AudioFlinger.cpp revision d65d73c4ae74d084751b417615a78cbe7a51372a
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, tid %d, calling pid %d", 876 ioHandle, keyValuePairs.string(), gettid(), 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, tid %d, calling pid %d", 961 ioHandle, keys.string(), gettid(), 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(uint32_t *halFrames, uint32_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, tid %d, calling tid %d", pid, gettid(), 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 >= (int) (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(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(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(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(uint32_t *halFrames, uint32_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 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2155 2156 if (isSuspended()) { 2157 // return an estimation of rendered frames when the output is suspended 2158 int32_t frames = mBytesWritten - latency_l(); 2159 if (frames < 0) { 2160 frames = 0; 2161 } 2162 *dspFrames = (uint32_t)frames; 2163 return NO_ERROR; 2164 } else { 2165 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2166 } 2167} 2168 2169uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2170{ 2171 Mutex::Autolock _l(mLock); 2172 uint32_t result = 0; 2173 if (getEffectChain_l(sessionId) != 0) { 2174 result = EFFECT_SESSION; 2175 } 2176 2177 for (size_t i = 0; i < mTracks.size(); ++i) { 2178 sp<Track> track = mTracks[i]; 2179 if (sessionId == track->sessionId() && 2180 !(track->mCblk->flags & CBLK_INVALID)) { 2181 result |= TRACK_SESSION; 2182 break; 2183 } 2184 } 2185 2186 return result; 2187} 2188 2189uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2190{ 2191 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2192 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2193 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2194 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2195 } 2196 for (size_t i = 0; i < mTracks.size(); i++) { 2197 sp<Track> track = mTracks[i]; 2198 if (sessionId == track->sessionId() && 2199 !(track->mCblk->flags & CBLK_INVALID)) { 2200 return AudioSystem::getStrategyForStream(track->streamType()); 2201 } 2202 } 2203 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2204} 2205 2206 2207AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2208{ 2209 Mutex::Autolock _l(mLock); 2210 return mOutput; 2211} 2212 2213AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2214{ 2215 Mutex::Autolock _l(mLock); 2216 AudioStreamOut *output = mOutput; 2217 mOutput = NULL; 2218 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2219 // must push a NULL and wait for ack 2220 mOutputSink.clear(); 2221 mPipeSink.clear(); 2222 mNormalSink.clear(); 2223 return output; 2224} 2225 2226// this method must always be called either with ThreadBase mLock held or inside the thread loop 2227audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2228{ 2229 if (mOutput == NULL) { 2230 return NULL; 2231 } 2232 return &mOutput->stream->common; 2233} 2234 2235uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2236{ 2237 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2238} 2239 2240status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2241{ 2242 if (!isValidSyncEvent(event)) { 2243 return BAD_VALUE; 2244 } 2245 2246 Mutex::Autolock _l(mLock); 2247 2248 for (size_t i = 0; i < mTracks.size(); ++i) { 2249 sp<Track> track = mTracks[i]; 2250 if (event->triggerSession() == track->sessionId()) { 2251 (void) track->setSyncEvent(event); 2252 return NO_ERROR; 2253 } 2254 } 2255 2256 return NAME_NOT_FOUND; 2257} 2258 2259bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2260{ 2261 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2262} 2263 2264void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2265 const Vector< sp<Track> >& tracksToRemove) 2266{ 2267 size_t count = tracksToRemove.size(); 2268 if (CC_UNLIKELY(count)) { 2269 for (size_t i = 0 ; i < count ; i++) { 2270 const sp<Track>& track = tracksToRemove.itemAt(i); 2271 if ((track->sharedBuffer() != 0) && 2272 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2273 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2274 } 2275 } 2276 } 2277 2278} 2279 2280// ---------------------------------------------------------------------------- 2281 2282AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2283 audio_io_handle_t id, audio_devices_t device, type_t type) 2284 : PlaybackThread(audioFlinger, output, id, device, type), 2285 // mAudioMixer below 2286 // mFastMixer below 2287 mFastMixerFutex(0) 2288 // mOutputSink below 2289 // mPipeSink below 2290 // mNormalSink below 2291{ 2292 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2293 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2294 "mFrameCount=%d, mNormalFrameCount=%d", 2295 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2296 mNormalFrameCount); 2297 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2298 2299 // FIXME - Current mixer implementation only supports stereo output 2300 if (mChannelCount != FCC_2) { 2301 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2302 } 2303 2304 // create an NBAIO sink for the HAL output stream, and negotiate 2305 mOutputSink = new AudioStreamOutSink(output->stream); 2306 size_t numCounterOffers = 0; 2307 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2308 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2309 ALOG_ASSERT(index == 0); 2310 2311 // initialize fast mixer depending on configuration 2312 bool initFastMixer; 2313 switch (kUseFastMixer) { 2314 case FastMixer_Never: 2315 initFastMixer = false; 2316 break; 2317 case FastMixer_Always: 2318 initFastMixer = true; 2319 break; 2320 case FastMixer_Static: 2321 case FastMixer_Dynamic: 2322 initFastMixer = mFrameCount < mNormalFrameCount; 2323 break; 2324 } 2325 if (initFastMixer) { 2326 2327 // create a MonoPipe to connect our submix to FastMixer 2328 NBAIO_Format format = mOutputSink->format(); 2329 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2330 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2331 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2332 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2333 const NBAIO_Format offers[1] = {format}; 2334 size_t numCounterOffers = 0; 2335 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2336 ALOG_ASSERT(index == 0); 2337 monoPipe->setAvgFrames((mScreenState & 1) ? 2338 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2339 mPipeSink = monoPipe; 2340 2341#ifdef TEE_SINK_FRAMES 2342 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2343 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2344 numCounterOffers = 0; 2345 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2346 ALOG_ASSERT(index == 0); 2347 mTeeSink = teeSink; 2348 PipeReader *teeSource = new PipeReader(*teeSink); 2349 numCounterOffers = 0; 2350 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2351 ALOG_ASSERT(index == 0); 2352 mTeeSource = teeSource; 2353#endif 2354 2355 // create fast mixer and configure it initially with just one fast track for our submix 2356 mFastMixer = new FastMixer(); 2357 FastMixerStateQueue *sq = mFastMixer->sq(); 2358#ifdef STATE_QUEUE_DUMP 2359 sq->setObserverDump(&mStateQueueObserverDump); 2360 sq->setMutatorDump(&mStateQueueMutatorDump); 2361#endif 2362 FastMixerState *state = sq->begin(); 2363 FastTrack *fastTrack = &state->mFastTracks[0]; 2364 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2365 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2366 fastTrack->mVolumeProvider = NULL; 2367 fastTrack->mGeneration++; 2368 state->mFastTracksGen++; 2369 state->mTrackMask = 1; 2370 // fast mixer will use the HAL output sink 2371 state->mOutputSink = mOutputSink.get(); 2372 state->mOutputSinkGen++; 2373 state->mFrameCount = mFrameCount; 2374 state->mCommand = FastMixerState::COLD_IDLE; 2375 // already done in constructor initialization list 2376 //mFastMixerFutex = 0; 2377 state->mColdFutexAddr = &mFastMixerFutex; 2378 state->mColdGen++; 2379 state->mDumpState = &mFastMixerDumpState; 2380 state->mTeeSink = mTeeSink.get(); 2381 sq->end(); 2382 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2383 2384 // start the fast mixer 2385 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2386 pid_t tid = mFastMixer->getTid(); 2387 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2388 if (err != 0) { 2389 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2390 kPriorityFastMixer, getpid_cached, tid, err); 2391 } 2392 2393#ifdef AUDIO_WATCHDOG 2394 // create and start the watchdog 2395 mAudioWatchdog = new AudioWatchdog(); 2396 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2397 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2398 tid = mAudioWatchdog->getTid(); 2399 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2400 if (err != 0) { 2401 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2402 kPriorityFastMixer, getpid_cached, tid, err); 2403 } 2404#endif 2405 2406 } else { 2407 mFastMixer = NULL; 2408 } 2409 2410 switch (kUseFastMixer) { 2411 case FastMixer_Never: 2412 case FastMixer_Dynamic: 2413 mNormalSink = mOutputSink; 2414 break; 2415 case FastMixer_Always: 2416 mNormalSink = mPipeSink; 2417 break; 2418 case FastMixer_Static: 2419 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2420 break; 2421 } 2422} 2423 2424AudioFlinger::MixerThread::~MixerThread() 2425{ 2426 if (mFastMixer != NULL) { 2427 FastMixerStateQueue *sq = mFastMixer->sq(); 2428 FastMixerState *state = sq->begin(); 2429 if (state->mCommand == FastMixerState::COLD_IDLE) { 2430 int32_t old = android_atomic_inc(&mFastMixerFutex); 2431 if (old == -1) { 2432 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2433 } 2434 } 2435 state->mCommand = FastMixerState::EXIT; 2436 sq->end(); 2437 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2438 mFastMixer->join(); 2439 // Though the fast mixer thread has exited, it's state queue is still valid. 2440 // We'll use that extract the final state which contains one remaining fast track 2441 // corresponding to our sub-mix. 2442 state = sq->begin(); 2443 ALOG_ASSERT(state->mTrackMask == 1); 2444 FastTrack *fastTrack = &state->mFastTracks[0]; 2445 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2446 delete fastTrack->mBufferProvider; 2447 sq->end(false /*didModify*/); 2448 delete mFastMixer; 2449#ifdef AUDIO_WATCHDOG 2450 if (mAudioWatchdog != 0) { 2451 mAudioWatchdog->requestExit(); 2452 mAudioWatchdog->requestExitAndWait(); 2453 mAudioWatchdog.clear(); 2454 } 2455#endif 2456 } 2457 delete mAudioMixer; 2458} 2459 2460class CpuStats { 2461public: 2462 CpuStats(); 2463 void sample(const String8 &title); 2464#ifdef DEBUG_CPU_USAGE 2465private: 2466 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2467 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2468 2469 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2470 2471 int mCpuNum; // thread's current CPU number 2472 int mCpukHz; // frequency of thread's current CPU in kHz 2473#endif 2474}; 2475 2476CpuStats::CpuStats() 2477#ifdef DEBUG_CPU_USAGE 2478 : mCpuNum(-1), mCpukHz(-1) 2479#endif 2480{ 2481} 2482 2483void CpuStats::sample(const String8 &title) { 2484#ifdef DEBUG_CPU_USAGE 2485 // get current thread's delta CPU time in wall clock ns 2486 double wcNs; 2487 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2488 2489 // record sample for wall clock statistics 2490 if (valid) { 2491 mWcStats.sample(wcNs); 2492 } 2493 2494 // get the current CPU number 2495 int cpuNum = sched_getcpu(); 2496 2497 // get the current CPU frequency in kHz 2498 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2499 2500 // check if either CPU number or frequency changed 2501 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2502 mCpuNum = cpuNum; 2503 mCpukHz = cpukHz; 2504 // ignore sample for purposes of cycles 2505 valid = false; 2506 } 2507 2508 // if no change in CPU number or frequency, then record sample for cycle statistics 2509 if (valid && mCpukHz > 0) { 2510 double cycles = wcNs * cpukHz * 0.000001; 2511 mHzStats.sample(cycles); 2512 } 2513 2514 unsigned n = mWcStats.n(); 2515 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2516 if ((n & 127) == 1) { 2517 long long elapsed = mCpuUsage.elapsed(); 2518 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2519 double perLoop = elapsed / (double) n; 2520 double perLoop100 = perLoop * 0.01; 2521 double perLoop1k = perLoop * 0.001; 2522 double mean = mWcStats.mean(); 2523 double stddev = mWcStats.stddev(); 2524 double minimum = mWcStats.minimum(); 2525 double maximum = mWcStats.maximum(); 2526 double meanCycles = mHzStats.mean(); 2527 double stddevCycles = mHzStats.stddev(); 2528 double minCycles = mHzStats.minimum(); 2529 double maxCycles = mHzStats.maximum(); 2530 mCpuUsage.resetElapsed(); 2531 mWcStats.reset(); 2532 mHzStats.reset(); 2533 ALOGD("CPU usage for %s over past %.1f secs\n" 2534 " (%u mixer loops at %.1f mean ms per loop):\n" 2535 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2536 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2537 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2538 title.string(), 2539 elapsed * .000000001, n, perLoop * .000001, 2540 mean * .001, 2541 stddev * .001, 2542 minimum * .001, 2543 maximum * .001, 2544 mean / perLoop100, 2545 stddev / perLoop100, 2546 minimum / perLoop100, 2547 maximum / perLoop100, 2548 meanCycles / perLoop1k, 2549 stddevCycles / perLoop1k, 2550 minCycles / perLoop1k, 2551 maxCycles / perLoop1k); 2552 2553 } 2554 } 2555#endif 2556}; 2557 2558void AudioFlinger::PlaybackThread::checkSilentMode_l() 2559{ 2560 if (!mMasterMute) { 2561 char value[PROPERTY_VALUE_MAX]; 2562 if (property_get("ro.audio.silent", value, "0") > 0) { 2563 char *endptr; 2564 unsigned long ul = strtoul(value, &endptr, 0); 2565 if (*endptr == '\0' && ul != 0) { 2566 ALOGD("Silence is golden"); 2567 // The setprop command will not allow a property to be changed after 2568 // the first time it is set, so we don't have to worry about un-muting. 2569 setMasterMute_l(true); 2570 } 2571 } 2572 } 2573} 2574 2575bool AudioFlinger::PlaybackThread::threadLoop() 2576{ 2577 Vector< sp<Track> > tracksToRemove; 2578 2579 standbyTime = systemTime(); 2580 2581 // MIXER 2582 nsecs_t lastWarning = 0; 2583 2584 // DUPLICATING 2585 // FIXME could this be made local to while loop? 2586 writeFrames = 0; 2587 2588 cacheParameters_l(); 2589 sleepTime = idleSleepTime; 2590 2591 if (mType == MIXER) { 2592 sleepTimeShift = 0; 2593 } 2594 2595 CpuStats cpuStats; 2596 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2597 2598 acquireWakeLock(); 2599 2600 while (!exitPending()) 2601 { 2602 cpuStats.sample(myName); 2603 2604 Vector< sp<EffectChain> > effectChains; 2605 2606 processConfigEvents(); 2607 2608 { // scope for mLock 2609 2610 Mutex::Autolock _l(mLock); 2611 2612 if (checkForNewParameters_l()) { 2613 cacheParameters_l(); 2614 } 2615 2616 saveOutputTracks(); 2617 2618 // put audio hardware into standby after short delay 2619 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2620 isSuspended())) { 2621 if (!mStandby) { 2622 2623 threadLoop_standby(); 2624 2625 mStandby = true; 2626 } 2627 2628 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2629 // we're about to wait, flush the binder command buffer 2630 IPCThreadState::self()->flushCommands(); 2631 2632 clearOutputTracks(); 2633 2634 if (exitPending()) { 2635 break; 2636 } 2637 2638 releaseWakeLock_l(); 2639 // wait until we have something to do... 2640 ALOGV("%s going to sleep", myName.string()); 2641 mWaitWorkCV.wait(mLock); 2642 ALOGV("%s waking up", myName.string()); 2643 acquireWakeLock_l(); 2644 2645 mMixerStatus = MIXER_IDLE; 2646 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2647 mBytesWritten = 0; 2648 2649 checkSilentMode_l(); 2650 2651 standbyTime = systemTime() + standbyDelay; 2652 sleepTime = idleSleepTime; 2653 if (mType == MIXER) { 2654 sleepTimeShift = 0; 2655 } 2656 2657 continue; 2658 } 2659 } 2660 2661 // mMixerStatusIgnoringFastTracks is also updated internally 2662 mMixerStatus = prepareTracks_l(&tracksToRemove); 2663 2664 // prevent any changes in effect chain list and in each effect chain 2665 // during mixing and effect process as the audio buffers could be deleted 2666 // or modified if an effect is created or deleted 2667 lockEffectChains_l(effectChains); 2668 } 2669 2670 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2671 threadLoop_mix(); 2672 } else { 2673 threadLoop_sleepTime(); 2674 } 2675 2676 if (isSuspended()) { 2677 sleepTime = suspendSleepTimeUs(); 2678 mBytesWritten += mixBufferSize; 2679 } 2680 2681 // only process effects if we're going to write 2682 if (sleepTime == 0) { 2683 for (size_t i = 0; i < effectChains.size(); i ++) { 2684 effectChains[i]->process_l(); 2685 } 2686 } 2687 2688 // enable changes in effect chain 2689 unlockEffectChains(effectChains); 2690 2691 // sleepTime == 0 means we must write to audio hardware 2692 if (sleepTime == 0) { 2693 2694 threadLoop_write(); 2695 2696if (mType == MIXER) { 2697 // write blocked detection 2698 nsecs_t now = systemTime(); 2699 nsecs_t delta = now - mLastWriteTime; 2700 if (!mStandby && delta > maxPeriod) { 2701 mNumDelayedWrites++; 2702 if ((now - lastWarning) > kWarningThrottleNs) { 2703#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2704 ScopedTrace st(ATRACE_TAG, "underrun"); 2705#endif 2706 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2707 ns2ms(delta), mNumDelayedWrites, this); 2708 lastWarning = now; 2709 } 2710 } 2711} 2712 2713 mStandby = false; 2714 } else { 2715 usleep(sleepTime); 2716 } 2717 2718 // Finally let go of removed track(s), without the lock held 2719 // since we can't guarantee the destructors won't acquire that 2720 // same lock. This will also mutate and push a new fast mixer state. 2721 threadLoop_removeTracks(tracksToRemove); 2722 tracksToRemove.clear(); 2723 2724 // FIXME I don't understand the need for this here; 2725 // it was in the original code but maybe the 2726 // assignment in saveOutputTracks() makes this unnecessary? 2727 clearOutputTracks(); 2728 2729 // Effect chains will be actually deleted here if they were removed from 2730 // mEffectChains list during mixing or effects processing 2731 effectChains.clear(); 2732 2733 // FIXME Note that the above .clear() is no longer necessary since effectChains 2734 // is now local to this block, but will keep it for now (at least until merge done). 2735 } 2736 2737 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2738 if (mType == MIXER || mType == DIRECT) { 2739 // put output stream into standby mode 2740 if (!mStandby) { 2741 mOutput->stream->common.standby(&mOutput->stream->common); 2742 } 2743 } 2744 2745 releaseWakeLock(); 2746 2747 ALOGV("Thread %p type %d exiting", this, mType); 2748 return false; 2749} 2750 2751void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2752{ 2753 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2754} 2755 2756void AudioFlinger::MixerThread::threadLoop_write() 2757{ 2758 // FIXME we should only do one push per cycle; confirm this is true 2759 // Start the fast mixer if it's not already running 2760 if (mFastMixer != NULL) { 2761 FastMixerStateQueue *sq = mFastMixer->sq(); 2762 FastMixerState *state = sq->begin(); 2763 if (state->mCommand != FastMixerState::MIX_WRITE && 2764 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2765 if (state->mCommand == FastMixerState::COLD_IDLE) { 2766 int32_t old = android_atomic_inc(&mFastMixerFutex); 2767 if (old == -1) { 2768 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2769 } 2770#ifdef AUDIO_WATCHDOG 2771 if (mAudioWatchdog != 0) { 2772 mAudioWatchdog->resume(); 2773 } 2774#endif 2775 } 2776 state->mCommand = FastMixerState::MIX_WRITE; 2777 sq->end(); 2778 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2779 if (kUseFastMixer == FastMixer_Dynamic) { 2780 mNormalSink = mPipeSink; 2781 } 2782 } else { 2783 sq->end(false /*didModify*/); 2784 } 2785 } 2786 PlaybackThread::threadLoop_write(); 2787} 2788 2789// shared by MIXER and DIRECT, overridden by DUPLICATING 2790void AudioFlinger::PlaybackThread::threadLoop_write() 2791{ 2792 // FIXME rewrite to reduce number of system calls 2793 mLastWriteTime = systemTime(); 2794 mInWrite = true; 2795 int bytesWritten; 2796 2797 // If an NBAIO sink is present, use it to write the normal mixer's submix 2798 if (mNormalSink != 0) { 2799#define mBitShift 2 // FIXME 2800 size_t count = mixBufferSize >> mBitShift; 2801#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2802 Tracer::traceBegin(ATRACE_TAG, "write"); 2803#endif 2804 // update the setpoint when gScreenState changes 2805 uint32_t screenState = gScreenState; 2806 if (screenState != mScreenState) { 2807 mScreenState = screenState; 2808 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2809 if (pipe != NULL) { 2810 pipe->setAvgFrames((mScreenState & 1) ? 2811 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2812 } 2813 } 2814 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2815#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2816 Tracer::traceEnd(ATRACE_TAG); 2817#endif 2818 if (framesWritten > 0) { 2819 bytesWritten = framesWritten << mBitShift; 2820 } else { 2821 bytesWritten = framesWritten; 2822 } 2823 // otherwise use the HAL / AudioStreamOut directly 2824 } else { 2825 // Direct output thread. 2826 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2827 } 2828 2829 if (bytesWritten > 0) { 2830 mBytesWritten += mixBufferSize; 2831 } 2832 mNumWrites++; 2833 mInWrite = false; 2834} 2835 2836void AudioFlinger::MixerThread::threadLoop_standby() 2837{ 2838 // Idle the fast mixer if it's currently running 2839 if (mFastMixer != NULL) { 2840 FastMixerStateQueue *sq = mFastMixer->sq(); 2841 FastMixerState *state = sq->begin(); 2842 if (!(state->mCommand & FastMixerState::IDLE)) { 2843 state->mCommand = FastMixerState::COLD_IDLE; 2844 state->mColdFutexAddr = &mFastMixerFutex; 2845 state->mColdGen++; 2846 mFastMixerFutex = 0; 2847 sq->end(); 2848 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2849 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2850 if (kUseFastMixer == FastMixer_Dynamic) { 2851 mNormalSink = mOutputSink; 2852 } 2853#ifdef AUDIO_WATCHDOG 2854 if (mAudioWatchdog != 0) { 2855 mAudioWatchdog->pause(); 2856 } 2857#endif 2858 } else { 2859 sq->end(false /*didModify*/); 2860 } 2861 } 2862 PlaybackThread::threadLoop_standby(); 2863} 2864 2865// shared by MIXER and DIRECT, overridden by DUPLICATING 2866void AudioFlinger::PlaybackThread::threadLoop_standby() 2867{ 2868 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2869 mOutput->stream->common.standby(&mOutput->stream->common); 2870} 2871 2872void AudioFlinger::MixerThread::threadLoop_mix() 2873{ 2874 // obtain the presentation timestamp of the next output buffer 2875 int64_t pts; 2876 status_t status = INVALID_OPERATION; 2877 2878 if (mNormalSink != 0) { 2879 status = mNormalSink->getNextWriteTimestamp(&pts); 2880 } else { 2881 status = mOutputSink->getNextWriteTimestamp(&pts); 2882 } 2883 2884 if (status != NO_ERROR) { 2885 pts = AudioBufferProvider::kInvalidPTS; 2886 } 2887 2888 // mix buffers... 2889 mAudioMixer->process(pts); 2890 // increase sleep time progressively when application underrun condition clears. 2891 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2892 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2893 // such that we would underrun the audio HAL. 2894 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2895 sleepTimeShift--; 2896 } 2897 sleepTime = 0; 2898 standbyTime = systemTime() + standbyDelay; 2899 //TODO: delay standby when effects have a tail 2900} 2901 2902void AudioFlinger::MixerThread::threadLoop_sleepTime() 2903{ 2904 // If no tracks are ready, sleep once for the duration of an output 2905 // buffer size, then write 0s to the output 2906 if (sleepTime == 0) { 2907 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2908 sleepTime = activeSleepTime >> sleepTimeShift; 2909 if (sleepTime < kMinThreadSleepTimeUs) { 2910 sleepTime = kMinThreadSleepTimeUs; 2911 } 2912 // reduce sleep time in case of consecutive application underruns to avoid 2913 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2914 // duration we would end up writing less data than needed by the audio HAL if 2915 // the condition persists. 2916 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2917 sleepTimeShift++; 2918 } 2919 } else { 2920 sleepTime = idleSleepTime; 2921 } 2922 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2923 memset (mMixBuffer, 0, mixBufferSize); 2924 sleepTime = 0; 2925 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), 2926 "anticipated start"); 2927 } 2928 // TODO add standby time extension fct of effect tail 2929} 2930 2931// prepareTracks_l() must be called with ThreadBase::mLock held 2932AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2933 Vector< sp<Track> > *tracksToRemove) 2934{ 2935 2936 mixer_state mixerStatus = MIXER_IDLE; 2937 // find out which tracks need to be processed 2938 size_t count = mActiveTracks.size(); 2939 size_t mixedTracks = 0; 2940 size_t tracksWithEffect = 0; 2941 // counts only _active_ fast tracks 2942 size_t fastTracks = 0; 2943 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2944 2945 float masterVolume = mMasterVolume; 2946 bool masterMute = mMasterMute; 2947 2948 if (masterMute) { 2949 masterVolume = 0; 2950 } 2951 // Delegate master volume control to effect in output mix effect chain if needed 2952 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2953 if (chain != 0) { 2954 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2955 chain->setVolume_l(&v, &v); 2956 masterVolume = (float)((v + (1 << 23)) >> 24); 2957 chain.clear(); 2958 } 2959 2960 // prepare a new state to push 2961 FastMixerStateQueue *sq = NULL; 2962 FastMixerState *state = NULL; 2963 bool didModify = false; 2964 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2965 if (mFastMixer != NULL) { 2966 sq = mFastMixer->sq(); 2967 state = sq->begin(); 2968 } 2969 2970 for (size_t i=0 ; i<count ; i++) { 2971 sp<Track> t = mActiveTracks[i].promote(); 2972 if (t == 0) { 2973 continue; 2974 } 2975 2976 // this const just means the local variable doesn't change 2977 Track* const track = t.get(); 2978 2979 // process fast tracks 2980 if (track->isFastTrack()) { 2981 2982 // It's theoretically possible (though unlikely) for a fast track to be created 2983 // and then removed within the same normal mix cycle. This is not a problem, as 2984 // the track never becomes active so it's fast mixer slot is never touched. 2985 // The converse, of removing an (active) track and then creating a new track 2986 // at the identical fast mixer slot within the same normal mix cycle, 2987 // is impossible because the slot isn't marked available until the end of each cycle. 2988 int j = track->mFastIndex; 2989 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2990 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2991 FastTrack *fastTrack = &state->mFastTracks[j]; 2992 2993 // Determine whether the track is currently in underrun condition, 2994 // and whether it had a recent underrun. 2995 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2996 FastTrackUnderruns underruns = ftDump->mUnderruns; 2997 uint32_t recentFull = (underruns.mBitFields.mFull - 2998 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2999 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3000 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3001 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3002 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3003 uint32_t recentUnderruns = recentPartial + recentEmpty; 3004 track->mObservedUnderruns = underruns; 3005 // don't count underruns that occur while stopping or pausing 3006 // or stopped which can occur when flush() is called while active 3007 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 3008 track->mUnderrunCount += recentUnderruns; 3009 } 3010 3011 // This is similar to the state machine for normal tracks, 3012 // with a few modifications for fast tracks. 3013 bool isActive = true; 3014 switch (track->mState) { 3015 case TrackBase::STOPPING_1: 3016 // track stays active in STOPPING_1 state until first underrun 3017 if (recentUnderruns > 0) { 3018 track->mState = TrackBase::STOPPING_2; 3019 } 3020 break; 3021 case TrackBase::PAUSING: 3022 // ramp down is not yet implemented 3023 track->setPaused(); 3024 break; 3025 case TrackBase::RESUMING: 3026 // ramp up is not yet implemented 3027 track->mState = TrackBase::ACTIVE; 3028 break; 3029 case TrackBase::ACTIVE: 3030 if (recentFull > 0 || recentPartial > 0) { 3031 // track has provided at least some frames recently: reset retry count 3032 track->mRetryCount = kMaxTrackRetries; 3033 } 3034 if (recentUnderruns == 0) { 3035 // no recent underruns: stay active 3036 break; 3037 } 3038 // there has recently been an underrun of some kind 3039 if (track->sharedBuffer() == 0) { 3040 // were any of the recent underruns "empty" (no frames available)? 3041 if (recentEmpty == 0) { 3042 // no, then ignore the partial underruns as they are allowed indefinitely 3043 break; 3044 } 3045 // there has recently been an "empty" underrun: decrement the retry counter 3046 if (--(track->mRetryCount) > 0) { 3047 break; 3048 } 3049 // indicate to client process that the track was disabled because of underrun; 3050 // it will then automatically call start() when data is available 3051 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 3052 // remove from active list, but state remains ACTIVE [confusing but true] 3053 isActive = false; 3054 break; 3055 } 3056 // fall through 3057 case TrackBase::STOPPING_2: 3058 case TrackBase::PAUSED: 3059 case TrackBase::TERMINATED: 3060 case TrackBase::STOPPED: 3061 case TrackBase::FLUSHED: // flush() while active 3062 // Check for presentation complete if track is inactive 3063 // We have consumed all the buffers of this track. 