AudioFlinger.cpp revision a42ff007a17d63df22c60dd5e5fd811ee45ca1b3
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) { 427 mLock.unlock(); 428 } 429 } 430 return NO_ERROR; 431} 432 433sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 434{ 435 // If pid is already in the mClients wp<> map, then use that entry 436 // (for which promote() is always != 0), otherwise create a new entry and Client. 437 sp<Client> client = mClients.valueFor(pid).promote(); 438 if (client == 0) { 439 client = new Client(this, pid); 440 mClients.add(pid, client); 441 } 442 443 return client; 444} 445 446// IAudioFlinger interface 447 448 449sp<IAudioTrack> AudioFlinger::createTrack( 450 pid_t pid, 451 audio_stream_type_t streamType, 452 uint32_t sampleRate, 453 audio_format_t format, 454 audio_channel_mask_t channelMask, 455 size_t frameCount, 456 IAudioFlinger::track_flags_t *flags, 457 const sp<IMemory>& sharedBuffer, 458 audio_io_handle_t output, 459 pid_t tid, 460 int *sessionId, 461 status_t *status) 462{ 463 sp<PlaybackThread::Track> track; 464 sp<TrackHandle> trackHandle; 465 sp<Client> client; 466 status_t lStatus; 467 int lSessionId; 468 469 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 470 // but if someone uses binder directly they could bypass that and cause us to crash 471 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 472 ALOGE("createTrack() invalid stream type %d", streamType); 473 lStatus = BAD_VALUE; 474 goto Exit; 475 } 476 477 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 478 // and we don't yet support 8.24 or 32-bit PCM 479 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 480 ALOGE("createTrack() invalid format %d", format); 481 lStatus = BAD_VALUE; 482 goto Exit; 483 } 484 485 { 486 Mutex::Autolock _l(mLock); 487 PlaybackThread *thread = checkPlaybackThread_l(output); 488 PlaybackThread *effectThread = NULL; 489 if (thread == NULL) { 490 ALOGE("unknown output thread"); 491 lStatus = BAD_VALUE; 492 goto Exit; 493 } 494 495 client = registerPid_l(pid); 496 497 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 498 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 499 // check if an effect chain with the same session ID is present on another 500 // output thread and move it here. 501 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 502 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 503 if (mPlaybackThreads.keyAt(i) != output) { 504 uint32_t sessions = t->hasAudioSession(*sessionId); 505 if (sessions & PlaybackThread::EFFECT_SESSION) { 506 effectThread = t.get(); 507 break; 508 } 509 } 510 } 511 lSessionId = *sessionId; 512 } else { 513 // if no audio session id is provided, create one here 514 lSessionId = nextUniqueId(); 515 if (sessionId != NULL) { 516 *sessionId = lSessionId; 517 } 518 } 519 ALOGV("createTrack() lSessionId: %d", lSessionId); 520 521 track = thread->createTrack_l(client, streamType, sampleRate, format, 522 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 523 524 // move effect chain to this output thread if an effect on same session was waiting 525 // for a track to be created 526 if (lStatus == NO_ERROR && effectThread != NULL) { 527 Mutex::Autolock _dl(thread->mLock); 528 Mutex::Autolock _sl(effectThread->mLock); 529 moveEffectChain_l(lSessionId, effectThread, thread, true); 530 } 531 532 // Look for sync events awaiting for a session to be used. 533 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 534 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 535 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 536 if (lStatus == NO_ERROR) { 537 (void) track->setSyncEvent(mPendingSyncEvents[i]); 538 } else { 539 mPendingSyncEvents[i]->cancel(); 540 } 541 mPendingSyncEvents.removeAt(i); 542 i--; 543 } 544 } 545 } 546 } 547 if (lStatus == NO_ERROR) { 548 trackHandle = new TrackHandle(track); 549 } else { 550 // remove local strong reference to Client before deleting the Track so that the Client 551 // destructor is called by the TrackBase destructor with mLock held 552 client.clear(); 553 track.clear(); 554 } 555 556Exit: 557 if (status != NULL) { 558 *status = lStatus; 559 } 560 return trackHandle; 561} 562 563uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("sampleRate() unknown thread %d", output); 569 return 0; 570 } 571 return thread->sampleRate(); 572} 573 574int AudioFlinger::channelCount(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("channelCount() unknown thread %d", output); 580 return 0; 581 } 582 return thread->channelCount(); 583} 584 585audio_format_t AudioFlinger::format(audio_io_handle_t output) const 586{ 587 Mutex::Autolock _l(mLock); 588 PlaybackThread *thread = checkPlaybackThread_l(output); 589 if (thread == NULL) { 590 ALOGW("format() unknown thread %d", output); 591 return AUDIO_FORMAT_INVALID; 592 } 593 return thread->format(); 594} 595 596size_t AudioFlinger::frameCount(audio_io_handle_t output) const 597{ 598 Mutex::Autolock _l(mLock); 599 PlaybackThread *thread = checkPlaybackThread_l(output); 600 if (thread == NULL) { 601 ALOGW("frameCount() unknown thread %d", output); 602 return 0; 603 } 604 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 605 // should examine all callers and fix them to handle smaller counts 606 return thread->frameCount(); 607} 608 609uint32_t AudioFlinger::latency(audio_io_handle_t output) const 610{ 611 Mutex::Autolock _l(mLock); 612 PlaybackThread *thread = checkPlaybackThread_l(output); 613 if (thread == NULL) { 614 ALOGW("latency() unknown thread %d", output); 615 return 0; 616 } 617 return thread->latency(); 618} 619 620status_t AudioFlinger::setMasterVolume(float value) 621{ 622 status_t ret = initCheck(); 623 if (ret != NO_ERROR) { 624 return ret; 625 } 626 627 // check calling permissions 628 if (!settingsAllowed()) { 629 return PERMISSION_DENIED; 630 } 631 632 Mutex::Autolock _l(mLock); 633 mMasterVolume = value; 634 635 // Set master volume in the HALs which support it. 636 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 637 AutoMutex lock(mHardwareLock); 638 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 639 640 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 641 if (dev->canSetMasterVolume()) { 642 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 643 } 644 mHardwareStatus = AUDIO_HW_IDLE; 645 } 646 647 // Now set the master volume in each playback thread. Playback threads 648 // assigned to HALs which do not have master volume support will apply 649 // master volume during the mix operation. Threads with HALs which do 650 // support master volume will simply ignore the setting. 651 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 652 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 653 654 return NO_ERROR; 655} 656 657status_t AudioFlinger::setMode(audio_mode_t mode) 658{ 659 status_t ret = initCheck(); 660 if (ret != NO_ERROR) { 661 return ret; 662 } 663 664 // check calling permissions 665 if (!settingsAllowed()) { 666 return PERMISSION_DENIED; 667 } 668 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 669 ALOGW("Illegal value: setMode(%d)", mode); 670 return BAD_VALUE; 671 } 672 673 { // scope for the lock 674 AutoMutex lock(mHardwareLock); 675 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 676 mHardwareStatus = AUDIO_HW_SET_MODE; 677 ret = dev->set_mode(dev, mode); 678 mHardwareStatus = AUDIO_HW_IDLE; 679 } 680 681 if (NO_ERROR == ret) { 682 Mutex::Autolock _l(mLock); 683 mMode = mode; 684 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 685 mPlaybackThreads.valueAt(i)->setMode(mode); 686 } 687 688 return ret; 689} 690 691status_t AudioFlinger::setMicMute(bool state) 692{ 693 status_t ret = initCheck(); 694 if (ret != NO_ERROR) { 695 return ret; 696 } 697 698 // check calling permissions 699 if (!settingsAllowed()) { 700 return PERMISSION_DENIED; 701 } 702 703 AutoMutex lock(mHardwareLock); 704 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 705 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 706 ret = dev->set_mic_mute(dev, state); 707 mHardwareStatus = AUDIO_HW_IDLE; 708 return ret; 709} 710 711bool AudioFlinger::getMicMute() const 712{ 713 status_t ret = initCheck(); 714 if (ret != NO_ERROR) { 715 return false; 716 } 717 718 bool state = AUDIO_MODE_INVALID; 719 AutoMutex lock(mHardwareLock); 720 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 721 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 722 dev->get_mic_mute(dev, &state); 723 mHardwareStatus = AUDIO_HW_IDLE; 724 return state; 725} 726 727status_t AudioFlinger::setMasterMute(bool muted) 728{ 729 status_t ret = initCheck(); 730 if (ret != NO_ERROR) { 731 return ret; 732 } 733 734 // check calling permissions 735 if (!settingsAllowed()) { 736 return PERMISSION_DENIED; 737 } 738 739 Mutex::Autolock _l(mLock); 740 mMasterMute = muted; 741 742 // Set master mute in the HALs which support it. 743 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 744 AutoMutex lock(mHardwareLock); 745 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 746 747 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 748 if (dev->canSetMasterMute()) { 749 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 750 } 751 mHardwareStatus = AUDIO_HW_IDLE; 752 } 753 754 // Now set the master mute in each playback thread. Playback threads 755 // assigned to HALs which do not have master mute support will apply master 756 // mute during the mix operation. Threads with HALs which do support master 757 // mute will simply ignore the setting. 758 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 759 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 760 761 return NO_ERROR; 762} 763 764float AudioFlinger::masterVolume() const 765{ 766 Mutex::Autolock _l(mLock); 767 return masterVolume_l(); 768} 769 770bool AudioFlinger::masterMute() const 771{ 772 Mutex::Autolock _l(mLock); 773 return masterMute_l(); 774} 775 776float AudioFlinger::masterVolume_l() const 777{ 778 return mMasterVolume; 779} 780 781bool AudioFlinger::masterMute_l() const 782{ 783 return mMasterMute; 784} 785 786status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 787 audio_io_handle_t output) 788{ 789 // check calling permissions 790 if (!settingsAllowed()) { 791 return PERMISSION_DENIED; 792 } 793 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 795 ALOGE("setStreamVolume() invalid stream %d", stream); 796 return BAD_VALUE; 797 } 798 799 AutoMutex lock(mLock); 800 PlaybackThread *thread = NULL; 801 if (output) { 802 thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return BAD_VALUE; 805 } 806 } 807 808 mStreamTypes[stream].volume = value; 809 810 if (thread == NULL) { 811 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 812 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 813 } 814 } else { 815 thread->setStreamVolume(stream, value); 816 } 817 818 return NO_ERROR; 819} 820 821status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 822{ 823 // check calling permissions 824 if (!settingsAllowed()) { 825 return PERMISSION_DENIED; 826 } 827 828 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 829 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 830 ALOGE("setStreamMute() invalid stream %d", stream); 831 return BAD_VALUE; 832 } 833 834 AutoMutex lock(mLock); 835 mStreamTypes[stream].mute = muted; 836 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 837 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 838 839 return NO_ERROR; 840} 841 842float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 843{ 844 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 845 return 0.0f; 846 } 847 848 AutoMutex lock(mLock); 849 float volume; 850 if (output) { 851 PlaybackThread *thread = checkPlaybackThread_l(output); 852 if (thread == NULL) { 853 return 0.0f; 854 } 855 volume = thread->streamVolume(stream); 856 } else { 857 volume = streamVolume_l(stream); 858 } 859 860 return volume; 861} 862 863bool AudioFlinger::streamMute(audio_stream_type_t stream) const 864{ 865 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 866 return true; 867 } 868 869 AutoMutex lock(mLock); 870 return streamMute_l(stream); 871} 872 873status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 874{ 875 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 876 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 877 // check calling permissions 878 if (!settingsAllowed()) { 879 return PERMISSION_DENIED; 880 } 881 882 // ioHandle == 0 means the parameters are global to the audio hardware interface 883 if (ioHandle == 0) { 884 Mutex::Autolock _l(mLock); 885 status_t final_result = NO_ERROR; 886 { 887 AutoMutex lock(mHardwareLock); 888 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 889 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 890 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 891 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 892 final_result = result ?: final_result; 893 } 894 mHardwareStatus = AUDIO_HW_IDLE; 895 } 896 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 897 AudioParameter param = AudioParameter(keyValuePairs); 898 String8 value; 899 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 900 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 901 if (mBtNrecIsOff != btNrecIsOff) { 902 for (size_t i = 0; i < mRecordThreads.size(); i++) { 903 sp<RecordThread> thread = mRecordThreads.valueAt(i); 904 audio_devices_t device = thread->inDevice(); 905 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 906 // collect all of the thread's session IDs 907 KeyedVector<int, bool> ids = thread->sessionIds(); 908 // suspend effects associated with those session IDs 909 for (size_t j = 0; j < ids.size(); ++j) { 910 int sessionId = ids.keyAt(j); 911 thread->setEffectSuspended(FX_IID_AEC, 912 suspend, 913 sessionId); 914 thread->setEffectSuspended(FX_IID_NS, 915 suspend, 916 sessionId); 917 } 918 } 919 mBtNrecIsOff = btNrecIsOff; 920 } 921 } 922 String8 screenState; 923 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 924 bool isOff = screenState == "off"; 925 if (isOff != (gScreenState & 1)) { 926 gScreenState = ((gScreenState & ~1) + 2) | isOff; 927 } 928 } 929 return final_result; 930 } 931 932 // hold a strong ref on thread in case closeOutput() or closeInput() is called 933 // and the thread is exited once the lock is released 934 sp<ThreadBase> thread; 935 { 936 Mutex::Autolock _l(mLock); 937 thread = checkPlaybackThread_l(ioHandle); 938 if (thread == 0) { 939 thread = checkRecordThread_l(ioHandle); 940 } else if (thread == primaryPlaybackThread_l()) { 941 // indicate output device change to all input threads for pre processing 942 AudioParameter param = AudioParameter(keyValuePairs); 943 int value; 944 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 945 (value != 0)) { 946 for (size_t i = 0; i < mRecordThreads.size(); i++) { 947 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 948 } 949 } 950 } 951 } 952 if (thread != 0) { 953 return thread->setParameters(keyValuePairs); 954 } 955 return BAD_VALUE; 956} 957 958String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 959{ 960 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 961 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 962 963 Mutex::Autolock _l(mLock); 964 965 if (ioHandle == 0) { 966 String8 out_s8; 967 968 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 969 char *s; 970 { 971 AutoMutex lock(mHardwareLock); 972 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 973 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 974 s = dev->get_parameters(dev, keys.string()); 975 mHardwareStatus = AUDIO_HW_IDLE; 976 } 977 out_s8 += String8(s ? s : ""); 978 free(s); 979 } 980 return out_s8; 981 } 982 983 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 984 if (playbackThread != NULL) { 985 return playbackThread->getParameters(keys); 986 } 987 RecordThread *recordThread = checkRecordThread_l(ioHandle); 988 if (recordThread != NULL) { 989 return recordThread->getParameters(keys); 990 } 991 return String8(""); 992} 993 994size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 995 audio_channel_mask_t channelMask) const 996{ 997 status_t ret = initCheck(); 998 if (ret != NO_ERROR) { 999 return 0; 1000 } 1001 1002 AutoMutex lock(mHardwareLock); 1003 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1004 struct audio_config config = { 1005 sample_rate: sampleRate, 1006 channel_mask: channelMask, 1007 format: format, 1008 }; 1009 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1010 size_t size = dev->get_input_buffer_size(dev, &config); 1011 mHardwareStatus = AUDIO_HW_IDLE; 1012 return size; 1013} 1014 1015unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1016{ 1017 Mutex::Autolock _l(mLock); 1018 1019 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1020 if (recordThread != NULL) { 1021 return recordThread->getInputFramesLost(); 1022 } 1023 return 0; 1024} 1025 1026status_t AudioFlinger::setVoiceVolume(float value) 1027{ 1028 status_t ret = initCheck(); 1029 if (ret != NO_ERROR) { 1030 return ret; 1031 } 1032 1033 // check calling permissions 1034 if (!settingsAllowed()) { 1035 return PERMISSION_DENIED; 1036 } 1037 1038 AutoMutex lock(mHardwareLock); 1039 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1040 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1041 ret = dev->set_voice_volume(dev, value); 1042 mHardwareStatus = AUDIO_HW_IDLE; 1043 1044 return ret; 1045} 1046 1047status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1048 audio_io_handle_t output) const 1049{ 1050 status_t status; 1051 1052 Mutex::Autolock _l(mLock); 1053 1054 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1055 if (playbackThread != NULL) { 1056 return playbackThread->getRenderPosition(halFrames, dspFrames); 1057 } 1058 1059 return BAD_VALUE; 1060} 1061 1062void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1063{ 1064 1065 Mutex::Autolock _l(mLock); 1066 1067 pid_t pid = IPCThreadState::self()->getCallingPid(); 1068 if (mNotificationClients.indexOfKey(pid) < 0) { 1069 sp<NotificationClient> notificationClient = new NotificationClient(this, 1070 client, 1071 pid); 1072 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1073 1074 mNotificationClients.add(pid, notificationClient); 1075 1076 sp<IBinder> binder = client->asBinder(); 1077 binder->linkToDeath(notificationClient); 1078 1079 // the config change is always sent from playback or record threads to avoid deadlock 1080 // with AudioSystem::gLock 1081 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1082 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1083 } 1084 1085 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1086 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1087 } 1088 } 1089} 1090 1091void AudioFlinger::removeNotificationClient(pid_t pid) 1092{ 1093 Mutex::Autolock _l(mLock); 1094 1095 mNotificationClients.removeItem(pid); 1096 1097 ALOGV("%d died, releasing its sessions", pid); 1098 size_t num = mAudioSessionRefs.size(); 1099 bool removed = false; 1100 for (size_t i = 0; i< num; ) { 1101 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1102 ALOGV(" pid %d @ %d", ref->mPid, i); 1103 if (ref->mPid == pid) { 1104 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1105 mAudioSessionRefs.removeAt(i); 1106 delete ref; 1107 removed = true; 1108 num--; 1109 } else { 1110 i++; 1111 } 1112 } 1113 if (removed) { 1114 purgeStaleEffects_l(); 1115 } 1116} 1117 1118// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1119void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1120{ 1121 size_t size = mNotificationClients.size(); 1122 for (size_t i = 0; i < size; i++) { 1123 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1124 param2); 1125 } 1126} 1127 1128// removeClient_l() must be called with AudioFlinger::mLock held 1129void AudioFlinger::removeClient_l(pid_t pid) 1130{ 1131 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1132 IPCThreadState::self()->getCallingPid()); 1133 mClients.removeItem(pid); 1134} 1135 1136// getEffectThread_l() must be called with AudioFlinger::mLock held 1137sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1138{ 1139 sp<PlaybackThread> thread; 1140 1141 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1142 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1143 ALOG_ASSERT(thread == 0); 1144 thread = mPlaybackThreads.valueAt(i); 1145 } 1146 } 1147 1148 return thread; 1149} 1150 1151// ---------------------------------------------------------------------------- 1152 1153AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1154 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1155 : Thread(false /*canCallJava*/), 1156 mType(type), 1157 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1158 // mChannelMask 1159 mChannelCount(0), 1160 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1161 mParamStatus(NO_ERROR), 1162 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1163 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1164 // mName will be set by concrete (non-virtual) subclass 1165 mDeathRecipient(new PMDeathRecipient(this)) 1166{ 1167} 1168 1169AudioFlinger::ThreadBase::~ThreadBase() 1170{ 1171 mParamCond.broadcast(); 1172 // do not lock the mutex in destructor 1173 releaseWakeLock_l(); 1174 if (mPowerManager != 0) { 1175 sp<IBinder> binder = mPowerManager->asBinder(); 1176 binder->unlinkToDeath(mDeathRecipient); 1177 } 1178} 1179 1180void AudioFlinger::ThreadBase::exit() 1181{ 1182 ALOGV("ThreadBase::exit"); 1183 // do any cleanup required for exit to succeed 1184 preExit(); 1185 { 1186 // This lock prevents the following race in thread (uniprocessor for illustration): 1187 // if (!exitPending()) { 1188 // // context switch from here to exit() 1189 // // exit() calls requestExit(), what exitPending() observes 1190 // // exit() calls signal(), which is dropped since no waiters 1191 // // context switch back from exit() to here 1192 // mWaitWorkCV.wait(...); 1193 // // now thread is hung 1194 // } 1195 AutoMutex lock(mLock); 1196 requestExit(); 1197 mWaitWorkCV.broadcast(); 1198 } 1199 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1200 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1201 requestExitAndWait(); 1202} 1203 1204status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1205{ 1206 status_t status; 1207 1208 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1209 Mutex::Autolock _l(mLock); 1210 1211 mNewParameters.add(keyValuePairs); 1212 mWaitWorkCV.signal(); 1213 // wait condition with timeout in case the thread loop has exited 1214 // before the request could be processed 1215 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1216 status = mParamStatus; 1217 mWaitWorkCV.signal(); 1218 } else { 1219 status = TIMED_OUT; 1220 } 1221 return status; 1222} 1223 1224void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 1225{ 1226 Mutex::Autolock _l(mLock); 1227 sendIoConfigEvent_l(event, param); 1228} 1229 1230// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 1231void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 1232{ 1233 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 1234 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 1235 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 1236 param); 1237 mWaitWorkCV.signal(); 1238} 1239 1240// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 1241void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 1242{ 1243 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 1244 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 1245 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 1246 mConfigEvents.size(), pid, tid, prio); 1247 mWaitWorkCV.signal(); 1248} 1249 1250void AudioFlinger::ThreadBase::processConfigEvents() 1251{ 1252 mLock.lock(); 1253 while (!mConfigEvents.isEmpty()) { 1254 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1255 ConfigEvent *event = mConfigEvents[0]; 1256 mConfigEvents.removeAt(0); 1257 // release mLock before locking AudioFlinger mLock: lock order is always 1258 // AudioFlinger then ThreadBase to avoid cross deadlock 1259 mLock.unlock(); 1260 switch(event->type()) { 1261 case CFG_EVENT_PRIO: { 1262 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 1263 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 1264 if (err != 0) { 1265 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 1266 "error %d", 1267 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 1268 } 1269 } break; 1270 case CFG_EVENT_IO: { 1271 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 1272 mAudioFlinger->mLock.lock(); 1273 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 1274 mAudioFlinger->mLock.unlock(); 1275 } break; 1276 default: 1277 ALOGE("processConfigEvents() unknown event type %d", event->type()); 1278 break; 1279 } 1280 delete event; 1281 mLock.lock(); 1282 } 1283 mLock.unlock(); 1284} 1285 1286void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1287{ 1288 const size_t SIZE = 256; 1289 char buffer[SIZE]; 1290 String8 result; 1291 1292 bool locked = tryLock(mLock); 1293 if (!locked) { 1294 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1295 write(fd, buffer, strlen(buffer)); 1296 } 1297 1298 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1299 result.append(buffer); 1300 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1301 result.append(buffer); 1302 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1303 result.append(buffer); 1304 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 1305 result.append(buffer); 1306 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1307 result.append(buffer); 1308 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1309 result.append(buffer); 1310 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1311 result.append(buffer); 1312 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1313 result.append(buffer); 1314 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1315 result.append(buffer); 1316 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1317 result.append(buffer); 1318 1319 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1320 result.append(buffer); 1321 result.append(" Index Command"); 1322 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1323 snprintf(buffer, SIZE, "\n %02d ", i); 1324 result.append(buffer); 1325 result.append(mNewParameters[i]); 1326 } 1327 1328 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1329 result.append(buffer); 1330 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1331 mConfigEvents[i]->dump(buffer, SIZE); 1332 result.append(buffer); 1333 } 1334 result.append("\n"); 1335 1336 write(fd, result.string(), result.size()); 1337 1338 if (locked) { 1339 mLock.unlock(); 1340 } 1341} 1342 1343void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1344{ 1345 const size_t SIZE = 256; 1346 char buffer[SIZE]; 1347 String8 result; 1348 1349 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1350 write(fd, buffer, strlen(buffer)); 1351 1352 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1353 sp<EffectChain> chain = mEffectChains[i]; 1354 if (chain != 0) { 1355 chain->dump(fd, args); 1356 } 1357 } 1358} 1359 1360void AudioFlinger::ThreadBase::acquireWakeLock() 1361{ 1362 Mutex::Autolock _l(mLock); 1363 acquireWakeLock_l(); 1364} 1365 1366void AudioFlinger::ThreadBase::acquireWakeLock_l() 1367{ 1368 if (mPowerManager == 0) { 1369 // use checkService() to avoid blocking if power service is not up yet 1370 sp<IBinder> binder = 1371 defaultServiceManager()->checkService(String16("power")); 1372 if (binder == 0) { 1373 ALOGW("Thread %s cannot connect to the power manager service", mName); 1374 } else { 1375 mPowerManager = interface_cast<IPowerManager>(binder); 1376 binder->linkToDeath(mDeathRecipient); 1377 } 1378 } 1379 if (mPowerManager != 0) { 1380 sp<IBinder> binder = new BBinder(); 1381 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1382 binder, 1383 String16(mName)); 1384 if (status == NO_ERROR) { 1385 mWakeLockToken = binder; 1386 } 1387 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1388 } 1389} 1390 1391void AudioFlinger::ThreadBase::releaseWakeLock() 1392{ 1393 Mutex::Autolock _l(mLock); 1394 releaseWakeLock_l(); 1395} 1396 1397void AudioFlinger::ThreadBase::releaseWakeLock_l() 1398{ 1399 if (mWakeLockToken != 0) { 1400 ALOGV("releaseWakeLock_l() %s", mName); 1401 if (mPowerManager != 0) { 1402 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1403 } 1404 mWakeLockToken.clear(); 1405 } 1406} 1407 1408void AudioFlinger::ThreadBase::clearPowerManager() 1409{ 1410 Mutex::Autolock _l(mLock); 1411 releaseWakeLock_l(); 1412 mPowerManager.clear(); 1413} 1414 1415void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1416{ 1417 sp<ThreadBase> thread = mThread.promote(); 1418 if (thread != 0) { 1419 thread->clearPowerManager(); 1420 } 1421 ALOGW("power manager service died !!!"); 1422} 1423 1424void AudioFlinger::ThreadBase::setEffectSuspended( 1425 const effect_uuid_t *type, bool suspend, int sessionId) 1426{ 1427 Mutex::Autolock _l(mLock); 1428 setEffectSuspended_l(type, suspend, sessionId); 1429} 1430 1431void AudioFlinger::ThreadBase::setEffectSuspended_l( 1432 const effect_uuid_t *type, bool suspend, int sessionId) 1433{ 1434 sp<EffectChain> chain = getEffectChain_l(sessionId); 1435 if (chain != 0) { 1436 if (type != NULL) { 1437 chain->setEffectSuspended_l(type, suspend); 1438 } else { 1439 chain->setEffectSuspendedAll_l(suspend); 1440 } 1441 } 1442 1443 updateSuspendedSessions_l(type, suspend, sessionId); 1444} 1445 1446void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1447{ 1448 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1449 if (index < 0) { 1450 return; 1451 } 1452 1453 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1454 mSuspendedSessions.valueAt(index); 1455 1456 for (size_t i = 0; i < sessionEffects.size(); i++) { 1457 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1458 for (int j = 0; j < desc->mRefCount; j++) { 1459 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1460 chain->setEffectSuspendedAll_l(true); 1461 } else { 1462 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1463 desc->mType.timeLow); 1464 chain->setEffectSuspended_l(&desc->mType, true); 1465 } 1466 } 1467 } 1468} 1469 1470void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1471 bool suspend, 1472 int sessionId) 1473{ 1474 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1475 1476 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1477 1478 if (suspend) { 1479 if (index >= 0) { 1480 sessionEffects = mSuspendedSessions.valueAt(index); 1481 } else { 1482 mSuspendedSessions.add(sessionId, sessionEffects); 1483 } 1484 } else { 1485 if (index < 0) { 1486 return; 1487 } 1488 sessionEffects = mSuspendedSessions.valueAt(index); 1489 } 1490 1491 1492 int key = EffectChain::kKeyForSuspendAll; 1493 if (type != NULL) { 1494 key = type->timeLow; 1495 } 1496 index = sessionEffects.indexOfKey(key); 1497 1498 sp<SuspendedSessionDesc> desc; 1499 if (suspend) { 1500 if (index >= 0) { 1501 desc = sessionEffects.valueAt(index); 1502 } else { 1503 desc = new SuspendedSessionDesc(); 1504 if (type != NULL) { 1505 desc->mType = *type; 1506 } 1507 sessionEffects.add(key, desc); 1508 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1509 } 1510 desc->mRefCount++; 1511 } else { 1512 if (index < 0) { 1513 return; 1514 } 1515 desc = sessionEffects.valueAt(index); 1516 if (--desc->mRefCount == 0) { 1517 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1518 sessionEffects.removeItemsAt(index); 1519 if (sessionEffects.isEmpty()) { 1520 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1521 sessionId); 1522 mSuspendedSessions.removeItem(sessionId); 1523 } 1524 } 1525 } 1526 if (!sessionEffects.isEmpty()) { 1527 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1528 } 1529} 1530 1531void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1532 bool enabled, 1533 int sessionId) 1534{ 1535 Mutex::Autolock _l(mLock); 1536 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1537} 1538 1539void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1540 bool enabled, 1541 int sessionId) 1542{ 1543 if (mType != RECORD) { 1544 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1545 // another session. This gives the priority to well behaved effect control panels 1546 // and applications not using global effects. 1547 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1548 // global effects 1549 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1550 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1551 } 1552 } 1553 1554 sp<EffectChain> chain = getEffectChain_l(sessionId); 1555 if (chain != 0) { 1556 chain->checkSuspendOnEffectEnabled(effect, enabled); 1557 } 1558} 1559 1560// ---------------------------------------------------------------------------- 1561 1562AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1563 AudioStreamOut* output, 1564 audio_io_handle_t id, 1565 audio_devices_t device, 1566 type_t type) 1567 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1568 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1569 // mStreamTypes[] initialized in constructor body 1570 mOutput(output), 1571 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1572 mMixerStatus(MIXER_IDLE), 1573 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1574 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1575 mScreenState(gScreenState), 1576 // index 0 is reserved for normal mixer's submix 1577 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1578{ 1579 snprintf(mName, kNameLength, "AudioOut_%X", id); 1580 1581 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1582 // it would be safer to explicitly pass initial masterVolume/masterMute as 1583 // parameter. 1584 // 1585 // If the HAL we are using has support for master volume or master mute, 1586 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1587 // and the mute set to false). 