AudioFlinger.cpp revision 77035d10a740914313500811b31a90ab948bd267
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include <media/nbaio/AudioStreamOutSink.h> 80#include <media/nbaio/MonoPipe.h> 81#include <media/nbaio/MonoPipeReader.h> 82#include <media/nbaio/Pipe.h> 83#include <media/nbaio/PipeReader.h> 84#include <media/nbaio/SourceAudioBufferProvider.h> 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 169// for the track. The client then sub-divides this into smaller buffers for its use. 170// Currently the client uses double-buffering by default, but doesn't tell us about that. 171// So for now we just assume that client is double-buffered. 172// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 173// N-buffering, so AudioFlinger could allocate the right amount of memory. 174// See the client's minBufCount and mNotificationFramesAct calculations for details. 175static const int kFastTrackMultiplier = 2; 176 177// ---------------------------------------------------------------------------- 178 179#ifdef ADD_BATTERY_DATA 180// To collect the amplifier usage 181static void addBatteryData(uint32_t params) { 182 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 183 if (service == NULL) { 184 // it already logged 185 return; 186 } 187 188 service->addBatteryData(params); 189} 190#endif 191 192static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 193{ 194 const hw_module_t *mod; 195 int rc; 196 197 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 198 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 199 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 200 if (rc) { 201 goto out; 202 } 203 rc = audio_hw_device_open(mod, dev); 204 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 205 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 206 if (rc) { 207 goto out; 208 } 209 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 210 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 211 rc = BAD_VALUE; 212 goto out; 213 } 214 return 0; 215 216out: 217 *dev = NULL; 218 return rc; 219} 220 221// ---------------------------------------------------------------------------- 222 223AudioFlinger::AudioFlinger() 224 : BnAudioFlinger(), 225 mPrimaryHardwareDev(NULL), 226 mHardwareStatus(AUDIO_HW_IDLE), 227 mMasterVolume(1.0f), 228 mMasterMute(false), 229 mNextUniqueId(1), 230 mMode(AUDIO_MODE_INVALID), 231 mBtNrecIsOff(false) 232{ 233} 234 235void AudioFlinger::onFirstRef() 236{ 237 int rc = 0; 238 239 Mutex::Autolock _l(mLock); 240 241 /* TODO: move all this work into an Init() function */ 242 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 243 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 244 uint32_t int_val; 245 if (1 == sscanf(val_str, "%u", &int_val)) { 246 mStandbyTimeInNsecs = milliseconds(int_val); 247 ALOGI("Using %u mSec as standby time.", int_val); 248 } else { 249 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 250 ALOGI("Using default %u mSec as standby time.", 251 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 252 } 253 } 254 255 mMode = AUDIO_MODE_NORMAL; 256} 257 258AudioFlinger::~AudioFlinger() 259{ 260 while (!mRecordThreads.isEmpty()) { 261 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 262 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 263 } 264 while (!mPlaybackThreads.isEmpty()) { 265 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 266 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 267 } 268 269 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 270 // no mHardwareLock needed, as there are no other references to this 271 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 272 delete mAudioHwDevs.valueAt(i); 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367static bool tryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = tryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = tryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 dumpClients(fd, args); 403 dumpInternals(fd, args); 404 405 // dump playback threads 406 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 407 mPlaybackThreads.valueAt(i)->dump(fd, args); 408 } 409 410 // dump record threads 411 for (size_t i = 0; i < mRecordThreads.size(); i++) { 412 mRecordThreads.valueAt(i)->dump(fd, args); 413 } 414 415 // dump all hardware devs 416 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 417 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 418 dev->dump(dev, fd); 419 } 420 421 // dump the serially shared record tee sink 422 if (mRecordTeeSource != 0) { 423 dumpTee(fd, mRecordTeeSource); 424 } 425 426 if (locked) mLock.unlock(); 427 } 428 return NO_ERROR; 429} 430 431sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 432{ 433 // If pid is already in the mClients wp<> map, then use that entry 434 // (for which promote() is always != 0), otherwise create a new entry and Client. 435 sp<Client> client = mClients.valueFor(pid).promote(); 436 if (client == 0) { 437 client = new Client(this, pid); 438 mClients.add(pid, client); 439 } 440 441 return client; 442} 443 444// IAudioFlinger interface 445 446 447sp<IAudioTrack> AudioFlinger::createTrack( 448 pid_t pid, 449 audio_stream_type_t streamType, 450 uint32_t sampleRate, 451 audio_format_t format, 452 audio_channel_mask_t channelMask, 453 size_t frameCount, 454 IAudioFlinger::track_flags_t *flags, 455 const sp<IMemory>& sharedBuffer, 456 audio_io_handle_t output, 457 pid_t tid, 458 int *sessionId, 459 status_t *status) 460{ 461 sp<PlaybackThread::Track> track; 462 sp<TrackHandle> trackHandle; 463 sp<Client> client; 464 status_t lStatus; 465 int lSessionId; 466 467 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 468 // but if someone uses binder directly they could bypass that and cause us to crash 469 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 470 ALOGE("createTrack() invalid stream type %d", streamType); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 476 // and we don't yet support 8.24 or 32-bit PCM 477 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 478 ALOGE("createTrack() invalid format %d", format); 479 lStatus = BAD_VALUE; 480 goto Exit; 481 } 482 483 { 484 Mutex::Autolock _l(mLock); 485 PlaybackThread *thread = checkPlaybackThread_l(output); 486 PlaybackThread *effectThread = NULL; 487 if (thread == NULL) { 488 ALOGE("unknown output thread"); 489 lStatus = BAD_VALUE; 490 goto Exit; 491 } 492 493 client = registerPid_l(pid); 494 495 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 496 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 497 // check if an effect chain with the same session ID is present on another 498 // output thread and move it here. 499 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 500 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 501 if (mPlaybackThreads.keyAt(i) != output) { 502 uint32_t sessions = t->hasAudioSession(*sessionId); 503 if (sessions & PlaybackThread::EFFECT_SESSION) { 504 effectThread = t.get(); 505 break; 506 } 507 } 508 } 509 lSessionId = *sessionId; 510 } else { 511 // if no audio session id is provided, create one here 512 lSessionId = nextUniqueId(); 513 if (sessionId != NULL) { 514 *sessionId = lSessionId; 515 } 516 } 517 ALOGV("createTrack() lSessionId: %d", lSessionId); 518 519 track = thread->createTrack_l(client, streamType, sampleRate, format, 520 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 521 522 // move effect chain to this output thread if an effect on same session was waiting 523 // for a track to be created 524 if (lStatus == NO_ERROR && effectThread != NULL) { 525 Mutex::Autolock _dl(thread->mLock); 526 Mutex::Autolock _sl(effectThread->mLock); 527 moveEffectChain_l(lSessionId, effectThread, thread, true); 528 } 529 530 // Look for sync events awaiting for a session to be used. 531 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 532 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 533 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 534 if (lStatus == NO_ERROR) { 535 (void) track->setSyncEvent(mPendingSyncEvents[i]); 536 } else { 537 mPendingSyncEvents[i]->cancel(); 538 } 539 mPendingSyncEvents.removeAt(i); 540 i--; 541 } 542 } 543 } 544 } 545 if (lStatus == NO_ERROR) { 546 trackHandle = new TrackHandle(track); 547 } else { 548 // remove local strong reference to Client before deleting the Track so that the Client 549 // destructor is called by the TrackBase destructor with mLock held 550 client.clear(); 551 track.clear(); 552 } 553 554Exit: 555 if (status != NULL) { 556 *status = lStatus; 557 } 558 return trackHandle; 559} 560 561uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 562{ 563 Mutex::Autolock _l(mLock); 564 PlaybackThread *thread = checkPlaybackThread_l(output); 565 if (thread == NULL) { 566 ALOGW("sampleRate() unknown thread %d", output); 567 return 0; 568 } 569 return thread->sampleRate(); 570} 571 572int AudioFlinger::channelCount(audio_io_handle_t output) const 573{ 574 Mutex::Autolock _l(mLock); 575 PlaybackThread *thread = checkPlaybackThread_l(output); 576 if (thread == NULL) { 577 ALOGW("channelCount() unknown thread %d", output); 578 return 0; 579 } 580 return thread->channelCount(); 581} 582 583audio_format_t AudioFlinger::format(audio_io_handle_t output) const 584{ 585 Mutex::Autolock _l(mLock); 586 PlaybackThread *thread = checkPlaybackThread_l(output); 587 if (thread == NULL) { 588 ALOGW("format() unknown thread %d", output); 589 return AUDIO_FORMAT_INVALID; 590 } 591 return thread->format(); 592} 593 594size_t AudioFlinger::frameCount(audio_io_handle_t output) const 595{ 596 Mutex::Autolock _l(mLock); 597 PlaybackThread *thread = checkPlaybackThread_l(output); 598 if (thread == NULL) { 599 ALOGW("frameCount() unknown thread %d", output); 600 return 0; 601 } 602 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 603 // should examine all callers and fix them to handle smaller counts 604 return thread->frameCount(); 605} 606 607uint32_t AudioFlinger::latency(audio_io_handle_t output) const 608{ 609 Mutex::Autolock _l(mLock); 610 PlaybackThread *thread = checkPlaybackThread_l(output); 611 if (thread == NULL) { 612 ALOGW("latency() unknown thread %d", output); 613 return 0; 614 } 615 return thread->latency(); 616} 617 618status_t AudioFlinger::setMasterVolume(float value) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 630 Mutex::Autolock _l(mLock); 631 mMasterVolume = value; 632 633 // Set master volume in the HALs which support it. 634 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 635 AutoMutex lock(mHardwareLock); 636 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 637 638 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 639 if (dev->canSetMasterVolume()) { 640 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 641 } 642 mHardwareStatus = AUDIO_HW_IDLE; 643 } 644 645 // Now set the master volume in each playback thread. Playback threads 646 // assigned to HALs which do not have master volume support will apply 647 // master volume during the mix operation. Threads with HALs which do 648 // support master volume will simply ignore the setting. 649 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 650 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 651 652 return NO_ERROR; 653} 654 655status_t AudioFlinger::setMode(audio_mode_t mode) 656{ 657 status_t ret = initCheck(); 658 if (ret != NO_ERROR) { 659 return ret; 660 } 661 662 // check calling permissions 663 if (!settingsAllowed()) { 664 return PERMISSION_DENIED; 665 } 666 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 667 ALOGW("Illegal value: setMode(%d)", mode); 668 return BAD_VALUE; 669 } 670 671 { // scope for the lock 672 AutoMutex lock(mHardwareLock); 673 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 674 mHardwareStatus = AUDIO_HW_SET_MODE; 675 ret = dev->set_mode(dev, mode); 676 mHardwareStatus = AUDIO_HW_IDLE; 677 } 678 679 if (NO_ERROR == ret) { 680 Mutex::Autolock _l(mLock); 681 mMode = mode; 682 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 683 mPlaybackThreads.valueAt(i)->setMode(mode); 684 } 685 686 return ret; 687} 688 689status_t AudioFlinger::setMicMute(bool state) 690{ 691 status_t ret = initCheck(); 692 if (ret != NO_ERROR) { 693 return ret; 694 } 695 696 // check calling permissions 697 if (!settingsAllowed()) { 698 return PERMISSION_DENIED; 699 } 700 701 AutoMutex lock(mHardwareLock); 702 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 703 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 704 ret = dev->set_mic_mute(dev, state); 705 mHardwareStatus = AUDIO_HW_IDLE; 706 return ret; 707} 708 709bool AudioFlinger::getMicMute() const 710{ 711 status_t ret = initCheck(); 712 if (ret != NO_ERROR) { 713 return false; 714 } 715 716 bool state = AUDIO_MODE_INVALID; 717 AutoMutex lock(mHardwareLock); 718 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 719 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 720 dev->get_mic_mute(dev, &state); 721 mHardwareStatus = AUDIO_HW_IDLE; 722 return state; 723} 724 725status_t AudioFlinger::setMasterMute(bool muted) 726{ 727 status_t ret = initCheck(); 728 if (ret != NO_ERROR) { 729 return ret; 730 } 731 732 // check calling permissions 733 if (!settingsAllowed()) { 734 return PERMISSION_DENIED; 735 } 736 737 Mutex::Autolock _l(mLock); 738 mMasterMute = muted; 739 740 // Set master mute in the HALs which support it. 741 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 742 AutoMutex lock(mHardwareLock); 743 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 744 745 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 746 if (dev->canSetMasterMute()) { 747 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 748 } 749 mHardwareStatus = AUDIO_HW_IDLE; 750 } 751 752 // Now set the master mute in each playback thread. Playback threads 753 // assigned to HALs which do not have master mute support will apply master 754 // mute during the mix operation. Threads with HALs which do support master 755 // mute will simply ignore the setting. 756 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 757 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 758 759 return NO_ERROR; 760} 761 762float AudioFlinger::masterVolume() const 763{ 764 Mutex::Autolock _l(mLock); 765 return masterVolume_l(); 766} 767 768bool AudioFlinger::masterMute() const 769{ 770 Mutex::Autolock _l(mLock); 771 return masterMute_l(); 772} 773 774float AudioFlinger::masterVolume_l() const 775{ 776 return mMasterVolume; 777} 778 779bool AudioFlinger::masterMute_l() const 780{ 781 return mMasterMute; 782} 783 784status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 785 audio_io_handle_t output) 786{ 787 // check calling permissions 788 if (!settingsAllowed()) { 789 return PERMISSION_DENIED; 790 } 791 792 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 793 ALOGE("setStreamVolume() invalid stream %d", stream); 794 return BAD_VALUE; 795 } 796 797 AutoMutex lock(mLock); 798 PlaybackThread *thread = NULL; 799 if (output) { 800 thread = checkPlaybackThread_l(output); 801 if (thread == NULL) { 802 return BAD_VALUE; 803 } 804 } 805 806 mStreamTypes[stream].volume = value; 807 808 if (thread == NULL) { 809 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 810 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 811 } 812 } else { 813 thread->setStreamVolume(stream, value); 814 } 815 816 return NO_ERROR; 817} 818 819status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 820{ 821 // check calling permissions 822 if (!settingsAllowed()) { 823 return PERMISSION_DENIED; 824 } 825 826 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 827 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 828 ALOGE("setStreamMute() invalid stream %d", stream); 829 return BAD_VALUE; 830 } 831 832 AutoMutex lock(mLock); 833 mStreamTypes[stream].mute = muted; 834 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 835 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 836 837 return NO_ERROR; 838} 839 840float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 841{ 842 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 843 return 0.0f; 844 } 845 846 AutoMutex lock(mLock); 847 float volume; 848 if (output) { 849 PlaybackThread *thread = checkPlaybackThread_l(output); 850 if (thread == NULL) { 851 return 0.0f; 852 } 853 volume = thread->streamVolume(stream); 854 } else { 855 volume = streamVolume_l(stream); 856 } 857 858 return volume; 859} 860 861bool AudioFlinger::streamMute(audio_stream_type_t stream) const 862{ 863 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 864 return true; 865 } 866 867 AutoMutex lock(mLock); 868 return streamMute_l(stream); 869} 870 871status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 872{ 873 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 874 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 875 // check calling permissions 876 if (!settingsAllowed()) { 877 return PERMISSION_DENIED; 878 } 879 880 // ioHandle == 0 means the parameters are global to the audio hardware interface 881 if (ioHandle == 0) { 882 Mutex::Autolock _l(mLock); 883 status_t final_result = NO_ERROR; 884 { 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 887 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 888 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 889 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 890 final_result = result ?: final_result; 891 } 892 mHardwareStatus = AUDIO_HW_IDLE; 893 } 894 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 895 AudioParameter param = AudioParameter(keyValuePairs); 896 String8 value; 897 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 898 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 899 if (mBtNrecIsOff != btNrecIsOff) { 900 for (size_t i = 0; i < mRecordThreads.size(); i++) { 901 sp<RecordThread> thread = mRecordThreads.valueAt(i); 902 audio_devices_t device = thread->inDevice(); 903 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 904 // collect all of the thread's session IDs 905 KeyedVector<int, bool> ids = thread->sessionIds(); 906 // suspend effects associated with those session IDs 907 for (size_t j = 0; j < ids.size(); ++j) { 908 int sessionId = ids.keyAt(j); 909 thread->setEffectSuspended(FX_IID_AEC, 910 suspend, 911 sessionId); 912 thread->setEffectSuspended(FX_IID_NS, 913 suspend, 914 sessionId); 915 } 916 } 917 mBtNrecIsOff = btNrecIsOff; 918 } 919 } 920 String8 screenState; 921 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 922 bool isOff = screenState == "off"; 923 if (isOff != (gScreenState & 1)) { 924 gScreenState = ((gScreenState & ~1) + 2) | isOff; 925 } 926 } 927 return final_result; 928 } 929 930 // hold a strong ref on thread in case closeOutput() or closeInput() is called 931 // and the thread is exited once the lock is released 932 sp<ThreadBase> thread; 933 { 934 Mutex::Autolock _l(mLock); 935 thread = checkPlaybackThread_l(ioHandle); 936 if (thread == 0) { 937 thread = checkRecordThread_l(ioHandle); 938 } else if (thread == primaryPlaybackThread_l()) { 939 // indicate output device change to all input threads for pre processing 940 AudioParameter param = AudioParameter(keyValuePairs); 941 int value; 942 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 943 (value != 0)) { 944 for (size_t i = 0; i < mRecordThreads.size(); i++) { 945 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 946 } 947 } 948 } 949 } 950 if (thread != 0) { 951 return thread->setParameters(keyValuePairs); 952 } 953 return BAD_VALUE; 954} 955 956String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 957{ 958 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 959 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 960 961 Mutex::Autolock _l(mLock); 962 963 if (ioHandle == 0) { 964 String8 out_s8; 965 966 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 967 char *s; 968 { 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 971 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 972 s = dev->get_parameters(dev, keys.string()); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 } 975 out_s8 += String8(s ? s : ""); 976 free(s); 977 } 978 return out_s8; 979 } 980 981 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 982 if (playbackThread != NULL) { 983 return playbackThread->getParameters(keys); 984 } 985 RecordThread *recordThread = checkRecordThread_l(ioHandle); 986 if (recordThread != NULL) { 987 return recordThread->getParameters(keys); 988 } 989 return String8(""); 990} 991 992size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 993 audio_channel_mask_t channelMask) const 994{ 995 status_t ret = initCheck(); 996 if (ret != NO_ERROR) { 997 return 0; 998 } 999 1000 AutoMutex lock(mHardwareLock); 1001 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1002 struct audio_config config = { 1003 sample_rate: sampleRate, 1004 channel_mask: channelMask, 1005 format: format, 1006 }; 1007 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1008 size_t size = dev->get_input_buffer_size(dev, &config); 1009 mHardwareStatus = AUDIO_HW_IDLE; 1010 return size; 1011} 1012 1013unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1014{ 1015 Mutex::Autolock _l(mLock); 1016 1017 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1018 if (recordThread != NULL) { 1019 return recordThread->getInputFramesLost(); 1020 } 1021 return 0; 1022} 1023 1024status_t AudioFlinger::setVoiceVolume(float value) 1025{ 1026 status_t ret = initCheck(); 1027 if (ret != NO_ERROR) { 1028 return ret; 1029 } 1030 1031 // check calling permissions 1032 if (!settingsAllowed()) { 1033 return PERMISSION_DENIED; 1034 } 1035 1036 AutoMutex lock(mHardwareLock); 1037 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1038 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1039 ret = dev->set_voice_volume(dev, value); 1040 mHardwareStatus = AUDIO_HW_IDLE; 1041 1042 return ret; 1043} 1044 1045status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1046 audio_io_handle_t output) const 1047{ 1048 status_t status; 1049 1050 Mutex::Autolock _l(mLock); 1051 1052 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1053 if (playbackThread != NULL) { 1054 return playbackThread->getRenderPosition(halFrames, dspFrames); 1055 } 1056 1057 return BAD_VALUE; 1058} 1059 1060void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1061{ 1062 1063 Mutex::Autolock _l(mLock); 1064 1065 pid_t pid = IPCThreadState::self()->getCallingPid(); 1066 if (mNotificationClients.indexOfKey(pid) < 0) { 1067 sp<NotificationClient> notificationClient = new NotificationClient(this, 1068 client, 1069 pid); 1070 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1071 1072 mNotificationClients.add(pid, notificationClient); 1073 1074 sp<IBinder> binder = client->asBinder(); 1075 binder->linkToDeath(notificationClient); 1076 1077 // the config change is always sent from playback or record threads to avoid deadlock 1078 // with AudioSystem::gLock 1079 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1080 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1081 } 1082 1083 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1084 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1085 } 1086 } 1087} 1088 1089void AudioFlinger::removeNotificationClient(pid_t pid) 1090{ 1091 Mutex::Autolock _l(mLock); 1092 1093 mNotificationClients.removeItem(pid); 1094 1095 ALOGV("%d died, releasing its sessions", pid); 1096 size_t num = mAudioSessionRefs.size(); 1097 bool removed = false; 1098 for (size_t i = 0; i< num; ) { 1099 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1100 ALOGV(" pid %d @ %d", ref->mPid, i); 1101 if (ref->mPid == pid) { 1102 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1103 mAudioSessionRefs.removeAt(i); 1104 delete ref; 1105 removed = true; 1106 num--; 1107 } else { 1108 i++; 1109 } 1110 } 1111 if (removed) { 1112 purgeStaleEffects_l(); 1113 } 1114} 1115 1116// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1117void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1118{ 1119 size_t size = mNotificationClients.size(); 1120 for (size_t i = 0; i < size; i++) { 1121 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1122 param2); 1123 } 1124} 1125 1126// removeClient_l() must be called with AudioFlinger::mLock held 1127void AudioFlinger::removeClient_l(pid_t pid) 1128{ 1129 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1130 IPCThreadState::self()->getCallingPid()); 1131 mClients.removeItem(pid); 1132} 1133 1134// getEffectThread_l() must be called with AudioFlinger::mLock held 1135sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1136{ 1137 sp<PlaybackThread> thread; 1138 1139 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1140 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1141 ALOG_ASSERT(thread == 0); 1142 thread = mPlaybackThreads.valueAt(i); 1143 } 1144 } 1145 1146 return thread; 1147} 1148 1149// ---------------------------------------------------------------------------- 1150 1151AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1152 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1153 : Thread(false /*canCallJava*/), 1154 mType(type), 1155 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1156 // mChannelMask 1157 mChannelCount(0), 1158 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1159 mParamStatus(NO_ERROR), 1160 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1161 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1162 // mName will be set by concrete (non-virtual) subclass 1163 mDeathRecipient(new PMDeathRecipient(this)) 1164{ 1165} 1166 1167AudioFlinger::ThreadBase::~ThreadBase() 1168{ 1169 mParamCond.broadcast(); 1170 // do not lock the mutex in destructor 1171 releaseWakeLock_l(); 1172 if (mPowerManager != 0) { 1173 sp<IBinder> binder = mPowerManager->asBinder(); 1174 binder->unlinkToDeath(mDeathRecipient); 1175 } 1176} 1177 1178void AudioFlinger::ThreadBase::exit() 1179{ 1180 ALOGV("ThreadBase::exit"); 1181 // do any cleanup required for exit to succeed 1182 preExit(); 1183 { 1184 // This lock prevents the following race in thread (uniprocessor for illustration): 1185 // if (!exitPending()) { 1186 // // context switch from here to exit() 1187 // // exit() calls requestExit(), what exitPending() observes 1188 // // exit() calls signal(), which is dropped since no waiters 1189 // // context switch back from exit() to here 1190 // mWaitWorkCV.wait(...); 1191 // // now thread is hung 1192 // } 1193 AutoMutex lock(mLock); 1194 requestExit(); 1195 mWaitWorkCV.broadcast(); 1196 } 1197 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1198 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1199 requestExitAndWait(); 1200} 1201 1202status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1203{ 1204 status_t status; 1205 1206 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1207 Mutex::Autolock _l(mLock); 1208 1209 mNewParameters.add(keyValuePairs); 1210 mWaitWorkCV.signal(); 1211 // wait condition with timeout in case the thread loop has exited 1212 // before the request could be processed 1213 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1214 status = mParamStatus; 1215 mWaitWorkCV.signal(); 1216 } else { 1217 status = TIMED_OUT; 1218 } 1219 return status; 1220} 1221 1222void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 1223{ 1224 Mutex::Autolock _l(mLock); 1225 sendIoConfigEvent_l(event, param); 1226} 1227 1228// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 1229void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 1230{ 1231 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 1232 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 1233 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 1234 param); 1235 mWaitWorkCV.signal(); 1236} 1237 1238// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 1239void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 1240{ 1241 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 1242 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 1243 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 1244 mConfigEvents.size(), pid, tid, prio); 1245 mWaitWorkCV.signal(); 1246} 1247 1248void AudioFlinger::ThreadBase::processConfigEvents() 1249{ 1250 mLock.lock(); 1251 while (!mConfigEvents.isEmpty()) { 1252 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1253 ConfigEvent *event = mConfigEvents[0]; 1254 mConfigEvents.removeAt(0); 1255 // release mLock before locking AudioFlinger mLock: lock order is always 1256 // AudioFlinger then ThreadBase to avoid cross deadlock 1257 mLock.unlock(); 1258 switch(event->type()) { 1259 case CFG_EVENT_PRIO: { 1260 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 1261 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 1262 if (err != 0) { 1263 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 1264 "error %d", 1265 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 1266 } 1267 } break; 1268 case CFG_EVENT_IO: { 1269 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 1270 mAudioFlinger->mLock.lock(); 1271 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 1272 mAudioFlinger->mLock.unlock(); 1273 } break; 1274 default: 1275 ALOGE("processConfigEvents() unknown event type %d", event->type()); 1276 break; 1277 } 1278 delete event; 1279 mLock.lock(); 1280 } 1281 mLock.unlock(); 1282} 1283 1284void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1285{ 1286 const size_t SIZE = 256; 1287 char buffer[SIZE]; 1288 String8 result; 1289 1290 bool locked = tryLock(mLock); 1291 if (!locked) { 1292 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1293 write(fd, buffer, strlen(buffer)); 1294 } 1295 1296 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1297 result.append(buffer); 1298 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1299 result.append(buffer); 1300 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1301 result.append(buffer); 1302 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 1303 result.append(buffer); 1304 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1305 result.append(buffer); 1306 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1307 result.append(buffer); 1308 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1309 result.append(buffer); 1310 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1311 result.append(buffer); 1312 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1313 result.append(buffer); 1314 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1315 result.append(buffer); 1316 1317 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1318 result.append(buffer); 1319 result.append(" Index Command"); 1320 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1321 snprintf(buffer, SIZE, "\n %02d ", i); 1322 result.append(buffer); 1323 result.append(mNewParameters[i]); 1324 } 1325 1326 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1327 result.append(buffer); 1328 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1329 mConfigEvents[i]->dump(buffer, SIZE); 1330 result.append(buffer); 1331 } 1332 result.append("\n"); 1333 1334 write(fd, result.string(), result.size()); 1335 1336 if (locked) { 1337 mLock.unlock(); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1342{ 1343 const size_t SIZE = 256; 1344 char buffer[SIZE]; 1345 String8 result; 1346 1347 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1348 write(fd, buffer, strlen(buffer)); 1349 1350 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1351 sp<EffectChain> chain = mEffectChains[i]; 1352 if (chain != 0) { 1353 chain->dump(fd, args); 1354 } 1355 } 1356} 1357 1358void AudioFlinger::ThreadBase::acquireWakeLock() 1359{ 1360 Mutex::Autolock _l(mLock); 1361 acquireWakeLock_l(); 1362} 1363 1364void AudioFlinger::ThreadBase::acquireWakeLock_l() 1365{ 1366 if (mPowerManager == 0) { 1367 // use checkService() to avoid blocking if power service is not up yet 1368 sp<IBinder> binder = 1369 defaultServiceManager()->checkService(String16("power")); 1370 if (binder == 0) { 1371 ALOGW("Thread %s cannot connect to the power manager service", mName); 1372 } else { 1373 mPowerManager = interface_cast<IPowerManager>(binder); 1374 binder->linkToDeath(mDeathRecipient); 1375 } 1376 } 1377 if (mPowerManager != 0) { 1378 sp<IBinder> binder = new BBinder(); 1379 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1380 binder, 1381 String16(mName)); 1382 if (status == NO_ERROR) { 1383 mWakeLockToken = binder; 1384 } 1385 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1386 } 1387} 1388 1389void AudioFlinger::ThreadBase::releaseWakeLock() 1390{ 1391 Mutex::Autolock _l(mLock); 1392 releaseWakeLock_l(); 1393} 1394 1395void AudioFlinger::ThreadBase::releaseWakeLock_l() 1396{ 1397 if (mWakeLockToken != 0) { 1398 ALOGV("releaseWakeLock_l() %s", mName); 1399 if (mPowerManager != 0) { 1400 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1401 } 1402 mWakeLockToken.clear(); 1403 } 1404} 1405 1406void AudioFlinger::ThreadBase::clearPowerManager() 1407{ 1408 Mutex::Autolock _l(mLock); 1409 releaseWakeLock_l(); 1410 mPowerManager.clear(); 1411} 1412 1413void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1414{ 1415 sp<ThreadBase> thread = mThread.promote(); 1416 if (thread != 0) { 1417 thread->clearPowerManager(); 1418 } 1419 ALOGW("power manager service died !!!"); 1420} 1421 1422void AudioFlinger::ThreadBase::setEffectSuspended( 1423 const effect_uuid_t *type, bool suspend, int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 setEffectSuspended_l(type, suspend, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::setEffectSuspended_l( 1430 const effect_uuid_t *type, bool suspend, int sessionId) 1431{ 1432 sp<EffectChain> chain = getEffectChain_l(sessionId); 1433 if (chain != 0) { 1434 if (type != NULL) { 1435 chain->setEffectSuspended_l(type, suspend); 1436 } else { 1437 chain->setEffectSuspendedAll_l(suspend); 1438 } 1439 } 1440 1441 updateSuspendedSessions_l(type, suspend, sessionId); 1442} 1443 1444void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1445{ 1446 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1447 if (index < 0) { 1448 return; 1449 } 1450 1451 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1452 mSuspendedSessions.valueAt(index); 1453 1454 for (size_t i = 0; i < sessionEffects.size(); i++) { 1455 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1456 for (int j = 0; j < desc->mRefCount; j++) { 1457 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1458 chain->setEffectSuspendedAll_l(true); 1459 } else { 1460 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1461 desc->mType.timeLow); 1462 chain->setEffectSuspended_l(&desc->mType, true); 1463 } 1464 } 1465 } 1466} 1467 1468void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1469 bool suspend, 1470 int sessionId) 1471{ 1472 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1473 1474 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1475 1476 if (suspend) { 1477 if (index >= 0) { 1478 sessionEffects = mSuspendedSessions.valueAt(index); 1479 } else { 1480 mSuspendedSessions.add(sessionId, sessionEffects); 1481 } 1482 } else { 1483 if (index < 0) { 1484 return; 1485 } 1486 sessionEffects = mSuspendedSessions.valueAt(index); 1487 } 1488 1489 1490 int key = EffectChain::kKeyForSuspendAll; 1491 if (type != NULL) { 1492 key = type->timeLow; 1493 } 1494 index = sessionEffects.indexOfKey(key); 1495 1496 sp<SuspendedSessionDesc> desc; 1497 if (suspend) { 1498 if (index >= 0) { 1499 desc = sessionEffects.valueAt(index); 1500 } else { 1501 desc = new SuspendedSessionDesc(); 1502 if (type != NULL) { 1503 desc->mType = *type; 1504 } 1505 sessionEffects.add(key, desc); 1506 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1507 } 1508 desc->mRefCount++; 1509 } else { 1510 if (index < 0) { 1511 return; 1512 } 1513 desc = sessionEffects.valueAt(index); 1514 if (--desc->mRefCount == 0) { 1515 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1516 sessionEffects.removeItemsAt(index); 1517 if (sessionEffects.isEmpty()) { 1518 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1519 sessionId); 1520 mSuspendedSessions.removeItem(sessionId); 1521 } 1522 } 1523 } 1524 if (!sessionEffects.isEmpty()) { 1525 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1526 } 1527} 1528 1529void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1530 bool enabled, 1531 int sessionId) 1532{ 1533 Mutex::Autolock _l(mLock); 1534 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1535} 1536 1537void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1538 bool enabled, 1539 int sessionId) 1540{ 1541 if (mType != RECORD) { 1542 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1543 // another session. This gives the priority to well behaved effect control panels 1544 // and applications not using global effects. 