3064 // This would be incomplete if we auto-paused on underrun 3065 { 3066 size_t audioHALFrames = 3067 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3068 size_t framesWritten = 3069 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3070 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3071 // track stays in active list until presentation is complete 3072 break; 3073 } 3074 } 3075 if (track->isStopping_2()) { 3076 track->mState = TrackBase::STOPPED; 3077 } 3078 if (track->isStopped()) { 3079 // Can't reset directly, as fast mixer is still polling this track 3080 // track->reset(); 3081 // So instead mark this track as needing to be reset after push with ack 3082 resetMask |= 1 << i; 3083 } 3084 isActive = false; 3085 break; 3086 case TrackBase::IDLE: 3087 default: 3088 LOG_FATAL("unexpected track state %d", track->mState); 3089 } 3090 3091 if (isActive) { 3092 // was it previously inactive? 3093 if (!(state->mTrackMask & (1 << j))) { 3094 ExtendedAudioBufferProvider *eabp = track; 3095 VolumeProvider *vp = track; 3096 fastTrack->mBufferProvider = eabp; 3097 fastTrack->mVolumeProvider = vp; 3098 fastTrack->mSampleRate = track->mSampleRate; 3099 fastTrack->mChannelMask = track->mChannelMask; 3100 fastTrack->mGeneration++; 3101 state->mTrackMask |= 1 << j; 3102 didModify = true; 3103 // no acknowledgement required for newly active tracks 3104 } 3105 // cache the combined master volume and stream type volume for fast mixer; this 3106 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3107 track->mCachedVolume = track->isMuted() ? 3108 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3109 ++fastTracks; 3110 } else { 3111 // was it previously active? 3112 if (state->mTrackMask & (1 << j)) { 3113 fastTrack->mBufferProvider = NULL; 3114 fastTrack->mGeneration++; 3115 state->mTrackMask &= ~(1 << j); 3116 didModify = true; 3117 // If any fast tracks were removed, we must wait for acknowledgement 3118 // because we're about to decrement the last sp<> on those tracks. 3119 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3120 } else { 3121 LOG_FATAL("fast track %d should have been active", j); 3122 } 3123 tracksToRemove->add(track); 3124 // Avoids a misleading display in dumpsys 3125 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3126 } 3127 continue; 3128 } 3129 3130 { // local variable scope to avoid goto warning 3131 3132 audio_track_cblk_t* cblk = track->cblk(); 3133 3134 // The first time a track is added we wait 3135 // for all its buffers to be filled before processing it 3136 int name = track->name(); 3137 // make sure that we have enough frames to mix one full buffer. 3138 // enforce this condition only once to enable draining the buffer in case the client 3139 // app does not call stop() and relies on underrun to stop: 3140 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3141 // during last round 3142 uint32_t minFrames = 1; 3143 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3144 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3145 if (t->sampleRate() == mSampleRate) { 3146 minFrames = mNormalFrameCount; 3147 } else { 3148 // +1 for rounding and +1 for additional sample needed for interpolation 3149 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3150 // add frames already consumed but not yet released by the resampler 3151 // because cblk->framesReady() will include these frames 3152 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3153 // the minimum track buffer size is normally twice the number of frames necessary 3154 // to fill one buffer and the resampler should not leave more than one buffer worth 3155 // of unreleased frames after each pass, but just in case... 3156 ALOG_ASSERT(minFrames <= cblk->frameCount); 3157 } 3158 } 3159 if ((track->framesReady() >= minFrames) && track->isReady() && 3160 !track->isPaused() && !track->isTerminated()) 3161 { 3162 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 3163 this); 3164 3165 mixedTracks++; 3166 3167 // track->mainBuffer() != mMixBuffer means there is an effect chain 3168 // connected to the track 3169 chain.clear(); 3170 if (track->mainBuffer() != mMixBuffer) { 3171 chain = getEffectChain_l(track->sessionId()); 3172 // Delegate volume control to effect in track effect chain if needed 3173 if (chain != 0) { 3174 tracksWithEffect++; 3175 } else { 3176 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3177 "session %d", 3178 name, track->sessionId()); 3179 } 3180 } 3181 3182 3183 int param = AudioMixer::VOLUME; 3184 if (track->mFillingUpStatus == Track::FS_FILLED) { 3185 // no ramp for the first volume setting 3186 track->mFillingUpStatus = Track::FS_ACTIVE; 3187 if (track->mState == TrackBase::RESUMING) { 3188 track->mState = TrackBase::ACTIVE; 3189 param = AudioMixer::RAMP_VOLUME; 3190 } 3191 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3192 } else if (cblk->server != 0) { 3193 // If the track is stopped before the first frame was mixed, 3194 // do not apply ramp 3195 param = AudioMixer::RAMP_VOLUME; 3196 } 3197 3198 // compute volume for this track 3199 uint32_t vl, vr, va; 3200 if (track->isMuted() || track->isPausing() || 3201 mStreamTypes[track->streamType()].mute) { 3202 vl = vr = va = 0; 3203 if (track->isPausing()) { 3204 track->setPaused(); 3205 } 3206 } else { 3207 3208 // read original volumes with volume control 3209 float typeVolume = mStreamTypes[track->streamType()].volume; 3210 float v = masterVolume * typeVolume; 3211 uint32_t vlr = cblk->getVolumeLR(); 3212 vl = vlr & 0xFFFF; 3213 vr = vlr >> 16; 3214 // track volumes come from shared memory, so can't be trusted and must be clamped 3215 if (vl > MAX_GAIN_INT) { 3216 ALOGV("Track left volume out of range: %04X", vl); 3217 vl = MAX_GAIN_INT; 3218 } 3219 if (vr > MAX_GAIN_INT) { 3220 ALOGV("Track right volume out of range: %04X", vr); 3221 vr = MAX_GAIN_INT; 3222 } 3223 // now apply the master volume and stream type volume 3224 vl = (uint32_t)(v * vl) << 12; 3225 vr = (uint32_t)(v * vr) << 12; 3226 // assuming master volume and stream type volume each go up to 1.0, 3227 // vl and vr are now in 8.24 format 3228 3229 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3230 // send level comes from shared memory and so may be corrupt 3231 if (sendLevel > MAX_GAIN_INT) { 3232 ALOGV("Track send level out of range: %04X", sendLevel); 3233 sendLevel = MAX_GAIN_INT; 3234 } 3235 va = (uint32_t)(v * sendLevel); 3236 } 3237 // Delegate volume control to effect in track effect chain if needed 3238 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3239 // Do not ramp volume if volume is controlled by effect 3240 param = AudioMixer::VOLUME; 3241 track->mHasVolumeController = true; 3242 } else { 3243 // force no volume ramp when volume controller was just disabled or removed 3244 // from effect chain to avoid volume spike 3245 if (track->mHasVolumeController) { 3246 param = AudioMixer::VOLUME; 3247 } 3248 track->mHasVolumeController = false; 3249 } 3250 3251 // Convert volumes from 8.24 to 4.12 format 3252 // This additional clamping is needed in case chain->setVolume_l() overshot 3253 vl = (vl + (1 << 11)) >> 12; 3254 if (vl > MAX_GAIN_INT) { 3255 vl = MAX_GAIN_INT; 3256 } 3257 vr = (vr + (1 << 11)) >> 12; 3258 if (vr > MAX_GAIN_INT) { 3259 vr = MAX_GAIN_INT; 3260 } 3261 3262 if (va > MAX_GAIN_INT) { 3263 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3264 } 3265 3266 // XXX: these things DON'T need to be done each time 3267 mAudioMixer->setBufferProvider(name, track); 3268 mAudioMixer->enable(name); 3269 3270 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3271 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3272 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3273 mAudioMixer->setParameter( 3274 name, 3275 AudioMixer::TRACK, 3276 AudioMixer::FORMAT, (void *)track->format()); 3277 mAudioMixer->setParameter( 3278 name, 3279 AudioMixer::TRACK, 3280 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3281 mAudioMixer->setParameter( 3282 name, 3283 AudioMixer::RESAMPLE, 3284 AudioMixer::SAMPLE_RATE, 3285 (void *)(cblk->sampleRate)); 3286 mAudioMixer->setParameter( 3287 name, 3288 AudioMixer::TRACK, 3289 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3290 mAudioMixer->setParameter( 3291 name, 3292 AudioMixer::TRACK, 3293 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3294 3295 // reset retry count 3296 track->mRetryCount = kMaxTrackRetries; 3297 3298 // If one track is ready, set the mixer ready if: 3299 // - the mixer was not ready during previous round OR 3300 // - no other track is not ready 3301 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3302 mixerStatus != MIXER_TRACKS_ENABLED) { 3303 mixerStatus = MIXER_TRACKS_READY; 3304 } 3305 } else { 3306 // clear effect chain input buffer if an active track underruns to avoid sending 3307 // previous audio buffer again to effects 3308 chain = getEffectChain_l(track->sessionId()); 3309 if (chain != 0) { 3310 chain->clearInputBuffer(); 3311 } 3312 3313 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 3314 cblk->server, this); 3315 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3316 track->isStopped() || track->isPaused()) { 3317 // We have consumed all the buffers of this track. 3318 // Remove it from the list of active tracks. 3319 // TODO: use actual buffer filling status instead of latency when available from 3320 // audio HAL 3321 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3322 size_t framesWritten = 3323 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3324 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3325 if (track->isStopped()) { 3326 track->reset(); 3327 } 3328 tracksToRemove->add(track); 3329 } 3330 } else { 3331 track->mUnderrunCount++; 3332 // No buffers for this track. Give it a few chances to 3333 // fill a buffer, then remove it from active list. 3334 if (--(track->mRetryCount) <= 0) { 3335 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3336 tracksToRemove->add(track); 3337 // indicate to client process that the track was disabled because of underrun; 3338 // it will then automatically call start() when data is available 3339 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3340 // If one track is not ready, mark the mixer also not ready if: 3341 // - the mixer was ready during previous round OR 3342 // - no other track is ready 3343 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3344 mixerStatus != MIXER_TRACKS_READY) { 3345 mixerStatus = MIXER_TRACKS_ENABLED; 3346 } 3347 } 3348 mAudioMixer->disable(name); 3349 } 3350 3351 } // local variable scope to avoid goto warning 3352track_is_ready: ; 3353 3354 } 3355 3356 // Push the new FastMixer state if necessary 3357 bool pauseAudioWatchdog = false; 3358 if (didModify) { 3359 state->mFastTracksGen++; 3360 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3361 if (kUseFastMixer == FastMixer_Dynamic && 3362 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3363 state->mCommand = FastMixerState::COLD_IDLE; 3364 state->mColdFutexAddr = &mFastMixerFutex; 3365 state->mColdGen++; 3366 mFastMixerFutex = 0; 3367 if (kUseFastMixer == FastMixer_Dynamic) { 3368 mNormalSink = mOutputSink; 3369 } 3370 // If we go into cold idle, need to wait for acknowledgement 3371 // so that fast mixer stops doing I/O. 3372 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3373 pauseAudioWatchdog = true; 3374 } 3375 sq->end(); 3376 } 3377 if (sq != NULL) { 3378 sq->end(didModify); 3379 sq->push(block); 3380 } 3381#ifdef AUDIO_WATCHDOG 3382 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3383 mAudioWatchdog->pause(); 3384 } 3385#endif 3386 3387 // Now perform the deferred reset on fast tracks that have stopped 3388 while (resetMask != 0) { 3389 size_t i = __builtin_ctz(resetMask); 3390 ALOG_ASSERT(i < count); 3391 resetMask &= ~(1 << i); 3392 sp<Track> t = mActiveTracks[i].promote(); 3393 if (t == 0) { 3394 continue; 3395 } 3396 Track* track = t.get(); 3397 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3398 track->reset(); 3399 } 3400 3401 // remove all the tracks that need to be... 3402 count = tracksToRemove->size(); 3403 if (CC_UNLIKELY(count)) { 3404 for (size_t i=0 ; i<count ; i++) { 3405 const sp<Track>& track = tracksToRemove->itemAt(i); 3406 mActiveTracks.remove(track); 3407 if (track->mainBuffer() != mMixBuffer) { 3408 chain = getEffectChain_l(track->sessionId()); 3409 if (chain != 0) { 3410 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3411 track->sessionId()); 3412 chain->decActiveTrackCnt(); 3413 } 3414 } 3415 if (track->isTerminated()) { 3416 removeTrack_l(track); 3417 } 3418 } 3419 } 3420 3421 // mix buffer must be cleared if all tracks are connected to an 3422 // effect chain as in this case the mixer will not write to 3423 // mix buffer and track effects will accumulate into it 3424 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3425 (mixedTracks == 0 && fastTracks > 0)) { 3426 // FIXME as a performance optimization, should remember previous zero status 3427 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3428 } 3429 3430 // if any fast tracks, then status is ready 3431 mMixerStatusIgnoringFastTracks = mixerStatus; 3432 if (fastTracks > 0) { 3433 mixerStatus = MIXER_TRACKS_READY; 3434 } 3435 return mixerStatus; 3436} 3437 3438/* 3439The derived values that are cached: 3440 - mixBufferSize from frame count * frame size 3441 - activeSleepTime from activeSleepTimeUs() 3442 - idleSleepTime from idleSleepTimeUs() 3443 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3444 - maxPeriod from frame count and sample rate (MIXER only) 3445 3446The parameters that affect these derived values are: 3447 - frame count 3448 - frame size 3449 - sample rate 3450 - device type: A2DP or not 3451 - device latency 3452 - format: PCM or not 3453 - active sleep time 3454 - idle sleep time 3455*/ 3456 3457void AudioFlinger::PlaybackThread::cacheParameters_l() 3458{ 3459 mixBufferSize = mNormalFrameCount * mFrameSize; 3460 activeSleepTime = activeSleepTimeUs(); 3461 idleSleepTime = idleSleepTimeUs(); 3462} 3463 3464void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3465{ 3466 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3467 this, streamType, mTracks.size()); 3468 Mutex::Autolock _l(mLock); 3469 3470 size_t size = mTracks.size(); 3471 for (size_t i = 0; i < size; i++) { 3472 sp<Track> t = mTracks[i]; 3473 if (t->streamType() == streamType) { 3474 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 3475 t->mCblk->cv.signal(); 3476 } 3477 } 3478} 3479 3480// getTrackName_l() must be called with ThreadBase::mLock held 3481int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3482{ 3483 return mAudioMixer->getTrackName(channelMask, sessionId); 3484} 3485 3486// deleteTrackName_l() must be called with ThreadBase::mLock held 3487void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3488{ 3489 ALOGV("remove track (%d) and delete from mixer", name); 3490 mAudioMixer->deleteTrackName(name); 3491} 3492 3493// checkForNewParameters_l() must be called with ThreadBase::mLock held 3494bool AudioFlinger::MixerThread::checkForNewParameters_l() 3495{ 3496 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3497 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3498 bool reconfig = false; 3499 3500 while (!mNewParameters.isEmpty()) { 3501 3502 if (mFastMixer != NULL) { 3503 FastMixerStateQueue *sq = mFastMixer->sq(); 3504 FastMixerState *state = sq->begin(); 3505 if (!(state->mCommand & FastMixerState::IDLE)) { 3506 previousCommand = state->mCommand; 3507 state->mCommand = FastMixerState::HOT_IDLE; 3508 sq->end(); 3509 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3510 } else { 3511 sq->end(false /*didModify*/); 3512 } 3513 } 3514 3515 status_t status = NO_ERROR; 3516 String8 keyValuePair = mNewParameters[0]; 3517 AudioParameter param = AudioParameter(keyValuePair); 3518 int value; 3519 3520 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3521 reconfig = true; 3522 } 3523 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3524 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3525 status = BAD_VALUE; 3526 } else { 3527 reconfig = true; 3528 } 3529 } 3530 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3531 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3532 status = BAD_VALUE; 3533 } else { 3534 reconfig = true; 3535 } 3536 } 3537 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3538 // do not accept frame count changes if tracks are open as the track buffer 3539 // size depends on frame count and correct behavior would not be guaranteed 3540 // if frame count is changed after track creation 3541 if (!mTracks.isEmpty()) { 3542 status = INVALID_OPERATION; 3543 } else { 3544 reconfig = true; 3545 } 3546 } 3547 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3548#ifdef ADD_BATTERY_DATA 3549 // when changing the audio output device, call addBatteryData to notify 3550 // the change 3551 if (mOutDevice != value) { 3552 uint32_t params = 0; 3553 // check whether speaker is on 3554 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3555 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3556 } 3557 3558 audio_devices_t deviceWithoutSpeaker 3559 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3560 // check if any other device (except speaker) is on 3561 if (value & deviceWithoutSpeaker ) { 3562 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3563 } 3564 3565 if (params != 0) { 3566 addBatteryData(params); 3567 } 3568 } 3569#endif 3570 3571 // forward device change to effects that have requested to be 3572 // aware of attached audio device. 3573 mOutDevice = value; 3574 for (size_t i = 0; i < mEffectChains.size(); i++) { 3575 mEffectChains[i]->setDevice_l(mOutDevice); 3576 } 3577 } 3578 3579 if (status == NO_ERROR) { 3580 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3581 keyValuePair.string()); 3582 if (!mStandby && status == INVALID_OPERATION) { 3583 mOutput->stream->common.standby(&mOutput->stream->common); 3584 mStandby = true; 3585 mBytesWritten = 0; 3586 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3587 keyValuePair.string()); 3588 } 3589 if (status == NO_ERROR && reconfig) { 3590 delete mAudioMixer; 3591 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3592 mAudioMixer = NULL; 3593 readOutputParameters(); 3594 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3595 for (size_t i = 0; i < mTracks.size() ; i++) { 3596 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3597 if (name < 0) { 3598 break; 3599 } 3600 mTracks[i]->mName = name; 3601 // limit track sample rate to 2 x new output sample rate 3602 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3603 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3604 } 3605 } 3606 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3607 } 3608 } 3609 3610 mNewParameters.removeAt(0); 3611 3612 mParamStatus = status; 3613 mParamCond.signal(); 3614 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3615 // already timed out waiting for the status and will never signal the condition. 3616 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3617 } 3618 3619 if (!(previousCommand & FastMixerState::IDLE)) { 3620 ALOG_ASSERT(mFastMixer != NULL); 3621 FastMixerStateQueue *sq = mFastMixer->sq(); 3622 FastMixerState *state = sq->begin(); 3623 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3624 state->mCommand = previousCommand; 3625 sq->end(); 3626 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3627 } 3628 3629 return reconfig; 3630} 3631 3632void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3633{ 3634 NBAIO_Source *teeSource = source.get(); 3635 if (teeSource != NULL) { 3636 char teeTime[16]; 3637 struct timeval tv; 3638 gettimeofday(&tv, NULL); 3639 struct tm tm; 3640 localtime_r(&tv.tv_sec, &tm); 3641 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3642 char teePath[64]; 3643 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3644 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3645 if (teeFd >= 0) { 3646 char wavHeader[44]; 3647 memcpy(wavHeader, 3648 "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", 3649 sizeof(wavHeader)); 3650 NBAIO_Format format = teeSource->format(); 3651 unsigned channelCount = Format_channelCount(format); 3652 ALOG_ASSERT(channelCount <= FCC_2); 3653 uint32_t sampleRate = Format_sampleRate(format); 3654 wavHeader[22] = channelCount; // number of channels 3655 wavHeader[24] = sampleRate; // sample rate 3656 wavHeader[25] = sampleRate >> 8; 3657 wavHeader[32] = channelCount * 2; // block alignment 3658 write(teeFd, wavHeader, sizeof(wavHeader)); 3659 size_t total = 0; 3660 bool firstRead = true; 3661 for (;;) { 3662#define TEE_SINK_READ 1024 3663 short buffer[TEE_SINK_READ * FCC_2]; 3664 size_t count = TEE_SINK_READ; 3665 ssize_t actual = teeSource->read(buffer, count, 3666 AudioBufferProvider::kInvalidPTS); 3667 bool wasFirstRead = firstRead; 3668 firstRead = false; 3669 if (actual <= 0) { 3670 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3671 continue; 3672 } 3673 break; 3674 } 3675 ALOG_ASSERT(actual <= (ssize_t)count); 3676 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3677 total += actual; 3678 } 3679 lseek(teeFd, (off_t) 4, SEEK_SET); 3680 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3681 write(teeFd, &temp, sizeof(temp)); 3682 lseek(teeFd, (off_t) 40, SEEK_SET); 3683 temp = total * channelCount * sizeof(short); 3684 write(teeFd, &temp, sizeof(temp)); 3685 close(teeFd); 3686 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3687 } else { 3688 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3689 } 3690 } 3691} 3692 3693void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3694{ 3695 const size_t SIZE = 256; 3696 char buffer[SIZE]; 3697 String8 result; 3698 3699 PlaybackThread::dumpInternals(fd, args); 3700 3701 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3702 result.append(buffer); 3703 write(fd, result.string(), result.size()); 3704 3705 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3706 FastMixerDumpState copy = mFastMixerDumpState; 3707 copy.dump(fd); 3708 3709#ifdef STATE_QUEUE_DUMP 3710 // Similar for state queue 3711 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3712 observerCopy.dump(fd); 3713 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3714 mutatorCopy.dump(fd); 3715#endif 3716 3717 // Write the tee output to a .wav file 3718 dumpTee(fd, mTeeSource, mId); 3719 3720#ifdef AUDIO_WATCHDOG 3721 if (mAudioWatchdog != 0) { 3722 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3723 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3724 wdCopy.dump(fd); 3725 } 3726#endif 3727} 3728 3729uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3730{ 3731 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3732} 3733 3734uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3735{ 3736 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3737} 3738 3739void AudioFlinger::MixerThread::cacheParameters_l() 3740{ 3741 PlaybackThread::cacheParameters_l(); 3742 3743 // FIXME: Relaxed timing because of a certain device that can't meet latency 3744 // Should be reduced to 2x after the vendor fixes the driver issue 3745 // increase threshold again due to low power audio mode. The way this warning 3746 // threshold is calculated and its usefulness should be reconsidered anyway. 3747 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3748} 3749 3750// ---------------------------------------------------------------------------- 3751AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3752 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3753 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3754 // mLeftVolFloat, mRightVolFloat 3755{ 3756} 3757 3758AudioFlinger::DirectOutputThread::~DirectOutputThread() 3759{ 3760} 3761 3762AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3763 Vector< sp<Track> > *tracksToRemove 3764) 3765{ 3766 sp<Track> trackToRemove; 3767 3768 mixer_state mixerStatus = MIXER_IDLE; 3769 3770 // find out which tracks need to be processed 3771 if (mActiveTracks.size() != 0) { 3772 sp<Track> t = mActiveTracks[0].promote(); 3773 // The track died recently 3774 if (t == 0) { 3775 return MIXER_IDLE; 3776 } 3777 3778 Track* const track = t.get(); 3779 audio_track_cblk_t* cblk = track->cblk(); 3780 3781 // The first time a track is added we wait 3782 // for all its buffers to be filled before processing it 3783 uint32_t minFrames; 3784 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3785 minFrames = mNormalFrameCount; 3786 } else { 3787 minFrames = 1; 3788 } 3789 if ((track->framesReady() >= minFrames) && track->isReady() && 3790 !track->isPaused() && !track->isTerminated()) 3791 { 3792 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3793 3794 if (track->mFillingUpStatus == Track::FS_FILLED) { 3795 track->mFillingUpStatus = Track::FS_ACTIVE; 3796 mLeftVolFloat = mRightVolFloat = 0; 3797 if (track->mState == TrackBase::RESUMING) { 3798 track->mState = TrackBase::ACTIVE; 3799 } 3800 } 3801 3802 // compute volume for this track 3803 float left, right; 3804 if (track->isMuted() || mMasterMute || track->isPausing() || 3805 mStreamTypes[track->streamType()].mute) { 3806 left = right = 0; 3807 if (track->isPausing()) { 3808 track->setPaused(); 3809 } 3810 } else { 3811 float typeVolume = mStreamTypes[track->streamType()].volume; 3812 float v = mMasterVolume * typeVolume; 3813 uint32_t vlr = cblk->getVolumeLR(); 3814 float v_clamped = v * (vlr & 0xFFFF); 3815 if (v_clamped > MAX_GAIN) { 3816 v_clamped = MAX_GAIN; 3817 } 3818 left = v_clamped/MAX_GAIN; 3819 v_clamped = v * (vlr >> 16); 3820 if (v_clamped > MAX_GAIN) { 3821 v_clamped = MAX_GAIN; 3822 } 3823 right = v_clamped/MAX_GAIN; 3824 } 3825 3826 if (left != mLeftVolFloat || right != mRightVolFloat) { 3827 mLeftVolFloat = left; 3828 mRightVolFloat = right; 3829 3830 // Convert volumes from float to 8.24 3831 uint32_t vl = (uint32_t)(left * (1 << 24)); 3832 uint32_t vr = (uint32_t)(right * (1 << 24)); 3833 3834 // Delegate volume control to effect in track effect chain if needed 3835 // only one effect chain can be present on DirectOutputThread, so if 3836 // there is one, the track is connected to it 3837 if (!mEffectChains.isEmpty()) { 3838 // Do not ramp volume if volume is controlled by effect 3839 mEffectChains[0]->setVolume_l(&vl, &vr); 3840 left = (float)vl / (1 << 24); 3841 right = (float)vr / (1 << 24); 3842 } 3843 mOutput->stream->set_volume(mOutput->stream, left, right); 3844 } 3845 3846 // reset retry count 3847 track->mRetryCount = kMaxTrackRetriesDirect; 3848 mActiveTrack = t; 3849 mixerStatus = MIXER_TRACKS_READY; 3850 } else { 3851 // clear effect chain input buffer if an active track underruns to avoid sending 3852 // previous audio buffer again to effects 3853 if (!mEffectChains.isEmpty()) { 3854 mEffectChains[0]->clearInputBuffer(); 3855 } 3856 3857 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3858 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3859 track->isStopped() || track->isPaused()) { 3860 // We have consumed all the buffers of this track. 3861 // Remove it from the list of active tracks. 3862 // TODO: implement behavior for compressed audio 3863 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3864 size_t framesWritten = 3865 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3866 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3867 if (track->isStopped()) { 3868 track->reset(); 3869 } 3870 trackToRemove = track; 3871 } 3872 } else { 3873 // No buffers for this track. Give it a few chances to 3874 // fill a buffer, then remove it from active list. 3875 if (--(track->mRetryCount) <= 0) { 3876 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3877 trackToRemove = track; 3878 } else { 3879 mixerStatus = MIXER_TRACKS_ENABLED; 3880 } 3881 } 3882 } 3883 } 3884 3885 // FIXME merge this with similar code for removing multiple tracks 3886 // remove all the tracks that need to be... 3887 if (CC_UNLIKELY(trackToRemove != 0)) { 3888 tracksToRemove->add(trackToRemove); 3889 mActiveTracks.remove(trackToRemove); 3890 if (!mEffectChains.isEmpty()) { 3891 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3892 trackToRemove->sessionId()); 3893 mEffectChains[0]->decActiveTrackCnt(); 3894 } 3895 if (trackToRemove->isTerminated()) { 3896 removeTrack_l(trackToRemove); 3897 } 3898 } 3899 3900 return mixerStatus; 3901} 3902 3903void AudioFlinger::DirectOutputThread::threadLoop_mix() 3904{ 3905 AudioBufferProvider::Buffer buffer; 3906 size_t frameCount = mFrameCount; 3907 int8_t *curBuf = (int8_t *)mMixBuffer; 3908 // output audio to hardware 3909 while (frameCount) { 3910 buffer.frameCount = frameCount; 3911 mActiveTrack->getNextBuffer(&buffer); 3912 if (CC_UNLIKELY(buffer.raw == NULL)) { 3913 memset(curBuf, 0, frameCount * mFrameSize); 3914 break; 3915 } 3916 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3917 frameCount -= buffer.frameCount; 3918 curBuf += buffer.frameCount * mFrameSize; 3919 mActiveTrack->releaseBuffer(&buffer); 3920 } 3921 sleepTime = 0; 3922 standbyTime = systemTime() + standbyDelay; 3923 mActiveTrack.clear(); 3924 3925} 3926 3927void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3928{ 3929 if (sleepTime == 0) { 3930 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3931 sleepTime = activeSleepTime; 3932 } else { 3933 sleepTime = idleSleepTime; 3934 } 3935 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3936 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3937 sleepTime = 0; 3938 } 3939} 3940 3941// getTrackName_l() must be called with ThreadBase::mLock held 3942int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3943 int sessionId) 3944{ 3945 return 0; 3946} 3947 3948// deleteTrackName_l() must be called with ThreadBase::mLock held 3949void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3950{ 3951} 3952 3953// checkForNewParameters_l() must be called with ThreadBase::mLock held 3954bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3955{ 3956 bool reconfig = false; 3957 3958 while (!mNewParameters.isEmpty()) { 3959 status_t status = NO_ERROR; 3960 String8 keyValuePair = mNewParameters[0]; 3961 AudioParameter param = AudioParameter(keyValuePair); 3962 int value; 3963 3964 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3965 // do not accept frame count changes if tracks are open as the track buffer 3966 // size depends on frame count and correct behavior would not be garantied 3967 // if frame count is changed after track creation 3968 if (!mTracks.isEmpty()) { 3969 status = INVALID_OPERATION; 3970 } else { 3971 reconfig = true; 3972 } 3973 } 3974 if (status == NO_ERROR) { 3975 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3976 keyValuePair.string()); 3977 if (!mStandby && status == INVALID_OPERATION) { 3978 mOutput->stream->common.standby(&mOutput->stream->common); 3979 mStandby = true; 3980 mBytesWritten = 0; 3981 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3982 keyValuePair.string()); 3983 } 3984 if (status == NO_ERROR && reconfig) { 3985 readOutputParameters(); 3986 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3987 } 3988 } 3989 3990 mNewParameters.removeAt(0); 3991 3992 mParamStatus = status; 3993 mParamCond.signal(); 3994 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3995 // already timed out waiting for the status and will never signal the condition. 