 5347 audio_track_cblk_t* cblk = this->cblk(); 5348 uint32_t framesAvail; 5349 uint32_t framesReq = buffer->frameCount; 5350 5351 // Check if last stepServer failed, try to step now 5352 if (mStepServerFailed) { 5353 if (!step()) goto getNextBuffer_exit; 5354 ALOGV("stepServer recovered"); 5355 mStepServerFailed = false; 5356 } 5357 5358 framesAvail = cblk->framesAvailable_l(); 5359 5360 if (CC_LIKELY(framesAvail)) { 5361 uint32_t s = cblk->server; 5362 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5363 5364 if (framesReq > framesAvail) { 5365 framesReq = framesAvail; 5366 } 5367 if (framesReq > bufferEnd - s) { 5368 framesReq = bufferEnd - s; 5369 } 5370 5371 buffer->raw = getBuffer(s, framesReq); 5372 buffer->frameCount = framesReq; 5373 return NO_ERROR; 5374 } 5375 5376getNextBuffer_exit: 5377 buffer->raw = NULL; 5378 buffer->frameCount = 0; 5379 return NOT_ENOUGH_DATA; 5380} 5381 5382status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5383 int triggerSession) 5384{ 5385 sp<ThreadBase> thread = mThread.promote(); 5386 if (thread != 0) { 5387 RecordThread *recordThread = (RecordThread *)thread.get(); 5388 return recordThread->start(this, event, triggerSession); 5389 } else { 5390 return BAD_VALUE; 5391 } 5392} 5393 5394void AudioFlinger::RecordThread::RecordTrack::stop() 5395{ 5396 sp<ThreadBase> thread = mThread.promote(); 5397 if (thread != 0) { 5398 RecordThread *recordThread = (RecordThread *)thread.get(); 5399 recordThread->mLock.lock(); 5400 bool doStop = recordThread->stop_l(this); 5401 if (doStop) { 5402 TrackBase::reset(); 5403 // Force overrun condition to avoid false overrun callback until first data is 5404 // read from buffer 5405 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5406 } 5407 recordThread->mLock.unlock(); 5408 if (doStop) { 5409 AudioSystem::stopInput(recordThread->id()); 5410 } 5411 } 5412} 5413 5414/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5415{ 5416 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5417} 5418 5419void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5420{ 5421 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5422 (mClient == 0) ? getpid_cached : mClient->pid(), 5423 mFormat, 5424 mChannelMask, 5425 mSessionId, 5426 mFrameCount, 5427 mState, 5428 mCblk->sampleRate, 5429 mCblk->server, 5430 mCblk->user); 5431} 5432 5433 5434// ---------------------------------------------------------------------------- 5435 5436AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5437 PlaybackThread *playbackThread, 5438 DuplicatingThread *sourceThread, 5439 uint32_t sampleRate, 5440 audio_format_t format, 5441 audio_channel_mask_t channelMask, 5442 int frameCount) 5443 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5444 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5445 mActive(false), mSourceThread(sourceThread) 5446{ 5447 5448 if (mCblk != NULL) { 5449 mCblk->flags |= CBLK_DIRECTION_OUT; 5450 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5451 mOutBuffer.frameCount = 0; 5452 playbackThread->mTracks.add(this); 5453 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5454 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5455 mCblk, mBuffer, mCblk->buffers, 5456 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5457 } else { 5458 ALOGW("Error creating output track on thread %p", playbackThread); 5459 } 5460} 5461 5462AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5463{ 5464 clearBufferQueue(); 5465} 5466 5467status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5468 int triggerSession) 5469{ 5470 status_t status = Track::start(event, triggerSession); 5471 if (status != NO_ERROR) { 5472 return status; 5473 } 5474 5475 mActive = true; 5476 mRetryCount = 127; 5477 return status; 5478} 5479 5480void AudioFlinger::PlaybackThread::OutputTrack::stop() 5481{ 5482 Track::stop(); 5483 clearBufferQueue(); 5484 mOutBuffer.frameCount = 0; 5485 mActive = false; 5486} 5487 5488bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5489{ 5490 Buffer *pInBuffer; 5491 Buffer inBuffer; 5492 uint32_t channelCount = mChannelCount; 5493 bool outputBufferFull = false; 5494 inBuffer.frameCount = frames; 5495 inBuffer.i16 = data; 5496 5497 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5498 5499 if (!mActive && frames != 0) { 5500 start(); 5501 sp<ThreadBase> thread = mThread.promote(); 5502 if (thread != 0) { 5503 MixerThread *mixerThread = (MixerThread *)thread.get(); 5504 if (mCblk->frameCount > frames){ 5505 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5506 uint32_t startFrames = (mCblk->frameCount - frames); 5507 pInBuffer = new Buffer; 5508 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5509 pInBuffer->frameCount = startFrames; 5510 pInBuffer->i16 = pInBuffer->mBuffer; 5511 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5512 mBufferQueue.add(pInBuffer); 5513 } else { 5514 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5515 } 5516 } 5517 } 5518 } 5519 5520 while (waitTimeLeftMs) { 5521 // First write pending buffers, then new data 5522 if (mBufferQueue.size()) { 5523 pInBuffer = mBufferQueue.itemAt(0); 5524 } else { 5525 pInBuffer = &inBuffer; 5526 } 5527 5528 if (pInBuffer->frameCount == 0) { 5529 break; 5530 } 5531 5532 if (mOutBuffer.frameCount == 0) { 5533 mOutBuffer.frameCount = pInBuffer->frameCount; 5534 nsecs_t startTime = systemTime(); 5535 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5536 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5537 outputBufferFull = true; 5538 break; 5539 } 5540 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5541 if (waitTimeLeftMs >= waitTimeMs) { 5542 waitTimeLeftMs -= waitTimeMs; 5543 } else { 5544 waitTimeLeftMs = 0; 5545 } 5546 } 5547 5548 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5549 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5550 mCblk->stepUser(outFrames); 5551 pInBuffer->frameCount -= outFrames; 5552 pInBuffer->i16 += outFrames * channelCount; 5553 mOutBuffer.frameCount -= outFrames; 5554 mOutBuffer.i16 += outFrames * channelCount; 5555 5556 if (pInBuffer->frameCount == 0) { 5557 if (mBufferQueue.size()) { 5558 mBufferQueue.removeAt(0); 5559 delete [] pInBuffer->mBuffer; 5560 delete pInBuffer; 5561 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5562 } else { 5563 break; 5564 } 5565 } 5566 } 5567 5568 // If we could not write all frames, allocate a buffer and queue it for next time. 5569 if (inBuffer.frameCount) { 5570 sp<ThreadBase> thread = mThread.promote(); 5571 if (thread != 0 && !thread->standby()) { 5572 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5573 pInBuffer = new Buffer; 5574 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5575 pInBuffer->frameCount = inBuffer.frameCount; 5576 pInBuffer->i16 = pInBuffer->mBuffer; 5577 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5578 mBufferQueue.add(pInBuffer); 5579 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5580 } else { 5581 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5582 } 5583 } 5584 } 5585 5586 // Calling write() with a 0 length buffer, means that no more data will be written: 5587 // If no more buffers are pending, fill output track buffer to make sure it is started 5588 // by output mixer. 5589 if (frames == 0 && mBufferQueue.size() == 0) { 5590 if (mCblk->user < mCblk->frameCount) { 5591 frames = mCblk->frameCount - mCblk->user; 5592 pInBuffer = new Buffer; 5593 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5594 pInBuffer->frameCount = frames; 5595 pInBuffer->i16 = pInBuffer->mBuffer; 5596 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5597 mBufferQueue.add(pInBuffer); 5598 } else if (mActive) { 5599 stop(); 5600 } 5601 } 5602 5603 return outputBufferFull; 5604} 5605 5606status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5607{ 5608 int active; 5609 status_t result; 5610 audio_track_cblk_t* cblk = mCblk; 5611 uint32_t framesReq = buffer->frameCount; 5612 5613// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5614 buffer->frameCount = 0; 5615 5616 uint32_t framesAvail = cblk->framesAvailable(); 5617 5618 5619 if (framesAvail == 0) { 5620 Mutex::Autolock _l(cblk->lock); 5621 goto start_loop_here; 5622 while (framesAvail == 0) { 5623 active = mActive; 5624 if (CC_UNLIKELY(!active)) { 5625 ALOGV("Not active and NO_MORE_BUFFERS"); 5626 return NO_MORE_BUFFERS; 5627 } 5628 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5629 if (result != NO_ERROR) { 5630 return NO_MORE_BUFFERS; 5631 } 5632 // read the server count again 5633 start_loop_here: 5634 framesAvail = cblk->framesAvailable_l(); 5635 } 5636 } 5637 5638// if (framesAvail < framesReq) { 5639// return NO_MORE_BUFFERS; 5640// } 5641 5642 if (framesReq > framesAvail) { 5643 framesReq = framesAvail; 5644 } 5645 5646 uint32_t u = cblk->user; 5647 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5648 5649 if (framesReq > bufferEnd - u) { 5650 framesReq = bufferEnd - u; 5651 } 5652 5653 buffer->frameCount = framesReq; 5654 buffer->raw = (void *)cblk->buffer(u); 5655 return NO_ERROR; 5656} 5657 5658 5659void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5660{ 5661 size_t size = mBufferQueue.size(); 5662 5663 for (size_t i = 0; i < size; i++) { 5664 Buffer *pBuffer = mBufferQueue.itemAt(i); 5665 delete [] pBuffer->mBuffer; 5666 delete pBuffer; 5667 } 5668 mBufferQueue.clear(); 5669} 5670 5671// ---------------------------------------------------------------------------- 5672 5673AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5674 : RefBase(), 5675 mAudioFlinger(audioFlinger), 5676 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5677 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5678 mPid(pid), 5679 mTimedTrackCount(0) 5680{ 5681 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5682} 5683 5684// Client destructor must be called with AudioFlinger::mLock held 5685AudioFlinger::Client::~Client() 5686{ 5687 mAudioFlinger->removeClient_l(mPid); 5688} 5689 5690sp<MemoryDealer> AudioFlinger::Client::heap() const 5691{ 5692 return mMemoryDealer; 5693} 5694 5695// Reserve one of the limited slots for a timed audio track associated 5696// with this client 5697bool AudioFlinger::Client::reserveTimedTrack() 5698{ 5699 const int kMaxTimedTracksPerClient = 4; 5700 5701 Mutex::Autolock _l(mTimedTrackLock); 5702 5703 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5704 ALOGW("can not create timed track - pid %d has exceeded the limit", 5705 mPid); 5706 return false; 5707 } 5708 5709 mTimedTrackCount++; 5710 return true; 5711} 5712 5713// Release a slot for a timed audio track 5714void AudioFlinger::Client::releaseTimedTrack() 5715{ 5716 Mutex::Autolock _l(mTimedTrackLock); 5717 mTimedTrackCount--; 5718} 5719 5720// ---------------------------------------------------------------------------- 5721 5722AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5723 const sp<IAudioFlingerClient>& client, 5724 pid_t pid) 5725 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5726{ 5727} 5728 5729AudioFlinger::NotificationClient::~NotificationClient() 5730{ 5731} 5732 5733void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5734{ 5735 sp<NotificationClient> keep(this); 5736 mAudioFlinger->removeNotificationClient(mPid); 5737} 5738 5739// ---------------------------------------------------------------------------- 5740 5741AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5742 : BnAudioTrack(), 5743 mTrack(track) 5744{ 5745} 5746 5747AudioFlinger::TrackHandle::~TrackHandle() { 5748 // just stop the track on deletion, associated resources 5749 // will be freed from the main thread once all pending buffers have 5750 // been played. Unless it's not in the active track list, in which 5751 // case we free everything now... 5752 mTrack->destroy(); 5753} 5754 5755sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5756 return mTrack->getCblk(); 5757} 5758 5759status_t AudioFlinger::TrackHandle::start() { 5760 return mTrack->start(); 5761} 5762 5763void AudioFlinger::TrackHandle::stop() { 5764 mTrack->stop(); 5765} 5766 5767void AudioFlinger::TrackHandle::flush() { 5768 mTrack->flush(); 5769} 5770 5771void AudioFlinger::TrackHandle::mute(bool e) { 5772 mTrack->mute(e); 5773} 5774 5775void AudioFlinger::TrackHandle::pause() { 5776 mTrack->pause(); 5777} 5778 5779status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5780{ 5781 return mTrack->attachAuxEffect(EffectId); 5782} 5783 5784status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5785 sp<IMemory>* buffer) { 5786 if (!mTrack->isTimedTrack()) 5787 return INVALID_OPERATION; 5788 5789 PlaybackThread::TimedTrack* tt = 5790 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5791 return tt->allocateTimedBuffer(size, buffer); 5792} 5793 5794status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5795 int64_t pts) { 5796 if (!mTrack->isTimedTrack()) 5797 return INVALID_OPERATION; 5798 5799 PlaybackThread::TimedTrack* tt = 5800 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5801 return tt->queueTimedBuffer(buffer, pts); 5802} 5803 5804status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5805 const LinearTransform& xform, int target) { 5806 5807 if (!mTrack->isTimedTrack()) 5808 return INVALID_OPERATION; 5809 5810 PlaybackThread::TimedTrack* tt = 5811 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5812 return tt->setMediaTimeTransform( 5813 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5814} 5815 5816status_t AudioFlinger::TrackHandle::onTransact( 5817 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5818{ 5819 return BnAudioTrack::onTransact(code, data, reply, flags); 5820} 5821 5822// ---------------------------------------------------------------------------- 5823 5824sp<IAudioRecord> AudioFlinger::openRecord( 5825 pid_t pid, 5826 audio_io_handle_t input, 5827 uint32_t sampleRate, 5828 audio_format_t format, 5829 audio_channel_mask_t channelMask, 5830 int frameCount, 5831 IAudioFlinger::track_flags_t flags, 5832 pid_t tid, 5833 int *sessionId, 5834 status_t *status) 5835{ 5836 sp<RecordThread::RecordTrack> recordTrack; 5837 sp<RecordHandle> recordHandle; 5838 sp<Client> client; 5839 status_t lStatus; 5840 RecordThread *thread; 5841 size_t inFrameCount; 5842 int lSessionId; 5843 5844 // check calling permissions 5845 if (!recordingAllowed()) { 5846 lStatus = PERMISSION_DENIED; 5847 goto Exit; 5848 } 5849 5850 // add client to list 5851 { // scope for mLock 5852 Mutex::Autolock _l(mLock); 5853 thread = checkRecordThread_l(input); 5854 if (thread == NULL) { 5855 lStatus = BAD_VALUE; 5856 goto Exit; 5857 } 5858 5859 client = registerPid_l(pid); 5860 5861 // If no audio session id is provided, create one here 5862 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5863 lSessionId = *sessionId; 5864 } else { 5865 lSessionId = nextUniqueId(); 5866 if (sessionId != NULL) { 5867 *sessionId = lSessionId; 5868 } 5869 } 5870 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5871 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5872 frameCount, lSessionId, flags, tid, &lStatus); 5873 } 5874 if (lStatus != NO_ERROR) { 5875 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5876 // destructor is called by the TrackBase destructor with mLock held 5877 client.clear(); 5878 recordTrack.clear(); 5879 goto Exit; 5880 } 5881 5882 // return to handle to client 5883 recordHandle = new RecordHandle(recordTrack); 5884 lStatus = NO_ERROR; 5885 5886Exit: 5887 if (status) { 5888 *status = lStatus; 5889 } 5890 return recordHandle; 5891} 5892 5893// ---------------------------------------------------------------------------- 5894 5895AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5896 : BnAudioRecord(), 5897 mRecordTrack(recordTrack) 5898{ 5899} 5900 5901AudioFlinger::RecordHandle::~RecordHandle() { 5902 stop_nonvirtual(); 5903 mRecordTrack->destroy(); 5904} 5905 5906sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5907 return mRecordTrack->getCblk(); 5908} 5909 5910status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5911 ALOGV("RecordHandle::start()"); 5912 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5913} 