AudioFlinger.cpp revision e213c86d36414a8fc75e37c52999522fe09c7328
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 <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75#include "FastMixer.h" 76 77// NBAIO implementations 78#include "AudioStreamOutSink.h" 79#include "MonoPipe.h" 80#include "MonoPipeReader.h" 81#include "SourceAudioBufferProvider.h" 82 83#ifdef HAVE_REQUEST_PRIORITY 84#include "SchedulingPolicyService.h" 85#endif 86 87#ifdef SOAKER 88#include "Soaker.h" 89#endif 90 91// ---------------------------------------------------------------------------- 92 93// Note: the following macro is used for extremely verbose logging message. In 94// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 95// 0; but one side effect of this is to turn all LOGV's as well. Some messages 96// are so verbose that we want to suppress them even when we have ALOG_ASSERT 97// turned on. Do not uncomment the #def below unless you really know what you 98// are doing and want to see all of the extremely verbose messages. 99//#define VERY_VERY_VERBOSE_LOGGING 100#ifdef VERY_VERY_VERBOSE_LOGGING 101#define ALOGVV ALOGV 102#else 103#define ALOGVV(a...) do { } while(0) 104#endif 105 106namespace android { 107 108static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 109static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 110 111static const float MAX_GAIN = 4096.0f; 112static const uint32_t MAX_GAIN_INT = 0x1000; 113 114// retry counts for buffer fill timeout 115// 50 * ~20msecs = 1 second 116static const int8_t kMaxTrackRetries = 50; 117static const int8_t kMaxTrackStartupRetries = 50; 118// allow less retry attempts on direct output thread. 119// direct outputs can be a scarce resource in audio hardware and should 120// be released as quickly as possible. 121static const int8_t kMaxTrackRetriesDirect = 2; 122 123static const int kDumpLockRetries = 50; 124static const int kDumpLockSleepUs = 20000; 125 126// don't warn about blocked writes or record buffer overflows more often than this 127static const nsecs_t kWarningThrottleNs = seconds(5); 128 129// RecordThread loop sleep time upon application overrun or audio HAL read error 130static const int kRecordThreadSleepUs = 5000; 131 132// maximum time to wait for setParameters to complete 133static const nsecs_t kSetParametersTimeoutNs = seconds(2); 134 135// minimum sleep time for the mixer thread loop when tracks are active but in underrun 136static const uint32_t kMinThreadSleepTimeUs = 5000; 137// maximum divider applied to the active sleep time in the mixer thread loop 138static const uint32_t kMaxThreadSleepTimeShift = 2; 139 140// minimum normal mix buffer size, expressed in milliseconds rather than frames 141static const uint32_t kMinNormalMixBufferSizeMs = 20; 142 143nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 144 145// Whether to use fast mixer 146static const enum { 147 FastMixer_Never, // never initialize or use: for debugging only 148 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 149 // normal mixer multiplier is 1 150 FastMixer_Static, // initialize if needed, then use all the time if initialized, 151 // multipler is calculated based on minimum normal mixer buffer size 152 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 153 // multipler is calculated based on minimum normal mixer buffer size 154 // FIXME for FastMixer_Dynamic: 155 // Supporting this option will require fixing HALs that can't handle large writes. 156 // For example, one HAL implementation returns an error from a large write, 157 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 158 // We could either fix the HAL implementations, or provide a wrapper that breaks 159 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 160} kUseFastMixer = FastMixer_Static; 161 162// ---------------------------------------------------------------------------- 163 164#ifdef ADD_BATTERY_DATA 165// To collect the amplifier usage 166static void addBatteryData(uint32_t params) { 167 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 168 if (service == NULL) { 169 // it already logged 170 return; 171 } 172 173 service->addBatteryData(params); 174} 175#endif 176 177static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 178{ 179 const hw_module_t *mod; 180 int rc; 181 182 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 183 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 184 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 185 if (rc) { 186 goto out; 187 } 188 rc = audio_hw_device_open(mod, dev); 189 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 195 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 196 rc = BAD_VALUE; 197 goto out; 198 } 199 return 0; 200 201out: 202 *dev = NULL; 203 return rc; 204} 205 206// ---------------------------------------------------------------------------- 207 208AudioFlinger::AudioFlinger() 209 : BnAudioFlinger(), 210 mPrimaryHardwareDev(NULL), 211 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 212 mMasterVolume(1.0f), 213 mMasterVolumeSupportLvl(MVS_NONE), 214 mMasterMute(false), 215 mNextUniqueId(1), 216 mMode(AUDIO_MODE_INVALID), 217 mBtNrecIsOff(false) 218{ 219} 220 221void AudioFlinger::onFirstRef() 222{ 223 int rc = 0; 224 225 Mutex::Autolock _l(mLock); 226 227 /* TODO: move all this work into an Init() function */ 228 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 229 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 230 uint32_t int_val; 231 if (1 == sscanf(val_str, "%u", &int_val)) { 232 mStandbyTimeInNsecs = milliseconds(int_val); 233 ALOGI("Using %u mSec as standby time.", int_val); 234 } else { 235 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 236 ALOGI("Using default %u mSec as standby time.", 237 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 238 } 239 } 240 241 mMode = AUDIO_MODE_NORMAL; 242 mMasterVolumeSW = 1.0; 243 mMasterVolume = 1.0; 244 mHardwareStatus = AUDIO_HW_IDLE; 245} 246 247AudioFlinger::~AudioFlinger() 248{ 249 250 while (!mRecordThreads.isEmpty()) { 251 // closeInput() will remove first entry from mRecordThreads 252 closeInput(mRecordThreads.keyAt(0)); 253 } 254 while (!mPlaybackThreads.isEmpty()) { 255 // closeOutput() will remove first entry from mPlaybackThreads 256 closeOutput(mPlaybackThreads.keyAt(0)); 257 } 258 259 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 260 // no mHardwareLock needed, as there are no other references to this 261 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 262 delete mAudioHwDevs.valueAt(i); 263 } 264} 265 266static const char * const audio_interfaces[] = { 267 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 268 AUDIO_HARDWARE_MODULE_ID_A2DP, 269 AUDIO_HARDWARE_MODULE_ID_USB, 270}; 271#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 272 273audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 274{ 275 // if module is 0, the request comes from an old policy manager and we should load 276 // well known modules 277 if (module == 0) { 278 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 279 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 280 loadHwModule_l(audio_interfaces[i]); 281 } 282 } else { 283 // check a match for the requested module handle 284 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 285 if (audioHwdevice != NULL) { 286 return audioHwdevice->hwDevice(); 287 } 288 } 289 // then try to find a module supporting the requested device. 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 292 if ((dev->get_supported_devices(dev) & devices) == devices) 293 return dev; 294 } 295 296 return NULL; 297} 298 299status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 305 result.append("Clients:\n"); 306 for (size_t i = 0; i < mClients.size(); ++i) { 307 sp<Client> client = mClients.valueAt(i).promote(); 308 if (client != 0) { 309 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 310 result.append(buffer); 311 } 312 } 313 314 result.append("Global session refs:\n"); 315 result.append(" session pid count\n"); 316 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 317 AudioSessionRef *r = mAudioSessionRefs[i]; 318 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 319 result.append(buffer); 320 } 321 write(fd, result.string(), result.size()); 322 return NO_ERROR; 323} 324 325 326status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 327{ 328 const size_t SIZE = 256; 329 char buffer[SIZE]; 330 String8 result; 331 hardware_call_state hardwareStatus = mHardwareStatus; 332 333 snprintf(buffer, SIZE, "Hardware status: %d\n" 334 "Standby Time mSec: %u\n", 335 hardwareStatus, 336 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 337 result.append(buffer); 338 write(fd, result.string(), result.size()); 339 return NO_ERROR; 340} 341 342status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353 return NO_ERROR; 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 uint32_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 473 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 474 if (mPlaybackThreads.keyAt(i) != output) { 475 // prevent same audio session on different output threads 476 uint32_t sessions = t->hasAudioSession(*sessionId); 477 if (sessions & PlaybackThread::TRACK_SESSION) { 478 ALOGE("createTrack() session ID %d already in use", *sessionId); 479 lStatus = BAD_VALUE; 480 goto Exit; 481 } 482 // check if