3996 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3997 } 3998 return reconfig; 3999} 4000 4001uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4002{ 4003 uint32_t time; 4004 if (audio_is_linear_pcm(mFormat)) { 4005 time = PlaybackThread::activeSleepTimeUs(); 4006 } else { 4007 time = 10000; 4008 } 4009 return time; 4010} 4011 4012uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4013{ 4014 uint32_t time; 4015 if (audio_is_linear_pcm(mFormat)) { 4016 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4017 } else { 4018 time = 10000; 4019 } 4020 return time; 4021} 4022 4023uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4024{ 4025 uint32_t time; 4026 if (audio_is_linear_pcm(mFormat)) { 4027 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4028 } else { 4029 time = 10000; 4030 } 4031 return time; 4032} 4033 4034void AudioFlinger::DirectOutputThread::cacheParameters_l() 4035{ 4036 PlaybackThread::cacheParameters_l(); 4037 4038 // use shorter standby delay as on normal output to release 4039 // hardware resources as soon as possible 4040 standbyDelay = microseconds(activeSleepTime*2); 4041} 4042 4043// ---------------------------------------------------------------------------- 4044 4045AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4046 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4047 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4048 DUPLICATING), 4049 mWaitTimeMs(UINT_MAX) 4050{ 4051 addOutputTrack(mainThread); 4052} 4053 4054AudioFlinger::DuplicatingThread::~DuplicatingThread() 4055{ 4056 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4057 mOutputTracks[i]->destroy(); 4058 } 4059} 4060 4061void AudioFlinger::DuplicatingThread::threadLoop_mix() 4062{ 4063 // mix buffers... 4064 if (outputsReady(outputTracks)) { 4065 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4066 } else { 4067 memset(mMixBuffer, 0, mixBufferSize); 4068 } 4069 sleepTime = 0; 4070 writeFrames = mNormalFrameCount; 4071 standbyTime = systemTime() + standbyDelay; 4072} 4073 4074void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4075{ 4076 if (sleepTime == 0) { 4077 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4078 sleepTime = activeSleepTime; 4079 } else { 4080 sleepTime = idleSleepTime; 4081 } 4082 } else if (mBytesWritten != 0) { 4083 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4084 writeFrames = mNormalFrameCount; 4085 memset(mMixBuffer, 0, mixBufferSize); 4086 } else { 4087 // flush remaining overflow buffers in output tracks 4088 writeFrames = 0; 4089 } 4090 sleepTime = 0; 4091 } 4092} 4093 4094void AudioFlinger::DuplicatingThread::threadLoop_write() 4095{ 4096 for (size_t i = 0; i < outputTracks.size(); i++) { 4097 outputTracks[i]->write(mMixBuffer, writeFrames); 4098 } 4099 mBytesWritten += mixBufferSize; 4100} 4101 4102void AudioFlinger::DuplicatingThread::threadLoop_standby() 4103{ 4104 // DuplicatingThread implements standby by stopping all tracks 4105 for (size_t i = 0; i < outputTracks.size(); i++) { 4106 outputTracks[i]->stop(); 4107 } 4108} 4109 4110void AudioFlinger::DuplicatingThread::saveOutputTracks() 4111{ 4112 outputTracks = mOutputTracks; 4113} 4114 4115void AudioFlinger::DuplicatingThread::clearOutputTracks() 4116{ 4117 outputTracks.clear(); 4118} 4119 4120void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4121{ 4122 Mutex::Autolock _l(mLock); 4123 // FIXME explain this formula 4124 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4125 OutputTrack *outputTrack = new OutputTrack(thread, 4126 this, 4127 mSampleRate, 4128 mFormat, 4129 mChannelMask, 4130 frameCount); 4131 if (outputTrack->cblk() != NULL) { 4132 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4133 mOutputTracks.add(outputTrack); 4134 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4135 updateWaitTime_l(); 4136 } 4137} 4138 4139void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4140{ 4141 Mutex::Autolock _l(mLock); 4142 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4143 if (mOutputTracks[i]->thread() == thread) { 4144 mOutputTracks[i]->destroy(); 4145 mOutputTracks.removeAt(i); 4146 updateWaitTime_l(); 4147 return; 4148 } 4149 } 4150 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4151} 4152 4153// caller must hold mLock 4154void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4155{ 4156 mWaitTimeMs = UINT_MAX; 4157 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4158 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4159 if (strong != 0) { 4160 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4161 if (waitTimeMs < mWaitTimeMs) { 4162 mWaitTimeMs = waitTimeMs; 4163 } 4164 } 4165 } 4166} 4167 4168 4169bool AudioFlinger::DuplicatingThread::outputsReady( 4170 const SortedVector< sp<OutputTrack> > &outputTracks) 4171{ 4172 for (size_t i = 0; i < outputTracks.size(); i++) { 4173 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4174 if (thread == 0) { 4175 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4176 outputTracks[i].get()); 4177 return false; 4178 } 4179 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4180 // see note at standby() declaration 4181 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4182 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4183 thread.get()); 4184 return false; 4185 } 4186 } 4187 return true; 4188} 4189 4190uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4191{ 4192 return (mWaitTimeMs * 1000) / 2; 4193} 4194 4195void AudioFlinger::DuplicatingThread::cacheParameters_l() 4196{ 4197 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4198 updateWaitTime_l(); 4199 4200 MixerThread::cacheParameters_l(); 4201} 4202 4203// ---------------------------------------------------------------------------- 4204 4205// TrackBase constructor must be called with AudioFlinger::mLock held 4206AudioFlinger::ThreadBase::TrackBase::TrackBase( 4207 ThreadBase *thread, 4208 const sp<Client>& client, 4209 uint32_t sampleRate, 4210 audio_format_t format, 4211 audio_channel_mask_t channelMask, 4212 size_t frameCount, 4213 const sp<IMemory>& sharedBuffer, 4214 int sessionId) 4215 : RefBase(), 4216 mThread(thread), 4217 mClient(client), 4218 mCblk(NULL), 4219 // mBuffer 4220 // mBufferEnd 4221 mStepCount(0), 4222 mState(IDLE), 4223 mSampleRate(sampleRate), 4224 mFormat(format), 4225 mChannelMask(channelMask), 4226 mChannelCount(popcount(channelMask)), 4227 mFrameSize(audio_is_linear_pcm(format) ? 4228 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 4229 mStepServerFailed(false), 4230 mSessionId(sessionId) 4231{ 4232 // client == 0 implies sharedBuffer == 0 4233 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4234 4235 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4236 sharedBuffer->size()); 4237 4238 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4239 size_t size = sizeof(audio_track_cblk_t); 4240 size_t bufferSize = frameCount * mFrameSize; 4241 if (sharedBuffer == 0) { 4242 size += bufferSize; 4243 } 4244 4245 if (client != 0) { 4246 mCblkMemory = client->heap()->allocate(size); 4247 if (mCblkMemory != 0) { 4248 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4249 // can't assume mCblk != NULL 4250 } else { 4251 ALOGE("not enough memory for AudioTrack size=%u", size); 4252 client->heap()->dump("AudioTrack"); 4253 return; 4254 } 4255 } else { 4256 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4257 // assume mCblk != NULL 4258 } 4259 4260 // construct the shared structure in-place. 4261 if (mCblk != NULL) { 4262 new(mCblk) audio_track_cblk_t(); 4263 // clear all buffers 4264 mCblk->frameCount = frameCount; 4265 mCblk->sampleRate = sampleRate; 4266// uncomment the following lines to quickly test 32-bit wraparound 4267// mCblk->user = 0xffff0000; 4268// mCblk->server = 0xffff0000; 4269// mCblk->userBase = 0xffff0000; 4270// mCblk->serverBase = 0xffff0000; 4271 if (sharedBuffer == 0) { 4272 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4273 memset(mBuffer, 0, bufferSize); 4274 // Force underrun condition to avoid false underrun callback until first data is 4275 // written to buffer (other flags are cleared) 4276 mCblk->flags = CBLK_UNDERRUN; 4277 } else { 4278 mBuffer = sharedBuffer->pointer(); 4279 } 4280 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4281 } 4282} 4283 4284AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4285{ 4286 if (mCblk != NULL) { 4287 if (mClient == 0) { 4288 delete mCblk; 4289 } else { 4290 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4291 } 4292 } 4293 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4294 if (mClient != 0) { 4295 // Client destructor must run with AudioFlinger mutex locked 4296 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4297 // If the client's reference count drops to zero, the associated destructor 4298 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4299 // relying on the automatic clear() at end of scope. 4300 mClient.clear(); 4301 } 4302} 4303 4304// AudioBufferProvider interface 4305// getNextBuffer() = 0; 4306// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4307void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4308{ 4309 buffer->raw = NULL; 4310 mStepCount = buffer->frameCount; 4311 // FIXME See note at getNextBuffer() 4312 (void) step(); // ignore return value of step() 4313 buffer->frameCount = 0; 4314} 4315 4316bool AudioFlinger::ThreadBase::TrackBase::step() { 4317 bool result; 4318 audio_track_cblk_t* cblk = this->cblk(); 4319 4320 result = cblk->stepServer(mStepCount, isOut()); 4321 if (!result) { 4322 ALOGV("stepServer failed acquiring cblk mutex"); 4323 mStepServerFailed = true; 4324 } 4325 return result; 4326} 4327 4328void AudioFlinger::ThreadBase::TrackBase::reset() { 4329 audio_track_cblk_t* cblk = this->cblk(); 4330 4331 cblk->user = 0; 4332 cblk->server = 0; 4333 cblk->userBase = 0; 4334 cblk->serverBase = 0; 4335 mStepServerFailed = false; 4336 ALOGV("TrackBase::reset"); 4337} 4338 4339uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4340 return mCblk->sampleRate; 4341} 4342 4343void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4344 audio_track_cblk_t* cblk = this->cblk(); 4345 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; 4346 int8_t *bufferEnd = bufferStart + frames * mFrameSize; 4347 4348 // Check validity of returned pointer in case the track control block would have been corrupted. 4349 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4350 "TrackBase::getBuffer buffer out of range:\n" 4351 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4352 " server %u, serverBase %u, user %u, userBase %u, frameSize %u", 4353 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4354 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); 4355 4356 return bufferStart; 4357} 4358 4359status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4360{ 4361 mSyncEvents.add(event); 4362 return NO_ERROR; 4363} 4364 4365// ---------------------------------------------------------------------------- 4366 4367// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4368AudioFlinger::PlaybackThread::Track::Track( 4369 PlaybackThread *thread, 4370 const sp<Client>& client, 4371 audio_stream_type_t streamType, 4372 uint32_t sampleRate, 4373 audio_format_t format, 4374 audio_channel_mask_t channelMask, 4375 size_t frameCount, 4376 const sp<IMemory>& sharedBuffer, 4377 int sessionId, 4378 IAudioFlinger::track_flags_t flags) 4379 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4380 sessionId), 4381 mMute(false), 4382 mFillingUpStatus(FS_INVALID), 4383 // mRetryCount initialized later when needed 4384 mSharedBuffer(sharedBuffer), 4385 mStreamType(streamType), 4386 mName(-1), // see note below 4387 mMainBuffer(thread->mixBuffer()), 4388 mAuxBuffer(NULL), 4389 mAuxEffectId(0), mHasVolumeController(false), 4390 mPresentationCompleteFrames(0), 4391 mFlags(flags), 4392 mFastIndex(-1), 4393 mUnderrunCount(0), 4394 mCachedVolume(1.0) 4395{ 4396 if (mCblk != NULL) { 4397 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4398 mName = thread->getTrackName_l(channelMask, sessionId); 4399 mCblk->mName = mName; 4400 if (mName < 0) { 4401 ALOGE("no more track names available"); 4402 return; 4403 } 4404 // only allocate a fast track index if we were able to allocate a normal track name 4405 if (flags & IAudioFlinger::TRACK_FAST) { 4406 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4407 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4408 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4409 // FIXME This is too eager. We allocate a fast track index before the 4410 // fast track becomes active. Since fast tracks are a scarce resource, 4411 // this means we are potentially denying other more important fast tracks from 4412 // being created. It would be better to allocate the index dynamically. 4413 mFastIndex = i; 4414 mCblk->mName = i; 4415 // Read the initial underruns because this field is never cleared by the fast mixer 4416 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4417 thread->mFastTrackAvailMask &= ~(1 << i); 4418 } 4419 } 4420 ALOGV("Track constructor name %d, calling pid %d", mName, 4421 IPCThreadState::self()->getCallingPid()); 4422} 4423 4424AudioFlinger::PlaybackThread::Track::~Track() 4425{ 4426 ALOGV("PlaybackThread::Track destructor"); 4427} 4428 4429void AudioFlinger::PlaybackThread::Track::destroy() 4430{ 4431 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4432 // by removing it from mTracks vector, so there is a risk that this Tracks's 4433 // destructor is called. As the destructor needs to lock mLock, 4434 // we must acquire a strong reference on this Track before locking mLock 4435 // here so that the destructor is called only when exiting this function. 4436 // On the other hand, as long as Track::destroy() is only called by 4437 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4438 // this Track with its member mTrack. 4439 sp<Track> keep(this); 4440 { // scope for mLock 4441 sp<ThreadBase> thread = mThread.promote(); 4442 if (thread != 0) { 4443 if (!isOutputTrack()) { 4444 if (mState == ACTIVE || mState == RESUMING) { 4445 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4446 4447#ifdef ADD_BATTERY_DATA 4448 // to track the speaker usage 4449 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4450#endif 4451 } 4452 AudioSystem::releaseOutput(thread->id()); 4453 } 4454 Mutex::Autolock _l(thread->mLock); 4455 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4456 playbackThread->destroyTrack_l(this); 4457 } 4458 } 4459} 4460 4461/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4462{ 4463 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 4464 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4465} 4466 4467void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4468{ 4469 uint32_t vlr = mCblk->getVolumeLR(); 4470 if (isFastTrack()) { 4471 sprintf(buffer, " F %2d", mFastIndex); 4472 } else { 4473 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4474 } 4475 track_state state = mState; 4476 char stateChar; 4477 switch (state) { 4478 case IDLE: 4479 stateChar = 'I'; 4480 break; 4481 case TERMINATED: 4482 stateChar = 'T'; 4483 break; 4484 case STOPPING_1: 4485 stateChar = 's'; 4486 break; 4487 case STOPPING_2: 4488 stateChar = '5'; 4489 break; 4490 case STOPPED: 4491 stateChar = 'S'; 4492 break; 4493 case RESUMING: 4494 stateChar = 'R'; 4495 break; 4496 case ACTIVE: 4497 stateChar = 'A'; 4498 break; 4499 case PAUSING: 4500 stateChar = 'p'; 4501 break; 4502 case PAUSED: 4503 stateChar = 'P'; 4504 break; 4505 case FLUSHED: 4506 stateChar = 'F'; 4507 break; 4508 default: 4509 stateChar = '?'; 4510 break; 4511 } 4512 char nowInUnderrun; 4513 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4514 case UNDERRUN_FULL: 4515 nowInUnderrun = ' '; 4516 break; 4517 case UNDERRUN_PARTIAL: 4518 nowInUnderrun = '<'; 4519 break; 4520 case UNDERRUN_EMPTY: 4521 nowInUnderrun = '*'; 4522 break; 4523 default: 4524 nowInUnderrun = '?'; 4525 break; 4526 } 4527 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4528 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4529 (mClient == 0) ? getpid_cached : mClient->pid(), 4530 mStreamType, 4531 mFormat, 4532 mChannelMask, 4533 mSessionId, 4534 mStepCount, 4535 mCblk->frameCount, 4536 stateChar, 4537 mMute, 4538 mFillingUpStatus, 4539 mCblk->sampleRate, 4540 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4541 20.0 * log10((vlr >> 16) / 4096.0), 4542 mCblk->server, 4543 mCblk->user, 4544 (int)mMainBuffer, 4545 (int)mAuxBuffer, 4546 mCblk->flags, 4547 mUnderrunCount, 4548 nowInUnderrun); 4549} 4550 4551// AudioBufferProvider interface 4552status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4553 AudioBufferProvider::Buffer* buffer, int64_t pts) 4554{ 4555 audio_track_cblk_t* cblk = this->cblk(); 4556 uint32_t framesReady; 4557 uint32_t framesReq = buffer->frameCount; 4558 4559 // Check if last stepServer failed, try to step now 4560 if (mStepServerFailed) { 4561 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4562 // Since the fast mixer is higher priority than client callback thread, 4563 // it does not result in priority inversion for client. 4564 // But a non-blocking solution would be preferable to avoid 4565 // fast mixer being unable to tryLock(), and 4566 // to avoid the extra context switches if the client wakes up, 4567 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4568 if (!step()) goto getNextBuffer_exit; 4569 ALOGV("stepServer recovered"); 4570 mStepServerFailed = false; 4571 } 4572 4573 // FIXME Same as above 4574 framesReady = cblk->framesReadyOut(); 4575 4576 if (CC_LIKELY(framesReady)) { 4577 uint32_t s = cblk->server; 4578 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4579 4580 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4581 if (framesReq > framesReady) { 4582 framesReq = framesReady; 4583 } 4584 if (framesReq > bufferEnd - s) { 4585 framesReq = bufferEnd - s; 4586 } 4587 4588 buffer->raw = getBuffer(s, framesReq); 4589 buffer->frameCount = framesReq; 4590 return NO_ERROR; 4591 } 4592 4593getNextBuffer_exit: 4594 buffer->raw = NULL; 4595 buffer->frameCount = 0; 4596 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4597 return NOT_ENOUGH_DATA; 4598} 4599 4600// Note that framesReady() takes a mutex on the control block using tryLock(). 4601// This could result in priority inversion if framesReady() is called by the normal mixer, 4602// as the normal mixer thread runs at lower 4603// priority than the client's callback thread: there is a short window within framesReady() 4604// during which the normal mixer could be preempted, and the client callback would block. 4605// Another problem can occur if framesReady() is called by the fast mixer: 4606// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4607// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4608size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4609 return mCblk->framesReadyOut(); 4610} 4611 4612// Don't call for fast tracks; the framesReady() could result in priority inversion 4613bool AudioFlinger::PlaybackThread::Track::isReady() const { 4614 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 4615 return true; 4616 } 4617 4618 if (framesReady() >= mCblk->frameCount || 4619 (mCblk->flags & CBLK_FORCEREADY)) { 4620 mFillingUpStatus = FS_FILLED; 4621 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4622 return true; 4623 } 4624 return false; 4625} 4626 4627status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4628 int triggerSession) 4629{ 4630 status_t status = NO_ERROR; 4631 ALOGV("start(%d), calling pid %d session %d", 4632 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4633 4634 sp<ThreadBase> thread = mThread.promote(); 4635 if (thread != 0) { 4636 Mutex::Autolock _l(thread->mLock); 4637 track_state state = mState; 4638 // here the track could be either new, or restarted 4639 // in both cases "unstop" the track 4640 if (mState == PAUSED) { 4641 mState = TrackBase::RESUMING; 4642 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4643 } else { 4644 mState = TrackBase::ACTIVE; 4645 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4646 } 4647 4648 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4649 thread->mLock.unlock(); 4650 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4651 thread->mLock.lock(); 4652 4653#ifdef ADD_BATTERY_DATA 4654 // to track the speaker usage 4655 if (status == NO_ERROR) { 4656 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4657 } 4658#endif 4659 } 4660 if (status == NO_ERROR) { 4661 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4662 playbackThread->addTrack_l(this); 4663 } else { 4664 mState = state; 4665 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4666 } 4667 } else { 4668 status = BAD_VALUE; 4669 } 4670 return status; 4671} 4672 4673void AudioFlinger::PlaybackThread::Track::stop() 4674{ 4675 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4676 sp<ThreadBase> thread = mThread.promote(); 4677 if (thread != 0) { 4678 Mutex::Autolock _l(thread->mLock); 4679 track_state state = mState; 4680 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4681 // If the track is not active (PAUSED and buffers full), flush buffers 4682 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4683 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4684 reset(); 4685 mState = STOPPED; 4686 } else if (!isFastTrack()) { 4687 mState = STOPPED; 4688 } else { 4689 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4690 // and then to STOPPED and reset() when presentation is complete 4691 mState = STOPPING_1; 4692 } 4693 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4694 playbackThread); 4695 } 4696 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4697 thread->mLock.unlock(); 4698 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4699 thread->mLock.lock(); 4700 4701#ifdef ADD_BATTERY_DATA 4702 // to track the speaker usage 4703 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4704#endif 4705 } 4706 } 4707} 4708 4709void AudioFlinger::PlaybackThread::Track::pause() 4710{ 4711 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4712 sp<ThreadBase> thread = mThread.promote(); 4713 if (thread != 0) { 4714 Mutex::Autolock _l(thread->mLock); 4715 if (mState == ACTIVE || mState == RESUMING) { 4716 mState = PAUSING; 4717 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4718 if (!isOutputTrack()) { 4719 thread->mLock.unlock(); 4720 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4721 thread->mLock.lock(); 4722 4723#ifdef ADD_BATTERY_DATA 4724 // to track the speaker usage 4725 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4726#endif 4727 } 4728 } 4729 } 4730} 4731 4732void AudioFlinger::PlaybackThread::Track::flush() 4733{ 4734 ALOGV("flush(%d)", mName); 4735 sp<ThreadBase> thread = mThread.promote(); 4736 if (thread != 0) { 4737 Mutex::Autolock _l(thread->mLock); 4738 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4739 mState != PAUSING && mState != IDLE && mState != FLUSHED) { 4740 return; 4741 } 4742 // No point remaining in PAUSED state after a flush => go to 4743 // FLUSHED state 4744 mState = FLUSHED; 4745 // do not reset the track if it is still in the process of being stopped or paused. 4746 // this will be done by prepareTracks_l() when the track is stopped. 4747 // prepareTracks_l() will see mState == FLUSHED, then 4748 // remove from active track list, reset(), and trigger presentation complete 4749 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4750 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4751 reset(); 4752 } 4753 } 4754} 4755 4756void AudioFlinger::PlaybackThread::Track::reset() 4757{ 4758 // Do not reset twice to avoid discarding data written just after a flush and before 4759 // the audioflinger thread detects the track is stopped. 4760 if (!mResetDone) { 4761 TrackBase::reset(); 4762 // Force underrun condition to avoid false underrun callback until first data is 4763 // written to buffer 4764 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4765 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4766 mFillingUpStatus = FS_FILLING; 4767 mResetDone = true; 4768 if (mState == FLUSHED) { 4769 mState = IDLE; 4770 } 4771 } 4772} 4773 4774void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4775{ 4776 mMute = muted; 4777} 4778 4779status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4780{ 4781 status_t status = DEAD_OBJECT; 4782 sp<ThreadBase> thread = mThread.promote(); 4783 if (thread != 0) { 4784 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4785 sp<AudioFlinger> af = mClient->audioFlinger(); 4786 4787 Mutex::Autolock _l(af->mLock); 4788 4789 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4790 4791 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4792 Mutex::Autolock _dl(playbackThread->mLock); 4793 Mutex::Autolock _sl(srcThread->mLock); 4794 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4795 if (chain == 0) { 4796 return INVALID_OPERATION; 4797 } 4798 4799 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4800 if (effect == 0) { 4801 return INVALID_OPERATION; 4802 } 4803 srcThread->removeEffect_l(effect); 4804 playbackThread->addEffect_l(effect); 4805 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4806 if (effect->state() == EffectModule::ACTIVE || 4807 effect->state() == EffectModule::STOPPING) { 4808 effect->start(); 4809 } 4810 4811 sp<EffectChain> dstChain = effect->chain().promote(); 4812 if (dstChain == 0) { 4813 srcThread->addEffect_l(effect); 4814 return INVALID_OPERATION; 4815 } 4816 AudioSystem::unregisterEffect(effect->id()); 4817 AudioSystem::registerEffect(&effect->desc(), 4818 srcThread->id(), 4819 dstChain->strategy(), 4820 AUDIO_SESSION_OUTPUT_MIX, 4821 effect->id()); 4822 } 4823 status = playbackThread->attachAuxEffect(this, EffectId); 4824 } 4825 return status; 4826} 4827 4828void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4829{ 4830 mAuxEffectId = EffectId; 4831 mAuxBuffer = buffer; 4832} 4833 4834bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4835 size_t audioHalFrames) 4836{ 4837 // a track is considered presented when the total number of frames written to audio HAL 4838 // corresponds to the number of frames written when presentationComplete() is called for the 4839 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4840 if (mPresentationCompleteFrames == 0) { 4841 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4842 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4843 mPresentationCompleteFrames, audioHalFrames); 4844 } 4845 if (framesWritten >= mPresentationCompleteFrames) { 4846 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4847 mSessionId, framesWritten); 4848 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4849 return true; 4850 } 4851 return false; 4852} 4853 4854void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4855{ 4856 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4857 if (mSyncEvents[i]->type() == type) { 4858 mSyncEvents[i]->trigger(); 4859 mSyncEvents.removeAt(i); 4860 i--; 4861 } 4862 } 4863} 4864 4865// implement VolumeBufferProvider interface 4866 4867uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4868{ 4869 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4870 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4871 uint32_t vlr = mCblk->getVolumeLR(); 4872 uint32_t vl = vlr & 0xFFFF; 4873 uint32_t vr = vlr >> 16; 4874 // track volumes come from shared memory, so can't be trusted and must be clamped 4875 if (vl > MAX_GAIN_INT) { 4876 vl = MAX_GAIN_INT; 4877 } 4878 if (vr > MAX_GAIN_INT) { 4879 vr = MAX_GAIN_INT; 4880 } 4881 // now apply the cached master volume and stream type volume; 4882 // this is trusted but lacks any synchronization or barrier so may be stale 4883 float v = mCachedVolume; 4884 vl *= v; 4885 vr *= v; 4886 // re-combine into U4.16 4887 vlr = (vr << 16) | (vl & 0xFFFF); 4888 // FIXME look at mute, pause, and stop flags 4889 return vlr; 4890} 4891 4892status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4893{ 4894 if (mState == TERMINATED || mState == PAUSED || 4895 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4896 (mState == STOPPED)))) { 4897 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4898 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4899 event->cancel(); 4900 return INVALID_OPERATION; 4901 } 4902 (void) TrackBase::setSyncEvent(event); 4903 return NO_ERROR; 4904} 4905 4906bool AudioFlinger::PlaybackThread::Track::isOut() const 4907{ 4908 return true; 4909} 4910 4911// timed audio tracks 4912 4913sp<AudioFlinger::PlaybackThread::TimedTrack> 4914AudioFlinger::PlaybackThread::TimedTrack::create( 4915 PlaybackThread *thread, 4916 const sp<Client>& client, 4917 audio_stream_type_t streamType, 4918 uint32_t sampleRate, 4919 audio_format_t format, 4920 audio_channel_mask_t channelMask, 4921 size_t frameCount, 4922 const sp<IMemory>& sharedBuffer, 4923 int sessionId) { 4924 if (!client->reserveTimedTrack()) 4925 return 0; 4926 4927 return new TimedTrack( 4928 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4929 sharedBuffer, sessionId); 4930} 4931 4932AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4933 PlaybackThread *thread, 4934 const sp<Client>& client, 4935 audio_stream_type_t streamType, 4936 uint32_t sampleRate, 4937 audio_format_t format, 4938 audio_channel_mask_t channelMask, 4939 size_t frameCount, 4940 const sp<IMemory>& sharedBuffer, 4941 int sessionId) 4942 : Track(thread, client, streamType, sampleRate, format, channelMask, 4943 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4944 mQueueHeadInFlight(false), 4945 mTrimQueueHeadOnRelease(false), 4946 mFramesPendingInQueue(0), 4947 mTimedSilenceBuffer(NULL), 4948 mTimedSilenceBufferSize(0), 4949 mTimedAudioOutputOnTime(false), 4950 mMediaTimeTransformValid(false) 4951{ 4952 LocalClock lc; 4953 mLocalTimeFreq = lc.getLocalFreq(); 4954 4955 mLocalTimeToSampleTransform.a_zero = 0; 4956 mLocalTimeToSampleTransform.b_zero = 0; 4957 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4958 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4959 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4960 &mLocalTimeToSampleTransform.a_to_b_denom); 4961 4962 mMediaTimeToSampleTransform.a_zero = 0; 4963 mMediaTimeToSampleTransform.b_zero = 0; 4964 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4965 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4966 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4967 &mMediaTimeToSampleTransform.a_to_b_denom); 4968} 4969 4970AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4971 mClient->releaseTimedTrack(); 4972 delete [] mTimedSilenceBuffer; 4973} 4974 4975status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4976 size_t size, sp<IMemory>* buffer) { 4977 4978 Mutex::Autolock _l(mTimedBufferQueueLock); 4979 4980 trimTimedBufferQueue_l(); 4981 4982 // lazily initialize the shared memory heap for timed buffers 4983 if (mTimedMemoryDealer == NULL) { 4984 const int kTimedBufferHeapSize = 512 << 10; 4985 4986 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4987 "AudioFlingerTimed"); 4988 if (mTimedMemoryDealer == NULL) 4989 return NO_MEMORY; 4990 } 4991 4992 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4993 if (newBuffer == NULL) { 4994 newBuffer = mTimedMemoryDealer->allocate(size); 4995 if (newBuffer == NULL) 4996 return NO_MEMORY; 4997 } 4998 4999 *buffer = newBuffer; 5000 return NO_ERROR; 5001} 5002 5003// caller must hold mTimedBufferQueueLock 5004void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 5005 int64_t mediaTimeNow; 5006 { 5007 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5008 if (!mMediaTimeTransformValid) 5009 return; 5010 5011 int64_t targetTimeNow; 5012 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 5013 ? mCCHelper.getCommonTime(&targetTimeNow) 5014 : mCCHelper.getLocalTime(&targetTimeNow); 5015 5016 if (OK != res) 5017 return; 5018 5019 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 5020 &mediaTimeNow)) { 5021 return; 5022 } 5023 } 5024 5025 size_t trimEnd; 5026 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 5027 int64_t bufEnd; 5028 5029 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 5030 // We have a next buffer. Just use its PTS as the PTS of the frame 5031 // following the last frame in this buffer. If the stream is sparse 5032 // (ie, there are deliberate gaps left in the stream which should be 5033 // filled with silence by the TimedAudioTrack), then this can result 5034 // in one extra buffer being left un-trimmed when it could have 5035 // been. In general, this is not typical, and we would rather 5036 // optimized away the TS calculation below for the more common case 5037 // where PTSes are contiguous. 