5914 5915void AudioFlinger::RecordHandle::stop() { 5916 stop_nonvirtual(); 5917} 5918 5919void AudioFlinger::RecordHandle::stop_nonvirtual() { 5920 ALOGV("RecordHandle::stop()"); 5921 mRecordTrack->stop(); 5922} 5923 5924status_t AudioFlinger::RecordHandle::onTransact( 5925 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5926{ 5927 return BnAudioRecord::onTransact(code, data, reply, flags); 5928} 5929 5930// ---------------------------------------------------------------------------- 5931 5932AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5933 AudioStreamIn *input, 5934 uint32_t sampleRate, 5935 audio_channel_mask_t channelMask, 5936 audio_io_handle_t id, 5937 audio_devices_t device) : 5938 ThreadBase(audioFlinger, id, device, RECORD), 5939 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5940 // mRsmpInIndex and mInputBytes set by readInputParameters() 5941 mReqChannelCount(popcount(channelMask)), 5942 mReqSampleRate(sampleRate) 5943 // mBytesRead is only meaningful while active, and so is cleared in start() 5944 // (but might be better to also clear here for dump?) 5945{ 5946 snprintf(mName, kNameLength, "AudioIn_%X", id); 5947 5948 readInputParameters(); 5949} 5950 5951 5952AudioFlinger::RecordThread::~RecordThread() 5953{ 5954 delete[] mRsmpInBuffer; 5955 delete mResampler; 5956 delete[] mRsmpOutBuffer; 5957} 5958 5959void AudioFlinger::RecordThread::onFirstRef() 5960{ 5961 run(mName, PRIORITY_URGENT_AUDIO); 5962} 5963 5964status_t AudioFlinger::RecordThread::readyToRun() 5965{ 5966 status_t status = initCheck(); 5967 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5968 return status; 5969} 5970 5971bool AudioFlinger::RecordThread::threadLoop() 5972{ 5973 AudioBufferProvider::Buffer buffer; 5974 sp<RecordTrack> activeTrack; 5975 Vector< sp<EffectChain> > effectChains; 5976 5977 nsecs_t lastWarning = 0; 5978 5979 inputStandBy(); 5980 acquireWakeLock(); 5981 5982 // start recording 5983 while (!exitPending()) { 5984 5985 processConfigEvents(); 5986 5987 { // scope for mLock 5988 Mutex::Autolock _l(mLock); 5989 checkForNewParameters_l(); 5990 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5991 standby(); 5992 5993 if (exitPending()) break; 5994 5995 releaseWakeLock_l(); 5996 ALOGV("RecordThread: loop stopping"); 5997 // go to sleep 5998 mWaitWorkCV.wait(mLock); 5999 ALOGV("RecordThread: loop starting"); 6000 acquireWakeLock_l(); 6001 continue; 6002 } 6003 if (mActiveTrack != 0) { 6004 if (mActiveTrack->mState == TrackBase::PAUSING) { 6005 standby(); 6006 mActiveTrack.clear(); 6007 mStartStopCond.broadcast(); 6008 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6009 if (mReqChannelCount != mActiveTrack->channelCount()) { 6010 mActiveTrack.clear(); 6011 mStartStopCond.broadcast(); 6012 } else if (mBytesRead != 0) { 6013 // record start succeeds only if first read from audio input 6014 // succeeds 6015 if (mBytesRead > 0) { 6016 mActiveTrack->mState = TrackBase::ACTIVE; 6017 } else { 6018 mActiveTrack.clear(); 6019 } 6020 mStartStopCond.broadcast(); 6021 } 6022 mStandby = false; 6023 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6024 removeTrack_l(mActiveTrack); 6025 mActiveTrack.clear(); 6026 } 6027 } 6028 lockEffectChains_l(effectChains); 6029 } 6030 6031 if (mActiveTrack != 0) { 6032 if (mActiveTrack->mState != TrackBase::ACTIVE && 6033 mActiveTrack->mState != TrackBase::RESUMING) { 6034 unlockEffectChains(effectChains); 6035 usleep(kRecordThreadSleepUs); 6036 continue; 6037 } 6038 for (size_t i = 0; i < effectChains.size(); i ++) { 6039 effectChains[i]->process_l(); 6040 } 6041 6042 buffer.frameCount = mFrameCount; 6043 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6044 size_t framesOut = buffer.frameCount; 6045 if (mResampler == NULL) { 6046 // no resampling 6047 while (framesOut) { 6048 size_t framesIn = mFrameCount - mRsmpInIndex; 6049 if (framesIn) { 6050 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6051 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6052 if (framesIn > framesOut) 6053 framesIn = framesOut; 6054 mRsmpInIndex += framesIn; 6055 framesOut -= framesIn; 6056 if ((int)mChannelCount == mReqChannelCount || 6057 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6058 memcpy(dst, src, framesIn * mFrameSize); 6059 } else { 6060 if (mChannelCount == 1) { 6061 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6062 (int16_t *)src, framesIn); 6063 } else { 6064 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6065 (int16_t *)src, framesIn); 6066 } 6067 } 6068 } 6069 if (framesOut && mFrameCount == mRsmpInIndex) { 6070 if (framesOut == mFrameCount && 6071 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6072 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6073 framesOut = 0; 6074 } else { 6075 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6076 mRsmpInIndex = 0; 6077 } 6078 if (mBytesRead < 0) { 6079 ALOGE("Error reading audio input"); 6080 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6081 // Force input into standby so that it tries to 6082 // recover at next read attempt 6083 inputStandBy(); 6084 usleep(kRecordThreadSleepUs); 6085 } 6086 mRsmpInIndex = mFrameCount; 6087 framesOut = 0; 6088 buffer.frameCount = 0; 6089 } 6090 } 6091 } 6092 } else { 6093 // resampling 6094 6095 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6096 // alter output frame count as if we were expecting stereo samples 6097 if (mChannelCount == 1 && mReqChannelCount == 1) { 6098 framesOut >>= 1; 6099 } 6100 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6101 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6102 // are 32 bit aligned which should be always true. 6103 if (mChannelCount == 2 && mReqChannelCount == 1) { 6104 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6105 // the resampler always outputs stereo samples: do post stereo to mono conversion 6106 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6107 framesOut); 6108 } else { 6109 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6110 } 6111 6112 } 6113 if (mFramestoDrop == 0) { 6114 mActiveTrack->releaseBuffer(&buffer); 6115 } else { 6116 if (mFramestoDrop > 0) { 6117 mFramestoDrop -= buffer.frameCount; 6118 if (mFramestoDrop <= 0) { 6119 clearSyncStartEvent(); 6120 } 6121 } else { 6122 mFramestoDrop += buffer.frameCount; 6123 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6124 mSyncStartEvent->isCancelled()) { 6125 ALOGW("Synced record %s, session %d, trigger session %d", 6126 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6127 mActiveTrack->sessionId(), 6128 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6129 clearSyncStartEvent(); 6130 } 6131 } 6132 } 6133 mActiveTrack->clearOverflow(); 6134 } 6135 // client isn't retrieving buffers fast enough 6136 else { 6137 if (!mActiveTrack->setOverflow()) { 6138 nsecs_t now = systemTime(); 6139 if ((now - lastWarning) > kWarningThrottleNs) { 6140 ALOGW("RecordThread: buffer overflow"); 6141 lastWarning = now; 6142 } 6143 } 6144 // Release the processor for a while before asking for a new buffer. 6145 // This will give the application more chance to read from the buffer and 6146 // clear the overflow. 6147 usleep(kRecordThreadSleepUs); 6148 } 6149 } 6150 // enable changes in effect chain 6151 unlockEffectChains(effectChains); 6152 effectChains.clear(); 6153 } 6154 6155 standby(); 6156 6157 { 6158 Mutex::Autolock _l(mLock); 6159 mActiveTrack.clear(); 6160 mStartStopCond.broadcast(); 6161 } 6162 6163 releaseWakeLock(); 6164 6165 ALOGV("RecordThread %p exiting", this); 6166 return false; 6167} 6168 6169void AudioFlinger::RecordThread::standby() 6170{ 6171 if (!mStandby) { 6172 inputStandBy(); 6173 mStandby = true; 6174 } 6175} 6176 6177void AudioFlinger::RecordThread::inputStandBy() 6178{ 6179 mInput->stream->common.standby(&mInput->stream->common); 6180} 6181 6182sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6183 const sp<AudioFlinger::Client>& client, 6184 uint32_t sampleRate, 6185 audio_format_t format, 6186 audio_channel_mask_t channelMask, 6187 int frameCount, 6188 int sessionId, 6189 IAudioFlinger::track_flags_t flags, 6190 pid_t tid, 6191 status_t *status) 6192{ 6193 sp<RecordTrack> track; 6194 status_t lStatus; 6195 6196 lStatus = initCheck(); 6197 if (lStatus != NO_ERROR) { 6198 ALOGE("Audio driver not initialized."); 6199 goto Exit; 6200 } 6201 6202 // FIXME use flags and tid similar to createTrack_l() 6203 6204 { // scope for mLock 6205 Mutex::Autolock _l(mLock); 6206 6207 track = new RecordTrack(this, client, sampleRate, 6208 format, channelMask, frameCount, sessionId); 6209 6210 if (track->getCblk() == 0) { 6211 lStatus = NO_MEMORY; 6212 goto Exit; 6213 } 6214 mTracks.add(track); 6215 6216 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6217 bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) && 6218 mAudioFlinger->btNrecIsOff(); 6219 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6220 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6221 } 6222 lStatus = NO_ERROR; 6223 6224Exit: 6225 if (status) { 6226 *status = lStatus; 6227 } 6228 return track; 6229} 6230 6231status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6232 AudioSystem::sync_event_t event, 6233 int triggerSession) 6234{ 6235 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6236 sp<ThreadBase> strongMe = this; 6237 status_t status = NO_ERROR; 6238 6239 if (event == AudioSystem::SYNC_EVENT_NONE) { 6240 clearSyncStartEvent(); 6241 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6242 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6243 triggerSession, 6244 recordTrack->sessionId(), 6245 syncStartEventCallback, 6246 this); 6247 // Sync event can be cancelled by the trigger session if the track is not in a 6248 // compatible state in which case we start record immediately 6249 if (mSyncStartEvent->isCancelled()) { 6250 clearSyncStartEvent(); 6251 } else { 6252 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6253 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6254 } 6255 } 6256 6257 { 6258 AutoMutex lock(mLock); 6259 if (mActiveTrack != 0) { 6260 if (recordTrack != mActiveTrack.get()) { 6261 status = -EBUSY; 6262 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6263 mActiveTrack->mState = TrackBase::ACTIVE; 6264 } 6265 return status; 6266 } 6267 6268 recordTrack->mState = TrackBase::IDLE; 6269 mActiveTrack = recordTrack; 6270 mLock.unlock(); 6271 status_t status = AudioSystem::startInput(mId); 6272 mLock.lock(); 6273 if (status != NO_ERROR) { 6274 mActiveTrack.clear(); 6275 clearSyncStartEvent(); 6276 return status; 6277 } 6278 mRsmpInIndex = mFrameCount; 6279 mBytesRead = 0; 6280 if (mResampler != NULL) { 6281 mResampler->reset(); 6282 } 6283 mActiveTrack->mState = TrackBase::RESUMING; 6284 // signal thread to start 6285 ALOGV("Signal record thread"); 6286 mWaitWorkCV.signal(); 6287 // do not wait for mStartStopCond if exiting 6288 if (exitPending()) { 6289 mActiveTrack.clear(); 6290 status = INVALID_OPERATION; 6291 goto startError; 6292 } 6293 mStartStopCond.wait(mLock); 6294 if (mActiveTrack == 0) { 6295 ALOGV("Record failed to start"); 6296 status = BAD_VALUE; 6297 goto startError; 6298 } 6299 ALOGV("Record started OK"); 6300 return status; 6301 } 6302startError: 6303 AudioSystem::stopInput(mId); 6304 clearSyncStartEvent(); 6305 return status; 6306} 6307 6308void AudioFlinger::RecordThread::clearSyncStartEvent() 6309{ 6310 if (mSyncStartEvent != 0) { 6311 mSyncStartEvent->cancel(); 6312 } 6313 mSyncStartEvent.clear(); 6314 mFramestoDrop = 0; 6315} 6316 6317void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6318{ 6319 sp<SyncEvent> strongEvent = event.promote(); 6320 6321 if (strongEvent != 0) { 6322 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6323 me->handleSyncStartEvent(strongEvent); 6324 } 6325} 6326 6327void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6328{ 6329 if (event == mSyncStartEvent) { 6330 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6331 // from audio HAL 6332 mFramestoDrop = mFrameCount * 2; 6333 } 6334} 6335 6336bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6337 ALOGV("RecordThread::stop"); 6338 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6339 return false; 6340 } 6341 recordTrack->mState = TrackBase::PAUSING; 6342 // do not wait for mStartStopCond if exiting 6343 if (exitPending()) { 6344 return true; 6345 } 6346 mStartStopCond.wait(mLock); 6347 // if we have been restarted, recordTrack == mActiveTrack.get() here 6348 if (exitPending() || recordTrack != mActiveTrack.get()) { 6349 ALOGV("Record stopped OK"); 6350 return true; 6351 } 6352 return false; 6353} 6354 6355bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6356{ 6357 return false; 6358} 6359 6360status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6361{ 6362 if (!isValidSyncEvent(event)) { 6363 return BAD_VALUE; 6364 } 6365 6366 int eventSession = event->triggerSession(); 6367 status_t ret = NAME_NOT_FOUND; 6368 6369 Mutex::Autolock _l(mLock); 6370 6371 for (size_t i = 0; i < mTracks.size(); i++) { 6372 sp<RecordTrack> track = mTracks[i]; 6373 if (eventSession == track->sessionId()) { 6374 (void) track->setSyncEvent(event); 6375 ret = NO_ERROR; 6376 } 6377 } 6378 return ret; 6379} 6380 6381void AudioFlinger::RecordThread::RecordTrack::destroy() 6382{ 6383 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6384 sp<RecordTrack> keep(this); 6385 { 6386 sp<ThreadBase> thread = mThread.promote(); 6387 if (thread != 0) { 6388 if (mState == ACTIVE || mState == RESUMING) { 6389 AudioSystem::stopInput(thread->id()); 6390 } 6391 AudioSystem::releaseInput(thread->id()); 6392 Mutex::Autolock _l(thread->mLock); 6393 RecordThread *recordThread = (RecordThread *) thread.get(); 6394 recordThread->destroyTrack_l(this); 6395 } 6396 } 6397} 6398 6399// destroyTrack_l() must be called with ThreadBase::mLock held 6400void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6401{ 6402 track->mState = TrackBase::TERMINATED; 6403 // active tracks are removed by threadLoop() 6404 if (mActiveTrack != track) { 6405 removeTrack_l(track); 6406 } 6407} 6408 6409void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6410{ 6411 mTracks.remove(track); 6412 // need anything related to effects here? 6413} 6414 6415void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6416{ 6417 dumpInternals(fd, args); 6418 dumpTracks(fd, args); 6419 dumpEffectChains(fd, args); 6420} 6421 6422void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6423{ 6424 const size_t SIZE = 256; 6425 char buffer[SIZE]; 6426 String8 result; 6427 6428 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6429 result.append(buffer); 6430 6431 if (mActiveTrack != 0) { 6432 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6433 result.append(buffer); 6434 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6435 result.append(buffer); 6436 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6437 result.append(buffer); 6438 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6439 result.append(buffer); 6440 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6441 result.append(buffer); 6442 } else { 6443 result.append("No active record client\n"); 6444 } 6445 6446 write(fd, result.string(), result.size()); 6447 6448 dumpBase(fd, args); 6449} 6450 6451void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6452{ 6453 const size_t SIZE = 256; 6454 char buffer[SIZE]; 6455 String8 result; 6456 6457 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6458 result.append(buffer); 6459 RecordTrack::appendDumpHeader(result); 6460 for (size_t i = 0; i < mTracks.size(); ++i) { 6461 sp<RecordTrack> track = mTracks[i]; 6462 if (track != 0) { 6463 track->dump(buffer, SIZE); 6464 result.append(buffer); 6465 } 6466 } 6467 6468 if (mActiveTrack != 0) { 6469 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6470 result.append(buffer); 6471 RecordTrack::appendDumpHeader(result); 6472 mActiveTrack->dump(buffer, SIZE); 6473 result.append(buffer); 6474 6475 } 6476 write(fd, result.string(), result.size()); 6477} 6478 6479// AudioBufferProvider interface 