1588 mMasterVolume = audioFlinger->masterVolume_l(); 1589 mMasterMute = audioFlinger->masterMute_l(); 1590 if (mOutput && mOutput->audioHwDev) { 1591 if (mOutput->audioHwDev->canSetMasterVolume()) { 1592 mMasterVolume = 1.0; 1593 } 1594 1595 if (mOutput->audioHwDev->canSetMasterMute()) { 1596 mMasterMute = false; 1597 } 1598 } 1599 1600 readOutputParameters(); 1601 1602 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1603 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1604 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1605 stream = (audio_stream_type_t) (stream + 1)) { 1606 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1607 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1608 } 1609 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1610 // because mAudioFlinger doesn't have one to copy from 1611} 1612 1613AudioFlinger::PlaybackThread::~PlaybackThread() 1614{ 1615 delete [] mMixBuffer; 1616} 1617 1618void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1619{ 1620 dumpInternals(fd, args); 1621 dumpTracks(fd, args); 1622 dumpEffectChains(fd, args); 1623} 1624 1625void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1626{ 1627 const size_t SIZE = 256; 1628 char buffer[SIZE]; 1629 String8 result; 1630 1631 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1632 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1633 const stream_type_t *st = &mStreamTypes[i]; 1634 if (i > 0) { 1635 result.appendFormat(", "); 1636 } 1637 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1638 if (st->mute) { 1639 result.append("M"); 1640 } 1641 } 1642 result.append("\n"); 1643 write(fd, result.string(), result.length()); 1644 result.clear(); 1645 1646 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1647 result.append(buffer); 1648 Track::appendDumpHeader(result); 1649 for (size_t i = 0; i < mTracks.size(); ++i) { 1650 sp<Track> track = mTracks[i]; 1651 if (track != 0) { 1652 track->dump(buffer, SIZE); 1653 result.append(buffer); 1654 } 1655 } 1656 1657 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1658 result.append(buffer); 1659 Track::appendDumpHeader(result); 1660 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1661 sp<Track> track = mActiveTracks[i].promote(); 1662 if (track != 0) { 1663 track->dump(buffer, SIZE); 1664 result.append(buffer); 1665 } 1666 } 1667 write(fd, result.string(), result.size()); 1668 1669 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1670 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1671 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1672 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1673} 1674 1675void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1676{ 1677 const size_t SIZE = 256; 1678 char buffer[SIZE]; 1679 String8 result; 1680 1681 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1682 result.append(buffer); 1683 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1684 ns2ms(systemTime() - mLastWriteTime)); 1685 result.append(buffer); 1686 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1687 result.append(buffer); 1688 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1689 result.append(buffer); 1690 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1691 result.append(buffer); 1692 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1693 result.append(buffer); 1694 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1695 result.append(buffer); 1696 write(fd, result.string(), result.size()); 1697 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1698 1699 dumpBase(fd, args); 1700} 1701 1702// Thread virtuals 1703status_t AudioFlinger::PlaybackThread::readyToRun() 1704{ 1705 status_t status = initCheck(); 1706 if (status == NO_ERROR) { 1707 ALOGI("AudioFlinger's thread %p ready to run", this); 1708 } else { 1709 ALOGE("No working audio driver found."); 1710 } 1711 return status; 1712} 1713 1714void AudioFlinger::PlaybackThread::onFirstRef() 1715{ 1716 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1717} 1718 1719// ThreadBase virtuals 1720void AudioFlinger::PlaybackThread::preExit() 1721{ 1722 ALOGV(" preExit()"); 1723 // FIXME this is using hard-coded strings but in the future, this functionality will be 1724 // converted to use audio HAL extensions required to support tunneling 1725 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1726} 1727 1728// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1729sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1730 const sp<AudioFlinger::Client>& client, 1731 audio_stream_type_t streamType, 1732 uint32_t sampleRate, 1733 audio_format_t format, 1734 audio_channel_mask_t channelMask, 1735 size_t frameCount, 1736 const sp<IMemory>& sharedBuffer, 1737 int sessionId, 1738 IAudioFlinger::track_flags_t *flags, 1739 pid_t tid, 1740 status_t *status) 1741{ 1742 sp<Track> track; 1743 status_t lStatus; 1744 1745 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1746 1747 // client expresses a preference for FAST, but we get the final say 1748 if (*flags & IAudioFlinger::TRACK_FAST) { 1749 if ( 1750 // not timed 1751 (!isTimed) && 1752 // either of these use cases: 1753 ( 1754 // use case 1: shared buffer with any frame count 1755 ( 1756 (sharedBuffer != 0) 1757 ) || 1758 // use case 2: callback handler and frame count is default or at least as large as HAL 1759 ( 1760 (tid != -1) && 1761 ((frameCount == 0) || 1762 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1763 ) 1764 ) && 1765 // PCM data 1766 audio_is_linear_pcm(format) && 1767 // mono or stereo 1768 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1769 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1770#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1771 // hardware sample rate 1772 (sampleRate == mSampleRate) && 1773#endif 1774 // normal mixer has an associated fast mixer 1775 hasFastMixer() && 1776 // there are sufficient fast track slots available 1777 (mFastTrackAvailMask != 0) 1778 // FIXME test that MixerThread for this fast track has a capable output HAL 1779 // FIXME add a permission test also? 1780 ) { 1781 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1782 if (frameCount == 0) { 1783 frameCount = mFrameCount * kFastTrackMultiplier; 1784 } 1785 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1786 frameCount, mFrameCount); 1787 } else { 1788 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1789 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1790 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1791 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1792 audio_is_linear_pcm(format), 1793 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1794 *flags &= ~IAudioFlinger::TRACK_FAST; 1795 // For compatibility with AudioTrack calculation, buffer depth is forced 1796 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1797 // This is probably too conservative, but legacy application code may depend on it. 1798 // If you change this calculation, also review the start threshold which is related. 1799 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1800 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1801 if (minBufCount < 2) { 1802 minBufCount = 2; 1803 } 1804 size_t minFrameCount = mNormalFrameCount * minBufCount; 1805 if (frameCount < minFrameCount) { 1806 frameCount = minFrameCount; 1807 } 1808 } 1809 } 1810 1811 if (mType == DIRECT) { 1812 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1813 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1814 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1815 "for output %p with format %d", 1816 sampleRate, format, channelMask, mOutput, mFormat); 1817 lStatus = BAD_VALUE; 1818 goto Exit; 1819 } 1820 } 1821 } else { 1822 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1823 if (sampleRate > mSampleRate*2) { 1824 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1825 lStatus = BAD_VALUE; 1826 goto Exit; 1827 } 1828 } 1829 1830 lStatus = initCheck(); 1831 if (lStatus != NO_ERROR) { 1832 ALOGE("Audio driver not initialized."); 1833 goto Exit; 1834 } 1835 1836 { // scope for mLock 1837 Mutex::Autolock _l(mLock); 1838 1839 // all tracks in same audio session must share the same routing strategy otherwise 1840 // conflicts will happen when tracks are moved from one output to another by audio policy 1841 // manager 1842 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1843 for (size_t i = 0; i < mTracks.size(); ++i) { 1844 sp<Track> t = mTracks[i]; 1845 if (t != 0 && !t->isOutputTrack()) { 1846 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1847 if (sessionId == t->sessionId() && strategy != actual) { 1848 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1849 strategy, actual); 1850 lStatus = BAD_VALUE; 1851 goto Exit; 1852 } 1853 } 1854 } 1855 1856 if (!isTimed) { 1857 track = new Track(this, client, streamType, sampleRate, format, 1858 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1859 } else { 1860 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1861 channelMask, frameCount, sharedBuffer, sessionId); 1862 } 1863 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1864 lStatus = NO_MEMORY; 1865 goto Exit; 1866 } 1867 mTracks.add(track); 1868 1869 sp<EffectChain> chain = getEffectChain_l(sessionId); 1870 if (chain != 0) { 1871 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1872 track->setMainBuffer(chain->inBuffer()); 1873 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1874 chain->incTrackCnt(); 1875 } 1876 1877 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1878 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1879 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1880 // so ask activity manager to do this on our behalf 1881 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1882 } 1883 } 1884 1885 lStatus = NO_ERROR; 1886 1887Exit: 1888 if (status) { 1889 *status = lStatus; 1890 } 1891 return track; 1892} 1893 1894uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 1895{ 1896 if (mFastMixer != NULL) { 1897 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1898 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1899 } 1900 return latency; 1901} 1902 1903uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1904{ 1905 return latency; 1906} 1907 1908uint32_t AudioFlinger::PlaybackThread::latency() const 1909{ 1910 Mutex::Autolock _l(mLock); 1911 return latency_l(); 1912} 1913uint32_t AudioFlinger::PlaybackThread::latency_l() const 1914{ 1915 if (initCheck() == NO_ERROR) { 1916 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1917 } else { 1918 return 0; 1919 } 1920} 1921 1922void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1923{ 1924 Mutex::Autolock _l(mLock); 1925 // Don't apply master volume in SW if our HAL can do it for us. 1926 if (mOutput && mOutput->audioHwDev && 1927 mOutput->audioHwDev->canSetMasterVolume()) { 1928 mMasterVolume = 1.0; 1929 } else { 1930 mMasterVolume = value; 1931 } 1932} 1933 1934void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1935{ 1936 Mutex::Autolock _l(mLock); 1937 // Don't apply master mute in SW if our HAL can do it for us. 1938 if (mOutput && mOutput->audioHwDev && 1939 mOutput->audioHwDev->canSetMasterMute()) { 1940 mMasterMute = false; 1941 } else { 1942 mMasterMute = muted; 1943 } 1944} 1945 1946void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1947{ 1948 Mutex::Autolock _l(mLock); 1949 mStreamTypes[stream].volume = value; 1950} 1951 1952void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1953{ 1954 Mutex::Autolock _l(mLock); 1955 mStreamTypes[stream].mute = muted; 1956} 1957 1958float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1959{ 1960 Mutex::Autolock _l(mLock); 1961 return mStreamTypes[stream].volume; 1962} 1963 1964// addTrack_l() must be called with ThreadBase::mLock held 1965status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1966{ 1967 status_t status = ALREADY_EXISTS; 1968 1969 // set retry count for buffer fill 1970 track->mRetryCount = kMaxTrackStartupRetries; 1971 if (mActiveTracks.indexOf(track) < 0) { 1972 // the track is newly added, make sure it fills up all its 1973 // buffers before playing. This is to ensure the client will 1974 // effectively get the latency it requested. 1975 track->mFillingUpStatus = Track::FS_FILLING; 1976 track->mResetDone = false; 1977 track->mPresentationCompleteFrames = 0; 1978 mActiveTracks.add(track); 1979 if (track->mainBuffer() != mMixBuffer) { 1980 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1981 if (chain != 0) { 1982 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1983 track->sessionId()); 1984 chain->incActiveTrackCnt(); 1985 } 1986 } 1987 1988 status = NO_ERROR; 1989 } 1990 1991 ALOGV("mWaitWorkCV.broadcast"); 1992 mWaitWorkCV.broadcast(); 1993 1994 return status; 1995} 1996 1997// destroyTrack_l() must be called with ThreadBase::mLock held 1998void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1999{ 2000 track->mState = TrackBase::TERMINATED; 2001 // active tracks are removed by threadLoop() 2002 if (mActiveTracks.indexOf(track) < 0) { 2003 removeTrack_l(track); 2004 } 2005} 2006 2007void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2008{ 2009 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2010 mTracks.remove(track); 2011 deleteTrackName_l(track->name()); 2012 // redundant as track is about to be destroyed, for dumpsys only 2013 track->mName = -1; 2014 if (track->isFastTrack()) { 2015 int index = track->mFastIndex; 2016 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2017 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2018 mFastTrackAvailMask |= 1 << index; 2019 // redundant as track is about to be destroyed, for dumpsys only 2020 track->mFastIndex = -1; 2021 } 2022 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2023 if (chain != 0) { 2024 chain->decTrackCnt(); 2025 } 2026} 2027 2028String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2029{ 2030 String8 out_s8 = String8(""); 2031 char *s; 2032 2033 Mutex::Autolock _l(mLock); 2034 if (initCheck() != NO_ERROR) { 2035 return out_s8; 2036 } 2037 2038 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2039 out_s8 = String8(s); 2040 free(s); 2041 return out_s8; 2042} 2043 2044// audioConfigChanged_l() must be called with AudioFlinger::mLock held 2045void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 2046 AudioSystem::OutputDescriptor desc; 2047 void *param2 = NULL; 2048 2049 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 2050 param); 2051 2052 switch (event) { 2053 case AudioSystem::OUTPUT_OPENED: 2054 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2055 desc.channels = mChannelMask; 2056 desc.samplingRate = mSampleRate; 2057 desc.format = mFormat; 2058 desc.frameCount = mNormalFrameCount; // FIXME see 2059 // AudioFlinger::frameCount(audio_io_handle_t) 2060 desc.latency = latency(); 2061 param2 = &desc; 2062 break; 2063 2064 case AudioSystem::STREAM_CONFIG_CHANGED: 2065 param2 = ¶m; 2066 case AudioSystem::OUTPUT_CLOSED: 2067 default: 2068 break; 2069 } 2070 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2071} 2072 2073void AudioFlinger::PlaybackThread::readOutputParameters() 2074{ 2075 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2076 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2077 mChannelCount = (uint16_t)popcount(mChannelMask); 2078 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2079 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2080 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2081 if (mFrameCount & 15) { 2082 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2083 mFrameCount); 2084 } 2085 2086 // Calculate size of normal mix buffer relative to the HAL output buffer size 2087 double multiplier = 1.0; 2088 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2089 kUseFastMixer == FastMixer_Dynamic)) { 2090 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2091 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2092 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2093 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2094 maxNormalFrameCount = maxNormalFrameCount & ~15; 2095 if (maxNormalFrameCount < minNormalFrameCount) { 2096 maxNormalFrameCount = minNormalFrameCount; 2097 } 2098 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2099 if (multiplier <= 1.0) { 2100 multiplier = 1.0; 2101 } else if (multiplier <= 2.0) { 2102 if (2 * mFrameCount <= maxNormalFrameCount) { 2103 multiplier = 2.0; 2104 } else { 2105 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2106 } 2107 } else { 2108 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2109 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 2110 // track, but we sometimes have to do this to satisfy the maximum frame count 2111 // constraint) 2112 // FIXME this rounding up should not be done if no HAL SRC 2113 uint32_t truncMult = (uint32_t) multiplier; 2114 if ((truncMult & 1)) { 2115 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2116 ++truncMult; 2117 } 2118 } 2119 multiplier = (double) truncMult; 2120 } 2121 } 2122 mNormalFrameCount = multiplier * mFrameCount; 2123 // round up to nearest 16 frames to satisfy AudioMixer 2124 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2125 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 2126 mNormalFrameCount); 2127 2128 delete[] mMixBuffer; 2129 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2130 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2131 2132 // force reconfiguration of effect chains and engines to take new buffer size and audio 2133 // parameters into account 2134 // Note that mLock is not held when readOutputParameters() is called from the constructor 2135 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2136 // matter. 2137 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2138 Vector< sp<EffectChain> > effectChains = mEffectChains; 2139 for (size_t i = 0; i < effectChains.size(); i ++) { 2140 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2141 } 2142} 2143 2144 2145status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 2146{ 2147 if (halFrames == NULL || dspFrames == NULL) { 2148 return BAD_VALUE; 2149 } 2150 Mutex::Autolock _l(mLock); 2151 if (initCheck() != NO_ERROR) { 2152 return INVALID_OPERATION; 2153 } 2154 size_t framesWritten = mBytesWritten / mFrameSize; 2155 *halFrames = framesWritten; 2156 2157 if (isSuspended()) { 2158 // return an estimation of rendered frames when the output is suspended 2159 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2160 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2161 return NO_ERROR; 2162 } else { 2163 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2164 } 2165} 2166 2167uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2168{ 2169 Mutex::Autolock _l(mLock); 2170 uint32_t result = 0; 2171 if (getEffectChain_l(sessionId) != 0) { 2172 result = EFFECT_SESSION; 2173 } 2174 2175 for (size_t i = 0; i < mTracks.size(); ++i) { 2176 sp<Track> track = mTracks[i]; 2177 if (sessionId == track->sessionId() && 2178 !(track->mCblk->flags & CBLK_INVALID)) { 2179 result |= TRACK_SESSION; 2180 break; 2181 } 2182 } 2183 2184 return result; 2185} 2186 2187uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2188{ 2189 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2190 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2191 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2192 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2193 } 2194 for (size_t i = 0; i < mTracks.size(); i++) { 2195 sp<Track> track = mTracks[i]; 2196 if (sessionId == track->sessionId() && 2197 !(track->mCblk->flags & CBLK_INVALID)) { 2198 return AudioSystem::getStrategyForStream(track->streamType()); 2199 } 2200 } 2201 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2202} 2203 2204 2205AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2206{ 2207 Mutex::Autolock _l(mLock); 2208 return mOutput; 2209} 2210 2211AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2212{ 2213 Mutex::Autolock _l(mLock); 2214 AudioStreamOut *output = mOutput; 2215 mOutput = NULL; 2216 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2217 // must push a NULL and wait for ack 2218 mOutputSink.clear(); 2219 mPipeSink.clear(); 2220 mNormalSink.clear(); 2221 return output; 2222} 2223 2224// this method must always be called either with ThreadBase mLock held or inside the thread loop 2225audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2226{ 2227 if (mOutput == NULL) { 2228 return NULL; 2229 } 2230 return &mOutput->stream->common; 2231} 2232 2233uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2234{ 2235 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2236} 2237 2238status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2239{ 2240 if (!isValidSyncEvent(event)) { 2241 return BAD_VALUE; 2242 } 2243 2244 Mutex::Autolock _l(mLock); 2245 2246 for (size_t i = 0; i < mTracks.size(); ++i) { 2247 sp<Track> track = mTracks[i]; 2248 if (event->triggerSession() == track->sessionId()) { 2249 (void) track->setSyncEvent(event); 2250 return NO_ERROR; 2251 } 2252 } 2253 2254 return NAME_NOT_FOUND; 2255} 2256 2257bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2258{ 2259 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2260} 2261 2262void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2263 const Vector< sp<Track> >& tracksToRemove) 2264{ 2265 size_t count = tracksToRemove.size(); 2266 if (CC_UNLIKELY(count)) { 2267 for (size_t i = 0 ; i < count ; i++) { 2268 const sp<Track>& track = tracksToRemove.itemAt(i); 2269 if ((track->sharedBuffer() != 0) && 2270 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2271 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2272 } 2273 } 2274 } 2275 2276} 2277 2278// ---------------------------------------------------------------------------- 2279 2280AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2281 audio_io_handle_t id, audio_devices_t device, type_t type) 2282 : PlaybackThread(audioFlinger, output, id, device, type), 2283 // mAudioMixer below 2284 // mFastMixer below 2285 mFastMixerFutex(0) 2286 // mOutputSink below 2287 // mPipeSink below 2288 // mNormalSink below 2289{ 2290 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2291 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2292 "mFrameCount=%d, mNormalFrameCount=%d", 2293 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2294 mNormalFrameCount); 2295 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2296 2297 // FIXME - Current mixer implementation only supports stereo output 2298 if (mChannelCount != FCC_2) { 2299 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2300 } 2301 2302 // create an NBAIO sink for the HAL output stream, and negotiate 2303 mOutputSink = new AudioStreamOutSink(output->stream); 2304 size_t numCounterOffers = 0; 2305 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2306 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2307 ALOG_ASSERT(index == 0); 2308 2309 // initialize fast mixer depending on configuration 2310 bool initFastMixer; 2311 switch (kUseFastMixer) { 2312 case FastMixer_Never: 2313 initFastMixer = false; 2314 break; 2315 case FastMixer_Always: 2316 initFastMixer = true; 2317 break; 2318 case FastMixer_Static: 2319 case FastMixer_Dynamic: 2320 initFastMixer = mFrameCount < mNormalFrameCount; 2321 break; 2322 } 2323 if (initFastMixer) { 2324 2325 // create a MonoPipe to connect our submix to FastMixer 2326 NBAIO_Format format = mOutputSink->format(); 2327 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2328 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2329 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2330 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2331 const NBAIO_Format offers[1] = {format}; 2332 size_t numCounterOffers = 0; 2333 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2334 ALOG_ASSERT(index == 0); 2335 monoPipe->setAvgFrames((mScreenState & 1) ? 2336 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2337 mPipeSink = monoPipe; 2338 2339#ifdef TEE_SINK_FRAMES 2340 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2341 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2342 numCounterOffers = 0; 2343 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2344 ALOG_ASSERT(index == 0); 2345 mTeeSink = teeSink; 2346 PipeReader *teeSource = new PipeReader(*teeSink); 2347 numCounterOffers = 0; 2348 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2349 ALOG_ASSERT(index == 0); 2350 mTeeSource = teeSource; 2351#endif 2352 2353 // create fast mixer and configure it initially with just one fast track for our submix 2354 mFastMixer = new FastMixer(); 2355 FastMixerStateQueue *sq = mFastMixer->sq(); 2356#ifdef STATE_QUEUE_DUMP 2357 sq->setObserverDump(&mStateQueueObserverDump); 2358 sq->setMutatorDump(&mStateQueueMutatorDump); 2359#endif 2360 FastMixerState *state = sq->begin(); 2361 FastTrack *fastTrack = &state->mFastTracks[0]; 2362 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2363 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2364 fastTrack->mVolumeProvider = NULL; 2365 fastTrack->mGeneration++; 2366 state->mFastTracksGen++; 2367 state->mTrackMask = 1; 2368 // fast mixer will use the HAL output sink 2369 state->mOutputSink = mOutputSink.get(); 2370 state->mOutputSinkGen++; 2371 state->mFrameCount = mFrameCount; 2372 state->mCommand = FastMixerState::COLD_IDLE; 2373 // already done in constructor initialization list 2374 //mFastMixerFutex = 0; 2375 state->mColdFutexAddr = &mFastMixerFutex; 2376 state->mColdGen++; 2377 state->mDumpState = &mFastMixerDumpState; 2378 state->mTeeSink = mTeeSink.get(); 2379 sq->end(); 2380 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2381 2382 // start the fast mixer 2383 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2384 pid_t tid = mFastMixer->getTid(); 2385 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2386 if (err != 0) { 2387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2388 kPriorityFastMixer, getpid_cached, tid, err); 2389 } 2390 2391#ifdef AUDIO_WATCHDOG 2392 // create and start the watchdog 2393 mAudioWatchdog = new AudioWatchdog(); 2394 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2395 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2396 tid = mAudioWatchdog->getTid(); 2397 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2398 if (err != 0) { 2399 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2400 kPriorityFastMixer, getpid_cached, tid, err); 2401 } 2402#endif 2403 2404 } else { 2405 mFastMixer = NULL; 2406 } 2407 2408 switch (kUseFastMixer) { 2409 case FastMixer_Never: 2410 case FastMixer_Dynamic: 2411 mNormalSink = mOutputSink; 2412 break; 2413 case FastMixer_Always: 2414 mNormalSink = mPipeSink; 2415 break; 2416 case FastMixer_Static: 2417 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2418 break; 2419 } 2420} 2421 2422AudioFlinger::MixerThread::~MixerThread() 2423{ 2424 if (mFastMixer != NULL) { 2425 FastMixerStateQueue *sq = mFastMixer->sq(); 2426 FastMixerState *state = sq->begin(); 2427 if (state->mCommand == FastMixerState::COLD_IDLE) { 2428 int32_t old = android_atomic_inc(&mFastMixerFutex); 2429 if (old == -1) { 2430 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2431 } 2432 } 2433 state->mCommand = FastMixerState::EXIT; 2434 sq->end(); 2435 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2436 mFastMixer->join(); 2437 // Though the fast mixer thread has exited, it's state queue is still valid. 2438 // We'll use that extract the final state which contains one remaining fast track 2439 // corresponding to our sub-mix. 2440 state = sq->begin(); 2441 ALOG_ASSERT(state->mTrackMask == 1); 2442 FastTrack *fastTrack = &state->mFastTracks[0]; 2443 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2444 delete fastTrack->mBufferProvider; 2445 sq->end(false /*didModify*/); 2446 delete mFastMixer; 2447#ifdef AUDIO_WATCHDOG 2448 if (mAudioWatchdog != 0) { 2449 mAudioWatchdog->requestExit(); 2450 mAudioWatchdog->requestExitAndWait(); 2451 mAudioWatchdog.clear(); 2452 } 2453#endif 2454 } 2455 delete mAudioMixer; 2456} 2457 2458class CpuStats { 2459public: 2460 CpuStats(); 2461 void sample(const String8 &title); 2462#ifdef DEBUG_CPU_USAGE 2463private: 2464 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2465 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2466 2467 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2468 2469 int mCpuNum; // thread's current CPU number 2470 int mCpukHz; // frequency of thread's current CPU in kHz 2471#endif 2472}; 2473 2474CpuStats::CpuStats() 2475#ifdef DEBUG_CPU_USAGE 2476 : mCpuNum(-1), mCpukHz(-1) 2477#endif 2478{ 2479} 2480 2481void CpuStats::sample(const String8 &title) { 2482#ifdef DEBUG_CPU_USAGE 2483 // get current thread's delta CPU time in wall clock ns 2484 double wcNs; 2485 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2486 2487 // record sample for wall clock statistics 2488 if (valid) { 2489 mWcStats.sample(wcNs); 2490 } 2491 2492 // get the current CPU number 2493 int cpuNum = sched_getcpu(); 2494 2495 // get the current CPU frequency in kHz 2496 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2497 2498 // check if either CPU number or frequency changed 2499 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2500 mCpuNum = cpuNum; 2501 mCpukHz = cpukHz; 2502 // ignore sample for purposes of cycles 2503 valid = false; 2504 } 2505 2506 // if no change in CPU number or frequency, then record sample for cycle statistics 2507 if (valid && mCpukHz > 0) { 2508 double cycles = wcNs * cpukHz * 0.000001; 2509 mHzStats.sample(cycles); 2510 } 2511 2512 unsigned n = mWcStats.n(); 2513 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2514 if ((n & 127) == 1) { 2515 long long elapsed = mCpuUsage.elapsed(); 2516 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2517 double perLoop = elapsed / (double) n; 2518 double perLoop100 = perLoop * 0.01; 2519 double perLoop1k = perLoop * 0.001; 2520 double mean = mWcStats.mean(); 2521 double stddev = mWcStats.stddev(); 2522 double minimum = mWcStats.minimum(); 2523 double maximum = mWcStats.maximum(); 2524 double meanCycles = mHzStats.mean(); 2525 double stddevCycles = mHzStats.stddev(); 2526 double minCycles = mHzStats.minimum(); 2527 double maxCycles = mHzStats.maximum(); 2528 mCpuUsage.resetElapsed(); 2529 mWcStats.reset(); 2530 mHzStats.reset(); 2531 ALOGD("CPU usage for %s over past %.1f secs\n" 2532 " (%u mixer loops at %.1f mean ms per loop):\n" 2533 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2534 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2535 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2536 title.string(), 2537 elapsed * .000000001, n, perLoop * .000001, 2538 mean * .001, 2539 stddev * .001, 2540 minimum * .001, 2541 maximum * .001, 2542 mean / perLoop100, 2543 stddev / perLoop100, 2544 minimum / perLoop100, 2545 maximum / perLoop100, 2546 meanCycles / perLoop1k, 2547 stddevCycles / perLoop1k, 2548 minCycles / perLoop1k, 2549 maxCycles / perLoop1k); 2550 2551 } 2552 } 2553#endif 2554}; 2555 2556void AudioFlinger::PlaybackThread::checkSilentMode_l() 2557{ 2558 if (!mMasterMute) { 2559 char value[PROPERTY_VALUE_MAX]; 2560 if (property_get("ro.audio.silent", value, "0") > 0) { 2561 char *endptr; 2562 unsigned long ul = strtoul(value, &endptr, 0); 2563 if (*endptr == '\0' && ul != 0) { 2564 ALOGD("Silence is golden"); 2565 // The setprop command will not allow a property to be changed after 2566 // the first time it is set, so we don't have to worry about un-muting. 2567 setMasterMute_l(true); 2568 } 2569 } 2570 } 2571} 2572 2573bool AudioFlinger::PlaybackThread::threadLoop() 2574{ 2575 Vector< sp<Track> > tracksToRemove; 2576 2577 standbyTime = systemTime(); 2578 2579 // MIXER 2580 nsecs_t lastWarning = 0; 2581 2582 // DUPLICATING 2583 // FIXME could this be made local to while loop? 2584 writeFrames = 0; 2585 2586 cacheParameters_l(); 2587 sleepTime = idleSleepTime; 2588 2589 if (mType == MIXER) { 2590 sleepTimeShift = 0; 2591 } 2592 2593 CpuStats cpuStats; 2594 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2595 2596 acquireWakeLock(); 2597 2598 while (!exitPending()) 2599 { 2600 cpuStats.sample(myName); 2601 2602 Vector< sp<EffectChain> > effectChains; 2603 2604 processConfigEvents(); 2605 2606 { // scope for mLock 2607 2608 Mutex::Autolock _l(mLock); 2609 2610 if (checkForNewParameters_l()) { 2611 cacheParameters_l(); 2612 } 2613 2614 saveOutputTracks(); 2615 2616 // put audio hardware into standby after short delay 2617 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2618 isSuspended())) { 2619 if (!mStandby) { 2620 2621 threadLoop_standby(); 2622 2623 mStandby = true; 2624 } 2625 2626 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2627 // we're about to wait, flush the binder command buffer 2628 IPCThreadState::self()->flushCommands(); 2629 2630 clearOutputTracks(); 2631 2632 if (exitPending()) { 2633 break; 2634 } 2635 2636 releaseWakeLock_l(); 2637 // wait until we have something to do... 2638 ALOGV("%s going to sleep", myName.string()); 2639 mWaitWorkCV.wait(mLock); 2640 ALOGV("%s waking up", myName.string()); 2641 acquireWakeLock_l(); 2642 2643 mMixerStatus = MIXER_IDLE; 2644 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2645 mBytesWritten = 0; 2646 2647 checkSilentMode_l(); 2648 2649 standbyTime = systemTime() + standbyDelay; 2650 sleepTime = idleSleepTime; 2651 if (mType == MIXER) { 2652 sleepTimeShift = 0; 2653 } 2654 2655 continue; 2656 } 2657 } 2658 2659 // mMixerStatusIgnoringFastTracks is also updated internally 2660 mMixerStatus = prepareTracks_l(&tracksToRemove); 2661 2662 // prevent any changes in effect chain list and in each effect chain 2663 // during mixing and effect process as the audio buffers could be deleted 2664 // or modified if an effect is created or deleted 2665 lockEffectChains_l(effectChains); 2666 } 2667 2668 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2669 threadLoop_mix(); 2670 } else { 2671 threadLoop_sleepTime(); 2672 } 2673 2674 if (isSuspended()) { 2675 sleepTime = suspendSleepTimeUs(); 2676 mBytesWritten += mixBufferSize; 2677 } 2678 2679 // only process effects if we're going to write 2680 if (sleepTime == 0) { 2681 for (size_t i = 0; i < effectChains.size(); i ++) { 2682 effectChains[i]->process_l(); 2683 } 2684 } 2685 2686 // enable changes in effect chain 2687 unlockEffectChains(effectChains); 2688 2689 // sleepTime == 0 means we must write to audio hardware 2690 if (sleepTime == 0) { 2691 2692 threadLoop_write(); 2693 2694if (mType == MIXER) { 2695 // write blocked detection 2696 nsecs_t now = systemTime(); 2697 nsecs_t delta = now - mLastWriteTime; 2698 if (!mStandby && delta > maxPeriod) { 2699 mNumDelayedWrites++; 2700 if ((now - lastWarning) > kWarningThrottleNs) { 2701#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2702 ScopedTrace st(ATRACE_TAG, "underrun"); 2703#endif 2704 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2705 ns2ms(delta), mNumDelayedWrites, this); 2706 lastWarning = now; 2707 } 2708 } 2709} 2710 2711 mStandby = false; 2712 } else { 2713 usleep(sleepTime); 2714 } 2715 2716 // Finally let go of removed track(s), without the lock held 2717 // since we can't guarantee the destructors won't acquire that 2718 // same lock. This will also mutate and push a new fast mixer state. 2719 threadLoop_removeTracks(tracksToRemove); 2720 tracksToRemove.clear(); 2721 2722 // FIXME I don't understand the need for this here; 2723 // it was in the original code but maybe the 2724 // assignment in saveOutputTracks() makes this unnecessary? 2725 clearOutputTracks(); 2726 2727 // Effect chains will be actually deleted here if they were removed from 2728 // mEffectChains list during mixing or effects processing 2729 effectChains.clear(); 2730 2731 // FIXME Note that the above .clear() is no longer necessary since effectChains 2732 // is now local to this block, but will keep it for now (at least until merge done). 2733 } 2734 2735 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2736 if (mType == MIXER || mType == DIRECT) { 2737 // put output stream into standby mode 2738 if (!mStandby) { 2739 mOutput->stream->common.standby(&mOutput->stream->common); 2740 } 2741 } 2742 2743 releaseWakeLock(); 2744 2745 ALOGV("Thread %p type %d exiting", this, mType); 2746 return false; 2747} 2748 2749void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2750{ 2751 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2752} 2753 2754void AudioFlinger::MixerThread::threadLoop_write() 2755{ 2756 // FIXME we should only do one push per cycle; confirm this is true 2757 // Start the fast mixer if it's not already running 2758 if (mFastMixer != NULL) { 2759 FastMixerStateQueue *sq = mFastMixer->sq(); 2760 FastMixerState *state = sq->begin(); 2761 if (state->mCommand != FastMixerState::MIX_WRITE && 2762 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2763 if (state->mCommand == FastMixerState::COLD_IDLE) { 2764 int32_t old = android_atomic_inc(&mFastMixerFutex); 2765 if (old == -1) { 2766 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2767 } 2768#ifdef AUDIO_WATCHDOG 2769 if (mAudioWatchdog != 0) { 2770 mAudioWatchdog->resume(); 2771 } 2772#endif 2773 } 2774 state->mCommand = FastMixerState::MIX_WRITE; 2775 sq->end(); 2776 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2777 if (kUseFastMixer == FastMixer_Dynamic) { 2778 mNormalSink = mPipeSink; 2779 } 2780 } else { 2781 sq->end(false /*didModify*/); 2782 } 2783 } 2784 PlaybackThread::threadLoop_write(); 2785} 2786 2787// shared by MIXER and DIRECT, overridden by DUPLICATING 2788void AudioFlinger::PlaybackThread::threadLoop_write() 2789{ 2790 // FIXME rewrite to reduce number of system calls 2791 mLastWriteTime = systemTime(); 2792 mInWrite = true; 2793 int bytesWritten; 2794 2795 // If an NBAIO sink is present, use it to write the normal mixer's submix 2796 if (mNormalSink != 0) { 2797#define mBitShift 2 // FIXME 2798 size_t count = mixBufferSize >> mBitShift; 2799#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2800 Tracer::traceBegin(ATRACE_TAG, "write"); 2801#endif 2802 // update the setpoint when gScreenState changes 2803 uint32_t screenState = gScreenState; 2804 if (screenState != mScreenState) { 2805 mScreenState = screenState; 2806 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2807 if (pipe != NULL) { 2808 pipe->setAvgFrames((mScreenState & 1) ? 2809 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2810 } 2811 } 2812 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2813#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2814 Tracer::traceEnd(ATRACE_TAG); 2815#endif 2816 if (framesWritten > 0) { 2817 bytesWritten = framesWritten << mBitShift; 2818 } else { 2819 bytesWritten = framesWritten; 2820 } 2821 // otherwise use the HAL / AudioStreamOut directly 2822 } else { 2823 // Direct output thread. 2824 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2825 } 2826 2827 if (bytesWritten > 0) { 2828 mBytesWritten += mixBufferSize; 2829 } 2830 mNumWrites++; 2831 mInWrite = false; 2832} 2833 2834void AudioFlinger::MixerThread::threadLoop_standby() 2835{ 2836 // Idle the fast mixer if it's currently running 2837 if (mFastMixer != NULL) { 2838 FastMixerStateQueue *sq = mFastMixer->sq(); 2839 FastMixerState *state = sq->begin(); 2840 if (!