1545 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1546 // global effects 1547 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1548 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1549 } 1550 } 1551 1552 sp<EffectChain> chain = getEffectChain_l(sessionId); 1553 if (chain != 0) { 1554 chain->checkSuspendOnEffectEnabled(effect, enabled); 1555 } 1556} 1557 1558// ---------------------------------------------------------------------------- 1559 1560AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1561 AudioStreamOut* output, 1562 audio_io_handle_t id, 1563 audio_devices_t device, 1564 type_t type) 1565 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1566 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1567 // mStreamTypes[] initialized in constructor body 1568 mOutput(output), 1569 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1570 mMixerStatus(MIXER_IDLE), 1571 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1572 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1573 mScreenState(gScreenState), 1574 // index 0 is reserved for normal mixer's submix 1575 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1576{ 1577 snprintf(mName, kNameLength, "AudioOut_%X", id); 1578 1579 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1580 // it would be safer to explicitly pass initial masterVolume/masterMute as 1581 // parameter. 1582 // 1583 // If the HAL we are using has support for master volume or master mute, 1584 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1585 // and the mute set to false). 1586 mMasterVolume = audioFlinger->masterVolume_l(); 1587 mMasterMute = audioFlinger->masterMute_l(); 1588 if (mOutput && mOutput->audioHwDev) { 1589 if (mOutput->audioHwDev->canSetMasterVolume()) { 1590 mMasterVolume = 1.0; 1591 } 1592 1593 if (mOutput->audioHwDev->canSetMasterMute()) { 1594 mMasterMute = false; 1595 } 1596 } 1597 1598 readOutputParameters(); 1599 1600 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1601 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1602 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1603 stream = (audio_stream_type_t) (stream + 1)) { 1604 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1605 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1606 } 1607 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1608 // because mAudioFlinger doesn't have one to copy from 1609} 1610 1611AudioFlinger::PlaybackThread::~PlaybackThread() 1612{ 1613 delete [] mMixBuffer; 1614} 1615 1616void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1617{ 1618 dumpInternals(fd, args); 1619 dumpTracks(fd, args); 1620 dumpEffectChains(fd, args); 1621} 1622 1623void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1624{ 1625 const size_t SIZE = 256; 1626 char buffer[SIZE]; 1627 String8 result; 1628 1629 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1630 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1631 const stream_type_t *st = &mStreamTypes[i]; 1632 if (i > 0) { 1633 result.appendFormat(", "); 1634 } 1635 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1636 if (st->mute) { 1637 result.append("M"); 1638 } 1639 } 1640 result.append("\n"); 1641 write(fd, result.string(), result.length()); 1642 result.clear(); 1643 1644 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1645 result.append(buffer); 1646 Track::appendDumpHeader(result); 1647 for (size_t i = 0; i < mTracks.size(); ++i) { 1648 sp<Track> track = mTracks[i]; 1649 if (track != 0) { 1650 track->dump(buffer, SIZE); 1651 result.append(buffer); 1652 } 1653 } 1654 1655 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1656 result.append(buffer); 1657 Track::appendDumpHeader(result); 1658 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1659 sp<Track> track = mActiveTracks[i].promote(); 1660 if (track != 0) { 1661 track->dump(buffer, SIZE); 1662 result.append(buffer); 1663 } 1664 } 1665 write(fd, result.string(), result.size()); 1666 1667 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1668 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1669 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1670 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1671} 1672 1673void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1674{ 1675 const size_t SIZE = 256; 1676 char buffer[SIZE]; 1677 String8 result; 1678 1679 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1680 result.append(buffer); 1681 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1682 ns2ms(systemTime() - mLastWriteTime)); 1683 result.append(buffer); 1684 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1685 result.append(buffer); 1686 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1687 result.append(buffer); 1688 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1689 result.append(buffer); 1690 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1691 result.append(buffer); 1692 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1693 result.append(buffer); 1694 write(fd, result.string(), result.size()); 1695 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1696 1697 dumpBase(fd, args); 1698} 1699 1700// Thread virtuals 1701status_t AudioFlinger::PlaybackThread::readyToRun() 1702{ 1703 status_t status = initCheck(); 1704 if (status == NO_ERROR) { 1705 ALOGI("AudioFlinger's thread %p ready to run", this); 1706 } else { 1707 ALOGE("No working audio driver found."); 1708 } 1709 return status; 1710} 1711 1712void AudioFlinger::PlaybackThread::onFirstRef() 1713{ 1714 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1715} 1716 1717// ThreadBase virtuals 1718void AudioFlinger::PlaybackThread::preExit() 1719{ 1720 ALOGV(" preExit()"); 1721 // FIXME this is using hard-coded strings but in the future, this functionality will be 1722 // converted to use audio HAL extensions required to support tunneling 1723 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1724} 1725 1726// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1727sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1728 const sp<AudioFlinger::Client>& client, 1729 audio_stream_type_t streamType, 1730 uint32_t sampleRate, 1731 audio_format_t format, 1732 audio_channel_mask_t channelMask, 1733 size_t frameCount, 1734 const sp<IMemory>& sharedBuffer, 1735 int sessionId, 1736 IAudioFlinger::track_flags_t *flags, 1737 pid_t tid, 1738 status_t *status) 1739{ 1740 sp<Track> track; 1741 status_t lStatus; 1742 1743 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1744 1745 // client expresses a preference for FAST, but we get the final say 1746 if (*flags & IAudioFlinger::TRACK_FAST) { 1747 if ( 1748 // not timed 1749 (!isTimed) && 1750 // either of these use cases: 1751 ( 1752 // use case 1: shared buffer with any frame count 1753 ( 1754 (sharedBuffer != 0) 1755 ) || 1756 // use case 2: callback handler and frame count is default or at least as large as HAL 1757 ( 1758 (tid != -1) && 1759 ((frameCount == 0) || 1760 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1761 ) 1762 ) && 1763 // PCM data 1764 audio_is_linear_pcm(format) && 1765 // mono or stereo 1766 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1767 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1768#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1769 // hardware sample rate 1770 (sampleRate == mSampleRate) && 1771#endif 1772 // normal mixer has an associated fast mixer 1773 hasFastMixer() && 1774 // there are sufficient fast track slots available 1775 (mFastTrackAvailMask != 0) 1776 // FIXME test that MixerThread for this fast track has a capable output HAL 1777 // FIXME add a permission test also? 1778 ) { 1779 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1780 if (frameCount == 0) { 1781 frameCount = mFrameCount * kFastTrackMultiplier; 1782 } 1783 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1784 frameCount, mFrameCount); 1785 } else { 1786 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1787 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1788 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1789 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1790 audio_is_linear_pcm(format), 1791 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1792 *flags &= ~IAudioFlinger::TRACK_FAST; 1793 // For compatibility with AudioTrack calculation, buffer depth is forced 1794 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1795 // This is probably too conservative, but legacy application code may depend on it. 1796 // If you change this calculation, also review the start threshold which is related. 1797 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1798 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1799 if (minBufCount < 2) { 1800 minBufCount = 2; 1801 } 1802 size_t minFrameCount = mNormalFrameCount * minBufCount; 1803 if (frameCount < minFrameCount) { 1804 frameCount = minFrameCount; 1805 } 1806 } 1807 } 1808 1809 if (mType == DIRECT) { 1810 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1811 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1812 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1813 "for output %p with format %d", 1814 sampleRate, format, channelMask, mOutput, mFormat); 1815 lStatus = BAD_VALUE; 1816 goto Exit; 1817 } 1818 } 1819 } else { 1820 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1821 if (sampleRate > mSampleRate*2) { 1822 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1823 lStatus = BAD_VALUE; 1824 goto Exit; 1825 } 1826 } 1827 1828 lStatus = initCheck(); 1829 if (lStatus != NO_ERROR) { 1830 ALOGE("Audio driver not initialized."); 1831 goto Exit; 1832 } 1833 1834 { // scope for mLock 1835 Mutex::Autolock _l(mLock); 1836 1837 // all tracks in same audio session must share the same routing strategy otherwise 1838 // conflicts will happen when tracks are moved from one output to another by audio policy 1839 // manager 1840 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1841 for (size_t i = 0; i < mTracks.size(); ++i) { 1842 sp<Track> t = mTracks[i]; 1843 if (t != 0 && !t->isOutputTrack()) { 1844 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1845 if (sessionId == t->sessionId() && strategy != actual) { 1846 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1847 strategy, actual); 1848 lStatus = BAD_VALUE; 1849 goto Exit; 1850 } 1851 } 1852 } 1853 1854 if (!isTimed) { 1855 track = new Track(this, client, streamType, sampleRate, format, 1856 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1857 } else { 1858 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1859 channelMask, frameCount, sharedBuffer, sessionId); 1860 } 1861 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1862 lStatus = NO_MEMORY; 1863 goto Exit; 1864 } 1865 mTracks.add(track); 1866 1867 sp<EffectChain> chain = getEffectChain_l(sessionId); 1868 if (chain != 0) { 1869 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1870 track->setMainBuffer(chain->inBuffer()); 1871 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1872 chain->incTrackCnt(); 1873 } 1874 1875 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1876 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1877 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1878 // so ask activity manager to do this on our behalf 1879 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1880 } 1881 } 1882 1883 lStatus = NO_ERROR; 1884 1885Exit: 1886 if (status) { 1887 *status = lStatus; 1888 } 1889 return track; 1890} 1891 1892uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 1893{ 1894 if (mFastMixer != NULL) { 1895 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1896 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1897 } 1898 return latency; 1899} 1900 1901uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1902{ 1903 return latency; 1904} 1905 1906uint32_t AudioFlinger::PlaybackThread::latency() const 1907{ 1908 Mutex::Autolock _l(mLock); 1909 return latency_l(); 1910} 1911uint32_t AudioFlinger::PlaybackThread::latency_l() const 1912{ 1913 if (initCheck() == NO_ERROR) { 1914 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1915 } else { 1916 return 0; 1917 } 1918} 1919 1920void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1921{ 1922 Mutex::Autolock _l(mLock); 1923 // Don't apply master volume in SW if our HAL can do it for us. 1924 if (mOutput && mOutput->audioHwDev && 1925 mOutput->audioHwDev->canSetMasterVolume()) { 1926 mMasterVolume = 1.0; 1927 } else { 1928 mMasterVolume = value; 1929 } 1930} 1931 1932void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1933{ 1934 Mutex::Autolock _l(mLock); 1935 // Don't apply master mute in SW if our HAL can do it for us. 1936 if (mOutput && mOutput->audioHwDev && 1937 mOutput->audioHwDev->canSetMasterMute()) { 1938 mMasterMute = false; 1939 } else { 1940 mMasterMute = muted; 1941 } 1942} 1943 1944void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1945{ 1946 Mutex::Autolock _l(mLock); 1947 mStreamTypes[stream].volume = value; 1948} 1949 1950void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1951{ 1952 Mutex::Autolock _l(mLock); 1953 mStreamTypes[stream].mute = muted; 1954} 1955 1956float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1957{ 1958 Mutex::Autolock _l(mLock); 1959 return mStreamTypes[stream].volume; 1960} 1961 1962// addTrack_l() must be called with ThreadBase::mLock held 1963status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1964{ 1965 status_t status = ALREADY_EXISTS; 1966 1967 // set retry count for buffer fill 1968 track->mRetryCount = kMaxTrackStartupRetries; 1969 if (mActiveTracks.indexOf(track) < 0) { 1970 // the track is newly added, make sure it fills up all its 1971 // buffers before playing. This is to ensure the client will 1972 // effectively get the latency it requested. 1973 track->mFillingUpStatus = Track::FS_FILLING; 1974 track->mResetDone = false; 1975 track->mPresentationCompleteFrames = 0; 1976 mActiveTracks.add(track); 1977 if (track->mainBuffer() != mMixBuffer) { 1978 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1979 if (chain != 0) { 1980 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1981 track->sessionId()); 1982 chain->incActiveTrackCnt(); 1983 } 1984 } 1985 1986 status = NO_ERROR; 1987 } 1988 1989 ALOGV("mWaitWorkCV.broadcast"); 1990 mWaitWorkCV.broadcast(); 1991 1992 return status; 1993} 1994 1995// destroyTrack_l() must be called with ThreadBase::mLock held 1996void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1997{ 1998 track->mState = TrackBase::TERMINATED; 1999 // active tracks are removed by threadLoop() 2000 if (mActiveTracks.indexOf(track) < 0) { 2001 removeTrack_l(track); 2002 } 2003} 2004 2005void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2006{ 2007 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2008 mTracks.remove(track); 2009 deleteTrackName_l(track->name()); 2010 // redundant as track is about to be destroyed, for dumpsys only 2011 track->mName = -1; 2012 if (track->isFastTrack()) { 2013 int index = track->mFastIndex; 2014 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2015 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2016 mFastTrackAvailMask |= 1 << index; 2017 // redundant as track is about to be destroyed, for dumpsys only 2018 track->mFastIndex = -1; 2019 } 2020 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2021 if (chain != 0) { 2022 chain->decTrackCnt(); 2023 } 2024} 2025 2026String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2027{ 2028 String8 out_s8 = String8(""); 2029 char *s; 2030 2031 Mutex::Autolock _l(mLock); 2032 if (initCheck() != NO_ERROR) { 2033 return out_s8; 2034 } 2035 2036 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2037 out_s8 = String8(s); 2038 free(s); 2039 return out_s8; 2040} 2041 2042// audioConfigChanged_l() must be called with AudioFlinger::mLock held 2043void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 2044 AudioSystem::OutputDescriptor desc; 2045 void *param2 = NULL; 2046 2047 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 2048 param); 2049 2050 switch (event) { 2051 case AudioSystem::OUTPUT_OPENED: 2052 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2053 desc.channels = mChannelMask; 2054 desc.samplingRate = mSampleRate; 2055 desc.format = mFormat; 2056 desc.frameCount = mNormalFrameCount; // FIXME see 2057 // AudioFlinger::frameCount(audio_io_handle_t) 2058 desc.latency = latency(); 2059 param2 = &desc; 2060 break; 2061 2062 case AudioSystem::STREAM_CONFIG_CHANGED: 2063 param2 = ¶m; 2064 case AudioSystem::OUTPUT_CLOSED: 2065 default: 2066 break; 2067 } 2068 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2069} 2070 2071void AudioFlinger::PlaybackThread::readOutputParameters() 2072{ 2073 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2074 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2075 mChannelCount = (uint16_t)popcount(mChannelMask); 2076 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2077 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2078 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2079 if (mFrameCount & 15) { 2080 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2081 mFrameCount); 2082 } 2083 2084 // Calculate size of normal mix buffer relative to the HAL output buffer size 2085 double multiplier = 1.0; 2086 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2087 kUseFastMixer == FastMixer_Dynamic)) { 2088 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2089 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2090 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2091 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2092 maxNormalFrameCount = maxNormalFrameCount & ~15; 2093 if (maxNormalFrameCount < minNormalFrameCount) { 2094 maxNormalFrameCount = minNormalFrameCount; 2095 } 2096 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2097 if (multiplier <= 1.0) { 2098 multiplier = 1.0; 2099 } else if (multiplier <= 2.0) { 2100 if (2 * mFrameCount <= maxNormalFrameCount) { 2101 multiplier = 2.0; 2102 } else { 2103 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2104 } 2105 } else { 2106 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2107 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 2108 // track, but we sometimes have to do this to satisfy the maximum frame count 2109 // constraint) 2110 // FIXME this rounding up should not be done if no HAL SRC 2111 uint32_t truncMult = (uint32_t) multiplier; 2112 if ((truncMult & 1)) { 2113 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2114 ++truncMult; 2115 } 2116 } 2117 multiplier = (double) truncMult; 2118 } 2119 } 2120 mNormalFrameCount = multiplier * mFrameCount; 2121 // round up to nearest 16 frames to satisfy AudioMixer 2122 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2123 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 2124 mNormalFrameCount); 2125 2126 delete[] mMixBuffer; 2127 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2128 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2129 2130 // force reconfiguration of effect chains and engines to take new buffer size and audio 2131 // parameters into account 2132 // Note that mLock is not held when readOutputParameters() is called from the constructor 2133 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2134 // matter. 2135 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2136 Vector< sp<EffectChain> > effectChains = mEffectChains; 2137 for (size_t i = 0; i < effectChains.size(); i ++) { 2138 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2139 } 2140} 2141 2142 2143status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 2144{ 2145 if (halFrames == NULL || dspFrames == NULL) { 2146 return BAD_VALUE; 2147 } 2148 Mutex::Autolock _l(mLock); 2149 if (initCheck() != NO_ERROR) { 2150 return INVALID_OPERATION; 2151 } 2152 size_t framesWritten = mBytesWritten / mFrameSize; 2153 *halFrames = framesWritten; 2154 2155 if (isSuspended()) { 2156 // return an estimation of rendered frames when the output is suspended 2157 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2158 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2159 return NO_ERROR; 2160 } else { 2161 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2162 } 2163} 2164 2165uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2166{ 2167 Mutex::Autolock _l(mLock); 2168 uint32_t result = 0; 2169 if (getEffectChain_l(sessionId) != 0) { 2170 result = EFFECT_SESSION; 2171 } 2172 2173 for (size_t i = 0; i < mTracks.size(); ++i) { 2174 sp<Track> track = mTracks[i]; 2175 if (sessionId == track->sessionId() && 2176 !(track->mCblk->flags & CBLK_INVALID)) { 2177 result |= TRACK_SESSION; 2178 break; 2179 } 2180 } 2181 2182 return result; 2183} 2184 2185uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2186{ 2187 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2188 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2189 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2190 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2191 } 2192 for (size_t i = 0; i < mTracks.size(); i++) { 2193 sp<Track> track = mTracks[i]; 2194 if (sessionId == track->sessionId() && 2195 !(track->mCblk->flags & CBLK_INVALID)) { 2196 return AudioSystem::getStrategyForStream(track->streamType()); 2197 } 2198 } 2199 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2200} 2201 2202 2203AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2204{ 2205 Mutex::Autolock _l(mLock); 2206 return mOutput; 2207} 2208 2209AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2210{ 2211 Mutex::Autolock _l(mLock); 2212 AudioStreamOut *output = mOutput; 2213 mOutput = NULL; 2214 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2215 // must push a NULL and wait for ack 2216 mOutputSink.clear(); 2217 mPipeSink.clear(); 2218 mNormalSink.clear(); 2219 return output; 2220} 2221 2222// this method must always be called either with ThreadBase mLock held or inside the thread loop 2223audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2224{ 2225 if (mOutput == NULL) { 2226 return NULL; 2227 } 2228 return &mOutput->stream->common; 2229} 2230 2231uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2232{ 2233 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2234} 2235 2236status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2237{ 2238 if (!isValidSyncEvent(event)) { 2239 return BAD_VALUE; 2240 } 2241 2242 Mutex::Autolock _l(mLock); 2243 2244 for (size_t i = 0; i < mTracks.size(); ++i) { 2245 sp<Track> track = mTracks[i]; 2246 if (event->triggerSession() == track->sessionId()) { 2247 (void) track->setSyncEvent(event); 2248 return NO_ERROR; 2249 } 2250 } 2251 2252 return NAME_NOT_FOUND; 2253} 2254 2255bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2256{ 2257 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2258} 2259 2260void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2261 const Vector< sp<Track> >& tracksToRemove) 2262{ 2263 size_t count = tracksToRemove.size(); 2264 if (CC_UNLIKELY(count)) { 2265 for (size_t i = 0 ; i < count ; i++) { 2266 const sp<Track>& track = tracksToRemove.itemAt(i); 2267 if ((track->sharedBuffer() != 0) && 2268 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2269 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2270 } 2271 } 2272 } 2273 2274} 2275 2276// ---------------------------------------------------------------------------- 2277 2278AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2279 audio_io_handle_t id, audio_devices_t device, type_t type) 2280 : PlaybackThread(audioFlinger, output, id, device, type), 2281 // mAudioMixer below 2282 // mFastMixer below 2283 mFastMixerFutex(0) 2284 // mOutputSink below 2285 // mPipeSink below 2286 // mNormalSink below 2287{ 2288 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2289 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2290 "mFrameCount=%d, mNormalFrameCount=%d", 2291 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2292 mNormalFrameCount); 2293 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2294 2295 // FIXME - Current mixer implementation only supports stereo output 2296 if (mChannelCount != FCC_2) { 2297 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2298 } 2299 2300 // create an NBAIO sink for the HAL output stream, and negotiate 2301 mOutputSink = new AudioStreamOutSink(output->stream); 2302 size_t numCounterOffers = 0; 2303 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2304 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2305 ALOG_ASSERT(index == 0); 2306 2307 // initialize fast mixer depending on configuration 2308 bool initFastMixer; 2309 switch (kUseFastMixer) { 2310 case FastMixer_Never: 2311 initFastMixer = false; 2312 break; 2313 case FastMixer_Always: 2314 initFastMixer = true; 2315 break; 2316 case FastMixer_Static: 2317 case FastMixer_Dynamic: 2318 initFastMixer = mFrameCount < mNormalFrameCount; 2319 break; 2320 } 2321 if (initFastMixer) { 2322 2323 // create a MonoPipe to connect our submix to FastMixer 2324 NBAIO_Format format = mOutputSink->format(); 2325 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2326 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2327 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2328 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2329 const NBAIO_Format offers[1] = {format}; 2330 size_t numCounterOffers = 0; 2331 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2332 ALOG_ASSERT(index == 0); 2333 monoPipe->setAvgFrames((mScreenState & 1) ? 2334 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2335 mPipeSink = monoPipe; 2336 2337#ifdef TEE_SINK_FRAMES 2338 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2339 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2340 numCounterOffers = 0; 2341 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2342 ALOG_ASSERT(index == 0); 2343 mTeeSink = teeSink; 2344 PipeReader *teeSource = new PipeReader(*teeSink); 2345 numCounterOffers = 0; 2346 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2347 ALOG_ASSERT(index == 0); 2348 mTeeSource = teeSource; 2349#endif 2350 2351 // create fast mixer and configure it initially with just one fast track for our submix 2352 mFastMixer = new FastMixer(); 2353 FastMixerStateQueue *sq = mFastMixer->sq(); 2354#ifdef STATE_QUEUE_DUMP 2355 sq->setObserverDump(&mStateQueueObserverDump); 2356 sq->setMutatorDump(&mStateQueueMutatorDump); 2357#endif 2358 FastMixerState *state = sq->begin(); 2359 FastTrack *fastTrack = &state->mFastTracks[0]; 2360 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2361 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2362 fastTrack->mVolumeProvider = NULL; 2363 fastTrack->mGeneration++; 2364 state->mFastTracksGen++; 2365 state->mTrackMask = 1; 2366 // fast mixer will use the HAL output sink 2367 state->mOutputSink = mOutputSink.get(); 2368 state->mOutputSinkGen++; 2369 state->mFrameCount = mFrameCount; 2370 state->mCommand = FastMixerState::COLD_IDLE; 2371 // already done in constructor initialization list 2372 //mFastMixerFutex = 0; 2373 state->mColdFutexAddr = &mFastMixerFutex; 2374 state->mColdGen++; 2375 state->mDumpState = &mFastMixerDumpState; 2376 state->mTeeSink = mTeeSink.get(); 2377 sq->end(); 2378 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2379 2380 // start the fast mixer 2381 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2382 pid_t tid = mFastMixer->getTid(); 2383 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2384 if (err != 0) { 2385 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2386 kPriorityFastMixer, getpid_cached, tid, err); 2387 } 2388 2389#ifdef AUDIO_WATCHDOG 2390 // create and start the watchdog 2391 mAudioWatchdog = new AudioWatchdog(); 2392 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2393 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2394 tid = mAudioWatchdog->getTid(); 2395 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2396 if (err != 0) { 2397 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2398 kPriorityFastMixer, getpid_cached, tid, err); 2399 } 2400#endif 2401 2402 } else { 2403 mFastMixer = NULL; 2404 } 2405 2406 switch (kUseFastMixer) { 2407 case FastMixer_Never: 2408 case FastMixer_Dynamic: 2409 mNormalSink = mOutputSink; 2410 break; 2411 case FastMixer_Always: 2412 mNormalSink = mPipeSink; 2413 break; 2414 case FastMixer_Static: 2415 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2416 break; 2417 } 2418} 2419 2420AudioFlinger::MixerThread::~MixerThread() 2421{ 2422 if (mFastMixer != NULL) { 2423 FastMixerStateQueue *sq = mFastMixer->sq(); 2424 FastMixerState *state = sq->begin(); 2425 if (state->mCommand == FastMixerState::COLD_IDLE) { 2426 int32_t old = android_atomic_inc(&mFastMixerFutex); 2427 if (old == -1) { 2428 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2429 } 2430 } 2431 state->mCommand = FastMixerState::EXIT; 2432 sq->end(); 2433 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2434 mFastMixer->join(); 2435 // Though the fast mixer thread has exited, it's state queue is still valid. 2436 // We'll use that extract the final state which contains one remaining fast track 2437 // corresponding to our sub-mix. 2438 state = sq->begin(); 2439 ALOG_ASSERT(state->mTrackMask == 1); 2440 FastTrack *fastTrack = &state->mFastTracks[0]; 2441 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2442 delete fastTrack->mBufferProvider; 2443 sq->end(false /*didModify*/); 2444 delete mFastMixer; 2445#ifdef AUDIO_WATCHDOG 2446 if (mAudioWatchdog != 0) { 2447 mAudioWatchdog->requestExit(); 2448 mAudioWatchdog->requestExitAndWait(); 2449 mAudioWatchdog.clear(); 2450 } 2451#endif 2452 } 2453 delete mAudioMixer; 2454} 2455 2456class CpuStats { 2457public: 2458 CpuStats(); 2459 void sample(const String8 &title); 2460#ifdef DEBUG_CPU_USAGE 2461private: 2462 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2463 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2464 2465 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2466 2467 int mCpuNum; // thread's current CPU number 2468 int mCpukHz; // frequency of thread's current CPU in kHz 2469#endif 2470}; 2471 2472CpuStats::CpuStats() 2473#ifdef DEBUG_CPU_USAGE 2474 : mCpuNum(-1), mCpukHz(-1) 2475#endif 2476{ 2477} 2478 2479void CpuStats::sample(const String8 &title) { 2480#ifdef DEBUG_CPU_USAGE 2481 // get current thread's delta CPU time in wall clock ns 2482 double wcNs; 2483 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2484 2485 // record sample for wall clock statistics 2486 if (valid) { 2487 mWcStats.sample(wcNs); 2488 } 2489 2490 // get the current CPU number 2491 int cpuNum = sched_getcpu(); 2492 2493 // get the current CPU frequency in kHz 2494 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2495 2496 // check if either CPU number or frequency changed 2497 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2498 mCpuNum = cpuNum; 2499 mCpukHz = cpukHz; 2500 // ignore sample for purposes of cycles 2501 valid = false; 2502 } 2503 2504 // if no change in CPU number or frequency, then record sample for cycle statistics 2505 if (valid && mCpukHz > 0) { 2506 double cycles = wcNs * cpukHz * 0.000001; 2507 mHzStats.sample(cycles); 2508 } 2509 2510 unsigned n = mWcStats.n(); 2511 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2512 if ((n & 127) == 1) { 2513 long long elapsed = mCpuUsage.elapsed(); 2514 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2515 double perLoop = elapsed / (double) n; 2516 double perLoop100 = perLoop * 0.01; 2517 double perLoop1k = perLoop * 0.001; 2518 double mean = mWcStats.mean(); 2519 double stddev = mWcStats.stddev(); 2520 double minimum = mWcStats.minimum(); 2521 double maximum = mWcStats.maximum(); 2522 double meanCycles = mHzStats.mean(); 2523 double stddevCycles = mHzStats.stddev(); 2524 double minCycles = mHzStats.minimum(); 2525 double maxCycles = mHzStats.maximum(); 2526 mCpuUsage.resetElapsed(); 2527 mWcStats.reset(); 2528 mHzStats.reset(); 2529 ALOGD("CPU usage for %s over past %.1f secs\n" 2530 " (%u mixer loops at %.1f mean ms per loop):\n" 2531 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2532 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2533 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2534 title.string(), 2535 elapsed * .000000001, n, perLoop * .000001, 2536 mean * .001, 2537 stddev * .001, 2538 minimum * .001, 2539 maximum * .001, 2540 mean / perLoop100, 2541 stddev / perLoop100, 2542 minimum / perLoop100, 2543 maximum / perLoop100, 2544 meanCycles / perLoop1k, 2545 stddevCycles / perLoop1k, 2546 minCycles / perLoop1k, 2547 maxCycles / perLoop1k); 2548 2549 } 2550 } 2551#endif 2552}; 2553 2554void AudioFlinger::PlaybackThread::checkSilentMode_l() 2555{ 2556 if (!mMasterMute) { 2557 char value[PROPERTY_VALUE_MAX]; 2558 if (property_get("ro.audio.silent", value, "0") > 0) { 2559 char *endptr; 2560 unsigned long ul = strtoul(value, &endptr, 0); 2561 if (*endptr == '\0' && ul != 0) { 2562 ALOGD("Silence is golden"); 2563 // The setprop command will not allow a property to be changed after 2564 // the first time it is set, so we don't have to worry about un-muting. 2565 setMasterMute_l(true); 2566 } 2567 } 2568 } 2569} 2570 2571bool AudioFlinger::PlaybackThread::threadLoop() 2572{ 2573 Vector< sp<Track> > tracksToRemove; 2574 2575 standbyTime = systemTime(); 2576 2577 // MIXER 2578 nsecs_t lastWarning = 0; 2579 2580 // DUPLICATING 2581 // FIXME could this be made local to while loop? 2582 writeFrames = 0; 2583 2584 cacheParameters_l(); 2585 sleepTime = idleSleepTime; 2586 2587 if (mType == MIXER) { 2588 sleepTimeShift = 0; 2589 } 2590 2591 CpuStats cpuStats; 2592 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2593 2594 acquireWakeLock(); 2595 2596 while (!exitPending()) 2597 { 2598 cpuStats.sample(myName); 2599 2600 Vector< sp<EffectChain> > effectChains; 2601 2602 processConfigEvents(); 2603 2604 { // scope for mLock 2605 2606 Mutex::Autolock _l(mLock); 2607 2608 if (checkForNewParameters_l()) { 2609 cacheParameters_l(); 2610 } 2611 2612 saveOutputTracks(); 2613 2614 // put audio hardware into standby after short delay 2615 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2616 isSuspended())) { 2617 if (!mStandby) { 2618 2619 threadLoop_standby(); 2620 2621 mStandby = true; 2622 } 2623 2624 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2625 // we're about to wait, flush the binder command buffer 2626 IPCThreadState::self()->flushCommands(); 2627 2628 clearOutputTracks(); 2629 2630 if (exitPending()) break; 2631 2632 releaseWakeLock_l(); 2633 // wait until we have something to do... 2634 ALOGV("%s going to sleep", myName.string()); 2635 mWaitWorkCV.wait(mLock); 2636 ALOGV("%s waking up", myName.string()); 2637 acquireWakeLock_l(); 2638 2639 mMixerStatus = MIXER_IDLE; 2640 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2641 mBytesWritten = 0; 2642 2643 checkSilentMode_l(); 2644 2645 standbyTime = systemTime() + standbyDelay; 2646 sleepTime = idleSleepTime; 2647 if (mType == MIXER) { 2648 sleepTimeShift = 0; 2649 } 2650 2651 continue; 2652 } 2653 } 2654 2655 // mMixerStatusIgnoringFastTracks is also updated internally 2656 mMixerStatus = prepareTracks_l(&tracksToRemove); 2657 2658 // prevent any changes in effect chain list and in each effect chain 2659 // during mixing and effect process as the audio buffers could be deleted 2660 // or modified if an effect is created or deleted 2661 lockEffectChains_l(effectChains); 2662 } 2663 2664 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2665 threadLoop_mix(); 2666 } else { 2667 threadLoop_sleepTime(); 2668 } 2669 2670 if (isSuspended()) { 2671 sleepTime = suspendSleepTimeUs(); 2672 mBytesWritten += mixBufferSize; 2673 } 2674 2675 // only process effects if we're going to write 2676 if (sleepTime == 0) { 2677 for (size_t i = 0; i < effectChains.size(); i ++) { 2678 effectChains[i]->process_l(); 2679 } 2680 } 2681 2682 // enable changes in effect chain 2683 unlockEffectChains(effectChains); 2684 2685 // sleepTime == 0 means we must write to audio hardware 2686 if (sleepTime == 0) { 2687 2688 threadLoop_write(); 2689 2690if (mType == MIXER) { 2691 // write blocked detection 2692 nsecs_t now = systemTime(); 2693 nsecs_t delta = now - mLastWriteTime; 2694 if (!mStandby && delta > maxPeriod) { 2695 mNumDelayedWrites++; 2696 if ((now - lastWarning) > kWarningThrottleNs) { 2697#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2698 ScopedTrace st(ATRACE_TAG, "underrun"); 2699#endif 2700 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2701 ns2ms(delta), mNumDelayedWrites, this); 2702 lastWarning = now; 2703 } 2704 } 2705} 2706 2707 mStandby = false; 2708 } else { 2709 usleep(sleepTime); 2710 } 2711 2712 // Finally let go of removed track(s), without the lock held 2713 // since we can't guarantee the destructors won't acquire that 2714 // same lock. This will also mutate and push a new fast mixer state. 2715 threadLoop_removeTracks(tracksToRemove); 2716 tracksToRemove.clear(); 2717 2718 // FIXME I don't understand the need for this here; 2719 // it was in the original code but maybe the 2720 // assignment in saveOutputTracks() makes this unnecessary? 2721 clearOutputTracks(); 2722 2723 // Effect chains will be actually deleted here if they were removed from 2724 // mEffectChains list during mixing or effects processing 2725 effectChains.clear(); 2726 2727 // FIXME Note that the above .clear() is no longer necessary since effectChains 2728 // is now local to this block, but will keep it for now (at least until merge done). 