an effect with same session ID is waiting for a track to be created 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 } 486 } 487 } 488 lSessionId = *sessionId; 489 } else { 490 // if no audio session id is provided, create one here 491 lSessionId = nextUniqueId(); 492 if (sessionId != NULL) { 493 *sessionId = lSessionId; 494 } 495 } 496 ALOGV("createTrack() lSessionId: %d", lSessionId); 497 498 track = thread->createTrack_l(client, streamType, sampleRate, format, 499 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 500 501 // move effect chain to this output thread if an effect on same session was waiting 502 // for a track to be created 503 if (lStatus == NO_ERROR && effectThread != NULL) { 504 Mutex::Autolock _dl(thread->mLock); 505 Mutex::Autolock _sl(effectThread->mLock); 506 moveEffectChain_l(lSessionId, effectThread, thread, true); 507 } 508 509 // Look for sync events awaiting for a session to be used. 510 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 511 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 512 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 513 track->setSyncEvent(mPendingSyncEvents[i]); 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 RecordThread::RecordTrack *track = thread->track(); 872 if (track != NULL) { 873 audio_devices_t device = (audio_devices_t)( 874 thread->device() & AUDIO_DEVICE_IN_ALL); 875 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 876 thread->setEffectSuspended(FX_IID_AEC, 877 suspend, 878 track->sessionId()); 879 thread->setEffectSuspended(FX_IID_NS, 880 suspend, 881 track->sessionId()); 882 } 883 } 884 mBtNrecIsOff = btNrecIsOff; 885 } 886 } 887 return final_result; 888 } 889 890 // hold a strong ref on thread in case closeOutput() or closeInput() is called 891 // and the thread is exited once the lock is released 892 sp<ThreadBase> thread; 893 { 894 Mutex::Autolock _l(mLock); 895 thread = checkPlaybackThread_l(ioHandle); 896 if (thread == NULL) { 897 thread = checkRecordThread_l(ioHandle); 898 } else if (thread == primaryPlaybackThread_l()) { 899 // indicate output device change to all input threads for pre processing 900 AudioParameter param = AudioParameter(keyValuePairs); 901 int value; 902 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 903 (value != 0)) { 904 for (size_t i = 0; i < mRecordThreads.size(); i++) { 905 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 906 } 907 } 908 } 909 } 910 if (thread != 0) { 911 return thread->setParameters(keyValuePairs); 912 } 913 return BAD_VALUE; 914} 915 916String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 917{ 918// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 919// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 920 921 Mutex::Autolock _l(mLock); 922 923 if (ioHandle == 0) { 924 String8 out_s8; 925 926 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 927 char *s; 928 { 929 AutoMutex lock(mHardwareLock); 930 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 931 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 932 s = dev->get_parameters(dev, keys.string()); 933 mHardwareStatus = AUDIO_HW_IDLE; 934 } 935 out_s8 += String8(s ? s : ""); 936 free(s); 937 } 938 return out_s8; 939 } 940 941 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 942 if (playbackThread != NULL) { 943 return playbackThread->getParameters(keys); 944 } 945 RecordThread *recordThread = checkRecordThread_l(ioHandle); 946 if (recordThread != NULL) { 947 return recordThread->getParameters(keys); 948 } 949 return String8(""); 950} 951 952size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 953{ 954 status_t ret = initCheck(); 955 if (ret != NO_ERROR) { 956 return 0; 957 } 958 959 AutoMutex lock(mHardwareLock); 960 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 961 struct audio_config config = { 962 sample_rate: sampleRate, 963 channel_mask: audio_channel_in_mask_from_count(channelCount), 964 format: format, 965 }; 966 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 967 mHardwareStatus = AUDIO_HW_IDLE; 968 return size; 969} 970 971unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 972{ 973 if (ioHandle == 0) { 974 return 0; 975 } 976 977 Mutex::Autolock _l(mLock); 978 979 RecordThread *recordThread = checkRecordThread_l(ioHandle); 980 if (recordThread != NULL) { 981 return recordThread->getInputFramesLost(); 982 } 983 return 0; 984} 985 986status_t AudioFlinger::setVoiceVolume(float value) 987{ 988 status_t ret = initCheck(); 989 if (ret != NO_ERROR) { 990 return ret; 991 } 992 993 // check calling permissions 994 if (!settingsAllowed()) { 995 return PERMISSION_DENIED; 996 } 997 998 AutoMutex lock(mHardwareLock); 999 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1000 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 1003 return ret; 1004} 1005 1006status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1007 audio_io_handle_t output) const 1008{ 1009 status_t status; 1010 1011 Mutex::Autolock _l(mLock); 1012 1013 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1014 if (playbackThread != NULL) { 1015 return playbackThread->getRenderPosition(halFrames, dspFrames); 1016 } 1017 1018 return BAD_VALUE; 1019} 1020 1021void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1022{ 1023 1024 Mutex::Autolock _l(mLock); 1025 1026 pid_t pid = IPCThreadState::self()->getCallingPid(); 1027 if (mNotificationClients.indexOfKey(pid) < 0) { 1028 sp<NotificationClient> notificationClient = new NotificationClient(this, 1029 client, 1030 pid); 1031 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1032 1033 mNotificationClients.add(pid, notificationClient); 1034 1035 sp<IBinder> binder = client->asBinder(); 1036 binder->linkToDeath(notificationClient); 1037 1038 // the config change is always sent from playback or record threads to avoid deadlock 1039 // with AudioSystem::gLock 1040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1041 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1042 } 1043 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1046 } 1047 } 1048} 1049 1050void AudioFlinger::removeNotificationClient(pid_t pid) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 1054 mNotificationClients.removeItem(pid); 1055 1056 ALOGV("%d died, releasing its sessions", pid); 1057 size_t num = mAudioSessionRefs.size(); 1058 bool removed = false; 1059 for (size_t i = 0; i< num; ) { 1060 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1061 ALOGV(" pid %d @ %d", ref->mPid, i); 1062 if (ref->mPid == pid) { 1063 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1064 mAudioSessionRefs.removeAt(i); 1065 delete ref; 1066 removed = true; 1067 num--; 1068 } else { 1069 i++; 1070 } 1071 } 1072 if (removed) { 1073 purgeStaleEffects_l(); 1074 } 1075} 1076 1077// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1079{ 1080 size_t size = mNotificationClients.size(); 1081 for (size_t i = 0; i < size; i++) { 1082 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1083 param2); 1084 } 1085} 1086 1087// removeClient_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::removeClient_l(pid_t pid) 1089{ 1090 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1091 mClients.removeItem(pid); 1092} 1093 1094 1095// ---------------------------------------------------------------------------- 1096 1097AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1098 uint32_t device, type_t type) 1099 : Thread(false), 1100 mType(type), 1101 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1102 // mChannelMask 1103 mChannelCount(0), 1104 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1105 mParamStatus(NO_ERROR), 1106 mStandby(false), mId(id), 1107 mDevice(device), 1108 mDeathRecipient(new PMDeathRecipient(this)) 1109{ 1110} 1111 1112AudioFlinger::ThreadBase::~ThreadBase() 1113{ 1114 mParamCond.broadcast(); 1115 // do not lock the mutex in destructor 1116 releaseWakeLock_l(); 1117 if (mPowerManager != 0) { 1118 sp<IBinder> binder = mPowerManager->asBinder(); 1119 binder->unlinkToDeath(mDeathRecipient); 1120 } 1121} 1122 1123void AudioFlinger::ThreadBase::exit() 1124{ 1125 ALOGV("ThreadBase::exit"); 1126 { 1127 // This lock prevents the following race in thread (uniprocessor for illustration): 1128 // if (!exitPending()) { 1129 // // context switch from here to exit() 1130 // // exit() calls requestExit(), what exitPending() observes 1131 // // exit() calls signal(), which is dropped since no waiters 1132 // // context switch back from exit() to here 1133 // mWaitWorkCV.wait(...); 1134 // // now thread is hung 1135 // } 1136 AutoMutex lock(mLock); 1137 requestExit(); 1138 mWaitWorkCV.signal(); 1139 } 1140 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1141 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1142 requestExitAndWait(); 1143} 1144 1145status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1146{ 1147 status_t status; 1148 1149 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1150 Mutex::Autolock _l(mLock); 1151 1152 mNewParameters.add(keyValuePairs); 1153 mWaitWorkCV.signal(); 1154 // wait condition with timeout in case the thread loop has exited 1155 // before the request could be processed 1156 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1157 status = mParamStatus; 1158 mWaitWorkCV.signal(); 1159 } else { 1160 status = TIMED_OUT; 1161 } 1162 return status; 1163} 1164 