(state->mCommand & FastMixerState::IDLE)) { 2841 state->mCommand = FastMixerState::COLD_IDLE; 2842 state->mColdFutexAddr = &mFastMixerFutex; 2843 state->mColdGen++; 2844 mFastMixerFutex = 0; 2845 sq->end(); 2846 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2847 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2848 if (kUseFastMixer == FastMixer_Dynamic) { 2849 mNormalSink = mOutputSink; 2850 } 2851#ifdef AUDIO_WATCHDOG 2852 if (mAudioWatchdog != 0) { 2853 mAudioWatchdog->pause(); 2854 } 2855#endif 2856 } else { 2857 sq->end(false /*didModify*/); 2858 } 2859 } 2860 PlaybackThread::threadLoop_standby(); 2861} 2862 2863// shared by MIXER and DIRECT, overridden by DUPLICATING 2864void AudioFlinger::PlaybackThread::threadLoop_standby() 2865{ 2866 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2867 mOutput->stream->common.standby(&mOutput->stream->common); 2868} 2869 2870void AudioFlinger::MixerThread::threadLoop_mix() 2871{ 2872 // obtain the presentation timestamp of the next output buffer 2873 int64_t pts; 2874 status_t status = INVALID_OPERATION; 2875 2876 if (mNormalSink != 0) { 2877 status = mNormalSink->getNextWriteTimestamp(&pts); 2878 } else { 2879 status = mOutputSink->getNextWriteTimestamp(&pts); 2880 } 2881 2882 if (status != NO_ERROR) { 2883 pts = AudioBufferProvider::kInvalidPTS; 2884 } 2885 2886 // mix buffers... 2887 mAudioMixer->process(pts); 2888 // increase sleep time progressively when application underrun condition clears. 2889 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2890 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2891 // such that we would underrun the audio HAL. 2892 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2893 sleepTimeShift--; 2894 } 2895 sleepTime = 0; 2896 standbyTime = systemTime() + standbyDelay; 2897 //TODO: delay standby when effects have a tail 2898} 2899 2900void AudioFlinger::MixerThread::threadLoop_sleepTime() 2901{ 2902 // If no tracks are ready, sleep once for the duration of an output 2903 // buffer size, then write 0s to the output 2904 if (sleepTime == 0) { 2905 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2906 sleepTime = activeSleepTime >> sleepTimeShift; 2907 if (sleepTime < kMinThreadSleepTimeUs) { 2908 sleepTime = kMinThreadSleepTimeUs; 2909 } 2910 // reduce sleep time in case of consecutive application underruns to avoid 2911 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2912 // duration we would end up writing less data than needed by the audio HAL if 2913 // the condition persists. 2914 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2915 sleepTimeShift++; 2916 } 2917 } else { 2918 sleepTime = idleSleepTime; 2919 } 2920 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2921 memset (mMixBuffer, 0, mixBufferSize); 2922 sleepTime = 0; 2923 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2924 "anticipated start"); 2925 } 2926 // TODO add standby time extension fct of effect tail 2927} 2928 2929// prepareTracks_l() must be called with ThreadBase::mLock held 2930AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2931 Vector< sp<Track> > *tracksToRemove) 2932{ 2933 2934 mixer_state mixerStatus = MIXER_IDLE; 2935 // find out which tracks need to be processed 2936 size_t count = mActiveTracks.size(); 2937 size_t mixedTracks = 0; 2938 size_t tracksWithEffect = 0; 2939 // counts only _active_ fast tracks 2940 size_t fastTracks = 0; 2941 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2942 2943 float masterVolume = mMasterVolume; 2944 bool masterMute = mMasterMute; 2945 2946 if (masterMute) { 2947 masterVolume = 0; 2948 } 2949 // Delegate master volume control to effect in output mix effect chain if needed 2950 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2951 if (chain != 0) { 2952 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2953 chain->setVolume_l(&v, &v); 2954 masterVolume = (float)((v + (1 << 23)) >> 24); 2955 chain.clear(); 2956 } 2957 2958 // prepare a new state to push 2959 FastMixerStateQueue *sq = NULL; 2960 FastMixerState *state = NULL; 2961 bool didModify = false; 2962 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2963 if (mFastMixer != NULL) { 2964 sq = mFastMixer->sq(); 2965 state = sq->begin(); 2966 } 2967 2968 for (size_t i=0 ; i<count ; i++) { 2969 sp<Track> t = mActiveTracks[i].promote(); 2970 if (t == 0) { 2971 continue; 2972 } 2973 2974 // this const just means the local variable doesn't change 2975 Track* const track = t.get(); 2976 2977 // process fast tracks 2978 if (track->isFastTrack()) { 2979 2980 // It's theoretically possible (though unlikely) for a fast track to be created 2981 // and then removed within the same normal mix cycle. This is not a problem, as 2982 // the track never becomes active so it's fast mixer slot is never touched. 2983 // The converse, of removing an (active) track and then creating a new track 2984 // at the identical fast mixer slot within the same normal mix cycle, 2985 // is impossible because the slot isn't marked available until the end of each cycle. 2986 int j = track->mFastIndex; 2987 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2988 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2989 FastTrack *fastTrack = &state->mFastTracks[j]; 2990 2991 // Determine whether the track is currently in underrun condition, 2992 // and whether it had a recent underrun. 2993 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2994 FastTrackUnderruns underruns = ftDump->mUnderruns; 2995 uint32_t recentFull = (underruns.mBitFields.mFull - 2996 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2997 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2998 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2999 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3000 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3001 uint32_t recentUnderruns = recentPartial + recentEmpty; 3002 track->mObservedUnderruns = underruns; 3003 // don't count underruns that occur while stopping or pausing 3004 // or stopped which can occur when flush() is called while active 3005 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 3006 track->mUnderrunCount += recentUnderruns; 3007 } 3008 3009 // This is similar to the state machine for normal tracks, 3010 // with a few modifications for fast tracks. 3011 bool isActive = true; 3012 switch (track->mState) { 3013 case TrackBase::STOPPING_1: 3014 // track stays active in STOPPING_1 state until first underrun 3015 if (recentUnderruns > 0) { 3016 track->mState = TrackBase::STOPPING_2; 3017 } 3018 break; 3019 case TrackBase::PAUSING: 3020 // ramp down is not yet implemented 3021 track->setPaused(); 3022 break; 3023 case TrackBase::RESUMING: 3024 // ramp up is not yet implemented 3025 track->mState = TrackBase::ACTIVE; 3026 break; 3027 case TrackBase::ACTIVE: 3028 if (recentFull > 0 || recentPartial > 0) { 3029 // track has provided at least some frames recently: reset retry count 3030 track->mRetryCount = kMaxTrackRetries; 3031 } 3032 if (recentUnderruns == 0) { 3033 // no recent underruns: stay active 3034 break; 3035 } 3036 // there has recently been an underrun of some kind 3037 if (track->sharedBuffer() == 0) { 3038 // were any of the recent underruns "empty" (no frames available)? 3039 if (recentEmpty == 0) { 3040 // no, then ignore the partial underruns as they are allowed indefinitely 3041 break; 3042 } 3043 // there has recently been an "empty" underrun: decrement the retry counter 3044 if (--(track->mRetryCount) > 0) { 3045 break; 3046 } 3047 // indicate to client process that the track was disabled because of underrun; 3048 // it will then automatically call start() when data is available 3049 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 3050 // remove from active list, but state remains ACTIVE [confusing but true] 3051 isActive = false; 3052 break; 3053 } 3054 // fall through 3055 case TrackBase::STOPPING_2: 3056 case TrackBase::PAUSED: 3057 case TrackBase::TERMINATED: 3058 case TrackBase::STOPPED: 3059 case TrackBase::FLUSHED: // flush() while active 3060 // Check for presentation complete if track is inactive 3061 // We have consumed all the buffers of this track. 3062 // This would be incomplete if we auto-paused on underrun 3063 { 3064 size_t audioHALFrames = 3065 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3066 size_t framesWritten = mBytesWritten / mFrameSize; 3067 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3068 // track stays in active list until presentation is complete 3069 break; 3070 } 3071 } 3072 if (track->isStopping_2()) { 3073 track->mState = TrackBase::STOPPED; 3074 } 3075 if (track->isStopped()) { 3076 // Can't reset directly, as fast mixer is still polling this track 3077 // track->reset(); 3078 // So instead mark this track as needing to be reset after push with ack 3079 resetMask |= 1 << i; 3080 } 3081 isActive = false; 3082 break; 3083 case TrackBase::IDLE: 3084 default: 3085 LOG_FATAL("unexpected track state %d", track->mState); 3086 } 3087 3088 if (isActive) { 3089 // was it previously inactive? 3090 if (!(state->mTrackMask & (1 << j))) { 3091 ExtendedAudioBufferProvider *eabp = track; 3092 VolumeProvider *vp = track; 3093 fastTrack->mBufferProvider = eabp; 3094 fastTrack->mVolumeProvider = vp; 3095 fastTrack->mSampleRate = track->mSampleRate; 3096 fastTrack->mChannelMask = track->mChannelMask; 3097 fastTrack->mGeneration++; 3098 state->mTrackMask |= 1 << j; 3099 didModify = true; 3100 // no acknowledgement required for newly active tracks 3101 } 3102 // cache the combined master volume and stream type volume for fast mixer; this 3103 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3104 track->mCachedVolume = track->isMuted() ? 3105 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3106 ++fastTracks; 3107 } else { 3108 // was it previously active? 3109 if (state->mTrackMask & (1 << j)) { 3110 fastTrack->mBufferProvider = NULL; 3111 fastTrack->mGeneration++; 3112 state->mTrackMask &= ~(1 << j); 3113 didModify = true; 3114 // If any fast tracks were removed, we must wait for acknowledgement 3115 // because we're about to decrement the last sp<> on those tracks. 3116 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3117 } else { 3118 LOG_FATAL("fast track %d should have been active", j); 3119 } 3120 tracksToRemove->add(track); 3121 // Avoids a misleading display in dumpsys 3122 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3123 } 3124 continue; 3125 } 3126 3127 { // local variable scope to avoid goto warning 3128 3129 audio_track_cblk_t* cblk = track->cblk(); 3130 3131 // The first time a track is added we wait 3132 // for all its buffers to be filled before processing it 3133 int name = track->name(); 3134 // make sure that we have enough frames to mix one full buffer. 3135 // enforce this condition only once to enable draining the buffer in case the client 3136 // app does not call stop() and relies on underrun to stop: 3137 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3138 // during last round 3139 uint32_t minFrames = 1; 3140 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3141 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3142 if (t->sampleRate() == mSampleRate) { 3143 minFrames = mNormalFrameCount; 3144 } else { 3145 // +1 for rounding and +1 for additional sample needed for interpolation 3146 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3147 // add frames already consumed but not yet released by the resampler 3148 // because cblk->framesReady() will include these frames 3149 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3150 // the minimum track buffer size is normally twice the number of frames necessary 3151 // to fill one buffer and the resampler should not leave more than one buffer worth 3152 // of unreleased frames after each pass, but just in case... 3153 ALOG_ASSERT(minFrames <= cblk->frameCount); 3154 } 3155 } 3156 if ((track->framesReady() >= minFrames) && track->isReady() && 3157 !track->isPaused() && !track->isTerminated()) 3158 { 3159 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 3160 this); 3161 3162 mixedTracks++; 3163 3164 // track->mainBuffer() != mMixBuffer means there is an effect chain 3165 // connected to the track 3166 chain.clear(); 3167 if (track->mainBuffer() != mMixBuffer) { 3168 chain = getEffectChain_l(track->sessionId()); 3169 // Delegate volume control to effect in track effect chain if needed 3170 if (chain != 0) { 3171 tracksWithEffect++; 3172 } else { 3173 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3174 "session %d", 3175 name, track->sessionId()); 3176 } 3177 } 3178 3179 3180 int param = AudioMixer::VOLUME; 3181 if (track->mFillingUpStatus == Track::FS_FILLED) { 3182 // no ramp for the first volume setting 3183 track->mFillingUpStatus = Track::FS_ACTIVE; 3184 if (track->mState == TrackBase::RESUMING) { 3185 track->mState = TrackBase::ACTIVE; 3186 param = AudioMixer::RAMP_VOLUME; 3187 } 3188 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3189 } else if (cblk->server != 0) { 3190 // If the track is stopped before the first frame was mixed, 3191 // do not apply ramp 3192 param = AudioMixer::RAMP_VOLUME; 3193 } 3194 3195 // compute volume for this track 3196 uint32_t vl, vr, va; 3197 if (track->isMuted() || track->isPausing() || 3198 mStreamTypes[track->streamType()].mute) { 3199 vl = vr = va = 0; 3200 if (track->isPausing()) { 3201 track->setPaused(); 3202 } 3203 } else { 3204 3205 // read original volumes with volume control 3206 float typeVolume = mStreamTypes[track->streamType()].volume; 3207 float v = masterVolume * typeVolume; 3208 uint32_t vlr = cblk->getVolumeLR(); 3209 vl = vlr & 0xFFFF; 3210 vr = vlr >> 16; 3211 // track volumes come from shared memory, so can't be trusted and must be clamped 3212 if (vl > MAX_GAIN_INT) { 3213 ALOGV("Track left volume out of range: %04X", vl); 3214 vl = MAX_GAIN_INT; 3215 } 3216 if (vr > MAX_GAIN_INT) { 3217 ALOGV("Track right volume out of range: %04X", vr); 3218 vr = MAX_GAIN_INT; 3219 } 3220 // now apply the master volume and stream type volume 3221 vl = (uint32_t)(v * vl) << 12; 3222 vr = (uint32_t)(v * vr) << 12; 3223 // assuming master volume and stream type volume each go up to 1.0, 3224 // vl and vr are now in 8.24 format 3225 3226 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3227 // send level comes from shared memory and so may be corrupt 3228 if (sendLevel > MAX_GAIN_INT) { 3229 ALOGV("Track send level out of range: %04X", sendLevel); 3230 sendLevel = MAX_GAIN_INT; 3231 } 3232 va = (uint32_t)(v * sendLevel); 3233 } 3234 // Delegate volume control to effect in track effect chain if needed 3235 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3236 // Do not ramp volume if volume is controlled by effect 3237 param = AudioMixer::VOLUME; 3238 track->mHasVolumeController = true; 3239 } else { 3240 // force no volume ramp when volume controller was just disabled or removed 3241 // from effect chain to avoid volume spike 3242 if (track->mHasVolumeController) { 3243 param = AudioMixer::VOLUME; 3244 } 3245 track->mHasVolumeController = false; 3246 } 3247 3248 // Convert volumes from 8.24 to 4.12 format 3249 // This additional clamping is needed in case chain->setVolume_l() overshot 3250 vl = (vl + (1 << 11)) >> 12; 3251 if (vl > MAX_GAIN_INT) { 3252 vl = MAX_GAIN_INT; 3253 } 3254 vr = (vr + (1 << 11)) >> 12; 3255 if (vr > MAX_GAIN_INT) { 3256 vr = MAX_GAIN_INT; 3257 } 3258 3259 if (va > MAX_GAIN_INT) { 3260 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3261 } 3262 3263 // XXX: these things DON'T need to be done each time 3264 mAudioMixer->setBufferProvider(name, track); 3265 mAudioMixer->enable(name); 3266 3267 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3268 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3269 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3270 mAudioMixer->setParameter( 3271 name, 3272 AudioMixer::TRACK, 3273 AudioMixer::FORMAT, (void *)track->format()); 3274 mAudioMixer->setParameter( 3275 name, 3276 AudioMixer::TRACK, 3277 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3278 mAudioMixer->setParameter( 3279 name, 3280 AudioMixer::RESAMPLE, 3281 AudioMixer::SAMPLE_RATE, 3282 (void *)(cblk->sampleRate)); 3283 mAudioMixer->setParameter( 3284 name, 3285 AudioMixer::TRACK, 3286 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3287 mAudioMixer->setParameter( 3288 name, 3289 AudioMixer::TRACK, 3290 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3291 3292 // reset retry count 3293 track->mRetryCount = kMaxTrackRetries; 3294 3295 // If one track is ready, set the mixer ready if: 3296 // - the mixer was not ready during previous round OR 3297 // - no other track is not ready 3298 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3299 mixerStatus != MIXER_TRACKS_ENABLED) { 3300 mixerStatus = MIXER_TRACKS_READY; 3301 } 3302 } else { 3303 // clear effect chain input buffer if an active track underruns to avoid sending 3304 // previous audio buffer again to effects 3305 chain = getEffectChain_l(track->sessionId()); 3306 if (chain != 0) { 3307 chain->clearInputBuffer(); 3308 } 3309 3310 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 3311 cblk->server, this); 3312 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3313 track->isStopped() || track->isPaused()) { 3314 // We have consumed all the buffers of this track. 3315 // Remove it from the list of active tracks. 3316 // TODO: use actual buffer filling status instead of latency when available from 3317 // audio HAL 3318 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3319 size_t framesWritten = mBytesWritten / mFrameSize; 3320 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3321 if (track->isStopped()) { 3322 track->reset(); 3323 } 3324 tracksToRemove->add(track); 3325 } 3326 } else { 3327 track->mUnderrunCount++; 3328 // No buffers for this track. Give it a few chances to 3329 // fill a buffer, then remove it from active list. 3330 if (--(track->mRetryCount) <= 0) { 3331 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3332 tracksToRemove->add(track); 3333 // indicate to client process that the track was disabled because of underrun; 3334 // it will then automatically call start() when data is available 3335 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3336 // If one track is not ready, mark the mixer also not ready if: 3337 // - the mixer was ready during previous round OR 3338 // - no other track is ready 3339 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3340 mixerStatus != MIXER_TRACKS_READY) { 3341 mixerStatus = MIXER_TRACKS_ENABLED; 3342 } 3343 } 3344 mAudioMixer->disable(name); 3345 } 3346 3347 } // local variable scope to avoid goto warning 3348track_is_ready: ; 3349 3350 } 3351 3352 // Push the new FastMixer state if necessary 3353 bool pauseAudioWatchdog = false; 3354 if (didModify) { 3355 state->mFastTracksGen++; 3356 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3357 if (kUseFastMixer == FastMixer_Dynamic && 3358 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3359 state->mCommand = FastMixerState::COLD_IDLE; 3360 state->mColdFutexAddr = &mFastMixerFutex; 3361 state->mColdGen++; 3362 mFastMixerFutex = 0; 3363 if (kUseFastMixer == FastMixer_Dynamic) { 3364 mNormalSink = mOutputSink; 3365 } 3366 // If we go into cold idle, need to wait for acknowledgement 3367 // so that fast mixer stops doing I/O. 3368 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3369 pauseAudioWatchdog = true; 3370 } 3371 sq->end(); 3372 } 3373 if (sq != NULL) { 3374 sq->end(didModify); 3375 sq->push(block); 3376 } 3377#ifdef AUDIO_WATCHDOG 3378 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3379 mAudioWatchdog->pause(); 3380 } 3381#endif 3382 3383 // Now perform the deferred reset on fast tracks that have stopped 3384 while (resetMask != 0) { 3385 size_t i = __builtin_ctz(resetMask); 3386 ALOG_ASSERT(i < count); 3387 resetMask &= ~(1 << i); 3388 sp<Track> t = mActiveTracks[i].promote(); 3389 if (t == 0) { 3390 continue; 3391 } 3392 Track* track = t.get(); 3393 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3394 track->reset(); 3395 } 3396 3397 // remove all the tracks that need to be... 3398 count = tracksToRemove->size(); 3399 if (CC_UNLIKELY(count)) { 3400 for (size_t i=0 ; i<count ; i++) { 3401 const sp<Track>& track = tracksToRemove->itemAt(i); 3402 mActiveTracks.remove(track); 3403 if (track->mainBuffer() != mMixBuffer) { 3404 chain = getEffectChain_l(track->sessionId()); 3405 if (chain != 0) { 3406 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3407 track->sessionId()); 3408 chain->decActiveTrackCnt(); 3409 } 3410 } 3411 if (track->isTerminated()) { 3412 removeTrack_l(track); 3413 } 3414 } 3415 } 3416 3417 // mix buffer must be cleared if all tracks are connected to an 3418 // effect chain as in this case the mixer will not write to 3419 // mix buffer and track effects will accumulate into it 3420 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3421 (mixedTracks == 0 && fastTracks > 0)) { 3422 // FIXME as a performance optimization, should remember previous zero status 3423 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3424 } 3425 3426 // if any fast tracks, then status is ready 3427 mMixerStatusIgnoringFastTracks = mixerStatus; 3428 if (fastTracks > 0) { 3429 mixerStatus = MIXER_TRACKS_READY; 3430 } 3431 return mixerStatus; 3432} 3433 3434/* 3435The derived values that are cached: 3436 - mixBufferSize from frame count * frame size 3437 - activeSleepTime from activeSleepTimeUs() 3438 - idleSleepTime from idleSleepTimeUs() 3439 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3440 - maxPeriod from frame count and sample rate (MIXER only) 3441 3442The parameters that affect these derived values are: 3443 - frame count 3444 - frame size 3445 - sample rate 3446 - device type: A2DP or not 3447 - device latency 3448 - format: PCM or not 3449 - active sleep time 3450 - idle sleep time 3451*/ 3452 3453void AudioFlinger::PlaybackThread::cacheParameters_l() 3454{ 3455 mixBufferSize = mNormalFrameCount * mFrameSize; 3456 activeSleepTime = activeSleepTimeUs(); 3457 idleSleepTime = idleSleepTimeUs(); 3458} 3459 3460void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3461{ 3462 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3463 this, streamType, mTracks.size()); 3464 Mutex::Autolock _l(mLock); 3465 3466 size_t size = mTracks.size(); 3467 for (size_t i = 0; i < size; i++) { 3468 sp<Track> t = mTracks[i]; 3469 if (t->streamType() == streamType) { 3470 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 3471 t->mCblk->cv.signal(); 3472 } 3473 } 3474} 3475 3476// getTrackName_l() must be called with ThreadBase::mLock held 3477int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3478{ 3479 return mAudioMixer->getTrackName(channelMask, sessionId); 3480} 3481 3482// deleteTrackName_l() must be called with ThreadBase::mLock held 3483void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3484{ 3485 ALOGV("remove track (%d) and delete from mixer", name); 3486 mAudioMixer->deleteTrackName(name); 3487} 3488 3489// checkForNewParameters_l() must be called with ThreadBase::mLock held 3490bool AudioFlinger::MixerThread::checkForNewParameters_l() 3491{ 3492 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3493 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3494 bool reconfig = false; 3495 3496 while (!mNewParameters.isEmpty()) { 3497 3498 if (mFastMixer != NULL) { 3499 FastMixerStateQueue *sq = mFastMixer->sq(); 3500 FastMixerState *state = sq->begin(); 3501 if (!(state->mCommand & FastMixerState::IDLE)) { 3502 previousCommand = state->mCommand; 3503 state->mCommand = FastMixerState::HOT_IDLE; 3504 sq->end(); 3505 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3506 } else { 3507 sq->end(false /*didModify*/); 3508 } 3509 } 3510 3511 status_t status = NO_ERROR; 3512 String8 keyValuePair = mNewParameters[0]; 3513 AudioParameter param = AudioParameter(keyValuePair); 3514 int value; 3515 3516 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3517 reconfig = true; 3518 } 3519 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3520 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3521 status = BAD_VALUE; 3522 } else { 3523 reconfig = true; 3524 } 3525 } 3526 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3527 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3528 status = BAD_VALUE; 3529 } else { 3530 reconfig = true; 3531 } 3532 } 3533 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3534 // do not accept frame count changes if tracks are open as the track buffer 3535 // size depends on frame count and correct behavior would not be guaranteed 3536 // if frame count is changed after track creation 3537 if (!mTracks.isEmpty()) { 3538 status = INVALID_OPERATION; 3539 } else { 3540 reconfig = true; 3541 } 3542 } 3543 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3544#ifdef ADD_BATTERY_DATA 3545 // when changing the audio output device, call addBatteryData to notify 3546 // the change 3547 if (mOutDevice != value) { 3548 uint32_t params = 0; 3549 // check whether speaker is on 3550 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3551 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3552 } 3553 3554 audio_devices_t deviceWithoutSpeaker 3555 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3556 // check if any other device (except speaker) is on 3557 if (value & deviceWithoutSpeaker ) { 3558 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3559 } 3560 3561 if (params != 0) { 3562 addBatteryData(params); 3563 } 3564 } 3565#endif 3566 3567 // forward device change to effects that have requested to be 3568 // aware of attached audio device. 3569 mOutDevice = value; 3570 for (size_t i = 0; i < mEffectChains.size(); i++) { 3571 mEffectChains[i]->setDevice_l(mOutDevice); 3572 } 3573 } 3574 3575 if (status == NO_ERROR) { 3576 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3577 keyValuePair.string()); 3578 if (!mStandby && status == INVALID_OPERATION) { 3579 mOutput->stream->common.standby(&mOutput->stream->common); 3580 mStandby = true; 3581 mBytesWritten = 0; 3582 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3583 keyValuePair.string()); 3584 } 3585 if (status == NO_ERROR && reconfig) { 3586 delete mAudioMixer; 3587 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3588 mAudioMixer = NULL; 3589 readOutputParameters(); 3590 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3591 for (size_t i = 0; i < mTracks.size() ; i++) { 3592 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3593 if (name < 0) { 3594 break; 3595 } 3596 mTracks[i]->mName = name; 3597 // limit track sample rate to 2 x new output sample rate 3598 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3599 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3600 } 3601 } 3602 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3603 } 3604 } 3605 3606 mNewParameters.removeAt(0); 3607 3608 mParamStatus = status; 3609 mParamCond.signal(); 3610 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3611 // already timed out waiting for the status and will never signal the condition. 3612 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3613 } 3614 3615 if (!(previousCommand & FastMixerState::IDLE)) { 3616 ALOG_ASSERT(mFastMixer != NULL); 3617 FastMixerStateQueue *sq = mFastMixer->sq(); 3618 FastMixerState *state = sq->begin(); 3619 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3620 state->mCommand = previousCommand; 3621 sq->end(); 3622 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3623 } 3624 3625 return reconfig; 3626} 3627 3628void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3629{ 3630 NBAIO_Source *teeSource = source.get(); 3631 if (teeSource != NULL) { 3632 char teeTime[16]; 3633 struct timeval tv; 3634 gettimeofday(&tv, NULL); 3635 struct tm tm; 3636 localtime_r(&tv.tv_sec, &tm); 3637 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3638 char teePath[64]; 3639 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3640 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3641 if (teeFd >= 0) { 3642 char wavHeader[44]; 3643 memcpy(wavHeader, 3644 "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", 3645 sizeof(wavHeader)); 3646 NBAIO_Format format = teeSource->format(); 3647 unsigned channelCount = Format_channelCount(format); 3648 ALOG_ASSERT(channelCount <= FCC_2); 3649 uint32_t sampleRate = Format_sampleRate(format); 3650 wavHeader[22] = channelCount; // number of channels 3651 wavHeader[24] = sampleRate; // sample rate 3652 wavHeader[25] = sampleRate >> 8; 3653 wavHeader[32] = channelCount * 2; // block alignment 3654 write(teeFd, wavHeader, sizeof(wavHeader)); 3655 size_t total = 0; 3656 bool firstRead = true; 3657 for (;;) { 3658#define TEE_SINK_READ 1024 3659 short buffer[TEE_SINK_READ * FCC_2]; 3660 size_t count = TEE_SINK_READ; 3661 ssize_t actual = teeSource->read(buffer, count, 3662 AudioBufferProvider::kInvalidPTS); 3663 bool wasFirstRead = firstRead; 3664 firstRead = false; 3665 if (actual <= 0) { 3666 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3667 continue; 3668 } 3669 break; 3670 } 3671 ALOG_ASSERT(actual <= (ssize_t)count); 3672 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3673 total += actual; 3674 } 3675 lseek(teeFd, (off_t) 4, SEEK_SET); 3676 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3677 write(teeFd, &temp, sizeof(temp)); 3678 lseek(teeFd, (off_t) 40, SEEK_SET); 3679 temp = total * channelCount * sizeof(short); 3680 write(teeFd, &temp, sizeof(temp)); 3681 close(teeFd); 3682 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3683 } else { 3684 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3685 } 3686 } 3687} 3688 3689void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3690{ 3691 const size_t SIZE = 256; 3692 char buffer[SIZE]; 3693 String8 result; 3694 3695 PlaybackThread::dumpInternals(fd, args); 3696 3697 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3698 result.append(buffer); 3699 write(fd, result.string(), result.size()); 3700 3701 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3702 FastMixerDumpState copy = mFastMixerDumpState; 3703 copy.dump(fd); 3704 3705#ifdef STATE_QUEUE_DUMP 3706 // Similar for state queue 3707 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3708 observerCopy.dump(fd); 3709 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3710 mutatorCopy.dump(fd); 3711#endif 3712 3713 // Write the tee output to a .wav file 3714 dumpTee(fd, mTeeSource, mId); 3715 3716#ifdef AUDIO_WATCHDOG 3717 if (mAudioWatchdog != 0) { 3718 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3719 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3720 wdCopy.dump(fd); 3721 } 3722#endif 3723} 3724 3725uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3726{ 3727 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3728} 3729 3730uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3731{ 3732 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3733} 3734 3735void AudioFlinger::MixerThread::cacheParameters_l() 3736{ 3737 PlaybackThread::cacheParameters_l(); 3738 3739 // FIXME: Relaxed timing because of a certain device that can't meet latency 3740 // Should be reduced to 2x after the vendor fixes the driver issue 3741 // increase threshold again due to low power audio mode. The way this warning 3742 // threshold is calculated and its usefulness should be reconsidered anyway. 3743 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3744} 3745 3746// ---------------------------------------------------------------------------- 3747AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3748 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3749 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3750 // mLeftVolFloat, mRightVolFloat 3751{ 3752} 3753 3754AudioFlinger::DirectOutputThread::~DirectOutputThread() 3755{ 3756} 3757 3758AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3759 Vector< sp<Track> > *tracksToRemove 3760) 3761{ 3762 sp<Track> trackToRemove; 3763 3764 mixer_state mixerStatus = MIXER_IDLE; 3765 3766 // find out which tracks need to be processed 3767 if (mActiveTracks.size() != 0) { 3768 sp<Track> t = mActiveTracks[0].promote(); 3769 // The track died recently 3770 if (t == 0) { 3771 return MIXER_IDLE; 3772 } 3773 3774 Track* const track = t.get(); 3775 audio_track_cblk_t* cblk = track->cblk(); 3776 3777 // The first time a track is added we wait 3778 // for all its buffers to be filled before processing it 3779 uint32_t minFrames; 3780 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3781 minFrames = mNormalFrameCount; 3782 } else { 3783 minFrames = 1; 3784 } 3785 if ((track->framesReady() >= minFrames) && track->isReady() && 3786 !track->isPaused() && !track->isTerminated()) 3787 { 3788 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3789 3790 if (track->mFillingUpStatus == Track::FS_FILLED) { 3791 track->mFillingUpStatus = Track::FS_ACTIVE; 3792 mLeftVolFloat = mRightVolFloat = 0; 3793 if (track->mState == TrackBase::RESUMING) { 3794 track->mState = TrackBase::ACTIVE; 3795 } 3796 } 3797 3798 // compute volume for this track 3799 float left, right; 3800 if (track->isMuted() || mMasterMute || track->isPausing() || 3801 mStreamTypes[track->streamType()].mute) { 3802 left = right = 0; 3803 if (track->isPausing()) { 3804 track->setPaused(); 3805 } 3806 } else { 3807 float typeVolume = mStreamTypes[track->streamType()].volume; 3808 float v = mMasterVolume * typeVolume; 3809 uint32_t vlr = cblk->getVolumeLR(); 3810 float v_clamped = v * (vlr & 0xFFFF); 3811 if (v_clamped > MAX_GAIN) { 3812 v_clamped = MAX_GAIN; 3813 } 3814 left = v_clamped/MAX_GAIN; 3815 v_clamped = v * (vlr >> 16); 3816 if (v_clamped > MAX_GAIN) { 3817 v_clamped = MAX_GAIN; 3818 } 3819 right = v_clamped/MAX_GAIN; 3820 } 3821 3822 if (left != mLeftVolFloat || right != mRightVolFloat) { 3823 mLeftVolFloat = left; 3824 mRightVolFloat = right; 3825 3826 // Convert volumes from float to 8.24 3827 uint32_t vl = (uint32_t)(left * (1 << 24)); 3828 uint32_t vr = (uint32_t)(right * (1 << 24)); 3829 3830 // Delegate volume control to effect in track effect chain if needed 3831 // only one effect chain can be present on DirectOutputThread, so if 3832 // there is one, the track is connected to it 3833 if (!mEffectChains.isEmpty()) { 3834 // Do not ramp volume if volume is controlled by effect 3835 mEffectChains[0]->setVolume_l(&vl, &vr); 3836 left = (float)vl / (1 << 24); 3837 right = (float)vr / (1 << 24); 3838 } 3839 mOutput->stream->set_volume(mOutput->stream, left, right); 3840 } 3841 3842 // reset retry count 3843 track->mRetryCount = kMaxTrackRetriesDirect; 3844 mActiveTrack = t; 3845 mixerStatus = MIXER_TRACKS_READY; 3846 } else { 3847 // clear effect chain input buffer if an active track underruns to avoid sending 3848 // previous audio buffer again to effects 3849 if (!mEffectChains.isEmpty()) { 3850 mEffectChains[0]->clearInputBuffer(); 3851 } 3852 3853 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3854 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3855 track->isStopped() || track->isPaused()) { 3856 // We have consumed all the buffers of this track. 3857 // Remove it from the list of active tracks. 