5038 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5039 } else { 5040 // We have no next buffer. Compute the PTS of the frame following 5041 // the last frame in this buffer by computing the duration of of 5042 // this frame in media time units and adding it to the PTS of the 5043 // buffer. 5044 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5045 / mFrameSize; 5046 5047 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5048 &bufEnd)) { 5049 ALOGE("Failed to convert frame count of %lld to media time" 5050 " duration" " (scale factor %d/%u) in %s", 5051 frameCount, 5052 mMediaTimeToSampleTransform.a_to_b_numer, 5053 mMediaTimeToSampleTransform.a_to_b_denom, 5054 __PRETTY_FUNCTION__); 5055 break; 5056 } 5057 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5058 } 5059 5060 if (bufEnd > mediaTimeNow) 5061 break; 5062 5063 // Is the buffer we want to use in the middle of a mix operation right 5064 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5065 // from the mixer which should be coming back shortly. 5066 if (!trimEnd && mQueueHeadInFlight) { 5067 mTrimQueueHeadOnRelease = true; 5068 } 5069 } 5070 5071 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5072 if (trimStart < trimEnd) { 5073 // Update the bookkeeping for framesReady() 5074 for (size_t i = trimStart; i < trimEnd; ++i) { 5075 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5076 } 5077 5078 // Now actually remove the buffers from the queue. 5079 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5080 } 5081} 5082 5083void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5084 const char* logTag) { 5085 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5086 "%s called (reason \"%s\"), but timed buffer queue has no" 5087 " elements to trim.", __FUNCTION__, logTag); 5088 5089 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5090 mTimedBufferQueue.removeAt(0); 5091} 5092 5093void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5094 const TimedBuffer& buf, 5095 const char* logTag) { 5096 uint32_t bufBytes = buf.buffer()->size(); 5097 uint32_t consumedAlready = buf.position(); 5098 5099 ALOG_ASSERT(consumedAlready <= bufBytes, 5100 "Bad bookkeeping while updating frames pending. Timed buffer is" 5101 " only %u bytes long, but claims to have consumed %u" 5102 " bytes. (update reason: \"%s\")", 5103 bufBytes, consumedAlready, logTag); 5104 5105 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 5106 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5107 "Bad bookkeeping while updating frames pending. Should have at" 5108 " least %u queued frames, but we think we have only %u. (update" 5109 " reason: \"%s\")", 5110 bufFrames, mFramesPendingInQueue, logTag); 5111 5112 mFramesPendingInQueue -= bufFrames; 5113} 5114 5115status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5116 const sp<IMemory>& buffer, int64_t pts) { 5117 5118 { 5119 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5120 if (!mMediaTimeTransformValid) 5121 return INVALID_OPERATION; 5122 } 5123 5124 Mutex::Autolock _l(mTimedBufferQueueLock); 5125 5126 uint32_t bufFrames = buffer->size() / mFrameSize; 5127 mFramesPendingInQueue += bufFrames; 5128 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5129 5130 return NO_ERROR; 5131} 5132 5133status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5134 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5135 5136 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5137 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5138 target); 5139 5140 if (!(target == TimedAudioTrack::LOCAL_TIME || 5141 target == TimedAudioTrack::COMMON_TIME)) { 5142 return BAD_VALUE; 5143 } 5144 5145 Mutex::Autolock lock(mMediaTimeTransformLock); 5146 mMediaTimeTransform = xform; 5147 mMediaTimeTransformTarget = target; 5148 mMediaTimeTransformValid = true; 5149 5150 return NO_ERROR; 5151} 5152 5153#define min(a, b) ((a) < (b) ? (a) : (b)) 5154 5155// implementation of getNextBuffer for tracks whose buffers have timestamps 5156status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5157 AudioBufferProvider::Buffer* buffer, int64_t pts) 5158{ 5159 if (pts == AudioBufferProvider::kInvalidPTS) { 5160 buffer->raw = NULL; 5161 buffer->frameCount = 0; 5162 mTimedAudioOutputOnTime = false; 5163 return INVALID_OPERATION; 5164 } 5165 5166 Mutex::Autolock _l(mTimedBufferQueueLock); 5167 5168 ALOG_ASSERT(!mQueueHeadInFlight, 5169 "getNextBuffer called without releaseBuffer!"); 5170 5171 while (true) { 5172 5173 // if we have no timed buffers, then fail 5174 if (mTimedBufferQueue.isEmpty()) { 5175 buffer->raw = NULL; 5176 buffer->frameCount = 0; 5177 return NOT_ENOUGH_DATA; 5178 } 5179 5180 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5181 5182 // calculate the PTS of the head of the timed buffer queue expressed in 5183 // local time 5184 int64_t headLocalPTS; 5185 { 5186 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5187 5188 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5189 5190 if (mMediaTimeTransform.a_to_b_denom == 0) { 5191 // the transform represents a pause, so yield silence 5192 timedYieldSilence_l(buffer->frameCount, buffer); 5193 return NO_ERROR; 5194 } 5195 5196 int64_t transformedPTS; 5197 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5198 &transformedPTS)) { 5199 // the transform failed. this shouldn't happen, but if it does 5200 // then just drop this buffer 5201 ALOGW("timedGetNextBuffer transform failed"); 5202 buffer->raw = NULL; 5203 buffer->frameCount = 0; 5204 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5205 return NO_ERROR; 5206 } 5207 5208 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5209 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5210 &headLocalPTS)) { 5211 buffer->raw = NULL; 5212 buffer->frameCount = 0; 5213 return INVALID_OPERATION; 5214 } 5215 } else { 5216 headLocalPTS = transformedPTS; 5217 } 5218 } 5219 5220 // adjust the head buffer's PTS to reflect the portion of the head buffer 5221 // that has already been consumed 5222 int64_t effectivePTS = headLocalPTS + 5223 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 5224 5225 // Calculate the delta in samples between the head of the input buffer 5226 // queue and the start of the next output buffer that will be written. 5227 // If the transformation fails because of over or underflow, it means 5228 // that the sample's position in the output stream is so far out of 5229 // whack that it should just be dropped. 5230 int64_t sampleDelta; 5231 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5232 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5233 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5234 " mix"); 5235 continue; 5236 } 5237 if (!mLocalTimeToSampleTransform.doForwardTransform( 5238 (effectivePTS - pts) << 32, &sampleDelta)) { 5239 ALOGV("*** too late during sample rate transform: dropped buffer"); 5240 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5241 continue; 5242 } 5243 5244 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5245 " sampleDelta=[%d.%08x]", 5246 head.pts(), head.position(), pts, 5247 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5248 + (sampleDelta >> 32)), 5249 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5250 5251 // if the delta between the ideal placement for the next input sample and 5252 // the current output position is within this threshold, then we will 5253 // concatenate the next input samples to the previous output 5254 const int64_t kSampleContinuityThreshold = 5255 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5256 5257 // if this is the first buffer of audio that we're emitting from this track 5258 // then it should be almost exactly on time. 5259 const int64_t kSampleStartupThreshold = 1LL << 32; 5260 5261 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5262 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5263 // the next input is close enough to being on time, so concatenate it 5264 // with the last output 5265 timedYieldSamples_l(buffer); 5266 5267 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5268 head.position(), buffer->frameCount); 5269 return NO_ERROR; 5270 } 5271 5272 // Looks like our output is not on time. Reset our on timed status. 5273 // Next time we mix samples from our input queue, then should be within 5274 // the StartupThreshold. 5275 mTimedAudioOutputOnTime = false; 5276 if (sampleDelta > 0) { 5277 // the gap between the current output position and the proper start of 5278 // the next input sample is too big, so fill it with silence 5279 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5280 5281 timedYieldSilence_l(framesUntilNextInput, buffer); 5282 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5283 return NO_ERROR; 5284 } else { 5285 // the next input sample is late 5286 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5287 size_t onTimeSamplePosition = 5288 head.position() + lateFrames * mFrameSize; 5289 5290 if (onTimeSamplePosition > head.buffer()->size()) { 5291 // all the remaining samples in the head are too late, so 5292 // drop it and move on 5293 ALOGV("*** too late: dropped buffer"); 5294 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5295 continue; 5296 } else { 5297 // skip over the late samples 5298 head.setPosition(onTimeSamplePosition); 5299 5300 // yield the available samples 5301 timedYieldSamples_l(buffer); 5302 5303 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5304 return NO_ERROR; 5305 } 5306 } 5307 } 5308} 5309 5310// Yield samples from the timed buffer queue head up to the given output 5311// buffer's capacity. 5312// 5313// Caller must hold mTimedBufferQueueLock 5314void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5315 AudioBufferProvider::Buffer* buffer) { 5316 5317 const TimedBuffer& head = mTimedBufferQueue[0]; 5318 5319 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5320 head.position()); 5321 5322 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5323 mFrameSize); 5324 size_t framesRequested = buffer->frameCount; 5325 buffer->frameCount = min(framesLeftInHead, framesRequested); 5326 5327 mQueueHeadInFlight = true; 5328 mTimedAudioOutputOnTime = true; 5329} 5330 5331// Yield samples of silence up to the given output buffer's capacity 5332// 5333// Caller must hold mTimedBufferQueueLock 5334void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5335 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5336 5337 // lazily allocate a buffer filled with silence 5338 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 5339 delete [] mTimedSilenceBuffer; 5340 mTimedSilenceBufferSize = numFrames * mFrameSize; 5341 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5342 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5343 } 5344 5345 buffer->raw = mTimedSilenceBuffer; 5346 size_t framesRequested = buffer->frameCount; 5347 buffer->frameCount = min(numFrames, framesRequested); 5348 5349 mTimedAudioOutputOnTime = false; 5350} 5351 5352// AudioBufferProvider interface 5353void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5354 AudioBufferProvider::Buffer* buffer) { 5355 5356 Mutex::Autolock _l(mTimedBufferQueueLock); 5357 5358 // If the buffer which was just released is part of the buffer at the head 5359 // of the queue, be sure to update the amt of the buffer which has been 5360 // consumed. If the buffer being returned is not part of the head of the 5361 // queue, its either because the buffer is part of the silence buffer, or 5362 // because the head of the timed queue was trimmed after the mixer called 5363 // getNextBuffer but before the mixer called releaseBuffer. 5364 if (buffer->raw == mTimedSilenceBuffer) { 5365 ALOG_ASSERT(!mQueueHeadInFlight, 5366 "Queue head in flight during release of silence buffer!"); 5367 goto done; 5368 } 5369 5370 ALOG_ASSERT(mQueueHeadInFlight, 5371 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5372 " head in flight."); 5373 5374 if (mTimedBufferQueue.size()) { 5375 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5376 5377 void* start = head.buffer()->pointer(); 5378 void* end = reinterpret_cast<void*>( 5379 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5380 + head.buffer()->size()); 5381 5382 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5383 "released buffer not within the head of the timed buffer" 5384 " queue; qHead = [%p, %p], released buffer = %p", 5385 start, end, buffer->raw); 5386 5387 head.setPosition(head.position() + 5388 (buffer->frameCount * mFrameSize)); 5389 mQueueHeadInFlight = false; 5390 5391 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5392 "Bad bookkeeping during releaseBuffer! Should have at" 5393 " least %u queued frames, but we think we have only %u", 5394 buffer->frameCount, mFramesPendingInQueue); 5395 5396 mFramesPendingInQueue -= buffer->frameCount; 5397 5398 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5399 || mTrimQueueHeadOnRelease) { 5400 trimTimedBufferQueueHead_l("releaseBuffer"); 5401 mTrimQueueHeadOnRelease = false; 5402 } 5403 } else { 5404 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5405 " buffers in the timed buffer queue"); 5406 } 5407 5408done: 5409 buffer->raw = 0; 5410 buffer->frameCount = 0; 5411} 5412 5413size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5414 Mutex::Autolock _l(mTimedBufferQueueLock); 5415 return mFramesPendingInQueue; 5416} 5417 5418AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5419 : mPTS(0), mPosition(0) {} 5420 5421AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5422 const sp<IMemory>& buffer, int64_t pts) 5423 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5424 5425// ---------------------------------------------------------------------------- 5426 5427// RecordTrack constructor must be called with AudioFlinger::mLock held 5428AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5429 RecordThread *thread, 5430 const sp<Client>& client, 5431 uint32_t sampleRate, 5432 audio_format_t format, 5433 audio_channel_mask_t channelMask, 5434 size_t frameCount, 5435 int sessionId) 5436 : TrackBase(thread, client, sampleRate, format, 5437 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5438 mOverflow(false) 5439{ 5440 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5441} 5442 5443AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5444{ 5445 ALOGV("%s", __func__); 5446} 5447 5448// AudioBufferProvider interface 5449status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5450 int64_t pts) 5451{ 5452 audio_track_cblk_t* cblk = this->cblk(); 5453 uint32_t framesAvail; 5454 uint32_t framesReq = buffer->frameCount; 5455 5456 // Check if last stepServer failed, try to step now 5457 if (mStepServerFailed) { 5458 if (!step()) { 5459 goto getNextBuffer_exit; 5460 } 5461 ALOGV("stepServer recovered"); 5462 mStepServerFailed = false; 5463 } 5464 5465 // FIXME lock is not actually held, so overrun is possible 5466 framesAvail = cblk->framesAvailableIn_l(); 5467 5468 if (CC_LIKELY(framesAvail)) { 5469 uint32_t s = cblk->server; 5470 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5471 5472 if (framesReq > framesAvail) { 5473 framesReq = framesAvail; 5474 } 5475 if (framesReq > bufferEnd - s) { 5476 framesReq = bufferEnd - s; 5477 } 5478 5479 buffer->raw = getBuffer(s, framesReq); 5480 buffer->frameCount = framesReq; 5481 return NO_ERROR; 5482 } 5483 5484getNextBuffer_exit: 5485 buffer->raw = NULL; 5486 buffer->frameCount = 0; 5487 return NOT_ENOUGH_DATA; 5488} 5489 5490status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5491 int triggerSession) 5492{ 5493 sp<ThreadBase> thread = mThread.promote(); 5494 if (thread != 0) { 5495 RecordThread *recordThread = (RecordThread *)thread.get(); 5496 return recordThread->start(this, event, triggerSession); 5497 } else { 5498 return BAD_VALUE; 5499 } 5500} 5501 5502void AudioFlinger::RecordThread::RecordTrack::stop() 5503{ 5504 sp<ThreadBase> thread = mThread.promote(); 5505 if (thread != 0) { 5506 RecordThread *recordThread = (RecordThread *)thread.get(); 5507 recordThread->mLock.lock(); 5508 bool doStop = recordThread->stop_l(this); 5509 if (doStop) { 5510 TrackBase::reset(); 5511 // Force overrun condition to avoid false overrun callback until first data is 5512 // read from buffer 5513 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5514 } 5515 recordThread->mLock.unlock(); 5516 if (doStop) { 5517 AudioSystem::stopInput(recordThread->id()); 5518 } 5519 } 5520} 5521 5522/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5523{ 5524 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 5525} 5526 5527void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5528{ 5529 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5530 (mClient == 0) ? getpid_cached : mClient->pid(), 5531 mFormat, 5532 mChannelMask, 5533 mSessionId, 5534 mStepCount, 5535 mState, 5536 mCblk->sampleRate, 5537 mCblk->server, 5538 mCblk->user, 5539 mCblk->frameCount); 5540} 5541 5542bool AudioFlinger::RecordThread::RecordTrack::isOut() const 5543{ 5544 return false; 5545} 5546 5547// ---------------------------------------------------------------------------- 5548 5549AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5550 PlaybackThread *playbackThread, 5551 DuplicatingThread *sourceThread, 5552 uint32_t sampleRate, 5553 audio_format_t format, 5554 audio_channel_mask_t channelMask, 5555 size_t frameCount) 5556 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5557 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5558 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 5559{ 5560 5561 if (mCblk != NULL) { 5562 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5563 mOutBuffer.frameCount = 0; 5564 playbackThread->mTracks.add(this); 5565 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5566 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 5567 mCblk, mBuffer, mCblk->buffers, 5568 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5569 } else { 5570 ALOGW("Error creating output track on thread %p", playbackThread); 5571 } 5572} 5573 5574AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5575{ 5576 clearBufferQueue(); 5577} 5578 5579status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5580 int triggerSession) 5581{ 5582 status_t status = Track::start(event, triggerSession); 5583 if (status != NO_ERROR) { 5584 return status; 5585 } 5586 5587 mActive = true; 5588 mRetryCount = 127; 5589 return status; 5590} 5591 5592void AudioFlinger::PlaybackThread::OutputTrack::stop() 5593{ 5594 Track::stop(); 5595 clearBufferQueue(); 5596 mOutBuffer.frameCount = 0; 5597 mActive = false; 5598} 5599 5600bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5601{ 5602 Buffer *pInBuffer; 5603 Buffer inBuffer; 5604 uint32_t channelCount = mChannelCount; 5605 bool outputBufferFull = false; 5606 inBuffer.frameCount = frames; 5607 inBuffer.i16 = data; 5608 5609 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5610 5611 if (!mActive && frames != 0) { 5612 start(); 5613 sp<ThreadBase> thread = mThread.promote(); 5614 if (thread != 0) { 5615 MixerThread *mixerThread = (MixerThread *)thread.get(); 5616 if (mCblk->frameCount > frames){ 5617 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5618 uint32_t startFrames = (mCblk->frameCount - frames); 5619 pInBuffer = new Buffer; 5620 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5621 pInBuffer->frameCount = startFrames; 5622 pInBuffer->i16 = pInBuffer->mBuffer; 5623 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5624 mBufferQueue.add(pInBuffer); 5625 } else { 5626 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5627 } 5628 } 5629 } 5630 } 5631 5632 while (waitTimeLeftMs) { 5633 // First write pending buffers, then new data 5634 if (mBufferQueue.size()) { 5635 pInBuffer = mBufferQueue.itemAt(0); 5636 } else { 5637 pInBuffer = &inBuffer; 5638 } 5639 5640 if (pInBuffer->frameCount == 0) { 5641 break; 5642 } 5643 5644 if (mOutBuffer.frameCount == 0) { 5645 mOutBuffer.frameCount = pInBuffer->frameCount; 5646 nsecs_t startTime = systemTime(); 5647 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5648 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5649 mThread.unsafe_get()); 5650 outputBufferFull = true; 5651 break; 5652 } 5653 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5654 if (waitTimeLeftMs >= waitTimeMs) { 5655 waitTimeLeftMs -= waitTimeMs; 5656 } else { 5657 waitTimeLeftMs = 0; 5658 } 5659 } 5660 5661 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5662 pInBuffer->frameCount; 5663 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5664 mCblk->stepUserOut(outFrames); 5665 pInBuffer->frameCount -= outFrames; 5666 pInBuffer->i16 += outFrames * channelCount; 5667 mOutBuffer.frameCount -= outFrames; 5668 mOutBuffer.i16 += outFrames * channelCount; 5669 5670 if (pInBuffer->frameCount == 0) { 5671 if (mBufferQueue.size()) { 5672 mBufferQueue.removeAt(0); 5673 delete [] pInBuffer->mBuffer; 5674 delete pInBuffer; 5675 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5676 mThread.unsafe_get(), mBufferQueue.size()); 5677 } else { 5678 break; 5679 } 5680 } 5681 } 5682 5683 // If we could not write all frames, allocate a buffer and queue it for next time. 5684 if (inBuffer.frameCount) { 5685 sp<ThreadBase> thread = mThread.promote(); 5686 if (thread != 0 && !thread->standby()) { 5687 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5688 pInBuffer = new Buffer; 5689 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5690 pInBuffer->frameCount = inBuffer.frameCount; 5691 pInBuffer->i16 = pInBuffer->mBuffer; 5692 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5693 sizeof(int16_t)); 5694 mBufferQueue.add(pInBuffer); 5695 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5696 mThread.unsafe_get(), mBufferQueue.size()); 5697 } else { 5698 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5699 mThread.unsafe_get(), this); 5700 } 5701 } 5702 } 5703 5704 // Calling write() with a 0 length buffer, means that no more data will be written: 5705 // If no more buffers are pending, fill output track buffer to make sure it is started 5706 // by output mixer. 5707 if (frames == 0 && mBufferQueue.size() == 0) { 5708 if (mCblk->user < mCblk->frameCount) { 5709 frames = mCblk->frameCount - mCblk->user; 5710 pInBuffer = new Buffer; 5711 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5712 pInBuffer->frameCount = frames; 5713 pInBuffer->i16 = pInBuffer->mBuffer; 5714 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5715 mBufferQueue.add(pInBuffer); 5716 } else if (mActive) { 5717 stop(); 5718 } 5719 } 5720 5721 return outputBufferFull; 5722} 5723 5724status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5725 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5726{ 5727 int active; 5728 status_t result; 5729 audio_track_cblk_t* cblk = mCblk; 5730 uint32_t framesReq = buffer->frameCount; 5731 5732 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5733 buffer->frameCount = 0; 5734 5735 uint32_t framesAvail = cblk->framesAvailableOut(); 5736 5737 5738 if (framesAvail == 0) { 5739 Mutex::Autolock _l(cblk->lock); 5740 goto start_loop_here; 5741 while (framesAvail == 0) { 5742 active = mActive; 5743 if (CC_UNLIKELY(!active)) { 5744 ALOGV("Not active and NO_MORE_BUFFERS"); 5745 return NO_MORE_BUFFERS; 5746 } 5747 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5748 if (result != NO_ERROR) { 5749 return NO_MORE_BUFFERS; 5750 } 5751 // read the server count again 5752 start_loop_here: 5753 framesAvail = cblk->framesAvailableOut_l(); 5754 } 5755 } 5756 5757// if (framesAvail < framesReq) { 5758// return NO_MORE_BUFFERS; 5759// } 5760 5761 if (framesReq > framesAvail) { 5762 framesReq = framesAvail; 5763 } 5764 5765 uint32_t u = cblk->user; 5766 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5767 5768 if (framesReq > bufferEnd - u) { 5769 framesReq = bufferEnd - u; 5770 } 5771 5772 buffer->frameCount = framesReq; 5773 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 5774 return NO_ERROR; 5775} 5776 5777 5778void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5779{ 5780 size_t size = mBufferQueue.size(); 5781 5782 for (size_t i = 0; i < size; i++) { 5783 Buffer *pBuffer = mBufferQueue.itemAt(i); 5784 delete [] pBuffer->mBuffer; 5785 delete pBuffer; 5786 } 5787 mBufferQueue.clear(); 5788} 5789 5790// ---------------------------------------------------------------------------- 5791 5792AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5793 : RefBase(), 5794 mAudioFlinger(audioFlinger), 5795 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5796 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5797 mPid(pid), 5798 mTimedTrackCount(0) 5799{ 5800 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5801} 5802 5803// Client destructor must be called with AudioFlinger::mLock held 5804AudioFlinger::Client::~Client() 5805{ 5806 mAudioFlinger->removeClient_l(mPid); 5807} 5808 5809sp<MemoryDealer> AudioFlinger::Client::heap() const 5810{ 5811 return mMemoryDealer; 5812} 5813 5814// Reserve one of the limited slots for a timed audio track associated 5815// with this client 5816bool AudioFlinger::Client::reserveTimedTrack() 5817{ 5818 const int kMaxTimedTracksPerClient = 4; 5819 5820 Mutex::Autolock _l(mTimedTrackLock); 5821 5822 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5823 ALOGW("can not create timed track - pid %d has exceeded the limit", 5824 mPid); 5825 return false; 5826 } 5827 5828 mTimedTrackCount++; 5829 return true; 5830} 5831 5832// Release a slot for a timed audio track 5833void AudioFlinger::Client::releaseTimedTrack() 5834{ 5835 Mutex::Autolock _l(mTimedTrackLock); 5836 mTimedTrackCount--; 5837} 5838 5839// ---------------------------------------------------------------------------- 5840 5841AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5842 const sp<IAudioFlingerClient>& client, 5843 pid_t pid) 5844 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5845{ 5846} 5847 5848AudioFlinger::NotificationClient::~NotificationClient() 5849{ 5850} 5851 5852void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5853{ 5854 sp<NotificationClient> keep(this); 5855 mAudioFlinger->removeNotificationClient(mPid); 5856} 5857 5858// ---------------------------------------------------------------------------- 5859 5860AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5861 : BnAudioTrack(), 5862 mTrack(track) 5863{ 5864} 5865 5866AudioFlinger::TrackHandle::~TrackHandle() { 5867 // just stop the track on deletion, associated resources 5868 // will be freed from the main thread once all pending buffers have 5869 // been played. Unless it's not in the active track list, in which 5870 // case we free everything now... 5871 mTrack->destroy(); 5872} 5873 5874sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5875 return mTrack->getCblk(); 5876} 5877 5878status_t AudioFlinger::TrackHandle::start() { 5879 return mTrack->start(); 5880} 5881 5882void AudioFlinger::TrackHandle::stop() { 5883 mTrack->stop(); 5884} 5885 5886void AudioFlinger::TrackHandle::flush() { 5887 mTrack->flush(); 5888} 5889 5890void AudioFlinger::TrackHandle::mute(bool e) { 5891 mTrack->mute(e); 5892} 5893 5894void AudioFlinger::TrackHandle::pause() { 5895 mTrack->pause(); 5896} 5897 5898status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5899{ 5900 return mTrack->attachAuxEffect(EffectId); 5901} 5902 5903status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5904 sp<IMemory>* buffer) { 5905 if (!mTrack->isTimedTrack()) 5906 return INVALID_OPERATION; 5907 5908 PlaybackThread::TimedTrack* tt = 5909 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5910 return tt->allocateTimedBuffer(size, buffer); 5911} 5912 5913status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5914 int64_t pts) { 5915 if (!mTrack->isTimedTrack()) 5916 return INVALID_OPERATION; 5917 5918 PlaybackThread::TimedTrack* tt = 5919 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5920 return tt->queueTimedBuffer(buffer, pts); 5921} 5922 5923status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5924 const LinearTransform& xform, int target) { 5925 5926 if (!mTrack->isTimedTrack()) 5927 return INVALID_OPERATION; 5928 5929 PlaybackThread::TimedTrack* tt = 5930 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5931 return tt->setMediaTimeTransform( 5932 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5933} 5934 5935status_t AudioFlinger::TrackHandle::onTransact( 5936 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5937{ 5938 return BnAudioTrack::onTransact(code, data, reply, flags); 5939} 5940 5941// ---------------------------------------------------------------------------- 5942 5943sp<IAudioRecord> AudioFlinger::openRecord( 5944 pid_t pid, 5945 audio_io_handle_t input, 5946 uint32_t sampleRate, 5947 audio_format_t format, 5948 audio_channel_mask_t channelMask, 5949 size_t frameCount, 5950 IAudioFlinger::track_flags_t flags, 5951 pid_t tid, 5952 int *sessionId, 5953 status_t *status) 5954{ 5955 sp<RecordThread::RecordTrack> recordTrack; 5956 sp<RecordHandle> recordHandle; 5957 sp<Client> client; 5958 status_t lStatus; 5959 RecordThread *thread; 5960 size_t inFrameCount; 5961 int lSessionId; 5962 5963 // check calling permissions 5964 if (!recordingAllowed()) { 5965 lStatus = PERMISSION_DENIED; 5966 goto Exit; 5967 } 5968 5969 // add client to list 5970 { // scope for mLock 5971 Mutex::Autolock _l(mLock); 5972 thread = checkRecordThread_l(input); 5973 if (thread == NULL) { 5974 lStatus = BAD_VALUE; 5975 goto Exit; 5976 } 5977 5978 client = registerPid_l(pid); 5979 5980 // If no audio session id is provided, create one here 5981 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5982 lSessionId = *sessionId; 5983 } else { 5984 lSessionId = nextUniqueId(); 5985 if (sessionId != NULL) { 5986 *sessionId = lSessionId; 5987 } 5988 } 5989 // create new record track. 