1165void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1166{ 1167 Mutex::Autolock _l(mLock); 1168 sendConfigEvent_l(event, param); 1169} 1170 1171// sendConfigEvent_l() must be called with ThreadBase::mLock held 1172void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1173{ 1174 ConfigEvent configEvent; 1175 configEvent.mEvent = event; 1176 configEvent.mParam = param; 1177 mConfigEvents.add(configEvent); 1178 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1179 mWaitWorkCV.signal(); 1180} 1181 1182void AudioFlinger::ThreadBase::processConfigEvents() 1183{ 1184 mLock.lock(); 1185 while (!mConfigEvents.isEmpty()) { 1186 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1187 ConfigEvent configEvent = mConfigEvents[0]; 1188 mConfigEvents.removeAt(0); 1189 // release mLock before locking AudioFlinger mLock: lock order is always 1190 // AudioFlinger then ThreadBase to avoid cross deadlock 1191 mLock.unlock(); 1192 mAudioFlinger->mLock.lock(); 1193 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1194 mAudioFlinger->mLock.unlock(); 1195 mLock.lock(); 1196 } 1197 mLock.unlock(); 1198} 1199 1200status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1201{ 1202 const size_t SIZE = 256; 1203 char buffer[SIZE]; 1204 String8 result; 1205 1206 bool locked = tryLock(mLock); 1207 if (!locked) { 1208 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1209 write(fd, buffer, strlen(buffer)); 1210 } 1211 1212 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1221 result.append(buffer); 1222 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1223 result.append(buffer); 1224 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1225 result.append(buffer); 1226 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1227 result.append(buffer); 1228 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1229 result.append(buffer); 1230 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1231 result.append(buffer); 1232 1233 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1234 result.append(buffer); 1235 result.append(" Index Command"); 1236 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1237 snprintf(buffer, SIZE, "\n %02d ", i); 1238 result.append(buffer); 1239 result.append(mNewParameters[i]); 1240 } 1241 1242 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, " Index event param\n"); 1245 result.append(buffer); 1246 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1247 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1248 result.append(buffer); 1249 } 1250 result.append("\n"); 1251 1252 write(fd, result.string(), result.size()); 1253 1254 if (locked) { 1255 mLock.unlock(); 1256 } 1257 return NO_ERROR; 1258} 1259 1260status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1261{ 1262 const size_t SIZE = 256; 1263 char buffer[SIZE]; 1264 String8 result; 1265 1266 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1267 write(fd, buffer, strlen(buffer)); 1268 1269 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1270 sp<EffectChain> chain = mEffectChains[i]; 1271 if (chain != 0) { 1272 chain->dump(fd, args); 1273 } 1274 } 1275 return NO_ERROR; 1276} 1277 1278void AudioFlinger::ThreadBase::acquireWakeLock() 1279{ 1280 Mutex::Autolock _l(mLock); 1281 acquireWakeLock_l(); 1282} 1283 1284void AudioFlinger::ThreadBase::acquireWakeLock_l() 1285{ 1286 if (mPowerManager == 0) { 1287 // use checkService() to avoid blocking if power service is not up yet 1288 sp<IBinder> binder = 1289 defaultServiceManager()->checkService(String16("power")); 1290 if (binder == 0) { 1291 ALOGW("Thread %s cannot connect to the power manager service", mName); 1292 } else { 1293 mPowerManager = interface_cast<IPowerManager>(binder); 1294 binder->linkToDeath(mDeathRecipient); 1295 } 1296 } 1297 if (mPowerManager != 0) { 1298 sp<IBinder> binder = new BBinder(); 1299 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1300 binder, 1301 String16(mName)); 1302 if (status == NO_ERROR) { 1303 mWakeLockToken = binder; 1304 } 1305 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1306 } 1307} 1308 1309void AudioFlinger::ThreadBase::releaseWakeLock() 1310{ 1311 Mutex::Autolock _l(mLock); 1312 releaseWakeLock_l(); 1313} 1314 1315void AudioFlinger::ThreadBase::releaseWakeLock_l() 1316{ 1317 if (mWakeLockToken != 0) { 1318 ALOGV("releaseWakeLock_l() %s", mName); 1319 if (mPowerManager != 0) { 1320 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1321 } 1322 mWakeLockToken.clear(); 1323 } 1324} 1325 1326void AudioFlinger::ThreadBase::clearPowerManager() 1327{ 1328 Mutex::Autolock _l(mLock); 1329 releaseWakeLock_l(); 1330 mPowerManager.clear(); 1331} 1332 1333void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1334{ 1335 sp<ThreadBase> thread = mThread.promote(); 1336 if (thread != 0) { 1337 thread->clearPowerManager(); 1338 } 1339 ALOGW("power manager service died !!!"); 1340} 1341 1342void AudioFlinger::ThreadBase::setEffectSuspended( 1343 const effect_uuid_t *type, bool suspend, int sessionId) 1344{ 1345 Mutex::Autolock _l(mLock); 1346 setEffectSuspended_l(type, suspend, sessionId); 1347} 1348 1349void AudioFlinger::ThreadBase::setEffectSuspended_l( 1350 const effect_uuid_t *type, bool suspend, int sessionId) 1351{ 1352 sp<EffectChain> chain = getEffectChain_l(sessionId); 1353 if (chain != 0) { 1354 if (type != NULL) { 1355 chain->setEffectSuspended_l(type, suspend); 1356 } else { 1357 chain->setEffectSuspendedAll_l(suspend); 1358 } 1359 } 1360 1361 updateSuspendedSessions_l(type, suspend, sessionId); 1362} 1363 1364void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1365{ 1366 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1367 if (index < 0) { 1368 return; 1369 } 1370 1371 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1372 mSuspendedSessions.editValueAt(index); 1373 1374 for (size_t i = 0; i < sessionEffects.size(); i++) { 1375 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1376 for (int j = 0; j < desc->mRefCount; j++) { 1377 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1378 chain->setEffectSuspendedAll_l(true); 1379 } else { 1380 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1381 desc->mType.timeLow); 1382 chain->setEffectSuspended_l(&desc->mType, true); 1383 } 1384 } 1385 } 1386} 1387 1388void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1389 bool suspend, 1390 int sessionId) 1391{ 1392 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1393 1394 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1395 1396 if (suspend) { 1397 if (index >= 0) { 1398 sessionEffects = mSuspendedSessions.editValueAt(index); 1399 } else { 1400 mSuspendedSessions.add(sessionId, sessionEffects); 1401 } 1402 } else { 1403 if (index < 0) { 1404 return; 1405 } 1406 sessionEffects = mSuspendedSessions.editValueAt(index); 1407 } 1408 1409 1410 int key = EffectChain::kKeyForSuspendAll; 1411 if (type != NULL) { 1412 key = type->timeLow; 1413 } 1414 index = sessionEffects.indexOfKey(key); 1415 1416 sp<SuspendedSessionDesc> desc; 1417 if (suspend) { 1418 if (index >= 0) { 1419 desc = sessionEffects.valueAt(index); 1420 } else { 1421 desc = new SuspendedSessionDesc(); 1422 if (type != NULL) { 1423 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1424 } 1425 sessionEffects.add(key, desc); 1426 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1427 } 1428 desc->mRefCount++; 1429 } else { 1430 if (index < 0) { 1431 return; 1432 } 1433 desc = sessionEffects.valueAt(index); 1434 if (--desc->mRefCount == 0) { 1435 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1436 sessionEffects.removeItemsAt(index); 1437 if (sessionEffects.isEmpty()) { 1438 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1439 sessionId); 1440 mSuspendedSessions.removeItem(sessionId); 1441 } 1442 } 1443 } 1444 if (!sessionEffects.isEmpty()) { 1445 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1446 } 1447} 1448 1449void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1450 bool enabled, 1451 int sessionId) 1452{ 1453 Mutex::Autolock _l(mLock); 1454 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1455} 1456 1457void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1458 bool enabled, 1459 int sessionId) 1460{ 1461 if (mType != RECORD) { 1462 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1463 // another session. This gives the priority to well behaved effect control panels 1464 // and applications not using global effects. 