3858 // TODO: implement behavior for compressed audio 3859 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3860 size_t framesWritten = mBytesWritten / mFrameSize; 3861 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3862 if (track->isStopped()) { 3863 track->reset(); 3864 } 3865 trackToRemove = track; 3866 } 3867 } else { 3868 // No buffers for this track. Give it a few chances to 3869 // fill a buffer, then remove it from active list. 3870 if (--(track->mRetryCount) <= 0) { 3871 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3872 trackToRemove = track; 3873 } else { 3874 mixerStatus = MIXER_TRACKS_ENABLED; 3875 } 3876 } 3877 } 3878 } 3879 3880 // FIXME merge this with similar code for removing multiple tracks 3881 // remove all the tracks that need to be... 3882 if (CC_UNLIKELY(trackToRemove != 0)) { 3883 tracksToRemove->add(trackToRemove); 3884 mActiveTracks.remove(trackToRemove); 3885 if (!mEffectChains.isEmpty()) { 3886 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3887 trackToRemove->sessionId()); 3888 mEffectChains[0]->decActiveTrackCnt(); 3889 } 3890 if (trackToRemove->isTerminated()) { 3891 removeTrack_l(trackToRemove); 3892 } 3893 } 3894 3895 return mixerStatus; 3896} 3897 3898void AudioFlinger::DirectOutputThread::threadLoop_mix() 3899{ 3900 AudioBufferProvider::Buffer buffer; 3901 size_t frameCount = mFrameCount; 3902 int8_t *curBuf = (int8_t *)mMixBuffer; 3903 // output audio to hardware 3904 while (frameCount) { 3905 buffer.frameCount = frameCount; 3906 mActiveTrack->getNextBuffer(&buffer); 3907 if (CC_UNLIKELY(buffer.raw == NULL)) { 3908 memset(curBuf, 0, frameCount * mFrameSize); 3909 break; 3910 } 3911 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3912 frameCount -= buffer.frameCount; 3913 curBuf += buffer.frameCount * mFrameSize; 3914 mActiveTrack->releaseBuffer(&buffer); 3915 } 3916 sleepTime = 0; 3917 standbyTime = systemTime() + standbyDelay; 3918 mActiveTrack.clear(); 3919 3920} 3921 3922void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3923{ 3924 if (sleepTime == 0) { 3925 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3926 sleepTime = activeSleepTime; 3927 } else { 3928 sleepTime = idleSleepTime; 3929 } 3930 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3931 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3932 sleepTime = 0; 3933 } 3934} 3935 3936// getTrackName_l() must be called with ThreadBase::mLock held 3937int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3938 int sessionId) 3939{ 3940 return 0; 3941} 3942 3943// deleteTrackName_l() must be called with ThreadBase::mLock held 3944void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3945{ 3946} 3947 3948// checkForNewParameters_l() must be called with ThreadBase::mLock held 3949bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3950{ 3951 bool reconfig = false; 3952 3953 while (!mNewParameters.isEmpty()) { 3954 status_t status = NO_ERROR; 3955 String8 keyValuePair = mNewParameters[0]; 3956 AudioParameter param = AudioParameter(keyValuePair); 3957 int value; 3958 3959 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3960 // do not accept frame count changes if tracks are open as the track buffer 3961 // size depends on frame count and correct behavior would not be garantied 3962 // if frame count is changed after track creation 3963 if (!mTracks.isEmpty()) { 3964 status = INVALID_OPERATION; 3965 } else { 3966 reconfig = true; 3967 } 3968 } 3969 if (status == NO_ERROR) { 3970 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3971 keyValuePair.string()); 3972 if (!mStandby && status == INVALID_OPERATION) { 3973 mOutput->stream->common.standby(&mOutput->stream->common); 3974 mStandby = true; 3975 mBytesWritten = 0; 3976 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3977 keyValuePair.string()); 3978 } 3979 if (status == NO_ERROR && reconfig) { 3980 readOutputParameters(); 3981 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3982 } 3983 } 3984 3985 mNewParameters.removeAt(0); 3986 3987 mParamStatus = status; 3988 mParamCond.signal(); 3989 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3990 // already timed out waiting for the status and will never signal the condition. 3991 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3992 } 3993 return reconfig; 3994} 3995 3996uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3997{ 3998 uint32_t time; 3999 if (audio_is_linear_pcm(mFormat)) { 4000 time = PlaybackThread::activeSleepTimeUs(); 4001 } else { 4002 time = 10000; 4003 } 4004 return time; 4005} 4006 4007uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4008{ 4009 uint32_t time; 4010 if (audio_is_linear_pcm(mFormat)) { 4011 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4012 } else { 4013 time = 10000; 4014 } 4015 return time; 4016} 4017 4018uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4019{ 4020 uint32_t time; 4021 if (audio_is_linear_pcm(mFormat)) { 4022 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4023 } else { 4024 time = 10000; 4025 } 4026 return time; 4027} 4028 4029void AudioFlinger::DirectOutputThread::cacheParameters_l() 4030{ 4031 PlaybackThread::cacheParameters_l(); 4032 4033 // use shorter standby delay as on normal output to release 4034 // hardware resources as soon as possible 4035 standbyDelay = microseconds(activeSleepTime*2); 4036} 4037 4038// ---------------------------------------------------------------------------- 4039 4040AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4041 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4042 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4043 DUPLICATING), 4044 mWaitTimeMs(UINT_MAX) 4045{ 4046 addOutputTrack(mainThread); 4047} 4048 4049AudioFlinger::DuplicatingThread::~DuplicatingThread() 4050{ 4051 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4052 mOutputTracks[i]->destroy(); 4053 } 4054} 4055 4056void AudioFlinger::DuplicatingThread::threadLoop_mix() 4057{ 4058 // mix buffers... 4059 if (outputsReady(outputTracks)) { 4060 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4061 } else { 4062 memset(mMixBuffer, 0, mixBufferSize); 4063 } 4064 sleepTime = 0; 4065 writeFrames = mNormalFrameCount; 4066 standbyTime = systemTime() + standbyDelay; 4067} 4068 4069void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4070{ 4071 if (sleepTime == 0) { 4072 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4073 sleepTime = activeSleepTime; 4074 } else { 4075 sleepTime = idleSleepTime; 4076 } 4077 } else if (mBytesWritten != 0) { 4078 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4079 writeFrames = mNormalFrameCount; 4080 memset(mMixBuffer, 0, mixBufferSize); 4081 } else { 4082 // flush remaining overflow buffers in output tracks 4083 writeFrames = 0; 4084 } 4085 sleepTime = 0; 4086 } 4087} 4088 4089void AudioFlinger::DuplicatingThread::threadLoop_write() 4090{ 4091 for (size_t i = 0; i < outputTracks.size(); i++) { 4092 outputTracks[i]->write(mMixBuffer, writeFrames); 4093 } 4094 mBytesWritten += mixBufferSize; 4095} 4096 4097void AudioFlinger::DuplicatingThread::threadLoop_standby() 4098{ 4099 // DuplicatingThread implements standby by stopping all tracks 4100 for (size_t i = 0; i < outputTracks.size(); i++) { 4101 outputTracks[i]->stop(); 4102 } 4103} 4104 4105void AudioFlinger::DuplicatingThread::saveOutputTracks() 4106{ 4107 outputTracks = mOutputTracks; 4108} 4109 4110void AudioFlinger::DuplicatingThread::clearOutputTracks() 4111{ 4112 outputTracks.clear(); 4113} 4114 4115void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4116{ 4117 Mutex::Autolock _l(mLock); 4118 // FIXME explain this formula 4119 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4120 OutputTrack *outputTrack = new OutputTrack(thread, 4121 this, 4122 mSampleRate, 4123 mFormat, 4124 mChannelMask, 4125 frameCount); 4126 if (outputTrack->cblk() != NULL) { 4127 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4128 mOutputTracks.add(outputTrack); 4129 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4130 updateWaitTime_l(); 4131 } 4132} 4133 4134void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4135{ 4136 Mutex::Autolock _l(mLock); 4137 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4138 if (mOutputTracks[i]->thread() == thread) { 4139 mOutputTracks[i]->destroy(); 4140 mOutputTracks.removeAt(i); 4141 updateWaitTime_l(); 4142 return; 4143 } 4144 } 4145 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4146} 4147 4148// caller must hold mLock 4149void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4150{ 4151 mWaitTimeMs = UINT_MAX; 4152 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4153 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4154 if (strong != 0) { 4155 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4156 if (waitTimeMs < mWaitTimeMs) { 4157 mWaitTimeMs = waitTimeMs; 4158 } 4159 } 4160 } 4161} 4162 4163 4164bool AudioFlinger::DuplicatingThread::outputsReady( 4165 const SortedVector< sp<OutputTrack> > &outputTracks) 4166{ 4167 for (size_t i = 0; i < outputTracks.size(); i++) { 4168 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4169 if (thread == 0) { 4170 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4171 outputTracks[i].get()); 4172 return false; 4173 } 4174 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4175 // see note at standby() declaration 4176 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4177 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4178 thread.get()); 4179 return false; 4180 } 4181 } 4182 return true; 4183} 4184 4185uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4186{ 4187 return (mWaitTimeMs * 1000) / 2; 4188} 4189 4190void AudioFlinger::DuplicatingThread::cacheParameters_l() 4191{ 4192 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4193 updateWaitTime_l(); 4194 4195 MixerThread::cacheParameters_l(); 4196} 4197 4198// ---------------------------------------------------------------------------- 4199 4200// TrackBase constructor must be called with AudioFlinger::mLock held 4201AudioFlinger::ThreadBase::TrackBase::TrackBase( 4202 ThreadBase *thread, 4203 const sp<Client>& client, 4204 uint32_t sampleRate, 4205 audio_format_t format, 4206 audio_channel_mask_t channelMask, 4207 size_t frameCount, 4208 const sp<IMemory>& sharedBuffer, 4209 int sessionId) 4210 : RefBase(), 4211 mThread(thread), 4212 mClient(client), 4213 mCblk(NULL), 4214 // mBuffer 4215 // mBufferEnd 4216 mStepCount(0), 4217 mState(IDLE), 4218 mSampleRate(sampleRate), 4219 mFormat(format), 4220 mChannelMask(channelMask), 4221 mChannelCount(popcount(channelMask)), 4222 mFrameSize(audio_is_linear_pcm(format) ? 4223 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 4224 mFrameCount(frameCount), 4225 mStepServerFailed(false), 4226 mSessionId(sessionId) 4227{ 4228 // client == 0 implies sharedBuffer == 0 4229 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4230 4231 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4232 sharedBuffer->size()); 4233 4234 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4235 size_t size = sizeof(audio_track_cblk_t); 4236 size_t bufferSize = frameCount * mFrameSize; 4237 if (sharedBuffer == 0) { 4238 size += bufferSize; 4239 } 4240 4241 if (client != 0) { 4242 mCblkMemory = client->heap()->allocate(size); 4243 if (mCblkMemory != 0) { 4244 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4245 // can't assume mCblk != NULL 4246 } else { 4247 ALOGE("not enough memory for AudioTrack size=%u", size); 4248 client->heap()->dump("AudioTrack"); 4249 return; 4250 } 4251 } else { 4252 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4253 // assume mCblk != NULL 4254 } 4255 4256 // construct the shared structure in-place. 4257 if (mCblk != NULL) { 4258 new(mCblk) audio_track_cblk_t(); 4259 // clear all buffers 4260 mCblk->frameCount_ = frameCount; 4261 mCblk->sampleRate = sampleRate; 4262// uncomment the following lines to quickly test 32-bit wraparound 4263// mCblk->user = 0xffff0000; 4264// mCblk->server = 0xffff0000; 4265// mCblk->userBase = 0xffff0000; 4266// mCblk->serverBase = 0xffff0000; 4267 if (sharedBuffer == 0) { 4268 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4269 memset(mBuffer, 0, bufferSize); 4270 // Force underrun condition to avoid false underrun callback until first data is 4271 // written to buffer (other flags are cleared) 4272 mCblk->flags = CBLK_UNDERRUN; 4273 } else { 4274 mBuffer = sharedBuffer->pointer(); 4275 } 4276 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4277 } 4278} 4279 4280AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4281{ 4282 if (mCblk != NULL) { 4283 if (mClient == 0) { 4284 delete mCblk; 4285 } else { 4286 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4287 } 4288 } 4289 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4290 if (mClient != 0) { 4291 // Client destructor must run with AudioFlinger mutex locked 4292 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4293 // If the client's reference count drops to zero, the associated destructor 4294 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4295 // relying on the automatic clear() at end of scope. 4296 mClient.clear(); 4297 } 4298} 4299 4300// AudioBufferProvider interface 4301// getNextBuffer() = 0; 4302// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4303void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4304{ 4305 buffer->raw = NULL; 4306 mStepCount = buffer->frameCount; 4307 // FIXME See note at getNextBuffer() 4308 (void) step(); // ignore return value of step() 4309 buffer->frameCount = 0; 4310} 4311 4312bool AudioFlinger::ThreadBase::TrackBase::step() { 4313 bool result; 4314 audio_track_cblk_t* cblk = this->cblk(); 4315 4316 result = cblk->stepServer(mStepCount, mFrameCount, isOut()); 4317 if (!result) { 4318 ALOGV("stepServer failed acquiring cblk mutex"); 4319 mStepServerFailed = true; 4320 } 4321 return result; 4322} 4323 4324void AudioFlinger::ThreadBase::TrackBase::reset() { 4325 audio_track_cblk_t* cblk = this->cblk(); 4326 4327 cblk->user = 0; 4328 cblk->server = 0; 4329 cblk->userBase = 0; 4330 cblk->serverBase = 0; 4331 mStepServerFailed = false; 4332 ALOGV("TrackBase::reset"); 4333} 4334 4335uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4336 return mCblk->sampleRate; 4337} 4338 4339void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4340 audio_track_cblk_t* cblk = this->cblk(); 4341 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; 4342 int8_t *bufferEnd = bufferStart + frames * mFrameSize; 4343 4344 // Check validity of returned pointer in case the track control block would have been corrupted. 4345 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4346 "TrackBase::getBuffer buffer out of range:\n" 4347 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4348 " server %u, serverBase %u, user %u, userBase %u, frameSize %u", 4349 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4350 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); 4351 4352 return bufferStart; 4353} 4354 4355status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4356{ 4357 mSyncEvents.add(event); 4358 return NO_ERROR; 4359} 4360 4361// ---------------------------------------------------------------------------- 4362 4363// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4364AudioFlinger::PlaybackThread::Track::Track( 4365 PlaybackThread *thread, 4366 const sp<Client>& client, 4367 audio_stream_type_t streamType, 4368 uint32_t sampleRate, 4369 audio_format_t format, 4370 audio_channel_mask_t channelMask, 4371 size_t frameCount, 4372 const sp<IMemory>& sharedBuffer, 4373 int sessionId, 4374 IAudioFlinger::track_flags_t flags) 4375 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4376 sessionId), 4377 mMute(false), 4378 mFillingUpStatus(FS_INVALID), 4379 // mRetryCount initialized later when needed 4380 mSharedBuffer(sharedBuffer), 4381 mStreamType(streamType), 4382 mName(-1), // see note below 4383 mMainBuffer(thread->mixBuffer()), 4384 mAuxBuffer(NULL), 4385 mAuxEffectId(0), mHasVolumeController(false), 4386 mPresentationCompleteFrames(0), 4387 mFlags(flags), 4388 mFastIndex(-1), 4389 mUnderrunCount(0), 4390 mCachedVolume(1.0) 4391{ 4392 if (mCblk != NULL) { 4393 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4394 mName = thread->getTrackName_l(channelMask, sessionId); 4395 mCblk->mName = mName; 4396 if (mName < 0) { 4397 ALOGE("no more track names available"); 4398 return; 4399 } 4400 // only allocate a fast track index if we were able to allocate a normal track name 4401 if (flags & IAudioFlinger::TRACK_FAST) { 4402 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4403 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4404 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4405 // FIXME This is too eager. We allocate a fast track index before the 4406 // fast track becomes active. Since fast tracks are a scarce resource, 4407 // this means we are potentially denying other more important fast tracks from 4408 // being created. It would be better to allocate the index dynamically. 4409 mFastIndex = i; 4410 mCblk->mName = i; 4411 // Read the initial underruns because this field is never cleared by the fast mixer 4412 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4413 thread->mFastTrackAvailMask &= ~(1 << i); 4414 } 4415 } 4416 ALOGV("Track constructor name %d, calling pid %d", mName, 4417 IPCThreadState::self()->getCallingPid()); 4418} 4419 4420AudioFlinger::PlaybackThread::Track::~Track() 4421{ 4422 ALOGV("PlaybackThread::Track destructor"); 4423} 4424 4425void AudioFlinger::PlaybackThread::Track::destroy() 4426{ 4427 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4428 // by removing it from mTracks vector, so there is a risk that this Tracks's 4429 // destructor is called. As the destructor needs to lock mLock, 4430 // we must acquire a strong reference on this Track before locking mLock 4431 // here so that the destructor is called only when exiting this function. 4432 // On the other hand, as long as Track::destroy() is only called by 4433 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4434 // this Track with its member mTrack. 4435 sp<Track> keep(this); 4436 { // scope for mLock 4437 sp<ThreadBase> thread = mThread.promote(); 4438 if (thread != 0) { 4439 if (!isOutputTrack()) { 4440 if (mState == ACTIVE || mState == RESUMING) { 4441 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4442 4443#ifdef ADD_BATTERY_DATA 4444 // to track the speaker usage 4445 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4446#endif 4447 } 4448 AudioSystem::releaseOutput(thread->id()); 4449 } 4450 Mutex::Autolock _l(thread->mLock); 4451 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4452 playbackThread->destroyTrack_l(this); 4453 } 4454 } 4455} 4456 4457/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4458{ 4459 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 4460 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4461} 4462 4463void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4464{ 4465 uint32_t vlr = mCblk->getVolumeLR(); 4466 if (isFastTrack()) { 4467 sprintf(buffer, " F %2d", mFastIndex); 4468 } else { 4469 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4470 } 4471 track_state state = mState; 4472 char stateChar; 4473 switch (state) { 4474 case IDLE: 4475 stateChar = 'I'; 4476 break; 4477 case TERMINATED: 4478 stateChar = 'T'; 4479 break; 4480 case STOPPING_1: 4481 stateChar = 's'; 4482 break; 4483 case STOPPING_2: 4484 stateChar = '5'; 4485 break; 4486 case STOPPED: 4487 stateChar = 'S'; 4488 break; 4489 case RESUMING: 4490 stateChar = 'R'; 4491 break; 4492 case ACTIVE: 4493 stateChar = 'A'; 4494 break; 4495 case PAUSING: 4496 stateChar = 'p'; 4497 break; 4498 case PAUSED: 4499 stateChar = 'P'; 4500 break; 4501 case FLUSHED: 4502 stateChar = 'F'; 4503 break; 4504 default: 4505 stateChar = '?'; 4506 break; 4507 } 4508 char nowInUnderrun; 4509 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4510 case UNDERRUN_FULL: 4511 nowInUnderrun = ' '; 4512 break; 4513 case UNDERRUN_PARTIAL: 4514 nowInUnderrun = '<'; 4515 break; 4516 case UNDERRUN_EMPTY: 4517 nowInUnderrun = '*'; 4518 break; 4519 default: 4520 nowInUnderrun = '?'; 4521 break; 4522 } 4523 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4524 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4525 (mClient == 0) ? getpid_cached : mClient->pid(), 4526 mStreamType, 4527 mFormat, 4528 mChannelMask, 4529 mSessionId, 4530 mStepCount, 4531 mFrameCount, 4532 stateChar, 4533 mMute, 4534 mFillingUpStatus, 4535 mCblk->sampleRate, 4536 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4537 20.0 * log10((vlr >> 16) / 4096.0), 4538 mCblk->server, 4539 mCblk->user, 4540 (int)mMainBuffer, 4541 (int)mAuxBuffer, 4542 mCblk->flags, 4543 mUnderrunCount, 4544 nowInUnderrun); 4545} 4546 4547// AudioBufferProvider interface 4548status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4549 AudioBufferProvider::Buffer* buffer, int64_t pts) 4550{ 4551 audio_track_cblk_t* cblk = this->cblk(); 4552 uint32_t framesReady; 4553 uint32_t framesReq = buffer->frameCount; 4554 4555 // Check if last stepServer failed, try to step now 4556 if (mStepServerFailed) { 4557 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4558 // Since the fast mixer is higher priority than client callback thread, 4559 // it does not result in priority inversion for client. 4560 // But a non-blocking solution would be preferable to avoid 4561 // fast mixer being unable to tryLock(), and 4562 // to avoid the extra context switches if the client wakes up, 4563 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4564 if (!step()) goto getNextBuffer_exit; 4565 ALOGV("stepServer recovered"); 4566 mStepServerFailed = false; 4567 } 4568 4569 // FIXME Same as above 4570 framesReady = cblk->framesReadyOut(); 4571 4572 if (CC_LIKELY(framesReady)) { 4573 uint32_t s = cblk->server; 4574 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 4575 4576 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4577 if (framesReq > framesReady) { 4578 framesReq = framesReady; 4579 } 4580 if (framesReq > bufferEnd - s) { 4581 framesReq = bufferEnd - s; 4582 } 4583 4584 buffer->raw = getBuffer(s, framesReq); 4585 buffer->frameCount = framesReq; 4586 return NO_ERROR; 4587 } 4588 4589getNextBuffer_exit: 4590 buffer->raw = NULL; 4591 buffer->frameCount = 0; 4592 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4593 return NOT_ENOUGH_DATA; 4594} 4595 4596// Note that framesReady() takes a mutex on the control block using tryLock(). 4597// This could result in priority inversion if framesReady() is called by the normal mixer, 4598// as the normal mixer thread runs at lower 4599// priority than the client's callback thread: there is a short window within framesReady() 4600// during which the normal mixer could be preempted, and the client callback would block. 4601// Another problem can occur if framesReady() is called by the fast mixer: 4602// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4603// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4604size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4605 return mCblk->framesReadyOut(); 4606} 4607 4608// Don't call for fast tracks; the framesReady() could result in priority inversion 4609bool AudioFlinger::PlaybackThread::Track::isReady() const { 4610 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 4611 return true; 4612 } 4613 4614 if (framesReady() >= mFrameCount || 4615 (mCblk->flags & CBLK_FORCEREADY)) { 4616 mFillingUpStatus = FS_FILLED; 4617 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4618 return true; 4619 } 4620 return false; 4621} 4622 4623status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4624 int triggerSession) 4625{ 4626 status_t status = NO_ERROR; 4627 ALOGV("start(%d), calling pid %d session %d", 4628 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4629 4630 sp<ThreadBase> thread = mThread.promote(); 4631 if (thread != 0) { 4632 Mutex::Autolock _l(thread->mLock); 4633 track_state state = mState; 4634 // here the track could be either new, or restarted 4635 // in both cases "unstop" the track 4636 if (mState == PAUSED) { 4637 mState = TrackBase::RESUMING; 4638 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4639 } else { 4640 mState = TrackBase::ACTIVE; 4641 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4642 } 4643 4644 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4645 thread->mLock.unlock(); 4646 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4647 thread->mLock.lock(); 4648 4649#ifdef ADD_BATTERY_DATA 4650 // to track the speaker usage 4651 if (status == NO_ERROR) { 4652 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4653 } 4654#endif 4655 } 4656 if (status == NO_ERROR) { 4657 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4658 playbackThread->addTrack_l(this); 4659 } else { 4660 mState = state; 4661 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4662 } 4663 } else { 4664 status = BAD_VALUE; 4665 } 4666 return status; 4667} 4668 4669void AudioFlinger::PlaybackThread::Track::stop() 4670{ 4671 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4672 sp<ThreadBase> thread = mThread.promote(); 4673 if (thread != 0) { 4674 Mutex::Autolock _l(thread->mLock); 4675 track_state state = mState; 4676 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4677 // If the track is not active (PAUSED and buffers full), flush buffers 4678 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4679 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4680 reset(); 4681 mState = STOPPED; 4682 } else if (!isFastTrack()) { 4683 mState = STOPPED; 4684 } else { 4685 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4686 // and then to STOPPED and reset() when presentation is complete 4687 mState = STOPPING_1; 4688 } 4689 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4690 playbackThread); 4691 } 4692 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4693 thread->mLock.unlock(); 4694 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4695 thread->mLock.lock(); 4696 4697#ifdef ADD_BATTERY_DATA 4698 // to track the speaker usage 4699 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4700#endif 4701 } 4702 } 4703} 4704 4705void AudioFlinger::PlaybackThread::Track::pause() 4706{ 4707 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4708 sp<ThreadBase> thread = mThread.promote(); 4709 if (thread != 0) { 4710 Mutex::Autolock _l(thread->mLock); 4711 if (mState == ACTIVE || mState == RESUMING) { 4712 mState = PAUSING; 4713 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4714 if (!isOutputTrack()) { 4715 thread->mLock.unlock(); 4716 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4717 thread->mLock.lock(); 4718 4719#ifdef ADD_BATTERY_DATA 4720 // to track the speaker usage 4721 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4722#endif 4723 } 4724 } 4725 } 4726} 4727 4728void AudioFlinger::PlaybackThread::Track::flush() 4729{ 4730 ALOGV("flush(%d)", mName); 4731 sp<ThreadBase> thread = mThread.promote(); 4732 if (thread != 0) { 4733 Mutex::Autolock _l(thread->mLock); 4734 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4735 mState != PAUSING && mState != IDLE && mState != FLUSHED) { 4736 return; 4737 } 4738 // No point remaining in PAUSED state after a flush => go to 4739 // FLUSHED state 4740 mState = FLUSHED; 4741 // do not reset the track if it is still in the process of being stopped or paused. 4742 // this will be done by prepareTracks_l() when the track is stopped. 4743 // prepareTracks_l() will see mState == FLUSHED, then 4744 // remove from active track list, reset(), and trigger presentation complete 4745 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4746 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4747 reset(); 4748 } 4749 } 4750} 4751 4752void AudioFlinger::PlaybackThread::Track::reset() 4753{ 4754 // Do not reset twice to avoid discarding data written just after a flush and before 4755 // the audioflinger thread detects the track is stopped. 4756 if (!mResetDone) { 4757 TrackBase::reset(); 4758 // Force underrun condition to avoid false underrun callback until first data is 4759 // written to buffer 4760 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4761 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4762 mFillingUpStatus = FS_FILLING; 4763 mResetDone = true; 4764 if (mState == FLUSHED) { 4765 mState = IDLE; 4766 } 4767 } 4768} 4769 4770void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4771{ 4772 mMute = muted; 4773} 4774 4775status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4776{ 4777 status_t status = DEAD_OBJECT; 4778 sp<ThreadBase> thread = mThread.promote(); 4779 if (thread != 0) { 4780 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4781 sp<AudioFlinger> af = mClient->audioFlinger(); 4782 4783 Mutex::Autolock _l(af->mLock); 4784 4785 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4786 4787 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4788 Mutex::Autolock _dl(playbackThread->mLock); 4789 Mutex::Autolock _sl(srcThread->mLock); 4790 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4791 if (chain == 0) { 4792 return INVALID_OPERATION; 4793 } 4794 4795 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4796 if (effect == 0) { 4797 return INVALID_OPERATION; 4798 } 4799 srcThread->removeEffect_l(effect); 4800 playbackThread->addEffect_l(effect); 4801 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4802 if (effect->state() == EffectModule::ACTIVE || 4803 effect->state() == EffectModule::STOPPING) { 4804 effect->start(); 4805 } 4806 4807 sp<EffectChain> dstChain = effect->chain().promote(); 4808 if (dstChain == 0) { 4809 srcThread->addEffect_l(effect); 4810 return INVALID_OPERATION; 4811 } 4812 AudioSystem::unregisterEffect(effect->id()); 4813 AudioSystem::registerEffect(&effect->desc(), 4814 srcThread->id(), 4815 dstChain->strategy(), 4816 AUDIO_SESSION_OUTPUT_MIX, 4817 effect->id()); 4818 } 4819 status = playbackThread->attachAuxEffect(this, EffectId); 4820 } 4821 return status; 4822} 4823 4824void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4825{ 4826 mAuxEffectId = EffectId; 4827 mAuxBuffer = buffer; 4828} 4829 4830bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4831 size_t audioHalFrames) 4832{ 4833 // a track is considered presented when the total number of frames written to audio HAL 4834 // corresponds to the number of frames written when presentationComplete() is called for the 4835 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4836 if (mPresentationCompleteFrames == 0) { 4837 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4838 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4839 mPresentationCompleteFrames, audioHalFrames); 4840 } 4841 if (framesWritten >= mPresentationCompleteFrames) { 4842 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4843 mSessionId, framesWritten); 4844 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4845 return true; 4846 } 4847 return false; 4848} 4849 4850void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4851{ 4852 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4853 if (mSyncEvents[i]->type() == type) { 4854 mSyncEvents[i]->trigger(); 4855 mSyncEvents.removeAt(i); 4856 i--; 4857 } 4858 } 4859} 4860 4861// implement VolumeBufferProvider interface 4862 4863uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4864{ 4865 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4866 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4867 uint32_t vlr = mCblk->getVolumeLR(); 4868 uint32_t vl = vlr & 0xFFFF; 4869 uint32_t vr = vlr >> 16; 4870 // track volumes come from shared memory, so can't be trusted and must be clamped 4871 if (vl > MAX_GAIN_INT) { 4872 vl = MAX_GAIN_INT; 4873 } 4874 if (vr > MAX_GAIN_INT) { 4875 vr = MAX_GAIN_INT; 4876 } 4877 // now apply the cached master volume and stream type volume; 4878 // this is trusted but lacks any synchronization or barrier so may be stale 4879 float v = mCachedVolume; 4880 vl *= v; 4881 vr *= v; 4882 // re-combine into U4.16 4883 vlr = (vr << 16) | (vl & 0xFFFF); 4884 // FIXME look at mute, pause, and stop flags 4885 return vlr; 4886} 4887 4888status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4889{ 4890 if (mState == TERMINATED || mState == PAUSED || 4891 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4892 (mState == STOPPED)))) { 4893 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4894 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4895 event->cancel(); 4896 return INVALID_OPERATION; 4897 } 4898 (void) TrackBase::setSyncEvent(event); 4899 return NO_ERROR; 4900} 4901 4902bool AudioFlinger::PlaybackThread::Track::isOut() const 4903{ 4904 return true; 4905} 4906 4907// timed audio tracks 4908 4909sp<AudioFlinger::PlaybackThread::TimedTrack> 4910AudioFlinger::PlaybackThread::TimedTrack::create( 4911 PlaybackThread *thread, 4912 const sp<Client>& client, 4913 audio_stream_type_t streamType, 4914 uint32_t sampleRate, 4915 audio_format_t format, 4916 audio_channel_mask_t channelMask, 4917 size_t frameCount, 4918 const sp<IMemory>& sharedBuffer, 4919 int sessionId) { 4920 if (!client->reserveTimedTrack()) 4921 return 0; 4922 4923 return new TimedTrack( 4924 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4925 sharedBuffer, sessionId); 4926} 4927 4928AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4929 PlaybackThread *thread, 4930 const sp<Client>& client, 4931 audio_stream_type_t streamType, 4932 uint32_t sampleRate, 4933 audio_format_t format, 4934 audio_channel_mask_t channelMask, 4935 size_t frameCount, 4936 const sp<IMemory>& sharedBuffer, 4937 int sessionId) 4938 : Track(thread, client, streamType, sampleRate, format, channelMask, 4939 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4940 mQueueHeadInFlight(false), 4941 mTrimQueueHeadOnRelease(false), 4942 mFramesPendingInQueue(0), 4943 mTimedSilenceBuffer(NULL), 4944 mTimedSilenceBufferSize(0), 4945 mTimedAudioOutputOnTime(false), 4946 mMediaTimeTransformValid(false) 4947{ 4948 LocalClock lc; 4949 mLocalTimeFreq = lc.getLocalFreq(); 4950 4951 mLocalTimeToSampleTransform.a_zero = 0; 4952 mLocalTimeToSampleTransform.b_zero = 0; 4953 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4954 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4955 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4956 &mLocalTimeToSampleTransform.a_to_b_denom); 4957 4958 mMediaTimeToSampleTransform.a_zero = 0; 4959 mMediaTimeToSampleTransform.b_zero = 0; 4960 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4961 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4962 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4963 &mMediaTimeToSampleTransform.a_to_b_denom); 4964} 4965 4966AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4967 mClient->releaseTimedTrack(); 4968 delete [] mTimedSilenceBuffer; 4969} 4970 4971status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4972 size_t size, sp<IMemory>* buffer) { 4973 4974 Mutex::Autolock _l(mTimedBufferQueueLock); 4975 4976 trimTimedBufferQueue_l(); 4977 4978 // lazily initialize the shared memory heap for timed buffers 4979 if (mTimedMemoryDealer == NULL) { 4980 const int kTimedBufferHeapSize = 512 << 10; 4981 4982 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4983 "AudioFlingerTimed"); 4984 if (mTimedMemoryDealer == NULL) 4985 return NO_MEMORY; 4986 } 4987 4988 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4989 if (newBuffer == NULL) { 4990 newBuffer = mTimedMemoryDealer->allocate(size); 4991 if (newBuffer == NULL) 4992 return NO_MEMORY; 4993 } 4994 4995 *buffer = newBuffer; 4996 return NO_ERROR; 4997} 4998 4999// caller must hold mTimedBufferQueueLock 5000void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 5001 int64_t mediaTimeNow; 5002 { 5003 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5004 if (!mMediaTimeTransformValid) 5005 return; 5006 5007 int64_t targetTimeNow; 5008 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 5009 ? mCCHelper.getCommonTime(&targetTimeNow) 5010 : mCCHelper.getLocalTime(&targetTimeNow); 5011 5012 if (OK != res) 5013 return; 5014 5015 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 5016 &mediaTimeNow)) { 5017 return; 5018 } 5019 } 5020 5021 size_t trimEnd; 5022 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 5023 int64_t bufEnd; 5024 5025 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 5026 // We have a next buffer. Just use its PTS as the PTS of the frame 5027 // following the last frame in this buffer. If the stream is sparse 5028 // (ie, there are deliberate gaps left in the stream which should be 5029 // filled with silence by the TimedAudioTrack), then this can result 5030 // in one extra buffer being left un-trimmed when it could have 5031 // been. In general, this is not typical, and we would rather 5032 // optimized away the TS calculation below for the more common case 5033 // where PTSes are contiguous. 5034 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5035 } else { 5036 // We have no next buffer. Compute the PTS of the frame following 5037 // the last frame in this buffer by computing the duration of of 5038 // this frame in media time units and adding it to the PTS of the 5039 // buffer. 5040 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5041 / mFrameSize; 5042 5043 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5044 &bufEnd)) { 5045 ALOGE("Failed to convert frame count of %lld to media time" 5046 " duration" " (scale factor %d/%u) in %s", 5047 frameCount, 5048 mMediaTimeToSampleTransform.a_to_b_numer, 5049 mMediaTimeToSampleTransform.a_to_b_denom, 5050 __PRETTY_FUNCTION__); 5051 break; 5052 } 5053 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5054 } 5055 5056 if (bufEnd > mediaTimeNow) 5057 break; 5058 5059 // Is the buffer we want to use in the middle of a mix operation right 5060 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5061 // from the mixer which should be coming back shortly. 5062 if (!trimEnd && mQueueHeadInFlight) { 5063 mTrimQueueHeadOnRelease = true; 5064 } 5065 } 5066 5067 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5068 if (trimStart < trimEnd) { 5069 // Update the bookkeeping for framesReady() 5070 for (size_t i = trimStart; i < trimEnd; ++i) { 5071 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5072 } 5073 5074 // Now actually remove the buffers from the queue. 