5990 // The record track uses one track in mHardwareMixerThread by convention. 5991 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5992 frameCount, lSessionId, flags, tid, &lStatus); 5993 } 5994 if (lStatus != NO_ERROR) { 5995 // remove local strong reference to Client before deleting the RecordTrack so that the 5996 // Client destructor is called by the TrackBase destructor with mLock held 5997 client.clear(); 5998 recordTrack.clear(); 5999 goto Exit; 6000 } 6001 6002 // return to handle to client 6003 recordHandle = new RecordHandle(recordTrack); 6004 lStatus = NO_ERROR; 6005 6006Exit: 6007 if (status) { 6008 *status = lStatus; 6009 } 6010 return recordHandle; 6011} 6012 6013// ---------------------------------------------------------------------------- 6014 6015AudioFlinger::RecordHandle::RecordHandle( 6016 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 6017 : BnAudioRecord(), 6018 mRecordTrack(recordTrack) 6019{ 6020} 6021 6022AudioFlinger::RecordHandle::~RecordHandle() { 6023 stop_nonvirtual(); 6024 mRecordTrack->destroy(); 6025} 6026 6027sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 6028 return mRecordTrack->getCblk(); 6029} 6030 6031status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6032 int triggerSession) { 6033 ALOGV("RecordHandle::start()"); 6034 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6035} 6036 6037void AudioFlinger::RecordHandle::stop() { 6038 stop_nonvirtual(); 6039} 6040 6041void AudioFlinger::RecordHandle::stop_nonvirtual() { 6042 ALOGV("RecordHandle::stop()"); 6043 mRecordTrack->stop(); 6044} 6045 6046status_t AudioFlinger::RecordHandle::onTransact( 6047 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6048{ 6049 return BnAudioRecord::onTransact(code, data, reply, flags); 6050} 6051 6052// ---------------------------------------------------------------------------- 6053 6054AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6055 AudioStreamIn *input, 6056 uint32_t sampleRate, 6057 audio_channel_mask_t channelMask, 6058 audio_io_handle_t id, 6059 audio_devices_t device, 6060 const sp<NBAIO_Sink>& teeSink) : 6061 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6062 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6063 // mRsmpInIndex and mInputBytes set by readInputParameters() 6064 mReqChannelCount(popcount(channelMask)), 6065 mReqSampleRate(sampleRate), 6066 // mBytesRead is only meaningful while active, and so is cleared in start() 6067 // (but might be better to also clear here for dump?) 6068 mTeeSink(teeSink) 6069{ 6070 snprintf(mName, kNameLength, "AudioIn_%X", id); 6071 6072 readInputParameters(); 6073 6074} 6075 6076 6077AudioFlinger::RecordThread::~RecordThread() 6078{ 6079 delete[] mRsmpInBuffer; 6080 delete mResampler; 6081 delete[] mRsmpOutBuffer; 6082} 6083 6084void AudioFlinger::RecordThread::onFirstRef() 6085{ 6086 run(mName, PRIORITY_URGENT_AUDIO); 6087} 6088 6089status_t AudioFlinger::RecordThread::readyToRun() 6090{ 6091 status_t status = initCheck(); 6092 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6093 return status; 6094} 6095 6096bool AudioFlinger::RecordThread::threadLoop() 6097{ 6098 AudioBufferProvider::Buffer buffer; 6099 sp<RecordTrack> activeTrack; 6100 Vector< sp<EffectChain> > effectChains; 6101 6102 nsecs_t lastWarning = 0; 6103 6104 inputStandBy(); 6105 acquireWakeLock(); 6106 6107 // used to verify we've read at least once before evaluating how many bytes were read 6108 bool readOnce = false; 6109 6110 // start recording 6111 while (!exitPending()) { 6112 6113 processConfigEvents(); 6114 6115 { // scope for mLock 6116 Mutex::Autolock _l(mLock); 6117 checkForNewParameters_l(); 6118 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6119 standby(); 6120 6121 if (exitPending()) { 6122 break; 6123 } 6124 6125 releaseWakeLock_l(); 6126 ALOGV("RecordThread: loop stopping"); 6127 // go to sleep 6128 mWaitWorkCV.wait(mLock); 6129 ALOGV("RecordThread: loop starting"); 6130 acquireWakeLock_l(); 6131 continue; 6132 } 6133 if (mActiveTrack != 0) { 6134 if (mActiveTrack->mState == TrackBase::PAUSING) { 6135 standby(); 6136 mActiveTrack.clear(); 6137 mStartStopCond.broadcast(); 6138 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6139 if (mReqChannelCount != mActiveTrack->channelCount()) { 6140 mActiveTrack.clear(); 6141 mStartStopCond.broadcast(); 6142 } else if (readOnce) { 6143 // record start succeeds only if first read from audio input 6144 // succeeds 6145 if (mBytesRead >= 0) { 6146 mActiveTrack->mState = TrackBase::ACTIVE; 6147 } else { 6148 mActiveTrack.clear(); 6149 } 6150 mStartStopCond.broadcast(); 6151 } 6152 mStandby = false; 6153 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6154 removeTrack_l(mActiveTrack); 6155 mActiveTrack.clear(); 6156 } 6157 } 6158 lockEffectChains_l(effectChains); 6159 } 6160 6161 if (mActiveTrack != 0) { 6162 if (mActiveTrack->mState != TrackBase::ACTIVE && 6163 mActiveTrack->mState != TrackBase::RESUMING) { 6164 unlockEffectChains(effectChains); 6165 usleep(kRecordThreadSleepUs); 6166 continue; 6167 } 6168 for (size_t i = 0; i < effectChains.size(); i ++) { 6169 effectChains[i]->process_l(); 6170 } 6171 6172 buffer.frameCount = mFrameCount; 6173 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6174 readOnce = true; 6175 size_t framesOut = buffer.frameCount; 6176 if (mResampler == NULL) { 6177 // no resampling 6178 while (framesOut) { 6179 size_t framesIn = mFrameCount - mRsmpInIndex; 6180 if (framesIn) { 6181 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6182 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6183 mActiveTrack->mFrameSize; 6184 if (framesIn > framesOut) 6185 framesIn = framesOut; 6186 mRsmpInIndex += framesIn; 6187 framesOut -= framesIn; 6188 if ((int)mChannelCount == mReqChannelCount || 6189 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6190 memcpy(dst, src, framesIn * mFrameSize); 6191 } else { 6192 if (mChannelCount == 1) { 6193 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6194 (int16_t *)src, framesIn); 6195 } else { 6196 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6197 (int16_t *)src, framesIn); 6198 } 6199 } 6200 } 6201 if (framesOut && mFrameCount == mRsmpInIndex) { 6202 void *readInto; 6203 if (framesOut == mFrameCount && 6204 ((int)mChannelCount == mReqChannelCount || 6205 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6206 readInto = buffer.raw; 6207 framesOut = 0; 6208 } else { 6209 readInto = mRsmpInBuffer; 6210 mRsmpInIndex = 0; 6211 } 6212 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6213 if (mBytesRead <= 0) { 6214 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6215 { 6216 ALOGE("Error reading audio input"); 6217 // Force input into standby so that it tries to 6218 // recover at next read attempt 6219 inputStandBy(); 6220 usleep(kRecordThreadSleepUs); 6221 } 6222 mRsmpInIndex = mFrameCount; 6223 framesOut = 0; 6224 buffer.frameCount = 0; 6225 } else if (mTeeSink != 0) { 6226 (void) mTeeSink->write(readInto, 6227 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6228 } 6229 } 6230 } 6231 } else { 6232 // resampling 6233 6234 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6235 // alter output frame count as if we were expecting stereo samples 6236 if (mChannelCount == 1 && mReqChannelCount == 1) { 6237 framesOut >>= 1; 6238 } 6239 mResampler->resample(mRsmpOutBuffer, framesOut, 6240 this /* AudioBufferProvider* */); 6241 // ditherAndClamp() works as long as all buffers returned by 6242 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6243 if (mChannelCount == 2 && mReqChannelCount == 1) { 6244 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6245 // the resampler always outputs stereo samples: 6246 // do post stereo to mono conversion 6247 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6248 framesOut); 6249 } else { 6250 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6251 } 6252 6253 } 6254 if (mFramestoDrop == 0) { 6255 mActiveTrack->releaseBuffer(&buffer); 6256 } else { 6257 if (mFramestoDrop > 0) { 6258 mFramestoDrop -= buffer.frameCount; 6259 if (mFramestoDrop <= 0) { 6260 clearSyncStartEvent(); 6261 } 6262 } else { 6263 mFramestoDrop += buffer.frameCount; 6264 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6265 mSyncStartEvent->isCancelled()) { 6266 ALOGW("Synced record %s, session %d, trigger session %d", 6267 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6268 mActiveTrack->sessionId(), 6269 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6270 clearSyncStartEvent(); 6271 } 6272 } 6273 } 6274 mActiveTrack->clearOverflow(); 6275 } 6276 // client isn't retrieving buffers fast enough 6277 else { 6278 if (!mActiveTrack->setOverflow()) { 6279 nsecs_t now = systemTime(); 6280 if ((now - lastWarning) > kWarningThrottleNs) { 6281 ALOGW("RecordThread: buffer overflow"); 6282 lastWarning = now; 6283 } 6284 } 6285 // Release the processor for a while before asking for a new buffer. 6286 // This will give the application more chance to read from the buffer and 6287 // clear the overflow. 6288 usleep(kRecordThreadSleepUs); 6289 } 6290 } 6291 // enable changes in effect chain 6292 unlockEffectChains(effectChains); 6293 effectChains.clear(); 6294 } 6295 6296 standby(); 6297 6298 { 6299 Mutex::Autolock _l(mLock); 6300 mActiveTrack.clear(); 6301 mStartStopCond.broadcast(); 6302 } 6303 6304 releaseWakeLock(); 6305 6306 ALOGV("RecordThread %p exiting", this); 6307 return false; 6308} 6309 6310void AudioFlinger::RecordThread::standby() 6311{ 6312 if (!mStandby) { 6313 inputStandBy(); 6314 mStandby = true; 6315 } 6316} 6317 6318void AudioFlinger::RecordThread::inputStandBy() 6319{ 6320 mInput->stream->common.standby(&mInput->stream->common); 6321} 6322 6323sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6324 const sp<AudioFlinger::Client>& client, 6325 uint32_t sampleRate, 6326 audio_format_t format, 6327 audio_channel_mask_t channelMask, 6328 size_t frameCount, 6329 int sessionId, 6330 IAudioFlinger::track_flags_t flags, 6331 pid_t tid, 6332 status_t *status) 6333{ 6334 sp<RecordTrack> track; 6335 status_t lStatus; 6336 6337 lStatus = initCheck(); 6338 if (lStatus != NO_ERROR) { 6339 ALOGE("Audio driver not initialized."); 6340 goto Exit; 6341 } 6342 6343 // FIXME use flags and tid similar to createTrack_l() 6344 6345 { // scope for mLock 6346 Mutex::Autolock _l(mLock); 6347 6348 track = new RecordTrack(this, client, sampleRate, 6349 format, channelMask, frameCount, sessionId); 6350 6351 if (track->getCblk() == 0) { 6352 lStatus = NO_MEMORY; 6353 goto Exit; 6354 } 6355 mTracks.add(track); 6356 6357 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6358 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6359 mAudioFlinger->btNrecIsOff(); 6360 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6361 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6362 } 6363 lStatus = NO_ERROR; 6364 6365Exit: 6366 if (status) { 6367 *status = lStatus; 6368 } 6369 return track; 6370} 6371 6372status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6373 AudioSystem::sync_event_t event, 6374 int triggerSession) 6375{ 6376 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6377 sp<ThreadBase> strongMe = this; 6378 status_t status = NO_ERROR; 6379 6380 if (event == AudioSystem::SYNC_EVENT_NONE) { 6381 clearSyncStartEvent(); 6382 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6383 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6384 triggerSession, 6385 recordTrack->sessionId(), 6386 syncStartEventCallback, 6387 this); 6388 // Sync event can be cancelled by the trigger session if the track is not in a 6389 // compatible state in which case we start record immediately 6390 if (mSyncStartEvent->isCancelled()) { 6391 clearSyncStartEvent(); 6392 } else { 6393 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6394 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6395 } 6396 } 6397 6398 { 6399 AutoMutex lock(mLock); 6400 if (mActiveTrack != 0) { 6401 if (recordTrack != mActiveTrack.get()) { 6402 status = -EBUSY; 6403 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6404 mActiveTrack->mState = TrackBase::ACTIVE; 6405 } 6406 return status; 6407 } 6408 6409 recordTrack->mState = TrackBase::IDLE; 6410 mActiveTrack = recordTrack; 6411 mLock.unlock(); 6412 status_t status = AudioSystem::startInput(mId); 6413 mLock.lock(); 6414 if (status != NO_ERROR) { 6415 mActiveTrack.clear(); 6416 clearSyncStartEvent(); 6417 return status; 6418 } 6419 mRsmpInIndex = mFrameCount; 6420 mBytesRead = 0; 6421 if (mResampler != NULL) { 6422 mResampler->reset(); 6423 } 6424 mActiveTrack->mState = TrackBase::RESUMING; 6425 // signal thread to start 6426 ALOGV("Signal record thread"); 6427 mWaitWorkCV.broadcast(); 6428 // do not wait for mStartStopCond if exiting 6429 if (exitPending()) { 6430 mActiveTrack.clear(); 6431 status = INVALID_OPERATION; 6432 goto startError; 6433 } 6434 mStartStopCond.wait(mLock); 6435 if (mActiveTrack == 0) { 6436 ALOGV("Record failed to start"); 6437 status = BAD_VALUE; 6438 goto startError; 6439 } 6440 ALOGV("Record started OK"); 6441 return status; 6442 } 6443startError: 6444 AudioSystem::stopInput(mId); 6445 clearSyncStartEvent(); 6446 return status; 6447} 6448 6449void AudioFlinger::RecordThread::clearSyncStartEvent() 6450{ 6451 if (mSyncStartEvent != 0) { 6452 mSyncStartEvent->cancel(); 6453 } 6454 mSyncStartEvent.clear(); 6455 mFramestoDrop = 0; 6456} 6457 6458void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6459{ 6460 sp<SyncEvent> strongEvent = event.promote(); 6461 6462 if (strongEvent != 0) { 6463 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6464 me->handleSyncStartEvent(strongEvent); 6465 } 6466} 6467 6468void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6469{ 6470 if (event == mSyncStartEvent) { 6471 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6472 // from audio HAL 6473 mFramestoDrop = mFrameCount * 2; 6474 } 6475} 6476 6477bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6478 ALOGV("RecordThread::stop"); 6479 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6480 return false; 6481 } 6482 recordTrack->mState = TrackBase::PAUSING; 6483 // do not wait for mStartStopCond if exiting 6484 if (exitPending()) { 6485 return true; 6486 } 6487 mStartStopCond.wait(mLock); 6488 // if we have been restarted, recordTrack == mActiveTrack.get() here 6489 if (exitPending() || recordTrack != mActiveTrack.get()) { 6490 ALOGV("Record stopped OK"); 6491 return true; 6492 } 6493 return false; 6494} 6495 6496bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6497{ 6498 return false; 6499} 6500 6501status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6502{ 6503#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6504 if (!isValidSyncEvent(event)) { 6505 return BAD_VALUE; 6506 } 6507 6508 int eventSession = event->triggerSession(); 6509 status_t ret = NAME_NOT_FOUND; 6510 6511 Mutex::Autolock _l(mLock); 6512 6513 for (size_t i = 0; i < mTracks.size(); i++) { 6514 sp<RecordTrack> track = mTracks[i]; 6515 if (eventSession == track->sessionId()) { 6516 (void) track->setSyncEvent(event); 6517 ret = NO_ERROR; 6518 } 6519 } 6520 return ret; 6521#else 6522 return BAD_VALUE; 6523#endif 6524} 6525 6526void AudioFlinger::RecordThread::RecordTrack::destroy() 6527{ 6528 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6529 sp<RecordTrack> keep(this); 6530 { 6531 sp<ThreadBase> thread = mThread.promote(); 6532 if (thread != 0) { 6533 if (mState == ACTIVE || mState == RESUMING) { 6534 AudioSystem::stopInput(thread->id()); 6535 } 6536 AudioSystem::releaseInput(thread->id()); 6537 Mutex::Autolock _l(thread->mLock); 6538 RecordThread *recordThread = (RecordThread *) thread.get(); 6539 recordThread->destroyTrack_l(this); 6540 } 6541 } 6542} 6543 6544// destroyTrack_l() must be called with ThreadBase::mLock held 6545void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6546{ 6547 track->mState = TrackBase::TERMINATED; 6548 // active tracks are removed by threadLoop() 6549 if (mActiveTrack != track) { 6550 removeTrack_l(track); 6551 } 6552} 6553 6554void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6555{ 6556 mTracks.remove(track); 6557 // need anything related to effects here? 6558} 6559 6560void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6561{ 6562 dumpInternals(fd, args); 6563 dumpTracks(fd, args); 6564 dumpEffectChains(fd, args); 6565} 6566 6567void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6568{ 6569 const size_t SIZE = 256; 6570 char buffer[SIZE]; 6571 String8 result; 6572 6573 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6574 result.append(buffer); 6575 6576 if (mActiveTrack != 0) { 6577 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6578 result.append(buffer); 6579 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6580 result.append(buffer); 6581 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6582 result.append(buffer); 6583 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6584 result.append(buffer); 6585 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 6586 result.append(buffer); 6587 } else { 6588 result.append("No active record client\n"); 6589 } 6590 6591 write(fd, result.string(), result.size()); 6592 6593 dumpBase(fd, args); 6594} 6595 6596void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6597{ 6598 const size_t SIZE = 256; 6599 char buffer[SIZE]; 6600 String8 result; 6601 6602 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6603 result.append(buffer); 6604 RecordTrack::appendDumpHeader(result); 6605 for (size_t i = 0; i < mTracks.size(); ++i) { 6606 sp<RecordTrack> track = mTracks[i]; 6607 if (track != 0) { 6608 track->dump(buffer, SIZE); 6609 result.append(buffer); 6610 } 6611 } 6612 6613 if (mActiveTrack != 0) { 6614 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6615 result.append(buffer); 6616 RecordTrack::appendDumpHeader(result); 6617 mActiveTrack->dump(buffer, SIZE); 6618 result.append(buffer); 6619 6620 } 6621 write(fd, result.string(), result.size()); 6622} 6623 6624// AudioBufferProvider interface 6625status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6626{ 6627 size_t framesReq = buffer->frameCount; 6628 size_t framesReady = mFrameCount - mRsmpInIndex; 6629 int channelCount; 6630 6631 if (framesReady == 0) { 6632 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6633 if (mBytesRead <= 0) { 6634 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6635 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6636 // Force input into standby so that it tries to 6637 // recover at next read attempt 6638 inputStandBy(); 6639 usleep(kRecordThreadSleepUs); 6640 } 6641 buffer->raw = NULL; 6642 buffer->frameCount = 0; 6643 return NOT_ENOUGH_DATA; 6644 } 6645 mRsmpInIndex = 0; 6646 framesReady = mFrameCount; 6647 } 6648 6649 if (framesReq > framesReady) { 6650 framesReq = framesReady; 6651 } 6652 6653 if (mChannelCount == 1 && mReqChannelCount == 2) { 6654 channelCount = 1; 6655 } else { 6656 channelCount = 2; 6657 } 6658 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6659 buffer->frameCount = framesReq; 6660 return NO_ERROR; 6661} 6662 6663// AudioBufferProvider interface 6664void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6665{ 6666 mRsmpInIndex += buffer->frameCount; 6667 buffer->frameCount = 0; 6668} 6669 6670bool AudioFlinger::RecordThread::checkForNewParameters_l() 6671{ 6672 bool reconfig = false; 6673 6674 while (!mNewParameters.isEmpty()) { 6675 status_t status = NO_ERROR; 6676 String8 keyValuePair = mNewParameters[0]; 6677 AudioParameter param = AudioParameter(keyValuePair); 6678 int value; 6679 audio_format_t reqFormat = mFormat; 6680 uint32_t reqSamplingRate = mReqSampleRate; 6681 int reqChannelCount = mReqChannelCount; 6682 6683 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6684 reqSamplingRate = value; 6685 reconfig = true; 6686 } 6687 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6688 reqFormat = (audio_format_t) value; 6689 reconfig = true; 6690 } 6691 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6692 reqChannelCount = popcount(value); 6693 reconfig = true; 6694 } 6695 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6696 // do not accept frame count changes if tracks are open as the track buffer 6697 // size depends on frame count and correct behavior would not be guaranteed 6698 // if frame count is changed after track creation 6699 if (mActiveTrack != 0) { 6700 status = INVALID_OPERATION; 6701 } else { 6702 reconfig = true; 6703 } 6704 } 6705 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6706 // forward device change to effects that have requested to be 6707 // aware of attached audio device. 6708 for (size_t i = 0; i < mEffectChains.size(); i++) { 6709 mEffectChains[i]->setDevice_l(value); 6710 } 6711 6712 // store input device and output device but do not forward output device to audio HAL. 6713 // Note that status is ignored by the caller for output device 6714 // (see AudioFlinger::setParameters() 6715 if (audio_is_output_devices(value)) { 6716 mOutDevice = value; 6717 status = BAD_VALUE; 6718 } else { 6719 mInDevice = value; 6720 // disable AEC and NS if the device is a BT SCO headset supporting those 6721 // pre processings 6722 if (mTracks.size() > 0) { 6723 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6724 mAudioFlinger->btNrecIsOff(); 6725 for (size_t i = 0; i < mTracks.size(); i++) { 6726 sp<RecordTrack> track = mTracks[i]; 6727 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6728 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6729 } 6730 } 6731 } 6732 } 6733 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6734 mAudioSource != (audio_source_t)value) { 6735 // forward device change to effects that have requested to be 6736 // aware of attached audio device. 6737 for (size_t i = 0; i < mEffectChains.size(); i++) { 6738 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6739 } 6740 mAudioSource = (audio_source_t)value; 6741 } 6742 if (status == NO_ERROR) { 6743 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6744 keyValuePair.string()); 6745 if (status == INVALID_OPERATION) { 6746 inputStandBy(); 6747 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6748 keyValuePair.string()); 6749 } 6750 if (reconfig) { 6751 if (status == BAD_VALUE && 6752 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6753 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6754 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6755 <= (2 * reqSamplingRate)) && 6756 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6757 <= FCC_2 && 6758 (reqChannelCount <= FCC_2)) { 6759 status = NO_ERROR; 6760 } 6761 if (status == NO_ERROR) { 6762 readInputParameters(); 6763 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6764 } 6765 } 6766 } 6767 6768 mNewParameters.removeAt(0); 6769 6770 mParamStatus = status; 6771 mParamCond.signal(); 6772 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6773 // already timed out waiting for the status and will never signal the condition. 6774 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6775 } 6776 return reconfig; 6777} 6778 6779String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6780{ 6781 char *s; 6782 String8 out_s8 = String8(); 6783 6784 Mutex::Autolock _l(mLock); 6785 if (initCheck() != NO_ERROR) { 6786 return out_s8; 6787 } 6788 6789 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6790 out_s8 = String8(s); 6791 free(s); 6792 return out_s8; 6793} 6794 6795void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6796 AudioSystem::OutputDescriptor desc; 6797 void *param2 = NULL; 6798 6799 switch (event) { 6800 case AudioSystem::INPUT_OPENED: 6801 case AudioSystem::INPUT_CONFIG_CHANGED: 6802 desc.channels = mChannelMask; 6803 desc.samplingRate = mSampleRate; 6804 desc.format = mFormat; 6805 desc.frameCount = mFrameCount; 6806 desc.latency = 0; 6807 param2 = &desc; 6808 break; 6809 6810 case AudioSystem::INPUT_CLOSED: 6811 default: 6812 break; 6813 } 6814 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6815} 6816 6817void AudioFlinger::RecordThread::readInputParameters() 6818{ 6819 delete mRsmpInBuffer; 6820 // mRsmpInBuffer is always assigned a new[] below 6821 delete mRsmpOutBuffer; 6822 mRsmpOutBuffer = NULL; 6823 delete mResampler; 6824 mResampler = NULL; 6825 6826 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6827 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6828 mChannelCount = (uint16_t)popcount(mChannelMask); 6829 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6830 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6831 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6832 mFrameCount = mInputBytes / mFrameSize; 6833 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6834 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6835 6836 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6837 { 6838 int channelCount; 6839 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6840 // stereo to mono post process as the resampler always outputs stereo. 6841 if (mChannelCount == 1 && mReqChannelCount == 2) { 6842 channelCount = 1; 6843 } else { 6844 channelCount = 2; 6845 } 6846 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6847 mResampler->setSampleRate(mSampleRate); 6848 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6849 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6850 6851 // optmization: if mono to mono, alter input frame count as if we were inputing 6852 // stereo samples 6853 if (mChannelCount == 1 && mReqChannelCount == 1) { 6854 mFrameCount >>= 1; 6855 } 6856 6857 } 6858 mRsmpInIndex = mFrameCount; 6859} 6860 6861unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6862{ 6863 Mutex::Autolock _l(mLock); 6864 if (initCheck() != NO_ERROR) { 6865 return 0; 6866 } 6867 6868 return mInput->stream->get_input_frames_lost(mInput->stream); 6869} 6870 6871uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6872{ 6873 Mutex::Autolock _l(mLock); 6874 uint32_t result = 0; 6875 if (getEffectChain_l(sessionId) != 0) { 6876 result = EFFECT_SESSION; 6877 } 6878 6879 for (size_t i = 0; i < mTracks.size(); ++i) { 6880 if (sessionId == mTracks[i]->sessionId()) { 6881 result |= TRACK_SESSION; 6882 break; 6883 } 6884 } 6885 6886 return result; 6887} 6888 6889KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6890{ 6891 KeyedVector<int, bool> ids; 6892 Mutex::Autolock _l(mLock); 6893 for (size_t j = 0; j < mTracks.size(); ++j) { 6894 sp<RecordThread::RecordTrack> track = mTracks[j]; 6895 int sessionId = track->sessionId(); 6896 if (ids.indexOfKey(sessionId) < 0) { 6897 ids.add(sessionId, true); 6898 } 6899 } 6900 return ids; 6901} 6902 6903AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6904{ 6905 Mutex::Autolock _l(mLock); 6906 AudioStreamIn *input = mInput; 6907 mInput = NULL; 6908 return input; 6909} 6910 6911// this method must always be called either with ThreadBase mLock held or inside the thread loop 6912audio_stream_t* AudioFlinger::RecordThread::stream() const 6913{ 6914 if (mInput == NULL) { 6915 return NULL; 6916 } 6917 return &mInput->stream->common; 6918} 6919 6920 6921// ---------------------------------------------------------------------------- 6922 6923audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6924{ 6925 if (!settingsAllowed()) { 6926 return 0; 6927 } 6928 Mutex::Autolock _l(mLock); 6929 return loadHwModule_l(name); 6930} 6931 6932// loadHwModule_l() must be called with AudioFlinger::mLock held 6933audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6934{ 6935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6936 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6937 ALOGW("loadHwModule() module %s already loaded", name); 6938 return mAudioHwDevs.keyAt(i); 6939 } 6940 } 6941 6942 audio_hw_device_t *dev; 6943 6944 int rc = load_audio_interface(name, &dev); 6945 if (rc) { 6946 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6947 return 0; 6948 } 6949 6950 mHardwareStatus = AUDIO_HW_INIT; 6951 rc = dev->init_check(dev); 6952 mHardwareStatus = AUDIO_HW_IDLE; 6953 if (rc) { 6954 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6955 return 0; 6956 } 6957 6958 // Check and cache this HAL's level of support for master mute and master 6959 // volume. If this is the first HAL opened, and it supports the get 6960 // methods, use the initial values provided by the HAL as the current 6961 // master mute and volume settings. 