1465 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1466 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1467 } 1468 } 1469 1470 sp<EffectChain> chain = getEffectChain_l(sessionId); 1471 if (chain != 0) { 1472 chain->checkSuspendOnEffectEnabled(effect, enabled); 1473 } 1474} 1475 1476// ---------------------------------------------------------------------------- 1477 1478AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1479 AudioStreamOut* output, 1480 audio_io_handle_t id, 1481 uint32_t device, 1482 type_t type) 1483 : ThreadBase(audioFlinger, id, device, type), 1484 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1485 // Assumes constructor is called by AudioFlinger with it's mLock held, 1486 // but it would be safer to explicitly pass initial masterMute as parameter 1487 mMasterMute(audioFlinger->masterMute_l()), 1488 // mStreamTypes[] initialized in constructor body 1489 mOutput(output), 1490 // Assumes constructor is called by AudioFlinger with it's mLock held, 1491 // but it would be safer to explicitly pass initial masterVolume as parameter 1492 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1493 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1494 mMixerStatus(MIXER_IDLE), 1495 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1496 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1497 // index 0 is reserved for normal mixer's submix 1498 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1499{ 1500 snprintf(mName, kNameLength, "AudioOut_%X", id); 1501 1502 readOutputParameters(); 1503 1504 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1505 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1506 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1507 stream = (audio_stream_type_t) (stream + 1)) { 1508 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1509 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1510 } 1511 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1512 // because mAudioFlinger doesn't have one to copy from 1513} 1514 1515AudioFlinger::PlaybackThread::~PlaybackThread() 1516{ 1517 delete [] mMixBuffer; 1518} 1519 1520status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1521{ 1522 dumpInternals(fd, args); 1523 dumpTracks(fd, args); 1524 dumpEffectChains(fd, args); 1525 return NO_ERROR; 1526} 1527 1528status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1529{ 1530 const size_t SIZE = 256; 1531 char buffer[SIZE]; 1532 String8 result; 1533 1534 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1535 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1536 const stream_type_t *st = &mStreamTypes[i]; 1537 if (i > 0) { 1538 result.appendFormat(", "); 1539 } 1540 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1541 if (st->mute) { 1542 result.append("M"); 1543 } 1544 } 1545 result.append("\n"); 1546 write(fd, result.string(), result.length()); 1547 result.clear(); 1548 1549 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1550 result.append(buffer); 1551 Track::appendDumpHeader(result); 1552 for (size_t i = 0; i < mTracks.size(); ++i) { 1553 sp<Track> track = mTracks[i]; 1554 if (track != 0) { 1555 track->dump(buffer, SIZE); 1556 result.append(buffer); 1557 } 1558 } 1559 1560 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1561 result.append(buffer); 1562 Track::appendDumpHeader(result); 1563 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1564 sp<Track> track = mActiveTracks[i].promote(); 1565 if (track != 0) { 1566 track->dump(buffer, SIZE); 1567 result.append(buffer); 1568 } 1569 } 1570 write(fd, result.string(), result.size()); 1571 return NO_ERROR; 1572} 1573 1574status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1575{ 1576 const size_t SIZE = 256; 1577 char buffer[SIZE]; 1578 String8 result; 1579 1580 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1581 result.append(buffer); 1582 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1583 result.append(buffer); 1584 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1585 result.append(buffer); 1586 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1587 result.append(buffer); 1588 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1589 result.append(buffer); 1590 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1591 result.append(buffer); 1592 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1593 result.append(buffer); 1594 write(fd, result.string(), result.size()); 1595 1596 dumpBase(fd, args); 1597 1598 return NO_ERROR; 1599} 1600 1601// Thread virtuals 1602status_t AudioFlinger::PlaybackThread::readyToRun() 1603{ 1604 status_t status = initCheck(); 1605 if (status == NO_ERROR) { 1606 ALOGI("AudioFlinger's thread %p ready to run", this); 1607 } else { 1608 ALOGE("No working audio driver found."); 1609 } 1610 return status; 1611} 1612 1613void AudioFlinger::PlaybackThread::onFirstRef() 1614{ 1615 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1616} 1617 1618// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1619sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1620 const sp<AudioFlinger::Client>& client, 1621 audio_stream_type_t streamType, 1622 uint32_t sampleRate, 1623 audio_format_t format, 1624 uint32_t channelMask, 1625 int frameCount, 1626 const sp<IMemory>& sharedBuffer, 1627 int sessionId, 1628 IAudioFlinger::track_flags_t flags, 1629 pid_t tid, 1630 status_t *status) 1631{ 1632 sp<Track> track; 1633 status_t lStatus; 1634 1635 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1636 1637 // client expresses a preference for FAST, but we get the final say 1638 if (flags & IAudioFlinger::TRACK_FAST) { 1639 if ( 1640 // not timed 1641 (!isTimed) && 1642 // either of these use cases: 1643 ( 1644 // use case 1: shared buffer with any frame count 1645 ( 1646 (sharedBuffer != 0) 1647 ) || 1648 // use case 2: callback handler and frame count is default or at least as large as HAL 1649 ( 1650 (tid != -1) && 1651 ((frameCount == 0) || 1652 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1653 ) 1654 ) && 1655 // PCM data 1656 audio_is_linear_pcm(format) && 1657 // mono or stereo 1658 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1659 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1660#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1661 // hardware sample rate 1662 (sampleRate == mSampleRate) && 1663#endif 1664 // normal mixer has an associated fast mixer 1665 hasFastMixer() && 1666 // there are sufficient fast track slots available 1667 (mFastTrackAvailMask != 0) 1668 // FIXME test that MixerThread for this fast track has a capable output HAL 1669 // FIXME add a permission test also? 1670 ) { 1671 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1672 if (frameCount == 0) { 1673 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1674 } 1675 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1676 frameCount, mFrameCount); 1677 } else { 1678 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1679 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1680 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1681 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1682 audio_is_linear_pcm(format), 1683 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1684 flags &= ~IAudioFlinger::TRACK_FAST; 1685 // For compatibility with AudioTrack calculation, buffer depth is forced 1686 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1687 // This is probably too conservative, but legacy application code may depend on it. 1688 // If you change this calculation, also review the start threshold which is related. 1689 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1690 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1691 if (minBufCount < 2) { 1692 minBufCount = 2; 1693 } 1694 int minFrameCount = mNormalFrameCount * minBufCount; 1695 if (frameCount < minFrameCount) { 1696 frameCount = minFrameCount; 1697 } 1698 } 1699 } 1700 1701 if (mType == DIRECT) { 1702 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1703 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1704 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1705 "for output %p with format %d", 1706 sampleRate, format, channelMask, mOutput, mFormat); 1707 lStatus = BAD_VALUE; 1708 goto Exit; 1709 } 1710 } 1711 } else { 1712 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1713 if (sampleRate > mSampleRate*2) { 1714 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1715 lStatus = BAD_VALUE; 1716 goto Exit; 1717 } 1718 } 1719 1720 lStatus = initCheck(); 1721 if (lStatus != NO_ERROR) { 1722 ALOGE("Audio driver not initialized."); 1723 goto Exit; 1724 } 1725 1726 { // scope for mLock 1727 Mutex::Autolock _l(mLock); 1728 1729 // all tracks in same audio session must share the same routing strategy otherwise 1730 // conflicts will happen when tracks are moved from one output to another by audio policy 1731 // manager 1732 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1733 for (size_t i = 0; i < mTracks.size(); ++i) { 1734 sp<Track> t = mTracks[i]; 1735 if (t != 0 && !t->isOutputTrack()) { 1736 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1737 if (sessionId == t->sessionId() && strategy != actual) { 1738 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1739 strategy, actual); 1740 lStatus = BAD_VALUE; 1741 goto Exit; 1742 } 1743 } 1744 } 1745 1746 if (!isTimed) { 1747 track = new Track(this, client, streamType, sampleRate, format, 1748 channelMask, frameCount, sharedBuffer, sessionId, flags); 1749 } else { 1750 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1751 channelMask, frameCount, sharedBuffer, sessionId); 1752 } 1753 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1754 lStatus = NO_MEMORY; 1755 goto Exit; 1756 } 1757 mTracks.add(track); 1758 1759 sp<EffectChain> chain = getEffectChain_l(sessionId); 1760 if (chain != 0) { 1761 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1762 track->setMainBuffer(chain->inBuffer()); 1763 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1764 chain->incTrackCnt(); 1765 } 1766 } 1767 1768#ifdef HAVE_REQUEST_PRIORITY 1769 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1770 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1771 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1772 // so ask activity manager