6480status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6481{ 6482 size_t framesReq = buffer->frameCount; 6483 size_t framesReady = mFrameCount - mRsmpInIndex; 6484 int channelCount; 6485 6486 if (framesReady == 0) { 6487 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6488 if (mBytesRead < 0) { 6489 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6490 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6491 // Force input into standby so that it tries to 6492 // recover at next read attempt 6493 inputStandBy(); 6494 usleep(kRecordThreadSleepUs); 6495 } 6496 buffer->raw = NULL; 6497 buffer->frameCount = 0; 6498 return NOT_ENOUGH_DATA; 6499 } 6500 mRsmpInIndex = 0; 6501 framesReady = mFrameCount; 6502 } 6503 6504 if (framesReq > framesReady) { 6505 framesReq = framesReady; 6506 } 6507 6508 if (mChannelCount == 1 && mReqChannelCount == 2) { 6509 channelCount = 1; 6510 } else { 6511 channelCount = 2; 6512 } 6513 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6514 buffer->frameCount = framesReq; 6515 return NO_ERROR; 6516} 6517 6518// AudioBufferProvider interface 6519void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6520{ 6521 mRsmpInIndex += buffer->frameCount; 6522 buffer->frameCount = 0; 6523} 6524 6525bool AudioFlinger::RecordThread::checkForNewParameters_l() 6526{ 6527 bool reconfig = false; 6528 6529 while (!mNewParameters.isEmpty()) { 6530 status_t status = NO_ERROR; 6531 String8 keyValuePair = mNewParameters[0]; 6532 AudioParameter param = AudioParameter(keyValuePair); 6533 int value; 6534 audio_format_t reqFormat = mFormat; 6535 int reqSamplingRate = mReqSampleRate; 6536 int reqChannelCount = mReqChannelCount; 6537 6538 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6539 reqSamplingRate = value; 6540 reconfig = true; 6541 } 6542 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6543 reqFormat = (audio_format_t) value; 6544 reconfig = true; 6545 } 6546 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6547 reqChannelCount = popcount(value); 6548 reconfig = true; 6549 } 6550 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6551 // do not accept frame count changes if tracks are open as the track buffer 6552 // size depends on frame count and correct behavior would not be guaranteed 6553 // if frame count is changed after track creation 6554 if (mActiveTrack != 0) { 6555 status = INVALID_OPERATION; 6556 } else { 6557 reconfig = true; 6558 } 6559 } 6560 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6561 // forward device change to effects that have requested to be 6562 // aware of attached audio device. 6563 for (size_t i = 0; i < mEffectChains.size(); i++) { 6564 mEffectChains[i]->setDevice_l(value); 6565 } 6566 // store input device and output device but do not forward output device to audio HAL. 6567 // Note that status is ignored by the caller for output device 6568 // (see AudioFlinger::setParameters() 6569 audio_devices_t newDevice = mDevice; 6570 if (value & AUDIO_DEVICE_OUT_ALL) { 6571 newDevice &= ~(value & AUDIO_DEVICE_OUT_ALL); 6572 status = BAD_VALUE; 6573 } else { 6574 newDevice &= ~(value & AUDIO_DEVICE_IN_ALL); 6575 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6576 if (mTracks.size() > 0) { 6577 bool suspend = audio_is_bluetooth_sco_device( 6578 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6579 for (size_t i = 0; i < mTracks.size(); i++) { 6580 sp<RecordTrack> track = mTracks[i]; 6581 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6582 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6583 } 6584 } 6585 } 6586 newDevice |= value; 6587 mDevice = newDevice; // since mDevice is read by other threads, only write to it once 6588 } 6589 if (status == NO_ERROR) { 6590 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6591 if (status == INVALID_OPERATION) { 6592 inputStandBy(); 6593 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6594 keyValuePair.string()); 6595 } 6596 if (reconfig) { 6597 if (status == BAD_VALUE && 6598 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6599 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6600 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6601 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6602 (reqChannelCount <= FCC_2)) { 6603 status = NO_ERROR; 6604 } 6605 if (status == NO_ERROR) { 6606 readInputParameters(); 6607 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6608 } 6609 } 6610 } 6611 6612 mNewParameters.removeAt(0); 6613 6614 mParamStatus = status; 6615 mParamCond.signal(); 6616 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6617 // already timed out waiting for the status and will never signal the condition. 6618 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6619 } 6620 return reconfig; 6621} 6622 6623String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6624{ 6625 char *s; 6626 String8 out_s8 = String8(); 6627 6628 Mutex::Autolock _l(mLock); 6629 if (initCheck() != NO_ERROR) { 6630 return out_s8; 6631 } 6632 6633 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6634 out_s8 = String8(s); 6635 free(s); 6636 return out_s8; 6637} 6638 6639void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6640 AudioSystem::OutputDescriptor desc; 6641 void *param2 = NULL; 6642 6643 switch (event) { 6644 case AudioSystem::INPUT_OPENED: 6645 case AudioSystem::INPUT_CONFIG_CHANGED: 6646 desc.channels = mChannelMask; 6647 desc.samplingRate = mSampleRate; 6648 desc.format = mFormat; 6649 desc.frameCount = mFrameCount; 6650 desc.latency = 0; 6651 param2 = &desc; 6652 break; 6653 6654 case AudioSystem::INPUT_CLOSED: 6655 default: 6656 break; 6657 } 6658 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6659} 6660 6661void AudioFlinger::RecordThread::readInputParameters() 6662{ 6663 delete mRsmpInBuffer; 6664 // mRsmpInBuffer is always assigned a new[] below 6665 delete mRsmpOutBuffer; 6666 mRsmpOutBuffer = NULL; 6667 delete mResampler; 6668 mResampler = NULL; 6669 6670 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6671 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6672 mChannelCount = (uint16_t)popcount(mChannelMask); 6673 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6674 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6675 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6676 mFrameCount = mInputBytes / mFrameSize; 6677 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6678 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6679 6680 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6681 { 6682 int channelCount; 6683 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6684 // stereo to mono post process as the resampler always outputs stereo. 6685 if (mChannelCount == 1 && mReqChannelCount == 2) { 6686 channelCount = 1; 6687 } else { 6688 channelCount = 2; 6689 } 6690 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6691 mResampler->setSampleRate(mSampleRate); 6692 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6693 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6694 6695 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6696 if (mChannelCount == 1 && mReqChannelCount == 1) { 6697 mFrameCount >>= 1; 6698 } 6699 6700 } 6701 mRsmpInIndex = mFrameCount; 6702} 6703 6704unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6705{ 6706 Mutex::Autolock _l(mLock); 6707 if (initCheck() != NO_ERROR) { 6708 return 0; 6709 } 6710 6711 return mInput->stream->get_input_frames_lost(mInput->stream); 6712} 6713 6714uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6715{ 6716 Mutex::Autolock _l(mLock); 6717 uint32_t result = 0; 6718 if (getEffectChain_l(sessionId) != 0) { 6719 result = EFFECT_SESSION; 6720 } 6721 6722 for (size_t i = 0; i < mTracks.size(); ++i) { 6723 if (sessionId == mTracks[i]->sessionId()) { 6724 result |= TRACK_SESSION; 6725 break; 6726 } 6727 } 6728 6729 return result; 6730} 6731 6732KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6733{ 6734 KeyedVector<int, bool> ids; 6735 Mutex::Autolock _l(mLock); 6736 for (size_t j = 0; j < mTracks.size(); ++j) { 6737 sp<RecordThread::RecordTrack> track = mTracks[j]; 6738 int sessionId = track->sessionId(); 6739 if (ids.indexOfKey(sessionId) < 0) { 6740 ids.add(sessionId, true); 6741 } 6742 } 6743 return ids; 6744} 6745 6746AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6747{ 6748 Mutex::Autolock _l(mLock); 6749 AudioStreamIn *input = mInput; 6750 mInput = NULL; 6751 return input; 6752} 6753 6754// this method must always be called either with ThreadBase mLock held or inside the thread loop 6755audio_stream_t* AudioFlinger::RecordThread::stream() const 6756{ 6757 if (mInput == NULL) { 6758 return NULL; 6759 } 6760 return &mInput->stream->common; 6761} 6762 6763 6764// ---------------------------------------------------------------------------- 6765 6766audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6767{ 6768 if (!settingsAllowed()) { 6769 return 0; 6770 } 6771 Mutex::Autolock _l(mLock); 6772 return loadHwModule_l(name); 6773} 6774 6775// loadHwModule_l() must be called with AudioFlinger::mLock held 6776audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6777{ 6778 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6779 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6780 ALOGW("loadHwModule() module %s already loaded", name); 6781 return mAudioHwDevs.keyAt(i); 6782 } 6783 } 6784 6785 audio_hw_device_t *dev; 6786 6787 int rc = load_audio_interface(name, &dev); 6788 if (rc) { 6789 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6790 return 0; 6791 } 6792 6793 mHardwareStatus = AUDIO_HW_INIT; 6794 rc = dev->init_check(dev); 6795 mHardwareStatus = AUDIO_HW_IDLE; 6796 if (rc) { 6797 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6798 return 0; 6799 } 6800 6801 // Check and cache this HAL's level of support for master mute and master 6802 // volume. If this is the first HAL opened, and it supports the get 6803 // methods, use the initial values provided by the HAL as the current 6804 // master mute and volume settings. 6805 6806 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6807 { // scope for auto-lock pattern 6808 AutoMutex lock(mHardwareLock); 6809 6810 if (0 == mAudioHwDevs.size()) { 6811 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6812 if (NULL != dev->get_master_volume) { 6813 float mv; 6814 if (OK == dev->get_master_volume(dev, &mv)) { 6815 mMasterVolume = mv; 6816 } 6817 } 6818 6819 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6820 if (NULL != dev->get_master_mute) { 6821 bool mm; 6822 if (OK == dev->get_master_mute(dev, &mm)) { 6823 mMasterMute = mm; 6824 } 6825 } 6826 } 6827 6828 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6829 if ((NULL != dev->set_master_volume) && 6830 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6831 flags = static_cast<AudioHwDevice::Flags>(flags | 6832 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6833 } 6834 6835 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6836 if ((NULL != dev->set_master_mute) && 6837 (OK == dev->set_master_mute(dev, mMasterMute))) { 6838 flags = static_cast<AudioHwDevice::Flags>(flags | 6839 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6840 } 6841 6842 mHardwareStatus = AUDIO_HW_IDLE; 6843 } 6844 6845 audio_module_handle_t handle = nextUniqueId(); 6846 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6847 6848 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6849 name, dev->common.module->name, dev->common.module->id, handle); 6850 6851 return handle; 6852 6853} 6854 6855audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6856 audio_devices_t *pDevices, 6857 uint32_t *pSamplingRate, 6858 audio_format_t *pFormat, 6859 audio_channel_mask_t *pChannelMask, 6860 uint32_t *pLatencyMs, 6861 audio_output_flags_t flags) 6862{ 6863 status_t status; 6864 PlaybackThread *thread = NULL; 6865 struct audio_config config = { 6866 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6867 channel_mask: pChannelMask ? *pChannelMask : 0, 6868 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6869 }; 6870 audio_stream_out_t *outStream = NULL; 6871 AudioHwDevice *outHwDev; 6872 6873 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6874 module, 6875 (pDevices != NULL) ? *pDevices : 0, 6876 config.sample_rate, 6877 config.format, 6878 config.channel_mask, 6879 flags); 6880 6881 if (pDevices == NULL || *pDevices == 0) { 6882 return 0; 6883 } 6884 6885 Mutex::Autolock _l(mLock); 6886 6887 outHwDev = findSuitableHwDev_l(module, *pDevices); 6888 if (outHwDev == NULL) 6889 return 0; 6890 6891 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 6892 audio_io_handle_t id = nextUniqueId(); 6893 6894 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6895 6896 status = hwDevHal->open_output_stream(hwDevHal, 6897 id, 6898 *pDevices, 6899 (audio_output_flags_t)flags, 6900 &config, 6901 &outStream); 6902 6903 mHardwareStatus = AUDIO_HW_IDLE; 6904 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6905 outStream, 6906 config.sample_rate, 6907 config.format, 6908 config.channel_mask, 6909 status); 6910 6911 if (status == NO_ERROR && outStream != NULL) { 6912 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6913 6914 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6915 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6916 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6917 thread = new DirectOutputThread(this, output, id, *pDevices); 6918 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6919 } else { 6920 thread = new MixerThread(this, output, id, *pDevices); 6921 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6922 } 6923 mPlaybackThreads.add(id, thread); 6924 6925 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6926 if (pFormat != NULL) *pFormat = config.format; 6927 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6928 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6929 6930 // notify client processes of the new output creation 6931 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6932 6933 // the first primary output opened designates the primary hw device 6934 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6935 ALOGI("Using module %d has the primary audio interface", module); 6936 mPrimaryHardwareDev = outHwDev; 6937 6938 AutoMutex lock(mHardwareLock); 6939 mHardwareStatus = AUDIO_HW_SET_MODE; 6940 hwDevHal->set_mode(hwDevHal, mMode); 6941 mHardwareStatus = AUDIO_HW_IDLE; 6942 } 6943 return id; 6944 } 6945 6946 return 0; 6947} 6948 6949audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6950 audio_io_handle_t output2) 6951{ 6952 Mutex::Autolock _l(mLock); 6953 MixerThread *thread1 = checkMixerThread_l(output1); 6954 MixerThread *thread2 = checkMixerThread_l(output2); 6955 6956 if (thread1 == NULL || thread2 == NULL) { 6957 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6958 return 0; 6959 } 6960 6961 audio_io_handle_t id = nextUniqueId(); 6962 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6963 thread->addOutputTrack(thread2); 6964 mPlaybackThreads.add(id, thread); 6965 // notify client processes of the new output creation 6966 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6967 return id; 6968} 6969 6970status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6971{ 6972 return closeOutput_nonvirtual(output); 6973} 6974 6975status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6976{ 6977 // keep strong reference on the playback thread so that 6978 // it is not destroyed while exit() is executed 6979 sp<PlaybackThread> thread; 6980 { 6981 Mutex::Autolock _l(mLock); 6982 thread = checkPlaybackThread_l(output); 6983 if (thread == NULL) { 6984 return BAD_VALUE; 6985 } 6986 6987 ALOGV("closeOutput() %d", output); 6988 6989 if (thread->type() == ThreadBase::MIXER) { 6990 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6991 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6992 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6993 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6994 } 6995 } 6996 } 6997 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6998 mPlaybackThreads.removeItem(output); 6999 } 7000 thread->exit(); 7001 // The thread entity (active unit of execution) is no longer running here, 7002 // but the ThreadBase container still exists. 