2729 } 2730 2731 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2732 if (mType == MIXER || mType == DIRECT) { 2733 // put output stream into standby mode 2734 if (!mStandby) { 2735 mOutput->stream->common.standby(&mOutput->stream->common); 2736 } 2737 } 2738 2739 releaseWakeLock(); 2740 2741 ALOGV("Thread %p type %d exiting", this, mType); 2742 return false; 2743} 2744 2745void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2746{ 2747 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2748} 2749 2750void AudioFlinger::MixerThread::threadLoop_write() 2751{ 2752 // FIXME we should only do one push per cycle; confirm this is true 2753 // Start the fast mixer if it's not already running 2754 if (mFastMixer != NULL) { 2755 FastMixerStateQueue *sq = mFastMixer->sq(); 2756 FastMixerState *state = sq->begin(); 2757 if (state->mCommand != FastMixerState::MIX_WRITE && 2758 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2759 if (state->mCommand == FastMixerState::COLD_IDLE) { 2760 int32_t old = android_atomic_inc(&mFastMixerFutex); 2761 if (old == -1) { 2762 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2763 } 2764#ifdef AUDIO_WATCHDOG 2765 if (mAudioWatchdog != 0) { 2766 mAudioWatchdog->resume(); 2767 } 2768#endif 2769 } 2770 state->mCommand = FastMixerState::MIX_WRITE; 2771 sq->end(); 2772 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2773 if (kUseFastMixer == FastMixer_Dynamic) { 2774 mNormalSink = mPipeSink; 2775 } 2776 } else { 2777 sq->end(false /*didModify*/); 2778 } 2779 } 2780 PlaybackThread::threadLoop_write(); 2781} 2782 2783// shared by MIXER and DIRECT, overridden by DUPLICATING 2784void AudioFlinger::PlaybackThread::threadLoop_write() 2785{ 2786 // FIXME rewrite to reduce number of system calls 2787 mLastWriteTime = systemTime(); 2788 mInWrite = true; 2789 int bytesWritten; 2790 2791 // If an NBAIO sink is present, use it to write the normal mixer's submix 2792 if (mNormalSink != 0) { 2793#define mBitShift 2 // FIXME 2794 size_t count = mixBufferSize >> mBitShift; 2795#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2796 Tracer::traceBegin(ATRACE_TAG, "write"); 2797#endif 2798 // update the setpoint when gScreenState changes 2799 uint32_t screenState = gScreenState; 2800 if (screenState != mScreenState) { 2801 mScreenState = screenState; 2802 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2803 if (pipe != NULL) { 2804 pipe->setAvgFrames((mScreenState & 1) ? 2805 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2806 } 2807 } 2808 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2809#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2810 Tracer::traceEnd(ATRACE_TAG); 2811#endif 2812 if (framesWritten > 0) { 2813 bytesWritten = framesWritten << mBitShift; 2814 } else { 2815 bytesWritten = framesWritten; 2816 } 2817 // otherwise use the HAL / AudioStreamOut directly 2818 } else { 2819 // Direct output thread. 2820 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2821 } 2822 2823 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2824 mNumWrites++; 2825 mInWrite = false; 2826} 2827 2828void AudioFlinger::MixerThread::threadLoop_standby() 2829{ 2830 // Idle the fast mixer if it's currently running 2831 if (mFastMixer != NULL) { 2832 FastMixerStateQueue *sq = mFastMixer->sq(); 2833 FastMixerState *state = sq->begin(); 2834 if (!(state->mCommand & FastMixerState::IDLE)) { 2835 state->mCommand = FastMixerState::COLD_IDLE; 2836 state->mColdFutexAddr = &mFastMixerFutex; 2837 state->mColdGen++; 2838 mFastMixerFutex = 0; 2839 sq->end(); 2840 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2841 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2842 if (kUseFastMixer == FastMixer_Dynamic) { 2843 mNormalSink = mOutputSink; 2844 } 2845#ifdef AUDIO_WATCHDOG 2846 if (mAudioWatchdog != 0) { 2847 mAudioWatchdog->pause(); 2848 } 2849#endif 2850 } else { 2851 sq->end(false /*didModify*/); 2852 } 2853 } 2854 PlaybackThread::threadLoop_standby(); 2855} 2856 2857// shared by MIXER and DIRECT, overridden by DUPLICATING 2858void AudioFlinger::PlaybackThread::threadLoop_standby() 2859{ 2860 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2861 mOutput->stream->common.standby(&mOutput->stream->common); 2862} 2863 2864void AudioFlinger::MixerThread::threadLoop_mix() 2865{ 2866 // obtain the presentation timestamp of the next output buffer 2867 int64_t pts; 2868 status_t status = INVALID_OPERATION; 2869 2870 if (mNormalSink != 0) { 2871 status = mNormalSink->getNextWriteTimestamp(&pts); 2872 } else { 2873 status = mOutputSink->getNextWriteTimestamp(&pts); 2874 } 2875 2876 if (status != NO_ERROR) { 2877 pts = AudioBufferProvider::kInvalidPTS; 2878 } 2879 2880 // mix buffers... 2881 mAudioMixer->process(pts); 2882 // increase sleep time progressively when application underrun condition clears. 2883 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2884 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2885 // such that we would underrun the audio HAL. 2886 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2887 sleepTimeShift--; 2888 } 2889 sleepTime = 0; 2890 standbyTime = systemTime() + standbyDelay; 2891 //TODO: delay standby when effects have a tail 2892} 2893 2894void AudioFlinger::MixerThread::threadLoop_sleepTime() 2895{ 2896 // If no tracks are ready, sleep once for the duration of an output 2897 // buffer size, then write 0s to the output 2898 if (sleepTime == 0) { 2899 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2900 sleepTime = activeSleepTime >> sleepTimeShift; 2901 if (sleepTime < kMinThreadSleepTimeUs) { 2902 sleepTime = kMinThreadSleepTimeUs; 2903 } 2904 // reduce sleep time in case of consecutive application underruns to avoid 2905 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2906 // duration we would end up writing less data than needed by the audio HAL if 2907 // the condition persists. 2908 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2909 sleepTimeShift++; 2910 } 2911 } else { 2912 sleepTime = idleSleepTime; 2913 } 2914 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2915 memset (mMixBuffer, 0, mixBufferSize); 2916 sleepTime = 0; 2917 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2918 "anticipated start"); 2919 } 2920 // TODO add standby time extension fct of effect tail 2921} 2922 2923// prepareTracks_l() must be called with ThreadBase::mLock held 2924AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2925 Vector< sp<Track> > *tracksToRemove) 2926{ 2927 2928 mixer_state mixerStatus = MIXER_IDLE; 2929 // find out which tracks need to be processed 2930 size_t count = mActiveTracks.size(); 2931 size_t mixedTracks = 0; 2932 size_t tracksWithEffect = 0; 2933 // counts only _active_ fast tracks 2934 size_t fastTracks = 0; 2935 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2936 2937 float masterVolume = mMasterVolume; 2938 bool masterMute = mMasterMute; 2939 2940 if (masterMute) { 2941 masterVolume = 0; 2942 } 2943 // Delegate master volume control to effect in output mix effect chain if needed 2944 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2945 if (chain != 0) { 2946 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2947 chain->setVolume_l(&v, &v); 2948 masterVolume = (float)((v + (1 << 23)) >> 24); 2949 chain.clear(); 2950 } 2951 2952 // prepare a new state to push 2953 FastMixerStateQueue *sq = NULL; 2954 FastMixerState *state = NULL; 2955 bool didModify = false; 2956 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2957 if (mFastMixer != NULL) { 2958 sq = mFastMixer->sq(); 2959 state = sq->begin(); 2960 } 2961 2962 for (size_t i=0 ; i<count ; i++) { 2963 sp<Track> t = mActiveTracks[i].promote(); 2964 if (t == 0) continue; 2965 2966 // this const just means the local variable doesn't change 2967 Track* const track = t.get(); 2968 2969 // process fast tracks 2970 if (track->isFastTrack()) { 2971 2972 // It's theoretically possible (though unlikely) for a fast track to be created 2973 // and then removed within the same normal mix cycle. This is not a problem, as 2974 // the track never becomes active so it's fast mixer slot is never touched. 2975 // The converse, of removing an (active) track and then creating a new track 2976 // at the identical fast mixer slot within the same normal mix cycle, 2977 // is impossible because the slot isn't marked available until the end of each cycle. 2978 int j = track->mFastIndex; 2979 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2980 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2981 FastTrack *fastTrack = &state->mFastTracks[j]; 2982 2983 // Determine whether the track is currently in underrun condition, 2984 // and whether it had a recent underrun. 2985 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2986 FastTrackUnderruns underruns = ftDump->mUnderruns; 2987 uint32_t recentFull = (underruns.mBitFields.mFull - 2988 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2989 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2990 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2991 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2992 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2993 uint32_t recentUnderruns = recentPartial + recentEmpty; 2994 track->mObservedUnderruns = underruns; 2995 // don't count underruns that occur while stopping or pausing 2996 // or stopped which can occur when flush() is called while active 2997 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2998 track->mUnderrunCount += recentUnderruns; 2999 } 3000 3001 // This is similar to the state machine for normal tracks, 3002 // with a few modifications for fast tracks. 3003 bool isActive = true; 3004 switch (track->mState) { 3005 case TrackBase::STOPPING_1: 3006 // track stays active in STOPPING_1 state until first underrun 3007 if (recentUnderruns > 0) { 3008 track->mState = TrackBase::STOPPING_2; 3009 } 3010 break; 3011 case TrackBase::PAUSING: 3012 // ramp down is not yet implemented 3013 track->setPaused(); 3014 break; 3015 case TrackBase::RESUMING: 3016 // ramp up is not yet implemented 3017 track->mState = TrackBase::ACTIVE; 3018 break; 3019 case TrackBase::ACTIVE: 3020 if (recentFull > 0 || recentPartial > 0) { 3021 // track has provided at least some frames recently: reset retry count 3022 track->mRetryCount = kMaxTrackRetries; 3023 } 3024 if (recentUnderruns == 0) { 3025 // no recent underruns: stay active 3026 break; 3027 } 3028 // there has recently been an underrun of some kind 3029 if (track->sharedBuffer() == 0) { 3030 // were any of the recent underruns "empty" (no frames available)? 3031 if (recentEmpty == 0) { 3032 // no, then ignore the partial underruns as they are allowed indefinitely 3033 break; 3034 } 3035 // there has recently been an "empty" underrun: decrement the retry counter 3036 if (--(track->mRetryCount) > 0) { 3037 break; 3038 } 3039 // indicate to client process that the track was disabled because of underrun; 3040 // it will then automatically call start() when data is available 3041 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 3042 // remove from active list, but state remains ACTIVE [confusing but true] 3043 isActive = false; 3044 break; 3045 } 3046 // fall through 3047 case TrackBase::STOPPING_2: 3048 case TrackBase::PAUSED: 3049 case TrackBase::TERMINATED: 3050 case TrackBase::STOPPED: 3051 case TrackBase::FLUSHED: // flush() while active 3052 // Check for presentation complete if track is inactive 3053 // We have consumed all the buffers of this track. 3054 // This would be incomplete if we auto-paused on underrun 3055 { 3056 size_t audioHALFrames = 3057 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3058 size_t framesWritten = mBytesWritten / mFrameSize; 3059 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3060 // track stays in active list until presentation is complete 3061 break; 3062 } 3063 } 3064 if (track->isStopping_2()) { 3065 track->mState = TrackBase::STOPPED; 3066 } 3067 if (track->isStopped()) { 3068 // Can't reset directly, as fast mixer is still polling this track 3069 // track->reset(); 3070 // So instead mark this track as needing to be reset after push with ack 3071 resetMask |= 1 << i; 3072 } 3073 isActive = false; 3074 break; 3075 case TrackBase::IDLE: 3076 default: 3077 LOG_FATAL("unexpected track state %d", track->mState); 3078 } 3079 3080 if (isActive) { 3081 // was it previously inactive? 3082 if (!(state->mTrackMask & (1 << j))) { 3083 ExtendedAudioBufferProvider *eabp = track; 3084 VolumeProvider *vp = track; 3085 fastTrack->mBufferProvider = eabp; 3086 fastTrack->mVolumeProvider = vp; 3087 fastTrack->mSampleRate = track->mSampleRate; 3088 fastTrack->mChannelMask = track->mChannelMask; 3089 fastTrack->mGeneration++; 3090 state->mTrackMask |= 1 << j; 3091 didModify = true; 3092 // no acknowledgement required for newly active tracks 3093 } 3094 // cache the combined master volume and stream type volume for fast mixer; this 3095 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3096 track->mCachedVolume = track->isMuted() ? 3097 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3098 ++fastTracks; 3099 } else { 3100 // was it previously active? 3101 if (state->mTrackMask & (1 << j)) { 3102 fastTrack->mBufferProvider = NULL; 3103 fastTrack->mGeneration++; 3104 state->mTrackMask &= ~(1 << j); 3105 didModify = true; 3106 // If any fast tracks were removed, we must wait for acknowledgement 3107 // because we're about to decrement the last sp<> on those tracks. 3108 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3109 } else { 3110 LOG_FATAL("fast track %d should have been active", j); 3111 } 3112 tracksToRemove->add(track); 3113 // Avoids a misleading display in dumpsys 3114 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3115 } 3116 continue; 3117 } 3118 3119 { // local variable scope to avoid goto warning 3120 3121 audio_track_cblk_t* cblk = track->cblk(); 3122 3123 // The first time a track is added we wait 3124 // for all its buffers to be filled before processing it 3125 int name = track->name(); 3126 // make sure that we have enough frames to mix one full buffer. 3127 // enforce this condition only once to enable draining the buffer in case the client 3128 // app does not call stop() and relies on underrun to stop: 3129 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3130 // during last round 3131 uint32_t minFrames = 1; 3132 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3133 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3134 if (t->sampleRate() == mSampleRate) { 3135 minFrames = mNormalFrameCount; 3136 } else { 3137 // +1 for rounding and +1 for additional sample needed for interpolation 3138 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3139 // add frames already consumed but not yet released by the resampler 3140 // because cblk->framesReady() will include these frames 3141 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3142 // the minimum track buffer size is normally twice the number of frames necessary 3143 // to fill one buffer and the resampler should not leave more than one buffer worth 3144 // of unreleased frames after each pass, but just in case... 3145 ALOG_ASSERT(minFrames <= cblk->frameCount); 3146 } 3147 } 3148 if ((track->framesReady() >= minFrames) && track->isReady() && 3149 !track->isPaused() && !track->isTerminated()) 3150 { 3151 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 3152 this); 3153 3154 mixedTracks++; 3155 3156 // track->mainBuffer() != mMixBuffer means there is an effect chain 3157 // connected to the track 3158 chain.clear(); 3159 if (track->mainBuffer() != mMixBuffer) { 3160 chain = getEffectChain_l(track->sessionId()); 3161 // Delegate volume control to effect in track effect chain if needed 3162 if (chain != 0) { 3163 tracksWithEffect++; 3164 } else { 3165 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3166 "session %d", 3167 name, track->sessionId()); 3168 } 3169 } 3170 3171 3172 int param = AudioMixer::VOLUME; 3173 if (track->mFillingUpStatus == Track::FS_FILLED) { 3174 // no ramp for the first volume setting 3175 track->mFillingUpStatus = Track::FS_ACTIVE; 3176 if (track->mState == TrackBase::RESUMING) { 3177 track->mState = TrackBase::ACTIVE; 3178 param = AudioMixer::RAMP_VOLUME; 3179 } 3180 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3181 } else if (cblk->server != 0) { 3182 // If the track is stopped before the first frame was mixed, 3183 // do not apply ramp 3184 param = AudioMixer::RAMP_VOLUME; 3185 } 3186 3187 // compute volume for this track 3188 uint32_t vl, vr, va; 3189 if (track->isMuted() || track->isPausing() || 3190 mStreamTypes[track->streamType()].mute) { 3191 vl = vr = va = 0; 3192 if (track->isPausing()) { 3193 track->setPaused(); 3194 } 3195 } else { 3196 3197 // read original volumes with volume control 3198 float typeVolume = mStreamTypes[track->streamType()].volume; 3199 float v = masterVolume * typeVolume; 3200 uint32_t vlr = cblk->getVolumeLR(); 3201 vl = vlr & 0xFFFF; 3202 vr = vlr >> 16; 3203 // track volumes come from shared memory, so can't be trusted and must be clamped 3204 if (vl > MAX_GAIN_INT) { 3205 ALOGV("Track left volume out of range: %04X", vl); 3206 vl = MAX_GAIN_INT; 3207 } 3208 if (vr > MAX_GAIN_INT) { 3209 ALOGV("Track right volume out of range: %04X", vr); 3210 vr = MAX_GAIN_INT; 3211 } 3212 // now apply the master volume and stream type volume 3213 vl = (uint32_t)(v * vl) << 12; 3214 vr = (uint32_t)(v * vr) << 12; 3215 // assuming master volume and stream type volume each go up to 1.0, 3216 // vl and vr are now in 8.24 format 3217 3218 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3219 // send level comes from shared memory and so may be corrupt 3220 if (sendLevel > MAX_GAIN_INT) { 3221 ALOGV("Track send level out of range: %04X", sendLevel); 3222 sendLevel = MAX_GAIN_INT; 3223 } 3224 va = (uint32_t)(v * sendLevel); 3225 } 3226 // Delegate volume control to effect in track effect chain if needed 3227 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3228 // Do not ramp volume if volume is controlled by effect 3229 param = AudioMixer::VOLUME; 3230 track->mHasVolumeController = true; 3231 } else { 3232 // force no volume ramp when volume controller was just disabled or removed 3233 // from effect chain to avoid volume spike 3234 if (track->mHasVolumeController) { 3235 param = AudioMixer::VOLUME; 3236 } 3237 track->mHasVolumeController = false; 3238 } 3239 3240 // Convert volumes from 8.24 to 4.12 format 3241 // This additional clamping is needed in case chain->setVolume_l() overshot 3242 vl = (vl + (1 << 11)) >> 12; 3243 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3244 vr = (vr + (1 << 11)) >> 12; 3245 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3246 3247 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3248 3249 // XXX: these things DON'T need to be done each time 3250 mAudioMixer->setBufferProvider(name, track); 3251 mAudioMixer->enable(name); 3252 3253 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3254 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3255 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3256 mAudioMixer->setParameter( 3257 name, 3258 AudioMixer::TRACK, 3259 AudioMixer::FORMAT, (void *)track->format()); 3260 mAudioMixer->setParameter( 3261 name, 3262 AudioMixer::TRACK, 3263 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3264 mAudioMixer->setParameter( 3265 name, 3266 AudioMixer::RESAMPLE, 3267 AudioMixer::SAMPLE_RATE, 3268 (void *)(cblk->sampleRate)); 3269 mAudioMixer->setParameter( 3270 name, 3271 AudioMixer::TRACK, 3272 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3273 mAudioMixer->setParameter( 3274 name, 3275 AudioMixer::TRACK, 3276 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3277 3278 // reset retry count 3279 track->mRetryCount = kMaxTrackRetries; 3280 3281 // If one track is ready, set the mixer ready if: 3282 // - the mixer was not ready during previous round OR 3283 // - no other track is not ready 3284 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3285 mixerStatus != MIXER_TRACKS_ENABLED) { 3286 mixerStatus = MIXER_TRACKS_READY; 3287 } 3288 } else { 3289 // clear effect chain input buffer if an active track underruns to avoid sending 3290 // previous audio buffer again to effects 3291 chain = getEffectChain_l(track->sessionId()); 3292 if (chain != 0) { 3293 chain->clearInputBuffer(); 3294 } 3295 3296 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 3297 cblk->server, this); 3298 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3299 track->isStopped() || track->isPaused()) { 3300 // We have consumed all the buffers of this track. 3301 // Remove it from the list of active tracks. 3302 // TODO: use actual buffer filling status instead of latency when available from 3303 // audio HAL 3304 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3305 size_t framesWritten = mBytesWritten / mFrameSize; 3306 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3307 if (track->isStopped()) { 3308 track->reset(); 3309 } 3310 tracksToRemove->add(track); 3311 } 3312 } else { 3313 track->mUnderrunCount++; 3314 // No buffers for this track. Give it a few chances to 3315 // fill a buffer, then remove it from active list. 3316 if (--(track->mRetryCount) <= 0) { 3317 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3318 tracksToRemove->add(track); 3319 // indicate to client process that the track was disabled because of underrun; 3320 // it will then automatically call start() when data is available 3321 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3322 // If one track is not ready, mark the mixer also not ready if: 3323 // - the mixer was ready during previous round OR 3324 // - no other track is ready 3325 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3326 mixerStatus != MIXER_TRACKS_READY) { 3327 mixerStatus = MIXER_TRACKS_ENABLED; 3328 } 3329 } 3330 mAudioMixer->disable(name); 3331 } 3332 3333 } // local variable scope to avoid goto warning 3334track_is_ready: ; 3335 3336 } 3337 3338 // Push the new FastMixer state if necessary 3339 bool pauseAudioWatchdog = false; 3340 if (didModify) { 3341 state->mFastTracksGen++; 3342 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3343 if (kUseFastMixer == FastMixer_Dynamic && 3344 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3345 state->mCommand = FastMixerState::COLD_IDLE; 3346 state->mColdFutexAddr = &mFastMixerFutex; 3347 state->mColdGen++; 3348 mFastMixerFutex = 0; 3349 if (kUseFastMixer == FastMixer_Dynamic) { 3350 mNormalSink = mOutputSink; 3351 } 3352 // If we go into cold idle, need to wait for acknowledgement 3353 // so that fast mixer stops doing I/O. 3354 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3355 pauseAudioWatchdog = true; 3356 } 3357 sq->end(); 3358 } 3359 if (sq != NULL) { 3360 sq->end(didModify); 3361 sq->push(block); 3362 } 3363#ifdef AUDIO_WATCHDOG 3364 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3365 mAudioWatchdog->pause(); 3366 } 3367#endif 3368 3369 // Now perform the deferred reset on fast tracks that have stopped 3370 while (resetMask != 0) { 3371 size_t i = __builtin_ctz(resetMask); 3372 ALOG_ASSERT(i < count); 3373 resetMask &= ~(1 << i); 3374 sp<Track> t = mActiveTracks[i].promote(); 3375 if (t == 0) continue; 3376 Track* track = t.get(); 3377 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3378 track->reset(); 3379 } 3380 3381 // remove all the tracks that need to be... 3382 count = tracksToRemove->size(); 3383 if (CC_UNLIKELY(count)) { 3384 for (size_t i=0 ; i<count ; i++) { 3385 const sp<Track>& track = tracksToRemove->itemAt(i); 3386 mActiveTracks.remove(track); 3387 if (track->mainBuffer() != mMixBuffer) { 3388 chain = getEffectChain_l(track->sessionId()); 3389 if (chain != 0) { 3390 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3391 track->sessionId()); 3392 chain->decActiveTrackCnt(); 3393 } 3394 } 3395 if (track->isTerminated()) { 3396 removeTrack_l(track); 3397 } 3398 } 3399 } 3400 3401 // mix buffer must be cleared if all tracks are connected to an 3402 // effect chain as in this case the mixer will not write to 3403 // mix buffer and track effects will accumulate into it 3404 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3405 (mixedTracks == 0 && fastTracks > 0)) { 3406 // FIXME as a performance optimization, should remember previous zero status 3407 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3408 } 3409 3410 // if any fast tracks, then status is ready 3411 mMixerStatusIgnoringFastTracks = mixerStatus; 3412 if (fastTracks > 0) { 3413 mixerStatus = MIXER_TRACKS_READY; 3414 } 3415 return mixerStatus; 3416} 3417 3418/* 3419The derived values that are cached: 3420 - mixBufferSize from frame count * frame size 3421 - activeSleepTime from activeSleepTimeUs() 3422 - idleSleepTime from idleSleepTimeUs() 3423 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3424 - maxPeriod from frame count and sample rate (MIXER only) 3425 3426The parameters that affect these derived values are: 3427 - frame count 3428 - frame size 3429 - sample rate 3430 - device type: A2DP or not 3431 - device latency 3432 - format: PCM or not 3433 - active sleep time 3434 - idle sleep time 3435*/ 3436 3437void AudioFlinger::PlaybackThread::cacheParameters_l() 3438{ 3439 mixBufferSize = mNormalFrameCount * mFrameSize; 3440 activeSleepTime = activeSleepTimeUs(); 3441 idleSleepTime = idleSleepTimeUs(); 3442} 3443 3444void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3445{ 3446 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3447 this, streamType, mTracks.size()); 3448 Mutex::Autolock _l(mLock); 3449 3450 size_t size = mTracks.size(); 3451 for (size_t i = 0; i < size; i++) { 3452 sp<Track> t = mTracks[i]; 3453 if (t->streamType() == streamType) { 3454 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 3455 t->mCblk->cv.signal(); 3456 } 3457 } 3458} 3459 3460// getTrackName_l() must be called with ThreadBase::mLock held 3461int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3462{ 3463 return mAudioMixer->getTrackName(channelMask, sessionId); 3464} 3465 3466// deleteTrackName_l() must be called with ThreadBase::mLock held 3467void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3468{ 3469 ALOGV("remove track (%d) and delete from mixer", name); 3470 mAudioMixer->deleteTrackName(name); 3471} 3472 3473// checkForNewParameters_l() must be called with ThreadBase::mLock held 3474bool AudioFlinger::MixerThread::checkForNewParameters_l() 3475{ 3476 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3477 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3478 bool reconfig = false; 3479 3480 while (!mNewParameters.isEmpty()) { 3481 3482 if (mFastMixer != NULL) { 3483 FastMixerStateQueue *sq = mFastMixer->sq(); 3484 FastMixerState *state = sq->begin(); 3485 if (!(state->mCommand & FastMixerState::IDLE)) { 3486 previousCommand = state->mCommand; 3487 state->mCommand = FastMixerState::HOT_IDLE; 3488 sq->end(); 3489 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3490 } else { 3491 sq->end(false /*didModify*/); 3492 } 3493 } 3494 3495 status_t status = NO_ERROR; 3496 String8 keyValuePair = mNewParameters[0]; 3497 AudioParameter param = AudioParameter(keyValuePair); 3498 int value; 3499 3500 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3501 reconfig = true; 3502 } 3503 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3504 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3505 status = BAD_VALUE; 3506 } else { 3507 reconfig = true; 3508 } 3509 } 3510 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3511 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3512 status = BAD_VALUE; 3513 } else { 3514 reconfig = true; 3515 } 3516 } 3517 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3518 // do not accept frame count changes if tracks are open as the track buffer 3519 // size depends on frame count and correct behavior would not be guaranteed 3520 // if frame count is changed after track creation 3521 if (!mTracks.isEmpty()) { 3522 status = INVALID_OPERATION; 3523 } else { 3524 reconfig = true; 3525 } 3526 } 3527 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3528#ifdef ADD_BATTERY_DATA 3529 // when changing the audio output device, call addBatteryData to notify 3530 // the change 3531 if (mOutDevice != value) { 3532 uint32_t params = 0; 3533 // check whether speaker is on 3534 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3535 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3536 } 3537 3538 audio_devices_t deviceWithoutSpeaker 3539 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3540 // check if any other device (except speaker) is on 3541 if (value & deviceWithoutSpeaker ) { 3542 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3543 } 3544 3545 if (params != 0) { 3546 addBatteryData(params); 3547 } 3548 } 3549#endif 3550 3551 // forward device change to effects that have requested to be 3552 // aware of attached audio device. 3553 mOutDevice = value; 3554 for (size_t i = 0; i < mEffectChains.size(); i++) { 3555 mEffectChains[i]->setDevice_l(mOutDevice); 3556 } 3557 } 3558 3559 if (status == NO_ERROR) { 3560 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3561 keyValuePair.string()); 3562 if (!mStandby && status == INVALID_OPERATION) { 3563 mOutput->stream->common.standby(&mOutput->stream->common); 3564 mStandby = true; 3565 mBytesWritten = 0; 3566 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3567 keyValuePair.string()); 3568 } 3569 if (status == NO_ERROR && reconfig) { 3570 delete mAudioMixer; 3571 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3572 mAudioMixer = NULL; 3573 readOutputParameters(); 3574 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3575 for (size_t i = 0; i < mTracks.size() ; i++) { 3576 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3577 if (name < 0) break; 3578 mTracks[i]->mName = name; 3579 // limit track sample rate to 2 x new output sample rate 3580 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3581 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3582 } 3583 } 3584 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3585 } 3586 } 3587 3588 mNewParameters.removeAt(0); 3589 3590 mParamStatus = status; 3591 mParamCond.signal(); 3592 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3593 // already timed out waiting for the status and will never signal the condition. 3594 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3595 } 3596 3597 if (!(previousCommand & FastMixerState::IDLE)) { 3598 ALOG_ASSERT(mFastMixer != NULL); 3599 FastMixerStateQueue *sq = mFastMixer->sq(); 3600 FastMixerState *state = sq->begin(); 3601 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3602 state->mCommand = previousCommand; 3603 sq->end(); 3604 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3605 } 3606 3607 return reconfig; 3608} 3609 3610void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3611{ 3612 NBAIO_Source *teeSource = source.get(); 3613 if (teeSource != NULL) { 3614 char teeTime[16]; 3615 struct timeval tv; 3616 gettimeofday(&tv, NULL); 3617 struct tm tm; 3618 localtime_r(&tv.tv_sec, &tm); 3619 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3620 char teePath[64]; 3621 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3622 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3623 if (teeFd >= 0) { 3624 char wavHeader[44]; 3625 memcpy(wavHeader, 3626 "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", 3627 sizeof(wavHeader)); 3628 NBAIO_Format format = teeSource->format(); 3629 unsigned channelCount = Format_channelCount(format); 3630 ALOG_ASSERT(channelCount <= FCC_2); 3631 uint32_t sampleRate = Format_sampleRate(format); 3632 wavHeader[22] = channelCount; // number of channels 3633 wavHeader[24] = sampleRate; // sample rate 3634 wavHeader[25] = sampleRate >> 8; 3635 wavHeader[32] = channelCount * 2; // block alignment 3636 write(teeFd, wavHeader, sizeof(wavHeader)); 3637 size_t total = 0; 3638 bool firstRead = true; 3639 for (;;) { 3640#define TEE_SINK_READ 1024 3641 short buffer[TEE_SINK_READ * FCC_2]; 3642 size_t count = TEE_SINK_READ; 3643 ssize_t actual = teeSource->read(buffer, count, 3644 AudioBufferProvider::kInvalidPTS); 3645 bool wasFirstRead = firstRead; 3646 firstRead = false; 3647 if (actual <= 0) { 3648 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3649 continue; 3650 } 3651 break; 3652 } 3653 ALOG_ASSERT(actual <= (ssize_t)count); 3654 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3655 total += actual; 3656 } 3657 lseek(teeFd, (off_t) 4, SEEK_SET); 3658 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3659 write(teeFd, &temp, sizeof(temp)); 3660 lseek(teeFd, (off_t) 40, SEEK_SET); 3661 temp = total * channelCount * sizeof(short); 3662 write(teeFd, &temp, sizeof(temp)); 3663 close(teeFd); 3664 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3665 } else { 3666 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3667 } 3668 } 3669} 3670 3671void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3672{ 3673 const size_t SIZE = 256; 3674 char buffer[SIZE]; 3675 String8 result; 3676 3677 PlaybackThread::dumpInternals(fd, args); 3678 3679 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3680 result.append(buffer); 3681 write(fd, result.string(), result.size()); 3682 3683 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3684 FastMixerDumpState copy = mFastMixerDumpState; 3685 copy.dump(fd); 3686 3687#ifdef STATE_QUEUE_DUMP 3688 // Similar for state queue 3689 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3690 observerCopy.dump(fd); 3691 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3692 mutatorCopy.dump(fd); 3693#endif 3694 3695 // Write the tee output to a .wav file 3696 dumpTee(fd, mTeeSource, mId); 3697 3698#ifdef AUDIO_WATCHDOG 3699 if (mAudioWatchdog != 0) { 3700 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3701 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3702 wdCopy.dump(fd); 3703 } 3704#endif 3705} 3706 3707uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3708{ 3709 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3710} 3711 3712uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3713{ 3714 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3715} 3716 3717void AudioFlinger::MixerThread::cacheParameters_l() 3718{ 3719 PlaybackThread::cacheParameters_l(); 3720 3721 // FIXME: Relaxed timing because of a certain device that can't meet latency 3722 // Should be reduced to 2x after the vendor fixes the driver issue 3723 // increase threshold again due to low power audio mode. The way this warning 3724 // threshold is calculated and its usefulness should be reconsidered anyway. 3725 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3726} 3727 3728// ---------------------------------------------------------------------------- 3729AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3730 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3731 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3732 // mLeftVolFloat, mRightVolFloat 3733{ 3734} 3735 3736AudioFlinger::DirectOutputThread::~DirectOutputThread() 3737{ 3738} 3739 3740AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3741 Vector< sp<Track> > *tracksToRemove 3742) 3743{ 3744 sp<Track> trackToRemove; 3745 3746 mixer_state mixerStatus = MIXER_IDLE; 3747 3748 // find out which tracks need to be processed 3749 if (mActiveTracks.size() != 0) { 3750 sp<Track> t = mActiveTracks[0].promote(); 3751 // The track died recently 3752 if (t == 0) return MIXER_IDLE; 3753 3754 Track* const track = t.get(); 3755 audio_track_cblk_t* cblk = track->cblk(); 3756 3757 // The first time a track is added we wait 3758 // for all its buffers to be filled before processing it 3759 uint32_t minFrames; 3760 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3761 minFrames = mNormalFrameCount; 3762 } else { 3763 minFrames = 1; 3764 } 3765 if ((track->framesReady() >= minFrames) && track->isReady() && 3766 !track->isPaused() && !track->isTerminated()) 3767 { 3768 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3769 3770 if (track->mFillingUpStatus == Track::FS_FILLED) { 3771 track->mFillingUpStatus = Track::FS_ACTIVE; 3772 mLeftVolFloat = mRightVolFloat = 0; 3773 if (track->mState == TrackBase::RESUMING) { 3774 track->mState = TrackBase::ACTIVE; 3775 } 3776 } 3777 3778 // compute volume for this track 3779 float left, right; 3780 if (track->isMuted() || mMasterMute || track->isPausing() || 3781 mStreamTypes[track->streamType()].mute) { 3782 left = right = 0; 3783 if (track->isPausing()) { 3784 track->setPaused(); 3785 } 3786 } else { 3787 float typeVolume = mStreamTypes[track->streamType()].volume; 3788 float v = mMasterVolume * typeVolume; 3789 uint32_t vlr = cblk->getVolumeLR(); 3790 float v_clamped = v * (vlr & 0xFFFF); 3791 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3792 left = v_clamped/MAX_GAIN; 3793 v_clamped = v * (vlr >> 16); 3794 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3795 right = v_clamped/MAX_GAIN; 3796 } 3797 3798 if (left != mLeftVolFloat || right != mRightVolFloat) { 3799 mLeftVolFloat = left; 3800 mRightVolFloat = right; 3801 3802 // Convert volumes from float to 8.24 3803 uint32_t vl = (uint32_t)(left * (1 << 24)); 3804 uint32_t vr = (uint32_t)(right * (1 << 24)); 3805 3806 // Delegate volume control to effect in track effect chain if needed 3807 // only one effect chain can be present on DirectOutputThread, so if 3808 // there is one, the track is connected to it 3809 if (!mEffectChains.isEmpty()) { 3810 // Do not ramp volume if volume is controlled by effect 3811 mEffectChains[0]->setVolume_l(&vl, &vr); 3812 left = (float)vl / (1 << 24); 3813 right = (float)vr / (1 << 24); 3814 } 3815 mOutput->stream->set_volume(mOutput->stream, left, right); 3816 } 3817 3818 // reset retry count 3819 track->mRetryCount = kMaxTrackRetriesDirect; 3820 mActiveTrack = t; 3821 mixerStatus = MIXER_TRACKS_READY; 3822 } else { 3823 // clear effect chain input buffer if an active track underruns to avoid sending 3824 // previous audio buffer again to effects 3825 if (!mEffectChains.isEmpty()) { 3826 mEffectChains[0]->clearInputBuffer(); 3827 } 3828 3829 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3830 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3831 track->isStopped() || track->isPaused()) { 3832 // We have consumed all the buffers of this track. 