to do this on our behalf 1773 int err = requestPriority(callingPid, tid, 1); 1774 if (err != 0) { 1775 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1776 1, callingPid, tid, err); 1777 } 1778 } 1779#endif 1780 1781 lStatus = NO_ERROR; 1782 1783Exit: 1784 if (status) { 1785 *status = lStatus; 1786 } 1787 return track; 1788} 1789 1790uint32_t AudioFlinger::PlaybackThread::latency() const 1791{ 1792 Mutex::Autolock _l(mLock); 1793 if (initCheck() == NO_ERROR) { 1794 return mOutput->stream->get_latency(mOutput->stream); 1795 } else { 1796 return 0; 1797 } 1798} 1799 1800void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1801{ 1802 Mutex::Autolock _l(mLock); 1803 mMasterVolume = value; 1804} 1805 1806void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1807{ 1808 Mutex::Autolock _l(mLock); 1809 setMasterMute_l(muted); 1810} 1811 1812void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1813{ 1814 Mutex::Autolock _l(mLock); 1815 mStreamTypes[stream].volume = value; 1816} 1817 1818void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1819{ 1820 Mutex::Autolock _l(mLock); 1821 mStreamTypes[stream].mute = muted; 1822} 1823 1824float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1825{ 1826 Mutex::Autolock _l(mLock); 1827 return mStreamTypes[stream].volume; 1828} 1829 1830// addTrack_l() must be called with ThreadBase::mLock held 1831status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1832{ 1833 status_t status = ALREADY_EXISTS; 1834 1835 // set retry count for buffer fill 1836 track->mRetryCount = kMaxTrackStartupRetries; 1837 if (mActiveTracks.indexOf(track) < 0) { 1838 // the track is newly added, make sure it fills up all its 1839 // buffers before playing. This is to ensure the client will 1840 // effectively get the latency it requested. 1841 track->mFillingUpStatus = Track::FS_FILLING; 1842 track->mResetDone = false; 1843 mActiveTracks.add(track); 1844 if (track->mainBuffer() != mMixBuffer) { 1845 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1846 if (chain != 0) { 1847 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1848 chain->incActiveTrackCnt(); 1849 } 1850 } 1851 1852 status = NO_ERROR; 1853 } 1854 1855 ALOGV("mWaitWorkCV.broadcast"); 1856 mWaitWorkCV.broadcast(); 1857 1858 return status; 1859} 1860 1861// destroyTrack_l() must be called with ThreadBase::mLock held 1862void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1863{ 1864 track->mState = TrackBase::TERMINATED; 1865 // active tracks are removed by threadLoop() 1866 if (mActiveTracks.indexOf(track) < 0) { 1867 removeTrack_l(track); 1868 } 1869} 1870 1871void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1872{ 1873 mTracks.remove(track); 1874 deleteTrackName_l(track->name()); 1875 // redundant as track is about to be destroyed, for dumpsys only 1876 track->mName = -1; 1877 if (track->isFastTrack()) { 1878 int index = track->mFastIndex; 1879 ALOG_ASSERT(0 < index && index < FastMixerState::kMaxFastTracks); 1880 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1881 mFastTrackAvailMask |= 1 << index; 1882 // redundant as track is about to be destroyed, for dumpsys only 1883 track->mFastIndex = -1; 1884 } 1885 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1886 if (chain != 0) { 1887 chain->decTrackCnt(); 1888 } 1889} 1890 1891String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1892{ 1893 String8 out_s8 = String8(""); 1894 char *s; 1895 1896 Mutex::Autolock _l(mLock); 1897 if (initCheck() != NO_ERROR) { 1898 return out_s8; 1899 } 1900 1901 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1902 out_s8 = String8(s); 1903 free(s); 1904 return out_s8; 1905} 1906 1907// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1908void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1909 AudioSystem::OutputDescriptor desc; 1910 void *param2 = NULL; 1911 1912 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1913 1914 switch (event) { 1915 case AudioSystem::OUTPUT_OPENED: 1916 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1917 desc.channels = mChannelMask; 1918 desc.samplingRate = mSampleRate; 1919 desc.format = mFormat; 1920 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1921 desc.latency = latency(); 1922 param2 = &desc; 1923 break; 1924 1925 case AudioSystem::STREAM_CONFIG_CHANGED: 1926 param2 = ¶m; 1927 case AudioSystem::OUTPUT_CLOSED: 1928 default: 1929 break; 1930 } 1931 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1932} 1933 1934void AudioFlinger::PlaybackThread::readOutputParameters() 1935{ 1936 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1937 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1938 mChannelCount = (uint16_t)popcount(mChannelMask); 1939 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1940 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1941 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1942 if (mFrameCount & 15) { 1943 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1944 mFrameCount); 1945 } 1946 1947 // Calculate size of normal mix buffer relative to the HAL output buffer size 1948 uint32_t multiple = 1; 1949 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1950 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1951 multiple = (minNormalFrameCount + mFrameCount - 1) / mFrameCount; 1952 // force multiple to be even, for compatibility with doubling of fast tracks due to HAL SRC 1953 // (it would be unusual for the normal mix buffer size to not be a multiple of fast track) 1954 // FIXME this rounding up should not be done if no HAL SRC 1955 if ((multiple > 2) && (multiple & 1)) { 1956 ++multiple; 1957 } 1958 } 1959 mNormalFrameCount = multiple * mFrameCount; 1960 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 1961 1962 // FIXME - Current mixer implementation only supports stereo output: Always 1963 // Allocate a stereo buffer even if HW output is mono. 1964 delete[] mMixBuffer; 1965 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 1966 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 1967 1968 // force reconfiguration of effect chains and engines to take new buffer size and audio 1969 // parameters into account 1970 // Note that mLock is not held when readOutputParameters() is called from the constructor 1971 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1972 // matter. 1973 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1974 Vector< sp<EffectChain> > effectChains = mEffectChains; 1975 for (size_t i = 0; i < effectChains.size(); i ++) { 1976 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1977 } 1978} 1979 1980status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1981{ 1982 if (halFrames == NULL || dspFrames == NULL) { 1983 return BAD_VALUE; 1984 } 1985 Mutex::Autolock _l(mLock); 1986 if (initCheck() != NO_ERROR) { 1987 return INVALID_OPERATION; 1988 } 1989 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1990 1991 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1992} 1993 1994uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1995{ 1996 Mutex::Autolock _l(mLock); 1997 uint32_t result = 0; 1998 if (getEffectChain_l(sessionId) != 0) { 1999 result = EFFECT_SESSION; 2000 } 2001 2002 for (size_t i = 0; i < mTracks.size(); ++i) { 2003 sp<Track> track = mTracks[i]; 2004 if (sessionId == track->sessionId() && 2005 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2006 result |= TRACK_SESSION; 2007 break; 2008 } 2009 } 2010 2011 return result; 2012} 2013 2014uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2015{ 2016 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2017 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2018 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2019 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2020 } 2021 for (size_t i = 0; i < mTracks.size(); i++) { 2022 sp<Track> track = mTracks[i]; 2023 if (sessionId == track->sessionId() && 2024 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2025 return AudioSystem::getStrategyForStream(track->streamType()); 2026 } 2027 } 2028 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2029} 2030 2031 2032AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2033{ 2034 Mutex::Autolock _l(mLock); 2035 return mOutput; 2036} 2037 2038AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2039{ 2040 Mutex::Autolock _l(mLock); 2041 AudioStreamOut *output = mOutput; 2042 mOutput = NULL; 2043 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2044 // must push a NULL and wait for ack 2045 mOutputSink.clear(); 2046 mPipeSink.clear(); 2047 mNormalSink.clear(); 2048 return output; 2049} 2050 2051// this method must always be called either with ThreadBase mLock held or inside the thread loop 2052audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2053{ 2054 if (mOutput == NULL) { 2055 return NULL; 2056 } 2057 return &mOutput->stream->common; 2058} 2059 2060uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2061{ 2062 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2063 // decoding and transfer time. So sleeping for half of the latency would likely cause 2064 // underruns 2065 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2066 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2067 } else { 2068 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2069 } 2070} 2071 2072status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2073{ 