5075 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5076 } 5077} 5078 5079void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5080 const char* logTag) { 5081 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5082 "%s called (reason \"%s\"), but timed buffer queue has no" 5083 " elements to trim.", __FUNCTION__, logTag); 5084 5085 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5086 mTimedBufferQueue.removeAt(0); 5087} 5088 5089void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5090 const TimedBuffer& buf, 5091 const char* logTag) { 5092 uint32_t bufBytes = buf.buffer()->size(); 5093 uint32_t consumedAlready = buf.position(); 5094 5095 ALOG_ASSERT(consumedAlready <= bufBytes, 5096 "Bad bookkeeping while updating frames pending. Timed buffer is" 5097 " only %u bytes long, but claims to have consumed %u" 5098 " bytes. (update reason: \"%s\")", 5099 bufBytes, consumedAlready, logTag); 5100 5101 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 5102 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5103 "Bad bookkeeping while updating frames pending. Should have at" 5104 " least %u queued frames, but we think we have only %u. (update" 5105 " reason: \"%s\")", 5106 bufFrames, mFramesPendingInQueue, logTag); 5107 5108 mFramesPendingInQueue -= bufFrames; 5109} 5110 5111status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5112 const sp<IMemory>& buffer, int64_t pts) { 5113 5114 { 5115 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5116 if (!mMediaTimeTransformValid) 5117 return INVALID_OPERATION; 5118 } 5119 5120 Mutex::Autolock _l(mTimedBufferQueueLock); 5121 5122 uint32_t bufFrames = buffer->size() / mFrameSize; 5123 mFramesPendingInQueue += bufFrames; 5124 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5125 5126 return NO_ERROR; 5127} 5128 5129status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5130 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5131 5132 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5133 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5134 target); 5135 5136 if (!(target == TimedAudioTrack::LOCAL_TIME || 5137 target == TimedAudioTrack::COMMON_TIME)) { 5138 return BAD_VALUE; 5139 } 5140 5141 Mutex::Autolock lock(mMediaTimeTransformLock); 5142 mMediaTimeTransform = xform; 5143 mMediaTimeTransformTarget = target; 5144 mMediaTimeTransformValid = true; 5145 5146 return NO_ERROR; 5147} 5148 5149#define min(a, b) ((a) < (b) ? (a) : (b)) 5150 5151// implementation of getNextBuffer for tracks whose buffers have timestamps 5152status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5153 AudioBufferProvider::Buffer* buffer, int64_t pts) 5154{ 5155 if (pts == AudioBufferProvider::kInvalidPTS) { 5156 buffer->raw = NULL; 5157 buffer->frameCount = 0; 5158 mTimedAudioOutputOnTime = false; 5159 return INVALID_OPERATION; 5160 } 5161 5162 Mutex::Autolock _l(mTimedBufferQueueLock); 5163 5164 ALOG_ASSERT(!mQueueHeadInFlight, 5165 "getNextBuffer called without releaseBuffer!"); 5166 5167 while (true) { 5168 5169 // if we have no timed buffers, then fail 5170 if (mTimedBufferQueue.isEmpty()) { 5171 buffer->raw = NULL; 5172 buffer->frameCount = 0; 5173 return NOT_ENOUGH_DATA; 5174 } 5175 5176 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5177 5178 // calculate the PTS of the head of the timed buffer queue expressed in 5179 // local time 5180 int64_t headLocalPTS; 5181 { 5182 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5183 5184 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5185 5186 if (mMediaTimeTransform.a_to_b_denom == 0) { 5187 // the transform represents a pause, so yield silence 5188 timedYieldSilence_l(buffer->frameCount, buffer); 5189 return NO_ERROR; 5190 } 5191 5192 int64_t transformedPTS; 5193 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5194 &transformedPTS)) { 5195 // the transform failed. this shouldn't happen, but if it does 5196 // then just drop this buffer 5197 ALOGW("timedGetNextBuffer transform failed"); 5198 buffer->raw = NULL; 5199 buffer->frameCount = 0; 5200 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5201 return NO_ERROR; 5202 } 5203 5204 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5205 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5206 &headLocalPTS)) { 5207 buffer->raw = NULL; 5208 buffer->frameCount = 0; 5209 return INVALID_OPERATION; 5210 } 5211 } else { 5212 headLocalPTS = transformedPTS; 5213 } 5214 } 5215 5216 // adjust the head buffer's PTS to reflect the portion of the head buffer 5217 // that has already been consumed 5218 int64_t effectivePTS = headLocalPTS + 5219 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 5220 5221 // Calculate the delta in samples between the head of the input buffer 5222 // queue and the start of the next output buffer that will be written. 5223 // If the transformation fails because of over or underflow, it means 5224 // that the sample's position in the output stream is so far out of 5225 // whack that it should just be dropped. 5226 int64_t sampleDelta; 5227 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5228 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5229 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5230 " mix"); 5231 continue; 5232 } 5233 if (!mLocalTimeToSampleTransform.doForwardTransform( 5234 (effectivePTS - pts) << 32, &sampleDelta)) { 5235 ALOGV("*** too late during sample rate transform: dropped buffer"); 5236 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5237 continue; 5238 } 5239 5240 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5241 " sampleDelta=[%d.%08x]", 5242 head.pts(), head.position(), pts, 5243 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5244 + (sampleDelta >> 32)), 5245 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5246 5247 // if the delta between the ideal placement for the next input sample and 5248 // the current output position is within this threshold, then we will 5249 // concatenate the next input samples to the previous output 5250 const int64_t kSampleContinuityThreshold = 5251 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5252 5253 // if this is the first buffer of audio that we're emitting from this track 5254 // then it should be almost exactly on time. 5255 const int64_t kSampleStartupThreshold = 1LL << 32; 5256 5257 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5258 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5259 // the next input is close enough to being on time, so concatenate it 5260 // with the last output 5261 timedYieldSamples_l(buffer); 5262 5263 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5264 head.position(), buffer->frameCount); 5265 return NO_ERROR; 5266 } 5267 5268 // Looks like our output is not on time. Reset our on timed status. 5269 // Next time we mix samples from our input queue, then should be within 5270 // the StartupThreshold. 5271 mTimedAudioOutputOnTime = false; 5272 if (sampleDelta > 0) { 5273 // the gap between the current output position and the proper start of 5274 // the next input sample is too big, so fill it with silence 5275 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5276 5277 timedYieldSilence_l(framesUntilNextInput, buffer); 5278 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5279 return NO_ERROR; 5280 } else { 5281 // the next input sample is late 5282 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5283 size_t onTimeSamplePosition = 5284 head.position() + lateFrames * mFrameSize; 5285 5286 if (onTimeSamplePosition > head.buffer()->size()) { 5287 // all the remaining samples in the head are too late, so 5288 // drop it and move on 5289 ALOGV("*** too late: dropped buffer"); 5290 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5291 continue; 5292 } else { 5293 // skip over the late samples 5294 head.setPosition(onTimeSamplePosition); 5295 5296 // yield the available samples 5297 timedYieldSamples_l(buffer); 5298 5299 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5300 return NO_ERROR; 5301 } 5302 } 5303 } 5304} 5305 5306// Yield samples from the timed buffer queue head up to the given output 5307// buffer's capacity. 5308// 5309// Caller must hold mTimedBufferQueueLock 5310void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5311 AudioBufferProvider::Buffer* buffer) { 5312 5313 const TimedBuffer& head = mTimedBufferQueue[0]; 5314 5315 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5316 head.position()); 5317 5318 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5319 mFrameSize); 5320 size_t framesRequested = buffer->frameCount; 5321 buffer->frameCount = min(framesLeftInHead, framesRequested); 5322 5323 mQueueHeadInFlight = true; 5324 mTimedAudioOutputOnTime = true; 5325} 5326 5327// Yield samples of silence up to the given output buffer's capacity 5328// 5329// Caller must hold mTimedBufferQueueLock 5330void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5331 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5332 5333 // lazily allocate a buffer filled with silence 5334 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 5335 delete [] mTimedSilenceBuffer; 5336 mTimedSilenceBufferSize = numFrames * mFrameSize; 5337 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5338 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5339 } 5340 5341 buffer->raw = mTimedSilenceBuffer; 5342 size_t framesRequested = buffer->frameCount; 5343 buffer->frameCount = min(numFrames, framesRequested); 5344 5345 mTimedAudioOutputOnTime = false; 5346} 5347 5348// AudioBufferProvider interface 5349void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5350 AudioBufferProvider::Buffer* buffer) { 5351 5352 Mutex::Autolock _l(mTimedBufferQueueLock); 5353 5354 // If the buffer which was just released is part of the buffer at the head 5355 // of the queue, be sure to update the amt of the buffer which has been 5356 // consumed. If the buffer being returned is not part of the head of the 5357 // queue, its either because the buffer is part of the silence buffer, or 5358 // because the head of the timed queue was trimmed after the mixer called 5359 // getNextBuffer but before the mixer called releaseBuffer. 5360 if (buffer->raw == mTimedSilenceBuffer) { 5361 ALOG_ASSERT(!mQueueHeadInFlight, 5362 "Queue head in flight during release of silence buffer!"); 5363 goto done; 5364 } 5365 5366 ALOG_ASSERT(mQueueHeadInFlight, 5367 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5368 " head in flight."); 5369 5370 if (mTimedBufferQueue.size()) { 5371 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5372 5373 void* start = head.buffer()->pointer(); 5374 void* end = reinterpret_cast<void*>( 5375 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5376 + head.buffer()->size()); 5377 5378 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5379 "released buffer not within the head of the timed buffer" 5380 " queue; qHead = [%p, %p], released buffer = %p", 5381 start, end, buffer->raw); 5382 5383 head.setPosition(head.position() + 5384 (buffer->frameCount * mFrameSize)); 5385 mQueueHeadInFlight = false; 5386 5387 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5388 "Bad bookkeeping during releaseBuffer! Should have at" 5389 " least %u queued frames, but we think we have only %u", 5390 buffer->frameCount, mFramesPendingInQueue); 5391 5392 mFramesPendingInQueue -= buffer->frameCount; 5393 5394 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5395 || mTrimQueueHeadOnRelease) { 5396 trimTimedBufferQueueHead_l("releaseBuffer"); 5397 mTrimQueueHeadOnRelease = false; 5398 } 5399 } else { 5400 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5401 " buffers in the timed buffer queue"); 5402 } 5403 5404done: 5405 buffer->raw = 0; 5406 buffer->frameCount = 0; 5407} 5408 5409size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5410 Mutex::Autolock _l(mTimedBufferQueueLock); 5411 return mFramesPendingInQueue; 5412} 5413 5414AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5415 : mPTS(0), mPosition(0) {} 5416 5417AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5418 const sp<IMemory>& buffer, int64_t pts) 5419 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5420 5421// ---------------------------------------------------------------------------- 5422 5423// RecordTrack constructor must be called with AudioFlinger::mLock held 5424AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5425 RecordThread *thread, 5426 const sp<Client>& client, 5427 uint32_t sampleRate, 5428 audio_format_t format, 5429 audio_channel_mask_t channelMask, 5430 size_t frameCount, 5431 int sessionId) 5432 : TrackBase(thread, client, sampleRate, format, 5433 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5434 mOverflow(false) 5435{ 5436 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5437} 5438 5439AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5440{ 5441 ALOGV("%s", __func__); 5442} 5443 5444// AudioBufferProvider interface 5445status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5446 int64_t pts) 5447{ 5448 audio_track_cblk_t* cblk = this->cblk(); 5449 uint32_t framesAvail; 5450 uint32_t framesReq = buffer->frameCount; 5451 5452 // Check if last stepServer failed, try to step now 5453 if (mStepServerFailed) { 5454 if (!step()) { 5455 goto getNextBuffer_exit; 5456 } 5457 ALOGV("stepServer recovered"); 5458 mStepServerFailed = false; 5459 } 5460 5461 // FIXME lock is not actually held, so overrun is possible 5462 framesAvail = cblk->framesAvailableIn_l(mFrameCount); 5463 5464 if (CC_LIKELY(framesAvail)) { 5465 uint32_t s = cblk->server; 5466 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 5467 5468 if (framesReq > framesAvail) { 5469 framesReq = framesAvail; 5470 } 5471 if (framesReq > bufferEnd - s) { 5472 framesReq = bufferEnd - s; 5473 } 5474 5475 buffer->raw = getBuffer(s, framesReq); 5476 buffer->frameCount = framesReq; 5477 return NO_ERROR; 5478 } 5479 5480getNextBuffer_exit: 5481 buffer->raw = NULL; 5482 buffer->frameCount = 0; 5483 return NOT_ENOUGH_DATA; 5484} 5485 5486status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5487 int triggerSession) 5488{ 5489 sp<ThreadBase> thread = mThread.promote(); 5490 if (thread != 0) { 5491 RecordThread *recordThread = (RecordThread *)thread.get(); 5492 return recordThread->start(this, event, triggerSession); 5493 } else { 5494 return BAD_VALUE; 5495 } 5496} 5497 5498void AudioFlinger::RecordThread::RecordTrack::stop() 5499{ 5500 sp<ThreadBase> thread = mThread.promote(); 5501 if (thread != 0) { 5502 RecordThread *recordThread = (RecordThread *)thread.get(); 5503 recordThread->mLock.lock(); 5504 bool doStop = recordThread->stop_l(this); 5505 if (doStop) { 5506 TrackBase::reset(); 5507 // Force overrun condition to avoid false overrun callback until first data is 5508 // read from buffer 5509 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5510 } 5511 recordThread->mLock.unlock(); 5512 if (doStop) { 5513 AudioSystem::stopInput(recordThread->id()); 5514 } 5515 } 5516} 5517 5518/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5519{ 5520 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 5521} 5522 5523void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5524{ 5525 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5526 (mClient == 0) ? getpid_cached : mClient->pid(), 5527 mFormat, 5528 mChannelMask, 5529 mSessionId, 5530 mStepCount, 5531 mState, 5532 mCblk->sampleRate, 5533 mCblk->server, 5534 mCblk->user, 5535 mFrameCount); 5536} 5537 5538bool AudioFlinger::RecordThread::RecordTrack::isOut() const 5539{ 5540 return false; 5541} 5542 5543// ---------------------------------------------------------------------------- 5544 5545AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5546 PlaybackThread *playbackThread, 5547 DuplicatingThread *sourceThread, 5548 uint32_t sampleRate, 5549 audio_format_t format, 5550 audio_channel_mask_t channelMask, 5551 size_t frameCount) 5552 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5553 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5554 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 5555{ 5556 5557 if (mCblk != NULL) { 5558 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5559 mOutBuffer.frameCount = 0; 5560 playbackThread->mTracks.add(this); 5561 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \ 5562 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 5563 mCblk, mBuffer, mBuffers, 5564 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5565 } else { 5566 ALOGW("Error creating output track on thread %p", playbackThread); 5567 } 5568} 5569 5570AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5571{ 5572 clearBufferQueue(); 5573} 5574 5575status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5576 int triggerSession) 5577{ 5578 status_t status = Track::start(event, triggerSession); 5579 if (status != NO_ERROR) { 5580 return status; 5581 } 5582 5583 mActive = true; 5584 mRetryCount = 127; 5585 return status; 5586} 5587 5588void AudioFlinger::PlaybackThread::OutputTrack::stop() 5589{ 5590 Track::stop(); 5591 clearBufferQueue(); 5592 mOutBuffer.frameCount = 0; 5593 mActive = false; 5594} 5595 5596bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5597{ 5598 Buffer *pInBuffer; 5599 Buffer inBuffer; 5600 uint32_t channelCount = mChannelCount; 5601 bool outputBufferFull = false; 5602 inBuffer.frameCount = frames; 5603 inBuffer.i16 = data; 5604 5605 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5606 5607 if (!mActive && frames != 0) { 5608 start(); 5609 sp<ThreadBase> thread = mThread.promote(); 5610 if (thread != 0) { 5611 MixerThread *mixerThread = (MixerThread *)thread.get(); 5612 if (mFrameCount > frames){ 5613 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5614 uint32_t startFrames = (mFrameCount - frames); 5615 pInBuffer = new Buffer; 5616 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5617 pInBuffer->frameCount = startFrames; 5618 pInBuffer->i16 = pInBuffer->mBuffer; 5619 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5620 mBufferQueue.add(pInBuffer); 5621 } else { 5622 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5623 } 5624 } 5625 } 5626 } 5627 5628 while (waitTimeLeftMs) { 5629 // First write pending buffers, then new data 5630 if (mBufferQueue.size()) { 5631 pInBuffer = mBufferQueue.itemAt(0); 5632 } else { 5633 pInBuffer = &inBuffer; 5634 } 5635 5636 if (pInBuffer->frameCount == 0) { 5637 break; 5638 } 5639 5640 if (mOutBuffer.frameCount == 0) { 5641 mOutBuffer.frameCount = pInBuffer->frameCount; 5642 nsecs_t startTime = systemTime(); 5643 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5644 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5645 mThread.unsafe_get()); 5646 outputBufferFull = true; 5647 break; 5648 } 5649 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5650 if (waitTimeLeftMs >= waitTimeMs) { 5651 waitTimeLeftMs -= waitTimeMs; 5652 } else { 5653 waitTimeLeftMs = 0; 5654 } 5655 } 5656 5657 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5658 pInBuffer->frameCount; 5659 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5660 mCblk->stepUserOut(outFrames, mFrameCount); 5661 pInBuffer->frameCount -= outFrames; 5662 pInBuffer->i16 += outFrames * channelCount; 5663 mOutBuffer.frameCount -= outFrames; 5664 mOutBuffer.i16 += outFrames * channelCount; 5665 5666 if (pInBuffer->frameCount == 0) { 5667 if (mBufferQueue.size()) { 5668 mBufferQueue.removeAt(0); 5669 delete [] pInBuffer->mBuffer; 5670 delete pInBuffer; 5671 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5672 mThread.unsafe_get(), mBufferQueue.size()); 5673 } else { 5674 break; 5675 } 5676 } 5677 } 5678 5679 // If we could not write all frames, allocate a buffer and queue it for next time. 5680 if (inBuffer.frameCount) { 5681 sp<ThreadBase> thread = mThread.promote(); 5682 if (thread != 0 && !thread->standby()) { 5683 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5684 pInBuffer = new Buffer; 5685 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5686 pInBuffer->frameCount = inBuffer.frameCount; 5687 pInBuffer->i16 = pInBuffer->mBuffer; 5688 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5689 sizeof(int16_t)); 5690 mBufferQueue.add(pInBuffer); 5691 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5692 mThread.unsafe_get(), mBufferQueue.size()); 5693 } else { 5694 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5695 mThread.unsafe_get(), this); 5696 } 5697 } 5698 } 5699 5700 // Calling write() with a 0 length buffer, means that no more data will be written: 5701 // If no more buffers are pending, fill output track buffer to make sure it is started 5702 // by output mixer. 5703 if (frames == 0 && mBufferQueue.size() == 0) { 5704 if (mCblk->user < mFrameCount) { 5705 frames = mFrameCount - mCblk->user; 5706 pInBuffer = new Buffer; 5707 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5708 pInBuffer->frameCount = frames; 5709 pInBuffer->i16 = pInBuffer->mBuffer; 5710 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5711 mBufferQueue.add(pInBuffer); 5712 } else if (mActive) { 5713 stop(); 5714 } 5715 } 5716 5717 return outputBufferFull; 5718} 5719 5720status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5721 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5722{ 5723 int active; 5724 status_t result; 5725 audio_track_cblk_t* cblk = mCblk; 5726 uint32_t framesReq = buffer->frameCount; 5727 5728 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5729 buffer->frameCount = 0; 5730 5731 uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); 5732 5733 5734 if (framesAvail == 0) { 5735 Mutex::Autolock _l(cblk->lock); 5736 goto start_loop_here; 5737 while (framesAvail == 0) { 5738 active = mActive; 5739 if (CC_UNLIKELY(!active)) { 5740 ALOGV("Not active and NO_MORE_BUFFERS"); 5741 return NO_MORE_BUFFERS; 5742 } 5743 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5744 if (result != NO_ERROR) { 5745 return NO_MORE_BUFFERS; 5746 } 5747 // read the server count again 5748 start_loop_here: 5749 framesAvail = cblk->framesAvailableOut_l(mFrameCount); 5750 } 5751 } 5752 5753// if (framesAvail < framesReq) { 5754// return NO_MORE_BUFFERS; 5755// } 5756 5757 if (framesReq > framesAvail) { 5758 framesReq = framesAvail; 5759 } 5760 5761 uint32_t u = cblk->user; 5762 uint32_t bufferEnd = cblk->userBase + mFrameCount; 5763 5764 if (framesReq > bufferEnd - u) { 5765 framesReq = bufferEnd - u; 5766 } 5767 5768 buffer->frameCount = framesReq; 5769 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 5770 return NO_ERROR; 5771} 5772 5773 5774void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5775{ 5776 size_t size = mBufferQueue.size(); 5777 5778 for (size_t i = 0; i < size; i++) { 5779 Buffer *pBuffer = mBufferQueue.itemAt(i); 5780 delete [] pBuffer->mBuffer; 5781 delete pBuffer; 5782 } 5783 mBufferQueue.clear(); 5784} 5785 5786// ---------------------------------------------------------------------------- 5787 5788AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5789 : RefBase(), 5790 mAudioFlinger(audioFlinger), 5791 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5792 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5793 mPid(pid), 5794 mTimedTrackCount(0) 5795{ 5796 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5797} 5798 5799// Client destructor must be called with AudioFlinger::mLock held 5800AudioFlinger::Client::~Client() 5801{ 5802 mAudioFlinger->removeClient_l(mPid); 5803} 5804 5805sp<MemoryDealer> AudioFlinger::Client::heap() const 5806{ 5807 return mMemoryDealer; 5808} 5809 5810// Reserve one of the limited slots for a timed audio track associated 5811// with this client 5812bool AudioFlinger::Client::reserveTimedTrack() 5813{ 5814 const int kMaxTimedTracksPerClient = 4; 5815 5816 Mutex::Autolock _l(mTimedTrackLock); 5817 5818 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5819 ALOGW("can not create timed track - pid %d has exceeded the limit", 5820 mPid); 5821 return false; 5822 } 5823 5824 mTimedTrackCount++; 5825 return true; 5826} 5827 5828// Release a slot for a timed audio track 5829void AudioFlinger::Client::releaseTimedTrack() 5830{ 5831 Mutex::Autolock _l(mTimedTrackLock); 5832 mTimedTrackCount--; 5833} 5834 5835// ---------------------------------------------------------------------------- 5836 5837AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5838 const sp<IAudioFlingerClient>& client, 5839 pid_t pid) 5840 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5841{ 5842} 5843 5844AudioFlinger::NotificationClient::~NotificationClient() 5845{ 5846} 5847 5848void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5849{ 5850 sp<NotificationClient> keep(this); 5851 mAudioFlinger->removeNotificationClient(mPid); 5852} 5853 5854// ---------------------------------------------------------------------------- 5855 5856AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5857 : BnAudioTrack(), 5858 mTrack(track) 5859{ 5860} 5861 5862AudioFlinger::TrackHandle::~TrackHandle() { 5863 // just stop the track on deletion, associated resources 5864 // will be freed from the main thread once all pending buffers have 5865 // been played. Unless it's not in the active track list, in which 5866 // case we free everything now... 5867 mTrack->destroy(); 5868} 5869 5870sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5871 return mTrack->getCblk(); 5872} 5873 5874status_t AudioFlinger::TrackHandle::start() { 5875 return mTrack->start(); 5876} 5877 5878void AudioFlinger::TrackHandle::stop() { 5879 mTrack->stop(); 5880} 5881 5882void AudioFlinger::TrackHandle::flush() { 5883 mTrack->flush(); 5884} 5885 5886void AudioFlinger::TrackHandle::mute(bool e) { 5887 mTrack->mute(e); 5888} 5889 5890void AudioFlinger::TrackHandle::pause() { 5891 mTrack->pause(); 5892} 5893 5894status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5895{ 5896 return mTrack->attachAuxEffect(EffectId); 5897} 5898 5899status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5900 sp<IMemory>* buffer) { 5901 if (!mTrack->isTimedTrack()) 5902 return INVALID_OPERATION; 5903 5904 PlaybackThread::TimedTrack* tt = 5905 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5906 return tt->allocateTimedBuffer(size, buffer); 5907} 5908 5909status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5910 int64_t pts) { 5911 if (!mTrack->isTimedTrack()) 5912 return INVALID_OPERATION; 5913 5914 PlaybackThread::TimedTrack* tt = 5915 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5916 return tt->queueTimedBuffer(buffer, pts); 5917} 5918 5919status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5920 const LinearTransform& xform, int target) { 5921 5922 if (!mTrack->isTimedTrack()) 5923 return INVALID_OPERATION; 5924 5925 PlaybackThread::TimedTrack* tt = 5926 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5927 return tt->setMediaTimeTransform( 5928 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5929} 5930 5931status_t AudioFlinger::TrackHandle::onTransact( 5932 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5933{ 5934 return BnAudioTrack::onTransact(code, data, reply, flags); 5935} 5936 5937// ---------------------------------------------------------------------------- 5938 5939sp<IAudioRecord> AudioFlinger::openRecord( 5940 pid_t pid, 5941 audio_io_handle_t input, 5942 uint32_t sampleRate, 5943 audio_format_t format, 5944 audio_channel_mask_t channelMask, 5945 size_t frameCount, 5946 IAudioFlinger::track_flags_t flags, 5947 pid_t tid, 5948 int *sessionId, 5949 status_t *status) 5950{ 5951 sp<RecordThread::RecordTrack> recordTrack; 5952 sp<RecordHandle> recordHandle; 5953 sp<Client> client; 5954 status_t lStatus; 5955 RecordThread *thread; 5956 size_t inFrameCount; 5957 int lSessionId; 5958 5959 // check calling permissions 5960 if (!recordingAllowed()) { 5961 lStatus = PERMISSION_DENIED; 5962 goto Exit; 5963 } 5964 5965 // add client to list 5966 { // scope for mLock 5967 Mutex::Autolock _l(mLock); 5968 thread = checkRecordThread_l(input); 5969 if (thread == NULL) { 5970 lStatus = BAD_VALUE; 5971 goto Exit; 5972 } 5973 5974 client = registerPid_l(pid); 5975 5976 // If no audio session id is provided, create one here 5977 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5978 lSessionId = *sessionId; 5979 } else { 5980 lSessionId = nextUniqueId(); 5981 if (sessionId != NULL) { 5982 *sessionId = lSessionId; 5983 } 5984 } 5985 // create new record track. 5986 // The record track uses one track in mHardwareMixerThread by convention. 5987 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5988 frameCount, lSessionId, flags, tid, &lStatus); 5989 } 5990 if (lStatus != NO_ERROR) { 5991 // remove local strong reference to Client before deleting the RecordTrack so that the 5992 // Client destructor is called by the TrackBase destructor with mLock held 5993 client.clear(); 5994 recordTrack.clear(); 5995 goto Exit; 5996 } 5997 5998 // return to handle to client 5999 recordHandle = new RecordHandle(recordTrack); 6000 lStatus = NO_ERROR; 6001 6002Exit: 6003 if (status) { 6004 *status = lStatus; 6005 } 6006 return recordHandle; 6007} 6008 6009// ---------------------------------------------------------------------------- 6010 6011AudioFlinger::RecordHandle::RecordHandle( 6012 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 6013 : BnAudioRecord(), 6014 mRecordTrack(recordTrack) 6015{ 6016} 6017 6018AudioFlinger::RecordHandle::~RecordHandle() { 6019 stop_nonvirtual(); 6020 mRecordTrack->destroy(); 6021} 6022 6023sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 6024 return mRecordTrack->getCblk(); 6025} 6026 6027status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6028 int triggerSession) { 6029 ALOGV("RecordHandle::start()"); 6030 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6031} 6032 6033void AudioFlinger::RecordHandle::stop() { 6034 stop_nonvirtual(); 6035} 6036 6037void AudioFlinger::RecordHandle::stop_nonvirtual() { 6038 ALOGV("RecordHandle::stop()"); 6039 mRecordTrack->stop(); 6040} 6041 6042status_t AudioFlinger::RecordHandle::onTransact( 6043 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6044{ 6045 return BnAudioRecord::onTransact(code, data, reply, flags); 6046} 6047 6048// ---------------------------------------------------------------------------- 6049 6050AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6051 AudioStreamIn *input, 6052 uint32_t sampleRate, 6053 audio_channel_mask_t channelMask, 6054 audio_io_handle_t id, 6055 audio_devices_t device, 6056 const sp<NBAIO_Sink>& teeSink) : 6057 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6058 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6059 // mRsmpInIndex and mInputBytes set by readInputParameters() 6060 mReqChannelCount(popcount(channelMask)), 6061 mReqSampleRate(sampleRate), 6062 // mBytesRead is only meaningful while active, and so is cleared in start() 6063 // (but might be better to also clear here for dump?) 6064 mTeeSink(teeSink) 6065{ 6066 snprintf(mName, kNameLength, "AudioIn_%X", id); 6067 6068 readInputParameters(); 6069 6070} 6071 6072 6073AudioFlinger::RecordThread::~RecordThread() 6074{ 6075 delete[] mRsmpInBuffer; 6076 delete mResampler; 6077 delete[] mRsmpOutBuffer; 6078} 6079 6080void AudioFlinger::RecordThread::onFirstRef() 6081{ 6082 run(mName, PRIORITY_URGENT_AUDIO); 6083} 6084 6085status_t AudioFlinger::RecordThread::readyToRun() 6086{ 6087 status_t status = initCheck(); 6088 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6089 return status; 6090} 6091 6092bool AudioFlinger::RecordThread::threadLoop() 6093{ 6094 AudioBufferProvider::Buffer buffer; 6095 sp<RecordTrack> activeTrack; 6096 Vector< sp<EffectChain> > effectChains; 6097 6098 nsecs_t lastWarning = 0; 6099 6100 inputStandBy(); 6101 acquireWakeLock(); 6102 6103 // used to verify we've read at least once before evaluating how many bytes were read 6104 bool readOnce = false; 6105 6106 // start recording 6107 while (!exitPending()) { 6108 6109 processConfigEvents(); 6110 6111 { // scope for mLock 6112 Mutex::Autolock _l(mLock); 6113 checkForNewParameters_l(); 6114 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6115 standby(); 6116 6117 if (exitPending()) { 6118 break; 6119 } 6120 6121 releaseWakeLock_l(); 6122 ALOGV("RecordThread: loop stopping"); 6123 // go to sleep 6124 mWaitWorkCV.wait(mLock); 6125 ALOGV("RecordThread: loop starting"); 6126 acquireWakeLock_l(); 6127 continue; 6128 } 6129 if (mActiveTrack != 0) { 6130 if (mActiveTrack->mState == TrackBase::PAUSING) { 6131 standby(); 6132 mActiveTrack.clear(); 6133 mStartStopCond.broadcast(); 6134 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6135 if (mReqChannelCount != mActiveTrack->channelCount()) { 6136 mActiveTrack.clear(); 6137 mStartStopCond.broadcast(); 6138 } else if (readOnce) { 6139 // record start succeeds only if first read from audio input 6140 // succeeds 6141 if (mBytesRead >= 0) { 6142 mActiveTrack->mState = TrackBase::ACTIVE; 6143 } else { 6144 mActiveTrack.clear(); 6145 } 6146 mStartStopCond.broadcast(); 6147 } 6148 mStandby = false; 6149 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6150 removeTrack_l(mActiveTrack); 6151 mActiveTrack.clear(); 6152 } 6153 } 6154 lockEffectChains_l(effectChains); 6155 } 6156 6157 if (mActiveTrack != 0) { 6158 if (mActiveTrack->mState != TrackBase::ACTIVE && 6159 mActiveTrack->mState != TrackBase::RESUMING) { 6160 unlockEffectChains(effectChains); 6161 usleep(kRecordThreadSleepUs); 6162 continue; 6163 } 6164 for (size_t i = 0; i < effectChains.size(); i ++) { 6165 effectChains[i]->process_l(); 6166 } 6167 6168 buffer.frameCount = mFrameCount; 6169 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6170 readOnce = true; 6171 size_t framesOut = buffer.frameCount; 6172 if (mResampler == NULL) { 6173 // no resampling 6174 while (framesOut) { 6175 size_t framesIn = mFrameCount - mRsmpInIndex; 6176 if (framesIn) { 6177 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6178 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6179 mActiveTrack->mFrameSize; 6180 if (framesIn > framesOut) 6181 framesIn = framesOut; 6182 mRsmpInIndex += framesIn; 6183 framesOut -= framesIn; 6184 if (mChannelCount == mReqChannelCount || 6185 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6186 memcpy(dst, src, framesIn * mFrameSize); 6187 } else { 6188 if (mChannelCount == 1) { 6189 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6190 (int16_t *)src, framesIn); 6191 } else { 6192 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6193 (int16_t *)src, framesIn); 6194 } 6195 } 6196 } 6197 if (framesOut && mFrameCount == mRsmpInIndex) { 6198 void *readInto; 6199 if (framesOut == mFrameCount && 6200 (mChannelCount == mReqChannelCount || 6201 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6202 readInto = buffer.raw; 6203 framesOut = 0; 6204 } else { 6205 readInto = mRsmpInBuffer; 6206 mRsmpInIndex = 0; 6207 } 6208 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6209 if (mBytesRead <= 0) { 6210 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6211 { 6212 ALOGE("Error reading audio input"); 6213 // Force input into standby so that it tries to 6214 // recover at next read attempt 6215 inputStandBy(); 6216 usleep(kRecordThreadSleepUs); 6217 } 6218 mRsmpInIndex = mFrameCount; 6219 framesOut = 0; 6220 buffer.frameCount = 0; 6221 } else if (mTeeSink != 0) { 6222 (void) mTeeSink->write(readInto, 6223 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6224 } 6225 } 6226 } 6227 } else { 6228 // resampling 6229 6230 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6231 // alter output frame count as if we were expecting stereo samples 6232 if (mChannelCount == 1 && mReqChannelCount == 1) { 6233 framesOut >>= 1; 6234 } 6235 mResampler->resample(mRsmpOutBuffer, framesOut, 6236 this /* AudioBufferProvider* */); 6237 // ditherAndClamp() works as long as all buffers returned by 6238 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6239 if (mChannelCount == 2 && mReqChannelCount == 1) { 6240 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6241 // the resampler always outputs stereo samples: 6242 // do post stereo to mono conversion 6243 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6244 framesOut); 6245 } else { 6246 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6247 } 6248 6249 } 6250 if (mFramestoDrop == 0) { 6251 mActiveTrack->releaseBuffer(&buffer); 6252 } else { 6253 if (mFramestoDrop > 0) { 6254 mFramestoDrop -= buffer.frameCount; 6255 if (mFramestoDrop <= 0) { 6256 clearSyncStartEvent(); 6257 } 6258 } else { 6259 mFramestoDrop += buffer.frameCount; 6260 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6261 mSyncStartEvent->isCancelled()) { 6262 ALOGW("Synced record %s, session %d, trigger session %d", 6263 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6264 mActiveTrack->sessionId(), 6265 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6266 clearSyncStartEvent(); 6267 } 6268 } 6269 } 6270 mActiveTrack->clearOverflow(); 6271 } 6272 // client isn't retrieving buffers fast enough 6273 else { 6274 if (!mActiveTrack->setOverflow()) { 6275 nsecs_t now = systemTime(); 6276 if ((now - lastWarning) > kWarningThrottleNs) { 6277 ALOGW("RecordThread: buffer overflow"); 6278 lastWarning = now; 6279 } 6280 } 6281 // Release the processor for a while before asking for a new buffer. 