3833 // Remove it from the list of active tracks. 3834 // TODO: implement behavior for compressed audio 3835 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3836 size_t framesWritten = mBytesWritten / mFrameSize; 3837 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3838 if (track->isStopped()) { 3839 track->reset(); 3840 } 3841 trackToRemove = track; 3842 } 3843 } else { 3844 // No buffers for this track. Give it a few chances to 3845 // fill a buffer, then remove it from active list. 3846 if (--(track->mRetryCount) <= 0) { 3847 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3848 trackToRemove = track; 3849 } else { 3850 mixerStatus = MIXER_TRACKS_ENABLED; 3851 } 3852 } 3853 } 3854 } 3855 3856 // FIXME merge this with similar code for removing multiple tracks 3857 // remove all the tracks that need to be... 3858 if (CC_UNLIKELY(trackToRemove != 0)) { 3859 tracksToRemove->add(trackToRemove); 3860 mActiveTracks.remove(trackToRemove); 3861 if (!mEffectChains.isEmpty()) { 3862 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3863 trackToRemove->sessionId()); 3864 mEffectChains[0]->decActiveTrackCnt(); 3865 } 3866 if (trackToRemove->isTerminated()) { 3867 removeTrack_l(trackToRemove); 3868 } 3869 } 3870 3871 return mixerStatus; 3872} 3873 3874void AudioFlinger::DirectOutputThread::threadLoop_mix() 3875{ 3876 AudioBufferProvider::Buffer buffer; 3877 size_t frameCount = mFrameCount; 3878 int8_t *curBuf = (int8_t *)mMixBuffer; 3879 // output audio to hardware 3880 while (frameCount) { 3881 buffer.frameCount = frameCount; 3882 mActiveTrack->getNextBuffer(&buffer); 3883 if (CC_UNLIKELY(buffer.raw == NULL)) { 3884 memset(curBuf, 0, frameCount * mFrameSize); 3885 break; 3886 } 3887 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3888 frameCount -= buffer.frameCount; 3889 curBuf += buffer.frameCount * mFrameSize; 3890 mActiveTrack->releaseBuffer(&buffer); 3891 } 3892 sleepTime = 0; 3893 standbyTime = systemTime() + standbyDelay; 3894 mActiveTrack.clear(); 3895 3896} 3897 3898void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3899{ 3900 if (sleepTime == 0) { 3901 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3902 sleepTime = activeSleepTime; 3903 } else { 3904 sleepTime = idleSleepTime; 3905 } 3906 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3907 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3908 sleepTime = 0; 3909 } 3910} 3911 3912// getTrackName_l() must be called with ThreadBase::mLock held 3913int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3914 int sessionId) 3915{ 3916 return 0; 3917} 3918 3919// deleteTrackName_l() must be called with ThreadBase::mLock held 3920void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3921{ 3922} 3923 3924// checkForNewParameters_l() must be called with ThreadBase::mLock held 3925bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3926{ 3927 bool reconfig = false; 3928 3929 while (!mNewParameters.isEmpty()) { 3930 status_t status = NO_ERROR; 3931 String8 keyValuePair = mNewParameters[0]; 3932 AudioParameter param = AudioParameter(keyValuePair); 3933 int value; 3934 3935 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3936 // do not accept frame count changes if tracks are open as the track buffer 3937 // size depends on frame count and correct behavior would not be garantied 3938 // if frame count is changed after track creation 3939 if (!mTracks.isEmpty()) { 3940 status = INVALID_OPERATION; 3941 } else { 3942 reconfig = true; 3943 } 3944 } 3945 if (status == NO_ERROR) { 3946 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3947 keyValuePair.string()); 3948 if (!mStandby && status == INVALID_OPERATION) { 3949 mOutput->stream->common.standby(&mOutput->stream->common); 3950 mStandby = true; 3951 mBytesWritten = 0; 3952 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3953 keyValuePair.string()); 3954 } 3955 if (status == NO_ERROR && reconfig) { 3956 readOutputParameters(); 3957 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3958 } 3959 } 3960 3961 mNewParameters.removeAt(0); 3962 3963 mParamStatus = status; 3964 mParamCond.signal(); 3965 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3966 // already timed out waiting for the status and will never signal the condition. 3967 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3968 } 3969 return reconfig; 3970} 3971 3972uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3973{ 3974 uint32_t time; 3975 if (audio_is_linear_pcm(mFormat)) { 3976 time = PlaybackThread::activeSleepTimeUs(); 3977 } else { 3978 time = 10000; 3979 } 3980 return time; 3981} 3982 3983uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3984{ 3985 uint32_t time; 3986 if (audio_is_linear_pcm(mFormat)) { 3987 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3988 } else { 3989 time = 10000; 3990 } 3991 return time; 3992} 3993 3994uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3995{ 3996 uint32_t time; 3997 if (audio_is_linear_pcm(mFormat)) { 3998 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3999 } else { 4000 time = 10000; 4001 } 4002 return time; 4003} 4004 4005void AudioFlinger::DirectOutputThread::cacheParameters_l() 4006{ 4007 PlaybackThread::cacheParameters_l(); 4008 4009 // use shorter standby delay as on normal output to release 4010 // hardware resources as soon as possible 4011 standbyDelay = microseconds(activeSleepTime*2); 4012} 4013 4014// ---------------------------------------------------------------------------- 4015 4016AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4017 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4018 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4019 DUPLICATING), 4020 mWaitTimeMs(UINT_MAX) 4021{ 4022 addOutputTrack(mainThread); 4023} 4024 4025AudioFlinger::DuplicatingThread::~DuplicatingThread() 4026{ 4027 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4028 mOutputTracks[i]->destroy(); 4029 } 4030} 4031 4032void AudioFlinger::DuplicatingThread::threadLoop_mix() 4033{ 4034 // mix buffers... 4035 if (outputsReady(outputTracks)) { 4036 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4037 } else { 4038 memset(mMixBuffer, 0, mixBufferSize); 4039 } 4040 sleepTime = 0; 4041 writeFrames = mNormalFrameCount; 4042 standbyTime = systemTime() + standbyDelay; 4043} 4044 4045void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4046{ 4047 if (sleepTime == 0) { 4048 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4049 sleepTime = activeSleepTime; 4050 } else { 4051 sleepTime = idleSleepTime; 4052 } 4053 } else if (mBytesWritten != 0) { 4054 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4055 writeFrames = mNormalFrameCount; 4056 memset(mMixBuffer, 0, mixBufferSize); 4057 } else { 4058 // flush remaining overflow buffers in output tracks 4059 writeFrames = 0; 4060 } 4061 sleepTime = 0; 4062 } 4063} 4064 4065void AudioFlinger::DuplicatingThread::threadLoop_write() 4066{ 4067 for (size_t i = 0; i < outputTracks.size(); i++) { 4068 outputTracks[i]->write(mMixBuffer, writeFrames); 4069 } 4070 mBytesWritten += mixBufferSize; 4071} 4072 4073void AudioFlinger::DuplicatingThread::threadLoop_standby() 4074{ 4075 // DuplicatingThread implements standby by stopping all tracks 4076 for (size_t i = 0; i < outputTracks.size(); i++) { 4077 outputTracks[i]->stop(); 4078 } 4079} 4080 4081void AudioFlinger::DuplicatingThread::saveOutputTracks() 4082{ 4083 outputTracks = mOutputTracks; 4084} 4085 4086void AudioFlinger::DuplicatingThread::clearOutputTracks() 4087{ 4088 outputTracks.clear(); 4089} 4090 4091void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4092{ 4093 Mutex::Autolock _l(mLock); 4094 // FIXME explain this formula 4095 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4096 OutputTrack *outputTrack = new OutputTrack(thread, 4097 this, 4098 mSampleRate, 4099 mFormat, 4100 mChannelMask, 4101 frameCount); 4102 if (outputTrack->cblk() != NULL) { 4103 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4104 mOutputTracks.add(outputTrack); 4105 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4106 updateWaitTime_l(); 4107 } 4108} 4109 4110void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4111{ 4112 Mutex::Autolock _l(mLock); 4113 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4114 if (mOutputTracks[i]->thread() == thread) { 4115 mOutputTracks[i]->destroy(); 4116 mOutputTracks.removeAt(i); 4117 updateWaitTime_l(); 4118 return; 4119 } 4120 } 4121 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4122} 4123 4124// caller must hold mLock 4125void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4126{ 4127 mWaitTimeMs = UINT_MAX; 4128 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4129 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4130 if (strong != 0) { 4131 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4132 if (waitTimeMs < mWaitTimeMs) { 4133 mWaitTimeMs = waitTimeMs; 4134 } 4135 } 4136 } 4137} 4138 4139 4140bool AudioFlinger::DuplicatingThread::outputsReady( 4141 const SortedVector< sp<OutputTrack> > &outputTracks) 4142{ 4143 for (size_t i = 0; i < outputTracks.size(); i++) { 4144 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4145 if (thread == 0) { 4146 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4147 outputTracks[i].get()); 4148 return false; 4149 } 4150 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4151 // see note at standby() declaration 4152 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4153 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4154 thread.get()); 4155 return false; 4156 } 4157 } 4158 return true; 4159} 4160 4161uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4162{ 4163 return (mWaitTimeMs * 1000) / 2; 4164} 4165 4166void AudioFlinger::DuplicatingThread::cacheParameters_l() 4167{ 4168 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4169 updateWaitTime_l(); 4170 4171 MixerThread::cacheParameters_l(); 4172} 4173 4174// ---------------------------------------------------------------------------- 4175 4176// TrackBase constructor must be called with AudioFlinger::mLock held 4177AudioFlinger::ThreadBase::TrackBase::TrackBase( 4178 ThreadBase *thread, 4179 const sp<Client>& client, 4180 uint32_t sampleRate, 4181 audio_format_t format, 4182 audio_channel_mask_t channelMask, 4183 size_t frameCount, 4184 const sp<IMemory>& sharedBuffer, 4185 int sessionId) 4186 : RefBase(), 4187 mThread(thread), 4188 mClient(client), 4189 mCblk(NULL), 4190 // mBuffer 4191 // mBufferEnd 4192 mStepCount(0), 4193 mState(IDLE), 4194 mSampleRate(sampleRate), 4195 mFormat(format), 4196 mChannelMask(channelMask), 4197 mChannelCount(popcount(channelMask)), 4198 mFrameSize(audio_is_linear_pcm(format) ? 4199 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 4200 mFrameCount(frameCount), 4201 mStepServerFailed(false), 4202 mSessionId(sessionId) 4203{ 4204 // client == 0 implies sharedBuffer == 0 4205 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4206 4207 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4208 sharedBuffer->size()); 4209 4210 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4211 size_t size = sizeof(audio_track_cblk_t); 4212 size_t bufferSize = frameCount * mFrameSize; 4213 if (sharedBuffer == 0) { 4214 size += bufferSize; 4215 } 4216 4217 if (client != 0) { 4218 mCblkMemory = client->heap()->allocate(size); 4219 if (mCblkMemory != 0) { 4220 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4221 // can't assume mCblk != NULL 4222 } else { 4223 ALOGE("not enough memory for AudioTrack size=%u", size); 4224 client->heap()->dump("AudioTrack"); 4225 return; 4226 } 4227 } else { 4228 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4229 // assume mCblk != NULL 4230 } 4231 4232 // construct the shared structure in-place. 4233 if (mCblk != NULL) { 4234 new(mCblk) audio_track_cblk_t(); 4235 // clear all buffers 4236 mCblk->frameCount_ = frameCount; 4237 mCblk->sampleRate = sampleRate; 4238// uncomment the following lines to quickly test 32-bit wraparound 4239// mCblk->user = 0xffff0000; 4240// mCblk->server = 0xffff0000; 4241// mCblk->userBase = 0xffff0000; 4242// mCblk->serverBase = 0xffff0000; 4243 if (sharedBuffer == 0) { 4244 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4245 memset(mBuffer, 0, bufferSize); 4246 // Force underrun condition to avoid false underrun callback until first data is 4247 // written to buffer (other flags are cleared) 4248 mCblk->flags = CBLK_UNDERRUN; 4249 } else { 4250 mBuffer = sharedBuffer->pointer(); 4251 } 4252 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4253 } 4254} 4255 4256AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4257{ 4258 if (mCblk != NULL) { 4259 if (mClient == 0) { 4260 delete mCblk; 4261 } else { 4262 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4263 } 4264 } 4265 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4266 if (mClient != 0) { 4267 // Client destructor must run with AudioFlinger mutex locked 4268 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4269 // If the client's reference count drops to zero, the associated destructor 4270 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4271 // relying on the automatic clear() at end of scope. 4272 mClient.clear(); 4273 } 4274} 4275 4276// AudioBufferProvider interface 4277// getNextBuffer() = 0; 4278// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4279void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4280{ 4281 buffer->raw = NULL; 4282 mStepCount = buffer->frameCount; 4283 // FIXME See note at getNextBuffer() 4284 (void) step(); // ignore return value of step() 4285 buffer->frameCount = 0; 4286} 4287 4288bool AudioFlinger::ThreadBase::TrackBase::step() { 4289 bool result; 4290 audio_track_cblk_t* cblk = this->cblk(); 4291 4292 result = cblk->stepServer(mStepCount, mFrameCount, isOut()); 4293 if (!result) { 4294 ALOGV("stepServer failed acquiring cblk mutex"); 4295 mStepServerFailed = true; 4296 } 4297 return result; 4298} 4299 4300void AudioFlinger::ThreadBase::TrackBase::reset() { 4301 audio_track_cblk_t* cblk = this->cblk(); 4302 4303 cblk->user = 0; 4304 cblk->server = 0; 4305 cblk->userBase = 0; 4306 cblk->serverBase = 0; 4307 mStepServerFailed = false; 4308 ALOGV("TrackBase::reset"); 4309} 4310 4311uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4312 return mCblk->sampleRate; 4313} 4314 4315void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4316 audio_track_cblk_t* cblk = this->cblk(); 4317 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; 4318 int8_t *bufferEnd = bufferStart + frames * mFrameSize; 4319 4320 // Check validity of returned pointer in case the track control block would have been corrupted. 4321 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4322 "TrackBase::getBuffer buffer out of range:\n" 4323 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4324 " server %u, serverBase %u, user %u, userBase %u, frameSize %u", 4325 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4326 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); 4327 4328 return bufferStart; 4329} 4330 4331status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4332{ 4333 mSyncEvents.add(event); 4334 return NO_ERROR; 4335} 4336 4337// ---------------------------------------------------------------------------- 4338 4339// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4340AudioFlinger::PlaybackThread::Track::Track( 4341 PlaybackThread *thread, 4342 const sp<Client>& client, 4343 audio_stream_type_t streamType, 4344 uint32_t sampleRate, 4345 audio_format_t format, 4346 audio_channel_mask_t channelMask, 4347 size_t frameCount, 4348 const sp<IMemory>& sharedBuffer, 4349 int sessionId, 4350 IAudioFlinger::track_flags_t flags) 4351 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4352 sessionId), 4353 mMute(false), 4354 mFillingUpStatus(FS_INVALID), 4355 // mRetryCount initialized later when needed 4356 mSharedBuffer(sharedBuffer), 4357 mStreamType(streamType), 4358 mName(-1), // see note below 4359 mMainBuffer(thread->mixBuffer()), 4360 mAuxBuffer(NULL), 4361 mAuxEffectId(0), mHasVolumeController(false), 4362 mPresentationCompleteFrames(0), 4363 mFlags(flags), 4364 mFastIndex(-1), 4365 mUnderrunCount(0), 4366 mCachedVolume(1.0) 4367{ 4368 if (mCblk != NULL) { 4369 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4370 mName = thread->getTrackName_l(channelMask, sessionId); 4371 mCblk->mName = mName; 4372 if (mName < 0) { 4373 ALOGE("no more track names available"); 4374 return; 4375 } 4376 // only allocate a fast track index if we were able to allocate a normal track name 4377 if (flags & IAudioFlinger::TRACK_FAST) { 4378 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4379 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4380 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4381 // FIXME This is too eager. We allocate a fast track index before the 4382 // fast track becomes active. Since fast tracks are a scarce resource, 4383 // this means we are potentially denying other more important fast tracks from 4384 // being created. It would be better to allocate the index dynamically. 4385 mFastIndex = i; 4386 mCblk->mName = i; 4387 // Read the initial underruns because this field is never cleared by the fast mixer 4388 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4389 thread->mFastTrackAvailMask &= ~(1 << i); 4390 } 4391 } 4392 ALOGV("Track constructor name %d, calling pid %d", mName, 4393 IPCThreadState::self()->getCallingPid()); 4394} 4395 4396AudioFlinger::PlaybackThread::Track::~Track() 4397{ 4398 ALOGV("PlaybackThread::Track destructor"); 4399} 4400 4401void AudioFlinger::PlaybackThread::Track::destroy() 4402{ 4403 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4404 // by removing it from mTracks vector, so there is a risk that this Tracks's 4405 // destructor is called. As the destructor needs to lock mLock, 4406 // we must acquire a strong reference on this Track before locking mLock 4407 // here so that the destructor is called only when exiting this function. 4408 // On the other hand, as long as Track::destroy() is only called by 4409 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4410 // this Track with its member mTrack. 4411 sp<Track> keep(this); 4412 { // scope for mLock 4413 sp<ThreadBase> thread = mThread.promote(); 4414 if (thread != 0) { 4415 if (!isOutputTrack()) { 4416 if (mState == ACTIVE || mState == RESUMING) { 4417 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4418 4419#ifdef ADD_BATTERY_DATA 4420 // to track the speaker usage 4421 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4422#endif 4423 } 4424 AudioSystem::releaseOutput(thread->id()); 4425 } 4426 Mutex::Autolock _l(thread->mLock); 4427 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4428 playbackThread->destroyTrack_l(this); 4429 } 4430 } 4431} 4432 4433/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4434{ 4435 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 4436 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4437} 4438 4439void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4440{ 4441 uint32_t vlr = mCblk->getVolumeLR(); 4442 if (isFastTrack()) { 4443 sprintf(buffer, " F %2d", mFastIndex); 4444 } else { 4445 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4446 } 4447 track_state state = mState; 4448 char stateChar; 4449 switch (state) { 4450 case IDLE: 4451 stateChar = 'I'; 4452 break; 4453 case TERMINATED: 4454 stateChar = 'T'; 4455 break; 4456 case STOPPING_1: 4457 stateChar = 's'; 4458 break; 4459 case STOPPING_2: 4460 stateChar = '5'; 4461 break; 4462 case STOPPED: 4463 stateChar = 'S'; 4464 break; 4465 case RESUMING: 4466 stateChar = 'R'; 4467 break; 4468 case ACTIVE: 4469 stateChar = 'A'; 4470 break; 4471 case PAUSING: 4472 stateChar = 'p'; 4473 break; 4474 case PAUSED: 4475 stateChar = 'P'; 4476 break; 4477 case FLUSHED: 4478 stateChar = 'F'; 4479 break; 4480 default: 4481 stateChar = '?'; 4482 break; 4483 } 4484 char nowInUnderrun; 4485 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4486 case UNDERRUN_FULL: 4487 nowInUnderrun = ' '; 4488 break; 4489 case UNDERRUN_PARTIAL: 4490 nowInUnderrun = '<'; 4491 break; 4492 case UNDERRUN_EMPTY: 4493 nowInUnderrun = '*'; 4494 break; 4495 default: 4496 nowInUnderrun = '?'; 4497 break; 4498 } 4499 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4500 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4501 (mClient == 0) ? getpid_cached : mClient->pid(), 4502 mStreamType, 4503 mFormat, 4504 mChannelMask, 4505 mSessionId, 4506 mStepCount, 4507 mFrameCount, 4508 stateChar, 4509 mMute, 4510 mFillingUpStatus, 4511 mCblk->sampleRate, 4512 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4513 20.0 * log10((vlr >> 16) / 4096.0), 4514 mCblk->server, 4515 mCblk->user, 4516 (int)mMainBuffer, 4517 (int)mAuxBuffer, 4518 mCblk->flags, 4519 mUnderrunCount, 4520 nowInUnderrun); 4521} 4522 4523// AudioBufferProvider interface 4524status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4525 AudioBufferProvider::Buffer* buffer, int64_t pts) 4526{ 4527 audio_track_cblk_t* cblk = this->cblk(); 4528 uint32_t framesReady; 4529 uint32_t framesReq = buffer->frameCount; 4530 4531 // Check if last stepServer failed, try to step now 4532 if (mStepServerFailed) { 4533 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4534 // Since the fast mixer is higher priority than client callback thread, 4535 // it does not result in priority inversion for client. 4536 // But a non-blocking solution would be preferable to avoid 4537 // fast mixer being unable to tryLock(), and 4538 // to avoid the extra context switches if the client wakes up, 4539 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4540 if (!step()) goto getNextBuffer_exit; 4541 ALOGV("stepServer recovered"); 4542 mStepServerFailed = false; 4543 } 4544 4545 // FIXME Same as above 4546 framesReady = cblk->framesReadyOut(); 4547 4548 if (CC_LIKELY(framesReady)) { 4549 uint32_t s = cblk->server; 4550 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 4551 4552 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4553 if (framesReq > framesReady) { 4554 framesReq = framesReady; 4555 } 4556 if (framesReq > bufferEnd - s) { 4557 framesReq = bufferEnd - s; 4558 } 4559 4560 buffer->raw = getBuffer(s, framesReq); 4561 buffer->frameCount = framesReq; 4562 return NO_ERROR; 4563 } 4564 4565getNextBuffer_exit: 4566 buffer->raw = NULL; 4567 buffer->frameCount = 0; 4568 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4569 return NOT_ENOUGH_DATA; 4570} 4571 4572// Note that framesReady() takes a mutex on the control block using tryLock(). 4573// This could result in priority inversion if framesReady() is called by the normal mixer, 4574// as the normal mixer thread runs at lower 4575// priority than the client's callback thread: there is a short window within framesReady() 4576// during which the normal mixer could be preempted, and the client callback would block. 4577// Another problem can occur if framesReady() is called by the fast mixer: 4578// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4579// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4580size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4581 return mCblk->framesReadyOut(); 4582} 4583 4584// Don't call for fast tracks; the framesReady() could result in priority inversion 4585bool AudioFlinger::PlaybackThread::Track::isReady() const { 4586 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4587 4588 if (framesReady() >= mFrameCount || 4589 (mCblk->flags & CBLK_FORCEREADY)) { 4590 mFillingUpStatus = FS_FILLED; 4591 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4592 return true; 4593 } 4594 return false; 4595} 4596 4597status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4598 int triggerSession) 4599{ 4600 status_t status = NO_ERROR; 4601 ALOGV("start(%d), calling pid %d session %d", 4602 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4603 4604 sp<ThreadBase> thread = mThread.promote(); 4605 if (thread != 0) { 4606 Mutex::Autolock _l(thread->mLock); 4607 track_state state = mState; 4608 // here the track could be either new, or restarted 4609 // in both cases "unstop" the track 4610 if (mState == PAUSED) { 4611 mState = TrackBase::RESUMING; 4612 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4613 } else { 4614 mState = TrackBase::ACTIVE; 4615 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4616 } 4617 4618 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4619 thread->mLock.unlock(); 4620 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4621 thread->mLock.lock(); 4622 4623#ifdef ADD_BATTERY_DATA 4624 // to track the speaker usage 4625 if (status == NO_ERROR) { 4626 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4627 } 4628#endif 4629 } 4630 if (status == NO_ERROR) { 4631 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4632 playbackThread->addTrack_l(this); 4633 } else { 4634 mState = state; 4635 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4636 } 4637 } else { 4638 status = BAD_VALUE; 4639 } 4640 return status; 4641} 4642 4643void AudioFlinger::PlaybackThread::Track::stop() 4644{ 4645 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4646 sp<ThreadBase> thread = mThread.promote(); 4647 if (thread != 0) { 4648 Mutex::Autolock _l(thread->mLock); 4649 track_state state = mState; 4650 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4651 // If the track is not active (PAUSED and buffers full), flush buffers 4652 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4653 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4654 reset(); 4655 mState = STOPPED; 4656 } else if (!isFastTrack()) { 4657 mState = STOPPED; 4658 } else { 4659 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4660 // and then to STOPPED and reset() when presentation is complete 4661 mState = STOPPING_1; 4662 } 4663 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4664 playbackThread); 4665 } 4666 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4667 thread->mLock.unlock(); 4668 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4669 thread->mLock.lock(); 4670 4671#ifdef ADD_BATTERY_DATA 4672 // to track the speaker usage 4673 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4674#endif 4675 } 4676 } 4677} 4678 4679void AudioFlinger::PlaybackThread::Track::pause() 4680{ 4681 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4682 sp<ThreadBase> thread = mThread.promote(); 4683 if (thread != 0) { 4684 Mutex::Autolock _l(thread->mLock); 4685 if (mState == ACTIVE || mState == RESUMING) { 4686 mState = PAUSING; 4687 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4688 if (!isOutputTrack()) { 4689 thread->mLock.unlock(); 4690 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4691 thread->mLock.lock(); 4692 4693#ifdef ADD_BATTERY_DATA 4694 // to track the speaker usage 4695 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4696#endif 4697 } 4698 } 4699 } 4700} 4701 4702void AudioFlinger::PlaybackThread::Track::flush() 4703{ 4704 ALOGV("flush(%d)", mName); 4705 sp<ThreadBase> thread = mThread.promote(); 4706 if (thread != 0) { 4707 Mutex::Autolock _l(thread->mLock); 4708 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4709 mState != PAUSING && mState != IDLE && mState != FLUSHED) { 4710 return; 4711 } 4712 // No point remaining in PAUSED state after a flush => go to 4713 // FLUSHED state 4714 mState = FLUSHED; 4715 // do not reset the track if it is still in the process of being stopped or paused. 4716 // this will be done by prepareTracks_l() when the track is stopped. 4717 // prepareTracks_l() will see mState == FLUSHED, then 4718 // remove from active track list, reset(), and trigger presentation complete 4719 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4720 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4721 reset(); 4722 } 4723 } 4724} 4725 4726void AudioFlinger::PlaybackThread::Track::reset() 4727{ 4728 // Do not reset twice to avoid discarding data written just after a flush and before 4729 // the audioflinger thread detects the track is stopped. 4730 if (!mResetDone) { 4731 TrackBase::reset(); 4732 // Force underrun condition to avoid false underrun callback until first data is 4733 // written to buffer 4734 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4735 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4736 mFillingUpStatus = FS_FILLING; 4737 mResetDone = true; 4738 if (mState == FLUSHED) { 4739 mState = IDLE; 4740 } 4741 } 4742} 4743 4744void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4745{ 4746 mMute = muted; 4747} 4748 4749status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4750{ 4751 status_t status = DEAD_OBJECT; 4752 sp<ThreadBase> thread = mThread.promote(); 4753 if (thread != 0) { 4754 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4755 sp<AudioFlinger> af = mClient->audioFlinger(); 4756 4757 Mutex::Autolock _l(af->mLock); 4758 4759 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4760 4761 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4762 Mutex::Autolock _dl(playbackThread->mLock); 4763 Mutex::Autolock _sl(srcThread->mLock); 4764 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4765 if (chain == 0) { 4766 return INVALID_OPERATION; 4767 } 4768 4769 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4770 if (effect == 0) { 4771 return INVALID_OPERATION; 4772 } 4773 srcThread->removeEffect_l(effect); 4774 playbackThread->addEffect_l(effect); 4775 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4776 if (effect->state() == EffectModule::ACTIVE || 4777 effect->state() == EffectModule::STOPPING) { 4778 effect->start(); 4779 } 4780 4781 sp<EffectChain> dstChain = effect->chain().promote(); 4782 if (dstChain == 0) { 4783 srcThread->addEffect_l(effect); 4784 return INVALID_OPERATION; 4785 } 4786 AudioSystem::unregisterEffect(effect->id()); 4787 AudioSystem::registerEffect(&effect->desc(), 4788 srcThread->id(), 4789 dstChain->strategy(), 4790 AUDIO_SESSION_OUTPUT_MIX, 4791 effect->id()); 4792 } 4793 status = playbackThread->attachAuxEffect(this, EffectId); 4794 } 4795 return status; 4796} 4797 4798void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4799{ 4800 mAuxEffectId = EffectId; 4801 mAuxBuffer = buffer; 4802} 4803 4804bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4805 size_t audioHalFrames) 4806{ 4807 // a track is considered presented when the total number of frames written to audio HAL 4808 // corresponds to the number of frames written when presentationComplete() is called for the 4809 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4810 if (mPresentationCompleteFrames == 0) { 4811 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4812 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4813 mPresentationCompleteFrames, audioHalFrames); 4814 } 4815 if (framesWritten >= mPresentationCompleteFrames) { 4816 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4817 mSessionId, framesWritten); 4818 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4819 return true; 4820 } 4821 return false; 4822} 4823 4824void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4825{ 4826 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4827 if (mSyncEvents[i]->type() == type) { 4828 mSyncEvents[i]->trigger(); 4829 mSyncEvents.removeAt(i); 4830 i--; 4831 } 4832 } 4833} 4834 4835// implement VolumeBufferProvider interface 4836 4837uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4838{ 4839 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4840 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4841 uint32_t vlr = mCblk->getVolumeLR(); 4842 uint32_t vl = vlr & 0xFFFF; 4843 uint32_t vr = vlr >> 16; 4844 // track volumes come from shared memory, so can't be trusted and must be clamped 4845 if (vl > MAX_GAIN_INT) { 4846 vl = MAX_GAIN_INT; 4847 } 4848 if (vr > MAX_GAIN_INT) { 4849 vr = MAX_GAIN_INT; 4850 } 4851 // now apply the cached master volume and stream type volume; 4852 // this is trusted but lacks any synchronization or barrier so may be stale 4853 float v = mCachedVolume; 4854 vl *= v; 4855 vr *= v; 4856 // re-combine into U4.16 4857 vlr = (vr << 16) | (vl & 0xFFFF); 4858 // FIXME look at mute, pause, and stop flags 4859 return vlr; 4860} 4861 4862status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4863{ 4864 if (mState == TERMINATED || mState == PAUSED || 4865 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4866 (mState == STOPPED)))) { 4867 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4868 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4869 event->cancel(); 4870 return INVALID_OPERATION; 4871 } 4872 (void) TrackBase::setSyncEvent(event); 4873 return NO_ERROR; 4874} 4875 4876bool AudioFlinger::PlaybackThread::Track::isOut() const 4877{ 4878 return true; 4879} 4880 4881// timed audio tracks 4882 4883sp<AudioFlinger::PlaybackThread::TimedTrack> 4884AudioFlinger::PlaybackThread::TimedTrack::create( 4885 PlaybackThread *thread, 4886 const sp<Client>& client, 4887 audio_stream_type_t streamType, 4888 uint32_t sampleRate, 4889 audio_format_t format, 4890 audio_channel_mask_t channelMask, 4891 size_t frameCount, 4892 const sp<IMemory>& sharedBuffer, 4893 int sessionId) { 4894 if (!client->reserveTimedTrack()) 4895 return 0; 4896 4897 return new TimedTrack( 4898 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4899 sharedBuffer, sessionId); 4900} 4901 4902AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4903 PlaybackThread *thread, 4904 const sp<Client>& client, 4905 audio_stream_type_t streamType, 4906 uint32_t sampleRate, 4907 audio_format_t format, 4908 audio_channel_mask_t channelMask, 4909 size_t frameCount, 4910 const sp<IMemory>& sharedBuffer, 4911 int sessionId) 4912 : Track(thread, client, streamType, sampleRate, format, channelMask, 4913 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4914 mQueueHeadInFlight(false), 4915 mTrimQueueHeadOnRelease(false), 4916 mFramesPendingInQueue(0), 4917 mTimedSilenceBuffer(NULL), 4918 mTimedSilenceBufferSize(0), 4919 mTimedAudioOutputOnTime(false), 4920 mMediaTimeTransformValid(false) 4921{ 4922 LocalClock lc; 4923 mLocalTimeFreq = lc.getLocalFreq(); 4924 4925 mLocalTimeToSampleTransform.a_zero = 0; 4926 mLocalTimeToSampleTransform.b_zero = 0; 4927 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4928 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4929 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4930 &mLocalTimeToSampleTransform.a_to_b_denom); 4931 4932 mMediaTimeToSampleTransform.a_zero = 0; 4933 mMediaTimeToSampleTransform.b_zero = 0; 4934 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4935 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4936 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4937 &mMediaTimeToSampleTransform.a_to_b_denom); 4938} 4939 4940AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4941 mClient->releaseTimedTrack(); 4942 delete [] mTimedSilenceBuffer; 4943} 4944 4945status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4946 size_t size, sp<IMemory>* buffer) { 4947 4948 Mutex::Autolock _l(mTimedBufferQueueLock); 4949 4950 trimTimedBufferQueue_l(); 4951 4952 // lazily initialize the shared memory heap for timed buffers 4953 if (mTimedMemoryDealer == NULL) { 4954 const int kTimedBufferHeapSize = 512 << 10; 4955 4956 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4957 "AudioFlingerTimed"); 4958 if (mTimedMemoryDealer == NULL) 4959 return NO_MEMORY; 4960 } 4961 4962 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4963 if (newBuffer == NULL) { 4964 newBuffer = mTimedMemoryDealer->allocate(size); 4965 if (newBuffer == NULL) 4966 return NO_MEMORY; 4967 } 4968 4969 *buffer = newBuffer; 4970 return NO_ERROR; 4971} 4972 4973// caller must hold mTimedBufferQueueLock 4974void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4975 int64_t mediaTimeNow; 4976 { 4977 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4978 if (!mMediaTimeTransformValid) 4979 return; 4980 4981 int64_t targetTimeNow; 4982 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4983 ? mCCHelper.getCommonTime(&targetTimeNow) 4984 : mCCHelper.getLocalTime(&targetTimeNow); 4985 4986 if (OK != res) 4987 return; 4988 4989 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4990 &mediaTimeNow)) { 4991 return; 4992 } 4993 } 4994 4995 size_t trimEnd; 4996 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4997 int64_t bufEnd; 4998 4999 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 5000 // We have a next buffer. Just use its PTS as the PTS of the frame 5001 // following the last frame in this buffer. If the stream is sparse 5002 // (ie, there are deliberate gaps left in the stream which should be 5003 // filled with silence by the TimedAudioTrack), then this can result 5004 // in one extra buffer being left un-trimmed when it could have 5005 // been. In general, this is not typical, and we would rather 5006 // optimized away the TS calculation below for the more common case 5007 // where PTSes are contiguous. 