7003 7004 if (thread->type() != ThreadBase::DUPLICATING) { 7005 AudioStreamOut *out = thread->clearOutput(); 7006 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7007 // from now on thread->mOutput is NULL 7008 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7009 delete out; 7010 } 7011 return NO_ERROR; 7012} 7013 7014status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7015{ 7016 Mutex::Autolock _l(mLock); 7017 PlaybackThread *thread = checkPlaybackThread_l(output); 7018 7019 if (thread == NULL) { 7020 return BAD_VALUE; 7021 } 7022 7023 ALOGV("suspendOutput() %d", output); 7024 thread->suspend(); 7025 7026 return NO_ERROR; 7027} 7028 7029status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7030{ 7031 Mutex::Autolock _l(mLock); 7032 PlaybackThread *thread = checkPlaybackThread_l(output); 7033 7034 if (thread == NULL) { 7035 return BAD_VALUE; 7036 } 7037 7038 ALOGV("restoreOutput() %d", output); 7039 7040 thread->restore(); 7041 7042 return NO_ERROR; 7043} 7044 7045audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7046 audio_devices_t *pDevices, 7047 uint32_t *pSamplingRate, 7048 audio_format_t *pFormat, 7049 audio_channel_mask_t *pChannelMask) 7050{ 7051 status_t status; 7052 RecordThread *thread = NULL; 7053 struct audio_config config = { 7054 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7055 channel_mask: pChannelMask ? *pChannelMask : 0, 7056 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7057 }; 7058 uint32_t reqSamplingRate = config.sample_rate; 7059 audio_format_t reqFormat = config.format; 7060 audio_channel_mask_t reqChannels = config.channel_mask; 7061 audio_stream_in_t *inStream = NULL; 7062 AudioHwDevice *inHwDev; 7063 7064 if (pDevices == NULL || *pDevices == 0) { 7065 return 0; 7066 } 7067 7068 Mutex::Autolock _l(mLock); 7069 7070 inHwDev = findSuitableHwDev_l(module, *pDevices); 7071 if (inHwDev == NULL) 7072 return 0; 7073 7074 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7075 audio_io_handle_t id = nextUniqueId(); 7076 7077 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7078 &inStream); 7079 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7080 inStream, 7081 config.sample_rate, 7082 config.format, 7083 config.channel_mask, 7084 status); 7085 7086 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7087 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7088 // or stereo to mono conversions on 16 bit PCM inputs. 7089 if (status == BAD_VALUE && 7090 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7091 (config.sample_rate <= 2 * reqSamplingRate) && 7092 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7093 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7094 inStream = NULL; 7095 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7096 } 7097 7098 if (status == NO_ERROR && inStream != NULL) { 7099 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7100 7101 // Start record thread 7102 // RecorThread require both input and output device indication to forward to audio 7103 // pre processing modules 7104 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7105 thread = new RecordThread(this, 7106 input, 7107 reqSamplingRate, 7108 reqChannels, 7109 id, 7110 device); 7111 mRecordThreads.add(id, thread); 7112 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7113 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7114 if (pFormat != NULL) *pFormat = config.format; 7115 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7116 7117 // notify client processes of the new input creation 7118 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7119 return id; 7120 } 7121 7122 return 0; 7123} 7124 7125status_t AudioFlinger::closeInput(audio_io_handle_t input) 7126{ 7127 return closeInput_nonvirtual(input); 7128} 7129 7130status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7131{ 7132 // keep strong reference on the record thread so that 7133 // it is not destroyed while exit() is executed 7134 sp<RecordThread> thread; 7135 { 7136 Mutex::Autolock _l(mLock); 7137 thread = checkRecordThread_l(input); 7138 if (thread == 0) { 7139 return BAD_VALUE; 7140 } 7141 7142 ALOGV("closeInput() %d", input); 7143 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7144 mRecordThreads.removeItem(input); 7145 } 7146 thread->exit(); 7147 // The thread entity (active unit of execution) is no longer running here, 7148 // but the ThreadBase container still exists. 7149 7150 AudioStreamIn *in = thread->clearInput(); 7151 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7152 // from now on thread->mInput is NULL 7153 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7154 delete in; 7155 7156 return NO_ERROR; 7157} 7158 7159status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7160{ 7161 Mutex::Autolock _l(mLock); 7162 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7163 7164 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7165 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7166 thread->invalidateTracks(stream); 7167 } 7168 7169 return NO_ERROR; 7170} 7171 7172 7173int AudioFlinger::newAudioSessionId() 7174{ 7175 return nextUniqueId(); 7176} 7177 7178void AudioFlinger::acquireAudioSessionId(int audioSession) 7179{ 7180 Mutex::Autolock _l(mLock); 7181 pid_t caller = IPCThreadState::self()->getCallingPid(); 7182 ALOGV("acquiring %d from %d", audioSession, caller); 7183 size_t num = mAudioSessionRefs.size(); 7184 for (size_t i = 0; i< num; i++) { 7185 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7186 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7187 ref->mCnt++; 7188 ALOGV(" incremented refcount to %d", ref->mCnt); 7189 return; 7190 } 7191 } 7192 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7193 ALOGV(" added new entry for %d", audioSession); 7194} 7195 7196void AudioFlinger::releaseAudioSessionId(int audioSession) 7197{ 7198 Mutex::Autolock _l(mLock); 7199 pid_t caller = IPCThreadState::self()->getCallingPid(); 7200 ALOGV("releasing %d from %d", audioSession, caller); 7201 size_t num = mAudioSessionRefs.size(); 7202 for (size_t i = 0; i< num; i++) { 7203 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7204 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7205 ref->mCnt--; 7206 ALOGV(" decremented refcount to %d", ref->mCnt); 7207 if (ref->mCnt == 0) { 7208 mAudioSessionRefs.removeAt(i); 7209 delete ref; 7210 purgeStaleEffects_l(); 7211 } 7212 return; 7213 } 7214 } 7215 ALOGW("session id %d not found for pid %d", audioSession, caller); 7216} 7217 7218void AudioFlinger::purgeStaleEffects_l() { 7219 7220 ALOGV("purging stale effects"); 7221 7222 Vector< sp<EffectChain> > chains; 7223 7224 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7225 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7226 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7227 sp<EffectChain> ec = t->mEffectChains[j]; 7228 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7229 chains.push(ec); 7230 } 7231 } 7232 } 7233 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7234 sp<RecordThread> t = mRecordThreads.valueAt(i); 7235 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7236 sp<EffectChain> ec = t->mEffectChains[j]; 7237 chains.push(ec); 7238 } 7239 } 7240 7241 for (size_t i = 0; i < chains.size(); i++) { 7242 sp<EffectChain> ec = chains[i]; 7243 int sessionid = ec->sessionId(); 7244 sp<ThreadBase> t = ec->mThread.promote(); 7245 if (t == 0) { 7246 continue; 7247 } 7248 size_t numsessionrefs = mAudioSessionRefs.size(); 7249 bool found = false; 7250 for (size_t k = 0; k < numsessionrefs; k++) { 7251 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7252 if (ref->mSessionid == sessionid) { 7253 ALOGV(" session %d still exists for %d with %d refs", 7254 sessionid, ref->mPid, ref->mCnt); 7255 found = true; 7256 break; 7257 } 7258 } 7259 if (!found) { 7260 Mutex::Autolock _l (t->mLock); 7261 // remove all effects from the chain 7262 while (ec->mEffects.size()) { 7263 sp<EffectModule> effect = ec->mEffects[0]; 7264 effect->unPin(); 7265 t->removeEffect_l(effect); 7266 if (effect->purgeHandles()) { 7267 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7268 } 7269 AudioSystem::unregisterEffect(effect->id()); 7270 } 7271 } 7272 } 7273 return; 7274} 7275 7276// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7277AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7278{ 7279 return mPlaybackThreads.valueFor(output).get(); 7280} 7281 7282// checkMixerThread_l() must be called with AudioFlinger::mLock held 7283AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7284{ 7285 PlaybackThread *thread = checkPlaybackThread_l(output); 7286 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7287} 7288 7289// checkRecordThread_l() must be called with AudioFlinger::mLock held 7290AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7291{ 7292 return mRecordThreads.valueFor(input).get(); 7293} 7294 7295uint32_t AudioFlinger::nextUniqueId() 7296{ 7297 return android_atomic_inc(&mNextUniqueId); 7298} 7299 7300AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7301{ 7302 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7303 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7304 AudioStreamOut *output = thread->getOutput(); 7305 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7306 return thread; 7307 } 7308 } 7309 return NULL; 7310} 7311 7312audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7313{ 7314 PlaybackThread *thread = primaryPlaybackThread_l(); 7315 7316 if (thread == NULL) { 7317 return 0; 7318 } 7319 7320 return thread->device(); 7321} 7322 7323sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7324 int triggerSession, 7325 int listenerSession, 7326 sync_event_callback_t callBack, 7327 void *cookie) 7328{ 7329 Mutex::Autolock _l(mLock); 7330 7331 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7332 status_t playStatus = NAME_NOT_FOUND; 7333 status_t recStatus = NAME_NOT_FOUND; 7334 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7335 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7336 if (playStatus == NO_ERROR) { 7337 return event; 7338 } 7339 } 7340 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7341 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7342 if (recStatus == NO_ERROR) { 7343 return event; 7344 } 7345 } 7346 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7347 mPendingSyncEvents.add(event); 7348 } else { 7349 ALOGV("createSyncEvent() invalid event %d", event->type()); 7350 event.clear(); 7351 } 7352 return event; 7353} 7354 7355// ---------------------------------------------------------------------------- 7356// Effect management 7357// ---------------------------------------------------------------------------- 7358 7359 7360status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7361{ 7362 Mutex::Autolock _l(mLock); 7363 return EffectQueryNumberEffects(numEffects); 7364} 7365 7366status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7367{ 7368 Mutex::Autolock _l(mLock); 7369 return EffectQueryEffect(index, descriptor); 7370} 7371 7372status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7373 effect_descriptor_t *descriptor) const 7374{ 7375 Mutex::Autolock _l(mLock); 7376 return EffectGetDescriptor(pUuid, descriptor); 7377} 7378 7379 7380sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7381 effect_descriptor_t *pDesc, 7382 const sp<IEffectClient>& effectClient, 7383 int32_t priority, 7384 audio_io_handle_t io, 7385 int sessionId, 7386 status_t *status, 7387 int *id, 7388 int *enabled) 7389{ 7390 status_t lStatus = NO_ERROR; 7391 sp<EffectHandle> handle; 7392 effect_descriptor_t desc; 7393 7394 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7395 pid, effectClient.get(), priority, sessionId, io); 7396 7397 if (pDesc == NULL) { 7398 lStatus = BAD_VALUE; 7399 goto Exit; 7400 } 7401 7402 // check audio settings permission for global effects 7403 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7404 lStatus = PERMISSION_DENIED; 7405 goto Exit; 7406 } 7407 7408 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7409 // that can only be created by audio policy manager (running in same process) 7410 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7411 lStatus = PERMISSION_DENIED; 7412 goto Exit; 7413 } 7414 7415 if (io == 0) { 7416 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7417 // output must be specified by AudioPolicyManager when using session 7418 // AUDIO_SESSION_OUTPUT_STAGE 7419 lStatus = BAD_VALUE; 7420 goto Exit; 7421 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7422 // if the output returned by getOutputForEffect() is removed before we lock the 7423 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7424 // and we will exit safely 7425 io = AudioSystem::getOutputForEffect(&desc); 7426 } 7427 } 7428 7429 { 7430 Mutex::Autolock _l(mLock); 7431 7432 7433 if (!EffectIsNullUuid(&pDesc->uuid)) { 7434 // if uuid is specified, request effect descriptor 7435 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7436 if (lStatus < 0) { 7437 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7438 goto Exit; 7439 } 7440 } else { 7441 // if uuid is not specified, look for an available implementation 7442 // of the required type in effect factory 7443 if (EffectIsNullUuid(&pDesc->type)) { 7444 ALOGW("createEffect() no effect type"); 7445 lStatus = BAD_VALUE; 7446 goto Exit; 7447 } 7448 uint32_t numEffects = 0; 7449 effect_descriptor_t d; 7450 d.flags = 0; // prevent compiler warning 7451 bool found = false; 7452 7453 lStatus = EffectQueryNumberEffects(&numEffects); 7454 if (lStatus < 0) { 7455 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7456 goto Exit; 7457 } 7458 for (uint32_t i = 0; i < numEffects; i++) { 7459 lStatus = EffectQueryEffect(i, &desc); 7460 if (lStatus < 0) { 7461 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7462 continue; 7463 } 7464 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7465 // If matching type found save effect descriptor. If the session is 7466 // 0 and the effect is not auxiliary, continue enumeration in case 7467 // an auxiliary version of this effect type is available 7468 found = true; 7469 d = desc; 7470 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7471 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7472 break; 7473 } 7474 } 7475 } 7476 if (!found) { 7477 lStatus = BAD_VALUE; 7478 ALOGW("createEffect() effect not found"); 7479 goto Exit; 7480 } 7481 // For same effect type, chose auxiliary version over insert version if 7482 // connect to output mix (Compliance to OpenSL ES) 7483 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7484 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7485 desc = d; 7486 } 7487 } 7488 7489 // Do not allow auxiliary effects on a session different from 0 (output mix) 7490 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7491 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7492 lStatus = INVALID_OPERATION; 7493 goto Exit; 7494 } 7495 7496 // check recording permission for visualizer 7497 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7498 !recordingAllowed()) { 7499 lStatus = PERMISSION_DENIED; 7500 goto Exit; 7501 } 7502 7503 // return effect descriptor 7504 *pDesc = desc; 7505 7506 // If output is not specified try to find a matching audio session ID in one of the 7507 // output threads. 7508 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7509 // because of code checking output when entering the function. 