6282 // This will give the application more chance to read from the buffer and 6283 // clear the overflow. 6284 usleep(kRecordThreadSleepUs); 6285 } 6286 } 6287 // enable changes in effect chain 6288 unlockEffectChains(effectChains); 6289 effectChains.clear(); 6290 } 6291 6292 standby(); 6293 6294 { 6295 Mutex::Autolock _l(mLock); 6296 mActiveTrack.clear(); 6297 mStartStopCond.broadcast(); 6298 } 6299 6300 releaseWakeLock(); 6301 6302 ALOGV("RecordThread %p exiting", this); 6303 return false; 6304} 6305 6306void AudioFlinger::RecordThread::standby() 6307{ 6308 if (!mStandby) { 6309 inputStandBy(); 6310 mStandby = true; 6311 } 6312} 6313 6314void AudioFlinger::RecordThread::inputStandBy() 6315{ 6316 mInput->stream->common.standby(&mInput->stream->common); 6317} 6318 6319sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6320 const sp<AudioFlinger::Client>& client, 6321 uint32_t sampleRate, 6322 audio_format_t format, 6323 audio_channel_mask_t channelMask, 6324 size_t frameCount, 6325 int sessionId, 6326 IAudioFlinger::track_flags_t flags, 6327 pid_t tid, 6328 status_t *status) 6329{ 6330 sp<RecordTrack> track; 6331 status_t lStatus; 6332 6333 lStatus = initCheck(); 6334 if (lStatus != NO_ERROR) { 6335 ALOGE("Audio driver not initialized."); 6336 goto Exit; 6337 } 6338 6339 // FIXME use flags and tid similar to createTrack_l() 6340 6341 { // scope for mLock 6342 Mutex::Autolock _l(mLock); 6343 6344 track = new RecordTrack(this, client, sampleRate, 6345 format, channelMask, frameCount, sessionId); 6346 6347 if (track->getCblk() == 0) { 6348 lStatus = NO_MEMORY; 6349 goto Exit; 6350 } 6351 mTracks.add(track); 6352 6353 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6354 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6355 mAudioFlinger->btNrecIsOff(); 6356 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6357 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6358 } 6359 lStatus = NO_ERROR; 6360 6361Exit: 6362 if (status) { 6363 *status = lStatus; 6364 } 6365 return track; 6366} 6367 6368status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6369 AudioSystem::sync_event_t event, 6370 int triggerSession) 6371{ 6372 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6373 sp<ThreadBase> strongMe = this; 6374 status_t status = NO_ERROR; 6375 6376 if (event == AudioSystem::SYNC_EVENT_NONE) { 6377 clearSyncStartEvent(); 6378 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6379 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6380 triggerSession, 6381 recordTrack->sessionId(), 6382 syncStartEventCallback, 6383 this); 6384 // Sync event can be cancelled by the trigger session if the track is not in a 6385 // compatible state in which case we start record immediately 6386 if (mSyncStartEvent->isCancelled()) { 6387 clearSyncStartEvent(); 6388 } else { 6389 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6390 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6391 } 6392 } 6393 6394 { 6395 AutoMutex lock(mLock); 6396 if (mActiveTrack != 0) { 6397 if (recordTrack != mActiveTrack.get()) { 6398 status = -EBUSY; 6399 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6400 mActiveTrack->mState = TrackBase::ACTIVE; 6401 } 6402 return status; 6403 } 6404 6405 recordTrack->mState = TrackBase::IDLE; 6406 mActiveTrack = recordTrack; 6407 mLock.unlock(); 6408 status_t status = AudioSystem::startInput(mId); 6409 mLock.lock(); 6410 if (status != NO_ERROR) { 6411 mActiveTrack.clear(); 6412 clearSyncStartEvent(); 6413 return status; 6414 } 6415 mRsmpInIndex = mFrameCount; 6416 mBytesRead = 0; 6417 if (mResampler != NULL) { 6418 mResampler->reset(); 6419 } 6420 mActiveTrack->mState = TrackBase::RESUMING; 6421 // signal thread to start 6422 ALOGV("Signal record thread"); 6423 mWaitWorkCV.broadcast(); 6424 // do not wait for mStartStopCond if exiting 6425 if (exitPending()) { 6426 mActiveTrack.clear(); 6427 status = INVALID_OPERATION; 6428 goto startError; 6429 } 6430 mStartStopCond.wait(mLock); 6431 if (mActiveTrack == 0) { 6432 ALOGV("Record failed to start"); 6433 status = BAD_VALUE; 6434 goto startError; 6435 } 6436 ALOGV("Record started OK"); 6437 return status; 6438 } 6439startError: 6440 AudioSystem::stopInput(mId); 6441 clearSyncStartEvent(); 6442 return status; 6443} 6444 6445void AudioFlinger::RecordThread::clearSyncStartEvent() 6446{ 6447 if (mSyncStartEvent != 0) { 6448 mSyncStartEvent->cancel(); 6449 } 6450 mSyncStartEvent.clear(); 6451 mFramestoDrop = 0; 6452} 6453 6454void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6455{ 6456 sp<SyncEvent> strongEvent = event.promote(); 6457 6458 if (strongEvent != 0) { 6459 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6460 me->handleSyncStartEvent(strongEvent); 6461 } 6462} 6463 6464void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6465{ 6466 if (event == mSyncStartEvent) { 6467 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6468 // from audio HAL 6469 mFramestoDrop = mFrameCount * 2; 6470 } 6471} 6472 6473bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6474 ALOGV("RecordThread::stop"); 6475 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6476 return false; 6477 } 6478 recordTrack->mState = TrackBase::PAUSING; 6479 // do not wait for mStartStopCond if exiting 6480 if (exitPending()) { 6481 return true; 6482 } 6483 mStartStopCond.wait(mLock); 6484 // if we have been restarted, recordTrack == mActiveTrack.get() here 6485 if (exitPending() || recordTrack != mActiveTrack.get()) { 6486 ALOGV("Record stopped OK"); 6487 return true; 6488 } 6489 return false; 6490} 6491 6492bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6493{ 6494 return false; 6495} 6496 6497status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6498{ 6499#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6500 if (!isValidSyncEvent(event)) { 6501 return BAD_VALUE; 6502 } 6503 6504 int eventSession = event->triggerSession(); 6505 status_t ret = NAME_NOT_FOUND; 6506 6507 Mutex::Autolock _l(mLock); 6508 6509 for (size_t i = 0; i < mTracks.size(); i++) { 6510 sp<RecordTrack> track = mTracks[i]; 6511 if (eventSession == track->sessionId()) { 6512 (void) track->setSyncEvent(event); 6513 ret = NO_ERROR; 6514 } 6515 } 6516 return ret; 6517#else 6518 return BAD_VALUE; 6519#endif 6520} 6521 6522void AudioFlinger::RecordThread::RecordTrack::destroy() 6523{ 6524 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6525 sp<RecordTrack> keep(this); 6526 { 6527 sp<ThreadBase> thread = mThread.promote(); 6528 if (thread != 0) { 6529 if (mState == ACTIVE || mState == RESUMING) { 6530 AudioSystem::stopInput(thread->id()); 6531 } 6532 AudioSystem::releaseInput(thread->id()); 6533 Mutex::Autolock _l(thread->mLock); 6534 RecordThread *recordThread = (RecordThread *) thread.get(); 6535 recordThread->destroyTrack_l(this); 6536 } 6537 } 6538} 6539 6540// destroyTrack_l() must be called with ThreadBase::mLock held 6541void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6542{ 6543 track->mState = TrackBase::TERMINATED; 6544 // active tracks are removed by threadLoop() 6545 if (mActiveTrack != track) { 6546 removeTrack_l(track); 6547 } 6548} 6549 6550void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6551{ 6552 mTracks.remove(track); 6553 // need anything related to effects here? 6554} 6555 6556void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6557{ 6558 dumpInternals(fd, args); 6559 dumpTracks(fd, args); 6560 dumpEffectChains(fd, args); 6561} 6562 6563void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6564{ 6565 const size_t SIZE = 256; 6566 char buffer[SIZE]; 6567 String8 result; 6568 6569 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6570 result.append(buffer); 6571 6572 if (mActiveTrack != 0) { 6573 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6574 result.append(buffer); 6575 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6576 result.append(buffer); 6577 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6578 result.append(buffer); 6579 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 6580 result.append(buffer); 6581 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 6582 result.append(buffer); 6583 } else { 6584 result.append("No active record client\n"); 6585 } 6586 6587 write(fd, result.string(), result.size()); 6588 6589 dumpBase(fd, args); 6590} 6591 6592void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6593{ 6594 const size_t SIZE = 256; 6595 char buffer[SIZE]; 6596 String8 result; 6597 6598 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6599 result.append(buffer); 6600 RecordTrack::appendDumpHeader(result); 6601 for (size_t i = 0; i < mTracks.size(); ++i) { 6602 sp<RecordTrack> track = mTracks[i]; 6603 if (track != 0) { 6604 track->dump(buffer, SIZE); 6605 result.append(buffer); 6606 } 6607 } 6608 6609 if (mActiveTrack != 0) { 6610 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6611 result.append(buffer); 6612 RecordTrack::appendDumpHeader(result); 6613 mActiveTrack->dump(buffer, SIZE); 6614 result.append(buffer); 6615 6616 } 6617 write(fd, result.string(), result.size()); 6618} 6619 6620// AudioBufferProvider interface 6621status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6622{ 6623 size_t framesReq = buffer->frameCount; 6624 size_t framesReady = mFrameCount - mRsmpInIndex; 6625 int channelCount; 6626 6627 if (framesReady == 0) { 6628 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6629 if (mBytesRead <= 0) { 6630 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6631 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6632 // Force input into standby so that it tries to 6633 // recover at next read attempt 6634 inputStandBy(); 6635 usleep(kRecordThreadSleepUs); 6636 } 6637 buffer->raw = NULL; 6638 buffer->frameCount = 0; 6639 return NOT_ENOUGH_DATA; 6640 } 6641 mRsmpInIndex = 0; 6642 framesReady = mFrameCount; 6643 } 6644 6645 if (framesReq > framesReady) { 6646 framesReq = framesReady; 6647 } 6648 6649 if (mChannelCount == 1 && mReqChannelCount == 2) { 6650 channelCount = 1; 6651 } else { 6652 channelCount = 2; 6653 } 6654 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6655 buffer->frameCount = framesReq; 6656 return NO_ERROR; 6657} 6658 6659// AudioBufferProvider interface 6660void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6661{ 6662 mRsmpInIndex += buffer->frameCount; 6663 buffer->frameCount = 0; 6664} 6665 6666bool AudioFlinger::RecordThread::checkForNewParameters_l() 6667{ 6668 bool reconfig = false; 6669 6670 while (!mNewParameters.isEmpty()) { 6671 status_t status = NO_ERROR; 6672 String8 keyValuePair = mNewParameters[0]; 6673 AudioParameter param = AudioParameter(keyValuePair); 6674 int value; 6675 audio_format_t reqFormat = mFormat; 6676 uint32_t reqSamplingRate = mReqSampleRate; 6677 uint32_t reqChannelCount = mReqChannelCount; 6678 6679 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6680 reqSamplingRate = value; 6681 reconfig = true; 6682 } 6683 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6684 reqFormat = (audio_format_t) value; 6685 reconfig = true; 6686 } 6687 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6688 reqChannelCount = popcount(value); 6689 reconfig = true; 6690 } 6691 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6692 // do not accept frame count changes if tracks are open as the track buffer 6693 // size depends on frame count and correct behavior would not be guaranteed 6694 // if frame count is changed after track creation 6695 if (mActiveTrack != 0) { 6696 status = INVALID_OPERATION; 6697 } else { 6698 reconfig = true; 6699 } 6700 } 6701 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6702 // forward device change to effects that have requested to be 6703 // aware of attached audio device. 6704 for (size_t i = 0; i < mEffectChains.size(); i++) { 6705 mEffectChains[i]->setDevice_l(value); 6706 } 6707 6708 // store input device and output device but do not forward output device to audio HAL. 6709 // Note that status is ignored by the caller for output device 6710 // (see AudioFlinger::setParameters() 6711 if (audio_is_output_devices(value)) { 6712 mOutDevice = value; 6713 status = BAD_VALUE; 6714 } else { 6715 mInDevice = value; 6716 // disable AEC and NS if the device is a BT SCO headset supporting those 6717 // pre processings 6718 if (mTracks.size() > 0) { 6719 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6720 mAudioFlinger->btNrecIsOff(); 6721 for (size_t i = 0; i < mTracks.size(); i++) { 6722 sp<RecordTrack> track = mTracks[i]; 6723 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6724 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6725 } 6726 } 6727 } 6728 } 6729 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6730 mAudioSource != (audio_source_t)value) { 6731 // forward device change to effects that have requested to be 6732 // aware of attached audio device. 6733 for (size_t i = 0; i < mEffectChains.size(); i++) { 6734 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6735 } 6736 mAudioSource = (audio_source_t)value; 6737 } 6738 if (status == NO_ERROR) { 6739 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6740 keyValuePair.string()); 6741 if (status == INVALID_OPERATION) { 6742 inputStandBy(); 6743 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6744 keyValuePair.string()); 6745 } 6746 if (reconfig) { 6747 if (status == BAD_VALUE && 6748 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6749 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6750 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6751 <= (2 * reqSamplingRate)) && 6752 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6753 <= FCC_2 && 6754 (reqChannelCount <= FCC_2)) { 6755 status = NO_ERROR; 6756 } 6757 if (status == NO_ERROR) { 6758 readInputParameters(); 6759 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6760 } 6761 } 6762 } 6763 6764 mNewParameters.removeAt(0); 6765 6766 mParamStatus = status; 6767 mParamCond.signal(); 6768 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6769 // already timed out waiting for the status and will never signal the condition. 6770 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6771 } 6772 return reconfig; 6773} 6774 6775String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6776{ 6777 char *s; 6778 String8 out_s8 = String8(); 6779 6780 Mutex::Autolock _l(mLock); 6781 if (initCheck() != NO_ERROR) { 6782 return out_s8; 6783 } 6784 6785 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6786 out_s8 = String8(s); 6787 free(s); 6788 return out_s8; 6789} 6790 6791void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6792 AudioSystem::OutputDescriptor desc; 6793 void *param2 = NULL; 6794 6795 switch (event) { 6796 case AudioSystem::INPUT_OPENED: 6797 case AudioSystem::INPUT_CONFIG_CHANGED: 6798 desc.channels = mChannelMask; 6799 desc.samplingRate = mSampleRate; 6800 desc.format = mFormat; 6801 desc.frameCount = mFrameCount; 6802 desc.latency = 0; 6803 param2 = &desc; 6804 break; 6805 6806 case AudioSystem::INPUT_CLOSED: 6807 default: 6808 break; 6809 } 6810 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6811} 6812 6813void AudioFlinger::RecordThread::readInputParameters() 6814{ 6815 delete mRsmpInBuffer; 6816 // mRsmpInBuffer is always assigned a new[] below 6817 delete mRsmpOutBuffer; 6818 mRsmpOutBuffer = NULL; 6819 delete mResampler; 6820 mResampler = NULL; 6821 6822 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6823 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6824 mChannelCount = (uint16_t)popcount(mChannelMask); 6825 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6826 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6827 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6828 mFrameCount = mInputBytes / mFrameSize; 6829 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6830 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6831 6832 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6833 { 6834 int channelCount; 6835 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6836 // stereo to mono post process as the resampler always outputs stereo. 6837 if (mChannelCount == 1 && mReqChannelCount == 2) { 6838 channelCount = 1; 6839 } else { 6840 channelCount = 2; 6841 } 6842 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6843 mResampler->setSampleRate(mSampleRate); 6844 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6845 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6846 6847 // optmization: if mono to mono, alter input frame count as if we were inputing 6848 // stereo samples 6849 if (mChannelCount == 1 && mReqChannelCount == 1) { 6850 mFrameCount >>= 1; 6851 } 6852 6853 } 6854 mRsmpInIndex = mFrameCount; 6855} 6856 6857unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6858{ 6859 Mutex::Autolock _l(mLock); 6860 if (initCheck() != NO_ERROR) { 6861 return 0; 6862 } 6863 6864 return mInput->stream->get_input_frames_lost(mInput->stream); 6865} 6866 6867uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6868{ 6869 Mutex::Autolock _l(mLock); 6870 uint32_t result = 0; 6871 if (getEffectChain_l(sessionId) != 0) { 6872 result = EFFECT_SESSION; 6873 } 6874 6875 for (size_t i = 0; i < mTracks.size(); ++i) { 6876 if (sessionId == mTracks[i]->sessionId()) { 6877 result |= TRACK_SESSION; 6878 break; 6879 } 6880 } 6881 6882 return result; 6883} 6884 6885KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6886{ 6887 KeyedVector<int, bool> ids; 6888 Mutex::Autolock _l(mLock); 6889 for (size_t j = 0; j < mTracks.size(); ++j) { 6890 sp<RecordThread::RecordTrack> track = mTracks[j]; 6891 int sessionId = track->sessionId(); 6892 if (ids.indexOfKey(sessionId) < 0) { 6893 ids.add(sessionId, true); 6894 } 6895 } 6896 return ids; 6897} 6898 6899AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6900{ 6901 Mutex::Autolock _l(mLock); 6902 AudioStreamIn *input = mInput; 6903 mInput = NULL; 6904 return input; 6905} 6906 6907// this method must always be called either with ThreadBase mLock held or inside the thread loop 6908audio_stream_t* AudioFlinger::RecordThread::stream() const 6909{ 6910 if (mInput == NULL) { 6911 return NULL; 6912 } 6913 return &mInput->stream->common; 6914} 6915 6916 6917// ---------------------------------------------------------------------------- 6918 6919audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6920{ 6921 if (!settingsAllowed()) { 6922 return 0; 6923 } 6924 Mutex::Autolock _l(mLock); 6925 return loadHwModule_l(name); 6926} 6927 6928// loadHwModule_l() must be called with AudioFlinger::mLock held 6929audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6930{ 6931 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6932 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6933 ALOGW("loadHwModule() module %s already loaded", name); 6934 return mAudioHwDevs.keyAt(i); 6935 } 6936 } 6937 6938 audio_hw_device_t *dev; 6939 6940 int rc = load_audio_interface(name, &dev); 6941 if (rc) { 6942 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6943 return 0; 6944 } 6945 6946 mHardwareStatus = AUDIO_HW_INIT; 6947 rc = dev->init_check(dev); 6948 mHardwareStatus = AUDIO_HW_IDLE; 6949 if (rc) { 6950 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6951 return 0; 6952 } 6953 6954 // Check and cache this HAL's level of support for master mute and master 6955 // volume. If this is the first HAL opened, and it supports the get 6956 // methods, use the initial values provided by the HAL as the current 6957 // master mute and volume settings. 6958 6959 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6960 { // scope for auto-lock pattern 6961 AutoMutex lock(mHardwareLock); 6962 6963 if (0 == mAudioHwDevs.size()) { 6964 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6965 if (NULL != dev->get_master_volume) { 6966 float mv; 6967 if (OK == dev->get_master_volume(dev, &mv)) { 6968 mMasterVolume = mv; 6969 } 6970 } 6971 6972 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6973 if (NULL != dev->get_master_mute) { 6974 bool mm; 6975 if (OK == dev->get_master_mute(dev, &mm)) { 6976 mMasterMute = mm; 6977 } 6978 } 6979 } 6980 6981 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6982 if ((NULL != dev->set_master_volume) && 6983 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6984 flags = static_cast<AudioHwDevice::Flags>(flags | 6985 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6986 } 6987 6988 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6989 if ((NULL != dev->set_master_mute) && 6990 (OK == dev->set_master_mute(dev, mMasterMute))) { 6991 flags = static_cast<AudioHwDevice::Flags>(flags | 6992 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6993 } 6994 6995 mHardwareStatus = AUDIO_HW_IDLE; 6996 } 6997 6998 audio_module_handle_t handle = nextUniqueId(); 6999 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 7000 7001 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 7002 name, dev->common.module->name, dev->common.module->id, handle); 7003 7004 return handle; 7005 7006} 7007 7008// ---------------------------------------------------------------------------- 7009 7010uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 7011{ 7012 Mutex::Autolock _l(mLock); 7013 PlaybackThread *thread = primaryPlaybackThread_l(); 7014 return thread != NULL ? thread->sampleRate() : 0; 7015} 7016 7017size_t AudioFlinger::getPrimaryOutputFrameCount() 7018{ 7019 Mutex::Autolock _l(mLock); 7020 PlaybackThread *thread = primaryPlaybackThread_l(); 7021 return thread != NULL ? thread->frameCountHAL() : 0; 7022} 7023 7024// ---------------------------------------------------------------------------- 7025 7026audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 7027 audio_devices_t *pDevices, 7028 uint32_t *pSamplingRate, 7029 audio_format_t *pFormat, 7030 audio_channel_mask_t *pChannelMask, 7031 uint32_t *pLatencyMs, 7032 audio_output_flags_t flags) 7033{ 7034 status_t status; 7035 PlaybackThread *thread = NULL; 7036 struct audio_config config = { 7037 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7038 channel_mask: pChannelMask ? *pChannelMask : 0, 7039 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7040 }; 7041 audio_stream_out_t *outStream = NULL; 7042 AudioHwDevice *outHwDev; 7043 7044 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7045 module, 7046 (pDevices != NULL) ? *pDevices : 0, 7047 config.sample_rate, 7048 config.format, 7049 config.channel_mask, 7050 flags); 7051 7052 if (pDevices == NULL || *pDevices == 0) { 7053 return 0; 7054 } 7055 7056 Mutex::Autolock _l(mLock); 7057 7058 outHwDev = findSuitableHwDev_l(module, *pDevices); 7059 if (outHwDev == NULL) 7060 return 0; 7061 7062 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7063 audio_io_handle_t id = nextUniqueId(); 7064 7065 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7066 7067 status = hwDevHal->open_output_stream(hwDevHal, 7068 id, 7069 *pDevices, 7070 (audio_output_flags_t)flags, 7071 &config, 7072 &outStream); 7073 7074 mHardwareStatus = AUDIO_HW_IDLE; 7075 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7076 "Channels %x, status %d", 7077 outStream, 7078 config.sample_rate, 7079 config.format, 7080 config.channel_mask, 7081 status); 7082 7083 if (status == NO_ERROR && outStream != NULL) { 7084 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7085 7086 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7087 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7088 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7089 thread = new DirectOutputThread(this, output, id, *pDevices); 7090 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7091 } else { 7092 thread = new MixerThread(this, output, id, *pDevices); 7093 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7094 } 7095 mPlaybackThreads.add(id, thread); 7096 7097 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7098 if (pFormat != NULL) *pFormat = config.format; 7099 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7100 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7101 7102 // notify client processes of the new output creation 7103 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7104 7105 // the first primary output opened designates the primary hw device 7106 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7107 ALOGI("Using module %d has the primary audio interface", module); 7108 mPrimaryHardwareDev = outHwDev; 7109 7110 AutoMutex lock(mHardwareLock); 7111 mHardwareStatus = AUDIO_HW_SET_MODE; 7112 hwDevHal->set_mode(hwDevHal, mMode); 7113 mHardwareStatus = AUDIO_HW_IDLE; 7114 } 7115 return id; 7116 } 7117 7118 return 0; 7119} 7120 7121audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7122 audio_io_handle_t output2) 7123{ 7124 Mutex::Autolock _l(mLock); 7125 MixerThread *thread1 = checkMixerThread_l(output1); 7126 MixerThread *thread2 = checkMixerThread_l(output2); 7127 7128 if (thread1 == NULL || thread2 == NULL) { 7129 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7130 output2); 7131 return 0; 7132 } 7133 7134 audio_io_handle_t id = nextUniqueId(); 7135 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7136 thread->addOutputTrack(thread2); 7137 mPlaybackThreads.add(id, thread); 7138 // notify client processes of the new output creation 7139 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7140 return id; 7141} 7142 7143status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7144{ 7145 return closeOutput_nonvirtual(output); 7146} 7147 7148status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7149{ 7150 // keep strong reference on the playback thread so that 7151 // it is not destroyed while exit() is executed 7152 sp<PlaybackThread> thread; 7153 { 7154 Mutex::Autolock _l(mLock); 7155 thread = checkPlaybackThread_l(output); 7156 if (thread == NULL) { 7157 return BAD_VALUE; 7158 } 7159 7160 ALOGV("closeOutput() %d", output); 7161 7162 if (thread->type() == ThreadBase::MIXER) { 7163 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7164 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7165 DuplicatingThread *dupThread = 7166 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7167 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7168 } 7169 } 7170 } 7171 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7172 mPlaybackThreads.removeItem(output); 7173 } 7174 thread->exit(); 7175 // The thread entity (active unit of execution) is no longer running here, 7176 // but the ThreadBase container still exists. 7177 7178 if (thread->type() != ThreadBase::DUPLICATING) { 7179 AudioStreamOut *out = thread->clearOutput(); 7180 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7181 // from now on thread->mOutput is NULL 7182 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7183 delete out; 7184 } 7185 return NO_ERROR; 7186} 7187 7188status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7189{ 7190 Mutex::Autolock _l(mLock); 7191 PlaybackThread *thread = checkPlaybackThread_l(output); 7192 7193 if (thread == NULL) { 7194 return BAD_VALUE; 7195 } 7196 7197 ALOGV("suspendOutput() %d", output); 7198 thread->suspend(); 7199 7200 return NO_ERROR; 7201} 7202 7203status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7204{ 7205 Mutex::Autolock _l(mLock); 7206 PlaybackThread *thread = checkPlaybackThread_l(output); 7207 7208 if (thread == NULL) { 7209 return BAD_VALUE; 7210 } 7211 7212 ALOGV("restoreOutput() %d", output); 7213 7214 thread->restore(); 7215 7216 return NO_ERROR; 7217} 7218 7219audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7220 audio_devices_t *pDevices, 7221 uint32_t *pSamplingRate, 7222 audio_format_t *pFormat, 7223 audio_channel_mask_t *pChannelMask) 7224{ 7225 status_t status; 7226 RecordThread *thread = NULL; 7227 struct audio_config config = { 7228 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7229 channel_mask: pChannelMask ? *pChannelMask : 0, 7230 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7231 }; 7232 uint32_t reqSamplingRate = config.sample_rate; 7233 audio_format_t reqFormat = config.format; 7234 audio_channel_mask_t reqChannels = config.channel_mask; 7235 audio_stream_in_t *inStream = NULL; 7236 AudioHwDevice *inHwDev; 7237 7238 if (pDevices == NULL || *pDevices == 0) { 7239 return 0; 7240 } 7241 7242 Mutex::Autolock _l(mLock); 7243 7244 inHwDev = findSuitableHwDev_l(module, *pDevices); 7245 if (inHwDev == NULL) 7246 return 0; 7247 7248 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7249 audio_io_handle_t id = nextUniqueId(); 7250 7251 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7252 &inStream); 7253 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7254 "status %d", 7255 inStream, 7256 config.sample_rate, 7257 config.format, 7258 config.channel_mask, 7259 status); 7260 7261 // If the input could not be opened with the requested parameters and we can handle the 7262 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7263 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7264 if (status == BAD_VALUE && 7265 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7266 (config.sample_rate <= 2 * reqSamplingRate) && 7267 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7268 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7269 inStream = NULL; 7270 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7271 } 7272 7273 if (status == NO_ERROR && inStream != NULL) { 7274 7275 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7276 // or (re-)create if current Pipe is idle and does not match the new format 7277 sp<NBAIO_Sink> teeSink; 7278#ifdef TEE_SINK_INPUT_FRAMES 7279 enum { 7280 TEE_SINK_NO, // don't copy input 7281 TEE_SINK_NEW, // copy input using a new pipe 7282 TEE_SINK_OLD, // copy input using an existing pipe 7283 } kind; 7284 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7285 popcount(inStream->common.get_channels(&inStream->common))); 7286 if (format == Format_Invalid) { 7287 kind = TEE_SINK_NO; 7288 } else if (mRecordTeeSink == 0) { 7289 kind = TEE_SINK_NEW; 7290 } else if (mRecordTeeSink->getStrongCount() != 1) { 7291 kind = TEE_SINK_NO; 7292 } else if (format == mRecordTeeSink->format()) { 7293 kind = TEE_SINK_OLD; 7294 } else { 7295 kind = TEE_SINK_NEW; 7296 } 7297 switch (kind) { 7298 case TEE_SINK_NEW: { 7299 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7300 size_t numCounterOffers = 0; 7301 const NBAIO_Format offers[1] = {format}; 7302 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7303 ALOG_ASSERT(index == 0); 7304 PipeReader *pipeReader = new PipeReader(*pipe); 7305 numCounterOffers = 0; 7306 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7307 ALOG_ASSERT(index == 0); 7308 mRecordTeeSink = pipe; 7309 mRecordTeeSource = pipeReader; 7310 teeSink = pipe; 7311 } 7312 break; 7313 case TEE_SINK_OLD: 7314 teeSink = mRecordTeeSink; 7315 break; 7316 case TEE_SINK_NO: 7317 default: 7318 break; 7319 } 7320#endif 7321 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7322 7323 // Start record thread 7324 // RecorThread require both input and output device indication to forward to audio 7325 // pre processing modules 7326 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7327 7328 thread = new RecordThread(this, 7329 input, 7330 reqSamplingRate, 7331 reqChannels, 7332 id, 7333 device, teeSink); 7334 mRecordThreads.add(id, thread); 7335 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7336 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7337 if (pFormat != NULL) *pFormat = config.format; 7338 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7339 7340 // notify client processes of the new input creation 7341 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7342 return id; 7343 } 7344 7345 return 0; 7346} 7347 7348status_t AudioFlinger::closeInput(audio_io_handle_t input) 7349{ 7350 return closeInput_nonvirtual(input); 7351} 7352 7353status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7354{ 7355 // keep strong reference on the record thread so that 7356 // it is not destroyed while exit() is executed 7357 sp<RecordThread> thread; 7358 { 7359 Mutex::Autolock _l(mLock); 7360 thread = checkRecordThread_l(input); 7361 if (thread == 0) { 7362 return BAD_VALUE; 7363 } 7364 7365 ALOGV("closeInput() %d", input); 7366 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7367 mRecordThreads.removeItem(input); 7368 } 7369 thread->exit(); 7370 // The thread entity (active unit of execution) is no longer running here, 7371 // but the ThreadBase container still exists. 