5008 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5009 } else { 5010 // We have no next buffer. Compute the PTS of the frame following 5011 // the last frame in this buffer by computing the duration of of 5012 // this frame in media time units and adding it to the PTS of the 5013 // buffer. 5014 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5015 / mFrameSize; 5016 5017 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5018 &bufEnd)) { 5019 ALOGE("Failed to convert frame count of %lld to media time" 5020 " duration" " (scale factor %d/%u) in %s", 5021 frameCount, 5022 mMediaTimeToSampleTransform.a_to_b_numer, 5023 mMediaTimeToSampleTransform.a_to_b_denom, 5024 __PRETTY_FUNCTION__); 5025 break; 5026 } 5027 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5028 } 5029 5030 if (bufEnd > mediaTimeNow) 5031 break; 5032 5033 // Is the buffer we want to use in the middle of a mix operation right 5034 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5035 // from the mixer which should be coming back shortly. 5036 if (!trimEnd && mQueueHeadInFlight) { 5037 mTrimQueueHeadOnRelease = true; 5038 } 5039 } 5040 5041 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5042 if (trimStart < trimEnd) { 5043 // Update the bookkeeping for framesReady() 5044 for (size_t i = trimStart; i < trimEnd; ++i) { 5045 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5046 } 5047 5048 // Now actually remove the buffers from the queue. 5049 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5050 } 5051} 5052 5053void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5054 const char* logTag) { 5055 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5056 "%s called (reason \"%s\"), but timed buffer queue has no" 5057 " elements to trim.", __FUNCTION__, logTag); 5058 5059 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5060 mTimedBufferQueue.removeAt(0); 5061} 5062 5063void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5064 const TimedBuffer& buf, 5065 const char* logTag) { 5066 uint32_t bufBytes = buf.buffer()->size(); 5067 uint32_t consumedAlready = buf.position(); 5068 5069 ALOG_ASSERT(consumedAlready <= bufBytes, 5070 "Bad bookkeeping while updating frames pending. Timed buffer is" 5071 " only %u bytes long, but claims to have consumed %u" 5072 " bytes. (update reason: \"%s\")", 5073 bufBytes, consumedAlready, logTag); 5074 5075 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 5076 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5077 "Bad bookkeeping while updating frames pending. Should have at" 5078 " least %u queued frames, but we think we have only %u. (update" 5079 " reason: \"%s\")", 5080 bufFrames, mFramesPendingInQueue, logTag); 5081 5082 mFramesPendingInQueue -= bufFrames; 5083} 5084 5085status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5086 const sp<IMemory>& buffer, int64_t pts) { 5087 5088 { 5089 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5090 if (!mMediaTimeTransformValid) 5091 return INVALID_OPERATION; 5092 } 5093 5094 Mutex::Autolock _l(mTimedBufferQueueLock); 5095 5096 uint32_t bufFrames = buffer->size() / mFrameSize; 5097 mFramesPendingInQueue += bufFrames; 5098 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5099 5100 return NO_ERROR; 5101} 5102 5103status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5104 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5105 5106 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5107 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5108 target); 5109 5110 if (!(target == TimedAudioTrack::LOCAL_TIME || 5111 target == TimedAudioTrack::COMMON_TIME)) { 5112 return BAD_VALUE; 5113 } 5114 5115 Mutex::Autolock lock(mMediaTimeTransformLock); 5116 mMediaTimeTransform = xform; 5117 mMediaTimeTransformTarget = target; 5118 mMediaTimeTransformValid = true; 5119 5120 return NO_ERROR; 5121} 5122 5123#define min(a, b) ((a) < (b) ? (a) : (b)) 5124 5125// implementation of getNextBuffer for tracks whose buffers have timestamps 5126status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5127 AudioBufferProvider::Buffer* buffer, int64_t pts) 5128{ 5129 if (pts == AudioBufferProvider::kInvalidPTS) { 5130 buffer->raw = NULL; 5131 buffer->frameCount = 0; 5132 mTimedAudioOutputOnTime = false; 5133 return INVALID_OPERATION; 5134 } 5135 5136 Mutex::Autolock _l(mTimedBufferQueueLock); 5137 5138 ALOG_ASSERT(!mQueueHeadInFlight, 5139 "getNextBuffer called without releaseBuffer!"); 5140 5141 while (true) { 5142 5143 // if we have no timed buffers, then fail 5144 if (mTimedBufferQueue.isEmpty()) { 5145 buffer->raw = NULL; 5146 buffer->frameCount = 0; 5147 return NOT_ENOUGH_DATA; 5148 } 5149 5150 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5151 5152 // calculate the PTS of the head of the timed buffer queue expressed in 5153 // local time 5154 int64_t headLocalPTS; 5155 { 5156 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5157 5158 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5159 5160 if (mMediaTimeTransform.a_to_b_denom == 0) { 5161 // the transform represents a pause, so yield silence 5162 timedYieldSilence_l(buffer->frameCount, buffer); 5163 return NO_ERROR; 5164 } 5165 5166 int64_t transformedPTS; 5167 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5168 &transformedPTS)) { 5169 // the transform failed. this shouldn't happen, but if it does 5170 // then just drop this buffer 5171 ALOGW("timedGetNextBuffer transform failed"); 5172 buffer->raw = NULL; 5173 buffer->frameCount = 0; 5174 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5175 return NO_ERROR; 5176 } 5177 5178 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5179 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5180 &headLocalPTS)) { 5181 buffer->raw = NULL; 5182 buffer->frameCount = 0; 5183 return INVALID_OPERATION; 5184 } 5185 } else { 5186 headLocalPTS = transformedPTS; 5187 } 5188 } 5189 5190 // adjust the head buffer's PTS to reflect the portion of the head buffer 5191 // that has already been consumed 5192 int64_t effectivePTS = headLocalPTS + 5193 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 5194 5195 // Calculate the delta in samples between the head of the input buffer 5196 // queue and the start of the next output buffer that will be written. 5197 // If the transformation fails because of over or underflow, it means 5198 // that the sample's position in the output stream is so far out of 5199 // whack that it should just be dropped. 5200 int64_t sampleDelta; 5201 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5202 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5203 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5204 " mix"); 5205 continue; 5206 } 5207 if (!mLocalTimeToSampleTransform.doForwardTransform( 5208 (effectivePTS - pts) << 32, &sampleDelta)) { 5209 ALOGV("*** too late during sample rate transform: dropped buffer"); 5210 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5211 continue; 5212 } 5213 5214 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5215 " sampleDelta=[%d.%08x]", 5216 head.pts(), head.position(), pts, 5217 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5218 + (sampleDelta >> 32)), 5219 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5220 5221 // if the delta between the ideal placement for the next input sample and 5222 // the current output position is within this threshold, then we will 5223 // concatenate the next input samples to the previous output 5224 const int64_t kSampleContinuityThreshold = 5225 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5226 5227 // if this is the first buffer of audio that we're emitting from this track 5228 // then it should be almost exactly on time. 5229 const int64_t kSampleStartupThreshold = 1LL << 32; 5230 5231 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5232 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5233 // the next input is close enough to being on time, so concatenate it 5234 // with the last output 5235 timedYieldSamples_l(buffer); 5236 5237 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5238 head.position(), buffer->frameCount); 5239 return NO_ERROR; 5240 } 5241 5242 // Looks like our output is not on time. Reset our on timed status. 5243 // Next time we mix samples from our input queue, then should be within 5244 // the StartupThreshold. 5245 mTimedAudioOutputOnTime = false; 5246 if (sampleDelta > 0) { 5247 // the gap between the current output position and the proper start of 5248 // the next input sample is too big, so fill it with silence 5249 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5250 5251 timedYieldSilence_l(framesUntilNextInput, buffer); 5252 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5253 return NO_ERROR; 5254 } else { 5255 // the next input sample is late 5256 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5257 size_t onTimeSamplePosition = 5258 head.position() + lateFrames * mFrameSize; 5259 5260 if (onTimeSamplePosition > head.buffer()->size()) { 5261 // all the remaining samples in the head are too late, so 5262 // drop it and move on 5263 ALOGV("*** too late: dropped buffer"); 5264 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5265 continue; 5266 } else { 5267 // skip over the late samples 5268 head.setPosition(onTimeSamplePosition); 5269 5270 // yield the available samples 5271 timedYieldSamples_l(buffer); 5272 5273 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5274 return NO_ERROR; 5275 } 5276 } 5277 } 5278} 5279 5280// Yield samples from the timed buffer queue head up to the given output 5281// buffer's capacity. 5282// 5283// Caller must hold mTimedBufferQueueLock 5284void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5285 AudioBufferProvider::Buffer* buffer) { 5286 5287 const TimedBuffer& head = mTimedBufferQueue[0]; 5288 5289 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5290 head.position()); 5291 5292 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5293 mFrameSize); 5294 size_t framesRequested = buffer->frameCount; 5295 buffer->frameCount = min(framesLeftInHead, framesRequested); 5296 5297 mQueueHeadInFlight = true; 5298 mTimedAudioOutputOnTime = true; 5299} 5300 5301// Yield samples of silence up to the given output buffer's capacity 5302// 5303// Caller must hold mTimedBufferQueueLock 5304void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5305 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5306 5307 // lazily allocate a buffer filled with silence 5308 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 5309 delete [] mTimedSilenceBuffer; 5310 mTimedSilenceBufferSize = numFrames * mFrameSize; 5311 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5312 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5313 } 5314 5315 buffer->raw = mTimedSilenceBuffer; 5316 size_t framesRequested = buffer->frameCount; 5317 buffer->frameCount = min(numFrames, framesRequested); 5318 5319 mTimedAudioOutputOnTime = false; 5320} 5321 5322// AudioBufferProvider interface 5323void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5324 AudioBufferProvider::Buffer* buffer) { 5325 5326 Mutex::Autolock _l(mTimedBufferQueueLock); 5327 5328 // If the buffer which was just released is part of the buffer at the head 5329 // of the queue, be sure to update the amt of the buffer which has been 5330 // consumed. If the buffer being returned is not part of the head of the 5331 // queue, its either because the buffer is part of the silence buffer, or 5332 // because the head of the timed queue was trimmed after the mixer called 5333 // getNextBuffer but before the mixer called releaseBuffer. 5334 if (buffer->raw == mTimedSilenceBuffer) { 5335 ALOG_ASSERT(!mQueueHeadInFlight, 5336 "Queue head in flight during release of silence buffer!"); 5337 goto done; 5338 } 5339 5340 ALOG_ASSERT(mQueueHeadInFlight, 5341 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5342 " head in flight."); 5343 5344 if (mTimedBufferQueue.size()) { 5345 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5346 5347 void* start = head.buffer()->pointer(); 5348 void* end = reinterpret_cast<void*>( 5349 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5350 + head.buffer()->size()); 5351 5352 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5353 "released buffer not within the head of the timed buffer" 5354 " queue; qHead = [%p, %p], released buffer = %p", 5355 start, end, buffer->raw); 5356 5357 head.setPosition(head.position() + 5358 (buffer->frameCount * mFrameSize)); 5359 mQueueHeadInFlight = false; 5360 5361 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5362 "Bad bookkeeping during releaseBuffer! Should have at" 5363 " least %u queued frames, but we think we have only %u", 5364 buffer->frameCount, mFramesPendingInQueue); 5365 5366 mFramesPendingInQueue -= buffer->frameCount; 5367 5368 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5369 || mTrimQueueHeadOnRelease) { 5370 trimTimedBufferQueueHead_l("releaseBuffer"); 5371 mTrimQueueHeadOnRelease = false; 5372 } 5373 } else { 5374 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5375 " buffers in the timed buffer queue"); 5376 } 5377 5378done: 5379 buffer->raw = 0; 5380 buffer->frameCount = 0; 5381} 5382 5383size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5384 Mutex::Autolock _l(mTimedBufferQueueLock); 5385 return mFramesPendingInQueue; 5386} 5387 5388AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5389 : mPTS(0), mPosition(0) {} 5390 5391AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5392 const sp<IMemory>& buffer, int64_t pts) 5393 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5394 5395// ---------------------------------------------------------------------------- 5396 5397// RecordTrack constructor must be called with AudioFlinger::mLock held 5398AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5399 RecordThread *thread, 5400 const sp<Client>& client, 5401 uint32_t sampleRate, 5402 audio_format_t format, 5403 audio_channel_mask_t channelMask, 5404 size_t frameCount, 5405 int sessionId) 5406 : TrackBase(thread, client, sampleRate, format, 5407 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5408 mOverflow(false) 5409{ 5410 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5411} 5412 5413AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5414{ 5415 ALOGV("%s", __func__); 5416} 5417 5418// AudioBufferProvider interface 5419status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5420 int64_t pts) 5421{ 5422 audio_track_cblk_t* cblk = this->cblk(); 5423 uint32_t framesAvail; 5424 uint32_t framesReq = buffer->frameCount; 5425 5426 // Check if last stepServer failed, try to step now 5427 if (mStepServerFailed) { 5428 if (!step()) goto getNextBuffer_exit; 5429 ALOGV("stepServer recovered"); 5430 mStepServerFailed = false; 5431 } 5432 5433 // FIXME lock is not actually held, so overrun is possible 5434 framesAvail = cblk->framesAvailableIn_l(mFrameCount); 5435 5436 if (CC_LIKELY(framesAvail)) { 5437 uint32_t s = cblk->server; 5438 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 5439 5440 if (framesReq > framesAvail) { 5441 framesReq = framesAvail; 5442 } 5443 if (framesReq > bufferEnd - s) { 5444 framesReq = bufferEnd - s; 5445 } 5446 5447 buffer->raw = getBuffer(s, framesReq); 5448 buffer->frameCount = framesReq; 5449 return NO_ERROR; 5450 } 5451 5452getNextBuffer_exit: 5453 buffer->raw = NULL; 5454 buffer->frameCount = 0; 5455 return NOT_ENOUGH_DATA; 5456} 5457 5458status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5459 int triggerSession) 5460{ 5461 sp<ThreadBase> thread = mThread.promote(); 5462 if (thread != 0) { 5463 RecordThread *recordThread = (RecordThread *)thread.get(); 5464 return recordThread->start(this, event, triggerSession); 5465 } else { 5466 return BAD_VALUE; 5467 } 5468} 5469 5470void AudioFlinger::RecordThread::RecordTrack::stop() 5471{ 5472 sp<ThreadBase> thread = mThread.promote(); 5473 if (thread != 0) { 5474 RecordThread *recordThread = (RecordThread *)thread.get(); 5475 recordThread->mLock.lock(); 5476 bool doStop = recordThread->stop_l(this); 5477 if (doStop) { 5478 TrackBase::reset(); 5479 // Force overrun condition to avoid false overrun callback until first data is 5480 // read from buffer 5481 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5482 } 5483 recordThread->mLock.unlock(); 5484 if (doStop) { 5485 AudioSystem::stopInput(recordThread->id()); 5486 } 5487 } 5488} 5489 5490/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5491{ 5492 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 5493} 5494 5495void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5496{ 5497 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5498 (mClient == 0) ? getpid_cached : mClient->pid(), 5499 mFormat, 5500 mChannelMask, 5501 mSessionId, 5502 mStepCount, 5503 mState, 5504 mCblk->sampleRate, 5505 mCblk->server, 5506 mCblk->user, 5507 mFrameCount); 5508} 5509 5510bool AudioFlinger::RecordThread::RecordTrack::isOut() const 5511{ 5512 return false; 5513} 5514 5515// ---------------------------------------------------------------------------- 5516 5517AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5518 PlaybackThread *playbackThread, 5519 DuplicatingThread *sourceThread, 5520 uint32_t sampleRate, 5521 audio_format_t format, 5522 audio_channel_mask_t channelMask, 5523 size_t frameCount) 5524 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5525 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5526 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 5527{ 5528 5529 if (mCblk != NULL) { 5530 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5531 mOutBuffer.frameCount = 0; 5532 playbackThread->mTracks.add(this); 5533 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \ 5534 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 5535 mCblk, mBuffer, mBuffers, 5536 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5537 } else { 5538 ALOGW("Error creating output track on thread %p", playbackThread); 5539 } 5540} 5541 5542AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5543{ 5544 clearBufferQueue(); 5545} 5546 5547status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5548 int triggerSession) 5549{ 5550 status_t status = Track::start(event, triggerSession); 5551 if (status != NO_ERROR) { 5552 return status; 5553 } 5554 5555 mActive = true; 5556 mRetryCount = 127; 5557 return status; 5558} 5559 5560void AudioFlinger::PlaybackThread::OutputTrack::stop() 5561{ 5562 Track::stop(); 5563 clearBufferQueue(); 5564 mOutBuffer.frameCount = 0; 5565 mActive = false; 5566} 5567 5568bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5569{ 5570 Buffer *pInBuffer; 5571 Buffer inBuffer; 5572 uint32_t channelCount = mChannelCount; 5573 bool outputBufferFull = false; 5574 inBuffer.frameCount = frames; 5575 inBuffer.i16 = data; 5576 5577 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5578 5579 if (!mActive && frames != 0) { 5580 start(); 5581 sp<ThreadBase> thread = mThread.promote(); 5582 if (thread != 0) { 5583 MixerThread *mixerThread = (MixerThread *)thread.get(); 5584 if (mFrameCount > frames){ 5585 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5586 uint32_t startFrames = (mFrameCount - frames); 5587 pInBuffer = new Buffer; 5588 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5589 pInBuffer->frameCount = startFrames; 5590 pInBuffer->i16 = pInBuffer->mBuffer; 5591 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5592 mBufferQueue.add(pInBuffer); 5593 } else { 5594 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5595 } 5596 } 5597 } 5598 } 5599 5600 while (waitTimeLeftMs) { 5601 // First write pending buffers, then new data 5602 if (mBufferQueue.size()) { 5603 pInBuffer = mBufferQueue.itemAt(0); 5604 } else { 5605 pInBuffer = &inBuffer; 5606 } 5607 5608 if (pInBuffer->frameCount == 0) { 5609 break; 5610 } 5611 5612 if (mOutBuffer.frameCount == 0) { 5613 mOutBuffer.frameCount = pInBuffer->frameCount; 5614 nsecs_t startTime = systemTime(); 5615 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5616 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5617 mThread.unsafe_get()); 5618 outputBufferFull = true; 5619 break; 5620 } 5621 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5622 if (waitTimeLeftMs >= waitTimeMs) { 5623 waitTimeLeftMs -= waitTimeMs; 5624 } else { 5625 waitTimeLeftMs = 0; 5626 } 5627 } 5628 5629 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5630 pInBuffer->frameCount; 5631 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5632 mCblk->stepUserOut(outFrames, mFrameCount); 5633 pInBuffer->frameCount -= outFrames; 5634 pInBuffer->i16 += outFrames * channelCount; 5635 mOutBuffer.frameCount -= outFrames; 5636 mOutBuffer.i16 += outFrames * channelCount; 5637 5638 if (pInBuffer->frameCount == 0) { 5639 if (mBufferQueue.size()) { 5640 mBufferQueue.removeAt(0); 5641 delete [] pInBuffer->mBuffer; 5642 delete pInBuffer; 5643 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5644 mThread.unsafe_get(), mBufferQueue.size()); 5645 } else { 5646 break; 5647 } 5648 } 5649 } 5650 5651 // If we could not write all frames, allocate a buffer and queue it for next time. 5652 if (inBuffer.frameCount) { 5653 sp<ThreadBase> thread = mThread.promote(); 5654 if (thread != 0 && !thread->standby()) { 5655 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5656 pInBuffer = new Buffer; 5657 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5658 pInBuffer->frameCount = inBuffer.frameCount; 5659 pInBuffer->i16 = pInBuffer->mBuffer; 5660 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5661 sizeof(int16_t)); 5662 mBufferQueue.add(pInBuffer); 5663 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5664 mThread.unsafe_get(), mBufferQueue.size()); 5665 } else { 5666 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5667 mThread.unsafe_get(), this); 5668 } 5669 } 5670 } 5671 5672 // Calling write() with a 0 length buffer, means that no more data will be written: 5673 // If no more buffers are pending, fill output track buffer to make sure it is started 5674 // by output mixer. 5675 if (frames == 0 && mBufferQueue.size() == 0) { 5676 if (mCblk->user < mFrameCount) { 5677 frames = mFrameCount - mCblk->user; 5678 pInBuffer = new Buffer; 5679 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5680 pInBuffer->frameCount = frames; 5681 pInBuffer->i16 = pInBuffer->mBuffer; 5682 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5683 mBufferQueue.add(pInBuffer); 5684 } else if (mActive) { 5685 stop(); 5686 } 5687 } 5688 5689 return outputBufferFull; 5690} 5691 5692status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5693 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5694{ 5695 int active; 5696 status_t result; 5697 audio_track_cblk_t* cblk = mCblk; 5698 uint32_t framesReq = buffer->frameCount; 5699 5700 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5701 buffer->frameCount = 0; 5702 5703 uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); 5704 5705 5706 if (framesAvail == 0) { 5707 Mutex::Autolock _l(cblk->lock); 5708 goto start_loop_here; 5709 while (framesAvail == 0) { 5710 active = mActive; 5711 if (CC_UNLIKELY(!active)) { 5712 ALOGV("Not active and NO_MORE_BUFFERS"); 5713 return NO_MORE_BUFFERS; 5714 } 5715 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5716 if (result != NO_ERROR) { 5717 return NO_MORE_BUFFERS; 5718 } 5719 // read the server count again 5720 start_loop_here: 5721 framesAvail = cblk->framesAvailableOut_l(mFrameCount); 5722 } 5723 } 5724 5725// if (framesAvail < framesReq) { 5726// return NO_MORE_BUFFERS; 5727// } 5728 5729 if (framesReq > framesAvail) { 5730 framesReq = framesAvail; 5731 } 5732 5733 uint32_t u = cblk->user; 5734 uint32_t bufferEnd = cblk->userBase + mFrameCount; 5735 5736 if (framesReq > bufferEnd - u) { 5737 framesReq = bufferEnd - u; 5738 } 5739 5740 buffer->frameCount = framesReq; 5741 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 5742 return NO_ERROR; 5743} 5744 5745 5746void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5747{ 5748 size_t size = mBufferQueue.size(); 5749 5750 for (size_t i = 0; i < size; i++) { 5751 Buffer *pBuffer = mBufferQueue.itemAt(i); 5752 delete [] pBuffer->mBuffer; 5753 delete pBuffer; 5754 } 5755 mBufferQueue.clear(); 5756} 5757 5758// ---------------------------------------------------------------------------- 5759 5760AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5761 : RefBase(), 5762 mAudioFlinger(audioFlinger), 5763 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5764 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5765 mPid(pid), 5766 mTimedTrackCount(0) 5767{ 5768 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5769} 5770 5771// Client destructor must be called with AudioFlinger::mLock held 5772AudioFlinger::Client::~Client() 5773{ 5774 mAudioFlinger->removeClient_l(mPid); 5775} 5776 5777sp<MemoryDealer> AudioFlinger::Client::heap() const 5778{ 5779 return mMemoryDealer; 5780} 5781 5782// Reserve one of the limited slots for a timed audio track associated 5783// with this client 5784bool AudioFlinger::Client::reserveTimedTrack() 5785{ 5786 const int kMaxTimedTracksPerClient = 4; 5787 5788 Mutex::Autolock _l(mTimedTrackLock); 5789 5790 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5791 ALOGW("can not create timed track - pid %d has exceeded the limit", 5792 mPid); 5793 return false; 5794 } 5795 5796 mTimedTrackCount++; 5797 return true; 5798} 5799 5800// Release a slot for a timed audio track 5801void AudioFlinger::Client::releaseTimedTrack() 5802{ 5803 Mutex::Autolock _l(mTimedTrackLock); 5804 mTimedTrackCount--; 5805} 5806 5807// ---------------------------------------------------------------------------- 5808 5809AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5810 const sp<IAudioFlingerClient>& client, 5811 pid_t pid) 5812 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5813{ 5814} 5815 5816AudioFlinger::NotificationClient::~NotificationClient() 5817{ 5818} 5819 5820void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5821{ 5822 sp<NotificationClient> keep(this); 5823 mAudioFlinger->removeNotificationClient(mPid); 5824} 5825 5826// ---------------------------------------------------------------------------- 5827 5828AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5829 : BnAudioTrack(), 5830 mTrack(track) 5831{ 5832} 5833 5834AudioFlinger::TrackHandle::~TrackHandle() { 5835 // just stop the track on deletion, associated resources 5836 // will be freed from the main thread once all pending buffers have 5837 // been played. Unless it's not in the active track list, in which 5838 // case we free everything now... 5839 mTrack->destroy(); 5840} 5841 5842sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5843 return mTrack->getCblk(); 5844} 5845 5846status_t AudioFlinger::TrackHandle::start() { 5847 return mTrack->start(); 5848} 5849 5850void AudioFlinger::TrackHandle::stop() { 5851 mTrack->stop(); 5852} 5853 5854void AudioFlinger::TrackHandle::flush() { 5855 mTrack->flush(); 5856} 5857 5858void AudioFlinger::TrackHandle::mute(bool e) { 5859 mTrack->mute(e); 5860} 5861 5862void AudioFlinger::TrackHandle::pause() { 5863 mTrack->pause(); 5864} 5865 5866status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5867{ 5868 return mTrack->attachAuxEffect(EffectId); 5869} 5870 5871status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5872 sp<IMemory>* buffer) { 5873 if (!mTrack->isTimedTrack()) 5874 return INVALID_OPERATION; 5875 5876 PlaybackThread::TimedTrack* tt = 5877 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5878 return tt->allocateTimedBuffer(size, buffer); 5879} 5880 5881status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5882 int64_t pts) { 5883 if (!mTrack->isTimedTrack()) 5884 return INVALID_OPERATION; 5885 5886 PlaybackThread::TimedTrack* tt = 5887 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5888 return tt->queueTimedBuffer(buffer, pts); 5889} 5890 5891status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5892 const LinearTransform& xform, int target) { 5893 5894 if (!mTrack->isTimedTrack()) 5895 return INVALID_OPERATION; 5896 5897 PlaybackThread::TimedTrack* tt = 5898 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5899 return tt->setMediaTimeTransform( 5900 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5901} 5902 5903status_t AudioFlinger::TrackHandle::onTransact( 5904 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5905{ 5906 return BnAudioTrack::onTransact(code, data, reply, flags); 5907} 5908 5909// ---------------------------------------------------------------------------- 5910 5911sp<IAudioRecord> AudioFlinger::openRecord( 5912 pid_t pid, 5913 audio_io_handle_t input, 5914 uint32_t sampleRate, 5915 audio_format_t format, 5916 audio_channel_mask_t channelMask, 5917 size_t frameCount, 5918 IAudioFlinger::track_flags_t flags, 5919 pid_t tid, 5920 int *sessionId, 5921 status_t *status) 5922{ 5923 sp<RecordThread::RecordTrack> recordTrack; 5924 sp<RecordHandle> recordHandle; 5925 sp<Client> client; 5926 status_t lStatus; 5927 RecordThread *thread; 5928 size_t inFrameCount; 5929 int lSessionId; 5930 5931 // check calling permissions 5932 if (!recordingAllowed()) { 5933 lStatus = PERMISSION_DENIED; 5934 goto Exit; 5935 } 5936 5937 // add client to list 5938 { // scope for mLock 5939 Mutex::Autolock _l(mLock); 5940 thread = checkRecordThread_l(input); 5941 if (thread == NULL) { 5942 lStatus = BAD_VALUE; 5943 goto Exit; 5944 } 5945 5946 client = registerPid_l(pid); 5947 5948 // If no audio session id is provided, create one here 5949 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5950 lSessionId = *sessionId; 5951 } else { 5952 lSessionId = nextUniqueId(); 5953 if (sessionId != NULL) { 5954 *sessionId = lSessionId; 5955 } 5956 } 5957 // create new record track. 5958 // The record track uses one track in mHardwareMixerThread by convention. 5959 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5960 frameCount, lSessionId, flags, tid, &lStatus); 5961 } 5962 if (lStatus != NO_ERROR) { 5963 // remove local strong reference to Client before deleting the RecordTrack so that the 5964 // Client destructor is called by the TrackBase destructor with mLock held 5965 client.clear(); 5966 recordTrack.clear(); 5967 goto Exit; 5968 } 5969 5970 // return to handle to client 5971 recordHandle = new RecordHandle(recordTrack); 5972 lStatus = NO_ERROR; 5973 5974Exit: 5975 if (status) { 5976 *status = lStatus; 5977 } 5978 return recordHandle; 5979} 5980 5981// ---------------------------------------------------------------------------- 5982 5983AudioFlinger::RecordHandle::RecordHandle( 5984 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5985 : BnAudioRecord(), 5986 mRecordTrack(recordTrack) 5987{ 5988} 5989 5990AudioFlinger::RecordHandle::~RecordHandle() { 5991 stop_nonvirtual(); 5992 mRecordTrack->destroy(); 5993} 5994 5995sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5996 return mRecordTrack->getCblk(); 5997} 5998 5999status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6000 int triggerSession) { 6001 ALOGV("RecordHandle::start()"); 6002 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6003} 6004 6005void AudioFlinger::RecordHandle::stop() { 6006 stop_nonvirtual(); 6007} 6008 6009void AudioFlinger::RecordHandle::stop_nonvirtual() { 6010 ALOGV("RecordHandle::stop()"); 6011 mRecordTrack->stop(); 6012} 6013 6014status_t AudioFlinger::RecordHandle::onTransact( 6015 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6016{ 6017 return BnAudioRecord::onTransact(code, data, reply, flags); 6018} 6019 6020// ---------------------------------------------------------------------------- 6021 6022AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6023 AudioStreamIn *input, 6024 uint32_t sampleRate, 6025 audio_channel_mask_t channelMask, 6026 audio_io_handle_t id, 6027 audio_devices_t device, 6028 const sp<NBAIO_Sink>& teeSink) : 6029 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6030 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6031 // mRsmpInIndex and mInputBytes set by readInputParameters() 6032 mReqChannelCount(popcount(channelMask)), 6033 mReqSampleRate(sampleRate), 6034 // mBytesRead is only meaningful while active, and so is cleared in start() 6035 // (but might be better to also clear here for dump?) 6036 mTeeSink(teeSink) 6037{ 6038 snprintf(mName, kNameLength, "AudioIn_%X", id); 6039 6040 readInputParameters(); 6041 6042} 6043 6044 6045AudioFlinger::RecordThread::~RecordThread() 6046{ 6047 delete[] mRsmpInBuffer; 6048 delete mResampler; 6049 delete[] mRsmpOutBuffer; 6050} 6051 6052void AudioFlinger::RecordThread::onFirstRef() 6053{ 6054 run(mName, PRIORITY_URGENT_AUDIO); 6055} 6056 6057status_t AudioFlinger::RecordThread::readyToRun() 6058{ 6059 status_t status = initCheck(); 6060 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6061 return status; 6062} 6063 6064bool AudioFlinger::RecordThread::threadLoop() 6065{ 6066 AudioBufferProvider::Buffer buffer; 6067 sp<RecordTrack> activeTrack; 6068 Vector< sp<EffectChain> > effectChains; 6069 6070 nsecs_t lastWarning = 0; 6071 6072 inputStandBy(); 6073 acquireWakeLock(); 6074 6075 // used to verify we've read at least once before evaluating how many bytes were read 6076 bool readOnce = false; 6077 6078 // start recording 6079 while (!exitPending()) { 6080 6081 processConfigEvents(); 6082 6083 { // scope for mLock 6084 Mutex::Autolock _l(mLock); 6085 checkForNewParameters_l(); 6086 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6087 standby(); 6088 6089 if (exitPending()) break; 6090 6091 releaseWakeLock_l(); 6092 ALOGV("RecordThread: loop stopping"); 6093 // go to sleep 6094 mWaitWorkCV.wait(mLock); 6095 ALOGV("RecordThread: loop starting"); 6096 acquireWakeLock_l(); 6097 continue; 6098 } 6099 if (mActiveTrack != 0) { 6100 if (mActiveTrack->mState == TrackBase::PAUSING) { 6101 standby(); 6102 mActiveTrack.clear(); 6103 mStartStopCond.broadcast(); 6104 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6105 if (mReqChannelCount != mActiveTrack->channelCount()) { 6106 mActiveTrack.clear(); 6107 mStartStopCond.broadcast(); 6108 } else if (readOnce) { 6109 // record start succeeds only if first read from audio input 6110 // succeeds 6111 if (mBytesRead >= 0) { 6112 mActiveTrack->mState = TrackBase::ACTIVE; 6113 } else { 6114 mActiveTrack.clear(); 6115 } 6116 mStartStopCond.broadcast(); 6117 } 6118 mStandby = false; 6119 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6120 removeTrack_l(mActiveTrack); 6121 mActiveTrack.clear(); 6122 } 6123 } 6124 lockEffectChains_l(effectChains); 6125 } 6126 6127 if (mActiveTrack != 0) { 6128 if (mActiveTrack->mState != TrackBase::ACTIVE && 6129 mActiveTrack->mState != TrackBase::RESUMING) { 6130 unlockEffectChains(effectChains); 6131 usleep(kRecordThreadSleepUs); 6132 continue; 6133 } 6134 for (size_t i = 0; i < effectChains.size(); i ++) { 6135 effectChains[i]->process_l(); 6136 } 6137 6138 buffer.frameCount = mFrameCount; 6139 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6140 readOnce = true; 6141 size_t framesOut = buffer.frameCount; 6142 if (mResampler == NULL) { 6143 // no resampling 6144 while (framesOut) { 6145 size_t framesIn = mFrameCount - mRsmpInIndex; 6146 if (framesIn) { 6147 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6148 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6149 mActiveTrack->mFrameSize; 6150 if (framesIn > framesOut) 6151 framesIn = framesOut; 6152 mRsmpInIndex += framesIn; 6153 framesOut -= framesIn; 6154 if ((int)mChannelCount == mReqChannelCount || 6155 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6156 memcpy(dst, src, framesIn * mFrameSize); 6157 } else { 6158 if (mChannelCount == 1) { 6159 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6160 (int16_t *)src, framesIn); 6161 } else { 6162 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6163 (int16_t *)src, framesIn); 6164 } 6165 } 6166 } 6167 if (framesOut && mFrameCount == mRsmpInIndex) { 6168 void *readInto; 6169 if (framesOut == mFrameCount && 6170 ((int)mChannelCount == mReqChannelCount || 6171 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6172 readInto = buffer.raw; 6173 framesOut = 0; 6174 } else { 6175 readInto = mRsmpInBuffer; 6176 mRsmpInIndex = 0; 6177 } 6178 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6179 if (mBytesRead <= 0) { 6180 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6181 { 6182 ALOGE("Error reading audio input"); 6183 // Force input into standby so that it tries to 6184 // recover at next read attempt 6185 inputStandBy(); 6186 usleep(kRecordThreadSleepUs); 6187 } 6188 mRsmpInIndex = mFrameCount; 6189 framesOut = 0; 6190 buffer.frameCount = 0; 6191 } else if (mTeeSink != 0) { 6192 (void) mTeeSink->write(readInto, 6193 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6194 } 6195 } 6196 } 6197 } else { 6198 // resampling 6199 6200 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6201 // alter output frame count as if we were expecting stereo samples 6202 if (mChannelCount == 1 && mReqChannelCount == 1) { 6203 framesOut >>= 1; 6204 } 6205 mResampler->resample(mRsmpOutBuffer, framesOut, 6206 this /* AudioBufferProvider* */); 6207 // ditherAndClamp() works as long as all buffers returned by 6208 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6209 if (mChannelCount == 2 && mReqChannelCount == 1) { 6210 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6211 // the resampler always outputs stereo samples: 6212 // do post stereo to mono conversion 6213 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6214 framesOut); 6215 } else { 6216 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6217 } 6218 6219 } 6220 if (mFramestoDrop == 0) { 6221 mActiveTrack->releaseBuffer(&buffer); 6222 } else { 6223 if (mFramestoDrop > 0) { 6224 mFramestoDrop -= buffer.frameCount; 6225 if (mFramestoDrop <= 0) { 6226 clearSyncStartEvent(); 6227 } 6228 } else { 6229 mFramestoDrop += buffer.frameCount; 6230 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6231 mSyncStartEvent->isCancelled()) { 6232 ALOGW("Synced record %s, session %d, trigger session %d", 6233 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6234 mActiveTrack->sessionId(), 6235 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6236 clearSyncStartEvent(); 6237 } 6238 } 6239 } 6240 mActiveTrack->clearOverflow(); 6241 } 6242 // client isn't retrieving buffers fast enough 6243 else { 6244 if (!mActiveTrack->setOverflow()) { 6245 nsecs_t now = systemTime(); 6246 if ((now - lastWarning) > kWarningThrottleNs) { 6247 ALOGW("RecordThread: buffer overflow"); 6248 lastWarning = now; 6249 } 6250 } 6251 // Release the processor for a while before asking for a new buffer. 