7510 // Note: io is never 0 when creating an effect on an input 7511 if (io == 0) { 7512 // look for the thread where the specified audio session is present 7513 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7514 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7515 io = mPlaybackThreads.keyAt(i); 7516 break; 7517 } 7518 } 7519 if (io == 0) { 7520 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7521 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7522 io = mRecordThreads.keyAt(i); 7523 break; 7524 } 7525 } 7526 } 7527 // If no output thread contains the requested session ID, default to 7528 // first output. The effect chain will be moved to the correct output 7529 // thread when a track with the same session ID is created 7530 if (io == 0 && mPlaybackThreads.size()) { 7531 io = mPlaybackThreads.keyAt(0); 7532 } 7533 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7534 } 7535 ThreadBase *thread = checkRecordThread_l(io); 7536 if (thread == NULL) { 7537 thread = checkPlaybackThread_l(io); 7538 if (thread == NULL) { 7539 ALOGE("createEffect() unknown output thread"); 7540 lStatus = BAD_VALUE; 7541 goto Exit; 7542 } 7543 } 7544 7545 sp<Client> client = registerPid_l(pid); 7546 7547 // create effect on selected output thread 7548 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7549 &desc, enabled, &lStatus); 7550 if (handle != 0 && id != NULL) { 7551 *id = handle->id(); 7552 } 7553 } 7554 7555Exit: 7556 if (status != NULL) { 7557 *status = lStatus; 7558 } 7559 return handle; 7560} 7561 7562status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7563 audio_io_handle_t dstOutput) 7564{ 7565 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7566 sessionId, srcOutput, dstOutput); 7567 Mutex::Autolock _l(mLock); 7568 if (srcOutput == dstOutput) { 7569 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7570 return NO_ERROR; 7571 } 7572 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7573 if (srcThread == NULL) { 7574 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7575 return BAD_VALUE; 7576 } 7577 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7578 if (dstThread == NULL) { 7579 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7580 return BAD_VALUE; 7581 } 7582 7583 Mutex::Autolock _dl(dstThread->mLock); 7584 Mutex::Autolock _sl(srcThread->mLock); 7585 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7586 7587 return NO_ERROR; 7588} 7589 7590// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7591status_t AudioFlinger::moveEffectChain_l(int sessionId, 7592 AudioFlinger::PlaybackThread *srcThread, 7593 AudioFlinger::PlaybackThread *dstThread, 7594 bool reRegister) 7595{ 7596 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7597 sessionId, srcThread, dstThread); 7598 7599 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7600 if (chain == 0) { 7601 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7602 sessionId, srcThread); 7603 return INVALID_OPERATION; 7604 } 7605 7606 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7607 // so that a new chain is created with correct parameters when first effect is added. This is 7608 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7609 // removed. 7610 srcThread->removeEffectChain_l(chain); 7611 7612 // transfer all effects one by one so that new effect chain is created on new thread with 7613 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7614 audio_io_handle_t dstOutput = dstThread->id(); 7615 sp<EffectChain> dstChain; 7616 uint32_t strategy = 0; // prevent compiler warning 7617 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7618 while (effect != 0) { 7619 srcThread->removeEffect_l(effect); 7620 dstThread->addEffect_l(effect); 7621 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7622 if (effect->state() == EffectModule::ACTIVE || 7623 effect->state() == EffectModule::STOPPING) { 7624 effect->start(); 7625 } 7626 // if the move request is not received from audio policy manager, the effect must be 7627 // re-registered with the new strategy and output 7628 if (dstChain == 0) { 7629 dstChain = effect->chain().promote(); 7630 if (dstChain == 0) { 7631 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7632 srcThread->addEffect_l(effect); 7633 return NO_INIT; 7634 } 7635 strategy = dstChain->strategy(); 7636 } 7637 if (reRegister) { 7638 AudioSystem::unregisterEffect(effect->id()); 7639 AudioSystem::registerEffect(&effect->desc(), 7640 dstOutput, 7641 strategy, 7642 sessionId, 7643 effect->id()); 7644 } 7645 effect = chain->getEffectFromId_l(0); 7646 } 7647 7648 return NO_ERROR; 7649} 7650 7651 7652// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7653sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7654 const sp<AudioFlinger::Client>& client, 7655 const sp<IEffectClient>& effectClient, 7656 int32_t priority, 7657 int sessionId, 7658 effect_descriptor_t *desc, 7659 int *enabled, 7660 status_t *status 7661 ) 7662{ 7663 sp<EffectModule> effect; 7664 sp<EffectHandle> handle; 7665 status_t lStatus; 7666 sp<EffectChain> chain; 7667 bool chainCreated = false; 7668 bool effectCreated = false; 7669 bool effectRegistered = false; 7670 7671 lStatus = initCheck(); 7672 if (lStatus != NO_ERROR) { 7673 ALOGW("createEffect_l() Audio driver not initialized."); 7674 goto Exit; 7675 } 7676 7677 // Do not allow effects with session ID 0 on direct output or duplicating threads 7678 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7679 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7680 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7681 desc->name, sessionId); 7682 lStatus = BAD_VALUE; 7683 goto Exit; 7684 } 7685 // Only Pre processor effects are allowed on input threads and only on input threads 7686 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7687 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7688 desc->name, desc->flags, mType); 7689 lStatus = BAD_VALUE; 7690 goto Exit; 7691 } 7692 7693 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7694 7695 { // scope for mLock 7696 Mutex::Autolock _l(mLock); 7697 7698 // check for existing effect chain with the requested audio session 7699 chain = getEffectChain_l(sessionId); 7700 if (chain == 0) { 7701 // create a new chain for this session 7702 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7703 chain = new EffectChain(this, sessionId); 7704 addEffectChain_l(chain); 7705 chain->setStrategy(getStrategyForSession_l(sessionId)); 7706 chainCreated = true; 7707 } else { 7708 effect = chain->getEffectFromDesc_l(desc); 7709 } 7710 7711 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7712 7713 if (effect == 0) { 7714 int id = mAudioFlinger->nextUniqueId(); 7715 // Check CPU and memory usage 7716 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7717 if (lStatus != NO_ERROR) { 7718 goto Exit; 7719 } 7720 effectRegistered = true; 7721 // create a new effect module if none present in the chain 7722 effect = new EffectModule(this, chain, desc, id, sessionId); 7723 lStatus = effect->status(); 7724 if (lStatus != NO_ERROR) { 7725 goto Exit; 7726 } 7727 lStatus = chain->addEffect_l(effect); 7728 if (lStatus != NO_ERROR) { 7729 goto Exit; 7730 } 7731 effectCreated = true; 7732 7733 effect->setDevice(mDevice); 7734 effect->setMode(mAudioFlinger->getMode()); 7735 } 7736 // create effect handle and connect it to effect module 7737 handle = new EffectHandle(effect, client, effectClient, priority); 7738 lStatus = effect->addHandle(handle.get()); 7739 if (enabled != NULL) { 7740 *enabled = (int)effect->isEnabled(); 7741 } 7742 } 7743 7744Exit: 7745 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7746 Mutex::Autolock _l(mLock); 7747 if (effectCreated) { 7748 chain->removeEffect_l(effect); 7749 } 7750 if (effectRegistered) { 7751 AudioSystem::unregisterEffect(effect->id()); 7752 } 7753 if (chainCreated) { 7754 removeEffectChain_l(chain); 7755 } 7756 handle.clear(); 7757 } 7758 7759 if (status != NULL) { 7760 *status = lStatus; 7761 } 7762 return handle; 7763} 7764 7765sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7766{ 7767 Mutex::Autolock _l(mLock); 7768 return getEffect_l(sessionId, effectId); 7769} 7770 7771sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7772{ 7773 sp<EffectChain> chain = getEffectChain_l(sessionId); 7774 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7775} 7776 7777// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7778// PlaybackThread::mLock held 7779status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7780{ 7781 // check for existing effect chain with the requested audio session 7782 int sessionId = effect->sessionId(); 7783 sp<EffectChain> chain = getEffectChain_l(sessionId); 7784 bool chainCreated = false; 7785 7786 if (chain == 0) { 7787 // create a new chain for this session 7788 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7789 chain = new EffectChain(this, sessionId); 7790 addEffectChain_l(chain); 7791 chain->setStrategy(getStrategyForSession_l(sessionId)); 7792 chainCreated = true; 7793 } 7794 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7795 7796 if (chain->getEffectFromId_l(effect->id()) != 0) { 7797 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7798 this, effect->desc().name, chain.get()); 7799 return BAD_VALUE; 7800 } 7801 7802 status_t status = chain->addEffect_l(effect); 7803 if (status != NO_ERROR) { 7804 if (chainCreated) { 7805 removeEffectChain_l(chain); 7806 } 7807 return status; 7808 } 7809 7810 effect->setDevice(mDevice); 7811 effect->setMode(mAudioFlinger->getMode()); 7812 return NO_ERROR; 7813} 7814 7815void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7816 7817 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7818 effect_descriptor_t desc = effect->desc(); 7819 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7820 detachAuxEffect_l(effect->id()); 7821 } 7822 7823 sp<EffectChain> chain = effect->chain().promote(); 7824 if (chain != 0) { 7825 // remove effect chain if removing last effect 7826 if (chain->removeEffect_l(effect) == 0) { 7827 removeEffectChain_l(chain); 7828 } 7829 } else { 7830 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7831 } 7832} 7833 7834void AudioFlinger::ThreadBase::lockEffectChains_l( 7835 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7836{ 7837 effectChains = mEffectChains; 7838 for (size_t i = 0; i < mEffectChains.size(); i++) { 7839 mEffectChains[i]->lock(); 7840 } 7841} 7842 7843void AudioFlinger::ThreadBase::unlockEffectChains( 7844 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7845{ 7846 for (size_t i = 0; i < effectChains.size(); i++) { 7847 effectChains[i]->unlock(); 7848 } 7849} 7850 7851sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7852{ 7853 Mutex::Autolock _l(mLock); 7854 return getEffectChain_l(sessionId); 7855} 7856 7857sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 7858{ 7859 size_t size = mEffectChains.size(); 7860 for (size_t i = 0; i < size; i++) { 7861 if (mEffectChains[i]->sessionId() == sessionId) { 7862 return mEffectChains[i]; 7863 } 7864 } 7865 return 0; 7866} 7867 7868void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7869{ 7870 Mutex::Autolock _l(mLock); 7871 size_t size = mEffectChains.size(); 7872 for (size_t i = 0; i < size; i++) { 7873 mEffectChains[i]->setMode_l(mode); 7874 } 7875} 7876 7877void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7878 EffectHandle *handle, 7879 bool unpinIfLast) { 7880 7881 Mutex::Autolock _l(mLock); 7882 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7883 // delete the effect module if removing last handle on it 7884 if (effect->removeHandle(handle) == 0) { 7885 if (!effect->isPinned() || unpinIfLast) { 7886 removeEffect_l(effect); 7887 AudioSystem::unregisterEffect(effect->id()); 7888 } 7889 } 7890} 7891 7892status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7893{ 7894 int session = chain->sessionId(); 7895 int16_t *buffer = mMixBuffer; 7896 bool ownsBuffer = false; 7897 7898 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7899 if (session > 0) { 7900 // Only one effect chain can be present in direct output thread and it uses 7901 // the mix buffer as input 7902 if (mType != DIRECT) { 7903 size_t numSamples = mNormalFrameCount * mChannelCount; 7904 buffer = new int16_t[numSamples]; 7905 memset(buffer, 0, numSamples * sizeof(int16_t)); 7906 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7907 ownsBuffer = true; 7908 } 7909 7910 // Attach all tracks with same session ID to this chain. 7911 for (size_t i = 0; i < mTracks.size(); ++i) { 7912 sp<Track> track = mTracks[i]; 7913 if (session == track->sessionId()) { 7914 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7915 track->setMainBuffer(buffer); 7916 chain->incTrackCnt(); 7917 } 7918 } 7919 7920 // indicate all active tracks in the chain 7921 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7922 sp<Track> track = mActiveTracks[i].promote(); 7923 if (track == 0) continue; 7924 if (session == track->sessionId()) { 7925 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7926 chain->incActiveTrackCnt(); 7927 } 7928 } 7929 } 7930 7931 chain->setInBuffer(buffer, ownsBuffer); 7932 chain->setOutBuffer(mMixBuffer); 7933 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7934 // chains list in order to be processed last as it contains output stage effects 7935 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7936 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7937 // after track specific effects and before output stage 7938 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7939 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7940 // Effect chain for other sessions are inserted at beginning of effect 7941 // chains list to be processed before output mix effects. Relative order between other 7942 // sessions is not important 7943 size_t size = mEffectChains.size(); 7944 size_t i = 0; 7945 for (i = 0; i < size; i++) { 7946 if (mEffectChains[i]->sessionId() < session) break; 7947 } 7948 mEffectChains.insertAt(chain, i); 7949 checkSuspendOnAddEffectChain_l(chain); 7950 7951 return NO_ERROR; 7952} 7953 7954size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7955{ 7956 int session = chain->sessionId(); 7957 7958 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7959 7960 for (size_t i = 0; i < mEffectChains.size(); i++) { 7961 if (chain == mEffectChains[i]) { 7962 mEffectChains.removeAt(i); 7963 // detach all active tracks from the chain 7964 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7965 sp<Track> track = mActiveTracks[i].promote(); 7966 if (track == 0) continue; 7967 if (session == track->sessionId()) { 7968 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7969 chain.get(), session); 7970 chain->decActiveTrackCnt(); 7971 } 7972 } 7973 7974 // detach all tracks with same session ID from this chain 7975 for (size_t i = 0; i < mTracks.size(); ++i) { 7976 sp<Track> track = mTracks[i]; 7977 if (session == track->sessionId()) { 7978 track->setMainBuffer(mMixBuffer); 7979 chain->decTrackCnt(); 7980 } 7981 } 7982 break; 7983 } 7984 } 7985 return mEffectChains.size(); 7986} 7987 7988status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7989 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7990{ 7991 Mutex::Autolock _l(mLock); 7992 return attachAuxEffect_l(track, EffectId); 7993} 7994 7995status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7996 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7997{ 7998 status_t status = NO_ERROR; 7999 8000 if (EffectId == 0) { 8001 track->setAuxBuffer(0, NULL); 8002 } else { 8003 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8004 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8005 if (effect != 0) { 8006 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8007 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8008 } else { 8009 status = INVALID_OPERATION; 8010 } 8011 } else { 8012 status = BAD_VALUE; 8013 } 8014 } 8015 return status; 8016} 8017 8018void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8019{ 8020 for (size_t i = 0; i < mTracks.size(); ++i) { 8021 sp<Track> track = mTracks[i]; 8022 if (track->auxEffectId() == effectId) { 8023 attachAuxEffect_l(track, 0); 8024 } 8025 } 8026} 8027 8028status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8029{ 8030 // only one chain per input thread 8031 if (mEffectChains.size() != 0) { 8032 return INVALID_OPERATION; 8033 } 8034 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8035 8036 chain->setInBuffer(NULL); 8037 chain->setOutBuffer(NULL); 8038 8039 