6962 6963 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6964 { // scope for auto-lock pattern 6965 AutoMutex lock(mHardwareLock); 6966 6967 if (0 == mAudioHwDevs.size()) { 6968 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6969 if (NULL != dev->get_master_volume) { 6970 float mv; 6971 if (OK == dev->get_master_volume(dev, &mv)) { 6972 mMasterVolume = mv; 6973 } 6974 } 6975 6976 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6977 if (NULL != dev->get_master_mute) { 6978 bool mm; 6979 if (OK == dev->get_master_mute(dev, &mm)) { 6980 mMasterMute = mm; 6981 } 6982 } 6983 } 6984 6985 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6986 if ((NULL != dev->set_master_volume) && 6987 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6988 flags = static_cast<AudioHwDevice::Flags>(flags | 6989 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6990 } 6991 6992 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6993 if ((NULL != dev->set_master_mute) && 6994 (OK == dev->set_master_mute(dev, mMasterMute))) { 6995 flags = static_cast<AudioHwDevice::Flags>(flags | 6996 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6997 } 6998 6999 mHardwareStatus = AUDIO_HW_IDLE; 7000 } 7001 7002 audio_module_handle_t handle = nextUniqueId(); 7003 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 7004 7005 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 7006 name, dev->common.module->name, dev->common.module->id, handle); 7007 7008 return handle; 7009 7010} 7011 7012// ---------------------------------------------------------------------------- 7013 7014uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 7015{ 7016 Mutex::Autolock _l(mLock); 7017 PlaybackThread *thread = primaryPlaybackThread_l(); 7018 return thread != NULL ? thread->sampleRate() : 0; 7019} 7020 7021size_t AudioFlinger::getPrimaryOutputFrameCount() 7022{ 7023 Mutex::Autolock _l(mLock); 7024 PlaybackThread *thread = primaryPlaybackThread_l(); 7025 return thread != NULL ? thread->frameCountHAL() : 0; 7026} 7027 7028// ---------------------------------------------------------------------------- 7029 7030audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 7031 audio_devices_t *pDevices, 7032 uint32_t *pSamplingRate, 7033 audio_format_t *pFormat, 7034 audio_channel_mask_t *pChannelMask, 7035 uint32_t *pLatencyMs, 7036 audio_output_flags_t flags) 7037{ 7038 status_t status; 7039 PlaybackThread *thread = NULL; 7040 struct audio_config config = { 7041 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7042 channel_mask: pChannelMask ? *pChannelMask : 0, 7043 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7044 }; 7045 audio_stream_out_t *outStream = NULL; 7046 AudioHwDevice *outHwDev; 7047 7048 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7049 module, 7050 (pDevices != NULL) ? *pDevices : 0, 7051 config.sample_rate, 7052 config.format, 7053 config.channel_mask, 7054 flags); 7055 7056 if (pDevices == NULL || *pDevices == 0) { 7057 return 0; 7058 } 7059 7060 Mutex::Autolock _l(mLock); 7061 7062 outHwDev = findSuitableHwDev_l(module, *pDevices); 7063 if (outHwDev == NULL) 7064 return 0; 7065 7066 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7067 audio_io_handle_t id = nextUniqueId(); 7068 7069 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7070 7071 status = hwDevHal->open_output_stream(hwDevHal, 7072 id, 7073 *pDevices, 7074 (audio_output_flags_t)flags, 7075 &config, 7076 &outStream); 7077 7078 mHardwareStatus = AUDIO_HW_IDLE; 7079 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7080 "Channels %x, status %d", 7081 outStream, 7082 config.sample_rate, 7083 config.format, 7084 config.channel_mask, 7085 status); 7086 7087 if (status == NO_ERROR && outStream != NULL) { 7088 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7089 7090 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7091 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7092 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7093 thread = new DirectOutputThread(this, output, id, *pDevices); 7094 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7095 } else { 7096 thread = new MixerThread(this, output, id, *pDevices); 7097 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7098 } 7099 mPlaybackThreads.add(id, thread); 7100 7101 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7102 if (pFormat != NULL) *pFormat = config.format; 7103 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7104 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7105 7106 // notify client processes of the new output creation 7107 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7108 7109 // the first primary output opened designates the primary hw device 7110 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7111 ALOGI("Using module %d has the primary audio interface", module); 7112 mPrimaryHardwareDev = outHwDev; 7113 7114 AutoMutex lock(mHardwareLock); 7115 mHardwareStatus = AUDIO_HW_SET_MODE; 7116 hwDevHal->set_mode(hwDevHal, mMode); 7117 mHardwareStatus = AUDIO_HW_IDLE; 7118 } 7119 return id; 7120 } 7121 7122 return 0; 7123} 7124 7125audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7126 audio_io_handle_t output2) 7127{ 7128 Mutex::Autolock _l(mLock); 7129 MixerThread *thread1 = checkMixerThread_l(output1); 7130 MixerThread *thread2 = checkMixerThread_l(output2); 7131 7132 if (thread1 == NULL || thread2 == NULL) { 7133 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7134 output2); 7135 return 0; 7136 } 7137 7138 audio_io_handle_t id = nextUniqueId(); 7139 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7140 thread->addOutputTrack(thread2); 7141 mPlaybackThreads.add(id, thread); 7142 // notify client processes of the new output creation 7143 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7144 return id; 7145} 7146 7147status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7148{ 7149 return closeOutput_nonvirtual(output); 7150} 7151 7152status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7153{ 7154 // keep strong reference on the playback thread so that 7155 // it is not destroyed while exit() is executed 7156 sp<PlaybackThread> thread; 7157 { 7158 Mutex::Autolock _l(mLock); 7159 thread = checkPlaybackThread_l(output); 7160 if (thread == NULL) { 7161 return BAD_VALUE; 7162 } 7163 7164 ALOGV("closeOutput() %d", output); 7165 7166 if (thread->type() == ThreadBase::MIXER) { 7167 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7168 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7169 DuplicatingThread *dupThread = 7170 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7171 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7172 } 7173 } 7174 } 7175 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7176 mPlaybackThreads.removeItem(output); 7177 } 7178 thread->exit(); 7179 // The thread entity (active unit of execution) is no longer running here, 7180 // but the ThreadBase container still exists. 7181 7182 if (thread->type() != ThreadBase::DUPLICATING) { 7183 AudioStreamOut *out = thread->clearOutput(); 7184 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7185 // from now on thread->mOutput is NULL 7186 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7187 delete out; 7188 } 7189 return NO_ERROR; 7190} 7191 7192status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7193{ 7194 Mutex::Autolock _l(mLock); 7195 PlaybackThread *thread = checkPlaybackThread_l(output); 7196 7197 if (thread == NULL) { 7198 return BAD_VALUE; 7199 } 7200 7201 ALOGV("suspendOutput() %d", output); 7202 thread->suspend(); 7203 7204 return NO_ERROR; 7205} 7206 7207status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7208{ 7209 Mutex::Autolock _l(mLock); 7210 PlaybackThread *thread = checkPlaybackThread_l(output); 7211 7212 if (thread == NULL) { 7213 return BAD_VALUE; 7214 } 7215 7216 ALOGV("restoreOutput() %d", output); 7217 7218 thread->restore(); 7219 7220 return NO_ERROR; 7221} 7222 7223audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7224 audio_devices_t *pDevices, 7225 uint32_t *pSamplingRate, 7226 audio_format_t *pFormat, 7227 audio_channel_mask_t *pChannelMask) 7228{ 7229 status_t status; 7230 RecordThread *thread = NULL; 7231 struct audio_config config = { 7232 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7233 channel_mask: pChannelMask ? *pChannelMask : 0, 7234 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7235 }; 7236 uint32_t reqSamplingRate = config.sample_rate; 7237 audio_format_t reqFormat = config.format; 7238 audio_channel_mask_t reqChannels = config.channel_mask; 7239 audio_stream_in_t *inStream = NULL; 7240 AudioHwDevice *inHwDev; 7241 7242 if (pDevices == NULL || *pDevices == 0) { 7243 return 0; 7244 } 7245 7246 Mutex::Autolock _l(mLock); 7247 7248 inHwDev = findSuitableHwDev_l(module, *pDevices); 7249 if (inHwDev == NULL) 7250 return 0; 7251 7252 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7253 audio_io_handle_t id = nextUniqueId(); 7254 7255 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7256 &inStream); 7257 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7258 "status %d", 7259 inStream, 7260 config.sample_rate, 7261 config.format, 7262 config.channel_mask, 7263 status); 7264 7265 // If the input could not be opened with the requested parameters and we can handle the 7266 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7267 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7268 if (status == BAD_VALUE && 7269 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7270 (config.sample_rate <= 2 * reqSamplingRate) && 7271 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7272 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7273 inStream = NULL; 7274 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7275 } 7276 7277 if (status == NO_ERROR && inStream != NULL) { 7278 7279 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7280 // or (re-)create if current Pipe is idle and does not match the new format 7281 sp<NBAIO_Sink> teeSink; 7282#ifdef TEE_SINK_INPUT_FRAMES 7283 enum { 7284 TEE_SINK_NO, // don't copy input 7285 TEE_SINK_NEW, // copy input using a new pipe 7286 TEE_SINK_OLD, // copy input using an existing pipe 7287 } kind; 7288 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7289 popcount(inStream->common.get_channels(&inStream->common))); 7290 if (format == Format_Invalid) { 7291 kind = TEE_SINK_NO; 7292 } else if (mRecordTeeSink == 0) { 7293 kind = TEE_SINK_NEW; 7294 } else if (mRecordTeeSink->getStrongCount() != 1) { 7295 kind = TEE_SINK_NO; 7296 } else if (format == mRecordTeeSink->format()) { 7297 kind = TEE_SINK_OLD; 7298 } else { 7299 kind = TEE_SINK_NEW; 7300 } 7301 switch (kind) { 7302 case TEE_SINK_NEW: { 7303 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7304 size_t numCounterOffers = 0; 7305 const NBAIO_Format offers[1] = {format}; 7306 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7307 ALOG_ASSERT(index == 0); 7308 PipeReader *pipeReader = new PipeReader(*pipe); 7309 numCounterOffers = 0; 7310 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7311 ALOG_ASSERT(index == 0); 7312 mRecordTeeSink = pipe; 7313 mRecordTeeSource = pipeReader; 7314 teeSink = pipe; 7315 } 7316 break; 7317 case TEE_SINK_OLD: 7318 teeSink = mRecordTeeSink; 7319 break; 7320 case TEE_SINK_NO: 7321 default: 7322 break; 7323 } 7324#endif 7325 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7326 7327 // Start record thread 7328 // RecorThread require both input and output device indication to forward to audio 7329 // pre processing modules 7330 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7331 7332 thread = new RecordThread(this, 7333 input, 7334 reqSamplingRate, 7335 reqChannels, 7336 id, 7337 device, teeSink); 7338 mRecordThreads.add(id, thread); 7339 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7340 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7341 if (pFormat != NULL) *pFormat = config.format; 7342 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7343 7344 // notify client processes of the new input creation 7345 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7346 return id; 7347 } 7348 7349 return 0; 7350} 7351 7352status_t AudioFlinger::closeInput(audio_io_handle_t input) 7353{ 7354 return closeInput_nonvirtual(input); 7355} 7356 7357status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7358{ 7359 // keep strong reference on the record thread so that 7360 // it is not destroyed while exit() is executed 7361 sp<RecordThread> thread; 7362 { 7363 Mutex::Autolock _l(mLock); 7364 thread = checkRecordThread_l(input); 7365 if (thread == 0) { 7366 return BAD_VALUE; 7367 } 7368 7369 ALOGV("closeInput() %d", input); 7370 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7371 mRecordThreads.removeItem(input); 7372 } 7373 thread->exit(); 7374 // The thread entity (active unit of execution) is no longer running here, 7375 // but the ThreadBase container still exists. 7376 7377 AudioStreamIn *in = thread->clearInput(); 7378 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7379 // from now on thread->mInput is NULL 7380 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7381 delete in; 7382 7383 return NO_ERROR; 7384} 7385 7386status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7387{ 7388 Mutex::Autolock _l(mLock); 7389 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7390 7391 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7392 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7393 thread->invalidateTracks(stream); 7394 } 7395 7396 return NO_ERROR; 7397} 7398 7399 7400int AudioFlinger::newAudioSessionId() 7401{ 7402 return nextUniqueId(); 7403} 7404 7405void AudioFlinger::acquireAudioSessionId(int audioSession) 7406{ 7407 Mutex::Autolock _l(mLock); 7408 pid_t caller = IPCThreadState::self()->getCallingPid(); 7409 ALOGV("acquiring %d from %d", audioSession, caller); 7410 size_t num = mAudioSessionRefs.size(); 7411 for (size_t i = 0; i< num; i++) { 7412 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7413 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7414 ref->mCnt++; 7415 ALOGV(" incremented refcount to %d", ref->mCnt); 7416 return; 7417 } 7418 } 7419 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7420 ALOGV(" added new entry for %d", audioSession); 7421} 7422 7423void AudioFlinger::releaseAudioSessionId(int audioSession) 7424{ 7425 Mutex::Autolock _l(mLock); 7426 pid_t caller = IPCThreadState::self()->getCallingPid(); 7427 ALOGV("releasing %d from %d", audioSession, caller); 7428 size_t num = mAudioSessionRefs.size(); 7429 for (size_t i = 0; i< num; i++) { 7430 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7431 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7432 ref->mCnt--; 7433 ALOGV(" decremented refcount to %d", ref->mCnt); 7434 if (ref->mCnt == 0) { 7435 mAudioSessionRefs.removeAt(i); 7436 delete ref; 7437 purgeStaleEffects_l(); 7438 } 7439 return; 7440 } 7441 } 7442 ALOGW("session id %d not found for pid %d", audioSession, caller); 7443} 7444 7445void AudioFlinger::purgeStaleEffects_l() { 7446 7447 ALOGV("purging stale effects"); 7448 7449 Vector< sp<EffectChain> > chains; 7450 7451 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7452 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7453 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7454 sp<EffectChain> ec = t->mEffectChains[j]; 7455 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7456 chains.push(ec); 7457 } 7458 } 7459 } 7460 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7461 sp<RecordThread> t = mRecordThreads.valueAt(i); 7462 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7463 sp<EffectChain> ec = t->mEffectChains[j]; 7464 chains.push(ec); 7465 } 7466 } 7467 7468 for (size_t i = 0; i < chains.size(); i++) { 7469 sp<EffectChain> ec = chains[i]; 7470 int sessionid = ec->sessionId(); 7471 sp<ThreadBase> t = ec->mThread.promote(); 7472 if (t == 0) { 7473 continue; 7474 } 7475 size_t numsessionrefs = mAudioSessionRefs.size(); 7476 bool found = false; 7477 for (size_t k = 0; k < numsessionrefs; k++) { 7478 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7479 if (ref->mSessionid == sessionid) { 7480 ALOGV(" session %d still exists for %d with %d refs", 7481 sessionid, ref->mPid, ref->mCnt); 7482 found = true; 7483 break; 7484 } 7485 } 7486 if (!found) { 7487 Mutex::Autolock _l (t->mLock); 7488 // remove all effects from the chain 7489 while (ec->mEffects.size()) { 7490 sp<EffectModule> effect = ec->mEffects[0]; 7491 effect->unPin(); 7492 t->removeEffect_l(effect); 7493 if (effect->purgeHandles()) { 7494 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7495 } 7496 AudioSystem::unregisterEffect(effect->id()); 7497 } 7498 } 7499 } 7500 return; 7501} 7502 7503// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7504AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7505{ 7506 return mPlaybackThreads.valueFor(output).get(); 7507} 7508 7509// checkMixerThread_l() must be called with AudioFlinger::mLock held 7510AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7511{ 7512 PlaybackThread *thread = checkPlaybackThread_l(output); 7513 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7514} 7515 7516// checkRecordThread_l() must be called with AudioFlinger::mLock held 7517AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7518{ 7519 return mRecordThreads.valueFor(input).get(); 7520} 7521 7522uint32_t AudioFlinger::nextUniqueId() 7523{ 7524 return android_atomic_inc(&mNextUniqueId); 7525} 7526 7527AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7528{ 7529 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7530 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7531 AudioStreamOut *output = thread->getOutput(); 7532 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7533 return thread; 7534 } 7535 } 7536 return NULL; 7537} 7538 7539audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7540{ 7541 PlaybackThread *thread = primaryPlaybackThread_l(); 7542 7543 if (thread == NULL) { 7544 return 0; 7545 } 7546 7547 return thread->outDevice(); 7548} 7549 7550sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7551 int triggerSession, 7552 int listenerSession, 7553 sync_event_callback_t callBack, 7554 void *cookie) 7555{ 7556 Mutex::Autolock _l(mLock); 7557 7558 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7559 status_t playStatus = NAME_NOT_FOUND; 7560 status_t recStatus = NAME_NOT_FOUND; 7561 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7562 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7563 if (playStatus == NO_ERROR) { 7564 return event; 7565 } 7566 } 7567 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7568 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7569 if (recStatus == NO_ERROR) { 7570 return event; 7571 } 7572 } 7573 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7574 mPendingSyncEvents.add(event); 7575 } else { 7576 ALOGV("createSyncEvent() invalid event %d", event->type()); 7577 event.clear(); 7578 } 7579 return event; 7580} 7581 7582// ---------------------------------------------------------------------------- 7583// Effect management 7584// ---------------------------------------------------------------------------- 7585 7586 7587status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7588{ 7589 Mutex::Autolock _l(mLock); 7590 return EffectQueryNumberEffects(numEffects); 7591} 7592 7593status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7594{ 7595 Mutex::Autolock _l(mLock); 7596 return EffectQueryEffect(index, descriptor); 7597} 7598 7599status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7600 effect_descriptor_t *descriptor) const 7601{ 7602 Mutex::Autolock _l(mLock); 7603 return EffectGetDescriptor(pUuid, descriptor); 7604} 7605 7606 7607sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7608 effect_descriptor_t *pDesc, 7609 const sp<IEffectClient>& effectClient, 7610 int32_t priority, 7611 audio_io_handle_t io, 7612 int sessionId, 7613 status_t *status, 7614 int *id, 7615 int *enabled) 7616{ 7617 status_t lStatus = NO_ERROR; 7618 sp<EffectHandle> handle; 7619 effect_descriptor_t desc; 7620 7621 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7622 pid, effectClient.get(), priority, sessionId, io); 7623 7624 if (pDesc == NULL) { 7625 lStatus = BAD_VALUE; 7626 goto Exit; 7627 } 7628 7629 // check audio settings permission for global effects 7630 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7631 lStatus = PERMISSION_DENIED; 7632 goto Exit; 7633 } 7634 7635 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7636 // that can only be created by audio policy manager (running in same process) 7637 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7638 lStatus = PERMISSION_DENIED; 7639 goto Exit; 7640 } 7641 7642 if (io == 0) { 7643 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7644 // output must be specified by AudioPolicyManager when using session 7645 // AUDIO_SESSION_OUTPUT_STAGE 7646 lStatus = BAD_VALUE; 7647 goto Exit; 7648 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7649 // if the output returned by getOutputForEffect() is removed before we lock the 7650 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7651 // and we will exit safely 7652 io = AudioSystem::getOutputForEffect(&desc); 7653 } 7654 } 7655 7656 { 7657 Mutex::Autolock _l(mLock); 7658 7659 7660 if (!EffectIsNullUuid(&pDesc->uuid)) { 7661 // if uuid is specified, request effect descriptor 7662 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7663 if (lStatus < 0) { 7664 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7665 goto Exit; 7666 } 7667 } else { 7668 // if uuid is not specified, look for an available implementation 7669 // of the required type in effect factory 7670 if (EffectIsNullUuid(&pDesc->type)) { 7671 ALOGW("createEffect() no effect type"); 7672 lStatus = BAD_VALUE; 7673 goto Exit; 7674 } 7675 uint32_t numEffects = 0; 7676 effect_descriptor_t d; 7677 d.flags = 0; // prevent compiler warning 7678 bool found = false; 7679 7680 lStatus = EffectQueryNumberEffects(&numEffects); 7681 if (lStatus < 0) { 7682 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7683 goto Exit; 7684 } 7685 for (uint32_t i = 0; i < numEffects; i++) { 7686 lStatus = EffectQueryEffect(i, &desc); 7687 if (lStatus < 0) { 7688 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7689 continue; 7690 } 7691 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7692 // If matching type found save effect descriptor. If the session is 7693 // 0 and the effect is not auxiliary, continue enumeration in case 7694 // an auxiliary version of this effect type is available 7695 found = true; 7696 d = desc; 7697 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7698 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7699 break; 7700 } 7701 } 7702 } 7703 if (!found) { 7704 lStatus = BAD_VALUE; 7705 ALOGW("createEffect() effect not found"); 7706 goto Exit; 7707 } 7708 // For same effect type, chose auxiliary version over insert version if 7709 // connect to output mix (Compliance to OpenSL ES) 7710 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7711 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7712 desc = d; 7713 } 7714 } 7715 7716 // Do not allow auxiliary effects on a session different from 0 (output mix) 7717 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7718 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7719 lStatus = INVALID_OPERATION; 7720 goto Exit; 7721 } 7722 7723 // check recording permission for visualizer 7724 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7725 !recordingAllowed()) { 7726 lStatus = PERMISSION_DENIED; 7727 goto Exit; 7728 } 7729 7730 // return effect descriptor 7731 *pDesc = desc; 7732 7733 // If output is not specified try to find a matching audio session ID in one of the 7734 // output threads. 7735 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7736 // because of code checking output when entering the function. 7737 // Note: io is never 0 when creating an effect on an input 7738 if (io == 0) { 7739 // look for the thread where the specified audio session is present 7740 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7741 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7742 io = mPlaybackThreads.keyAt(i); 7743 break; 7744 } 7745 } 7746 if (io == 0) { 7747 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7748 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7749 io = mRecordThreads.keyAt(i); 7750 break; 7751 } 7752 } 7753 } 7754 // If no output thread contains the requested session ID, default to 7755 // first output. The effect chain will be moved to the correct output 7756 // thread when a track with the same session ID is created 7757 if (io == 0 && mPlaybackThreads.size()) { 7758 io = mPlaybackThreads.keyAt(0); 7759 } 7760 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7761 } 7762 ThreadBase *thread = checkRecordThread_l(io); 7763 if (thread == NULL) { 7764 thread = checkPlaybackThread_l(io); 7765 if (thread == NULL) { 7766 ALOGE("createEffect() unknown output thread"); 7767 lStatus = BAD_VALUE; 7768 goto Exit; 7769 } 7770 } 7771 7772 sp<Client> client = registerPid_l(pid); 7773 7774 // create effect on selected output thread 7775 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7776 &desc, enabled, &lStatus); 7777 if (handle != 0 && id != NULL) { 7778 *id = handle->id(); 7779 } 7780 } 7781 7782Exit: 7783 if (status != NULL) { 7784 *status = lStatus; 7785 } 7786 return handle; 7787} 7788 7789status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7790 audio_io_handle_t dstOutput) 7791{ 7792 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7793 sessionId, srcOutput, dstOutput); 7794 Mutex::Autolock _l(mLock); 7795 if (srcOutput == dstOutput) { 7796 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7797 return NO_ERROR; 7798 } 7799 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7800 if (srcThread == NULL) { 7801 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7802 return BAD_VALUE; 7803 } 7804 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7805 if (dstThread == NULL) { 7806 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7807 return BAD_VALUE; 7808 } 7809 7810 Mutex::Autolock _dl(dstThread->mLock); 7811 Mutex::Autolock _sl(srcThread->mLock); 7812 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7813 7814 return NO_ERROR; 7815} 7816 7817// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7818status_t AudioFlinger::moveEffectChain_l(int sessionId, 7819 AudioFlinger::PlaybackThread *srcThread, 7820 AudioFlinger::PlaybackThread *dstThread, 7821 bool reRegister) 7822{ 7823 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7824 sessionId, srcThread, dstThread); 7825 7826 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7827 if (chain == 0) { 7828 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7829 sessionId, srcThread); 7830 return INVALID_OPERATION; 7831 } 7832 7833 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7834 // so that a new chain is created with correct parameters when first effect is added. This is 7835 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7836 // removed. 