6252 // This will give the application more chance to read from the buffer and 6253 // clear the overflow. 6254 usleep(kRecordThreadSleepUs); 6255 } 6256 } 6257 // enable changes in effect chain 6258 unlockEffectChains(effectChains); 6259 effectChains.clear(); 6260 } 6261 6262 standby(); 6263 6264 { 6265 Mutex::Autolock _l(mLock); 6266 mActiveTrack.clear(); 6267 mStartStopCond.broadcast(); 6268 } 6269 6270 releaseWakeLock(); 6271 6272 ALOGV("RecordThread %p exiting", this); 6273 return false; 6274} 6275 6276void AudioFlinger::RecordThread::standby() 6277{ 6278 if (!mStandby) { 6279 inputStandBy(); 6280 mStandby = true; 6281 } 6282} 6283 6284void AudioFlinger::RecordThread::inputStandBy() 6285{ 6286 mInput->stream->common.standby(&mInput->stream->common); 6287} 6288 6289sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6290 const sp<AudioFlinger::Client>& client, 6291 uint32_t sampleRate, 6292 audio_format_t format, 6293 audio_channel_mask_t channelMask, 6294 size_t frameCount, 6295 int sessionId, 6296 IAudioFlinger::track_flags_t flags, 6297 pid_t tid, 6298 status_t *status) 6299{ 6300 sp<RecordTrack> track; 6301 status_t lStatus; 6302 6303 lStatus = initCheck(); 6304 if (lStatus != NO_ERROR) { 6305 ALOGE("Audio driver not initialized."); 6306 goto Exit; 6307 } 6308 6309 // FIXME use flags and tid similar to createTrack_l() 6310 6311 { // scope for mLock 6312 Mutex::Autolock _l(mLock); 6313 6314 track = new RecordTrack(this, client, sampleRate, 6315 format, channelMask, frameCount, sessionId); 6316 6317 if (track->getCblk() == 0) { 6318 lStatus = NO_MEMORY; 6319 goto Exit; 6320 } 6321 mTracks.add(track); 6322 6323 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6324 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6325 mAudioFlinger->btNrecIsOff(); 6326 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6327 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6328 } 6329 lStatus = NO_ERROR; 6330 6331Exit: 6332 if (status) { 6333 *status = lStatus; 6334 } 6335 return track; 6336} 6337 6338status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6339 AudioSystem::sync_event_t event, 6340 int triggerSession) 6341{ 6342 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6343 sp<ThreadBase> strongMe = this; 6344 status_t status = NO_ERROR; 6345 6346 if (event == AudioSystem::SYNC_EVENT_NONE) { 6347 clearSyncStartEvent(); 6348 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6349 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6350 triggerSession, 6351 recordTrack->sessionId(), 6352 syncStartEventCallback, 6353 this); 6354 // Sync event can be cancelled by the trigger session if the track is not in a 6355 // compatible state in which case we start record immediately 6356 if (mSyncStartEvent->isCancelled()) { 6357 clearSyncStartEvent(); 6358 } else { 6359 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6360 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6361 } 6362 } 6363 6364 { 6365 AutoMutex lock(mLock); 6366 if (mActiveTrack != 0) { 6367 if (recordTrack != mActiveTrack.get()) { 6368 status = -EBUSY; 6369 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6370 mActiveTrack->mState = TrackBase::ACTIVE; 6371 } 6372 return status; 6373 } 6374 6375 recordTrack->mState = TrackBase::IDLE; 6376 mActiveTrack = recordTrack; 6377 mLock.unlock(); 6378 status_t status = AudioSystem::startInput(mId); 6379 mLock.lock(); 6380 if (status != NO_ERROR) { 6381 mActiveTrack.clear(); 6382 clearSyncStartEvent(); 6383 return status; 6384 } 6385 mRsmpInIndex = mFrameCount; 6386 mBytesRead = 0; 6387 if (mResampler != NULL) { 6388 mResampler->reset(); 6389 } 6390 mActiveTrack->mState = TrackBase::RESUMING; 6391 // signal thread to start 6392 ALOGV("Signal record thread"); 6393 mWaitWorkCV.broadcast(); 6394 // do not wait for mStartStopCond if exiting 6395 if (exitPending()) { 6396 mActiveTrack.clear(); 6397 status = INVALID_OPERATION; 6398 goto startError; 6399 } 6400 mStartStopCond.wait(mLock); 6401 if (mActiveTrack == 0) { 6402 ALOGV("Record failed to start"); 6403 status = BAD_VALUE; 6404 goto startError; 6405 } 6406 ALOGV("Record started OK"); 6407 return status; 6408 } 6409startError: 6410 AudioSystem::stopInput(mId); 6411 clearSyncStartEvent(); 6412 return status; 6413} 6414 6415void AudioFlinger::RecordThread::clearSyncStartEvent() 6416{ 6417 if (mSyncStartEvent != 0) { 6418 mSyncStartEvent->cancel(); 6419 } 6420 mSyncStartEvent.clear(); 6421 mFramestoDrop = 0; 6422} 6423 6424void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6425{ 6426 sp<SyncEvent> strongEvent = event.promote(); 6427 6428 if (strongEvent != 0) { 6429 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6430 me->handleSyncStartEvent(strongEvent); 6431 } 6432} 6433 6434void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6435{ 6436 if (event == mSyncStartEvent) { 6437 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6438 // from audio HAL 6439 mFramestoDrop = mFrameCount * 2; 6440 } 6441} 6442 6443bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6444 ALOGV("RecordThread::stop"); 6445 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6446 return false; 6447 } 6448 recordTrack->mState = TrackBase::PAUSING; 6449 // do not wait for mStartStopCond if exiting 6450 if (exitPending()) { 6451 return true; 6452 } 6453 mStartStopCond.wait(mLock); 6454 // if we have been restarted, recordTrack == mActiveTrack.get() here 6455 if (exitPending() || recordTrack != mActiveTrack.get()) { 6456 ALOGV("Record stopped OK"); 6457 return true; 6458 } 6459 return false; 6460} 6461 6462bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6463{ 6464 return false; 6465} 6466 6467status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6468{ 6469#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6470 if (!isValidSyncEvent(event)) { 6471 return BAD_VALUE; 6472 } 6473 6474 int eventSession = event->triggerSession(); 6475 status_t ret = NAME_NOT_FOUND; 6476 6477 Mutex::Autolock _l(mLock); 6478 6479 for (size_t i = 0; i < mTracks.size(); i++) { 6480 sp<RecordTrack> track = mTracks[i]; 6481 if (eventSession == track->sessionId()) { 6482 (void) track->setSyncEvent(event); 6483 ret = NO_ERROR; 6484 } 6485 } 6486 return ret; 6487#else 6488 return BAD_VALUE; 6489#endif 6490} 6491 6492void AudioFlinger::RecordThread::RecordTrack::destroy() 6493{ 6494 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6495 sp<RecordTrack> keep(this); 6496 { 6497 sp<ThreadBase> thread = mThread.promote(); 6498 if (thread != 0) { 6499 if (mState == ACTIVE || mState == RESUMING) { 6500 AudioSystem::stopInput(thread->id()); 6501 } 6502 AudioSystem::releaseInput(thread->id()); 6503 Mutex::Autolock _l(thread->mLock); 6504 RecordThread *recordThread = (RecordThread *) thread.get(); 6505 recordThread->destroyTrack_l(this); 6506 } 6507 } 6508} 6509 6510// destroyTrack_l() must be called with ThreadBase::mLock held 6511void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6512{ 6513 track->mState = TrackBase::TERMINATED; 6514 // active tracks are removed by threadLoop() 6515 if (mActiveTrack != track) { 6516 removeTrack_l(track); 6517 } 6518} 6519 6520void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6521{ 6522 mTracks.remove(track); 6523 // need anything related to effects here? 6524} 6525 6526void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6527{ 6528 dumpInternals(fd, args); 6529 dumpTracks(fd, args); 6530 dumpEffectChains(fd, args); 6531} 6532 6533void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6534{ 6535 const size_t SIZE = 256; 6536 char buffer[SIZE]; 6537 String8 result; 6538 6539 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6540 result.append(buffer); 6541 6542 if (mActiveTrack != 0) { 6543 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6544 result.append(buffer); 6545 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6546 result.append(buffer); 6547 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6548 result.append(buffer); 6549 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6550 result.append(buffer); 6551 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 6552 result.append(buffer); 6553 } else { 6554 result.append("No active record client\n"); 6555 } 6556 6557 write(fd, result.string(), result.size()); 6558 6559 dumpBase(fd, args); 6560} 6561 6562void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6563{ 6564 const size_t SIZE = 256; 6565 char buffer[SIZE]; 6566 String8 result; 6567 6568 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6569 result.append(buffer); 6570 RecordTrack::appendDumpHeader(result); 6571 for (size_t i = 0; i < mTracks.size(); ++i) { 6572 sp<RecordTrack> track = mTracks[i]; 6573 if (track != 0) { 6574 track->dump(buffer, SIZE); 6575 result.append(buffer); 6576 } 6577 } 6578 6579 if (mActiveTrack != 0) { 6580 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6581 result.append(buffer); 6582 RecordTrack::appendDumpHeader(result); 6583 mActiveTrack->dump(buffer, SIZE); 6584 result.append(buffer); 6585 6586 } 6587 write(fd, result.string(), result.size()); 6588} 6589 6590// AudioBufferProvider interface 6591status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6592{ 6593 size_t framesReq = buffer->frameCount; 6594 size_t framesReady = mFrameCount - mRsmpInIndex; 6595 int channelCount; 6596 6597 if (framesReady == 0) { 6598 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6599 if (mBytesRead <= 0) { 6600 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6601 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6602 // Force input into standby so that it tries to 6603 // recover at next read attempt 6604 inputStandBy(); 6605 usleep(kRecordThreadSleepUs); 6606 } 6607 buffer->raw = NULL; 6608 buffer->frameCount = 0; 6609 return NOT_ENOUGH_DATA; 6610 } 6611 mRsmpInIndex = 0; 6612 framesReady = mFrameCount; 6613 } 6614 6615 if (framesReq > framesReady) { 6616 framesReq = framesReady; 6617 } 6618 6619 if (mChannelCount == 1 && mReqChannelCount == 2) { 6620 channelCount = 1; 6621 } else { 6622 channelCount = 2; 6623 } 6624 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6625 buffer->frameCount = framesReq; 6626 return NO_ERROR; 6627} 6628 6629// AudioBufferProvider interface 6630void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6631{ 6632 mRsmpInIndex += buffer->frameCount; 6633 buffer->frameCount = 0; 6634} 6635 6636bool AudioFlinger::RecordThread::checkForNewParameters_l() 6637{ 6638 bool reconfig = false; 6639 6640 while (!mNewParameters.isEmpty()) { 6641 status_t status = NO_ERROR; 6642 String8 keyValuePair = mNewParameters[0]; 6643 AudioParameter param = AudioParameter(keyValuePair); 6644 int value; 6645 audio_format_t reqFormat = mFormat; 6646 uint32_t reqSamplingRate = mReqSampleRate; 6647 int reqChannelCount = mReqChannelCount; 6648 6649 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6650 reqSamplingRate = value; 6651 reconfig = true; 6652 } 6653 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6654 reqFormat = (audio_format_t) value; 6655 reconfig = true; 6656 } 6657 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6658 reqChannelCount = popcount(value); 6659 reconfig = true; 6660 } 6661 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6662 // do not accept frame count changes if tracks are open as the track buffer 6663 // size depends on frame count and correct behavior would not be guaranteed 6664 // if frame count is changed after track creation 6665 if (mActiveTrack != 0) { 6666 status = INVALID_OPERATION; 6667 } else { 6668 reconfig = true; 6669 } 6670 } 6671 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6672 // forward device change to effects that have requested to be 6673 // aware of attached audio device. 6674 for (size_t i = 0; i < mEffectChains.size(); i++) { 6675 mEffectChains[i]->setDevice_l(value); 6676 } 6677 6678 // store input device and output device but do not forward output device to audio HAL. 6679 // Note that status is ignored by the caller for output device 6680 // (see AudioFlinger::setParameters() 6681 if (audio_is_output_devices(value)) { 6682 mOutDevice = value; 6683 status = BAD_VALUE; 6684 } else { 6685 mInDevice = value; 6686 // disable AEC and NS if the device is a BT SCO headset supporting those 6687 // pre processings 6688 if (mTracks.size() > 0) { 6689 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6690 mAudioFlinger->btNrecIsOff(); 6691 for (size_t i = 0; i < mTracks.size(); i++) { 6692 sp<RecordTrack> track = mTracks[i]; 6693 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6694 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6695 } 6696 } 6697 } 6698 } 6699 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6700 mAudioSource != (audio_source_t)value) { 6701 // forward device change to effects that have requested to be 6702 // aware of attached audio device. 6703 for (size_t i = 0; i < mEffectChains.size(); i++) { 6704 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6705 } 6706 mAudioSource = (audio_source_t)value; 6707 } 6708 if (status == NO_ERROR) { 6709 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6710 keyValuePair.string()); 6711 if (status == INVALID_OPERATION) { 6712 inputStandBy(); 6713 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6714 keyValuePair.string()); 6715 } 6716 if (reconfig) { 6717 if (status == BAD_VALUE && 6718 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6719 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6720 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6721 <= (2 * reqSamplingRate)) && 6722 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6723 <= FCC_2 && 6724 (reqChannelCount <= FCC_2)) { 6725 status = NO_ERROR; 6726 } 6727 if (status == NO_ERROR) { 6728 readInputParameters(); 6729 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6730 } 6731 } 6732 } 6733 6734 mNewParameters.removeAt(0); 6735 6736 mParamStatus = status; 6737 mParamCond.signal(); 6738 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6739 // already timed out waiting for the status and will never signal the condition. 6740 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6741 } 6742 return reconfig; 6743} 6744 6745String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6746{ 6747 char *s; 6748 String8 out_s8 = String8(); 6749 6750 Mutex::Autolock _l(mLock); 6751 if (initCheck() != NO_ERROR) { 6752 return out_s8; 6753 } 6754 6755 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6756 out_s8 = String8(s); 6757 free(s); 6758 return out_s8; 6759} 6760 6761void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6762 AudioSystem::OutputDescriptor desc; 6763 void *param2 = NULL; 6764 6765 switch (event) { 6766 case AudioSystem::INPUT_OPENED: 6767 case AudioSystem::INPUT_CONFIG_CHANGED: 6768 desc.channels = mChannelMask; 6769 desc.samplingRate = mSampleRate; 6770 desc.format = mFormat; 6771 desc.frameCount = mFrameCount; 6772 desc.latency = 0; 6773 param2 = &desc; 6774 break; 6775 6776 case AudioSystem::INPUT_CLOSED: 6777 default: 6778 break; 6779 } 6780 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6781} 6782 6783void AudioFlinger::RecordThread::readInputParameters() 6784{ 6785 delete mRsmpInBuffer; 6786 // mRsmpInBuffer is always assigned a new[] below 6787 delete mRsmpOutBuffer; 6788 mRsmpOutBuffer = NULL; 6789 delete mResampler; 6790 mResampler = NULL; 6791 6792 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6793 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6794 mChannelCount = (uint16_t)popcount(mChannelMask); 6795 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6796 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6797 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6798 mFrameCount = mInputBytes / mFrameSize; 6799 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6800 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6801 6802 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6803 { 6804 int channelCount; 6805 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6806 // stereo to mono post process as the resampler always outputs stereo. 6807 if (mChannelCount == 1 && mReqChannelCount == 2) { 6808 channelCount = 1; 6809 } else { 6810 channelCount = 2; 6811 } 6812 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6813 mResampler->setSampleRate(mSampleRate); 6814 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6815 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6816 6817 // optmization: if mono to mono, alter input frame count as if we were inputing 6818 // stereo samples 6819 if (mChannelCount == 1 && mReqChannelCount == 1) { 6820 mFrameCount >>= 1; 6821 } 6822 6823 } 6824 mRsmpInIndex = mFrameCount; 6825} 6826 6827unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6828{ 6829 Mutex::Autolock _l(mLock); 6830 if (initCheck() != NO_ERROR) { 6831 return 0; 6832 } 6833 6834 return mInput->stream->get_input_frames_lost(mInput->stream); 6835} 6836 6837uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6838{ 6839 Mutex::Autolock _l(mLock); 6840 uint32_t result = 0; 6841 if (getEffectChain_l(sessionId) != 0) { 6842 result = EFFECT_SESSION; 6843 } 6844 6845 for (size_t i = 0; i < mTracks.size(); ++i) { 6846 if (sessionId == mTracks[i]->sessionId()) { 6847 result |= TRACK_SESSION; 6848 break; 6849 } 6850 } 6851 6852 return result; 6853} 6854 6855KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6856{ 6857 KeyedVector<int, bool> ids; 6858 Mutex::Autolock _l(mLock); 6859 for (size_t j = 0; j < mTracks.size(); ++j) { 6860 sp<RecordThread::RecordTrack> track = mTracks[j]; 6861 int sessionId = track->sessionId(); 6862 if (ids.indexOfKey(sessionId) < 0) { 6863 ids.add(sessionId, true); 6864 } 6865 } 6866 return ids; 6867} 6868 6869AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6870{ 6871 Mutex::Autolock _l(mLock); 6872 AudioStreamIn *input = mInput; 6873 mInput = NULL; 6874 return input; 6875} 6876 6877// this method must always be called either with ThreadBase mLock held or inside the thread loop 6878audio_stream_t* AudioFlinger::RecordThread::stream() const 6879{ 6880 if (mInput == NULL) { 6881 return NULL; 6882 } 6883 return &mInput->stream->common; 6884} 6885 6886 6887// ---------------------------------------------------------------------------- 6888 6889audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6890{ 6891 if (!settingsAllowed()) { 6892 return 0; 6893 } 6894 Mutex::Autolock _l(mLock); 6895 return loadHwModule_l(name); 6896} 6897 6898// loadHwModule_l() must be called with AudioFlinger::mLock held 6899audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6900{ 6901 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6902 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6903 ALOGW("loadHwModule() module %s already loaded", name); 6904 return mAudioHwDevs.keyAt(i); 6905 } 6906 } 6907 6908 audio_hw_device_t *dev; 6909 6910 int rc = load_audio_interface(name, &dev); 6911 if (rc) { 6912 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6913 return 0; 6914 } 6915 6916 mHardwareStatus = AUDIO_HW_INIT; 6917 rc = dev->init_check(dev); 6918 mHardwareStatus = AUDIO_HW_IDLE; 6919 if (rc) { 6920 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6921 return 0; 6922 } 6923 6924 // Check and cache this HAL's level of support for master mute and master 6925 // volume. If this is the first HAL opened, and it supports the get 6926 // methods, use the initial values provided by the HAL as the current 6927 // master mute and volume settings. 6928 6929 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6930 { // scope for auto-lock pattern 6931 AutoMutex lock(mHardwareLock); 6932 6933 if (0 == mAudioHwDevs.size()) { 6934 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6935 if (NULL != dev->get_master_volume) { 6936 float mv; 6937 if (OK == dev->get_master_volume(dev, &mv)) { 6938 mMasterVolume = mv; 6939 } 6940 } 6941 6942 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6943 if (NULL != dev->get_master_mute) { 6944 bool mm; 6945 if (OK == dev->get_master_mute(dev, &mm)) { 6946 mMasterMute = mm; 6947 } 6948 } 6949 } 6950 6951 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6952 if ((NULL != dev->set_master_volume) && 6953 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6954 flags = static_cast<AudioHwDevice::Flags>(flags | 6955 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6956 } 6957 6958 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6959 if ((NULL != dev->set_master_mute) && 6960 (OK == dev->set_master_mute(dev, mMasterMute))) { 6961 flags = static_cast<AudioHwDevice::Flags>(flags | 6962 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6963 } 6964 6965 mHardwareStatus = AUDIO_HW_IDLE; 6966 } 6967 6968 audio_module_handle_t handle = nextUniqueId(); 6969 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6970 6971 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6972 name, dev->common.module->name, dev->common.module->id, handle); 6973 6974 return handle; 6975 6976} 6977 6978// ---------------------------------------------------------------------------- 6979 6980uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 6981{ 6982 Mutex::Autolock _l(mLock); 6983 PlaybackThread *thread = primaryPlaybackThread_l(); 6984 return thread != NULL ? thread->sampleRate() : 0; 6985} 6986 6987size_t AudioFlinger::getPrimaryOutputFrameCount() 6988{ 6989 Mutex::Autolock _l(mLock); 6990 PlaybackThread *thread = primaryPlaybackThread_l(); 6991 return thread != NULL ? thread->frameCountHAL() : 0; 6992} 6993 6994// ---------------------------------------------------------------------------- 6995 6996audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6997 audio_devices_t *pDevices, 6998 uint32_t *pSamplingRate, 6999 audio_format_t *pFormat, 7000 audio_channel_mask_t *pChannelMask, 7001 uint32_t *pLatencyMs, 7002 audio_output_flags_t flags) 7003{ 7004 status_t status; 7005 PlaybackThread *thread = NULL; 7006 struct audio_config config = { 7007 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7008 channel_mask: pChannelMask ? *pChannelMask : 0, 7009 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7010 }; 7011 audio_stream_out_t *outStream = NULL; 7012 AudioHwDevice *outHwDev; 7013 7014 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7015 module, 7016 (pDevices != NULL) ? *pDevices : 0, 7017 config.sample_rate, 7018 config.format, 7019 config.channel_mask, 7020 flags); 7021 7022 if (pDevices == NULL || *pDevices == 0) { 7023 return 0; 7024 } 7025 7026 Mutex::Autolock _l(mLock); 7027 7028 outHwDev = findSuitableHwDev_l(module, *pDevices); 7029 if (outHwDev == NULL) 7030 return 0; 7031 7032 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7033 audio_io_handle_t id = nextUniqueId(); 7034 7035 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7036 7037 status = hwDevHal->open_output_stream(hwDevHal, 7038 id, 7039 *pDevices, 7040 (audio_output_flags_t)flags, 7041 &config, 7042 &outStream); 7043 7044 mHardwareStatus = AUDIO_HW_IDLE; 7045 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7046 "Channels %x, status %d", 7047 outStream, 7048 config.sample_rate, 7049 config.format, 7050 config.channel_mask, 7051 status); 7052 7053 if (status == NO_ERROR && outStream != NULL) { 7054 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7055 7056 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7057 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7058 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7059 thread = new DirectOutputThread(this, output, id, *pDevices); 7060 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7061 } else { 7062 thread = new MixerThread(this, output, id, *pDevices); 7063 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7064 } 7065 mPlaybackThreads.add(id, thread); 7066 7067 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7068 if (pFormat != NULL) *pFormat = config.format; 7069 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7070 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7071 7072 // notify client processes of the new output creation 7073 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7074 7075 // the first primary output opened designates the primary hw device 7076 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7077 ALOGI("Using module %d has the primary audio interface", module); 7078 mPrimaryHardwareDev = outHwDev; 7079 7080 AutoMutex lock(mHardwareLock); 7081 mHardwareStatus = AUDIO_HW_SET_MODE; 7082 hwDevHal->set_mode(hwDevHal, mMode); 7083 mHardwareStatus = AUDIO_HW_IDLE; 7084 } 7085 return id; 7086 } 7087 7088 return 0; 7089} 7090 7091audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7092 audio_io_handle_t output2) 7093{ 7094 Mutex::Autolock _l(mLock); 7095 MixerThread *thread1 = checkMixerThread_l(output1); 7096 MixerThread *thread2 = checkMixerThread_l(output2); 7097 7098 if (thread1 == NULL || thread2 == NULL) { 7099 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7100 output2); 7101 return 0; 7102 } 7103 7104 audio_io_handle_t id = nextUniqueId(); 7105 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7106 thread->addOutputTrack(thread2); 7107 mPlaybackThreads.add(id, thread); 7108 // notify client processes of the new output creation 7109 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7110 return id; 7111} 7112 7113status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7114{ 7115 return closeOutput_nonvirtual(output); 7116} 7117 7118status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7119{ 7120 // keep strong reference on the playback thread so that 7121 // it is not destroyed while exit() is executed 7122 sp<PlaybackThread> thread; 7123 { 7124 Mutex::Autolock _l(mLock); 7125 thread = checkPlaybackThread_l(output); 7126 if (thread == NULL) { 7127 return BAD_VALUE; 7128 } 7129 7130 ALOGV("closeOutput() %d", output); 7131 7132 if (thread->type() == ThreadBase::MIXER) { 7133 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7134 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7135 DuplicatingThread *dupThread = 7136 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7137 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7138 } 7139 } 7140 } 7141 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7142 mPlaybackThreads.removeItem(output); 7143 } 7144 thread->exit(); 7145 // The thread entity (active unit of execution) is no longer running here, 7146 // but the ThreadBase container still exists. 7147 7148 if (thread->type() != ThreadBase::DUPLICATING) { 7149 AudioStreamOut *out = thread->clearOutput(); 7150 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7151 // from now on thread->mOutput is NULL 7152 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7153 delete out; 7154 } 7155 return NO_ERROR; 7156} 7157 7158status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7159{ 7160 Mutex::Autolock _l(mLock); 7161 PlaybackThread *thread = checkPlaybackThread_l(output); 7162 7163 if (thread == NULL) { 7164 return BAD_VALUE; 7165 } 7166 7167 ALOGV("suspendOutput() %d", output); 7168 thread->suspend(); 7169 7170 return NO_ERROR; 7171} 7172 7173status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7174{ 7175 Mutex::Autolock _l(mLock); 7176 PlaybackThread *thread = checkPlaybackThread_l(output); 7177 7178 if (thread == NULL) { 7179 return BAD_VALUE; 7180 } 7181 7182 ALOGV("restoreOutput() %d", output); 7183 7184 thread->restore(); 7185 7186 return NO_ERROR; 7187} 7188 7189audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7190 audio_devices_t *pDevices, 7191 uint32_t *pSamplingRate, 7192 audio_format_t *pFormat, 7193 audio_channel_mask_t *pChannelMask) 7194{ 7195 status_t status; 7196 RecordThread *thread = NULL; 7197 struct audio_config config = { 7198 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7199 channel_mask: pChannelMask ? *pChannelMask : 0, 7200 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7201 }; 7202 uint32_t reqSamplingRate = config.sample_rate; 7203 audio_format_t reqFormat = config.format; 7204 audio_channel_mask_t reqChannels = config.channel_mask; 7205 audio_stream_in_t *inStream = NULL; 7206 AudioHwDevice *inHwDev; 7207 7208 if (pDevices == NULL || *pDevices == 0) { 7209 return 0; 7210 } 7211 7212 Mutex::Autolock _l(mLock); 7213 7214 inHwDev = findSuitableHwDev_l(module, *pDevices); 7215 if (inHwDev == NULL) 7216 return 0; 7217 7218 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7219 audio_io_handle_t id = nextUniqueId(); 7220 7221 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7222 &inStream); 7223 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7224 "status %d", 7225 inStream, 7226 config.sample_rate, 7227 config.format, 7228 config.channel_mask, 7229 status); 7230 7231 // If the input could not be opened with the requested parameters and we can handle the 7232 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7233 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7234 if (status == BAD_VALUE && 7235 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7236 (config.sample_rate <= 2 * reqSamplingRate) && 7237 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7238 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7239 inStream = NULL; 7240 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7241 } 7242 7243 if (status == NO_ERROR && inStream != NULL) { 7244 7245 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7246 // or (re-)create if current Pipe is idle and does not match the new format 7247 sp<NBAIO_Sink> teeSink; 7248#ifdef TEE_SINK_INPUT_FRAMES 7249 enum { 7250 TEE_SINK_NO, // don't copy input 7251 TEE_SINK_NEW, // copy input using a new pipe 7252 TEE_SINK_OLD, // copy input using an existing pipe 7253 } kind; 7254 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7255 popcount(inStream->common.get_channels(&inStream->common))); 7256 if (format == Format_Invalid) { 7257 kind = TEE_SINK_NO; 7258 } else if (mRecordTeeSink == 0) { 7259 kind = TEE_SINK_NEW; 7260 } else if (mRecordTeeSink->getStrongCount() != 1) { 7261 kind = TEE_SINK_NO; 7262 } else if (format == mRecordTeeSink->format()) { 7263 kind = TEE_SINK_OLD; 7264 } else { 7265 kind = TEE_SINK_NEW; 7266 } 7267 switch (kind) { 7268 case TEE_SINK_NEW: { 7269 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7270 size_t numCounterOffers = 0; 7271 const NBAIO_Format offers[1] = {format}; 7272 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7273 ALOG_ASSERT(index == 0); 7274 PipeReader *pipeReader = new PipeReader(*pipe); 7275 numCounterOffers = 0; 7276 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7277 ALOG_ASSERT(index == 0); 7278 mRecordTeeSink = pipe; 7279 mRecordTeeSource = pipeReader; 7280 teeSink = pipe; 7281 } 7282 break; 7283 case TEE_SINK_OLD: 7284 teeSink = mRecordTeeSink; 7285 break; 7286 case TEE_SINK_NO: 7287 default: 7288 break; 7289 } 7290#endif 7291 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7292 7293 // Start record thread 7294 // RecorThread require both input and output device indication to forward to audio 7295 // pre processing modules 7296 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7297 7298 thread = new RecordThread(this, 7299 input, 7300 reqSamplingRate, 7301 reqChannels, 7302 id, 7303 device, teeSink); 7304 mRecordThreads.add(id, thread); 7305 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7306 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7307 if (pFormat != NULL) *pFormat = config.format; 7308 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7309 7310 // notify client processes of the new input creation 7311 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7312 return id; 7313 } 7314 7315 return 0; 7316} 7317 7318status_t AudioFlinger::closeInput(audio_io_handle_t input) 7319{ 7320 return closeInput_nonvirtual(input); 7321} 7322 7323status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7324{ 7325 // keep strong reference on the record thread so that 7326 // it is not destroyed while exit() is executed 7327 sp<RecordThread> thread; 7328 { 7329 Mutex::Autolock _l(mLock); 7330 thread = checkRecordThread_l(input); 7331 if (thread == 0) { 7332 return BAD_VALUE; 7333 } 7334 7335 ALOGV("closeInput() %d", input); 7336 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7337 mRecordThreads.removeItem(input); 7338 } 7339 thread->exit(); 7340 // The thread entity (active unit of execution) is no longer running here, 7341 // but the ThreadBase container still exists. 