7372 7373 AudioStreamIn *in = thread->clearInput(); 7374 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7375 // from now on thread->mInput is NULL 7376 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7377 delete in; 7378 7379 return NO_ERROR; 7380} 7381 7382status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7383{ 7384 Mutex::Autolock _l(mLock); 7385 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7386 7387 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7388 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7389 thread->invalidateTracks(stream); 7390 } 7391 7392 return NO_ERROR; 7393} 7394 7395 7396int AudioFlinger::newAudioSessionId() 7397{ 7398 return nextUniqueId(); 7399} 7400 7401void AudioFlinger::acquireAudioSessionId(int audioSession) 7402{ 7403 Mutex::Autolock _l(mLock); 7404 pid_t caller = IPCThreadState::self()->getCallingPid(); 7405 ALOGV("acquiring %d from %d", audioSession, caller); 7406 size_t num = mAudioSessionRefs.size(); 7407 for (size_t i = 0; i< num; i++) { 7408 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7409 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7410 ref->mCnt++; 7411 ALOGV(" incremented refcount to %d", ref->mCnt); 7412 return; 7413 } 7414 } 7415 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7416 ALOGV(" added new entry for %d", audioSession); 7417} 7418 7419void AudioFlinger::releaseAudioSessionId(int audioSession) 7420{ 7421 Mutex::Autolock _l(mLock); 7422 pid_t caller = IPCThreadState::self()->getCallingPid(); 7423 ALOGV("releasing %d from %d", audioSession, caller); 7424 size_t num = mAudioSessionRefs.size(); 7425 for (size_t i = 0; i< num; i++) { 7426 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7427 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7428 ref->mCnt--; 7429 ALOGV(" decremented refcount to %d", ref->mCnt); 7430 if (ref->mCnt == 0) { 7431 mAudioSessionRefs.removeAt(i); 7432 delete ref; 7433 purgeStaleEffects_l(); 7434 } 7435 return; 7436 } 7437 } 7438 ALOGW("session id %d not found for pid %d", audioSession, caller); 7439} 7440 7441void AudioFlinger::purgeStaleEffects_l() { 7442 7443 ALOGV("purging stale effects"); 7444 7445 Vector< sp<EffectChain> > chains; 7446 7447 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7448 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7449 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7450 sp<EffectChain> ec = t->mEffectChains[j]; 7451 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7452 chains.push(ec); 7453 } 7454 } 7455 } 7456 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7457 sp<RecordThread> t = mRecordThreads.valueAt(i); 7458 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7459 sp<EffectChain> ec = t->mEffectChains[j]; 7460 chains.push(ec); 7461 } 7462 } 7463 7464 for (size_t i = 0; i < chains.size(); i++) { 7465 sp<EffectChain> ec = chains[i]; 7466 int sessionid = ec->sessionId(); 7467 sp<ThreadBase> t = ec->mThread.promote(); 7468 if (t == 0) { 7469 continue; 7470 } 7471 size_t numsessionrefs = mAudioSessionRefs.size(); 7472 bool found = false; 7473 for (size_t k = 0; k < numsessionrefs; k++) { 7474 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7475 if (ref->mSessionid == sessionid) { 7476 ALOGV(" session %d still exists for %d with %d refs", 7477 sessionid, ref->mPid, ref->mCnt); 7478 found = true; 7479 break; 7480 } 7481 } 7482 if (!found) { 7483 Mutex::Autolock _l (t->mLock); 7484 // remove all effects from the chain 7485 while (ec->mEffects.size()) { 7486 sp<EffectModule> effect = ec->mEffects[0]; 7487 effect->unPin(); 7488 t->removeEffect_l(effect); 7489 if (effect->purgeHandles()) { 7490 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7491 } 7492 AudioSystem::unregisterEffect(effect->id()); 7493 } 7494 } 7495 } 7496 return; 7497} 7498 7499// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7500AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7501{ 7502 return mPlaybackThreads.valueFor(output).get(); 7503} 7504 7505// checkMixerThread_l() must be called with AudioFlinger::mLock held 7506AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7507{ 7508 PlaybackThread *thread = checkPlaybackThread_l(output); 7509 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7510} 7511 7512// checkRecordThread_l() must be called with AudioFlinger::mLock held 7513AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7514{ 7515 return mRecordThreads.valueFor(input).get(); 7516} 7517 7518uint32_t AudioFlinger::nextUniqueId() 7519{ 7520 return android_atomic_inc(&mNextUniqueId); 7521} 7522 7523AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7524{ 7525 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7526 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7527 AudioStreamOut *output = thread->getOutput(); 7528 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7529 return thread; 7530 } 7531 } 7532 return NULL; 7533} 7534 7535audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7536{ 7537 PlaybackThread *thread = primaryPlaybackThread_l(); 7538 7539 if (thread == NULL) { 7540 return 0; 7541 } 7542 7543 return thread->outDevice(); 7544} 7545 7546sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7547 int triggerSession, 7548 int listenerSession, 7549 sync_event_callback_t callBack, 7550 void *cookie) 7551{ 7552 Mutex::Autolock _l(mLock); 7553 7554 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7555 status_t playStatus = NAME_NOT_FOUND; 7556 status_t recStatus = NAME_NOT_FOUND; 7557 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7558 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7559 if (playStatus == NO_ERROR) { 7560 return event; 7561 } 7562 } 7563 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7564 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7565 if (recStatus == NO_ERROR) { 7566 return event; 7567 } 7568 } 7569 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7570 mPendingSyncEvents.add(event); 7571 } else { 7572 ALOGV("createSyncEvent() invalid event %d", event->type()); 7573 event.clear(); 7574 } 7575 return event; 7576} 7577 7578// ---------------------------------------------------------------------------- 7579// Effect management 7580// ---------------------------------------------------------------------------- 7581 7582 7583status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7584{ 7585 Mutex::Autolock _l(mLock); 7586 return EffectQueryNumberEffects(numEffects); 7587} 7588 7589status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7590{ 7591 Mutex::Autolock _l(mLock); 7592 return EffectQueryEffect(index, descriptor); 7593} 7594 7595status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7596 effect_descriptor_t *descriptor) const 7597{ 7598 Mutex::Autolock _l(mLock); 7599 return EffectGetDescriptor(pUuid, descriptor); 7600} 7601 7602 7603sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7604 effect_descriptor_t *pDesc, 7605 const sp<IEffectClient>& effectClient, 7606 int32_t priority, 7607 audio_io_handle_t io, 7608 int sessionId, 7609 status_t *status, 7610 int *id, 7611 int *enabled) 7612{ 7613 status_t lStatus = NO_ERROR; 7614 sp<EffectHandle> handle; 7615 effect_descriptor_t desc; 7616 7617 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7618 pid, effectClient.get(), priority, sessionId, io); 7619 7620 if (pDesc == NULL) { 7621 lStatus = BAD_VALUE; 7622 goto Exit; 7623 } 7624 7625 // check audio settings permission for global effects 7626 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7627 lStatus = PERMISSION_DENIED; 7628 goto Exit; 7629 } 7630 7631 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7632 // that can only be created by audio policy manager (running in same process) 7633 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7634 lStatus = PERMISSION_DENIED; 7635 goto Exit; 7636 } 7637 7638 if (io == 0) { 7639 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7640 // output must be specified by AudioPolicyManager when using session 7641 // AUDIO_SESSION_OUTPUT_STAGE 7642 lStatus = BAD_VALUE; 7643 goto Exit; 7644 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7645 // if the output returned by getOutputForEffect() is removed before we lock the 7646 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7647 // and we will exit safely 7648 io = AudioSystem::getOutputForEffect(&desc); 7649 } 7650 } 7651 7652 { 7653 Mutex::Autolock _l(mLock); 7654 7655 7656 if (!EffectIsNullUuid(&pDesc->uuid)) { 7657 // if uuid is specified, request effect descriptor 7658 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7659 if (lStatus < 0) { 7660 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7661 goto Exit; 7662 } 7663 } else { 7664 // if uuid is not specified, look for an available implementation 7665 // of the required type in effect factory 7666 if (EffectIsNullUuid(&pDesc->type)) { 7667 ALOGW("createEffect() no effect type"); 7668 lStatus = BAD_VALUE; 7669 goto Exit; 7670 } 7671 uint32_t numEffects = 0; 7672 effect_descriptor_t d; 7673 d.flags = 0; // prevent compiler warning 7674 bool found = false; 7675 7676 lStatus = EffectQueryNumberEffects(&numEffects); 7677 if (lStatus < 0) { 7678 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7679 goto Exit; 7680 } 7681 for (uint32_t i = 0; i < numEffects; i++) { 7682 lStatus = EffectQueryEffect(i, &desc); 7683 if (lStatus < 0) { 7684 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7685 continue; 7686 } 7687 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7688 // If matching type found save effect descriptor. If the session is 7689 // 0 and the effect is not auxiliary, continue enumeration in case 7690 // an auxiliary version of this effect type is available 7691 found = true; 7692 d = desc; 7693 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7694 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7695 break; 7696 } 7697 } 7698 } 7699 if (!found) { 7700 lStatus = BAD_VALUE; 7701 ALOGW("createEffect() effect not found"); 7702 goto Exit; 7703 } 7704 // For same effect type, chose auxiliary version over insert version if 7705 // connect to output mix (Compliance to OpenSL ES) 7706 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7707 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7708 desc = d; 7709 } 7710 } 7711 7712 // Do not allow auxiliary effects on a session different from 0 (output mix) 7713 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7714 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7715 lStatus = INVALID_OPERATION; 7716 goto Exit; 7717 } 7718 7719 // check recording permission for visualizer 7720 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7721 !recordingAllowed()) { 7722 lStatus = PERMISSION_DENIED; 7723 goto Exit; 7724 } 7725 7726 // return effect descriptor 7727 *pDesc = desc; 7728 7729 // If output is not specified try to find a matching audio session ID in one of the 7730 // output threads. 7731 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7732 // because of code checking output when entering the function. 7733 // Note: io is never 0 when creating an effect on an input 7734 if (io == 0) { 7735 // look for the thread where the specified audio session is present 7736 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7737 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7738 io = mPlaybackThreads.keyAt(i); 7739 break; 7740 } 7741 } 7742 if (io == 0) { 7743 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7744 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7745 io = mRecordThreads.keyAt(i); 7746 break; 7747 } 7748 } 7749 } 7750 // If no output thread contains the requested session ID, default to 7751 // first output. The effect chain will be moved to the correct output 7752 // thread when a track with the same session ID is created 7753 if (io == 0 && mPlaybackThreads.size()) { 7754 io = mPlaybackThreads.keyAt(0); 7755 } 7756 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7757 } 7758 ThreadBase *thread = checkRecordThread_l(io); 7759 if (thread == NULL) { 7760 thread = checkPlaybackThread_l(io); 7761 if (thread == NULL) { 7762 ALOGE("createEffect() unknown output thread"); 7763 lStatus = BAD_VALUE; 7764 goto Exit; 7765 } 7766 } 7767 7768 sp<Client> client = registerPid_l(pid); 7769 7770 // create effect on selected output thread 7771 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7772 &desc, enabled, &lStatus); 7773 if (handle != 0 && id != NULL) { 7774 *id = handle->id(); 7775 } 7776 } 7777 7778Exit: 7779 if (status != NULL) { 7780 *status = lStatus; 7781 } 7782 return handle; 7783} 7784 7785status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7786 audio_io_handle_t dstOutput) 7787{ 7788 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7789 sessionId, srcOutput, dstOutput); 7790 Mutex::Autolock _l(mLock); 7791 if (srcOutput == dstOutput) { 7792 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7793 return NO_ERROR; 7794 } 7795 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7796 if (srcThread == NULL) { 7797 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7798 return BAD_VALUE; 7799 } 7800 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7801 if (dstThread == NULL) { 7802 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7803 return BAD_VALUE; 7804 } 7805 7806 Mutex::Autolock _dl(dstThread->mLock); 7807 Mutex::Autolock _sl(srcThread->mLock); 7808 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7809 7810 return NO_ERROR; 7811} 7812 7813// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7814status_t AudioFlinger::moveEffectChain_l(int sessionId, 7815 AudioFlinger::PlaybackThread *srcThread, 7816 AudioFlinger::PlaybackThread *dstThread, 7817 bool reRegister) 7818{ 7819 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7820 sessionId, srcThread, dstThread); 7821 7822 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7823 if (chain == 0) { 7824 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7825 sessionId, srcThread); 7826 return INVALID_OPERATION; 7827 } 7828 7829 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7830 // so that a new chain is created with correct parameters when first effect is added. This is 7831 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7832 // removed. 7833 srcThread->removeEffectChain_l(chain); 7834 7835 // transfer all effects one by one so that new effect chain is created on new thread with 7836 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7837 audio_io_handle_t dstOutput = dstThread->id(); 7838 sp<EffectChain> dstChain; 7839 uint32_t strategy = 0; // prevent compiler warning 7840 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7841 while (effect != 0) { 7842 srcThread->removeEffect_l(effect); 7843 dstThread->addEffect_l(effect); 7844 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7845 if (effect->state() == EffectModule::ACTIVE || 7846 effect->state() == EffectModule::STOPPING) { 7847 effect->start(); 7848 } 7849 // if the move request is not received from audio policy manager, the effect must be 7850 // re-registered with the new strategy and output 7851 if (dstChain == 0) { 7852 dstChain = effect->chain().promote(); 7853 if (dstChain == 0) { 7854 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7855 srcThread->addEffect_l(effect); 7856 return NO_INIT; 7857 } 7858 strategy = dstChain->strategy(); 7859 } 7860 if (reRegister) { 7861 AudioSystem::unregisterEffect(effect->id()); 7862 AudioSystem::registerEffect(&effect->desc(), 7863 dstOutput, 7864 strategy, 7865 sessionId, 7866 effect->id()); 7867 } 7868 effect = chain->getEffectFromId_l(0); 7869 } 7870 7871 return NO_ERROR; 7872} 7873 7874 7875// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7876sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7877 const sp<AudioFlinger::Client>& client, 7878 const sp<IEffectClient>& effectClient, 7879 int32_t priority, 7880 int sessionId, 7881 effect_descriptor_t *desc, 7882 int *enabled, 7883 status_t *status 7884 ) 7885{ 7886 sp<EffectModule> effect; 7887 sp<EffectHandle> handle; 7888 status_t lStatus; 7889 sp<EffectChain> chain; 7890 bool chainCreated = false; 7891 bool effectCreated = false; 7892 bool effectRegistered = false; 7893 7894 lStatus = initCheck(); 7895 if (lStatus != NO_ERROR) { 7896 ALOGW("createEffect_l() Audio driver not initialized."); 7897 goto Exit; 7898 } 7899 7900 // Do not allow effects with session ID 0 on direct output or duplicating threads 7901 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7902 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7903 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7904 desc->name, sessionId); 7905 lStatus = BAD_VALUE; 7906 goto Exit; 7907 } 7908 // Only Pre processor effects are allowed on input threads and only on input threads 7909 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7910 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7911 desc->name, desc->flags, mType); 7912 lStatus = BAD_VALUE; 7913 goto Exit; 7914 } 7915 7916 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7917 7918 { // scope for mLock 7919 Mutex::Autolock _l(mLock); 7920 7921 // check for existing effect chain with the requested audio session 7922 chain = getEffectChain_l(sessionId); 7923 if (chain == 0) { 7924 // create a new chain for this session 7925 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7926 chain = new EffectChain(this, sessionId); 7927 addEffectChain_l(chain); 7928 chain->setStrategy(getStrategyForSession_l(sessionId)); 7929 chainCreated = true; 7930 } else { 7931 effect = chain->getEffectFromDesc_l(desc); 7932 } 7933 7934 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7935 7936 if (effect == 0) { 7937 int id = mAudioFlinger->nextUniqueId(); 7938 // Check CPU and memory usage 7939 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7940 if (lStatus != NO_ERROR) { 7941 goto Exit; 7942 } 7943 effectRegistered = true; 7944 // create a new effect module if none present in the chain 7945 effect = new EffectModule(this, chain, desc, id, sessionId); 7946 lStatus = effect->status(); 7947 if (lStatus != NO_ERROR) { 7948 goto Exit; 7949 } 7950 lStatus = chain->addEffect_l(effect); 7951 if (lStatus != NO_ERROR) { 7952 goto Exit; 7953 } 7954 effectCreated = true; 7955 7956 effect->setDevice(mOutDevice); 7957 effect->setDevice(mInDevice); 7958 effect->setMode(mAudioFlinger->getMode()); 7959 effect->setAudioSource(mAudioSource); 7960 } 7961 // create effect handle and connect it to effect module 7962 handle = new EffectHandle(effect, client, effectClient, priority); 7963 lStatus = effect->addHandle(handle.get()); 7964 if (enabled != NULL) { 7965 *enabled = (int)effect->isEnabled(); 7966 } 7967 } 7968 7969Exit: 7970 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7971 Mutex::Autolock _l(mLock); 7972 if (effectCreated) { 7973 chain->removeEffect_l(effect); 7974 } 7975 if (effectRegistered) { 7976 AudioSystem::unregisterEffect(effect->id()); 7977 } 7978 if (chainCreated) { 7979 removeEffectChain_l(chain); 7980 } 7981 handle.clear(); 7982 } 7983 7984 if (status != NULL) { 7985 *status = lStatus; 7986 } 7987 return handle; 7988} 7989 7990sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7991{ 7992 Mutex::Autolock _l(mLock); 7993 return getEffect_l(sessionId, effectId); 7994} 7995 7996sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7997{ 7998 sp<EffectChain> chain = getEffectChain_l(sessionId); 7999 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 8000} 8001 8002// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 8003// PlaybackThread::mLock held 8004status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 8005{ 8006 // check for existing effect chain with the requested audio session 8007 int sessionId = effect->sessionId(); 8008 sp<EffectChain> chain = getEffectChain_l(sessionId); 8009 bool chainCreated = false; 8010 8011 if (chain == 0) { 8012 // create a new chain for this session 8013 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 8014 chain = new EffectChain(this, sessionId); 8015 addEffectChain_l(chain); 8016 chain->setStrategy(getStrategyForSession_l(sessionId)); 8017 chainCreated = true; 8018 } 8019 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 8020 8021 if (chain->getEffectFromId_l(effect->id()) != 0) { 8022 ALOGW("addEffect_l() %p effect %s already present in chain %p", 8023 this, effect->desc().name, chain.get()); 8024 return BAD_VALUE; 8025 } 8026 8027 status_t status = chain->addEffect_l(effect); 8028 if (status != NO_ERROR) { 8029 if (chainCreated) { 8030 removeEffectChain_l(chain); 8031 } 8032 return status; 8033 } 8034 8035 effect->setDevice(mOutDevice); 8036 effect->setDevice(mInDevice); 8037 effect->setMode(mAudioFlinger->getMode()); 8038 effect->setAudioSource(mAudioSource); 8039 return NO_ERROR; 8040} 8041 8042void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8043 8044 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8045 effect_descriptor_t desc = effect->desc(); 8046 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8047 detachAuxEffect_l(effect->id()); 8048 } 8049 8050 sp<EffectChain> chain = effect->chain().promote(); 8051 if (chain != 0) { 8052 // remove effect chain if removing last effect 8053 if (chain->removeEffect_l(effect) == 0) { 8054 removeEffectChain_l(chain); 8055 } 8056 } else { 8057 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8058 } 8059} 8060 8061void AudioFlinger::ThreadBase::lockEffectChains_l( 8062 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8063{ 8064 effectChains = mEffectChains; 8065 for (size_t i = 0; i < mEffectChains.size(); i++) { 8066 mEffectChains[i]->lock(); 8067 } 8068} 8069 8070void AudioFlinger::ThreadBase::unlockEffectChains( 8071 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8072{ 8073 for (size_t i = 0; i < effectChains.size(); i++) { 8074 effectChains[i]->unlock(); 8075 } 8076} 8077 8078sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8079{ 8080 Mutex::Autolock _l(mLock); 8081 return getEffectChain_l(sessionId); 8082} 8083 8084sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8085{ 8086 size_t size = mEffectChains.size(); 8087 for (size_t i = 0; i < size; i++) { 8088 if (mEffectChains[i]->sessionId() == sessionId) { 8089 return mEffectChains[i]; 8090 } 8091 } 8092 return 0; 8093} 8094 8095void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8096{ 8097 Mutex::Autolock _l(mLock); 8098 size_t size = mEffectChains.size(); 8099 for (size_t i = 0; i < size; i++) { 8100 mEffectChains[i]->setMode_l(mode); 8101 } 8102} 8103 8104void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8105 EffectHandle *handle, 8106 bool unpinIfLast) { 8107 8108 Mutex::Autolock _l(mLock); 8109 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8110 // delete the effect module if removing last handle on it 8111 if (effect->removeHandle(handle) == 0) { 8112 if (!effect->isPinned() || unpinIfLast) { 8113 removeEffect_l(effect); 8114 AudioSystem::unregisterEffect(effect->id()); 8115 } 8116 } 8117} 8118 8119status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8120{ 8121 int session = chain->sessionId(); 8122 int16_t *buffer = mMixBuffer; 8123 bool ownsBuffer = false; 8124 8125 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8126 if (session > 0) { 8127 // Only one effect chain can be present in direct output thread and it uses 8128 // the mix buffer as input 8129 if (mType != DIRECT) { 8130 size_t numSamples = mNormalFrameCount * mChannelCount; 8131 buffer = new int16_t[numSamples]; 8132 memset(buffer, 0, numSamples * sizeof(int16_t)); 8133 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8134 ownsBuffer = true; 8135 } 8136 8137 // Attach all tracks with same session ID to this chain. 8138 for (size_t i = 0; i < mTracks.size(); ++i) { 8139 sp<Track> track = mTracks[i]; 8140 if (session == track->sessionId()) { 8141 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8142 buffer); 8143 track->setMainBuffer(buffer); 8144 chain->incTrackCnt(); 8145 } 8146 } 8147 8148 // indicate all active tracks in the chain 8149 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8150 sp<Track> track = mActiveTracks[i].promote(); 8151 if (track == 0) { 8152 continue; 8153 } 8154 if (session == track->sessionId()) { 8155 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8156 chain->incActiveTrackCnt(); 8157 } 8158 } 8159 } 8160 8161 chain->setInBuffer(buffer, ownsBuffer); 8162 chain->setOutBuffer(mMixBuffer); 8163 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8164 // chains list in order to be processed last as it contains output stage effects 8165 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8166 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8167 // after track specific effects and before output stage 8168 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8169 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8170 // Effect chain for other sessions are inserted at beginning of effect 8171 // chains list to be processed before output mix effects. Relative order between other 8172 // sessions is not important 8173 size_t size = mEffectChains.size(); 8174 size_t i = 0; 8175 for (i = 0; i < size; i++) { 8176 if (mEffectChains[i]->sessionId() < session) { 8177 break; 8178 } 8179 } 8180 mEffectChains.insertAt(chain, i); 8181 checkSuspendOnAddEffectChain_l(chain); 8182 8183 return NO_ERROR; 8184} 8185 8186size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8187{ 8188 int session = chain->sessionId(); 8189 8190 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8191 8192 for (size_t i = 0; i < mEffectChains.size(); i++) { 8193 if (chain == mEffectChains[i]) { 8194 mEffectChains.removeAt(i); 8195 // detach all active tracks from the chain 8196 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8197 sp<Track> track = mActiveTracks[i].promote(); 8198 if (track == 0) { 8199 continue; 8200 } 8201 if (session == track->sessionId()) { 8202 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8203 chain.get(), session); 8204 chain->decActiveTrackCnt(); 8205 } 8206 } 8207 8208 // detach all tracks with same session ID from this chain 8209 for (size_t i = 0; i < mTracks.size(); ++i) { 8210 sp<Track> track = mTracks[i]; 8211 if (session == track->sessionId()) { 8212 track->setMainBuffer(mMixBuffer); 8213 chain->decTrackCnt(); 8214 } 8215 } 8216 break; 8217 } 8218 } 8219 return mEffectChains.size(); 8220} 8221 8222status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8223 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8224{ 8225 Mutex::Autolock _l(mLock); 8226 return attachAuxEffect_l(track, EffectId); 8227} 8228 8229status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8230 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8231{ 8232 status_t status = NO_ERROR; 8233 8234 if (EffectId == 0) { 8235 track->setAuxBuffer(0, NULL); 8236 } else { 8237 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8238 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8239 if (effect != 0) { 8240 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8241 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8242 } else { 8243 status = INVALID_OPERATION; 8244 } 8245 } else { 8246 status = BAD_VALUE; 8247 } 8248 } 8249 return status; 8250} 8251 8252void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8253{ 8254 for (size_t i = 0; i < mTracks.size(); ++i) { 8255 sp<Track> track = mTracks[i]; 8256 if (track->auxEffectId() == effectId) { 8257 attachAuxEffect_l(track, 0); 8258 } 8259 } 8260} 8261 8262status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8263{ 8264 // only one chain per input thread 8265 if (mEffectChains.size() != 0) { 8266 return INVALID_OPERATION; 8267 } 8268 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8269 8270 chain->setInBuffer(NULL); 8271 chain->setOutBuffer(NULL); 8272 8273 checkSuspendOnAddEffectChain_l(chain); 8274 8275 mEffectChains.add(chain); 8276 8277 return NO_ERROR; 8278} 8279 8280size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8281{ 8282 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8283 ALOGW_IF(mEffectChains.size() != 1, 8284 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8285 chain.get(), mEffectChains.size(), this); 8286 if (mEffectChains.size() == 1) { 8287 mEffectChains.removeAt(0); 8288 } 8289 return 0; 8290} 8291 8292// ---------------------------------------------------------------------------- 8293// EffectModule implementation 8294// ---------------------------------------------------------------------------- 8295 8296#undef LOG_TAG 8297#define LOG_TAG "AudioFlinger::EffectModule" 8298 8299AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8300 const wp<AudioFlinger::EffectChain>& chain, 8301 effect_descriptor_t *desc, 8302 int id, 8303 int sessionId) 8304 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8305 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8306 mDescriptor(*desc), 8307 // mConfig is set by configure() and not used before then 8308 mEffectInterface(NULL), 8309 mStatus(NO_INIT), mState(IDLE), 8310 // mMaxDisableWaitCnt is set by configure() and not used before then 8311 // mDisableWaitCnt is set by process() and updateState() and not used before then 8312 mSuspended(false) 8313{ 8314 ALOGV("Constructor %p", this); 8315 int lStatus; 8316 8317 // create effect engine from effect factory 8318 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8319 8320 if (mStatus != NO_ERROR) { 8321 return; 8322 } 8323 lStatus = init(); 8324 if (lStatus < 0) { 8325 mStatus = lStatus; 8326 goto Error; 8327 } 8328 8329 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8330 return; 8331Error: 8332 EffectRelease(mEffectInterface); 8333 mEffectInterface = NULL; 8334 ALOGV("Constructor Error %d", mStatus); 8335} 8336 8337AudioFlinger::EffectModule::~EffectModule() 8338{ 8339 ALOGV("Destructor %p", this); 8340 if (mEffectInterface != NULL) { 8341 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8342 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8343 sp<ThreadBase> thread = mThread.promote(); 8344 if (thread != 0) { 8345 audio_stream_t *stream = thread->stream(); 8346 if (stream != NULL) { 8347 stream->remove_audio_effect(stream, mEffectInterface); 8348 } 8349 } 8350 } 8351 // release effect engine 8352 EffectRelease(mEffectInterface); 8353 } 8354} 8355 8356status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8357{ 8358 status_t status; 8359 8360 Mutex::Autolock _l(mLock); 8361 int priority = handle->priority(); 8362 size_t size = mHandles.size(); 8363 EffectHandle *controlHandle = NULL; 8364 size_t i; 8365 for (i = 0; i < size; i++) { 8366 EffectHandle *h = mHandles[i]; 8367 if (h == NULL || h->destroyed_l()) { 8368 continue; 8369 } 8370 // first non destroyed handle is considered in control 8371 if (controlHandle == NULL) 8372 controlHandle = h; 8373 if (h->priority() <= priority) { 8374 break; 8375 } 8376 } 8377 // if inserted in first place, move effect control from previous owner to this handle 8378 if (i == 0) { 8379 bool enabled = false; 8380 if (controlHandle != NULL) { 8381 enabled = controlHandle->enabled(); 8382 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8383 } 8384 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8385 status = NO_ERROR; 8386 } else { 8387 status = ALREADY_EXISTS; 8388 } 8389 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8390 mHandles.insertAt(handle, i); 8391 return status; 8392} 8393 8394size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8395{ 8396 Mutex::Autolock _l(mLock); 8397 size_t size = mHandles.size(); 8398 size_t i; 8399 for (i = 0; i < size; i++) { 8400 if (mHandles[i] == handle) { 8401 break; 8402 } 8403 } 8404 if (i == size) { 8405 return size; 8406 } 8407 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8408 8409 mHandles.removeAt(i); 8410 // if removed from first place, move effect control from this handle to next in line 8411 if (i == 0) { 8412 EffectHandle *h = controlHandle_l(); 8413 if (h != NULL) { 8414 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8415 } 8416 } 8417 8418 // Prevent calls to process() and other functions on effect interface from now on. 8419 // The effect engine will be released by the destructor when the last strong reference on 8420 // this object is released which can happen after next process is called. 8421 if (mHandles.size() == 0 && !mPinned) { 8422 mState = DESTROYED; 8423 } 8424 8425 return mHandles.size(); 8426} 8427 8428// must be called with EffectModule::mLock held 8429AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8430{ 8431 // the first valid handle in the list has control over the module 8432 for (size_t i = 0; i < mHandles.size(); i++) { 8433 EffectHandle *h = mHandles[i]; 8434 if (h != NULL && !h->destroyed_l()) { 8435 return h; 8436 } 8437 } 8438 8439 return NULL; 8440} 8441 8442size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8443{ 8444 ALOGV("disconnect() %p handle %p", this, handle); 8445 // keep a strong reference on this EffectModule to avoid calling the 8446 // destructor before we exit 8447 sp<EffectModule> keep(this); 8448 { 8449 sp<ThreadBase> thread = mThread.promote(); 8450 if (thread != 0) { 8451 thread->disconnectEffect(keep, handle, unpinIfLast); 8452 } 8453 } 8454 return mHandles.size(); 8455} 8456 8457void AudioFlinger::EffectModule::updateState() { 8458 Mutex::Autolock _l(mLock); 8459 8460 switch (mState) { 8461 case RESTART: 8462 reset_l(); 8463 // FALL THROUGH 8464 8465 case STARTING: 8466 // clear auxiliary effect input buffer for next accumulation 8467 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8468 memset(mConfig.inputCfg.buffer.raw, 8469 0, 8470 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8471 } 8472 start_l(); 8473 mState = ACTIVE; 8474 break; 8475 case STOPPING: 8476 stop_l(); 8477 mDisableWaitCnt = mMaxDisableWaitCnt; 8478 mState = STOPPED; 8479 break; 8480 case STOPPED: 8481 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8482 // turn off sequence. 8483 if (--mDisableWaitCnt == 0) { 8484 reset_l(); 8485 mState = IDLE; 8486 } 8487 break; 8488 default: //IDLE , ACTIVE, DESTROYED 8489 break; 8490 } 8491} 8492 8493void AudioFlinger::EffectModule::process() 8494{ 8495 Mutex::Autolock _l(mLock); 8496 8497 if (mState == DESTROYED || mEffectInterface == NULL || 8498 mConfig.inputCfg.buffer.raw == NULL || 8499 mConfig.outputCfg.buffer.raw == NULL) { 8500 return; 8501 } 8502 8503 if (isProcessEnabled()) { 8504 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8505 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8506 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8507 mConfig.inputCfg.buffer.s32, 8508 mConfig.inputCfg.buffer.frameCount/2); 8509 } 8510 8511 // do the actual processing in the effect engine 8512 int ret = (*mEffectInterface)->process(mEffectInterface, 8513 &mConfig.inputCfg.buffer, 8514 &mConfig.outputCfg.buffer); 8515 8516 // force transition to IDLE state when engine is ready 8517 if (mState == STOPPED && ret == -ENODATA) { 8518 mDisableWaitCnt = 1; 8519 } 8520 8521 // clear auxiliary effect input buffer for next accumulation 8522 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8523 memset(mConfig.inputCfg.buffer.raw, 0, 8524 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8525 } 8526 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8527 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8528 // If an insert effect is idle and input buffer is different from output buffer, 8529 // accumulate input onto output 8530 sp<EffectChain> chain = mChain.promote(); 8531 if (chain != 0 && chain->activeTrackCnt() != 0) { 8532 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8533 int16_t *in = mConfig.inputCfg.buffer.s16; 8534 int16_t *out = mConfig.outputCfg.buffer.s16; 8535 for (size_t i = 0; i < frameCnt; i++) { 8536 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8537 } 8538 } 8539 } 8540} 8541 8542void AudioFlinger::EffectModule::reset_l() 8543{ 8544 if (mEffectInterface == NULL) { 8545 return; 8546 } 8547 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8548} 8549 8550status_t AudioFlinger::EffectModule::configure() 8551{ 8552 if (mEffectInterface == NULL) { 8553 return NO_INIT; 8554 } 8555 8556 sp<ThreadBase> thread = mThread.promote(); 8557 if (thread == 0) { 8558 return DEAD_OBJECT; 8559 } 8560 8561 // TODO: handle configuration of effects replacing track process 8562 audio_channel_mask_t channelMask = thread->channelMask(); 8563 8564 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8565 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8566 } else { 8567 mConfig.inputCfg.channels = channelMask; 8568 } 8569 mConfig.outputCfg.channels = channelMask; 8570 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8571 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8572 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8573 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8574 mConfig.inputCfg.bufferProvider.cookie = NULL; 8575 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8576 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8577 mConfig.outputCfg.bufferProvider.cookie = NULL; 8578 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8579 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8580 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8581 // Insert effect: 8582 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8583 // always overwrites output buffer: input buffer == output buffer 8584 // - in other sessions: 8585 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8586 // other effect: overwrites output buffer: input buffer == output buffer 8587 // Auxiliary effect: 8588 // accumulates in output buffer: input buffer != output buffer 8589 // Therefore: accumulate <=> input buffer != output buffer 8590 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8591 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8592 } else { 8593 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8594 } 8595 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8596 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8597 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8598 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8599 8600 ALOGV("configure() %p thread %p buffer %p framecount %d", 8601 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8602 8603 status_t cmdStatus; 8604 uint32_t size = sizeof(int); 8605 status_t status = (*mEffectInterface)->command(mEffectInterface, 8606 EFFECT_CMD_SET_CONFIG, 8607 sizeof(effect_config_t), 8608 &mConfig, 8609 &size, 8610 &cmdStatus); 8611 if (status == 0) { 8612 status = cmdStatus; 8613 } 8614 8615 if (status == 0 && 8616 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8617 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8618 effect_param_t *p = (effect_param_t *)buf32; 8619 8620 p->psize = sizeof(uint32_t); 8621 p->vsize = sizeof(uint32_t); 8622 size = sizeof(int); 8623 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8624 8625 uint32_t latency = 0; 8626 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8627 if (pbt != NULL) { 8628 latency = pbt->latency_l(); 8629 } 8630 8631 *((int32_t *)p->data + 1)= latency; 8632 (*mEffectInterface)->command(mEffectInterface, 8633 EFFECT_CMD_SET_PARAM, 8634 sizeof(effect_param_t) + 8, 8635 &buf32, 8636 &size, 8637 &cmdStatus); 8638 } 8639 8640 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8641 (1000 * mConfig.outputCfg.buffer.frameCount); 8642 8643 return status; 8644} 8645 8646status_t AudioFlinger::EffectModule::init() 8647{ 8648 Mutex::Autolock _l(mLock); 8649 if (mEffectInterface == NULL) { 8650 return NO_INIT; 8651 } 8652 status_t cmdStatus; 8653 uint32_t size = sizeof(status_t); 8654 status_t status = (*mEffectInterface)->command(mEffectInterface, 8655 EFFECT_CMD_INIT, 8656 0, 8657 NULL, 8658 &size, 8659 &cmdStatus); 8660 if (status == 0) { 8661 status = cmdStatus; 8662 } 8663 return status; 8664} 8665 8666status_t AudioFlinger::EffectModule::start() 8667{ 8668 Mutex::Autolock _l(mLock); 8669 return start_l(); 8670} 8671 8672status_t AudioFlinger::EffectModule::start_l() 8673{ 8674 