7837 srcThread->removeEffectChain_l(chain); 7838 7839 // transfer all effects one by one so that new effect chain is created on new thread with 7840 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7841 audio_io_handle_t dstOutput = dstThread->id(); 7842 sp<EffectChain> dstChain; 7843 uint32_t strategy = 0; // prevent compiler warning 7844 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7845 while (effect != 0) { 7846 srcThread->removeEffect_l(effect); 7847 dstThread->addEffect_l(effect); 7848 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7849 if (effect->state() == EffectModule::ACTIVE || 7850 effect->state() == EffectModule::STOPPING) { 7851 effect->start(); 7852 } 7853 // if the move request is not received from audio policy manager, the effect must be 7854 // re-registered with the new strategy and output 7855 if (dstChain == 0) { 7856 dstChain = effect->chain().promote(); 7857 if (dstChain == 0) { 7858 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7859 srcThread->addEffect_l(effect); 7860 return NO_INIT; 7861 } 7862 strategy = dstChain->strategy(); 7863 } 7864 if (reRegister) { 7865 AudioSystem::unregisterEffect(effect->id()); 7866 AudioSystem::registerEffect(&effect->desc(), 7867 dstOutput, 7868 strategy, 7869 sessionId, 7870 effect->id()); 7871 } 7872 effect = chain->getEffectFromId_l(0); 7873 } 7874 7875 return NO_ERROR; 7876} 7877 7878 7879// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7880sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7881 const sp<AudioFlinger::Client>& client, 7882 const sp<IEffectClient>& effectClient, 7883 int32_t priority, 7884 int sessionId, 7885 effect_descriptor_t *desc, 7886 int *enabled, 7887 status_t *status 7888 ) 7889{ 7890 sp<EffectModule> effect; 7891 sp<EffectHandle> handle; 7892 status_t lStatus; 7893 sp<EffectChain> chain; 7894 bool chainCreated = false; 7895 bool effectCreated = false; 7896 bool effectRegistered = false; 7897 7898 lStatus = initCheck(); 7899 if (lStatus != NO_ERROR) { 7900 ALOGW("createEffect_l() Audio driver not initialized."); 7901 goto Exit; 7902 } 7903 7904 // Do not allow effects with session ID 0 on direct output or duplicating threads 7905 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7906 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7907 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7908 desc->name, sessionId); 7909 lStatus = BAD_VALUE; 7910 goto Exit; 7911 } 7912 // Only Pre processor effects are allowed on input threads and only on input threads 7913 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7914 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7915 desc->name, desc->flags, mType); 7916 lStatus = BAD_VALUE; 7917 goto Exit; 7918 } 7919 7920 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7921 7922 { // scope for mLock 7923 Mutex::Autolock _l(mLock); 7924 7925 // check for existing effect chain with the requested audio session 7926 chain = getEffectChain_l(sessionId); 7927 if (chain == 0) { 7928 // create a new chain for this session 7929 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7930 chain = new EffectChain(this, sessionId); 7931 addEffectChain_l(chain); 7932 chain->setStrategy(getStrategyForSession_l(sessionId)); 7933 chainCreated = true; 7934 } else { 7935 effect = chain->getEffectFromDesc_l(desc); 7936 } 7937 7938 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7939 7940 if (effect == 0) { 7941 int id = mAudioFlinger->nextUniqueId(); 7942 // Check CPU and memory usage 7943 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7944 if (lStatus != NO_ERROR) { 7945 goto Exit; 7946 } 7947 effectRegistered = true; 7948 // create a new effect module if none present in the chain 7949 effect = new EffectModule(this, chain, desc, id, sessionId); 7950 lStatus = effect->status(); 7951 if (lStatus != NO_ERROR) { 7952 goto Exit; 7953 } 7954 lStatus = chain->addEffect_l(effect); 7955 if (lStatus != NO_ERROR) { 7956 goto Exit; 7957 } 7958 effectCreated = true; 7959 7960 effect->setDevice(mOutDevice); 7961 effect->setDevice(mInDevice); 7962 effect->setMode(mAudioFlinger->getMode()); 7963 effect->setAudioSource(mAudioSource); 7964 } 7965 // create effect handle and connect it to effect module 7966 handle = new EffectHandle(effect, client, effectClient, priority); 7967 lStatus = effect->addHandle(handle.get()); 7968 if (enabled != NULL) { 7969 *enabled = (int)effect->isEnabled(); 7970 } 7971 } 7972 7973Exit: 7974 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7975 Mutex::Autolock _l(mLock); 7976 if (effectCreated) { 7977 chain->removeEffect_l(effect); 7978 } 7979 if (effectRegistered) { 7980 AudioSystem::unregisterEffect(effect->id()); 7981 } 7982 if (chainCreated) { 7983 removeEffectChain_l(chain); 7984 } 7985 handle.clear(); 7986 } 7987 7988 if (status != NULL) { 7989 *status = lStatus; 7990 } 7991 return handle; 7992} 7993 7994sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7995{ 7996 Mutex::Autolock _l(mLock); 7997 return getEffect_l(sessionId, effectId); 7998} 7999 8000sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 8001{ 8002 sp<EffectChain> chain = getEffectChain_l(sessionId); 8003 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 8004} 8005 8006// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 8007// PlaybackThread::mLock held 8008status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 8009{ 8010 // check for existing effect chain with the requested audio session 8011 int sessionId = effect->sessionId(); 8012 sp<EffectChain> chain = getEffectChain_l(sessionId); 8013 bool chainCreated = false; 8014 8015 if (chain == 0) { 8016 // create a new chain for this session 8017 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 8018 chain = new EffectChain(this, sessionId); 8019 addEffectChain_l(chain); 8020 chain->setStrategy(getStrategyForSession_l(sessionId)); 8021 chainCreated = true; 8022 } 8023 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 8024 8025 if (chain->getEffectFromId_l(effect->id()) != 0) { 8026 ALOGW("addEffect_l() %p effect %s already present in chain %p", 8027 this, effect->desc().name, chain.get()); 8028 return BAD_VALUE; 8029 } 8030 8031 status_t status = chain->addEffect_l(effect); 8032 if (status != NO_ERROR) { 8033 if (chainCreated) { 8034 removeEffectChain_l(chain); 8035 } 8036 return status; 8037 } 8038 8039 effect->setDevice(mOutDevice); 8040 effect->setDevice(mInDevice); 8041 effect->setMode(mAudioFlinger->getMode()); 8042 effect->setAudioSource(mAudioSource); 8043 return NO_ERROR; 8044} 8045 8046void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8047 8048 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8049 effect_descriptor_t desc = effect->desc(); 8050 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8051 detachAuxEffect_l(effect->id()); 8052 } 8053 8054 sp<EffectChain> chain = effect->chain().promote(); 8055 if (chain != 0) { 8056 // remove effect chain if removing last effect 8057 if (chain->removeEffect_l(effect) == 0) { 8058 removeEffectChain_l(chain); 8059 } 8060 } else { 8061 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8062 } 8063} 8064 8065void AudioFlinger::ThreadBase::lockEffectChains_l( 8066 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8067{ 8068 effectChains = mEffectChains; 8069 for (size_t i = 0; i < mEffectChains.size(); i++) { 8070 mEffectChains[i]->lock(); 8071 } 8072} 8073 8074void AudioFlinger::ThreadBase::unlockEffectChains( 8075 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8076{ 8077 for (size_t i = 0; i < effectChains.size(); i++) { 8078 effectChains[i]->unlock(); 8079 } 8080} 8081 8082sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8083{ 8084 Mutex::Autolock _l(mLock); 8085 return getEffectChain_l(sessionId); 8086} 8087 8088sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8089{ 8090 size_t size = mEffectChains.size(); 8091 for (size_t i = 0; i < size; i++) { 8092 if (mEffectChains[i]->sessionId() == sessionId) { 8093 return mEffectChains[i]; 8094 } 8095 } 8096 return 0; 8097} 8098 8099void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8100{ 8101 Mutex::Autolock _l(mLock); 8102 size_t size = mEffectChains.size(); 8103 for (size_t i = 0; i < size; i++) { 8104 mEffectChains[i]->setMode_l(mode); 8105 } 8106} 8107 8108void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8109 EffectHandle *handle, 8110 bool unpinIfLast) { 8111 8112 Mutex::Autolock _l(mLock); 8113 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8114 // delete the effect module if removing last handle on it 8115 if (effect->removeHandle(handle) == 0) { 8116 if (!effect->isPinned() || unpinIfLast) { 8117 removeEffect_l(effect); 8118 AudioSystem::unregisterEffect(effect->id()); 8119 } 8120 } 8121} 8122 8123status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8124{ 8125 int session = chain->sessionId(); 8126 int16_t *buffer = mMixBuffer; 8127 bool ownsBuffer = false; 8128 8129 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8130 if (session > 0) { 8131 // Only one effect chain can be present in direct output thread and it uses 8132 // the mix buffer as input 8133 if (mType != DIRECT) { 8134 size_t numSamples = mNormalFrameCount * mChannelCount; 8135 buffer = new int16_t[numSamples]; 8136 memset(buffer, 0, numSamples * sizeof(int16_t)); 8137 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8138 ownsBuffer = true; 8139 } 8140 8141 // Attach all tracks with same session ID to this chain. 8142 for (size_t i = 0; i < mTracks.size(); ++i) { 8143 sp<Track> track = mTracks[i]; 8144 if (session == track->sessionId()) { 8145 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8146 buffer); 8147 track->setMainBuffer(buffer); 8148 chain->incTrackCnt(); 8149 } 8150 } 8151 8152 // indicate all active tracks in the chain 8153 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8154 sp<Track> track = mActiveTracks[i].promote(); 8155 if (track == 0) { 8156 continue; 8157 } 8158 if (session == track->sessionId()) { 8159 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8160 chain->incActiveTrackCnt(); 8161 } 8162 } 8163 } 8164 8165 chain->setInBuffer(buffer, ownsBuffer); 8166 chain->setOutBuffer(mMixBuffer); 8167 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8168 // chains list in order to be processed last as it contains output stage effects 8169 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8170 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8171 // after track specific effects and before output stage 8172 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8173 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8174 // Effect chain for other sessions are inserted at beginning of effect 8175 // chains list to be processed before output mix effects. Relative order between other 8176 // sessions is not important 8177 size_t size = mEffectChains.size(); 8178 size_t i = 0; 8179 for (i = 0; i < size; i++) { 8180 if (mEffectChains[i]->sessionId() < session) { 8181 break; 8182 } 8183 } 8184 mEffectChains.insertAt(chain, i); 8185 checkSuspendOnAddEffectChain_l(chain); 8186 8187 return NO_ERROR; 8188} 8189 8190size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8191{ 8192 int session = chain->sessionId(); 8193 8194 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8195 8196 for (size_t i = 0; i < mEffectChains.size(); i++) { 8197 if (chain == mEffectChains[i]) { 8198 mEffectChains.removeAt(i); 8199 // detach all active tracks from the chain 8200 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8201 sp<Track> track = mActiveTracks[i].promote(); 8202 if (track == 0) { 8203 continue; 8204 } 8205 if (session == track->sessionId()) { 8206 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8207 chain.get(), session); 8208 chain->decActiveTrackCnt(); 8209 } 8210 } 8211 8212 // detach all tracks with same session ID from this chain 8213 for (size_t i = 0; i < mTracks.size(); ++i) { 8214 sp<Track> track = mTracks[i]; 8215 if (session == track->sessionId()) { 8216 track->setMainBuffer(mMixBuffer); 8217 chain->decTrackCnt(); 8218 } 8219 } 8220 break; 8221 } 8222 } 8223 return mEffectChains.size(); 8224} 8225 8226status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8227 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8228{ 8229 Mutex::Autolock _l(mLock); 8230 return attachAuxEffect_l(track, EffectId); 8231} 8232 8233status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8234 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8235{ 8236 status_t status = NO_ERROR; 8237 8238 if (EffectId == 0) { 8239 track->setAuxBuffer(0, NULL); 8240 } else { 8241 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8242 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8243 if (effect != 0) { 8244 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8245 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8246 } else { 8247 status = INVALID_OPERATION; 8248 } 8249 } else { 8250 status = BAD_VALUE; 8251 } 8252 } 8253 return status; 8254} 8255 8256void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8257{ 8258 for (size_t i = 0; i < mTracks.size(); ++i) { 8259 sp<Track> track = mTracks[i]; 8260 if (track->auxEffectId() == effectId) { 8261 attachAuxEffect_l(track, 0); 8262 } 8263 } 8264} 8265 8266status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8267{ 8268 // only one chain per input thread 8269 if (mEffectChains.size() != 0) { 8270 return INVALID_OPERATION; 8271 } 8272 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8273 8274 chain->setInBuffer(NULL); 8275 chain->setOutBuffer(NULL); 8276 8277 checkSuspendOnAddEffectChain_l(chain); 8278 8279 mEffectChains.add(chain); 8280 8281 return NO_ERROR; 8282} 8283 8284size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8285{ 8286 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8287 ALOGW_IF(mEffectChains.size() != 1, 8288 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8289 chain.get(), mEffectChains.size(), this); 8290 if (mEffectChains.size() == 1) { 8291 mEffectChains.removeAt(0); 8292 } 8293 return 0; 8294} 8295 8296// ---------------------------------------------------------------------------- 8297// EffectModule implementation 8298// ---------------------------------------------------------------------------- 8299 8300#undef LOG_TAG 8301#define LOG_TAG "AudioFlinger::EffectModule" 8302 8303AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8304 const wp<AudioFlinger::EffectChain>& chain, 8305 effect_descriptor_t *desc, 8306 int id, 8307 int sessionId) 8308 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8309 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8310 mDescriptor(*desc), 8311 // mConfig is set by configure() and not used before then 8312 mEffectInterface(NULL), 8313 mStatus(NO_INIT), mState(IDLE), 8314 // mMaxDisableWaitCnt is set by configure() and not used before then 8315 // mDisableWaitCnt is set by process() and updateState() and not used before then 8316 mSuspended(false) 8317{ 8318 ALOGV("Constructor %p", this); 8319 int lStatus; 8320 8321 // create effect engine from effect factory 8322 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8323 8324 if (mStatus != NO_ERROR) { 8325 return; 8326 } 8327 lStatus = init(); 8328 if (lStatus < 0) { 8329 mStatus = lStatus; 8330 goto Error; 8331 } 8332 8333 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8334 return; 8335Error: 8336 EffectRelease(mEffectInterface); 8337 mEffectInterface = NULL; 8338 ALOGV("Constructor Error %d", mStatus); 8339} 8340 8341AudioFlinger::EffectModule::~EffectModule() 8342{ 8343 ALOGV("Destructor %p", this); 8344 if (mEffectInterface != NULL) { 8345 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8346 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8347 sp<ThreadBase> thread = mThread.promote(); 8348 if (thread != 0) { 8349 audio_stream_t *stream = thread->stream(); 8350 if (stream != NULL) { 8351 stream->remove_audio_effect(stream, mEffectInterface); 8352 } 8353 } 8354 } 8355 // release effect engine 8356 EffectRelease(mEffectInterface); 8357 } 8358} 8359 8360status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8361{ 8362 status_t status; 8363 8364 Mutex::Autolock _l(mLock); 8365 int priority = handle->priority(); 8366 size_t size = mHandles.size(); 8367 EffectHandle *controlHandle = NULL; 8368 size_t i; 8369 for (i = 0; i < size; i++) { 8370 EffectHandle *h = mHandles[i]; 8371 if (h == NULL || h->destroyed_l()) { 8372 continue; 8373 } 8374 // first non destroyed handle is considered in control 8375 if (controlHandle == NULL) 8376 controlHandle = h; 8377 if (h->priority() <= priority) { 8378 break; 8379 } 8380 } 8381 // if inserted in first place, move effect control from previous owner to this handle 8382 if (i == 0) { 8383 bool enabled = false; 8384 if (controlHandle != NULL) { 8385 enabled = controlHandle->enabled(); 8386 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8387 } 8388 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8389 status = NO_ERROR; 8390 } else { 8391 status = ALREADY_EXISTS; 8392 } 8393 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8394 mHandles.insertAt(handle, i); 8395 return status; 8396} 8397 8398size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8399{ 8400 Mutex::Autolock _l(mLock); 8401 size_t size = mHandles.size(); 8402 size_t i; 8403 for (i = 0; i < size; i++) { 8404 if (mHandles[i] == handle) { 8405 break; 8406 } 8407 } 8408 if (i == size) { 8409 return size; 8410 } 8411 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8412 8413 mHandles.removeAt(i); 8414 // if removed from first place, move effect control from this handle to next in line 8415 if (i == 0) { 8416 EffectHandle *h = controlHandle_l(); 8417 if (h != NULL) { 8418 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8419 } 8420 } 8421 8422 // Prevent calls to process() and other functions on effect interface from now on. 8423 // The effect engine will be released by the destructor when the last strong reference on 8424 // this object is released which can happen after next process is called. 8425 if (mHandles.size() == 0 && !mPinned) { 8426 mState = DESTROYED; 8427 } 8428 8429 return mHandles.size(); 8430} 8431 8432// must be called with EffectModule::mLock held 8433AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8434{ 8435 // the first valid handle in the list has control over the module 8436 for (size_t i = 0; i < mHandles.size(); i++) { 8437 EffectHandle *h = mHandles[i]; 8438 if (h != NULL && !h->destroyed_l()) { 8439 return h; 8440 } 8441 } 8442 8443 return NULL; 8444} 8445 8446size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8447{ 8448 ALOGV("disconnect() %p handle %p", this, handle); 8449 // keep a strong reference on this EffectModule to avoid calling the 8450 // destructor before we exit 8451 sp<EffectModule> keep(this); 8452 { 8453 sp<ThreadBase> thread = mThread.promote(); 8454 if (thread != 0) { 8455 thread->disconnectEffect(keep, handle, unpinIfLast); 8456 } 8457 } 8458 return mHandles.size(); 8459} 8460 8461void AudioFlinger::EffectModule::updateState() { 8462 Mutex::Autolock _l(mLock); 8463 8464 switch (mState) { 8465 case RESTART: 8466 reset_l(); 8467 // FALL THROUGH 8468 8469 case STARTING: 8470 // clear auxiliary effect input buffer for next accumulation 8471 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8472 memset(mConfig.inputCfg.buffer.raw, 8473 0, 8474 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8475 } 8476 start_l(); 8477 mState = ACTIVE; 8478 break; 8479 case STOPPING: 8480 stop_l(); 8481 mDisableWaitCnt = mMaxDisableWaitCnt; 8482 mState = STOPPED; 8483 break; 8484 case STOPPED: 8485 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8486 // turn off sequence. 8487 if (--mDisableWaitCnt == 0) { 8488 reset_l(); 8489 mState = IDLE; 8490 } 8491 break; 8492 default: //IDLE , ACTIVE, DESTROYED 8493 break; 8494 } 8495} 8496 8497void AudioFlinger::EffectModule::process() 8498{ 8499 Mutex::Autolock _l(mLock); 8500 8501 if (mState == DESTROYED || mEffectInterface == NULL || 8502 mConfig.inputCfg.buffer.raw == NULL || 8503 mConfig.outputCfg.buffer.raw == NULL) { 8504 return; 8505 } 8506 8507 if (isProcessEnabled()) { 8508 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8509 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8510 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8511 mConfig.inputCfg.buffer.s32, 8512 mConfig.inputCfg.buffer.frameCount/2); 8513 } 8514 8515 // do the actual processing in the effect engine 8516 int ret = (*mEffectInterface)->process(mEffectInterface, 8517 &mConfig.inputCfg.buffer, 8518 &mConfig.outputCfg.buffer); 8519 8520 // force transition to IDLE state when engine is ready 8521 if (mState == STOPPED && ret == -ENODATA) { 8522 mDisableWaitCnt = 1; 8523 } 8524 8525 // clear auxiliary effect input buffer for next accumulation 8526 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8527 memset(mConfig.inputCfg.buffer.raw, 0, 8528 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8529 } 8530 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8531 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8532 // If an insert effect is idle and input buffer is different from output buffer, 8533 // accumulate input onto output 8534 sp<EffectChain> chain = mChain.promote(); 8535 if (chain != 0 && chain->activeTrackCnt() != 0) { 8536 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8537 int16_t *in = mConfig.inputCfg.buffer.s16; 8538 int16_t *out = mConfig.outputCfg.buffer.s16; 8539 for (size_t i = 0; i < frameCnt; i++) { 8540 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8541 } 8542 } 8543 } 8544} 8545 8546void AudioFlinger::EffectModule::reset_l() 8547{ 8548 if (mEffectInterface == NULL) { 8549 return; 8550 } 8551 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8552} 8553 8554status_t AudioFlinger::EffectModule::configure() 8555{ 8556 if (mEffectInterface == NULL) { 8557 return NO_INIT; 8558 } 8559 8560 sp<ThreadBase> thread = mThread.promote(); 8561 if (thread == 0) { 8562 return DEAD_OBJECT; 8563 } 8564 8565 // TODO: handle configuration of effects replacing track process 8566 audio_channel_mask_t channelMask = thread->channelMask(); 8567 8568 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8569 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8570 } else { 8571 mConfig.inputCfg.channels = channelMask; 8572 } 8573 mConfig.outputCfg.channels = channelMask; 8574 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8575 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8576 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8577 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8578 mConfig.inputCfg.bufferProvider.cookie = NULL; 8579 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8580 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8581 mConfig.outputCfg.bufferProvider.cookie = NULL; 8582 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8583 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8584 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8585 // Insert effect: 8586 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8587 // always overwrites output buffer: input buffer == output buffer 8588 // - in other sessions: 8589 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8590 // other effect: overwrites output buffer: input buffer == output buffer 8591 // Auxiliary effect: 8592 // accumulates in output buffer: input buffer != output buffer 8593 // Therefore: accumulate <=> input buffer != output buffer 8594 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8595 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8596 } else { 8597 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8598 } 8599 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8600 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8601 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8602 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8603 8604 ALOGV("configure() %p thread %p buffer %p framecount %d", 8605 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8606 8607 status_t cmdStatus; 8608 uint32_t size = sizeof(int); 8609 status_t status = (*mEffectInterface)->command(mEffectInterface, 8610 EFFECT_CMD_SET_CONFIG, 8611 sizeof(effect_config_t), 8612 &mConfig, 8613 &size, 8614 &cmdStatus); 8615 if (status == 0) { 8616 status = cmdStatus; 8617 } 8618 8619 if (status == 0 && 8620 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8621 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8622 effect_param_t *p = (effect_param_t *)buf32; 8623 8624 p->psize = sizeof(uint32_t); 8625 p->vsize = sizeof(uint32_t); 8626 size = sizeof(int); 8627 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8628 8629 uint32_t latency = 0; 8630 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8631 if (pbt != NULL) { 8632 latency = pbt->latency_l(); 8633 } 8634 8635 *((int32_t *)p->data + 1)= latency; 8636 (*mEffectInterface)->command(mEffectInterface, 8637 EFFECT_CMD_SET_PARAM, 8638 sizeof(effect_param_t) + 8, 8639 &buf32, 8640 &size, 8641 &cmdStatus); 8642 } 8643 8644 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8645 (1000 * mConfig.outputCfg.buffer.frameCount); 8646 8647 return status; 8648} 8649 8650status_t AudioFlinger::EffectModule::init() 8651{ 8652 Mutex::Autolock _l(mLock); 8653 if (mEffectInterface == NULL) { 8654 return NO_INIT; 8655 } 8656 status_t cmdStatus; 8657 uint32_t size = sizeof(status_t); 8658 status_t status = (*mEffectInterface)->command(mEffectInterface, 8659 EFFECT_CMD_INIT, 8660 0, 8661 NULL, 8662 &size, 8663 &cmdStatus); 8664 if (status == 0) { 8665 status = cmdStatus; 8666 } 8667 return status; 8668} 8669 8670status_t AudioFlinger::EffectModule::start() 8671{ 8672 Mutex::Autolock _l(mLock); 8673 return start_l(); 8674} 8675 8676status_t AudioFlinger::EffectModule::start_l() 8677{ 8678 if (mEffectInterface == NULL) { 8679 return NO_INIT; 8680 } 8681 status_t cmdStatus; 8682 uint32_t size = sizeof(status_t); 8683 status_t status = (*mEffectInterface)->command(mEffectInterface, 8684 EFFECT_CMD_ENABLE, 8685 0, 8686 NULL, 8687 &size, 8688 &cmdStatus); 8689 if (status == 0) { 8690 status = cmdStatus; 8691 } 8692 if (status == 0 && 8693 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8694 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8695 sp<ThreadBase> thread = mThread.promote(); 8696 if (thread != 0) { 8697 audio_stream_t *stream = thread->stream(); 8698 if (stream != NULL) { 8699 stream->add_audio_effect(stream, mEffectInterface); 8700 } 8701 } 8702 } 8703 return status; 8704} 8705 8706status_t AudioFlinger::EffectModule::stop() 8707{ 8708 Mutex::Autolock _l(mLock); 8709 return stop_l(); 8710} 8711 8712status_t AudioFlinger::EffectModule::stop_l() 8713{ 8714 if (mEffectInterface == NULL) { 8715 return NO_INIT; 8716 } 8717 status_t cmdStatus; 8718 uint32_t size = sizeof(status_t); 8719 status_t status = (*mEffectInterface)->command(mEffectInterface, 8720 EFFECT_CMD_DISABLE, 8721 0, 8722 NULL, 8723 &size, 8724 &cmdStatus); 8725 if (status == 0) { 8726 status = cmdStatus; 8727 } 8728 if (status == 0 && 8729 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8730 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8731 sp<ThreadBase> thread = mThread.promote(); 8732 if (thread != 0) { 8733 audio_stream_t *stream = thread->stream(); 8734 if (stream != NULL) { 8735 stream->remove_audio_effect(stream, mEffectInterface); 8736 } 8737 } 8738 } 8739 return status; 8740} 8741 8742status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8743 uint32_t cmdSize, 8744 void *pCmdData, 8745 uint32_t *replySize, 8746 void *pReplyData) 8747{ 8748 Mutex::Autolock _l(mLock); 8749 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8750 8751 if (mState == DESTROYED || mEffectInterface == NULL) { 8752 return NO_INIT; 8753 } 8754 status_t status = (*mEffectInterface)->command(mEffectInterface, 8755 cmdCode, 8756 cmdSize, 8757 pCmdData, 8758 replySize, 8759 pReplyData); 8760 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8761 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8762 for (size_t i = 1; i < mHandles.size(); i++) { 8763 EffectHandle *h = mHandles[i]; 8764 if (h != NULL && !h->destroyed_l()) { 8765 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8766 } 8767 } 8768 } 8769 return status; 8770} 8771 8772status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8773{ 8774 Mutex::Autolock _l(mLock); 8775 return setEnabled_l(enabled); 8776} 8777 8778// must be called with EffectModule::mLock held 8779status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8780{ 8781 8782 ALOGV("setEnabled %p enabled %d", this, enabled); 8783 8784 if (enabled != isEnabled()) { 8785 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8786 if (enabled && status != NO_ERROR) { 8787 return status; 8788 } 8789 8790 switch (mState) { 8791 // going from disabled to enabled 8792 case IDLE: 8793 mState = STARTING; 8794 break; 8795 case STOPPED: 8796 mState = RESTART; 8797 break; 8798 case STOPPING: 8799 mState = ACTIVE; 8800 break; 8801 8802 // going from enabled to disabled 8803 case RESTART: 8804 mState = STOPPED; 8805 break; 8806 case STARTING: 8807 mState = IDLE; 8808 break; 8809 case ACTIVE: 8810 mState = STOPPING; 8811 break; 8812 case DESTROYED: 8813 return NO_ERROR; // simply ignore as we are being destroyed 8814 } 8815 for (size_t i = 1; i < mHandles.size(); i++) { 8816 EffectHandle *h = mHandles[i]; 8817 if (h != NULL && !h->destroyed_l()) { 8818 h->setEnabled(enabled); 8819 } 8820 } 8821 } 8822 return NO_ERROR; 8823} 8824 8825bool AudioFlinger::EffectModule::isEnabled() const 8826{ 8827 switch (mState) { 8828 case RESTART: 8829 case STARTING: 8830 case ACTIVE: 8831 return true; 8832 case IDLE: 8833 case STOPPING: 8834 case STOPPED: 8835 case DESTROYED: 8836 default: 8837 return false; 8838 } 8839} 8840 8841bool AudioFlinger::EffectModule::isProcessEnabled() const 8842{ 8843 switch (mState) { 8844 case RESTART: 8845 case ACTIVE: 8846 case STOPPING: 8847 case STOPPED: 8848 return true; 8849 case IDLE: 8850 case STARTING: 8851 case DESTROYED: 8852 default: 8853 return false; 8854 } 8855} 8856 8857status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8858{ 8859 Mutex::Autolock _l(mLock); 8860 status_t status = NO_ERROR; 8861 8862 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8863 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8864 if (isProcessEnabled() && 8865 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8866 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8867 status_t cmdStatus; 8868 uint32_t volume[2]; 8869 uint32_t *pVolume = NULL; 8870 uint32_t size = sizeof(volume); 8871 volume[0] = *left; 8872 volume[1] = *right; 8873 if (controller) { 8874 pVolume = volume; 8875 } 8876 status = (*mEffectInterface)->command(mEffectInterface, 8877 EFFECT_CMD_SET_VOLUME, 8878 size, 8879 volume, 8880 &size, 8881 pVolume); 8882 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8883 *left = volume[0]; 8884 *right = volume[1]; 8885 } 8886 } 8887 return status; 8888} 8889 8890status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8891{ 8892 if (device == AUDIO_DEVICE_NONE) { 8893 return NO_ERROR; 8894 } 8895 8896 Mutex::Autolock _l(mLock); 8897 status_t status = NO_ERROR; 8898 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8899 status_t cmdStatus; 8900 uint32_t size = sizeof(status_t); 8901 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8902 EFFECT_CMD_SET_INPUT_DEVICE; 8903 status = (*mEffectInterface)->command(mEffectInterface, 8904 cmd, 8905 sizeof(uint32_t), 8906 &device, 8907 &size, 8908 &cmdStatus); 8909 } 8910 return status; 8911} 8912 8913status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8914{ 8915 Mutex::Autolock _l(mLock); 8916 status_t status = NO_ERROR; 8917 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8918 status_t cmdStatus; 8919 uint32_t size = sizeof(status_t); 8920 status = (*mEffectInterface)->command(mEffectInterface, 8921 EFFECT_CMD_SET_AUDIO_MODE, 8922 sizeof(audio_mode_t), 8923 &mode, 8924 &size, 8925 &cmdStatus); 8926 if (status == NO_ERROR) { 8927 status = cmdStatus; 8928 } 8929 } 8930 return status; 8931} 8932 8933status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8934{ 8935 Mutex::Autolock _l(mLock); 8936 status_t status = NO_ERROR; 8937 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8938 uint32_t size = 0; 8939 status = (*mEffectInterface)->command(mEffectInterface, 8940 EFFECT_CMD_SET_AUDIO_SOURCE, 8941 sizeof(audio_source_t), 8942 &source, 8943 &size, 8944 NULL); 8945 } 8946 return status; 8947} 8948 8949void AudioFlinger::EffectModule::setSuspended(bool suspended) 8950{ 8951 Mutex::Autolock _l(mLock); 8952 mSuspended = suspended; 8953} 8954 8955bool AudioFlinger::EffectModule::suspended() const 8956{ 8957 Mutex::Autolock _l(mLock); 8958 return mSuspended; 8959} 8960 8961bool AudioFlinger::EffectModule::purgeHandles() 8962{ 8963 bool enabled = false; 8964 Mutex::Autolock _l(mLock); 8965 for (size_t i = 0; i < mHandles.size(); i++) { 8966 EffectHandle *handle = mHandles[i]; 8967 if (handle != NULL && !handle->destroyed_l()) { 8968 handle->effect().clear(); 8969 if (handle->hasControl()) { 8970 enabled = handle->enabled(); 8971 } 8972 } 8973 } 8974 return enabled; 8975} 8976 8977void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8978{ 8979 const size_t SIZE = 256; 8980 char buffer[SIZE]; 8981 String8 result; 8982 8983 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8984 result.append(buffer); 8985 8986 bool locked = tryLock(mLock); 8987 // failed to lock - AudioFlinger is probably deadlocked 8988 if (!locked) { 8989 result.append("\t\tCould not lock Fx mutex:\n"); 8990 } 8991 8992 result.append("\t\tSession Status State Engine:\n"); 8993 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8994 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8995 result.append(buffer); 8996 8997 result.append("\t\tDescriptor:\n"); 8998 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8999 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 9000 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 9001 mDescriptor.uuid.node[2], 9002 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 9003 result.append(buffer); 9004 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 9005 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 9006 mDescriptor.type.timeHiAndVersion, 9007 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 9008 mDescriptor.type.node[2], 9009 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 9010 result.append(buffer); 9011 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 9012 mDescriptor.apiVersion, 9013 mDescriptor.flags); 9014 result.append(buffer); 9015 snprintf(buffer, SIZE, "\t\t- name: %s\n", 9016 mDescriptor.name); 9017 result.append(buffer); 9018 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 9019 mDescriptor.implementor); 9020 result.append(buffer); 9021 9022 result.append("\t\t- Input configuration:\n"); 9023 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 9024 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 9025 (uint32_t)mConfig.inputCfg.buffer.raw, 9026 mConfig.inputCfg.buffer.frameCount, 9027 mConfig.inputCfg.samplingRate, 9028 mConfig.inputCfg.channels, 9029 mConfig.inputCfg.format); 9030 result.append(buffer); 9031 9032 result.append("\t\t- Output configuration:\n"); 9033 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 9034 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 9035 (uint32_t)mConfig.outputCfg.buffer.raw, 9036 mConfig.outputCfg.buffer.frameCount, 9037 mConfig.outputCfg.samplingRate, 9038 mConfig.outputCfg.channels, 9039 mConfig.outputCfg.format); 9040 result.append(buffer); 9041 9042 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 9043 result.append(buffer); 9044 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 9045 for (size_t i = 0; i < mHandles.size(); ++i) { 9046 EffectHandle *handle = mHandles[i]; 9047 if (handle != NULL && !handle->destroyed_l()) { 9048 handle->dump(buffer, SIZE); 9049 result.append(buffer); 9050 } 9051 } 9052 9053 result.append("\n"); 9054 9055 write(fd, result.string(), result.length()); 9056 9057 if (locked) { 9058 mLock.unlock(); 9059 } 9060} 9061 9062// ---------------------------------------------------------------------------- 9063// EffectHandle implementation 9064// ---------------------------------------------------------------------------- 9065 9066#undef LOG_TAG 9067#define LOG_TAG "AudioFlinger::EffectHandle" 9068 9069AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9070 const sp<AudioFlinger::Client>& client, 9071 const sp<IEffectClient>& effectClient, 9072 int32_t priority) 9073 : BnEffect(), 9074 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9075 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9076{ 9077 ALOGV("constructor %p", this); 9078 9079 if (client == 0) { 9080 return; 9081 } 9082 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9083 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9084 if (mCblkMemory != 0) { 9085 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9086 9087 if (mCblk != NULL) { 9088 new(mCblk) effect_param_cblk_t(); 9089 mBuffer = (uint8_t *)mCblk + bufOffset; 9090 } 9091 } else { 9092 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9093 sizeof(effect_param_cblk_t)); 9094 return; 9095 } 9096} 9097 9098AudioFlinger::EffectHandle::~EffectHandle() 9099{ 9100 ALOGV("Destructor %p", this); 9101 9102 if (mEffect == 0) { 9103 mDestroyed = true; 9104 return; 9105 } 9106 mEffect->lock(); 9107 mDestroyed = true; 9108 mEffect->unlock(); 9109 disconnect(false); 9110} 9111 9112status_t AudioFlinger::EffectHandle::enable() 9113{ 9114 ALOGV("enable %p", this); 9115 if (!mHasControl) { 9116 return INVALID_OPERATION; 9117 } 9118 if (mEffect == 0) { 9119 return DEAD_OBJECT; 9120 } 9121 9122 if (mEnabled) { 9123 return NO_ERROR; 9124 } 9125 9126 mEnabled = true; 9127 9128 sp<ThreadBase> thread = mEffect->thread().promote(); 9129 if (thread != 0) { 9130 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9131 } 9132 9133 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9134 if (mEffect->suspended()) { 9135 return NO_ERROR; 9136 } 9137 9138 status_t status = mEffect->setEnabled(true); 9139 if (status != NO_ERROR) { 9140 if (thread != 0) { 9141 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9142 } 9143 mEnabled = false; 9144 } 9145 return status; 9146} 9147 9148status_t AudioFlinger::EffectHandle::disable() 9149{ 9150 ALOGV("disable %p", this); 9151 if (!mHasControl) { 9152 return INVALID_OPERATION; 9153 } 9154 if (mEffect == 0) { 9155 return DEAD_OBJECT; 9156 } 9157 9158 if (!mEnabled) { 9159 return NO_ERROR; 9160 } 9161 mEnabled = false; 9162 9163 if (mEffect->suspended()) { 9164 return NO_ERROR; 9165 } 9166 9167 status_t status = mEffect->setEnabled(false); 9168 9169 sp<ThreadBase> thread = mEffect->thread().promote(); 9170 if (thread != 0) { 9171 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9172 } 9173 9174 return status; 9175} 9176 9177void AudioFlinger::EffectHandle::disconnect() 9178{ 9179 disconnect(true); 9180} 9181 9182void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9183{ 9184 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9185 if (mEffect == 0) { 9186 return; 9187 } 9188 // restore suspended effects if the disconnected handle was enabled and the last one. 9189 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9190 sp<ThreadBase> thread = mEffect->thread().promote(); 9191 if (thread != 0) { 9192 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9193 } 9194 } 9195 9196 // release sp on module => module destructor can be called now 9197 mEffect.clear(); 9198 if (mClient != 0) { 9199 if (mCblk != NULL) { 9200 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9201 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9202 } 9203 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9204 // Client destructor must run with AudioFlinger mutex locked 9205 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9206 mClient.clear(); 9207 } 9208} 9209 9210status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9211 uint32_t cmdSize, 9212 void *pCmdData, 9213 uint32_t *replySize, 9214 void *pReplyData) 9215{ 9216 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9217 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9218 9219 // only get parameter command is permitted for applications not controlling the effect 9220 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9221 return INVALID_OPERATION; 9222 } 9223 if (mEffect == 0) { 9224 return DEAD_OBJECT; 9225 } 9226 if (mClient == 0) { 9227 return INVALID_OPERATION; 9228 } 9229 9230 // handle commands that are not forwarded transparently to effect engine 9231 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9232 // No need to trylock() here as this function is executed in the binder thread serving a 9233 // particular client process: no risk to block the whole media server process or mixer 9234 // threads if we are stuck here 9235 Mutex::Autolock _l(mCblk->lock); 9236 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9237 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9238 mCblk->serverIndex = 0; 9239 mCblk->clientIndex = 0; 9240 return BAD_VALUE; 9241 } 9242 status_t status = NO_ERROR; 9243 while (mCblk->serverIndex < mCblk->clientIndex) { 9244 int reply; 9245 uint32_t rsize = sizeof(int); 9246 int *p = (int *)(mBuffer + mCblk->serverIndex); 9247 int size = *p++; 9248 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9249 ALOGW("command(): invalid parameter block size"); 9250 break; 9251 } 9252 effect_param_t *param = (effect_param_t *)p; 9253 if (param->psize == 0 || param->vsize == 0) { 9254 ALOGW("command(): null parameter or value size"); 9255 mCblk->serverIndex += size; 9256 continue; 9257 } 9258 uint32_t psize = sizeof(effect_param_t) + 9259 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9260 param->vsize; 9261 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9262 psize, 9263 p, 9264 &rsize, 9265 &reply); 9266 // stop at first error encountered 9267 if (ret != NO_ERROR) { 9268 status = ret; 9269 *(int *)pReplyData = reply; 9270 break; 9271 } else if (reply != NO_ERROR) { 9272 *(int *)pReplyData = reply; 9273 break; 9274 } 9275 mCblk->serverIndex += size; 9276 } 9277 mCblk->serverIndex = 0; 9278 mCblk->clientIndex = 0; 9279 return status; 9280 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9281 *(int *)pReplyData = NO_ERROR; 9282 return enable(); 9283 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9284 *(int *)pReplyData = NO_ERROR; 9285 return disable(); 9286 } 9287 9288 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9289} 9290 9291void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9292{ 9293 ALOGV("setControl %p control %d", this, hasControl); 9294 9295 mHasControl = hasControl; 9296 mEnabled = enabled; 9297 9298 if (signal && mEffectClient != 0) { 9299 mEffectClient->controlStatusChanged(hasControl); 9300 } 9301} 9302 9303void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9304 uint32_t cmdSize, 9305 void *pCmdData, 9306 uint32_t replySize, 9307 void *pReplyData) 9308{ 9309 if (mEffectClient != 0) { 9310 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9311 } 9312} 9313 9314 9315 9316void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9317{ 9318 if (mEffectClient != 0) { 9319 mEffectClient->enableStatusChanged(enabled); 9320 } 9321} 9322 9323status_t AudioFlinger::EffectHandle::onTransact( 9324 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9325{ 9326 return BnEffect::onTransact(code, data, reply, flags); 9327} 9328 9329 9330void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9331{ 9332 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9333 9334 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9335 (mClient == 0) ? getpid_cached : mClient->pid(), 9336 mPriority, 9337 mHasControl, 9338 !locked, 9339 mCblk ? mCblk->clientIndex : 0, 9340 mCblk ? mCblk->serverIndex : 0 9341 ); 9342 9343 if (locked) { 9344 mCblk->lock.unlock(); 9345 } 9346} 9347 9348#undef LOG_TAG 9349#define LOG_TAG "AudioFlinger::EffectChain" 9350 9351AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9352 int sessionId) 9353 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9354 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9355 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9356{ 9357 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9358 if (thread == NULL) { 9359 return; 9360 } 9361 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9362 thread->frameCount(); 9363} 9364 9365AudioFlinger::EffectChain::~EffectChain() 9366{ 9367 if (mOwnInBuffer) { 9368 delete mInBuffer; 9369 } 9370 9371} 9372 9373// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9374sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9375 effect_descriptor_t *descriptor) 9376{ 9377 size_t size = mEffects.size(); 9378 9379 for (size_t i = 0; i < size; i++) { 9380 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9381 return mEffects[i]; 9382 } 9383 } 9384 return 0; 9385} 9386 9387// getEffectFromId_l() must be called with ThreadBase::mLock held 9388sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9389{ 9390 size_t size = mEffects.size(); 9391 9392 for (size_t i = 0; i < size; i++) { 9393 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9394 if (id == 0 || mEffects[i]->id() == id) { 9395 return mEffects[i]; 9396 } 9397 } 9398 return 0; 9399} 9400 9401// getEffectFromType_l() must be called with ThreadBase::mLock held 9402sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9403 const effect_uuid_t *type) 9404{ 9405 size_t size = mEffects.size(); 9406 9407 for (size_t i = 0; i < size; i++) { 9408 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9409 return mEffects[i]; 9410 } 9411 } 9412 return 0; 9413} 9414 9415void AudioFlinger::EffectChain::clearInputBuffer() 9416{ 9417 Mutex::Autolock _l(mLock); 9418 sp<ThreadBase> thread = mThread.promote(); 9419 if (thread == 0) { 9420 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9421 return; 9422 } 9423 clearInputBuffer_l(thread); 9424} 9425 9426// Must be called with EffectChain::mLock locked 9427void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9428{ 9429 size_t numSamples = thread->frameCount() * thread->channelCount(); 9430 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9431 9432} 9433 9434// Must be called with EffectChain::mLock locked 9435void AudioFlinger::EffectChain::process_l() 9436{ 9437 sp<ThreadBase> thread = mThread.promote(); 9438 if (thread == 0) { 9439 ALOGW("process_l(): cannot promote mixer thread"); 9440 return; 9441 } 9442 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9443 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9444 // always process effects unless no more tracks are on the session and the effect tail 9445 // has been rendered 9446 bool doProcess = true; 9447 if (!isGlobalSession) { 9448 bool tracksOnSession = (trackCnt() != 0); 9449 9450 if (!tracksOnSession && mTailBufferCount == 0) { 9451 doProcess = false; 9452 } 9453 9454 if (activeTrackCnt() == 0) { 9455 // if no track is active and the effect tail has not been rendered, 9456 // the input buffer must be cleared here as the mixer process will not do it 9457 if (tracksOnSession || mTailBufferCount > 0) { 9458 clearInputBuffer_l(thread); 9459 if (mTailBufferCount > 0) { 9460 mTailBufferCount--; 9461 } 9462 } 9463 } 9464 } 9465 9466 size_t size = mEffects.size(); 9467 if (doProcess) { 9468 for (size_t i = 0; i < size; i++) { 9469 mEffects[i]->process(); 9470 } 9471 } 9472 for (size_t i = 0; i < size; i++) { 9473 mEffects[i]->updateState(); 9474 } 9475} 9476 9477// addEffect_l() must be called with PlaybackThread::mLock held 9478status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9479{ 9480 effect_descriptor_t desc = effect->desc(); 9481 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9482 9483 Mutex::Autolock _l(mLock); 9484 effect->setChain(this); 9485 sp<ThreadBase> thread = mThread.promote(); 9486 if (thread == 0) { 9487 return NO_INIT; 9488 } 9489 effect->setThread(thread); 9490 9491 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9492 // Auxiliary effects are inserted at the beginning of mEffects vector as 9493 // they are processed first and accumulated in chain input buffer 9494 mEffects.insertAt(effect, 0); 9495 9496 // the input buffer for auxiliary effect contains mono samples in 9497 // 32 bit format. This is to avoid saturation in AudoMixer 9498 // accumulation stage. Saturation is done in EffectModule::process() before 9499 // calling the process in effect engine 9500 size_t numSamples = thread->frameCount(); 9501 int32_t *buffer = new int32_t[numSamples]; 9502 memset(buffer, 0, numSamples * sizeof(int32_t)); 9503 effect->setInBuffer((int16_t *)buffer); 9504 // auxiliary effects output samples to chain input buffer for further processing 9505 // by insert effects 9506 effect->setOutBuffer(mInBuffer); 9507 } else { 9508 // Insert effects are inserted at the end of mEffects vector as they are processed 9509 // after track and auxiliary effects. 9510 // Insert effect order as a function of indicated preference: 9511 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9512 // another effect is present 9513 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9514 // last effect claiming first position 9515 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9516 // first effect claiming last position 9517 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9518 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9519 // already present 9520 9521 size_t size = mEffects.size(); 9522 size_t idx_insert = size; 9523 ssize_t idx_insert_first = -1; 9524 ssize_t idx_insert_last = -1; 9525 9526 for (size_t i = 0; i < size; i++) { 9527 effect_descriptor_t d = mEffects[i]->desc(); 9528 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9529 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9530 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9531 // check invalid effect chaining combinations 9532 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9533 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9534 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9535 desc.name, d.name); 9536 return INVALID_OPERATION; 9537 } 9538 // remember position of first insert effect and by default 9539 // select this as insert position for new effect 9540 if (idx_insert == size) { 9541 idx_insert = i; 9542 } 9543 // remember position of last insert effect claiming 9544 // first position 9545 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9546 idx_insert_first = i; 9547 } 9548 // remember position of first insert effect claiming 9549 // last position 9550 if (iPref == EFFECT_FLAG_INSERT_LAST && 9551 idx_insert_last == -1) { 9552 idx_insert_last = i; 9553 } 9554 } 9555 } 9556 9557 // modify idx_insert from first position if needed 9558 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9559 if (idx_insert_last != -1) { 9560 idx_insert = idx_insert_last; 9561 } else { 9562 idx_insert = size; 9563 } 9564 } else { 9565 if (idx_insert_first != -1) { 9566 idx_insert = idx_insert_first + 1; 9567 } 9568 } 9569 9570 // always read samples from chain input buffer 9571 effect->setInBuffer(mInBuffer); 9572 9573 // if last effect in the chain, output samples to chain 9574 // output buffer, otherwise to chain input buffer 9575 if (idx_insert == size) { 9576 if (idx_insert != 0) { 9577 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9578 mEffects[idx_insert-1]->configure(); 9579 } 9580 effect->setOutBuffer(mOutBuffer); 9581 } else { 9582 effect->setOutBuffer(mInBuffer); 9583 } 9584 mEffects.insertAt(effect, idx_insert); 9585 9586 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9587 idx_insert); 9588 } 9589 effect->configure(); 9590 return NO_ERROR; 9591} 9592 9593// removeEffect_l() must be called with PlaybackThread::mLock held 9594size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9595{ 9596 Mutex::Autolock _l(mLock); 9597 size_t size = mEffects.size(); 9598 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9599 9600 for (size_t i = 0; i < size; i++) { 9601 if (effect == mEffects[i]) { 9602 // calling stop here will remove pre-processing effect from the audio HAL. 9603 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9604 // the middle of a read from audio HAL 9605 if (mEffects[i]->state() == EffectModule::ACTIVE || 9606 mEffects[i]->state() == EffectModule::STOPPING) { 9607 mEffects[i]->stop(); 9608 } 9609 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9610 delete[] effect->inBuffer(); 9611 } else { 9612 if (i == size - 1 && i != 0) { 9613 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9614 mEffects[i - 1]->configure(); 9615 } 9616 } 9617 mEffects.removeAt(i); 9618 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9619 this, i); 9620 break; 9621 } 9622 } 9623 9624 return mEffects.size(); 9625} 9626 9627// setDevice_l() must be called with PlaybackThread::mLock held 9628void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9629{ 9630 size_t size = mEffects.size(); 9631 for (size_t i = 0; i < size; i++) { 9632 mEffects[i]->setDevice(device); 9633 } 9634} 9635 9636// setMode_l() must be called with PlaybackThread::mLock held 9637void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9638{ 9639 size_t size = mEffects.size(); 9640 for (size_t i = 0; i < size; i++) { 9641 mEffects[i]->setMode(mode); 9642 } 9643} 9644 9645// setAudioSource_l() must be called with PlaybackThread::mLock held 9646void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9647{ 9648 size_t size = mEffects.size(); 9649 for (size_t i = 0; i < size; i++) { 9650 mEffects[i]->setAudioSource(source); 9651 } 9652} 9653 9654// setVolume_l() must be called with PlaybackThread::mLock held 9655bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9656{ 9657 uint32_t newLeft = *left; 9658 uint32_t newRight = *right; 9659 bool hasControl = false; 9660 int ctrlIdx = -1; 9661 size_t size = mEffects.size(); 9662 9663 // first update volume controller 9664 for (size_t i = size; i > 0; i--) { 9665 if (mEffects[i - 1]->isProcessEnabled() && 9666 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9667 ctrlIdx = i - 1; 9668 hasControl = true; 9669 break; 9670 } 9671 } 9672 9673 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9674 if (hasControl) { 9675 *left = mNewLeftVolume; 9676 *right = mNewRightVolume; 9677 } 9678 return hasControl; 9679 } 9680 9681 mVolumeCtrlIdx = ctrlIdx; 9682 mLeftVolume = newLeft; 9683 mRightVolume = newRight; 9684 9685 // second get volume update from volume controller 9686 if (ctrlIdx >= 0) { 9687 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9688 mNewLeftVolume = newLeft; 9689 mNewRightVolume = newRight; 9690 } 9691 // then indicate volume to all other effects in chain. 9692 // Pass altered volume to effects before volume controller 9693 // and requested volume to effects after controller 9694 uint32_t lVol = newLeft; 9695 uint32_t rVol = newRight; 9696 9697 for (size_t i = 0; i < size; i++) { 9698 if ((int)i == ctrlIdx) { 9699 continue; 9700 } 9701 // this also works for ctrlIdx == -1 when there is no volume controller 9702 if ((int)i > ctrlIdx) { 9703 lVol = *left; 9704 rVol = *right; 9705 } 9706 mEffects[i]->setVolume(&lVol, &rVol, false); 9707 } 9708 *left = newLeft; 9709 *right = newRight; 9710 9711 return hasControl; 9712} 9713 9714void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9715{ 9716 const size_t SIZE = 256; 9717 char buffer[SIZE]; 9718 String8 result; 9719 9720 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9721 result.append(buffer); 9722 9723 bool locked = tryLock(mLock); 9724 // failed to lock - AudioFlinger is probably deadlocked 9725 if (!locked) { 9726 result.append("\tCould not lock mutex:\n"); 9727 } 9728 9729 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9730 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9731 mEffects.size(), 9732 (uint32_t)mInBuffer, 9733 (uint32_t)mOutBuffer, 9734 mActiveTrackCnt); 9735 result.append(buffer); 9736 write(fd, result.string(), result.size()); 9737 9738 for (size_t i = 0; i < mEffects.size(); ++i) { 9739 sp<EffectModule> effect = mEffects[i]; 9740 if (effect != 0) { 9741 effect->dump(fd, args); 9742 } 9743 } 9744 9745 if (locked) { 9746 mLock.unlock(); 9747 } 9748} 9749 9750// must be called with ThreadBase::mLock held 9751void AudioFlinger::EffectChain::setEffectSuspended_l( 9752 const effect_uuid_t *type, bool suspend) 9753{ 9754 sp<SuspendedEffectDesc> desc; 9755 // use effect type UUID timelow as key as there is no real risk of identical 9756 // timeLow fields among effect type UUIDs. 9757 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9758 if (suspend) { 9759 if (index >= 0) { 9760 desc = mSuspendedEffects.valueAt(index); 9761 } else { 9762 desc = new SuspendedEffectDesc(); 9763 desc->mType = *type; 9764 mSuspendedEffects.add(type->timeLow, desc); 9765 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9766 } 9767 if (desc->mRefCount++ == 0) { 9768 sp<EffectModule> effect = getEffectIfEnabled(type); 9769 if (effect != 0) { 9770 desc->mEffect = effect; 9771 effect->setSuspended(true); 9772 effect->setEnabled(false); 9773 } 9774 } 9775 } else { 9776 if (index < 0) { 9777 return; 9778 } 9779 desc = mSuspendedEffects.valueAt(index); 9780 if (desc->mRefCount <= 0) { 9781 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9782 desc->mRefCount = 1; 9783 } 9784 if (--desc->mRefCount == 0) { 9785 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9786 if (desc->mEffect != 0) { 9787 sp<EffectModule> effect = desc->mEffect.promote(); 9788 if (effect != 0) { 9789 effect->setSuspended(false); 9790 effect->lock(); 9791 EffectHandle *handle = effect->controlHandle_l(); 9792 if (handle != NULL && !handle->destroyed_l()) { 9793 effect->setEnabled_l(handle->enabled()); 9794 } 9795 effect->unlock(); 9796 } 9797 desc->mEffect.clear(); 9798 } 9799 mSuspendedEffects.removeItemsAt(index); 9800 } 9801 } 9802} 9803 9804// must be called with ThreadBase::mLock held 9805void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9806{ 9807 sp<SuspendedEffectDesc> desc; 9808 9809 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9810 if (suspend) { 9811 if (index >= 0) { 9812 desc = mSuspendedEffects.valueAt(index); 9813 } else { 9814 desc = new SuspendedEffectDesc(); 9815 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9816 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9817 } 9818 if (desc->mRefCount++ == 0) { 9819 Vector< sp<EffectModule> > effects; 9820 getSuspendEligibleEffects(effects); 9821 for (size_t i = 0; i < effects.size(); i++) { 9822 setEffectSuspended_l(&effects[i]->desc().type, true); 9823 } 9824 } 9825 } else { 9826 if (index < 0) { 9827 return; 9828 } 9829 desc = mSuspendedEffects.valueAt(index); 9830 if (desc->mRefCount <= 0) { 9831 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9832 desc->mRefCount = 1; 9833 } 9834 if (--desc->mRefCount == 0) { 9835 Vector<const effect_uuid_t *> types; 9836 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9837 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9838 continue; 9839 } 9840 types.add(&mSuspendedEffects.valueAt(i)->mType); 9841 } 9842 for (size_t i = 0; i < types.size(); i++) { 9843 setEffectSuspended_l(types[i], false); 9844 } 9845 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9846 mSuspendedEffects.keyAt(index)); 9847 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9848 } 9849 } 9850} 9851 9852 9853// The volume effect is used for automated tests only 9854#ifndef OPENSL_ES_H_ 9855static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9856 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9857const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9858#endif //OPENSL_ES_H_ 9859 9860bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9861{ 9862 // auxiliary effects and visualizer are never suspended on output mix 9863 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9864 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9865 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9866 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9867 return false; 9868 } 9869 return true; 9870} 9871 9872void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9873 Vector< sp<AudioFlinger::EffectModule> > &effects) 9874{ 9875 effects.clear(); 9876 for (size_t i = 0; i < mEffects.size(); i++) { 9877 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9878 effects.add(mEffects[i]); 9879 } 9880 } 9881} 9882 9883sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9884 const effect_uuid_t *type) 9885{ 9886 sp<EffectModule> effect = getEffectFromType_l(type); 9887 return effect != 0 && effect->isEnabled() ? effect : 0; 9888} 9889 9890void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9891 bool enabled) 9892{ 9893 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9894 if (enabled) { 9895 if (index < 0) { 9896 // if the effect is not suspend check if all effects are suspended 9897 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9898 if (index < 0) { 9899 return; 9900 } 9901 if (!isEffectEligibleForSuspend(effect->desc())) { 9902 return; 9903 } 9904 setEffectSuspended_l(&effect->desc().type, enabled); 9905 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9906 if (index < 0) { 9907 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9908 return; 9909 } 9910 } 9911 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9912 effect->desc().type.timeLow); 9913 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9914 // if effect is requested to suspended but was not yet enabled, supend it now. 9915 if (desc->mEffect == 0) { 9916 desc->mEffect = effect; 9917 effect->setEnabled(false); 9918 effect->setSuspended(true); 9919 } 9920 } else { 9921 if (index < 0) { 9922 return; 9923 } 9924 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9925 effect->desc().type.timeLow); 9926 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9927 desc->mEffect.clear(); 9928 effect->setSuspended(false); 9929 } 9930} 9931 9932#undef LOG_TAG 9933#define LOG_TAG "AudioFlinger" 9934 9935// ---------------------------------------------------------------------------- 9936 9937status_t AudioFlinger::onTransact( 9938 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9939{ 9940 return BnAudioFlinger::onTransact(code, data, reply, flags); 9941} 9942 9943}; // namespace android 9944