checkSuspendOnAddEffectChain_l(chain); 8040 8041 mEffectChains.add(chain); 8042 8043 return NO_ERROR; 8044} 8045 8046size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8047{ 8048 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8049 ALOGW_IF(mEffectChains.size() != 1, 8050 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8051 chain.get(), mEffectChains.size(), this); 8052 if (mEffectChains.size() == 1) { 8053 mEffectChains.removeAt(0); 8054 } 8055 return 0; 8056} 8057 8058// ---------------------------------------------------------------------------- 8059// EffectModule implementation 8060// ---------------------------------------------------------------------------- 8061 8062#undef LOG_TAG 8063#define LOG_TAG "AudioFlinger::EffectModule" 8064 8065AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8066 const wp<AudioFlinger::EffectChain>& chain, 8067 effect_descriptor_t *desc, 8068 int id, 8069 int sessionId) 8070 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8071 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8072 mDescriptor(*desc), 8073 // mConfig is set by configure() and not used before then 8074 mEffectInterface(NULL), 8075 mStatus(NO_INIT), mState(IDLE), 8076 // mMaxDisableWaitCnt is set by configure() and not used before then 8077 // mDisableWaitCnt is set by process() and updateState() and not used before then 8078 mSuspended(false) 8079{ 8080 ALOGV("Constructor %p", this); 8081 int lStatus; 8082 8083 // create effect engine from effect factory 8084 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8085 8086 if (mStatus != NO_ERROR) { 8087 return; 8088 } 8089 lStatus = init(); 8090 if (lStatus < 0) { 8091 mStatus = lStatus; 8092 goto Error; 8093 } 8094 8095 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8096 return; 8097Error: 8098 EffectRelease(mEffectInterface); 8099 mEffectInterface = NULL; 8100 ALOGV("Constructor Error %d", mStatus); 8101} 8102 8103AudioFlinger::EffectModule::~EffectModule() 8104{ 8105 ALOGV("Destructor %p", this); 8106 if (mEffectInterface != NULL) { 8107 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8108 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8109 sp<ThreadBase> thread = mThread.promote(); 8110 if (thread != 0) { 8111 audio_stream_t *stream = thread->stream(); 8112 if (stream != NULL) { 8113 stream->remove_audio_effect(stream, mEffectInterface); 8114 } 8115 } 8116 } 8117 // release effect engine 8118 EffectRelease(mEffectInterface); 8119 } 8120} 8121 8122status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8123{ 8124 status_t status; 8125 8126 Mutex::Autolock _l(mLock); 8127 int priority = handle->priority(); 8128 size_t size = mHandles.size(); 8129 EffectHandle *controlHandle = NULL; 8130 size_t i; 8131 for (i = 0; i < size; i++) { 8132 EffectHandle *h = mHandles[i]; 8133 if (h == NULL || h->destroyed_l()) continue; 8134 // first non destroyed handle is considered in control 8135 if (controlHandle == NULL) 8136 controlHandle = h; 8137 if (h->priority() <= priority) break; 8138 } 8139 // if inserted in first place, move effect control from previous owner to this handle 8140 if (i == 0) { 8141 bool enabled = false; 8142 if (controlHandle != NULL) { 8143 enabled = controlHandle->enabled(); 8144 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8145 } 8146 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8147 status = NO_ERROR; 8148 } else { 8149 status = ALREADY_EXISTS; 8150 } 8151 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8152 mHandles.insertAt(handle, i); 8153 return status; 8154} 8155 8156size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8157{ 8158 Mutex::Autolock _l(mLock); 8159 size_t size = mHandles.size(); 8160 size_t i; 8161 for (i = 0; i < size; i++) { 8162 if (mHandles[i] == handle) break; 8163 } 8164 if (i == size) { 8165 return size; 8166 } 8167 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8168 8169 mHandles.removeAt(i); 8170 // if removed from first place, move effect control from this handle to next in line 8171 if (i == 0) { 8172 EffectHandle *h = controlHandle_l(); 8173 if (h != NULL) { 8174 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8175 } 8176 } 8177 8178 // Prevent calls to process() and other functions on effect interface from now on. 8179 // The effect engine will be released by the destructor when the last strong reference on 8180 // this object is released which can happen after next process is called. 8181 if (mHandles.size() == 0 && !mPinned) { 8182 mState = DESTROYED; 8183 } 8184 8185 return mHandles.size(); 8186} 8187 8188// must be called with EffectModule::mLock held 8189AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8190{ 8191 // the first valid handle in the list has control over the module 8192 for (size_t i = 0; i < mHandles.size(); i++) { 8193 EffectHandle *h = mHandles[i]; 8194 if (h != NULL && !h->destroyed_l()) { 8195 return h; 8196 } 8197 } 8198 8199 return NULL; 8200} 8201 8202size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8203{ 8204 ALOGV("disconnect() %p handle %p", this, handle); 8205 // keep a strong reference on this EffectModule to avoid calling the 8206 // destructor before we exit 8207 sp<EffectModule> keep(this); 8208 { 8209 sp<ThreadBase> thread = mThread.promote(); 8210 if (thread != 0) { 8211 thread->disconnectEffect(keep, handle, unpinIfLast); 8212 } 8213 } 8214 return mHandles.size(); 8215} 8216 8217void AudioFlinger::EffectModule::updateState() { 8218 Mutex::Autolock _l(mLock); 8219 8220 switch (mState) { 8221 case RESTART: 8222 reset_l(); 8223 // FALL THROUGH 8224 8225 case STARTING: 8226 // clear auxiliary effect input buffer for next accumulation 8227 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8228 memset(mConfig.inputCfg.buffer.raw, 8229 0, 8230 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8231 } 8232 start_l(); 8233 mState = ACTIVE; 8234 break; 8235 case STOPPING: 8236 stop_l(); 8237 mDisableWaitCnt = mMaxDisableWaitCnt; 8238 mState = STOPPED; 8239 break; 8240 case STOPPED: 8241 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8242 // turn off sequence. 8243 if (--mDisableWaitCnt == 0) { 8244 reset_l(); 8245 mState = IDLE; 8246 } 8247 break; 8248 default: //IDLE , ACTIVE, DESTROYED 8249 break; 8250 } 8251} 8252 8253void AudioFlinger::EffectModule::process() 8254{ 8255 Mutex::Autolock _l(mLock); 8256 8257 if (mState == DESTROYED || mEffectInterface == NULL || 8258 mConfig.inputCfg.buffer.raw == NULL || 8259 mConfig.outputCfg.buffer.raw == NULL) { 8260 return; 8261 } 8262 8263 if (isProcessEnabled()) { 8264 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8265 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8266 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8267 mConfig.inputCfg.buffer.s32, 8268 mConfig.inputCfg.buffer.frameCount/2); 8269 } 8270 8271 // do the actual processing in the effect engine 8272 int ret = (*mEffectInterface)->process(mEffectInterface, 8273 &mConfig.inputCfg.buffer, 8274 &mConfig.outputCfg.buffer); 8275 8276 // force transition to IDLE state when engine is ready 8277 if (mState == STOPPED && ret == -ENODATA) { 8278 mDisableWaitCnt = 1; 8279 } 8280 8281 // clear auxiliary effect input buffer for next accumulation 8282 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8283 memset(mConfig.inputCfg.buffer.raw, 0, 8284 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8285 } 8286 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8287 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8288 // If an insert effect is idle and input buffer is different from output buffer, 8289 // accumulate input onto output 8290 sp<EffectChain> chain = mChain.promote(); 8291 if (chain != 0 && chain->activeTrackCnt() != 0) { 8292 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8293 int16_t *in = mConfig.inputCfg.buffer.s16; 8294 int16_t *out = mConfig.outputCfg.buffer.s16; 8295 for (size_t i = 0; i < frameCnt; i++) { 8296 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8297 } 8298 } 8299 } 8300} 8301 8302void AudioFlinger::EffectModule::reset_l() 8303{ 8304 if (mEffectInterface == NULL) { 8305 return; 8306 } 8307 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8308} 8309 8310status_t AudioFlinger::EffectModule::configure() 8311{ 8312 if (mEffectInterface == NULL) { 8313 return NO_INIT; 8314 } 8315 8316 sp<ThreadBase> thread = mThread.promote(); 8317 if (thread == 0) { 8318 return DEAD_OBJECT; 8319 } 8320 8321 // TODO: handle configuration of effects replacing track process 8322 audio_channel_mask_t channelMask = thread->channelMask(); 8323 8324 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8325 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8326 } else { 8327 mConfig.inputCfg.channels = channelMask; 8328 } 8329 mConfig.outputCfg.channels = channelMask; 8330 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8331 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8332 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8333 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8334 mConfig.inputCfg.bufferProvider.cookie = NULL; 8335 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8336 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8337 mConfig.outputCfg.bufferProvider.cookie = NULL; 8338 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8339 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8340 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8341 // Insert effect: 8342 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8343 // always overwrites output buffer: input buffer == output buffer 8344 // - in other sessions: 8345 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8346 // other effect: overwrites output buffer: input buffer == output buffer 8347 // Auxiliary effect: 8348 // accumulates in output buffer: input buffer != output buffer 8349 // Therefore: accumulate <=> input buffer != output buffer 8350 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8351 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8352 } else { 8353 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8354 } 8355 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8356 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8357 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8358 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8359 8360 ALOGV("configure() %p thread %p buffer %p framecount %d", 8361 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8362 8363 status_t cmdStatus; 8364 uint32_t size = sizeof(int); 8365 status_t status = (*mEffectInterface)->command(mEffectInterface, 8366 EFFECT_CMD_SET_CONFIG, 8367 sizeof(effect_config_t), 8368 &mConfig, 8369 &size, 8370 &cmdStatus); 8371 if (status == 0) { 8372 status = cmdStatus; 8373 } 8374 8375 if (status == 0 && 8376 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8377 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8378 effect_param_t *p = (effect_param_t *)buf32; 8379 8380 p->psize = sizeof(uint32_t); 8381 p->vsize = sizeof(uint32_t); 8382 size = sizeof(int); 8383 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8384 8385 uint32_t latency = 0; 8386 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8387 if (pbt != NULL) { 8388 latency = pbt->latency_l(); 8389 } 8390 8391 *((int32_t *)p->data + 1)= latency; 8392 (*mEffectInterface)->command(mEffectInterface, 8393 EFFECT_CMD_SET_PARAM, 8394 sizeof(effect_param_t) + 8, 8395 &buf32, 8396 &size, 8397 &cmdStatus); 8398 } 8399 8400 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8401 (1000 * mConfig.outputCfg.buffer.frameCount); 8402 8403 return status; 8404} 8405 8406status_t AudioFlinger::EffectModule::init() 8407{ 8408 Mutex::Autolock _l(mLock); 8409 if (mEffectInterface == NULL) { 8410 return NO_INIT; 8411 } 8412 status_t cmdStatus; 8413 uint32_t size = sizeof(status_t); 8414 status_t status = (*mEffectInterface)->command(mEffectInterface, 8415 EFFECT_CMD_INIT, 8416 0, 8417 NULL, 8418 &size, 8419 &cmdStatus); 8420 if (status == 0) { 8421 status = cmdStatus; 8422 } 8423 return status; 8424} 8425 8426status_t AudioFlinger::EffectModule::start() 8427{ 8428 Mutex::Autolock _l(mLock); 8429 return start_l(); 8430} 8431 8432status_t AudioFlinger::EffectModule::start_l() 8433{ 8434 if (mEffectInterface == NULL) { 8435 return NO_INIT; 8436 } 8437 status_t cmdStatus; 8438 uint32_t size = sizeof(status_t); 8439 status_t status = (*mEffectInterface)->command(mEffectInterface, 8440 EFFECT_CMD_ENABLE, 8441 0, 8442 NULL, 8443 &size, 8444 &cmdStatus); 8445 if (status == 0) { 8446 status = cmdStatus; 8447 } 8448 if (status == 0 && 8449 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8450 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8451 sp<ThreadBase> thread = mThread.promote(); 8452 if (thread != 0) { 8453 audio_stream_t *stream = thread->stream(); 8454 if (stream != NULL) { 8455 stream->add_audio_effect(stream, mEffectInterface); 8456 } 8457 } 8458 } 8459 return status; 8460} 8461 8462status_t AudioFlinger::EffectModule::stop() 8463{ 8464 Mutex::Autolock _l(mLock); 8465 return stop_l(); 8466} 8467 8468status_t AudioFlinger::EffectModule::stop_l() 8469{ 8470 if (mEffectInterface == NULL) { 8471 return NO_INIT; 8472 } 8473 status_t cmdStatus; 8474 uint32_t size = sizeof(status_t); 8475 status_t status = (*mEffectInterface)->command(mEffectInterface, 8476 EFFECT_CMD_DISABLE, 8477 0, 8478 NULL, 8479 &size, 8480 &cmdStatus); 8481 if (status == 0) { 8482 status = cmdStatus; 8483 } 8484 if (status == 0 && 8485 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8486 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8487 sp<ThreadBase> thread = mThread.promote(); 8488 if (thread != 0) { 8489 audio_stream_t *stream = thread->stream(); 8490 if (stream != NULL) { 8491 stream->remove_audio_effect(stream, mEffectInterface); 8492 } 8493 } 8494 } 8495 return status; 8496} 8497 8498status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8499 uint32_t cmdSize, 8500 void *pCmdData, 8501 uint32_t *replySize, 8502 void *pReplyData) 8503{ 8504 Mutex::Autolock _l(mLock); 8505// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8506 8507 if (mState == DESTROYED || mEffectInterface == NULL) { 8508 return NO_INIT; 8509 } 8510 status_t status = (*mEffectInterface)->command(mEffectInterface, 8511 cmdCode, 8512 cmdSize, 8513 pCmdData, 8514 replySize, 8515 pReplyData); 8516 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8517 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8518 for (size_t i = 1; i < mHandles.size(); i++) { 8519 EffectHandle *h = mHandles[i]; 8520 if (h != NULL && !h->destroyed_l()) { 8521 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8522 } 8523 } 8524 } 8525 return status; 8526} 8527 8528status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8529{ 8530 Mutex::Autolock _l(mLock); 8531 return setEnabled_l(enabled); 8532} 8533 8534// must be called with EffectModule::mLock held 8535status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8536{ 8537 8538 ALOGV("setEnabled %p enabled %d", this, enabled); 8539 8540 if (enabled != isEnabled()) { 8541 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8542 if (enabled && status != NO_ERROR) { 8543 return status; 8544 } 8545 8546 switch (mState) { 8547 // going from disabled to enabled 8548 case IDLE: 8549 mState = STARTING; 8550 break; 8551 case STOPPED: 8552 mState = RESTART; 8553 break; 8554 case STOPPING: 8555 mState = ACTIVE; 8556 break; 8557 8558 // going from enabled to disabled 8559 case RESTART: 8560 mState = STOPPED; 8561 break; 8562 case STARTING: 8563 mState = IDLE; 8564 break; 8565 case ACTIVE: 8566 mState = STOPPING; 8567 break; 8568 case DESTROYED: 8569 return NO_ERROR; // simply ignore as we are being destroyed 8570 } 8571 for (size_t i = 1; i < mHandles.size(); i++) { 8572 EffectHandle *h = mHandles[i]; 8573 if (h != NULL && !h->destroyed_l()) { 8574 h->setEnabled(enabled); 8575 } 8576 } 8577 } 8578 return NO_ERROR; 8579} 8580 8581bool AudioFlinger::EffectModule::isEnabled() const 8582{ 8583 switch (mState) { 8584 case RESTART: 8585 case STARTING: 8586 case ACTIVE: 8587 return true; 