if (mEffectInterface == NULL) { 8675 return NO_INIT; 8676 } 8677 status_t cmdStatus; 8678 uint32_t size = sizeof(status_t); 8679 status_t status = (*mEffectInterface)->command(mEffectInterface, 8680 EFFECT_CMD_ENABLE, 8681 0, 8682 NULL, 8683 &size, 8684 &cmdStatus); 8685 if (status == 0) { 8686 status = cmdStatus; 8687 } 8688 if (status == 0 && 8689 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8690 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8691 sp<ThreadBase> thread = mThread.promote(); 8692 if (thread != 0) { 8693 audio_stream_t *stream = thread->stream(); 8694 if (stream != NULL) { 8695 stream->add_audio_effect(stream, mEffectInterface); 8696 } 8697 } 8698 } 8699 return status; 8700} 8701 8702status_t AudioFlinger::EffectModule::stop() 8703{ 8704 Mutex::Autolock _l(mLock); 8705 return stop_l(); 8706} 8707 8708status_t AudioFlinger::EffectModule::stop_l() 8709{ 8710 if (mEffectInterface == NULL) { 8711 return NO_INIT; 8712 } 8713 status_t cmdStatus; 8714 uint32_t size = sizeof(status_t); 8715 status_t status = (*mEffectInterface)->command(mEffectInterface, 8716 EFFECT_CMD_DISABLE, 8717 0, 8718 NULL, 8719 &size, 8720 &cmdStatus); 8721 if (status == 0) { 8722 status = cmdStatus; 8723 } 8724 if (status == 0 && 8725 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8726 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8727 sp<ThreadBase> thread = mThread.promote(); 8728 if (thread != 0) { 8729 audio_stream_t *stream = thread->stream(); 8730 if (stream != NULL) { 8731 stream->remove_audio_effect(stream, mEffectInterface); 8732 } 8733 } 8734 } 8735 return status; 8736} 8737 8738status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8739 uint32_t cmdSize, 8740 void *pCmdData, 8741 uint32_t *replySize, 8742 void *pReplyData) 8743{ 8744 Mutex::Autolock _l(mLock); 8745 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8746 8747 if (mState == DESTROYED || mEffectInterface == NULL) { 8748 return NO_INIT; 8749 } 8750 status_t status = (*mEffectInterface)->command(mEffectInterface, 8751 cmdCode, 8752 cmdSize, 8753 pCmdData, 8754 replySize, 8755 pReplyData); 8756 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8757 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8758 for (size_t i = 1; i < mHandles.size(); i++) { 8759 EffectHandle *h = mHandles[i]; 8760 if (h != NULL && !h->destroyed_l()) { 8761 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8762 } 8763 } 8764 } 8765 return status; 8766} 8767 8768status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8769{ 8770 Mutex::Autolock _l(mLock); 8771 return setEnabled_l(enabled); 8772} 8773 8774// must be called with EffectModule::mLock held 8775status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8776{ 8777 8778 ALOGV("setEnabled %p enabled %d", this, enabled); 8779 8780 if (enabled != isEnabled()) { 8781 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8782 if (enabled && status != NO_ERROR) { 8783 return status; 8784 } 8785 8786 switch (mState) { 8787 // going from disabled to enabled 8788 case IDLE: 8789 mState = STARTING; 8790 break; 8791 case STOPPED: 8792 mState = RESTART; 8793 break; 8794 case STOPPING: 8795 mState = ACTIVE; 8796 break; 8797 8798 // going from enabled to disabled 8799 case RESTART: 8800 mState = STOPPED; 8801 break; 8802 case STARTING: 8803 mState = IDLE; 8804 break; 8805 case ACTIVE: 8806 mState = STOPPING; 8807 break; 8808 case DESTROYED: 8809 return NO_ERROR; // simply ignore as we are being destroyed 8810 } 8811 for (size_t i = 1; i < mHandles.size(); i++) { 8812 EffectHandle *h = mHandles[i]; 8813 if (h != NULL && !h->destroyed_l()) { 8814 h->setEnabled(enabled); 8815 } 8816 } 8817 } 8818 return NO_ERROR; 8819} 8820 8821bool AudioFlinger::EffectModule::isEnabled() const 8822{ 8823 switch (mState) { 8824 case RESTART: 8825 case STARTING: 8826 case ACTIVE: 8827 return true; 8828 case IDLE: 8829 case STOPPING: 8830 case STOPPED: 8831 case DESTROYED: 8832 default: 8833 return false; 8834 } 8835} 8836 8837bool AudioFlinger::EffectModule::isProcessEnabled() const 8838{ 8839 switch (mState) { 8840 case RESTART: 8841 case ACTIVE: 8842 case STOPPING: 8843 case STOPPED: 8844 return true; 8845 case IDLE: 8846 case STARTING: 8847 case DESTROYED: 8848 default: 8849 return false; 8850 } 8851} 8852 8853status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8854{ 8855 Mutex::Autolock _l(mLock); 8856 status_t status = NO_ERROR; 8857 8858 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8859 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8860 if (isProcessEnabled() && 8861 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8862 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8863 status_t cmdStatus; 8864 uint32_t volume[2]; 8865 uint32_t *pVolume = NULL; 8866 uint32_t size = sizeof(volume); 8867 volume[0] = *left; 8868 volume[1] = *right; 8869 if (controller) { 8870 pVolume = volume; 8871 } 8872 status = (*mEffectInterface)->command(mEffectInterface, 8873 EFFECT_CMD_SET_VOLUME, 8874 size, 8875 volume, 8876 &size, 8877 pVolume); 8878 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8879 *left = volume[0]; 8880 *right = volume[1]; 8881 } 8882 } 8883 return status; 8884} 8885 8886status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8887{ 8888 if (device == AUDIO_DEVICE_NONE) { 8889 return NO_ERROR; 8890 } 8891 8892 Mutex::Autolock _l(mLock); 8893 status_t status = NO_ERROR; 8894 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8895 status_t cmdStatus; 8896 uint32_t size = sizeof(status_t); 8897 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8898 EFFECT_CMD_SET_INPUT_DEVICE; 8899 status = (*mEffectInterface)->command(mEffectInterface, 8900 cmd, 8901 sizeof(uint32_t), 8902 &device, 8903 &size, 8904 &cmdStatus); 8905 } 8906 return status; 8907} 8908 8909status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8910{ 8911 Mutex::Autolock _l(mLock); 8912 status_t status = NO_ERROR; 8913 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8914 status_t cmdStatus; 8915 uint32_t size = sizeof(status_t); 8916 status = (*mEffectInterface)->command(mEffectInterface, 8917 EFFECT_CMD_SET_AUDIO_MODE, 8918 sizeof(audio_mode_t), 8919 &mode, 8920 &size, 8921 &cmdStatus); 8922 if (status == NO_ERROR) { 8923 status = cmdStatus; 8924 } 8925 } 8926 return status; 8927} 8928 8929status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8930{ 8931 Mutex::Autolock _l(mLock); 8932 status_t status = NO_ERROR; 8933 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8934 uint32_t size = 0; 8935 status = (*mEffectInterface)->command(mEffectInterface, 8936 EFFECT_CMD_SET_AUDIO_SOURCE, 8937 sizeof(audio_source_t), 8938 &source, 8939 &size, 8940 NULL); 8941 } 8942 return status; 8943} 8944 8945void AudioFlinger::EffectModule::setSuspended(bool suspended) 8946{ 8947 Mutex::Autolock _l(mLock); 8948 mSuspended = suspended; 8949} 8950 8951bool AudioFlinger::EffectModule::suspended() const 8952{ 8953 Mutex::Autolock _l(mLock); 8954 return mSuspended; 8955} 8956 8957bool AudioFlinger::EffectModule::purgeHandles() 8958{ 8959 bool enabled = false; 8960 Mutex::Autolock _l(mLock); 8961 for (size_t i = 0; i < mHandles.size(); i++) { 8962 EffectHandle *handle = mHandles[i]; 8963 if (handle != NULL && !handle->destroyed_l()) { 8964 handle->effect().clear(); 8965 if (handle->hasControl()) { 8966 enabled = handle->enabled(); 8967 } 8968 } 8969 } 8970 return enabled; 8971} 8972 8973void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8974{ 8975 const size_t SIZE = 256; 8976 char buffer[SIZE]; 8977 String8 result; 8978 8979 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8980 result.append(buffer); 8981 8982 bool locked = tryLock(mLock); 8983 // failed to lock - AudioFlinger is probably deadlocked 8984 if (!locked) { 8985 result.append("\t\tCould not lock Fx mutex:\n"); 8986 } 8987 8988 result.append("\t\tSession Status State Engine:\n"); 8989 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8990 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8991 result.append(buffer); 8992 8993 result.append("\t\tDescriptor:\n"); 8994 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8995 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8996 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 8997 mDescriptor.uuid.node[2], 8998 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8999 result.append(buffer); 9000 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 9001 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 9002 mDescriptor.type.timeHiAndVersion, 9003 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 9004 mDescriptor.type.node[2], 9005 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 9006 result.append(buffer); 9007 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 9008 mDescriptor.apiVersion, 9009 mDescriptor.flags); 9010 result.append(buffer); 9011 snprintf(buffer, SIZE, "\t\t- name: %s\n", 9012 mDescriptor.name); 9013 result.append(buffer); 9014 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 9015 mDescriptor.implementor); 9016 result.append(buffer); 9017 9018 result.append("\t\t- Input configuration:\n"); 9019 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 9020 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 9021 (uint32_t)mConfig.inputCfg.buffer.raw, 9022 mConfig.inputCfg.buffer.frameCount, 9023 mConfig.inputCfg.samplingRate, 9024 mConfig.inputCfg.channels, 9025 mConfig.inputCfg.format); 9026 result.append(buffer); 9027 9028 result.append("\t\t- Output configuration:\n"); 9029 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 9030 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 9031 (uint32_t)mConfig.outputCfg.buffer.raw, 9032 mConfig.outputCfg.buffer.frameCount, 9033 mConfig.outputCfg.samplingRate, 9034 mConfig.outputCfg.channels, 9035 mConfig.outputCfg.format); 9036 result.append(buffer); 9037 9038 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 9039 result.append(buffer); 9040 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 9041 for (size_t i = 0; i < mHandles.size(); ++i) { 9042 EffectHandle *handle = mHandles[i]; 9043 if (handle != NULL && !handle->destroyed_l()) { 9044 handle->dump(buffer, SIZE); 9045 result.append(buffer); 9046 } 9047 } 9048 9049 result.append("\n"); 9050 9051 write(fd, result.string(), result.length()); 9052 9053 if (locked) { 9054 mLock.unlock(); 9055 } 9056} 9057 9058// ---------------------------------------------------------------------------- 9059// EffectHandle implementation 9060// ---------------------------------------------------------------------------- 9061 9062#undef LOG_TAG 9063#define LOG_TAG "AudioFlinger::EffectHandle" 9064 9065AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9066 const sp<AudioFlinger::Client>& client, 9067 const sp<IEffectClient>& effectClient, 9068 int32_t priority) 9069 : BnEffect(), 9070 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9071 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9072{ 9073 ALOGV("constructor %p", this); 9074 9075 if (client == 0) { 9076 return; 9077 } 9078 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9079 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9080 if (mCblkMemory != 0) { 9081 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9082 9083 if (mCblk != NULL) { 9084 new(mCblk) effect_param_cblk_t(); 9085 mBuffer = (uint8_t *)mCblk + bufOffset; 9086 } 9087 } else { 9088 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9089 sizeof(effect_param_cblk_t)); 9090 return; 9091 } 9092} 9093 9094AudioFlinger::EffectHandle::~EffectHandle() 9095{ 9096 ALOGV("Destructor %p", this); 9097 9098 if (mEffect == 0) { 9099 mDestroyed = true; 9100 return; 9101 } 9102 mEffect->lock(); 9103 mDestroyed = true; 9104 mEffect->unlock(); 9105 disconnect(false); 9106} 9107 9108status_t AudioFlinger::EffectHandle::enable() 9109{ 9110 ALOGV("enable %p", this); 9111 if (!mHasControl) { 9112 return INVALID_OPERATION; 9113 } 9114 if (mEffect == 0) { 9115 return DEAD_OBJECT; 9116 } 9117 9118 if (mEnabled) { 9119 return NO_ERROR; 9120 } 9121 9122 mEnabled = true; 9123 9124 sp<ThreadBase> thread = mEffect->thread().promote(); 9125 if (thread != 0) { 9126 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9127 } 9128 9129 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9130 if (mEffect->suspended()) { 9131 return NO_ERROR; 9132 } 9133 9134 status_t status = mEffect->setEnabled(true); 9135 if (status != NO_ERROR) { 9136 if (thread != 0) { 9137 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9138 } 9139 mEnabled = false; 9140 } 9141 return status; 9142} 9143 9144status_t AudioFlinger::EffectHandle::disable() 9145{ 9146 ALOGV("disable %p", this); 9147 if (!mHasControl) { 9148 return INVALID_OPERATION; 9149 } 9150 if (mEffect == 0) { 9151 return DEAD_OBJECT; 9152 } 9153 9154 if (!mEnabled) { 9155 return NO_ERROR; 9156 } 9157 mEnabled = false; 9158 9159 if (mEffect->suspended()) { 9160 return NO_ERROR; 9161 } 9162 9163 status_t status = mEffect->setEnabled(false); 9164 9165 sp<ThreadBase> thread = mEffect->thread().promote(); 9166 if (thread != 0) { 9167 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9168 } 9169 9170 return status; 9171} 9172 9173void AudioFlinger::EffectHandle::disconnect() 9174{ 9175 disconnect(true); 9176} 9177 9178void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9179{ 9180 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9181 if (mEffect == 0) { 9182 return; 9183 } 9184 // restore suspended effects if the disconnected handle was enabled and the last one. 9185 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9186 sp<ThreadBase> thread = mEffect->thread().promote(); 9187 if (thread != 0) { 9188 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9189 } 9190 } 9191 9192 // release sp on module => module destructor can be called now 9193 mEffect.clear(); 9194 if (mClient != 0) { 9195 if (mCblk != NULL) { 9196 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9197 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9198 } 9199 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9200 // Client destructor must run with AudioFlinger mutex locked 9201 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9202 mClient.clear(); 9203 } 9204} 9205 9206status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9207 uint32_t cmdSize, 9208 void *pCmdData, 9209 uint32_t *replySize, 9210 void *pReplyData) 9211{ 9212 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9213 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9214 9215 // only get parameter command is permitted for applications not controlling the effect 9216 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9217 return INVALID_OPERATION; 9218 } 9219 if (mEffect == 0) { 9220 return DEAD_OBJECT; 9221 } 9222 if (mClient == 0) { 9223 return INVALID_OPERATION; 9224 } 9225 9226 // handle commands that are not forwarded transparently to effect engine 9227 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9228 // No need to trylock() here as this function is executed in the binder thread serving a 9229 // particular client process: no risk to block the whole media server process or mixer 9230 // threads if we are stuck here 9231 Mutex::Autolock _l(mCblk->lock); 9232 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9233 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9234 mCblk->serverIndex = 0; 9235 mCblk->clientIndex = 0; 9236 return BAD_VALUE; 9237 } 9238 status_t status = NO_ERROR; 9239 while (mCblk->serverIndex < mCblk->clientIndex) { 9240 int reply; 9241 uint32_t rsize = sizeof(int); 9242 int *p = (int *)(mBuffer + mCblk->serverIndex); 9243 int size = *p++; 9244 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9245 ALOGW("command(): invalid parameter block size"); 9246 break; 9247 } 9248 effect_param_t *param = (effect_param_t *)p; 9249 if (param->psize == 0 || param->vsize == 0) { 9250 ALOGW("command(): null parameter or value size"); 9251 mCblk->serverIndex += size; 9252 continue; 9253 } 9254 uint32_t psize = sizeof(effect_param_t) + 9255 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9256 param->vsize; 9257 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9258 psize, 9259 p, 9260 &rsize, 9261 &reply); 9262 // stop at first error encountered 9263 if (ret != NO_ERROR) { 9264 status = ret; 9265 *(int *)pReplyData = reply; 9266 break; 9267 } else if (reply != NO_ERROR) { 9268 *(int *)pReplyData = reply; 9269 break; 9270 } 9271 mCblk->serverIndex += size; 9272 } 9273 mCblk->serverIndex = 0; 9274 mCblk->clientIndex = 0; 9275 return status; 9276 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9277 *(int *)pReplyData = NO_ERROR; 9278 return enable(); 9279 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9280 *(int *)pReplyData = NO_ERROR; 9281 return disable(); 9282 } 9283 9284 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9285} 9286 9287void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9288{ 9289 ALOGV("setControl %p control %d", this, hasControl); 9290 9291 mHasControl = hasControl; 9292 mEnabled = enabled; 9293 9294 if (signal && mEffectClient != 0) { 9295 mEffectClient->controlStatusChanged(hasControl); 9296 } 9297} 9298 9299void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9300 uint32_t cmdSize, 9301 void *pCmdData, 9302 uint32_t replySize, 9303 void *pReplyData) 9304{ 9305 if (mEffectClient != 0) { 9306 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9307 } 9308} 9309 9310 9311 9312void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9313{ 9314 if (mEffectClient != 0) { 9315 mEffectClient->enableStatusChanged(enabled); 9316 } 9317} 9318 9319status_t AudioFlinger::EffectHandle::onTransact( 9320 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9321{ 9322 return BnEffect::onTransact(code, data, reply, flags); 9323} 9324 9325 9326void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9327{ 9328 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9329 9330 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9331 (mClient == 0) ? getpid_cached : mClient->pid(), 9332 mPriority, 9333 mHasControl, 9334 !locked, 9335 mCblk ? mCblk->clientIndex : 0, 9336 mCblk ? mCblk->serverIndex : 0 9337 ); 9338 9339 if (locked) { 9340 mCblk->lock.unlock(); 9341 } 9342} 9343 9344#undef LOG_TAG 9345#define LOG_TAG "AudioFlinger::EffectChain" 9346 9347AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9348 int sessionId) 9349 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9350 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9351 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9352{ 9353 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9354 if (thread == NULL) { 9355 return; 9356 } 9357 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9358 thread->frameCount(); 9359} 9360 9361AudioFlinger::EffectChain::~EffectChain() 9362{ 9363 if (mOwnInBuffer) { 9364 delete mInBuffer; 9365 } 9366 9367} 9368 9369// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9370sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9371 effect_descriptor_t *descriptor) 9372{ 9373 size_t size = mEffects.size(); 9374 9375 for (size_t i = 0; i < size; i++) { 9376 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9377 return mEffects[i]; 9378 } 9379 } 9380 return 0; 9381} 9382 9383// getEffectFromId_l() must be called with ThreadBase::mLock held 9384sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9385{ 9386 size_t size = mEffects.size(); 9387 9388 for (size_t i = 0; i < size; i++) { 9389 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9390 if (id == 0 || mEffects[i]->id() == id) { 9391 return mEffects[i]; 9392 } 9393 } 9394 return 0; 9395} 9396 9397// getEffectFromType_l() must be called with ThreadBase::mLock held 9398sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9399 const effect_uuid_t *type) 9400{ 9401 size_t size = mEffects.size(); 9402 9403 for (size_t i = 0; i < size; i++) { 9404 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9405 return mEffects[i]; 9406 } 9407 } 9408 return 0; 9409} 9410 9411void AudioFlinger::EffectChain::clearInputBuffer() 9412{ 9413 Mutex::Autolock _l(mLock); 9414 sp<ThreadBase> thread = mThread.promote(); 9415 if (thread == 0) { 9416 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9417 return; 9418 } 9419 clearInputBuffer_l(thread); 9420} 9421 9422// Must be called with EffectChain::mLock locked 9423void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9424{ 9425 size_t numSamples = thread->frameCount() * thread->channelCount(); 9426 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9427 9428} 9429 9430// Must be called with EffectChain::mLock locked 9431void AudioFlinger::EffectChain::process_l() 9432{ 9433 sp<ThreadBase> thread = mThread.promote(); 9434 if (thread == 0) { 9435 ALOGW("process_l(): cannot promote mixer thread"); 9436 return; 9437 } 9438 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9439 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9440 // always process effects unless no more tracks are on the session and the effect tail 9441 // has been rendered 9442 bool doProcess = true; 9443 if (!isGlobalSession) { 9444 bool tracksOnSession = (trackCnt() != 0); 9445 9446 if (!tracksOnSession && mTailBufferCount == 0) { 9447 doProcess = false; 9448 } 9449 9450 if (activeTrackCnt() == 0) { 9451 // if no track is active and the effect tail has not been rendered, 9452 // the input buffer must be cleared here as the mixer process will not do it 9453 if (tracksOnSession || mTailBufferCount > 0) { 9454 clearInputBuffer_l(thread); 9455 if (mTailBufferCount > 0) { 9456 mTailBufferCount--; 9457 } 9458 } 9459 } 9460 } 9461 9462 size_t size = mEffects.size(); 9463 if (doProcess) { 9464 for (size_t i = 0; i < size; i++) { 9465 mEffects[i]->process(); 9466 } 9467 } 9468 for (size_t i = 0; i < size; i++) { 9469 mEffects[i]->updateState(); 9470 } 9471} 9472 9473// addEffect_l() must be called with PlaybackThread::mLock held 9474status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9475{ 9476 effect_descriptor_t desc = effect->desc(); 9477 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9478 9479 Mutex::Autolock _l(mLock); 9480 effect->setChain(this); 9481 sp<ThreadBase> thread = mThread.promote(); 9482 if (thread == 0) { 9483 return NO_INIT; 9484 } 9485 effect->setThread(thread); 9486 9487 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9488 // Auxiliary effects are inserted at the beginning of mEffects vector as 9489 // they are processed first and accumulated in chain input buffer 9490 mEffects.insertAt(effect, 0); 9491 9492 // the input buffer for auxiliary effect contains mono samples in 9493 // 32 bit format. This is to avoid saturation in AudoMixer 9494 // accumulation stage. Saturation is done in EffectModule::process() before 9495 // calling the process in effect engine 9496 size_t numSamples = thread->frameCount(); 9497 int32_t *buffer = new int32_t[numSamples]; 9498 memset(buffer, 0, numSamples * sizeof(int32_t)); 9499 effect->setInBuffer((int16_t *)buffer); 9500 // auxiliary effects output samples to chain input buffer for further processing 9501 // by insert effects 9502 effect->setOutBuffer(mInBuffer); 9503 } else { 9504 // Insert effects are inserted at the end of mEffects vector as they are processed 9505 // after track and auxiliary effects. 9506 // Insert effect order as a function of indicated preference: 9507 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9508 // another effect is present 9509 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9510 // last effect claiming first position 9511 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9512 // first effect claiming last position 9513 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9514 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9515 // already present 9516 9517 size_t size = mEffects.size(); 9518 size_t idx_insert = size; 9519 ssize_t idx_insert_first = -1; 9520 ssize_t idx_insert_last = -1; 9521 9522 for (size_t i = 0; i < size; i++) { 9523 effect_descriptor_t d = mEffects[i]->desc(); 9524 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9525 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9526 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9527 // check invalid effect chaining combinations 9528 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9529 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9530 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9531 desc.name, d.name); 9532 return INVALID_OPERATION; 9533 } 9534 // remember position of first insert effect and by default 9535 // select this as insert position for new effect 9536 if (idx_insert == size) { 9537 idx_insert = i; 9538 } 9539 // remember position of last insert effect claiming 9540 // first position 9541 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9542 idx_insert_first = i; 9543 } 9544 // remember position of first insert effect claiming 9545 // last position 9546 if (iPref == EFFECT_FLAG_INSERT_LAST && 9547 idx_insert_last == -1) { 9548 idx_insert_last = i; 9549 } 9550 } 9551 } 9552 9553 // modify idx_insert from first position if needed 9554 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9555 if (idx_insert_last != -1) { 9556 idx_insert = idx_insert_last; 9557 } else { 9558 idx_insert = size; 9559 } 9560 } else { 9561 if (idx_insert_first != -1) { 9562 idx_insert = idx_insert_first + 1; 9563 } 9564 } 9565 9566 // always read samples from chain input buffer 9567 effect->setInBuffer(mInBuffer); 9568 9569 // if last effect in the chain, output samples to chain 9570 // output buffer, otherwise to chain input buffer 9571 if (idx_insert == size) { 9572 if (idx_insert != 0) { 9573 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9574 mEffects[idx_insert-1]->configure(); 9575 } 9576 effect->setOutBuffer(mOutBuffer); 9577 } else { 9578 effect->setOutBuffer(mInBuffer); 9579 } 9580 mEffects.insertAt(effect, idx_insert); 9581 9582 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9583 idx_insert); 9584 } 9585 effect->configure(); 9586 return NO_ERROR; 9587} 9588 9589// removeEffect_l() must be called with PlaybackThread::mLock held 9590size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9591{ 9592 Mutex::Autolock _l(mLock); 9593 size_t size = mEffects.size(); 9594 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9595 9596 for (size_t i = 0; i < size; i++) { 9597 if (effect == mEffects[i]) { 9598 // calling stop here will remove pre-processing effect from the audio HAL. 9599 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9600 // the middle of a read from audio HAL 9601 if (mEffects[i]->state() == EffectModule::ACTIVE || 9602 mEffects[i]->state() == EffectModule::STOPPING) { 9603 mEffects[i]->stop(); 9604 } 9605 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9606 delete[] effect->inBuffer(); 9607 } else { 9608 if (i == size - 1 && i != 0) { 9609 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9610 mEffects[i - 1]->configure(); 9611 } 9612 } 9613 mEffects.removeAt(i); 9614 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9615 this, i); 9616 break; 9617 } 9618 } 9619 9620 return mEffects.size(); 9621} 9622 9623// setDevice_l() must be called with PlaybackThread::mLock held 9624void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9625{ 9626 size_t size = mEffects.size(); 9627 for (size_t i = 0; i < size; i++) { 9628 mEffects[i]->setDevice(device); 9629 } 9630} 9631 9632// setMode_l() must be called with PlaybackThread::mLock held 9633void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9634{ 9635 size_t size = mEffects.size(); 9636 for (size_t i = 0; i < size; i++) { 9637 mEffects[i]->setMode(mode); 9638 } 9639} 9640 9641// setAudioSource_l() must be called with PlaybackThread::mLock held 9642void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9643{ 9644 size_t size = mEffects.size(); 9645 for (size_t i = 0; i < size; i++) { 9646 mEffects[i]->setAudioSource(source); 9647 } 9648} 9649 9650// setVolume_l() must be called with PlaybackThread::mLock held 9651bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9652{ 9653 uint32_t newLeft = *left; 9654 uint32_t newRight = *right; 9655 bool hasControl = false; 9656 int ctrlIdx = -1; 9657 size_t size = mEffects.size(); 9658 9659 // first update volume controller 9660 for (size_t i = size; i > 0; i--) { 9661 if (mEffects[i - 1]->isProcessEnabled() && 9662 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9663 ctrlIdx = i - 1; 9664 hasControl = true; 9665 break; 9666 } 9667 } 9668 9669 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9670 if (hasControl) { 9671 *left = mNewLeftVolume; 9672 *right = mNewRightVolume; 9673 } 9674 return hasControl; 9675 } 9676 9677 mVolumeCtrlIdx = ctrlIdx; 9678 mLeftVolume = newLeft; 9679 mRightVolume = newRight; 9680 9681 // second get volume update from volume controller 9682 if (ctrlIdx >= 0) { 9683 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9684 mNewLeftVolume = newLeft; 9685 mNewRightVolume = newRight; 9686 } 9687 // then indicate volume to all other effects in chain. 9688 // Pass altered volume to effects before volume controller 9689 // and requested volume to effects after controller 9690 uint32_t lVol = newLeft; 9691 uint32_t rVol = newRight; 9692 9693 for (size_t i = 0; i < size; i++) { 9694 if ((int)i == ctrlIdx) { 9695 continue; 9696 } 9697 // this also works for ctrlIdx == -1 when there is no volume controller 9698 if ((int)i > ctrlIdx) { 9699 lVol = *left; 9700 rVol = *right; 9701 } 9702 mEffects[i]->setVolume(&lVol, &rVol, false); 9703 } 9704 *left = newLeft; 9705 *right = newRight; 9706 9707 return hasControl; 9708} 9709 9710void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9711{ 9712 const size_t SIZE = 256; 9713 char buffer[SIZE]; 9714 String8 result; 9715 9716 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9717 result.append(buffer); 9718 9719 bool locked = tryLock(mLock); 9720 // failed to lock - AudioFlinger is probably deadlocked 9721 if (!locked) { 9722 result.append("\tCould not lock mutex:\n"); 9723 } 9724 9725 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9726 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9727 mEffects.size(), 9728 (uint32_t)mInBuffer, 9729 (uint32_t)mOutBuffer, 9730 mActiveTrackCnt); 9731 result.append(buffer); 9732 write(fd, result.string(), result.size()); 9733 9734 for (size_t i = 0; i < mEffects.size(); ++i) { 9735 sp<EffectModule> effect = mEffects[i]; 9736 if (effect != 0) { 9737 effect->dump(fd, args); 9738 } 9739 } 9740 9741 if (locked) { 9742 mLock.unlock(); 9743 } 9744} 9745 9746// must be called with ThreadBase::mLock held 9747void AudioFlinger::EffectChain::setEffectSuspended_l( 9748 const effect_uuid_t *type, bool suspend) 9749{ 9750 sp<SuspendedEffectDesc> desc; 9751 // use effect type UUID timelow as key as there is no real risk of identical 9752 // timeLow fields among effect type UUIDs. 9753 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9754 if (suspend) { 9755 if (index >= 0) { 9756 desc = mSuspendedEffects.valueAt(index); 9757 } else { 9758 desc = new SuspendedEffectDesc(); 9759 desc->mType = *type; 9760 mSuspendedEffects.add(type->timeLow, desc); 9761 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9762 } 9763 if (desc->mRefCount++ == 0) { 9764 sp<EffectModule> effect = getEffectIfEnabled(type); 9765 if (effect != 0) { 9766 desc->mEffect = effect; 9767 effect->setSuspended(true); 9768 effect->setEnabled(false); 9769 } 9770 } 9771 } else { 9772 if (index < 0) { 9773 return; 9774 } 9775 desc = mSuspendedEffects.valueAt(index); 9776 if (desc->mRefCount <= 0) { 9777 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9778 desc->mRefCount = 1; 9779 } 9780 if (--desc->mRefCount == 0) { 9781 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9782 if (desc->mEffect != 0) { 9783 sp<EffectModule> effect = desc->mEffect.promote(); 9784 if (effect != 0) { 9785 effect->setSuspended(false); 9786 effect->lock(); 9787 EffectHandle *handle = effect->controlHandle_l(); 9788 if (handle != NULL && !handle->destroyed_l()) { 9789 effect->setEnabled_l(handle->enabled()); 9790 } 9791 effect->unlock(); 9792 } 9793 desc->mEffect.clear(); 9794 } 9795 mSuspendedEffects.removeItemsAt(index); 9796 } 9797 } 9798} 9799 9800// must be called with ThreadBase::mLock held 9801void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9802{ 9803 sp<SuspendedEffectDesc> desc; 9804 9805 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9806 if (suspend) { 9807 if (index >= 0) { 9808 desc = mSuspendedEffects.valueAt(index); 9809 } else { 9810 desc = new SuspendedEffectDesc(); 9811 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9812 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9813 } 9814 if (desc->mRefCount++ == 0) { 9815 Vector< sp<EffectModule> > effects; 9816 getSuspendEligibleEffects(effects); 9817 for (size_t i = 0; i < effects.size(); i++) { 9818 setEffectSuspended_l(&effects[i]->desc().type, true); 9819 } 9820 } 9821 } else { 9822 if (index < 0) { 9823 return; 9824 } 9825 desc = mSuspendedEffects.valueAt(index); 9826 if (desc->mRefCount <= 0) { 9827 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9828 desc->mRefCount = 1; 9829 } 9830 if (--desc->mRefCount == 0) { 9831 Vector<const effect_uuid_t *> types; 9832 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9833 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9834 continue; 9835 } 9836 types.add(&mSuspendedEffects.valueAt(i)->mType); 9837 } 9838 for (size_t i = 0; i < types.size(); i++) { 9839 setEffectSuspended_l(types[i], false); 9840 } 9841 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9842 mSuspendedEffects.keyAt(index)); 9843 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9844 } 9845 } 9846} 9847 9848 9849// The volume effect is used for automated tests only 9850#ifndef OPENSL_ES_H_ 9851static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9852 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9853const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9854#endif //OPENSL_ES_H_ 9855 9856bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9857{ 9858 // auxiliary effects and visualizer are never suspended on output mix 9859 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9860 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9861 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9862 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9863 return false; 9864 } 9865 return true; 9866} 9867 9868void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9869 Vector< sp<AudioFlinger::EffectModule> > &effects) 9870{ 9871 effects.clear(); 9872 for (size_t i = 0; i < mEffects.size(); i++) { 9873 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9874 effects.add(mEffects[i]); 9875 } 9876 } 9877} 9878 9879sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9880 const effect_uuid_t *type) 9881{ 9882 sp<EffectModule> effect = getEffectFromType_l(type); 9883 return effect != 0 && effect->isEnabled() ? effect : 0; 9884} 9885 9886void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9887 bool enabled) 9888{ 9889 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9890 if (enabled) { 9891 if (index < 0) { 9892 // if the effect is not suspend check if all effects are suspended 9893 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9894 if (index < 0) { 9895 return; 9896 } 9897 if (!isEffectEligibleForSuspend(effect->desc())) { 9898 return; 9899 } 9900 setEffectSuspended_l(&effect->desc().type, enabled); 9901 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9902 if (index < 0) { 9903 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9904 return; 9905 } 9906 } 9907 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9908 effect->desc().type.timeLow); 9909 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9910 // if effect is requested to suspended but was not yet enabled, supend it now. 9911 if (desc->mEffect == 0) { 9912 desc->mEffect = effect; 9913 effect->setEnabled(false); 9914 effect->setSuspended(true); 9915 } 9916 } else { 9917 if (index < 0) { 9918 return; 9919 } 9920 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9921 effect->desc().type.timeLow); 9922 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9923 desc->mEffect.clear(); 9924 effect->setSuspended(false); 9925 } 9926} 9927 9928#undef LOG_TAG 9929#define LOG_TAG "AudioFlinger" 9930 9931// ---------------------------------------------------------------------------- 9932 9933status_t AudioFlinger::onTransact( 9934 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9935{ 9936 return BnAudioFlinger::onTransact(code, data, reply, flags); 9937} 9938 9939}; // namespace android 9940