7342 7343 AudioStreamIn *in = thread->clearInput(); 7344 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7345 // from now on thread->mInput is NULL 7346 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7347 delete in; 7348 7349 return NO_ERROR; 7350} 7351 7352status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7353{ 7354 Mutex::Autolock _l(mLock); 7355 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7356 7357 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7358 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7359 thread->invalidateTracks(stream); 7360 } 7361 7362 return NO_ERROR; 7363} 7364 7365 7366int AudioFlinger::newAudioSessionId() 7367{ 7368 return nextUniqueId(); 7369} 7370 7371void AudioFlinger::acquireAudioSessionId(int audioSession) 7372{ 7373 Mutex::Autolock _l(mLock); 7374 pid_t caller = IPCThreadState::self()->getCallingPid(); 7375 ALOGV("acquiring %d from %d", audioSession, caller); 7376 size_t num = mAudioSessionRefs.size(); 7377 for (size_t i = 0; i< num; i++) { 7378 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7379 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7380 ref->mCnt++; 7381 ALOGV(" incremented refcount to %d", ref->mCnt); 7382 return; 7383 } 7384 } 7385 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7386 ALOGV(" added new entry for %d", audioSession); 7387} 7388 7389void AudioFlinger::releaseAudioSessionId(int audioSession) 7390{ 7391 Mutex::Autolock _l(mLock); 7392 pid_t caller = IPCThreadState::self()->getCallingPid(); 7393 ALOGV("releasing %d from %d", audioSession, caller); 7394 size_t num = mAudioSessionRefs.size(); 7395 for (size_t i = 0; i< num; i++) { 7396 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7397 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7398 ref->mCnt--; 7399 ALOGV(" decremented refcount to %d", ref->mCnt); 7400 if (ref->mCnt == 0) { 7401 mAudioSessionRefs.removeAt(i); 7402 delete ref; 7403 purgeStaleEffects_l(); 7404 } 7405 return; 7406 } 7407 } 7408 ALOGW("session id %d not found for pid %d", audioSession, caller); 7409} 7410 7411void AudioFlinger::purgeStaleEffects_l() { 7412 7413 ALOGV("purging stale effects"); 7414 7415 Vector< sp<EffectChain> > chains; 7416 7417 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7418 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7419 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7420 sp<EffectChain> ec = t->mEffectChains[j]; 7421 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7422 chains.push(ec); 7423 } 7424 } 7425 } 7426 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7427 sp<RecordThread> t = mRecordThreads.valueAt(i); 7428 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7429 sp<EffectChain> ec = t->mEffectChains[j]; 7430 chains.push(ec); 7431 } 7432 } 7433 7434 for (size_t i = 0; i < chains.size(); i++) { 7435 sp<EffectChain> ec = chains[i]; 7436 int sessionid = ec->sessionId(); 7437 sp<ThreadBase> t = ec->mThread.promote(); 7438 if (t == 0) { 7439 continue; 7440 } 7441 size_t numsessionrefs = mAudioSessionRefs.size(); 7442 bool found = false; 7443 for (size_t k = 0; k < numsessionrefs; k++) { 7444 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7445 if (ref->mSessionid == sessionid) { 7446 ALOGV(" session %d still exists for %d with %d refs", 7447 sessionid, ref->mPid, ref->mCnt); 7448 found = true; 7449 break; 7450 } 7451 } 7452 if (!found) { 7453 Mutex::Autolock _l (t->mLock); 7454 // remove all effects from the chain 7455 while (ec->mEffects.size()) { 7456 sp<EffectModule> effect = ec->mEffects[0]; 7457 effect->unPin(); 7458 t->removeEffect_l(effect); 7459 if (effect->purgeHandles()) { 7460 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7461 } 7462 AudioSystem::unregisterEffect(effect->id()); 7463 } 7464 } 7465 } 7466 return; 7467} 7468 7469// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7470AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7471{ 7472 return mPlaybackThreads.valueFor(output).get(); 7473} 7474 7475// checkMixerThread_l() must be called with AudioFlinger::mLock held 7476AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7477{ 7478 PlaybackThread *thread = checkPlaybackThread_l(output); 7479 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7480} 7481 7482// checkRecordThread_l() must be called with AudioFlinger::mLock held 7483AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7484{ 7485 return mRecordThreads.valueFor(input).get(); 7486} 7487 7488uint32_t AudioFlinger::nextUniqueId() 7489{ 7490 return android_atomic_inc(&mNextUniqueId); 7491} 7492 7493AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7494{ 7495 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7496 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7497 AudioStreamOut *output = thread->getOutput(); 7498 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7499 return thread; 7500 } 7501 } 7502 return NULL; 7503} 7504 7505audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7506{ 7507 PlaybackThread *thread = primaryPlaybackThread_l(); 7508 7509 if (thread == NULL) { 7510 return 0; 7511 } 7512 7513 return thread->outDevice(); 7514} 7515 7516sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7517 int triggerSession, 7518 int listenerSession, 7519 sync_event_callback_t callBack, 7520 void *cookie) 7521{ 7522 Mutex::Autolock _l(mLock); 7523 7524 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7525 status_t playStatus = NAME_NOT_FOUND; 7526 status_t recStatus = NAME_NOT_FOUND; 7527 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7528 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7529 if (playStatus == NO_ERROR) { 7530 return event; 7531 } 7532 } 7533 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7534 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7535 if (recStatus == NO_ERROR) { 7536 return event; 7537 } 7538 } 7539 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7540 mPendingSyncEvents.add(event); 7541 } else { 7542 ALOGV("createSyncEvent() invalid event %d", event->type()); 7543 event.clear(); 7544 } 7545 return event; 7546} 7547 7548// ---------------------------------------------------------------------------- 7549// Effect management 7550// ---------------------------------------------------------------------------- 7551 7552 7553status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7554{ 7555 Mutex::Autolock _l(mLock); 7556 return EffectQueryNumberEffects(numEffects); 7557} 7558 7559status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7560{ 7561 Mutex::Autolock _l(mLock); 7562 return EffectQueryEffect(index, descriptor); 7563} 7564 7565status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7566 effect_descriptor_t *descriptor) const 7567{ 7568 Mutex::Autolock _l(mLock); 7569 return EffectGetDescriptor(pUuid, descriptor); 7570} 7571 7572 7573sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7574 effect_descriptor_t *pDesc, 7575 const sp<IEffectClient>& effectClient, 7576 int32_t priority, 7577 audio_io_handle_t io, 7578 int sessionId, 7579 status_t *status, 7580 int *id, 7581 int *enabled) 7582{ 7583 status_t lStatus = NO_ERROR; 7584 sp<EffectHandle> handle; 7585 effect_descriptor_t desc; 7586 7587 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7588 pid, effectClient.get(), priority, sessionId, io); 7589 7590 if (pDesc == NULL) { 7591 lStatus = BAD_VALUE; 7592 goto Exit; 7593 } 7594 7595 // check audio settings permission for global effects 7596 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7597 lStatus = PERMISSION_DENIED; 7598 goto Exit; 7599 } 7600 7601 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7602 // that can only be created by audio policy manager (running in same process) 7603 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7604 lStatus = PERMISSION_DENIED; 7605 goto Exit; 7606 } 7607 7608 if (io == 0) { 7609 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7610 // output must be specified by AudioPolicyManager when using session 7611 // AUDIO_SESSION_OUTPUT_STAGE 7612 lStatus = BAD_VALUE; 7613 goto Exit; 7614 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7615 // if the output returned by getOutputForEffect() is removed before we lock the 7616 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7617 // and we will exit safely 7618 io = AudioSystem::getOutputForEffect(&desc); 7619 } 7620 } 7621 7622 { 7623 Mutex::Autolock _l(mLock); 7624 7625 7626 if (!EffectIsNullUuid(&pDesc->uuid)) { 7627 // if uuid is specified, request effect descriptor 7628 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7629 if (lStatus < 0) { 7630 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7631 goto Exit; 7632 } 7633 } else { 7634 // if uuid is not specified, look for an available implementation 7635 // of the required type in effect factory 7636 if (EffectIsNullUuid(&pDesc->type)) { 7637 ALOGW("createEffect() no effect type"); 7638 lStatus = BAD_VALUE; 7639 goto Exit; 7640 } 7641 uint32_t numEffects = 0; 7642 effect_descriptor_t d; 7643 d.flags = 0; // prevent compiler warning 7644 bool found = false; 7645 7646 lStatus = EffectQueryNumberEffects(&numEffects); 7647 if (lStatus < 0) { 7648 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7649 goto Exit; 7650 } 7651 for (uint32_t i = 0; i < numEffects; i++) { 7652 lStatus = EffectQueryEffect(i, &desc); 7653 if (lStatus < 0) { 7654 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7655 continue; 7656 } 7657 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7658 // If matching type found save effect descriptor. If the session is 7659 // 0 and the effect is not auxiliary, continue enumeration in case 7660 // an auxiliary version of this effect type is available 7661 found = true; 7662 d = desc; 7663 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7664 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7665 break; 7666 } 7667 } 7668 } 7669 if (!found) { 7670 lStatus = BAD_VALUE; 7671 ALOGW("createEffect() effect not found"); 7672 goto Exit; 7673 } 7674 // For same effect type, chose auxiliary version over insert version if 7675 // connect to output mix (Compliance to OpenSL ES) 7676 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7677 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7678 desc = d; 7679 } 7680 } 7681 7682 // Do not allow auxiliary effects on a session different from 0 (output mix) 7683 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7684 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7685 lStatus = INVALID_OPERATION; 7686 goto Exit; 7687 } 7688 7689 // check recording permission for visualizer 7690 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7691 !recordingAllowed()) { 7692 lStatus = PERMISSION_DENIED; 7693 goto Exit; 7694 } 7695 7696 // return effect descriptor 7697 *pDesc = desc; 7698 7699 // If output is not specified try to find a matching audio session ID in one of the 7700 // output threads. 7701 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7702 // because of code checking output when entering the function. 7703 // Note: io is never 0 when creating an effect on an input 7704 if (io == 0) { 7705 // look for the thread where the specified audio session is present 7706 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7707 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7708 io = mPlaybackThreads.keyAt(i); 7709 break; 7710 } 7711 } 7712 if (io == 0) { 7713 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7714 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7715 io = mRecordThreads.keyAt(i); 7716 break; 7717 } 7718 } 7719 } 7720 // If no output thread contains the requested session ID, default to 7721 // first output. The effect chain will be moved to the correct output 7722 // thread when a track with the same session ID is created 7723 if (io == 0 && mPlaybackThreads.size()) { 7724 io = mPlaybackThreads.keyAt(0); 7725 } 7726 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7727 } 7728 ThreadBase *thread = checkRecordThread_l(io); 7729 if (thread == NULL) { 7730 thread = checkPlaybackThread_l(io); 7731 if (thread == NULL) { 7732 ALOGE("createEffect() unknown output thread"); 7733 lStatus = BAD_VALUE; 7734 goto Exit; 7735 } 7736 } 7737 7738 sp<Client> client = registerPid_l(pid); 7739 7740 // create effect on selected output thread 7741 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7742 &desc, enabled, &lStatus); 7743 if (handle != 0 && id != NULL) { 7744 *id = handle->id(); 7745 } 7746 } 7747 7748Exit: 7749 if (status != NULL) { 7750 *status = lStatus; 7751 } 7752 return handle; 7753} 7754 7755status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7756 audio_io_handle_t dstOutput) 7757{ 7758 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7759 sessionId, srcOutput, dstOutput); 7760 Mutex::Autolock _l(mLock); 7761 if (srcOutput == dstOutput) { 7762 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7763 return NO_ERROR; 7764 } 7765 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7766 if (srcThread == NULL) { 7767 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7768 return BAD_VALUE; 7769 } 7770 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7771 if (dstThread == NULL) { 7772 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7773 return BAD_VALUE; 7774 } 7775 7776 Mutex::Autolock _dl(dstThread->mLock); 7777 Mutex::Autolock _sl(srcThread->mLock); 7778 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7779 7780 return NO_ERROR; 7781} 7782 7783// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7784status_t AudioFlinger::moveEffectChain_l(int sessionId, 7785 AudioFlinger::PlaybackThread *srcThread, 7786 AudioFlinger::PlaybackThread *dstThread, 7787 bool reRegister) 7788{ 7789 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7790 sessionId, srcThread, dstThread); 7791 7792 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7793 if (chain == 0) { 7794 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7795 sessionId, srcThread); 7796 return INVALID_OPERATION; 7797 } 7798 7799 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7800 // so that a new chain is created with correct parameters when first effect is added. This is 7801 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7802 // removed. 7803 srcThread->removeEffectChain_l(chain); 7804 7805 // transfer all effects one by one so that new effect chain is created on new thread with 7806 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7807 audio_io_handle_t dstOutput = dstThread->id(); 7808 sp<EffectChain> dstChain; 7809 uint32_t strategy = 0; // prevent compiler warning 7810 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7811 while (effect != 0) { 7812 srcThread->removeEffect_l(effect); 7813 dstThread->addEffect_l(effect); 7814 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7815 if (effect->state() == EffectModule::ACTIVE || 7816 effect->state() == EffectModule::STOPPING) { 7817 effect->start(); 7818 } 7819 // if the move request is not received from audio policy manager, the effect must be 7820 // re-registered with the new strategy and output 7821 if (dstChain == 0) { 7822 dstChain = effect->chain().promote(); 7823 if (dstChain == 0) { 7824 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7825 srcThread->addEffect_l(effect); 7826 return NO_INIT; 7827 } 7828 strategy = dstChain->strategy(); 7829 } 7830 if (reRegister) { 7831 AudioSystem::unregisterEffect(effect->id()); 7832 AudioSystem::registerEffect(&effect->desc(), 7833 dstOutput, 7834 strategy, 7835 sessionId, 7836 effect->id()); 7837 } 7838 effect = chain->getEffectFromId_l(0); 7839 } 7840 7841 return NO_ERROR; 7842} 7843 7844 7845// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7846sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7847 const sp<AudioFlinger::Client>& client, 7848 const sp<IEffectClient>& effectClient, 7849 int32_t priority, 7850 int sessionId, 7851 effect_descriptor_t *desc, 7852 int *enabled, 7853 status_t *status 7854 ) 7855{ 7856 sp<EffectModule> effect; 7857 sp<EffectHandle> handle; 7858 status_t lStatus; 7859 sp<EffectChain> chain; 7860 bool chainCreated = false; 7861 bool effectCreated = false; 7862 bool effectRegistered = false; 7863 7864 lStatus = initCheck(); 7865 if (lStatus != NO_ERROR) { 7866 ALOGW("createEffect_l() Audio driver not initialized."); 7867 goto Exit; 7868 } 7869 7870 // Do not allow effects with session ID 0 on direct output or duplicating threads 7871 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7872 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7873 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7874 desc->name, sessionId); 7875 lStatus = BAD_VALUE; 7876 goto Exit; 7877 } 7878 // Only Pre processor effects are allowed on input threads and only on input threads 7879 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7880 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7881 desc->name, desc->flags, mType); 7882 lStatus = BAD_VALUE; 7883 goto Exit; 7884 } 7885 7886 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7887 7888 { // scope for mLock 7889 Mutex::Autolock _l(mLock); 7890 7891 // check for existing effect chain with the requested audio session 7892 chain = getEffectChain_l(sessionId); 7893 if (chain == 0) { 7894 // create a new chain for this session 7895 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7896 chain = new EffectChain(this, sessionId); 7897 addEffectChain_l(chain); 7898 chain->setStrategy(getStrategyForSession_l(sessionId)); 7899 chainCreated = true; 7900 } else { 7901 effect = chain->getEffectFromDesc_l(desc); 7902 } 7903 7904 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7905 7906 if (effect == 0) { 7907 int id = mAudioFlinger->nextUniqueId(); 7908 // Check CPU and memory usage 7909 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7910 if (lStatus != NO_ERROR) { 7911 goto Exit; 7912 } 7913 effectRegistered = true; 7914 // create a new effect module if none present in the chain 7915 effect = new EffectModule(this, chain, desc, id, sessionId); 7916 lStatus = effect->status(); 7917 if (lStatus != NO_ERROR) { 7918 goto Exit; 7919 } 7920 lStatus = chain->addEffect_l(effect); 7921 if (lStatus != NO_ERROR) { 7922 goto Exit; 7923 } 7924 effectCreated = true; 7925 7926 effect->setDevice(mOutDevice); 7927 effect->setDevice(mInDevice); 7928 effect->setMode(mAudioFlinger->getMode()); 7929 effect->setAudioSource(mAudioSource); 7930 } 7931 // create effect handle and connect it to effect module 7932 handle = new EffectHandle(effect, client, effectClient, priority); 7933 lStatus = effect->addHandle(handle.get()); 7934 if (enabled != NULL) { 7935 *enabled = (int)effect->isEnabled(); 7936 } 7937 } 7938 7939Exit: 7940 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7941 Mutex::Autolock _l(mLock); 7942 if (effectCreated) { 7943 chain->removeEffect_l(effect); 7944 } 7945 if (effectRegistered) { 7946 AudioSystem::unregisterEffect(effect->id()); 7947 } 7948 if (chainCreated) { 7949 removeEffectChain_l(chain); 7950 } 7951 handle.clear(); 7952 } 7953 7954 if (status != NULL) { 7955 *status = lStatus; 7956 } 7957 return handle; 7958} 7959 7960sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7961{ 7962 Mutex::Autolock _l(mLock); 7963 return getEffect_l(sessionId, effectId); 7964} 7965 7966sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7967{ 7968 sp<EffectChain> chain = getEffectChain_l(sessionId); 7969 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7970} 7971 7972// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7973// PlaybackThread::mLock held 7974status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7975{ 7976 // check for existing effect chain with the requested audio session 7977 int sessionId = effect->sessionId(); 7978 sp<EffectChain> chain = getEffectChain_l(sessionId); 7979 bool chainCreated = false; 7980 7981 if (chain == 0) { 7982 // create a new chain for this session 7983 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7984 chain = new EffectChain(this, sessionId); 7985 addEffectChain_l(chain); 7986 chain->setStrategy(getStrategyForSession_l(sessionId)); 7987 chainCreated = true; 7988 } 7989 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7990 7991 if (chain->getEffectFromId_l(effect->id()) != 0) { 7992 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7993 this, effect->desc().name, chain.get()); 7994 return BAD_VALUE; 7995 } 7996 7997 status_t status = chain->addEffect_l(effect); 7998 if (status != NO_ERROR) { 7999 if (chainCreated) { 8000 removeEffectChain_l(chain); 8001 } 8002 return status; 8003 } 8004 8005 effect->setDevice(mOutDevice); 8006 effect->setDevice(mInDevice); 8007 effect->setMode(mAudioFlinger->getMode()); 8008 effect->setAudioSource(mAudioSource); 8009 return NO_ERROR; 8010} 8011 8012void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8013 8014 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8015 effect_descriptor_t desc = effect->desc(); 8016 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8017 detachAuxEffect_l(effect->id()); 8018 } 8019 8020 sp<EffectChain> chain = effect->chain().promote(); 8021 if (chain != 0) { 8022 // remove effect chain if removing last effect 8023 if (chain->removeEffect_l(effect) == 0) { 8024 removeEffectChain_l(chain); 8025 } 8026 } else { 8027 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8028 } 8029} 8030 8031void AudioFlinger::ThreadBase::lockEffectChains_l( 8032 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8033{ 8034 effectChains = mEffectChains; 8035 for (size_t i = 0; i < mEffectChains.size(); i++) { 8036 mEffectChains[i]->lock(); 8037 } 8038} 8039 8040void AudioFlinger::ThreadBase::unlockEffectChains( 8041 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8042{ 8043 for (size_t i = 0; i < effectChains.size(); i++) { 8044 effectChains[i]->unlock(); 8045 } 8046} 8047 8048sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8049{ 8050 Mutex::Autolock _l(mLock); 8051 return getEffectChain_l(sessionId); 8052} 8053 8054sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8055{ 8056 size_t size = mEffectChains.size(); 8057 for (size_t i = 0; i < size; i++) { 8058 if (mEffectChains[i]->sessionId() == sessionId) { 8059 return mEffectChains[i]; 8060 } 8061 } 8062 return 0; 8063} 8064 8065void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8066{ 8067 Mutex::Autolock _l(mLock); 8068 size_t size = mEffectChains.size(); 8069 for (size_t i = 0; i < size; i++) { 8070 mEffectChains[i]->setMode_l(mode); 8071 } 8072} 8073 8074void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8075 EffectHandle *handle, 8076 bool unpinIfLast) { 8077 8078 Mutex::Autolock _l(mLock); 8079 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8080 // delete the effect module if removing last handle on it 8081 if (effect->removeHandle(handle) == 0) { 8082 if (!effect->isPinned() || unpinIfLast) { 8083 removeEffect_l(effect); 8084 AudioSystem::unregisterEffect(effect->id()); 8085 } 8086 } 8087} 8088 8089status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8090{ 8091 int session = chain->sessionId(); 8092 int16_t *buffer = mMixBuffer; 8093 bool ownsBuffer = false; 8094 8095 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8096 if (session > 0) { 8097 // Only one effect chain can be present in direct output thread and it uses 8098 // the mix buffer as input 8099 if (mType != DIRECT) { 8100 size_t numSamples = mNormalFrameCount * mChannelCount; 8101 buffer = new int16_t[numSamples]; 8102 memset(buffer, 0, numSamples * sizeof(int16_t)); 8103 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8104 ownsBuffer = true; 8105 } 8106 8107 // Attach all tracks with same session ID to this chain. 8108 for (size_t i = 0; i < mTracks.size(); ++i) { 8109 sp<Track> track = mTracks[i]; 8110 if (session == track->sessionId()) { 8111 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8112 buffer); 8113 track->setMainBuffer(buffer); 8114 chain->incTrackCnt(); 8115 } 8116 } 8117 8118 // indicate all active tracks in the chain 8119 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8120 sp<Track> track = mActiveTracks[i].promote(); 8121 if (track == 0) continue; 8122 if (session == track->sessionId()) { 8123 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8124 chain->incActiveTrackCnt(); 8125 } 8126 } 8127 } 8128 8129 chain->setInBuffer(buffer, ownsBuffer); 8130 chain->setOutBuffer(mMixBuffer); 8131 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8132 // chains list in order to be processed last as it contains output stage effects 8133 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8134 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8135 // after track specific effects and before output stage 8136 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8137 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8138 // Effect chain for other sessions are inserted at beginning of effect 8139 // chains list to be processed before output mix effects. Relative order between other 8140 // sessions is not important 8141 size_t size = mEffectChains.size(); 8142 size_t i = 0; 8143 for (i = 0; i < size; i++) { 8144 if (mEffectChains[i]->sessionId() < session) break; 8145 } 8146 mEffectChains.insertAt(chain, i); 8147 checkSuspendOnAddEffectChain_l(chain); 8148 8149 return NO_ERROR; 8150} 8151 8152size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8153{ 8154 int session = chain->sessionId(); 8155 8156 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8157 8158 for (size_t i = 0; i < mEffectChains.size(); i++) { 8159 if (chain == mEffectChains[i]) { 8160 mEffectChains.removeAt(i); 8161 // detach all active tracks from the chain 8162 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8163 sp<Track> track = mActiveTracks[i].promote(); 8164 if (track == 0) continue; 8165 if (session == track->sessionId()) { 8166 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8167 chain.get(), session); 8168 chain->decActiveTrackCnt(); 8169 } 8170 } 8171 8172 // detach all tracks with same session ID from this chain 8173 for (size_t i = 0; i < mTracks.size(); ++i) { 8174 sp<Track> track = mTracks[i]; 8175 if (session == track->sessionId()) { 8176 track->setMainBuffer(mMixBuffer); 8177 chain->decTrackCnt(); 8178 } 8179 } 8180 break; 8181 } 8182 } 8183 return mEffectChains.size(); 8184} 8185 8186status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8187 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8188{ 8189 Mutex::Autolock _l(mLock); 8190 return attachAuxEffect_l(track, EffectId); 8191} 8192 8193status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8194 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8195{ 8196 status_t status = NO_ERROR; 8197 8198 if (EffectId == 0) { 8199 track->setAuxBuffer(0, NULL); 8200 } else { 8201 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8202 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8203 if (effect != 0) { 8204 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8205 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8206 } else { 8207 status = INVALID_OPERATION; 8208 } 8209 } else { 8210 status = BAD_VALUE; 8211 } 8212 } 8213 return status; 8214} 8215 8216void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8217{ 8218 for (size_t i = 0; i < mTracks.size(); ++i) { 8219 sp<Track> track = mTracks[i]; 8220 if (track->auxEffectId() == effectId) { 8221 attachAuxEffect_l(track, 0); 8222 } 8223 } 8224} 8225 8226status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8227{ 8228 // only one chain per input thread 8229 if (mEffectChains.size() != 0) { 8230 return INVALID_OPERATION; 8231 } 8232 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8233 8234 chain->setInBuffer(NULL); 8235 chain->setOutBuffer(NULL); 8236 8237 checkSuspendOnAddEffectChain_l(chain); 8238 8239 mEffectChains.add(chain); 8240 8241 return NO_ERROR; 8242} 8243 8244size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8245{ 8246 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8247 ALOGW_IF(mEffectChains.size() != 1, 8248 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8249 chain.get(), mEffectChains.size(), this); 8250 if (mEffectChains.size() == 1) { 8251 mEffectChains.removeAt(0); 8252 } 8253 return 0; 8254} 8255 8256// ---------------------------------------------------------------------------- 8257// EffectModule implementation 8258// ---------------------------------------------------------------------------- 8259 8260#undef LOG_TAG 8261#define LOG_TAG "AudioFlinger::EffectModule" 8262 8263AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8264 const wp<AudioFlinger::EffectChain>& chain, 8265 effect_descriptor_t *desc, 8266 int id, 8267 int sessionId) 8268 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8269 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8270 mDescriptor(*desc), 8271 // mConfig is set by configure() and not used before then 8272 mEffectInterface(NULL), 8273 mStatus(NO_INIT), mState(IDLE), 8274 // mMaxDisableWaitCnt is set by configure() and not used before then 8275 // mDisableWaitCnt is set by process() and updateState() and not used before then 8276 mSuspended(false) 8277{ 8278 ALOGV("Constructor %p", this); 8279 int lStatus; 8280 8281 // create effect engine from effect factory 8282 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8283 8284 if (mStatus != NO_ERROR) { 8285 return; 8286 } 8287 lStatus = init(); 8288 if (lStatus < 0) { 8289 mStatus = lStatus; 8290 goto Error; 8291 } 8292 8293 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8294 return; 8295Error: 8296 EffectRelease(mEffectInterface); 8297 mEffectInterface = NULL; 8298 ALOGV("Constructor Error %d", mStatus); 8299} 8300 8301AudioFlinger::EffectModule::~EffectModule() 8302{ 8303 ALOGV("Destructor %p", this); 8304 if (mEffectInterface != NULL) { 8305 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8306 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8307 sp<ThreadBase> thread = mThread.promote(); 8308 if (thread != 0) { 8309 audio_stream_t *stream = thread->stream(); 8310 if (stream != NULL) { 8311 stream->remove_audio_effect(stream, mEffectInterface); 8312 } 8313 } 8314 } 8315 // release effect engine 8316 EffectRelease(mEffectInterface); 8317 } 8318} 8319 8320status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8321{ 8322 status_t status; 8323 8324 Mutex::Autolock _l(mLock); 8325 int priority = handle->priority(); 8326 size_t size = mHandles.size(); 8327 EffectHandle *controlHandle = NULL; 8328 size_t i; 8329 for (i = 0; i < size; i++) { 8330 EffectHandle *h = mHandles[i]; 8331 if (h == NULL || h->destroyed_l()) continue; 8332 // first non destroyed handle is considered in control 8333 if (controlHandle == NULL) 8334 controlHandle = h; 8335 if (h->priority() <= priority) break; 8336 } 8337 // if inserted in first place, move effect control from previous owner to this handle 8338 if (i == 0) { 8339 bool enabled = false; 8340 if (controlHandle != NULL) { 8341 enabled = controlHandle->enabled(); 8342 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8343 } 8344 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8345 status = NO_ERROR; 8346 } else { 8347 status = ALREADY_EXISTS; 8348 } 8349 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8350 mHandles.insertAt(handle, i); 8351 return status; 8352} 8353 8354size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8355{ 8356 Mutex::Autolock _l(mLock); 8357 size_t size = mHandles.size(); 8358 size_t i; 8359 for (i = 0; i < size; i++) { 8360 if (mHandles[i] == handle) break; 8361 } 8362 if (i == size) { 8363 return size; 8364 } 8365 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8366 8367 mHandles.removeAt(i); 8368 // if removed from first place, move effect control from this handle to next in line 8369 if (i == 0) { 8370 EffectHandle *h = controlHandle_l(); 8371 if (h != NULL) { 8372 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8373 } 8374 } 8375 8376 // Prevent calls to process() and other functions on effect interface from now on. 8377 // The effect engine will be released by the destructor when the last strong reference on 8378 // this object is released which can happen after next process is called. 8379 if (mHandles.size() == 0 && !mPinned) { 8380 mState = DESTROYED; 8381 } 8382 8383 return mHandles.size(); 8384} 8385 8386// must be called with EffectModule::mLock held 8387AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8388{ 8389 // the first valid handle in the list has control over the module 8390 for (size_t i = 0; i < mHandles.size(); i++) { 8391 EffectHandle *h = mHandles[i]; 8392 if (h != NULL && !h->destroyed_l()) { 8393 return h; 8394 } 8395 } 8396 8397 return NULL; 8398} 8399 8400size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8401{ 8402 ALOGV("disconnect() %p handle %p", this, handle); 8403 // keep a strong reference on this EffectModule to avoid calling the 8404 // destructor before we exit 8405 sp<EffectModule> keep(this); 8406 { 8407 sp<ThreadBase> thread = mThread.promote(); 8408 if (thread != 0) { 8409 thread->disconnectEffect(keep, handle, unpinIfLast); 8410 } 8411 } 8412 return mHandles.size(); 8413} 8414 8415void AudioFlinger::EffectModule::updateState() { 8416 Mutex::Autolock _l(mLock); 8417 8418 switch (mState) { 8419 case RESTART: 8420 reset_l(); 8421 // FALL THROUGH 8422 8423 case STARTING: 8424 // clear auxiliary effect input buffer for next accumulation 8425 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8426 memset(mConfig.inputCfg.buffer.raw, 8427 0, 8428 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8429 } 8430 start_l(); 8431 mState = ACTIVE; 8432 break; 8433 case STOPPING: 8434 stop_l(); 8435 mDisableWaitCnt = mMaxDisableWaitCnt; 8436 mState = STOPPED; 8437 break; 8438 case STOPPED: 8439 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8440 // turn off sequence. 