8588 case IDLE: 8589 case STOPPING: 8590 case STOPPED: 8591 case DESTROYED: 8592 default: 8593 return false; 8594 } 8595} 8596 8597bool AudioFlinger::EffectModule::isProcessEnabled() const 8598{ 8599 switch (mState) { 8600 case RESTART: 8601 case ACTIVE: 8602 case STOPPING: 8603 case STOPPED: 8604 return true; 8605 case IDLE: 8606 case STARTING: 8607 case DESTROYED: 8608 default: 8609 return false; 8610 } 8611} 8612 8613status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8614{ 8615 Mutex::Autolock _l(mLock); 8616 status_t status = NO_ERROR; 8617 8618 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8619 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8620 if (isProcessEnabled() && 8621 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8622 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8623 status_t cmdStatus; 8624 uint32_t volume[2]; 8625 uint32_t *pVolume = NULL; 8626 uint32_t size = sizeof(volume); 8627 volume[0] = *left; 8628 volume[1] = *right; 8629 if (controller) { 8630 pVolume = volume; 8631 } 8632 status = (*mEffectInterface)->command(mEffectInterface, 8633 EFFECT_CMD_SET_VOLUME, 8634 size, 8635 volume, 8636 &size, 8637 pVolume); 8638 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8639 *left = volume[0]; 8640 *right = volume[1]; 8641 } 8642 } 8643 return status; 8644} 8645 8646status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8647{ 8648 Mutex::Autolock _l(mLock); 8649 status_t status = NO_ERROR; 8650 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8651 // audio pre processing modules on RecordThread can receive both output and 8652 // input device indication in the same call 8653 audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL; 8654 if (dev) { 8655 status_t cmdStatus; 8656 uint32_t size = sizeof(status_t); 8657 8658 status = (*mEffectInterface)->command(mEffectInterface, 8659 EFFECT_CMD_SET_DEVICE, 8660 sizeof(uint32_t), 8661 &dev, 8662 &size, 8663 &cmdStatus); 8664 if (status == NO_ERROR) { 8665 status = cmdStatus; 8666 } 8667 } 8668 dev = device & AUDIO_DEVICE_IN_ALL; 8669 if (dev) { 8670 status_t cmdStatus; 8671 uint32_t size = sizeof(status_t); 8672 8673 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8674 EFFECT_CMD_SET_INPUT_DEVICE, 8675 sizeof(uint32_t), 8676 &dev, 8677 &size, 8678 &cmdStatus); 8679 if (status2 == NO_ERROR) { 8680 status2 = cmdStatus; 8681 } 8682 if (status == NO_ERROR) { 8683 status = status2; 8684 } 8685 } 8686 } 8687 return status; 8688} 8689 8690status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8691{ 8692 Mutex::Autolock _l(mLock); 8693 status_t status = NO_ERROR; 8694 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8695 status_t cmdStatus; 8696 uint32_t size = sizeof(status_t); 8697 status = (*mEffectInterface)->command(mEffectInterface, 8698 EFFECT_CMD_SET_AUDIO_MODE, 8699 sizeof(audio_mode_t), 8700 &mode, 8701 &size, 8702 &cmdStatus); 8703 if (status == NO_ERROR) { 8704 status = cmdStatus; 8705 } 8706 } 8707 return status; 8708} 8709 8710void AudioFlinger::EffectModule::setSuspended(bool suspended) 8711{ 8712 Mutex::Autolock _l(mLock); 8713 mSuspended = suspended; 8714} 8715 8716bool AudioFlinger::EffectModule::suspended() const 8717{ 8718 Mutex::Autolock _l(mLock); 8719 return mSuspended; 8720} 8721 8722bool AudioFlinger::EffectModule::purgeHandles() 8723{ 8724 bool enabled = false; 8725 Mutex::Autolock _l(mLock); 8726 for (size_t i = 0; i < mHandles.size(); i++) { 8727 EffectHandle *handle = mHandles[i]; 8728 if (handle != NULL && !handle->destroyed_l()) { 8729 handle->effect().clear(); 8730 if (handle->hasControl()) { 8731 enabled = handle->enabled(); 8732 } 8733 } 8734 } 8735 return enabled; 8736} 8737 8738void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8739{ 8740 const size_t SIZE = 256; 8741 char buffer[SIZE]; 8742 String8 result; 8743 8744 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8745 result.append(buffer); 8746 8747 bool locked = tryLock(mLock); 8748 // failed to lock - AudioFlinger is probably deadlocked 8749 if (!locked) { 8750 result.append("\t\tCould not lock Fx mutex:\n"); 8751 } 8752 8753 result.append("\t\tSession Status State Engine:\n"); 8754 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8755 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8756 result.append(buffer); 8757 8758 result.append("\t\tDescriptor:\n"); 8759 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8760 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8761 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8762 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8763 result.append(buffer); 8764 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8765 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8766 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8767 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8768 result.append(buffer); 8769 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8770 mDescriptor.apiVersion, 8771 mDescriptor.flags); 8772 result.append(buffer); 8773 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8774 mDescriptor.name); 8775 result.append(buffer); 8776 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8777 mDescriptor.implementor); 8778 result.append(buffer); 8779 8780 result.append("\t\t- Input configuration:\n"); 8781 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8782 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8783 (uint32_t)mConfig.inputCfg.buffer.raw, 8784 mConfig.inputCfg.buffer.frameCount, 8785 mConfig.inputCfg.samplingRate, 8786 mConfig.inputCfg.channels, 8787 mConfig.inputCfg.format); 8788 result.append(buffer); 8789 8790 result.append("\t\t- Output configuration:\n"); 8791 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8792 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8793 (uint32_t)mConfig.outputCfg.buffer.raw, 8794 mConfig.outputCfg.buffer.frameCount, 8795 mConfig.outputCfg.samplingRate, 8796 mConfig.outputCfg.channels, 8797 mConfig.outputCfg.format); 8798 result.append(buffer); 8799 8800 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8801 result.append(buffer); 8802 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8803 for (size_t i = 0; i < mHandles.size(); ++i) { 8804 EffectHandle *handle = mHandles[i]; 8805 if (handle != NULL && !handle->destroyed_l()) { 8806 handle->dump(buffer, SIZE); 8807 result.append(buffer); 8808 } 8809 } 8810 8811 result.append("\n"); 8812 8813 write(fd, result.string(), result.length()); 8814 8815 if (locked) { 8816 mLock.unlock(); 8817 } 8818} 8819 8820// ---------------------------------------------------------------------------- 8821// EffectHandle implementation 8822// ---------------------------------------------------------------------------- 8823 8824#undef LOG_TAG 8825#define LOG_TAG "AudioFlinger::EffectHandle" 8826 8827AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8828 const sp<AudioFlinger::Client>& client, 8829 const sp<IEffectClient>& effectClient, 8830 int32_t priority) 8831 : BnEffect(), 8832 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8833 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8834{ 8835 ALOGV("constructor %p", this); 8836 8837 if (client == 0) { 8838 return; 8839 } 8840 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8841 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8842 if (mCblkMemory != 0) { 8843 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8844 8845 if (mCblk != NULL) { 8846 new(mCblk) effect_param_cblk_t(); 8847 mBuffer = (uint8_t *)mCblk + bufOffset; 8848 } 8849 } else { 8850 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8851 return; 8852 } 8853} 8854 8855AudioFlinger::EffectHandle::~EffectHandle() 8856{ 8857 ALOGV("Destructor %p", this); 8858 8859 if (mEffect == 0) { 8860 mDestroyed = true; 8861 return; 8862 } 8863 mEffect->lock(); 8864 mDestroyed = true; 8865 mEffect->unlock(); 8866 disconnect(false); 8867} 8868 8869status_t AudioFlinger::EffectHandle::enable() 8870{ 8871 ALOGV("enable %p", this); 8872 if (!mHasControl) return INVALID_OPERATION; 8873 if (mEffect == 0) return DEAD_OBJECT; 8874 8875 if (mEnabled) { 8876 return NO_ERROR; 8877 } 8878 8879 mEnabled = true; 8880 8881 sp<ThreadBase> thread = mEffect->thread().promote(); 8882 if (thread != 0) { 8883 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8884 } 8885 8886 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8887 if (mEffect->suspended()) { 8888 return NO_ERROR; 8889 } 8890 8891 status_t status = mEffect->setEnabled(true); 8892 if (status != NO_ERROR) { 8893 if (thread != 0) { 8894 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8895 } 8896 mEnabled = false; 8897 } 8898 return status; 8899} 8900 8901status_t AudioFlinger::EffectHandle::disable() 8902{ 8903 ALOGV("disable %p", this); 8904 if (!mHasControl) return INVALID_OPERATION; 8905 if (mEffect == 0) return DEAD_OBJECT; 8906 8907 if (!mEnabled) { 8908 return NO_ERROR; 8909 } 8910 mEnabled = false; 8911 8912 if (mEffect->suspended()) { 8913 return NO_ERROR; 8914 } 8915 8916 status_t status = mEffect->setEnabled(false); 8917 8918 sp<ThreadBase> thread = mEffect->thread().promote(); 8919 if (thread != 0) { 8920 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8921 } 8922 8923 return status; 8924} 8925 8926void AudioFlinger::EffectHandle::disconnect() 8927{ 8928 disconnect(true); 8929} 8930 8931void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8932{ 8933 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8934 if (mEffect == 0) { 8935 return; 8936 } 8937 // restore suspended effects if the disconnected handle was enabled and the last one. 8938 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8939 sp<ThreadBase> thread = mEffect->thread().promote(); 8940 if (thread != 0) { 8941 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8942 } 8943 } 8944 8945 // release sp on module => module destructor can be called now 8946 mEffect.clear(); 8947 if (mClient != 0) { 8948 if (mCblk != NULL) { 8949 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8950 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8951 } 8952 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8953 // Client destructor must run with AudioFlinger mutex locked 8954 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8955 mClient.clear(); 8956 } 8957} 8958 8959status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8960 uint32_t cmdSize, 8961 void *pCmdData, 8962 uint32_t *replySize, 8963 void *pReplyData) 8964{ 8965// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8966// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8967 8968 // only get parameter command is permitted for applications not controlling the effect 8969 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8970 return INVALID_OPERATION; 8971 } 8972 if (mEffect == 0) return DEAD_OBJECT; 8973 if (mClient == 0) return INVALID_OPERATION; 8974 8975 // handle commands that are not forwarded transparently to effect engine 8976 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8977 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8978 // no risk to block the whole media server process or mixer threads is we are stuck here 8979 Mutex::Autolock _l(mCblk->lock); 8980 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8981 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8982 mCblk->serverIndex = 0; 8983 mCblk->clientIndex = 0; 8984 return BAD_VALUE; 8985 } 8986 status_t status = NO_ERROR; 8987 while (mCblk->serverIndex < mCblk->clientIndex) { 8988 int reply; 8989 uint32_t rsize = sizeof(int); 8990 int *p = (int *)(mBuffer + mCblk->serverIndex); 8991 int size = *p++; 8992 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8993 ALOGW("command(): invalid parameter block size"); 8994 break; 8995 } 8996 effect_param_t *param = (effect_param_t *)p; 8997 if (param->psize == 0 || param->vsize == 0) { 8998 ALOGW("command(): null parameter or value size"); 8999 mCblk->serverIndex += size; 9000 continue; 9001 } 9002 uint32_t psize = sizeof(effect_param_t) + 9003 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9004 param->vsize; 9005 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9006 psize, 9007 p, 9008 &rsize, 9009 &reply); 9010 // stop at first error encountered 9011 if (ret != NO_ERROR) { 9012 status = ret; 9013 *(int *)pReplyData = reply; 9014 break; 9015 } else if (reply != NO_ERROR) { 9016 *(int *)pReplyData = reply; 9017 break; 9018 } 9019 mCblk->serverIndex += size; 9020 } 9021 mCblk->serverIndex = 0; 9022 mCblk->clientIndex = 0; 9023 return status; 9024 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9025 *(int *)pReplyData = NO_ERROR; 9026 return enable(); 9027 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9028 *(int *)pReplyData = NO_ERROR; 9029 return disable(); 9030 } 9031 9032 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9033} 9034 9035void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9036{ 9037 ALOGV("setControl %p control %d", this, hasControl); 9038 9039 mHasControl = hasControl; 9040 mEnabled = enabled; 9041 9042 if (signal && mEffectClient != 0) { 9043 mEffectClient->controlStatusChanged(hasControl); 9044 } 9045} 9046 9047void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9048 uint32_t cmdSize, 9049 void *pCmdData, 9050 uint32_t replySize, 9051 void *pReplyData) 9052{ 9053 if (mEffectClient != 0) { 9054 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9055 } 9056} 9057 9058 9059 9060void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9061{ 9062 if (mEffectClient != 0) { 9063 mEffectClient->enableStatusChanged(enabled); 9064 } 9065} 9066 9067status_t AudioFlinger::EffectHandle::onTransact( 9068 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9069{ 9070 return BnEffect::onTransact(code, data, reply, flags); 9071} 9072 9073 9074void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9075{ 9076 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9077 9078 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9079 (mClient == 0) ? getpid_cached : mClient->pid(), 9080 mPriority, 9081 mHasControl, 9082 !locked, 9083 mCblk ? mCblk->clientIndex : 0, 9084 mCblk ? mCblk->serverIndex : 0 9085 ); 9086 9087 if (locked) { 9088 mCblk->lock.unlock(); 9089 } 9090} 9091 9092#undef LOG_TAG 9093#define LOG_TAG "AudioFlinger::EffectChain" 9094 9095AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9096 int sessionId) 9097 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9098 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9099 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9100{ 9101 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9102 if (thread == NULL) { 9103 return; 9104 } 9105 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9106 thread->frameCount(); 9107} 9108 9109AudioFlinger::EffectChain::~EffectChain() 9110{ 9111 if (mOwnInBuffer) { 9112 delete mInBuffer; 9113 } 9114 9115} 9116 9117// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9118sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9119{ 9120 size_t size = mEffects.size(); 9121 9122 for (size_t i = 0; i < size; i++) { 9123 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9124 return mEffects[i]; 9125 } 9126 } 9127 return 0; 9128} 9129 9130// getEffectFromId_l() must be called with ThreadBase::mLock held 9131sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9132{ 9133 size_t size = mEffects.size(); 9134 9135 for (size_t i = 0; i < size; i++) { 9136 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9137 if (id == 0 || mEffects[i]->id() == id) { 9138 return mEffects[i]; 9139 } 9140 } 9141 return 0; 9142} 9143 9144// getEffectFromType_l() must be called with ThreadBase::mLock held 9145sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9146 const effect_uuid_t *type) 9147{ 9148 size_t size = mEffects.size(); 9149 9150 for (size_t i = 0; i < size; i++) { 9151 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9152 return mEffects[i]; 9153 } 9154 } 9155 return 0; 9156} 9157 9158void AudioFlinger::EffectChain::clearInputBuffer() 9159{ 9160 Mutex::Autolock _l(mLock); 9161 sp<ThreadBase> thread = mThread.promote(); 9162 if (thread == 0) { 9163 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9164 return; 9165 } 9166 clearInputBuffer_l(thread); 9167} 9168 9169// Must be called with EffectChain::mLock locked 9170void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9171{ 9172 size_t numSamples = thread->frameCount() * thread->channelCount(); 9173 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9174 9175} 9176 9177// Must be called with EffectChain::mLock locked 9178void AudioFlinger::EffectChain::process_l() 9179{ 9180 sp<ThreadBase> thread = mThread.promote(); 9181 if (thread == 0) { 9182 ALOGW("process_l(): cannot promote mixer thread"); 9183 return; 9184 } 9185 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9186 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9187 // always process effects unless no more tracks are on the session and the effect tail 9188 // has been rendered 9189 bool doProcess = true; 9190 if (!isGlobalSession) { 9191 bool tracksOnSession = (trackCnt() != 0); 9192 9193 if (!tracksOnSession && mTailBufferCount == 0) { 9194 doProcess = false; 9195 } 9196 9197 if (activeTrackCnt() == 0) { 9198 // if no track is active and the effect tail has not been rendered, 9199 // the input buffer must be cleared here as the mixer process will not do it 9200 if (tracksOnSession || mTailBufferCount > 0) { 9201 clearInputBuffer_l(thread); 9202 if (mTailBufferCount > 0) { 9203 mTailBufferCount--; 9204 } 9205 } 9206 } 9207 } 9208 9209 size_t size = mEffects.size(); 9210 if (doProcess) { 9211 for (size_t i = 0; i < size; i++) { 9212 mEffects[i]->process(); 9213 } 9214 } 9215 for (size_t i = 0; i < size; i++) { 9216 mEffects[i]->updateState(); 9217 } 9218} 9219 9220// addEffect_l() must be called with PlaybackThread::mLock held 9221status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9222{ 9223 effect_descriptor_t desc = effect->desc(); 9224 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9225 9226 Mutex::Autolock _l(mLock); 9227 effect->setChain(this); 9228 sp<ThreadBase> thread = mThread.promote(); 9229 if (thread == 0) { 9230 return NO_INIT; 9231 } 9232 effect->setThread(thread); 9233 9234 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9235 // Auxiliary effects are inserted at the beginning of mEffects vector as 9236 // they are processed first and accumulated in chain input buffer 9237 mEffects.insertAt(effect, 0); 9238 9239 // the input buffer for auxiliary effect contains mono samples in 9240 // 32 bit format. This is to avoid saturation in AudoMixer 9241 // accumulation stage. Saturation is done in EffectModule::process() before 9242 // calling the process in effect engine 9243 size_t numSamples = thread->frameCount(); 9244 int32_t *buffer = new int32_t[numSamples]; 9245 memset(buffer, 0, numSamples * sizeof(int32_t)); 9246 effect->setInBuffer((int16_t *)buffer); 9247 // auxiliary effects output samples to chain input buffer for further processing 9248 // by insert effects 9249 effect->setOutBuffer(mInBuffer); 9250 } else { 9251 // Insert effects are inserted at the end of mEffects vector as they are processed 9252 // after track and auxiliary effects. 9253 // Insert effect order as a function of indicated preference: 9254 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9255 // another effect is present 9256 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9257 // last effect claiming first position 9258 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9259 // first effect claiming last position 9260 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9261 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9262 // already present 9263 9264 size_t size = mEffects.size(); 9265 size_t idx_insert = size; 9266 ssize_t idx_insert_first = -1; 9267 ssize_t idx_insert_last = -1; 9268 9269 for (size_t i = 0; i < size; i++) { 9270 effect_descriptor_t d = mEffects[i]->desc(); 9271 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9272 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9273 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9274 // check invalid effect chaining combinations 9275 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9276 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9277 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9278 return INVALID_OPERATION; 9279 } 9280 // remember position of first insert effect and by default 9281 // select this as insert position for new effect 9282 if (idx_insert == size) { 9283 idx_insert = i; 9284 } 9285 // remember position of last insert effect claiming 9286 // first position 9287 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9288 idx_insert_first = i; 9289 } 9290 // remember position of first insert effect claiming 9291 // last position 9292 if (iPref == EFFECT_FLAG_INSERT_LAST && 9293 idx_insert_last == -1) { 9294 idx_insert_last = i; 9295 } 9296 } 9297 } 9298 9299 // modify idx_insert from first position if needed 9300 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9301 if (idx_insert_last != -1) { 9302 idx_insert = idx_insert_last; 9303 } else { 9304 idx_insert = size; 9305 } 9306 } else { 9307 if (idx_insert_first != -1) { 9308 idx_insert = idx_insert_first + 1; 9309 } 9310 } 9311 9312 // always read samples from chain input buffer 9313 effect->setInBuffer(mInBuffer); 9314 9315 // if last effect in the chain, output samples to chain 9316 // output buffer, otherwise to chain input buffer 9317 if (idx_insert == size) { 9318 if (idx_insert != 0) { 9319 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9320 mEffects[idx_insert-1]->configure(); 9321 } 9322 effect->setOutBuffer(mOutBuffer); 9323 } else { 9324 effect->setOutBuffer(mInBuffer); 9325 } 9326 mEffects.insertAt(effect, idx_insert); 9327 9328 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9329 } 9330 effect->configure(); 9331 return NO_ERROR; 9332} 9333 9334// removeEffect_l() must be called with PlaybackThread::mLock held 9335size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9336{ 9337 Mutex::Autolock _l(mLock); 9338 size_t size = mEffects.size(); 9339 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9340 9341 for (size_t i = 0; i < size; i++) { 9342 if (effect == mEffects[i]) { 9343 // calling stop here will remove pre-processing effect from the audio HAL. 9344 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9345 // the middle of a read from audio HAL 9346 if (mEffects[i]->state() == EffectModule::ACTIVE || 9347 mEffects[i]->state() == EffectModule::STOPPING) { 9348 mEffects[i]->stop(); 9349 } 9350 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9351 delete[] effect->inBuffer(); 9352 } else { 9353 if (i == size - 1 && i != 0) { 9354 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9355 mEffects[i - 1]->configure(); 9356 } 9357 } 9358 mEffects.removeAt(i); 9359 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9360 break; 9361 } 9362 } 9363 9364 return mEffects.size(); 9365} 9366 9367// setDevice_l() must be called with PlaybackThread::mLock held 9368void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9369{ 9370 size_t size = mEffects.size(); 9371 for (size_t i = 0; i < size; i++) { 9372 mEffects[i]->setDevice(device); 9373 } 9374} 9375 9376// setMode_l() must be called with PlaybackThread::mLock held 9377void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9378{ 9379 size_t size = mEffects.size(); 9380 for (size_t i = 0; i < size; i++) { 9381 mEffects[i]->setMode(mode); 9382 } 9383} 9384 9385// setVolume_l() must be called with PlaybackThread::mLock held 9386bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9387{ 9388 uint32_t newLeft = *left; 9389 uint32_t newRight = *right; 9390 bool hasControl = false; 9391 int ctrlIdx = -1; 9392 size_t size = mEffects.size(); 9393 9394 // first update volume controller 9395 for (size_t i = size; i > 0; i--) { 9396 if (mEffects[i - 1]->isProcessEnabled() && 9397 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9398 ctrlIdx = i - 1; 9399 hasControl = true; 9400 break; 9401 } 9402 } 9403 9404 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9405 if (hasControl) { 9406 *left = mNewLeftVolume; 9407 *right = mNewRightVolume; 9408 } 9409 return hasControl; 9410 } 9411 9412 mVolumeCtrlIdx = ctrlIdx; 9413 mLeftVolume = newLeft; 9414 mRightVolume = newRight; 9415 9416 // second get volume update from volume controller 9417 if (ctrlIdx >= 0) { 9418 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9419 mNewLeftVolume = newLeft; 9420 mNewRightVolume = newRight; 9421 } 9422 // then indicate volume to all other effects in chain. 9423 // Pass altered volume to effects before volume controller 9424 // and requested volume to effects after controller 9425 uint32_t lVol = newLeft; 9426 uint32_t rVol = newRight; 9427 9428 for (size_t i = 0; i < size; i++) { 9429 if ((int)i == ctrlIdx) continue; 9430 // this also works for ctrlIdx == -1 when there is no volume controller 9431 if ((int)i > ctrlIdx) { 9432 lVol = *left; 9433 rVol = *right; 9434 } 9435 mEffects[i]->setVolume(&lVol, &rVol, false); 9436 } 9437 *left = newLeft; 9438 *right = newRight; 9439 9440 return hasControl; 9441} 9442 9443void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9444{ 9445 const size_t SIZE = 256; 9446 char buffer[SIZE]; 9447 String8 result; 9448 9449 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9450 result.append(buffer); 9451 9452 bool locked = tryLock(mLock); 9453 // failed to lock - AudioFlinger is probably deadlocked 9454 if (!locked) { 9455 result.append("\tCould not lock mutex:\n"); 9456 } 9457 9458 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9459 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9460 mEffects.size(), 9461 (uint32_t)mInBuffer, 9462 (uint32_t)mOutBuffer, 9463 mActiveTrackCnt); 9464 result.append(buffer); 9465 write(fd, result.string(), result.size()); 9466 9467 for (size_t i = 0; i < mEffects.size(); ++i) { 9468 sp<EffectModule> effect = mEffects[i]; 9469 if (effect != 0) { 9470 effect->dump(fd, args); 9471 } 9472 } 9473 9474 if (locked) { 9475 mLock.unlock(); 9476 } 9477} 9478 9479// must be called with ThreadBase::mLock held 9480void AudioFlinger::EffectChain::setEffectSuspended_l( 9481 const effect_uuid_t *type, bool suspend) 9482{ 9483 sp<SuspendedEffectDesc> desc; 9484 // use effect type UUID timelow as key as there is no real risk of identical 9485 // timeLow fields among effect type UUIDs. 9486 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9487 if (suspend) { 9488 if (index >= 0) { 9489 desc = mSuspendedEffects.valueAt(index); 9490 } else { 9491 desc = new SuspendedEffectDesc(); 9492 desc->mType = *type; 9493 mSuspendedEffects.add(type->timeLow, desc); 9494 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9495 } 9496 if (desc->mRefCount++ == 0) { 9497 sp<EffectModule> effect = getEffectIfEnabled(type); 9498 if (effect != 0) { 9499 desc->mEffect = effect; 9500 effect->setSuspended(true); 9501 effect->setEnabled(false); 9502 } 9503 } 9504 } else { 9505 if (index < 0) { 9506 return; 9507 } 9508 desc = mSuspendedEffects.valueAt(index); 9509 if (desc->mRefCount <= 0) { 9510 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9511 desc->mRefCount = 1; 9512 } 9513 if (--desc->mRefCount == 0) { 9514 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9515 if (desc->mEffect != 0) { 9516 sp<EffectModule> effect = desc->mEffect.promote(); 9517 if (effect != 0) { 9518 effect->setSuspended(false); 9519 effect->lock(); 9520 EffectHandle *handle = effect->controlHandle_l(); 9521 if (handle != NULL && !handle->destroyed_l()) { 9522 effect->setEnabled_l(handle->enabled()); 9523 } 9524 effect->unlock(); 9525 } 9526 desc->mEffect.clear(); 9527 } 9528 mSuspendedEffects.removeItemsAt(index); 9529 } 9530 } 9531} 9532 9533// must be called with ThreadBase::mLock held 9534void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9535{ 9536 sp<SuspendedEffectDesc> desc; 9537 9538 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9539 if (suspend) { 9540 if (index >= 0) { 9541 desc = mSuspendedEffects.valueAt(index); 9542 } else { 9543 desc = new SuspendedEffectDesc(); 9544 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9545 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9546 } 9547 if (desc->mRefCount++ == 0) { 9548 Vector< sp<EffectModule> > effects; 9549 getSuspendEligibleEffects(effects); 9550 for (size_t i = 0; i < effects.size(); i++) { 9551 setEffectSuspended_l(&effects[i]->desc().type, true); 9552 } 9553 } 9554 } else { 9555 if (index < 0) { 9556 return; 9557 } 9558 desc = mSuspendedEffects.valueAt(index); 9559 if (desc->mRefCount <= 0) { 9560 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9561 desc->mRefCount = 1; 9562 } 9563 if (--desc->mRefCount == 0) { 9564 Vector<const effect_uuid_t *> types; 9565 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9566 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9567 continue; 9568 } 9569 types.add(&mSuspendedEffects.valueAt(i)->mType); 9570 } 9571 for (size_t i = 0; i < types.size(); i++) { 9572 setEffectSuspended_l(types[i], false); 9573 } 9574 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9575 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9576 } 9577 } 9578} 9579 9580 9581// The volume effect is used for automated tests only 9582#ifndef OPENSL_ES_H_ 9583static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9584 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9585const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9586#endif //OPENSL_ES_H_ 9587 9588bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9589{ 9590 // auxiliary effects and visualizer are never suspended on output mix 9591 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9592 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9593 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9594 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9595 return false; 9596 } 9597 return true; 9598} 9599 9600void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9601{ 9602 effects.clear(); 9603 for (size_t i = 0; i < mEffects.size(); i++) { 9604 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9605 effects.add(mEffects[i]); 9606 } 9607 } 9608} 9609 9610sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9611 const effect_uuid_t *type) 9612{ 9613 sp<EffectModule> effect = getEffectFromType_l(type); 9614 return effect != 0 && effect->isEnabled() ? effect : 0; 9615} 9616 9617void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9618 bool enabled) 9619{ 9620 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9621 if (enabled) { 9622 if (index < 0) { 9623 // if the effect is not suspend check if all effects are suspended 9624 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9625 if (index < 0) { 9626 return; 9627 } 9628 if (!isEffectEligibleForSuspend(effect->desc())) { 9629 return; 9630 } 9631 setEffectSuspended_l(&effect->desc().type, enabled); 9632 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9633 if (index < 0) { 9634 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9635 return; 9636 } 9637 } 9638 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9639 effect->desc().type.timeLow); 9640 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9641 // if effect is requested to suspended but was not yet enabled, supend it now. 9642 if (desc->mEffect == 0) { 9643 desc->mEffect = effect; 9644 effect->setEnabled(false); 9645 effect->setSuspended(true); 9646 } 9647 } else { 9648 if (index < 0) { 9649 return; 9650 } 9651 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9652 effect->desc().type.timeLow); 9653 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9654 desc->mEffect.clear(); 9655 effect->setSuspended(false); 9656 } 9657} 9658 9659#undef LOG_TAG 9660#define LOG_TAG "AudioFlinger" 9661 9662// ---------------------------------------------------------------------------- 9663 9664status_t AudioFlinger::onTransact( 9665 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9666{ 9667 return BnAudioFlinger::onTransact(code, data, reply, flags); 9668} 9669 9670}; // namespace android 9671