2074 if (!isValidSyncEvent(event)) { 2075 return BAD_VALUE; 2076 } 2077 2078 Mutex::Autolock _l(mLock); 2079 2080 for (size_t i = 0; i < mTracks.size(); ++i) { 2081 sp<Track> track = mTracks[i]; 2082 if (event->triggerSession() == track->sessionId()) { 2083 track->setSyncEvent(event); 2084 return NO_ERROR; 2085 } 2086 } 2087 2088 return NAME_NOT_FOUND; 2089} 2090 2091bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2092{ 2093 switch (event->type()) { 2094 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2095 return true; 2096 default: 2097 break; 2098 } 2099 return false; 2100} 2101 2102// ---------------------------------------------------------------------------- 2103 2104AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2105 audio_io_handle_t id, uint32_t device, type_t type) 2106 : PlaybackThread(audioFlinger, output, id, device, type), 2107 // mAudioMixer below 2108#ifdef SOAKER 2109 mSoaker(NULL), 2110#endif 2111 // mFastMixer below 2112 mFastMixerFutex(0) 2113 // mOutputSink below 2114 // mPipeSink below 2115 // mNormalSink below 2116{ 2117 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2118 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2119 "mFrameCount=%d, mNormalFrameCount=%d", 2120 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2121 mNormalFrameCount); 2122 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2123 2124 // FIXME - Current mixer implementation only supports stereo output 2125 if (mChannelCount == 1) { 2126 ALOGE("Invalid audio hardware channel count"); 2127 } 2128 2129 // create an NBAIO sink for the HAL output stream, and negotiate 2130 mOutputSink = new AudioStreamOutSink(output->stream); 2131 size_t numCounterOffers = 0; 2132 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2133 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2134 ALOG_ASSERT(index == 0); 2135 2136 // initialize fast mixer depending on configuration 2137 bool initFastMixer; 2138 switch (kUseFastMixer) { 2139 case FastMixer_Never: 2140 initFastMixer = false; 2141 break; 2142 case FastMixer_Always: 2143 initFastMixer = true; 2144 break; 2145 case FastMixer_Static: 2146 case FastMixer_Dynamic: 2147 initFastMixer = mFrameCount < mNormalFrameCount; 2148 break; 2149 } 2150 if (initFastMixer) { 2151 2152 // create a MonoPipe to connect our submix to FastMixer 2153 NBAIO_Format format = mOutputSink->format(); 2154 // frame count will be rounded up to a power of 2, so this formula should work well 2155 MonoPipe *monoPipe = new MonoPipe((mNormalFrameCount * 3) / 2, format, 2156 true /*writeCanBlock*/); 2157 const NBAIO_Format offers[1] = {format}; 2158 size_t numCounterOffers = 0; 2159 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2160 ALOG_ASSERT(index == 0); 2161 mPipeSink = monoPipe; 2162 2163#ifdef SOAKER 2164 // create a soaker as workaround for governor issues 2165 mSoaker = new Soaker(); 2166 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2167 mSoaker->run("Soaker", PRIORITY_LOWEST); 2168#endif 2169 2170 // create fast mixer and configure it initially with just one fast track for our submix 2171 mFastMixer = new FastMixer(); 2172 FastMixerStateQueue *sq = mFastMixer->sq(); 2173 FastMixerState *state = sq->begin(); 2174 FastTrack *fastTrack = &state->mFastTracks[0]; 2175 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2176 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2177 fastTrack->mVolumeProvider = NULL; 2178 fastTrack->mGeneration++; 2179 state->mFastTracksGen++; 2180 state->mTrackMask = 1; 2181 // fast mixer will use the HAL output sink 2182 state->mOutputSink = mOutputSink.get(); 2183 state->mOutputSinkGen++; 2184 state->mFrameCount = mFrameCount; 2185 state->mCommand = FastMixerState::COLD_IDLE; 2186 // already done in constructor initialization list 2187 //mFastMixerFutex = 0; 2188 state->mColdFutexAddr = &mFastMixerFutex; 2189 state->mColdGen++; 2190 state->mDumpState = &mFastMixerDumpState; 2191 sq->end(); 2192 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2193 2194 // start the fast mixer 2195 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2196#ifdef HAVE_REQUEST_PRIORITY 2197 pid_t tid = mFastMixer->getTid(); 2198 int err = requestPriority(getpid_cached, tid, 2); 2199 if (err != 0) { 2200 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2201 2, getpid_cached, tid, err); 2202 } 2203#endif 2204 2205 } else { 2206 mFastMixer = NULL; 2207 } 2208 2209 switch (kUseFastMixer) { 2210 case FastMixer_Never: 2211 case FastMixer_Dynamic: 2212 mNormalSink = mOutputSink; 2213 break; 2214 case FastMixer_Always: 2215 mNormalSink = mPipeSink; 2216 break; 2217 case FastMixer_Static: 2218 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2219 break; 2220 } 2221} 2222 2223AudioFlinger::MixerThread::~MixerThread() 2224{ 2225 if (mFastMixer != NULL) { 2226 FastMixerStateQueue *sq = mFastMixer->sq(); 2227 FastMixerState *state = sq->begin(); 2228 if (state->mCommand == FastMixerState::COLD_IDLE) { 2229 int32_t old = android_atomic_inc(&mFastMixerFutex); 2230 if (old == -1) { 2231 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2232 } 2233 } 2234 state->mCommand = FastMixerState::EXIT; 2235 sq->end(); 2236 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2237 mFastMixer->join(); 2238 // Though the fast mixer thread has exited, it's state queue is still valid. 2239 // We'll use that extract the final state which contains one remaining fast track 2240 // corresponding to our sub-mix. 2241 state = sq->begin(); 2242 ALOG_ASSERT(state->mTrackMask == 1); 2243 FastTrack *fastTrack = &state->mFastTracks[0]; 2244 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2245 delete fastTrack->mBufferProvider; 2246 sq->end(false /*didModify*/); 2247 delete mFastMixer; 2248#ifdef SOAKER 2249 if (mSoaker != NULL) { 2250 mSoaker->requestExitAndWait(); 2251 } 2252 delete mSoaker; 2253#endif 2254 } 2255 delete mAudioMixer; 2256} 2257 2258class CpuStats { 2259public: 2260 CpuStats(); 2261 void sample(const String8 &title); 2262#ifdef DEBUG_CPU_USAGE 2263private: 2264 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2265 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2266 2267 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2268 2269 int mCpuNum; // thread's current CPU number 2270 int mCpukHz; // frequency of thread's current CPU in kHz 2271#endif 2272}; 2273 2274CpuStats::CpuStats() 2275#ifdef DEBUG_CPU_USAGE 2276 : mCpuNum(-1), mCpukHz(-1) 2277#endif 2278{ 2279} 2280 2281void CpuStats::sample(const String8 &title) { 2282#ifdef DEBUG_CPU_USAGE 2283 // get current thread's delta CPU time in wall clock ns 2284 double wcNs; 2285 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2286 2287 // record sample for wall clock statistics 2288 if (valid) { 2289 mWcStats.sample(wcNs); 2290 } 2291 2292 // get the current CPU number 2293 int cpuNum = sched_getcpu(); 2294 2295 // get the current CPU frequency in kHz 2296 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2297 2298 // check if either CPU number or frequency changed 2299 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2300 mCpuNum = cpuNum; 2301 mCpukHz = cpukHz; 2302 // ignore sample for purposes of cycles 2303 valid = false; 2304 } 2305 2306 // if no change in CPU number or frequency, then record sample for cycle statistics 2307 if (valid && mCpukHz > 0) { 2308 double cycles = wcNs * cpukHz * 0.000001; 2309 mHzStats.sample(cycles); 2310 } 2311 2312 unsigned n = mWcStats.n(); 2313 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2314 if ((n & 127) == 1) { 2315 long long elapsed = mCpuUsage.elapsed(); 2316 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2317 double perLoop = elapsed / (double) n; 2318 double perLoop100 = perLoop * 0.01; 2319 double perLoop1k = perLoop * 0.001; 2320 double mean = mWcStats.mean(); 2321 double stddev = mWcStats.stddev(); 2322 double minimum = mWcStats.minimum(); 2323 double maximum = mWcStats.maximum(); 2324 double meanCycles = mHzStats.mean(); 2325 double stddevCycles = mHzStats.stddev(); 2326 double minCycles = mHzStats.minimum(); 2327 double maxCycles = mHzStats.maximum(); 2328 mCpuUsage.resetElapsed(); 2329 mWcStats.reset(); 2330 mHzStats.reset(); 2331 ALOGD("CPU usage for %s over past %.1f secs\n" 2332 " (%u mixer loops at %.1f mean ms per loop):\n" 2333 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2334 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2335 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2336 title.string(), 2337 elapsed * .000000001, n, perLoop * .000001, 2338 mean * .001, 2339 stddev * .001, 2340 minimum * .001, 2341 maximum * .001, 2342 mean / perLoop100, 2343 stddev / perLoop100, 2344 minimum / perLoop100, 2345 maximum / perLoop100, 2346 meanCycles / perLoop1k, 2347 stddevCycles / perLoop1k, 2348 minCycles / perLoop1k, 2349 maxCycles / perLoop1k); 2350 2351 } 2352 } 2353#endif 2354}; 2355 2356void AudioFlinger::PlaybackThread::checkSilentMode_l() 2357{ 2358 if (!mMasterMute) { 2359 char value[PROPERTY_VALUE_MAX]; 2360 if (property_get("ro.audio.silent", value, "0") > 0) { 2361 char *endptr; 2362 unsigned long ul = strtoul(value, &endptr, 0); 2363 if (*endptr == '\0' && ul != 0) { 2364 ALOGD("Silence is golden"); 2365 // The setprop command will not allow a property to be changed after 2366 // the first time it is set, so we don't have to worry about un-muting. 