8441 if (--mDisableWaitCnt == 0) { 8442 reset_l(); 8443 mState = IDLE; 8444 } 8445 break; 8446 default: //IDLE , ACTIVE, DESTROYED 8447 break; 8448 } 8449} 8450 8451void AudioFlinger::EffectModule::process() 8452{ 8453 Mutex::Autolock _l(mLock); 8454 8455 if (mState == DESTROYED || mEffectInterface == NULL || 8456 mConfig.inputCfg.buffer.raw == NULL || 8457 mConfig.outputCfg.buffer.raw == NULL) { 8458 return; 8459 } 8460 8461 if (isProcessEnabled()) { 8462 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8463 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8464 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8465 mConfig.inputCfg.buffer.s32, 8466 mConfig.inputCfg.buffer.frameCount/2); 8467 } 8468 8469 // do the actual processing in the effect engine 8470 int ret = (*mEffectInterface)->process(mEffectInterface, 8471 &mConfig.inputCfg.buffer, 8472 &mConfig.outputCfg.buffer); 8473 8474 // force transition to IDLE state when engine is ready 8475 if (mState == STOPPED && ret == -ENODATA) { 8476 mDisableWaitCnt = 1; 8477 } 8478 8479 // clear auxiliary effect input buffer for next accumulation 8480 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8481 memset(mConfig.inputCfg.buffer.raw, 0, 8482 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8483 } 8484 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8485 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8486 // If an insert effect is idle and input buffer is different from output buffer, 8487 // accumulate input onto output 8488 sp<EffectChain> chain = mChain.promote(); 8489 if (chain != 0 && chain->activeTrackCnt() != 0) { 8490 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8491 int16_t *in = mConfig.inputCfg.buffer.s16; 8492 int16_t *out = mConfig.outputCfg.buffer.s16; 8493 for (size_t i = 0; i < frameCnt; i++) { 8494 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8495 } 8496 } 8497 } 8498} 8499 8500void AudioFlinger::EffectModule::reset_l() 8501{ 8502 if (mEffectInterface == NULL) { 8503 return; 8504 } 8505 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8506} 8507 8508status_t AudioFlinger::EffectModule::configure() 8509{ 8510 if (mEffectInterface == NULL) { 8511 return NO_INIT; 8512 } 8513 8514 sp<ThreadBase> thread = mThread.promote(); 8515 if (thread == 0) { 8516 return DEAD_OBJECT; 8517 } 8518 8519 // TODO: handle configuration of effects replacing track process 8520 audio_channel_mask_t channelMask = thread->channelMask(); 8521 8522 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8523 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8524 } else { 8525 mConfig.inputCfg.channels = channelMask; 8526 } 8527 mConfig.outputCfg.channels = channelMask; 8528 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8529 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8530 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8531 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8532 mConfig.inputCfg.bufferProvider.cookie = NULL; 8533 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8534 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8535 mConfig.outputCfg.bufferProvider.cookie = NULL; 8536 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8537 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8538 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8539 // Insert effect: 8540 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8541 // always overwrites output buffer: input buffer == output buffer 8542 // - in other sessions: 8543 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8544 // other effect: overwrites output buffer: input buffer == output buffer 8545 // Auxiliary effect: 8546 // accumulates in output buffer: input buffer != output buffer 8547 // Therefore: accumulate <=> input buffer != output buffer 8548 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8549 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8550 } else { 8551 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8552 } 8553 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8554 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8555 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8556 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8557 8558 ALOGV("configure() %p thread %p buffer %p framecount %d", 8559 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8560 8561 status_t cmdStatus; 8562 uint32_t size = sizeof(int); 8563 status_t status = (*mEffectInterface)->command(mEffectInterface, 8564 EFFECT_CMD_SET_CONFIG, 8565 sizeof(effect_config_t), 8566 &mConfig, 8567 &size, 8568 &cmdStatus); 8569 if (status == 0) { 8570 status = cmdStatus; 8571 } 8572 8573 if (status == 0 && 8574 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8575 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8576 effect_param_t *p = (effect_param_t *)buf32; 8577 8578 p->psize = sizeof(uint32_t); 8579 p->vsize = sizeof(uint32_t); 8580 size = sizeof(int); 8581 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8582 8583 uint32_t latency = 0; 8584 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8585 if (pbt != NULL) { 8586 latency = pbt->latency_l(); 8587 } 8588 8589 *((int32_t *)p->data + 1)= latency; 8590 (*mEffectInterface)->command(mEffectInterface, 8591 EFFECT_CMD_SET_PARAM, 8592 sizeof(effect_param_t) + 8, 8593 &buf32, 8594 &size, 8595 &cmdStatus); 8596 } 8597 8598 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8599 (1000 * mConfig.outputCfg.buffer.frameCount); 8600 8601 return status; 8602} 8603 8604status_t AudioFlinger::EffectModule::init() 8605{ 8606 Mutex::Autolock _l(mLock); 8607 if (mEffectInterface == NULL) { 8608 return NO_INIT; 8609 } 8610 status_t cmdStatus; 8611 uint32_t size = sizeof(status_t); 8612 status_t status = (*mEffectInterface)->command(mEffectInterface, 8613 EFFECT_CMD_INIT, 8614 0, 8615 NULL, 8616 &size, 8617 &cmdStatus); 8618 if (status == 0) { 8619 status = cmdStatus; 8620 } 8621 return status; 8622} 8623 8624status_t AudioFlinger::EffectModule::start() 8625{ 8626 Mutex::Autolock _l(mLock); 8627 return start_l(); 8628} 8629 8630status_t AudioFlinger::EffectModule::start_l() 8631{ 8632 if (mEffectInterface == NULL) { 8633 return NO_INIT; 8634 } 8635 status_t cmdStatus; 8636 uint32_t size = sizeof(status_t); 8637 status_t status = (*mEffectInterface)->command(mEffectInterface, 8638 EFFECT_CMD_ENABLE, 8639 0, 8640 NULL, 8641 &size, 8642 &cmdStatus); 8643 if (status == 0) { 8644 status = cmdStatus; 8645 } 8646 if (status == 0 && 8647 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8648 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8649 sp<ThreadBase> thread = mThread.promote(); 8650 if (thread != 0) { 8651 audio_stream_t *stream = thread->stream(); 8652 if (stream != NULL) { 8653 stream->add_audio_effect(stream, mEffectInterface); 8654 } 8655 } 8656 } 8657 return status; 8658} 8659 8660status_t AudioFlinger::EffectModule::stop() 8661{ 8662 Mutex::Autolock _l(mLock); 8663 return stop_l(); 8664} 8665 8666status_t AudioFlinger::EffectModule::stop_l() 8667{ 8668 if (mEffectInterface == NULL) { 8669 return NO_INIT; 8670 } 8671 status_t cmdStatus; 8672 uint32_t size = sizeof(status_t); 8673 status_t status = (*mEffectInterface)->command(mEffectInterface, 8674 EFFECT_CMD_DISABLE, 8675 0, 8676 NULL, 8677 &size, 8678 &cmdStatus); 8679 if (status == 0) { 8680 status = cmdStatus; 8681 } 8682 if (status == 0 && 8683 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8684 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8685 sp<ThreadBase> thread = mThread.promote(); 8686 if (thread != 0) { 8687 audio_stream_t *stream = thread->stream(); 8688 if (stream != NULL) { 8689 stream->remove_audio_effect(stream, mEffectInterface); 8690 } 8691 } 8692 } 8693 return status; 8694} 8695 8696status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8697 uint32_t cmdSize, 8698 void *pCmdData, 8699 uint32_t *replySize, 8700 void *pReplyData) 8701{ 8702 Mutex::Autolock _l(mLock); 8703 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8704 8705 if (mState == DESTROYED || mEffectInterface == NULL) { 8706 return NO_INIT; 8707 } 8708 status_t status = (*mEffectInterface)->command(mEffectInterface, 8709 cmdCode, 8710 cmdSize, 8711 pCmdData, 8712 replySize, 8713 pReplyData); 8714 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8715 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8716 for (size_t i = 1; i < mHandles.size(); i++) { 8717 EffectHandle *h = mHandles[i]; 8718 if (h != NULL && !h->destroyed_l()) { 8719 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8720 } 8721 } 8722 } 8723 return status; 8724} 8725 8726status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8727{ 8728 Mutex::Autolock _l(mLock); 8729 return setEnabled_l(enabled); 8730} 8731 8732// must be called with EffectModule::mLock held 8733status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8734{ 8735 8736 ALOGV("setEnabled %p enabled %d", this, enabled); 8737 8738 if (enabled != isEnabled()) { 8739 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8740 if (enabled && status != NO_ERROR) { 8741 return status; 8742 } 8743 8744 switch (mState) { 8745 // going from disabled to enabled 8746 case IDLE: 8747 mState = STARTING; 8748 break; 8749 case STOPPED: 8750 mState = RESTART; 8751 break; 8752 case STOPPING: 8753 mState = ACTIVE; 8754 break; 8755 8756 // going from enabled to disabled 8757 case RESTART: 8758 mState = STOPPED; 8759 break; 8760 case STARTING: 8761 mState = IDLE; 8762 break; 8763 case ACTIVE: 8764 mState = STOPPING; 8765 break; 8766 case DESTROYED: 8767 return NO_ERROR; // simply ignore as we are being destroyed 8768 } 8769 for (size_t i = 1; i < mHandles.size(); i++) { 8770 EffectHandle *h = mHandles[i]; 8771 if (h != NULL && !h->destroyed_l()) { 8772 h->setEnabled(enabled); 8773 } 8774 } 8775 } 8776 return NO_ERROR; 8777} 8778 8779bool AudioFlinger::EffectModule::isEnabled() const 8780{ 8781 switch (mState) { 8782 case RESTART: 8783 case STARTING: 8784 case ACTIVE: 8785 return true; 8786 case IDLE: 8787 case STOPPING: 8788 case STOPPED: 8789 case DESTROYED: 8790 default: 8791 return false; 8792 } 8793} 8794 8795bool AudioFlinger::EffectModule::isProcessEnabled() const 8796{ 8797 switch (mState) { 8798 case RESTART: 8799 case ACTIVE: 8800 case STOPPING: 8801 case STOPPED: 8802 return true; 8803 case IDLE: 8804 case STARTING: 8805 case DESTROYED: 8806 default: 8807 return false; 8808 } 8809} 8810 8811status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8812{ 8813 Mutex::Autolock _l(mLock); 8814 status_t status = NO_ERROR; 8815 8816 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8817 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8818 if (isProcessEnabled() && 8819 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8820 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8821 status_t cmdStatus; 8822 uint32_t volume[2]; 8823 uint32_t *pVolume = NULL; 8824 uint32_t size = sizeof(volume); 8825 volume[0] = *left; 8826 volume[1] = *right; 8827 if (controller) { 8828 pVolume = volume; 8829 } 8830 status = (*mEffectInterface)->command(mEffectInterface, 8831 EFFECT_CMD_SET_VOLUME, 8832 size, 8833 volume, 8834 &size, 8835 pVolume); 8836 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8837 *left = volume[0]; 8838 *right = volume[1]; 8839 } 8840 } 8841 return status; 8842} 8843 8844status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8845{ 8846 if (device == AUDIO_DEVICE_NONE) { 8847 return NO_ERROR; 8848 } 8849 8850 Mutex::Autolock _l(mLock); 8851 status_t status = NO_ERROR; 8852 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8853 status_t cmdStatus; 8854 uint32_t size = sizeof(status_t); 8855 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8856 EFFECT_CMD_SET_INPUT_DEVICE; 8857 status = (*mEffectInterface)->command(mEffectInterface, 8858 cmd, 8859 sizeof(uint32_t), 8860 &device, 8861 &size, 8862 &cmdStatus); 8863 } 8864 return status; 8865} 8866 8867status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8868{ 8869 Mutex::Autolock _l(mLock); 8870 status_t status = NO_ERROR; 8871 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8872 status_t cmdStatus; 8873 uint32_t size = sizeof(status_t); 8874 status = (*mEffectInterface)->command(mEffectInterface, 8875 EFFECT_CMD_SET_AUDIO_MODE, 8876 sizeof(audio_mode_t), 8877 &mode, 8878 &size, 8879 &cmdStatus); 8880 if (status == NO_ERROR) { 8881 status = cmdStatus; 8882 } 8883 } 8884 return status; 8885} 8886 8887status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8888{ 8889 Mutex::Autolock _l(mLock); 8890 status_t status = NO_ERROR; 8891 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8892 uint32_t size = 0; 8893 status = (*mEffectInterface)->command(mEffectInterface, 8894 EFFECT_CMD_SET_AUDIO_SOURCE, 8895 sizeof(audio_source_t), 8896 &source, 8897 &size, 8898 NULL); 8899 } 8900 return status; 8901} 8902 8903void AudioFlinger::EffectModule::setSuspended(bool suspended) 8904{ 8905 Mutex::Autolock _l(mLock); 8906 mSuspended = suspended; 8907} 8908 8909bool AudioFlinger::EffectModule::suspended() const 8910{ 8911 Mutex::Autolock _l(mLock); 8912 return mSuspended; 8913} 8914 8915bool AudioFlinger::EffectModule::purgeHandles() 8916{ 8917 bool enabled = false; 8918 Mutex::Autolock _l(mLock); 8919 for (size_t i = 0; i < mHandles.size(); i++) { 8920 EffectHandle *handle = mHandles[i]; 8921 if (handle != NULL && !handle->destroyed_l()) { 8922 handle->effect().clear(); 8923 if (handle->hasControl()) { 8924 enabled = handle->enabled(); 8925 } 8926 } 8927 } 8928 return enabled; 8929} 8930 8931void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8932{ 8933 const size_t SIZE = 256; 8934 char buffer[SIZE]; 8935 String8 result; 8936 8937 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8938 result.append(buffer); 8939 8940 bool locked = tryLock(mLock); 8941 // failed to lock - AudioFlinger is probably deadlocked 8942 if (!locked) { 8943 result.append("\t\tCould not lock Fx mutex:\n"); 8944 } 8945 8946 result.append("\t\tSession Status State Engine:\n"); 8947 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8948 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8949 result.append(buffer); 8950 8951 result.append("\t\tDescriptor:\n"); 8952 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8953 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8954 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 8955 mDescriptor.uuid.node[2], 8956 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8957 result.append(buffer); 8958 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8959 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 8960 mDescriptor.type.timeHiAndVersion, 8961 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 8962 mDescriptor.type.node[2], 8963 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8964 result.append(buffer); 8965 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8966 mDescriptor.apiVersion, 8967 mDescriptor.flags); 8968 result.append(buffer); 8969 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8970 mDescriptor.name); 8971 result.append(buffer); 8972 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8973 mDescriptor.implementor); 8974 result.append(buffer); 8975 8976 result.append("\t\t- Input configuration:\n"); 8977 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8978 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8979 (uint32_t)mConfig.inputCfg.buffer.raw, 8980 mConfig.inputCfg.buffer.frameCount, 8981 mConfig.inputCfg.samplingRate, 8982 mConfig.inputCfg.channels, 8983 mConfig.inputCfg.format); 8984 result.append(buffer); 8985 8986 result.append("\t\t- Output configuration:\n"); 8987 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8988 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8989 (uint32_t)mConfig.outputCfg.buffer.raw, 8990 mConfig.outputCfg.buffer.frameCount, 8991 mConfig.outputCfg.samplingRate, 8992 mConfig.outputCfg.channels, 8993 mConfig.outputCfg.format); 8994 result.append(buffer); 8995 8996 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8997 result.append(buffer); 8998 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8999 for (size_t i = 0; i < mHandles.size(); ++i) { 9000 EffectHandle *handle = mHandles[i]; 9001 if (handle != NULL && !handle->destroyed_l()) { 9002 handle->dump(buffer, SIZE); 9003 result.append(buffer); 9004 } 9005 } 9006 9007 result.append("\n"); 9008 9009 write(fd, result.string(), result.length()); 9010 9011 if (locked) { 9012 mLock.unlock(); 9013 } 9014} 9015 9016// ---------------------------------------------------------------------------- 9017// EffectHandle implementation 9018// ---------------------------------------------------------------------------- 9019 9020#undef LOG_TAG 9021#define LOG_TAG "AudioFlinger::EffectHandle" 9022 9023AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9024 const sp<AudioFlinger::Client>& client, 9025 const sp<IEffectClient>& effectClient, 9026 int32_t priority) 9027 : BnEffect(), 9028 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9029 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9030{ 9031 ALOGV("constructor %p", this); 9032 9033 if (client == 0) { 9034 return; 9035 } 9036 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9037 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9038 if (mCblkMemory != 0) { 9039 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9040 9041 if (mCblk != NULL) { 9042 new(mCblk) effect_param_cblk_t(); 9043 mBuffer = (uint8_t *)mCblk + bufOffset; 9044 } 9045 } else { 9046 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9047 sizeof(effect_param_cblk_t)); 9048 return; 9049 } 9050} 9051 9052AudioFlinger::EffectHandle::~EffectHandle() 9053{ 9054 ALOGV("Destructor %p", this); 9055 9056 if (mEffect == 0) { 9057 mDestroyed = true; 9058 return; 9059 } 9060 mEffect->lock(); 9061 mDestroyed = true; 9062 mEffect->unlock(); 9063 disconnect(false); 9064} 9065 9066status_t AudioFlinger::EffectHandle::enable() 9067{ 9068 ALOGV("enable %p", this); 9069 if (!mHasControl) return INVALID_OPERATION; 9070 if (mEffect == 0) return DEAD_OBJECT; 9071 9072 if (mEnabled) { 9073 return NO_ERROR; 9074 } 9075 9076 mEnabled = true; 9077 9078 sp<ThreadBase> thread = mEffect->thread().promote(); 9079 if (thread != 0) { 9080 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9081 } 9082 9083 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9084 if (mEffect->suspended()) { 9085 return NO_ERROR; 9086 } 9087 9088 status_t status = mEffect->setEnabled(true); 9089 if (status != NO_ERROR) { 9090 if (thread != 0) { 9091 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9092 } 9093 mEnabled = false; 9094 } 9095 return status; 9096} 9097 9098status_t AudioFlinger::EffectHandle::disable() 9099{ 9100 ALOGV("disable %p", this); 9101 if (!mHasControl) return INVALID_OPERATION; 9102 if (mEffect == 0) return DEAD_OBJECT; 9103 9104 if (!mEnabled) { 9105 return NO_ERROR; 9106 } 9107 mEnabled = false; 9108 9109 if (mEffect->suspended()) { 9110 return NO_ERROR; 9111 } 9112 9113 status_t status = mEffect->setEnabled(false); 9114 9115 sp<ThreadBase> thread = mEffect->thread().promote(); 9116 if (thread != 0) { 9117 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9118 } 9119 9120 return status; 9121} 9122 9123void AudioFlinger::EffectHandle::disconnect() 9124{ 9125 disconnect(true); 9126} 9127 9128void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9129{ 9130 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9131 if (mEffect == 0) { 9132 return; 9133 } 9134 // restore suspended effects if the disconnected handle was enabled and the last one. 9135 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9136 sp<ThreadBase> thread = mEffect->thread().promote(); 9137 if (thread != 0) { 9138 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9139 } 9140 } 9141 9142 // release sp on module => module destructor can be called now 9143 mEffect.clear(); 9144 if (mClient != 0) { 9145 if (mCblk != NULL) { 9146 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9147 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9148 } 9149 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9150 // Client destructor must run with AudioFlinger mutex locked 9151 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9152 mClient.clear(); 9153 } 9154} 9155 9156status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9157 uint32_t cmdSize, 9158 void *pCmdData, 9159 uint32_t *replySize, 9160 void *pReplyData) 9161{ 9162 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9163 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9164 9165 // only get parameter command is permitted for applications not controlling the effect 9166 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9167 return INVALID_OPERATION; 9168 } 9169 if (mEffect == 0) return DEAD_OBJECT; 9170 if (mClient == 0) return INVALID_OPERATION; 9171 9172 // handle commands that are not forwarded transparently to effect engine 9173 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9174 // No need to trylock() here as this function is executed in the binder thread serving a 9175 // particular client process: no risk to block the whole media server process or mixer 9176 // threads if we are stuck here 9177 Mutex::Autolock _l(mCblk->lock); 9178 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9179 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9180 mCblk->serverIndex = 0; 9181 mCblk->clientIndex = 0; 9182 return BAD_VALUE; 9183 } 9184 status_t status = NO_ERROR; 9185 while (mCblk->serverIndex < mCblk->clientIndex) { 9186 int reply; 9187 uint32_t rsize = sizeof(int); 9188 int *p = (int *)(mBuffer + mCblk->serverIndex); 9189 int size = *p++; 9190 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9191 ALOGW("command(): invalid parameter block size"); 9192 break; 9193 } 9194 effect_param_t *param = (effect_param_t *)p; 9195 if (param->psize == 0 || param->vsize == 0) { 9196 ALOGW("command(): null parameter or value size"); 9197 mCblk->serverIndex += size; 9198 continue; 9199 } 9200 uint32_t psize = sizeof(effect_param_t) + 9201 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9202 param->vsize; 9203 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9204 psize, 9205 p, 9206 &rsize, 9207 &reply); 9208 // stop at first error encountered 9209 if (ret != NO_ERROR) { 9210 status = ret; 9211 *(int *)pReplyData = reply; 9212 break; 9213 } else if (reply != NO_ERROR) { 9214 *(int *)pReplyData = reply; 9215 break; 9216 } 9217 mCblk->serverIndex += size; 9218 } 9219 mCblk->serverIndex = 0; 9220 mCblk->clientIndex = 0; 9221 return status; 9222 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9223 *(int *)pReplyData = NO_ERROR; 9224 return enable(); 9225 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9226 *(int *)pReplyData = NO_ERROR; 9227 return disable(); 9228 } 9229 9230 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9231} 9232 9233void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9234{ 9235 ALOGV("setControl %p control %d", this, hasControl); 9236 9237 mHasControl = hasControl; 9238 mEnabled = enabled; 9239 9240 if (signal && mEffectClient != 0) { 9241 mEffectClient->controlStatusChanged(hasControl); 9242 } 9243} 9244 9245void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9246 uint32_t cmdSize, 9247 void *pCmdData, 9248 uint32_t replySize, 9249 void *pReplyData) 9250{ 9251 if (mEffectClient != 0) { 9252 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9253 } 9254} 9255 9256 9257 9258void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9259{ 9260 if (mEffectClient != 0) { 9261 mEffectClient->enableStatusChanged(enabled); 9262 } 9263} 9264 9265status_t AudioFlinger::EffectHandle::onTransact( 9266 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9267{ 9268 return BnEffect::onTransact(code, data, reply, flags); 9269} 9270 9271 9272void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9273{ 9274 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9275 9276 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9277 (mClient == 0) ? getpid_cached : mClient->pid(), 9278 mPriority, 9279 mHasControl, 9280 !locked, 9281 mCblk ? mCblk->clientIndex : 0, 9282 mCblk ? mCblk->serverIndex : 0 9283 ); 9284 9285 if (locked) { 9286 mCblk->lock.unlock(); 9287 } 9288} 9289 9290#undef LOG_TAG 9291#define LOG_TAG "AudioFlinger::EffectChain" 9292 9293AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9294 int sessionId) 9295 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9296 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9297 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9298{ 9299 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9300 if (thread == NULL) { 9301 return; 9302 } 9303 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9304 thread->frameCount(); 9305} 9306 9307AudioFlinger::EffectChain::~EffectChain() 9308{ 9309 if (mOwnInBuffer) { 9310 delete mInBuffer; 9311 } 9312 9313} 9314 9315// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9316sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9317 effect_descriptor_t *descriptor) 9318{ 9319 size_t size = mEffects.size(); 9320 9321 for (size_t i = 0; i < size; i++) { 9322 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9323 return mEffects[i]; 9324 } 9325 } 9326 return 0; 9327} 9328 9329// getEffectFromId_l() must be called with ThreadBase::mLock held 9330sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9331{ 9332 size_t size = mEffects.size(); 9333 9334 for (size_t i = 0; i < size; i++) { 9335 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9336 if (id == 0 || mEffects[i]->id() == id) { 9337 return mEffects[i]; 9338 } 9339 } 9340 return 0; 9341} 9342 9343// getEffectFromType_l() must be called with ThreadBase::mLock held 9344sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9345 const effect_uuid_t *type) 9346{ 9347 size_t size = mEffects.size(); 9348 9349 for (size_t i = 0; i < size; i++) { 9350 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9351 return mEffects[i]; 9352 } 9353 } 9354 return 0; 9355} 9356 9357void AudioFlinger::EffectChain::clearInputBuffer() 9358{ 9359 Mutex::Autolock _l(mLock); 9360 sp<ThreadBase> thread = mThread.promote(); 9361 if (thread == 0) { 9362 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9363 return; 9364 } 9365 clearInputBuffer_l(thread); 9366} 9367 9368// Must be called with EffectChain::mLock locked 9369void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9370{ 9371 size_t numSamples = thread->frameCount() * thread->channelCount(); 9372 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9373 9374} 9375 9376// Must be called with EffectChain::mLock locked 9377void AudioFlinger::EffectChain::process_l() 9378{ 9379 sp<ThreadBase> thread = mThread.promote(); 9380 if (thread == 0) { 9381 ALOGW("process_l(): cannot promote mixer thread"); 9382 return; 9383 } 9384 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9385 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9386 // always process effects unless no more tracks are on the session and the effect tail 9387 // has been rendered 9388 bool doProcess = true; 9389 if (!isGlobalSession) { 9390 bool tracksOnSession = (trackCnt() != 0); 9391 9392 if (!tracksOnSession && mTailBufferCount == 0) { 9393 doProcess = false; 9394 } 9395 9396 if (activeTrackCnt() == 0) { 9397 // if no track is active and the effect tail has not been rendered, 9398 // the input buffer must be cleared here as the mixer process will not do it 9399 if (tracksOnSession || mTailBufferCount > 0) { 9400 clearInputBuffer_l(thread); 9401 if (mTailBufferCount > 0) { 9402 mTailBufferCount--; 9403 } 9404 } 9405 } 9406 } 9407 9408 size_t size = mEffects.size(); 9409 if (doProcess) { 9410 for (size_t i = 0; i < size; i++) { 9411 mEffects[i]->process(); 9412 } 9413 } 9414 for (size_t i = 0; i < size; i++) { 9415 mEffects[i]->updateState(); 9416 } 9417} 9418 9419// addEffect_l() must be called with PlaybackThread::mLock held 9420status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9421{ 9422 effect_descriptor_t desc = effect->desc(); 9423 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9424 9425 Mutex::Autolock _l(mLock); 9426 effect->setChain(this); 9427 sp<ThreadBase> thread = mThread.promote(); 9428 if (thread == 0) { 9429 return NO_INIT; 9430 } 9431 effect->setThread(thread); 9432 9433 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9434 // Auxiliary effects are inserted at the beginning of mEffects vector as 9435 // they are processed first and accumulated in chain input buffer 9436 mEffects.insertAt(effect, 0); 9437 9438 // the input buffer for auxiliary effect contains mono samples in 9439 // 32 bit format. This is to avoid saturation in AudoMixer 9440 // accumulation stage. Saturation is done in EffectModule::process() before 9441 // calling the process in effect engine 9442 size_t numSamples = thread->frameCount(); 9443 int32_t *buffer = new int32_t[numSamples]; 9444 memset(buffer, 0, numSamples * sizeof(int32_t)); 9445 effect->setInBuffer((int16_t *)buffer); 9446 // auxiliary effects output samples to chain input buffer for further processing 9447 // by insert effects 9448 effect->setOutBuffer(mInBuffer); 9449 } else { 9450 // Insert effects are inserted at the end of mEffects vector as they are processed 9451 // after track and auxiliary effects. 9452 // Insert effect order as a function of indicated preference: 9453 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9454 // another effect is present 9455 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9456 // last effect claiming first position 9457 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9458 // first effect claiming last position 9459 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9460 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9461 // already present 9462 9463 size_t size = mEffects.size(); 9464 size_t idx_insert = size; 9465 ssize_t idx_insert_first = -1; 9466 ssize_t idx_insert_last = -1; 9467 9468 for (size_t i = 0; i < size; i++) { 9469 effect_descriptor_t d = mEffects[i]->desc(); 9470 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9471 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9472 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9473 // check invalid effect chaining combinations 9474 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9475 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9476 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9477 desc.name, d.name); 9478 return INVALID_OPERATION; 9479 } 9480 // remember position of first insert effect and by default 9481 // select this as insert position for new effect 9482 if (idx_insert == size) { 9483 idx_insert = i; 9484 } 9485 // remember position of last insert effect claiming 9486 // first position 9487 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9488 idx_insert_first = i; 9489 } 9490 // remember position of first insert effect claiming 9491 // last position 9492 if (iPref == EFFECT_FLAG_INSERT_LAST && 9493 idx_insert_last == -1) { 9494 idx_insert_last = i; 9495 } 9496 } 9497 } 9498 9499 // modify idx_insert from first position if needed 9500 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9501 if (idx_insert_last != -1) { 9502 idx_insert = idx_insert_last; 9503 } else { 9504 idx_insert = size; 9505 } 9506 } else { 9507 if (idx_insert_first != -1) { 9508 idx_insert = idx_insert_first + 1; 9509 } 9510 } 9511 9512 // always read samples from chain input buffer 9513 effect->setInBuffer(mInBuffer); 9514 9515 // if last effect in the chain, output samples to chain 9516 // output buffer, otherwise to chain input buffer 9517 if (idx_insert == size) { 9518 if (idx_insert != 0) { 9519 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9520 mEffects[idx_insert-1]->configure(); 9521 } 9522 effect->setOutBuffer(mOutBuffer); 9523 } else { 9524 effect->setOutBuffer(mInBuffer); 9525 } 9526 mEffects.insertAt(effect, idx_insert); 9527 9528 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9529 idx_insert); 9530 } 9531 effect->configure(); 9532 return NO_ERROR; 9533} 9534 9535// removeEffect_l() must be called with PlaybackThread::mLock held 9536size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9537{ 9538 Mutex::Autolock _l(mLock); 9539 size_t size = mEffects.size(); 9540 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9541 9542 for (size_t i = 0; i < size; i++) { 9543 if (effect == mEffects[i]) { 9544 // calling stop here will remove pre-processing effect from the audio HAL. 9545 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9546 // the middle of a read from audio HAL 9547 if (mEffects[i]->state() == EffectModule::ACTIVE || 9548 mEffects[i]->state() == EffectModule::STOPPING) { 9549 mEffects[i]->stop(); 9550 } 9551 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9552 delete[] effect->inBuffer(); 9553 } else { 9554 if (i == size - 1 && i != 0) { 9555 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9556 mEffects[i - 1]->configure(); 9557 } 9558 } 9559 mEffects.removeAt(i); 9560 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9561 this, i); 9562 break; 9563 } 9564 } 9565 9566 return mEffects.size(); 9567} 9568 9569// setDevice_l() must be called with PlaybackThread::mLock held 9570void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9571{ 9572 size_t size = mEffects.size(); 9573 for (size_t i = 0; i < size; i++) { 9574 mEffects[i]->setDevice(device); 9575 } 9576} 9577 9578// setMode_l() must be called with PlaybackThread::mLock held 9579void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9580{ 9581 size_t size = mEffects.size(); 9582 for (size_t i = 0; i < size; i++) { 9583 mEffects[i]->setMode(mode); 9584 } 9585} 9586 9587// setAudioSource_l() must be called with PlaybackThread::mLock held 9588void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9589{ 9590 size_t size = mEffects.size(); 9591 for (size_t i = 0; i < size; i++) { 9592 mEffects[i]->setAudioSource(source); 9593 } 9594} 9595 9596// setVolume_l() must be called with PlaybackThread::mLock held 9597bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9598{ 9599 uint32_t newLeft = *left; 9600 uint32_t newRight = *right; 9601 bool hasControl = false; 9602 int ctrlIdx = -1; 9603 size_t size = mEffects.size(); 9604 9605 // first update volume controller 9606 for (size_t i = size; i > 0; i--) { 9607 if (mEffects[i - 1]->isProcessEnabled() && 9608 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9609 ctrlIdx = i - 1; 9610 hasControl = true; 9611 break; 9612 } 9613 } 9614 9615 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9616 if (hasControl) { 9617 *left = mNewLeftVolume; 9618 *right = mNewRightVolume; 9619 } 9620 return hasControl; 9621 } 9622 9623 mVolumeCtrlIdx = ctrlIdx; 9624 mLeftVolume = newLeft; 9625 mRightVolume = newRight; 9626 9627 // second get volume update from volume controller 9628 if (ctrlIdx >= 0) { 9629 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9630 mNewLeftVolume = newLeft; 9631 mNewRightVolume = newRight; 9632 } 9633 // then indicate volume to all other effects in chain. 9634 // Pass altered volume to effects before volume controller 9635 // and requested volume to effects after controller 9636 uint32_t lVol = newLeft; 9637 uint32_t rVol = newRight; 9638 9639 for (size_t i = 0; i < size; i++) { 9640 if ((int)i == ctrlIdx) continue; 9641 // this also works for ctrlIdx == -1 when there is no volume controller 9642 if ((int)i > ctrlIdx) { 9643 lVol = *left; 9644 rVol = *right; 9645 } 9646 mEffects[i]->setVolume(&lVol, &rVol, false); 9647 } 9648 *left = newLeft; 9649 *right = newRight; 9650 9651 return hasControl; 9652} 9653 9654void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9655{ 9656 const size_t SIZE = 256; 9657 char buffer[SIZE]; 9658 String8 result; 9659 9660 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9661 result.append(buffer); 9662 9663 bool locked = tryLock(mLock); 9664 // failed to lock - AudioFlinger is probably deadlocked 9665 if (!locked) { 9666 result.append("\tCould not lock mutex:\n"); 9667 } 9668 9669 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9670 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9671 mEffects.size(), 9672 (uint32_t)mInBuffer, 9673 (uint32_t)mOutBuffer, 9674 mActiveTrackCnt); 9675 result.append(buffer); 9676 write(fd, result.string(), result.size()); 9677 9678 for (size_t i = 0; i < mEffects.size(); ++i) { 9679 sp<EffectModule> effect = mEffects[i]; 9680 if (effect != 0) { 9681 effect->dump(fd, args); 9682 } 9683 } 9684 9685 if (locked) { 9686 mLock.unlock(); 9687 } 9688} 9689 9690// must be called with ThreadBase::mLock held 9691void AudioFlinger::EffectChain::setEffectSuspended_l( 9692 const effect_uuid_t *type, bool suspend) 9693{ 9694 sp<SuspendedEffectDesc> desc; 9695 // use effect type UUID timelow as key as there is no real risk of identical 9696 // timeLow fields among effect type UUIDs. 9697 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9698 if (suspend) { 9699 if (index >= 0) { 9700 desc = mSuspendedEffects.valueAt(index); 9701 } else { 9702 desc = new SuspendedEffectDesc(); 9703 desc->mType = *type; 9704 mSuspendedEffects.add(type->timeLow, desc); 9705 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9706 } 9707 if (desc->mRefCount++ == 0) { 9708 sp<EffectModule> effect = getEffectIfEnabled(type); 9709 if (effect != 0) { 9710 desc->mEffect = effect; 9711 effect->setSuspended(true); 9712 effect->setEnabled(false); 9713 } 9714 } 9715 } else { 9716 if (index < 0) { 9717 return; 9718 } 9719 desc = mSuspendedEffects.valueAt(index); 9720 if (desc->mRefCount <= 0) { 9721 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9722 desc->mRefCount = 1; 9723 } 9724 if (--desc->mRefCount == 0) { 9725 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9726 if (desc->mEffect != 0) { 9727 sp<EffectModule> effect = desc->mEffect.promote(); 9728 if (effect != 0) { 9729 effect->setSuspended(false); 9730 effect->lock(); 9731 EffectHandle *handle = effect->controlHandle_l(); 9732 if (handle != NULL && !handle->destroyed_l()) { 9733 effect->setEnabled_l(handle->enabled()); 9734 } 9735 effect->unlock(); 9736 } 9737 desc->mEffect.clear(); 9738 } 9739 mSuspendedEffects.removeItemsAt(index); 9740 } 9741 } 9742} 9743 9744// must be called with ThreadBase::mLock held 9745void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9746{ 9747 sp<SuspendedEffectDesc> desc; 9748 9749 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9750 if (suspend) { 9751 if (index >= 0) { 9752 desc = mSuspendedEffects.valueAt(index); 9753 } else { 9754 desc = new SuspendedEffectDesc(); 9755 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9756 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9757 } 9758 if (desc->mRefCount++ == 0) { 9759 Vector< sp<EffectModule> > effects; 9760 getSuspendEligibleEffects(effects); 9761 for (size_t i = 0; i < effects.size(); i++) { 9762 setEffectSuspended_l(&effects[i]->desc().type, true); 9763 } 9764 } 9765 } else { 9766 if (index < 0) { 9767 return; 9768 } 9769 desc = mSuspendedEffects.valueAt(index); 9770 if (desc->mRefCount <= 0) { 9771 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9772 desc->mRefCount = 1; 9773 } 9774 if (--desc->mRefCount == 0) { 9775 Vector<const effect_uuid_t *> types; 9776 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9777 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9778 continue; 9779 } 9780 types.add(&mSuspendedEffects.valueAt(i)->mType); 9781 } 9782 for (size_t i = 0; i < types.size(); i++) { 9783 setEffectSuspended_l(types[i], false); 9784 } 9785 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9786 mSuspendedEffects.keyAt(index)); 9787 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9788 } 9789 } 9790} 9791 9792 9793// The volume effect is used for automated tests only 9794#ifndef OPENSL_ES_H_ 9795static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9796 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9797const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9798#endif //OPENSL_ES_H_ 9799 9800bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9801{ 9802 // auxiliary effects and visualizer are never suspended on output mix 9803 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9804 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9805 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9806 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9807 return false; 9808 } 9809 return true; 9810} 9811 9812void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9813 Vector< sp<AudioFlinger::EffectModule> > &effects) 9814{ 9815 effects.clear(); 9816 for (size_t i = 0; i < mEffects.size(); i++) { 9817 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9818 effects.add(mEffects[i]); 9819 } 9820 } 9821} 9822 9823sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9824 const effect_uuid_t *type) 9825{ 9826 sp<EffectModule> effect = getEffectFromType_l(type); 9827 return effect != 0 && effect->isEnabled() ? effect : 0; 9828} 9829 9830void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9831 bool enabled) 9832{ 9833 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9834 if (enabled) { 9835 if (index < 0) { 9836 // if the effect is not suspend check if all effects are suspended 9837 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9838 if (index < 0) { 9839 return; 9840 } 9841 if (!isEffectEligibleForSuspend(effect->desc())) { 9842 return; 9843 } 9844 setEffectSuspended_l(&effect->desc().type, enabled); 9845 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9846 if (index < 0) { 9847 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9848 return; 9849 } 9850 } 9851 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9852 effect->desc().type.timeLow); 9853 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9854 // if effect is requested to suspended but was not yet enabled, supend it now. 9855 if (desc->mEffect == 0) { 9856 desc->mEffect = effect; 9857 effect->setEnabled(false); 9858 effect->setSuspended(true); 9859 } 9860 } else { 9861 if (index < 0) { 9862 return; 9863 } 9864 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9865 effect->desc().type.timeLow); 9866 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9867 desc->mEffect.clear(); 9868 effect->setSuspended(false); 9869 } 9870} 9871 9872#undef LOG_TAG 9873#define LOG_TAG "AudioFlinger" 9874 9875// ---------------------------------------------------------------------------- 9876 9877status_t AudioFlinger::onTransact( 9878 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9879{ 9880 return BnAudioFlinger::onTransact(code, data, reply, flags); 9881} 9882 9883}; // namespace android 9884