2367 setMasterMute_l(true); 2368 } 2369 } 2370 } 2371} 2372 2373bool AudioFlinger::PlaybackThread::threadLoop() 2374{ 2375 Vector< sp<Track> > tracksToRemove; 2376 2377 standbyTime = systemTime(); 2378 2379 // MIXER 2380 nsecs_t lastWarning = 0; 2381if (mType == MIXER) { 2382 longStandbyExit = false; 2383} 2384 2385 // DUPLICATING 2386 // FIXME could this be made local to while loop? 2387 writeFrames = 0; 2388 2389 cacheParameters_l(); 2390 sleepTime = idleSleepTime; 2391 2392if (mType == MIXER) { 2393 sleepTimeShift = 0; 2394} 2395 2396 CpuStats cpuStats; 2397 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2398 2399 acquireWakeLock(); 2400 2401 while (!exitPending()) 2402 { 2403 cpuStats.sample(myName); 2404 2405 Vector< sp<EffectChain> > effectChains; 2406 2407 processConfigEvents(); 2408 2409 { // scope for mLock 2410 2411 Mutex::Autolock _l(mLock); 2412 2413 if (checkForNewParameters_l()) { 2414 cacheParameters_l(); 2415 } 2416 2417 saveOutputTracks(); 2418 2419 // put audio hardware into standby after short delay 2420 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2421 mSuspended > 0)) { 2422 if (!mStandby) { 2423 2424 threadLoop_standby(); 2425 2426 mStandby = true; 2427 mBytesWritten = 0; 2428 } 2429 2430 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2431 // we're about to wait, flush the binder command buffer 2432 IPCThreadState::self()->flushCommands(); 2433 2434 clearOutputTracks(); 2435 2436 if (exitPending()) break; 2437 2438 releaseWakeLock_l(); 2439 // wait until we have something to do... 2440 ALOGV("%s going to sleep", myName.string()); 2441 mWaitWorkCV.wait(mLock); 2442 ALOGV("%s waking up", myName.string()); 2443 acquireWakeLock_l(); 2444 2445 mMixerStatus = MIXER_IDLE; 2446 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2447 2448 checkSilentMode_l(); 2449 2450 standbyTime = systemTime() + standbyDelay; 2451 sleepTime = idleSleepTime; 2452 if (mType == MIXER) { 2453 sleepTimeShift = 0; 2454 } 2455 2456 continue; 2457 } 2458 } 2459 2460 // mMixerStatusIgnoringFastTracks is also updated internally 2461 mMixerStatus = prepareTracks_l(&tracksToRemove); 2462 2463 // prevent any changes in effect chain list and in each effect chain 2464 // during mixing and effect process as the audio buffers could be deleted 2465 // or modified if an effect is created or deleted 2466 lockEffectChains_l(effectChains); 2467 } 2468 2469 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2470 threadLoop_mix(); 2471 } else { 2472 threadLoop_sleepTime(); 2473 } 2474 2475 if (mSuspended > 0) { 2476 sleepTime = suspendSleepTimeUs(); 2477 } 2478 2479 // only process effects if we're going to write 2480 if (sleepTime == 0) { 2481 for (size_t i = 0; i < effectChains.size(); i ++) { 2482 effectChains[i]->process_l(); 2483 } 2484 } 2485 2486 // enable changes in effect chain 2487 unlockEffectChains(effectChains); 2488 2489 // sleepTime == 0 means we must write to audio hardware 2490 if (sleepTime == 0) { 2491 2492 threadLoop_write(); 2493 2494if (mType == MIXER) { 2495 // write blocked detection 2496 nsecs_t now = systemTime(); 2497 nsecs_t delta = now - mLastWriteTime; 2498 if (!mStandby && delta > maxPeriod) { 2499 mNumDelayedWrites++; 2500 if ((now - lastWarning) > kWarningThrottleNs) { 2501 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2502 ns2ms(delta), mNumDelayedWrites, this); 2503 lastWarning = now; 2504 } 2505 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2506 // a different threshold. Or completely removed for what it is worth anyway... 2507 if (mStandby) { 2508 longStandbyExit = true; 2509 } 2510 } 2511} 2512 2513 mStandby = false; 2514 } else { 2515 usleep(sleepTime); 2516 } 2517 2518 // Finally let go of removed track(s), without the lock held 2519 // since we can't guarantee the destructors won't acquire that 2520 // same lock. This will also mutate and push a new fast mixer state. 2521 threadLoop_removeTracks(tracksToRemove); 2522 tracksToRemove.clear(); 2523 2524 // FIXME I don't understand the need for this here; 2525 // it was in the original code but maybe the 2526 // assignment in saveOutputTracks() makes this unnecessary? 2527 clearOutputTracks(); 2528 2529 // Effect chains will be actually deleted here if they were removed from 2530 // mEffectChains list during mixing or effects processing 2531 effectChains.clear(); 2532 2533 // FIXME Note that the above .clear() is no longer necessary since effectChains 2534 // is now local to this block, but will keep it for now (at least until merge done). 2535 } 2536 2537if (mType == MIXER || mType == DIRECT) { 2538 // put output stream into standby mode 2539 if (!mStandby) { 2540 mOutput->stream->common.standby(&mOutput->stream->common); 2541 } 2542} 2543if (mType == DUPLICATING) { 2544 // for DuplicatingThread, standby mode is handled by the outputTracks 2545} 2546 2547 releaseWakeLock(); 2548 2549 ALOGV("Thread %p type %d exiting", this, mType); 2550 return false; 2551} 2552 2553// returns (via tracksToRemove) a set of tracks to remove. 2554void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2555{ 2556 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2557} 2558 2559void AudioFlinger::MixerThread::threadLoop_write() 2560{ 2561 // FIXME we should only do one push per cycle; confirm this is true 2562 // Start the fast mixer if it's not already running 2563 if (mFastMixer != NULL) { 2564 FastMixerStateQueue *sq = mFastMixer->sq(); 2565 FastMixerState *state = sq->begin(); 2566 if (state->mCommand != FastMixerState::MIX_WRITE && 2567 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2568 if (state->mCommand == FastMixerState::COLD_IDLE) { 2569 int32_t old = android_atomic_inc(&mFastMixerFutex); 2570 if (old == -1) { 2571 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2572 } 2573 } 2574 state->mCommand = FastMixerState::MIX_WRITE; 2575 sq->end(); 2576 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2577 if (kUseFastMixer == FastMixer_Dynamic) { 2578 mNormalSink = mPipeSink; 2579 } 2580 } else { 2581 sq->end(false /*didModify*/); 2582 } 2583 } 2584 PlaybackThread::threadLoop_write(); 2585} 2586 2587// shared by MIXER and DIRECT, overridden by DUPLICATING 2588void AudioFlinger::PlaybackThread::threadLoop_write() 2589{ 2590 // FIXME rewrite to reduce number of system calls 2591 mLastWriteTime = systemTime(); 2592 mInWrite = true; 2593 2594#define mBitShift 2 // FIXME 2595 size_t count = mixBufferSize >> mBitShift; 2596 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2597 if (framesWritten > 0) { 2598 size_t bytesWritten = framesWritten << mBitShift; 2599 mBytesWritten += bytesWritten; 2600 } 2601 2602 mNumWrites++; 2603 mInWrite = false; 2604} 2605 2606void AudioFlinger::MixerThread::threadLoop_standby() 2607{ 2608 // Idle the fast mixer if it's currently running 2609 if (mFastMixer != NULL) { 2610 FastMixerStateQueue *sq = mFastMixer->sq(); 2611 FastMixerState *state = sq->begin(); 2612 if (!(state->mCommand & FastMixerState::IDLE)) { 2613 state->mCommand = FastMixerState::COLD_IDLE; 2614 state->mColdFutexAddr = &mFastMixerFutex; 2615 state->mColdGen++; 2616 mFastMixerFutex = 0; 2617 sq->end(); 2618 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2619 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2620 if (kUseFastMixer == FastMixer_Dynamic) { 2621 mNormalSink = mOutputSink; 2622 } 2623 } else { 2624 sq->end(false /*didModify*/); 2625 } 2626 } 2627 PlaybackThread::threadLoop_standby(); 2628} 2629 2630// shared by MIXER and DIRECT, overridden by DUPLICATING 2631void AudioFlinger::PlaybackThread::threadLoop_standby() 2632{ 2633 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2634 mOutput->stream->common.standby(&mOutput->stream->common); 2635} 2636 2637void AudioFlinger::MixerThread::threadLoop_mix() 2638{ 2639 // obtain the presentation timestamp of the next output buffer 2640 int64_t pts; 2641 status_t status = INVALID_OPERATION; 2642 2643 if (NULL != mOutput->stream->get_next_write_timestamp) { 2644 status = mOutput->stream->get_next_write_timestamp( 2645 mOutput->stream, &pts); 2646 } 2647 2648 if (status != NO_ERROR) { 2649 pts = AudioBufferProvider::kInvalidPTS; 2650 } 2651 2652 // mix buffers... 2653 mAudioMixer->process(pts); 2654 // increase sleep time progressively when application underrun condition clears. 2655 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2656 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2657 // such that we would underrun the audio HAL. 2658 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2659 sleepTimeShift--; 2660 } 2661 sleepTime = 0; 2662 standbyTime = systemTime() + standbyDelay; 2663 //TODO: delay standby when effects have a tail 2664} 2665 2666void AudioFlinger::MixerThread::threadLoop_sleepTime() 2667{ 2668 // If no tracks are ready, sleep once for the duration of an output 2669 // buffer size, then write 0s to the output 2670 if (sleepTime == 0) { 2671 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2672 sleepTime = activeSleepTime >> sleepTimeShift; 2673 if (sleepTime < kMinThreadSleepTimeUs) { 2674 sleepTime = kMinThreadSleepTimeUs; 2675 } 2676 // reduce sleep time in case of consecutive application underruns to avoid 2677 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2678 // duration we would end up writing less data than needed by the audio HAL if 2679 // the condition persists. 2680 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2681 sleepTimeShift++; 2682 } 2683 } else { 2684 sleepTime = idleSleepTime; 2685 } 2686 } else if (mBytesWritten != 0 || 2687 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2688 memset (mMixBuffer, 0, mixBufferSize); 2689 sleepTime = 0; 2690 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2691 } 2692 // TODO add standby time extension fct of effect tail 2693} 2694 2695// prepareTracks_l() must be called with ThreadBase::mLock held 2696AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2697 Vector< sp<Track> > *tracksToRemove) 2698{ 2699 2700 mixer_state mixerStatus = MIXER_IDLE; 2701 // find out which tracks need to be processed 2702 size_t count = mActiveTracks.size(); 2703 size_t mixedTracks = 0; 2704 size_t tracksWithEffect = 0; 2705 // counts only _active_ fast tracks 2706 size_t fastTracks = 0; 2707 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2708 2709 float masterVolume = mMasterVolume; 2710 bool masterMute = mMasterMute; 2711 2712 if (masterMute) { 2713 masterVolume = 0; 2714 } 2715 // Delegate master volume control to effect in output mix effect chain if needed 2716 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2717 if (chain != 0) { 2718 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2719 chain->setVolume_l(&v, &v); 2720 masterVolume = (float)((v + (1 << 23)) >> 24); 2721 chain.clear(); 2722 } 2723 2724 // prepare a new state to push 2725 FastMixerStateQueue *sq = NULL; 2726 FastMixerState *state = NULL; 2727 bool didModify = false; 2728 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2729 if (mFastMixer != NULL) { 2730 sq = mFastMixer->sq(); 2731 state = sq->begin(); 2732 } 2733 2734 for (size_t i=0 ; i<count ; i++) { 2735 sp<Track> t = mActiveTracks[i].promote(); 2736 if (t == 0) continue; 2737 2738 // this const just means the local variable doesn't change 2739 Track* const track = t.get(); 2740 2741 // process fast tracks 2742 if (track->isFastTrack()) { 2743 2744 // It's theoretically possible (though unlikely) for a fast track to be created 2745 // and then removed within the same normal mix cycle. This is not a problem, as 2746 // the track never becomes active so it's fast mixer slot is never touched. 2747 // The converse, of removing an (active) track and then creating a new track 2748 // at the identical fast mixer slot within the same normal mix cycle, 2749 // is impossible because the slot isn't marked available until the end of each cycle. 2750 int j = track->mFastIndex; 2751 FastTrack *fastTrack = &state->mFastTracks[j]; 2752 2753 // Determine whether the track is currently in underrun condition, 2754 // and whether it had a recent underrun. 2755 uint32_t underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2756 uint32_t recentUnderruns = (underruns - (track->mObservedUnderruns & ~1)) >> 1; 2757 // don't count underruns that occur while stopping or pausing 2758 if (!(track->isStopped() || track->isPausing())) { 2759 track->mUnderrunCount += recentUnderruns; 2760 } 2761 track->mObservedUnderruns = underruns; 2762 2763 // This is similar to the formula for normal tracks, 2764 // with a few modifications for fast tracks. 2765 bool isActive; 2766 if (track->isStopped()) { 2767 // track stays active after stop() until first underrun 2768 isActive = recentUnderruns == 0; 2769 } else if (track->isPaused() || track->isTerminated()) { 2770 isActive = false; 2771 } else if (track->isPausing()) { 2772 // ramp down is not yet implemented 2773 isActive = true; 2774 track->setPaused(); 2775 } else if (track->isResuming()) { 2776 // ramp up is not yet implemented 2777 isActive = true; 2778 track->mState = TrackBase::ACTIVE; 2779 } else { 2780 // no minimum frame count for fast tracks; continual underrun is allowed, 2781 // but later could implement automatic pause after several consecutive underruns, 2782 // or auto-mute yet still consider the track active and continue to service it 2783 isActive = true; 2784 } 2785 2786 if (isActive) { 2787 // was it previously inactive? 2788 if (!(state->mTrackMask & (1 << j))) { 2789 ExtendedAudioBufferProvider *eabp = track; 2790 VolumeProvider *vp = track; 2791 fastTrack->mBufferProvider = eabp; 2792 fastTrack->mVolumeProvider = vp; 2793 fastTrack->mSampleRate = track->mSampleRate; 2794 fastTrack->mChannelMask = track->mChannelMask; 2795 fastTrack->mGeneration++; 2796 state->mTrackMask |= 1 << j; 2797 didModify = true; 2798 // no acknowledgement required for newly active tracks 2799 } 2800 // cache the combined master volume and stream type volume for fast mixer; this 2801 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2802 track->mCachedVolume = track->isMuted() ? 2803 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2804 ++fastTracks; 2805 } else { 2806 // was it previously active? 2807 if (state->mTrackMask & (1 << j)) { 2808 fastTrack->mBufferProvider = NULL; 2809 fastTrack->mGeneration++; 2810 state->mTrackMask &= ~(1 << j); 2811 didModify = true; 2812 // If any fast tracks were removed, we must wait for acknowledgement 2813 // because we're about to decrement the last sp<> on those tracks. 2814 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2815 } 2816 // Remainder of this block is copied from similar code for normal tracks 2817 if (track->isStopped()) { 2818 // Can't reset directly, as fast mixer is still polling this track 2819 // track->reset(); 2820 // So instead mark this track as needing to be reset after push with ack 2821 resetMask |= 1 << i; 2822 } 2823 // This would be incomplete if we auto-paused on underrun 2824 size_t audioHALFrames = 2825 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2826 size_t framesWritten = 2827 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2828 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2829 tracksToRemove->add(track); 2830 } 2831 // Avoids a misleading display in dumpsys 2832 track->mObservedUnderruns &= ~1; 2833 } 2834 continue; 2835 } 2836 2837 { // local variable scope to avoid goto warning 2838 2839 audio_track_cblk_t* cblk = track->cblk(); 2840 2841 // The first time a track is added we wait 2842 // for all its buffers to be filled before processing it 2843 int name = track->name(); 2844 // make sure that we have enough frames to mix one full buffer. 2845 // enforce this condition only once to enable draining the buffer in case the client 2846 // app does not call stop() and relies on underrun to stop: 2847 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2848 // during last round 2849 uint32_t minFrames = 1; 2850 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2851 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2852 if (t->sampleRate() == (int)mSampleRate) { 2853 minFrames = mNormalFrameCount; 2854 } else { 2855 // +1 for rounding and +1 for additional sample needed for interpolation 2856 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2857 // add frames already consumed but not yet released by the resampler 2858 // because cblk->framesReady() will include these frames 2859 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2860 // the minimum track buffer size is normally twice the number of frames necessary 2861 // to fill one buffer and the resampler should not leave more than one buffer worth 2862 // of unreleased frames after each pass, but just in case... 2863 ALOG_ASSERT(minFrames <= cblk->frameCount); 2864 } 2865 } 2866 if ((track->framesReady() >= minFrames) && track->isReady() && 2867 !track->isPaused() && !track->isTerminated()) 2868 { 2869 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2870 2871 mixedTracks++; 2872 2873 // track->mainBuffer() != mMixBuffer means there is an effect chain 2874 // connected to the track 2875 chain.clear(); 2876 if (track->mainBuffer() != mMixBuffer) { 2877 chain = getEffectChain_l(track->sessionId()); 2878 // Delegate volume control to effect in track effect chain if needed 2879 if (chain != 0) { 2880 tracksWithEffect++; 2881 } else { 2882 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2883 name, track->sessionId()); 2884 } 2885 } 2886 2887 2888 int param = AudioMixer::VOLUME; 2889 if (track->mFillingUpStatus == Track::FS_FILLED) { 2890 // no ramp for the first volume setting 2891 track->mFillingUpStatus = Track::FS_ACTIVE; 2892 if (track->mState == TrackBase::RESUMING) { 2893 track->mState = TrackBase::ACTIVE; 2894 param = AudioMixer::RAMP_VOLUME; 2895 } 2896 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2897 } else if (cblk->server != 0) { 2898 // If the track is stopped before the first frame was mixed, 2899 // do not apply ramp 2900 param = AudioMixer::RAMP_VOLUME; 2901 } 2902 2903 // compute volume for this track 2904 uint32_t vl, vr, va; 2905 if (track->isMuted() || track->isPausing() || 2906 mStreamTypes[track->streamType()].mute) { 2907 vl = vr = va = 0; 2908 if (track->isPausing()) { 2909 track->setPaused(); 2910 } 2911 } else { 2912 2913 // read original volumes with volume control 2914 float typeVolume = mStreamTypes[track->streamType()].volume; 2915 float v = masterVolume * typeVolume; 2916 uint32_t vlr = cblk->getVolumeLR(); 2917 vl = vlr & 0xFFFF; 2918 vr = vlr >> 16; 2919 // track volumes come from shared memory, so can't be trusted and must be clamped 2920 if (vl > MAX_GAIN_INT) { 2921 ALOGV("Track left volume out of range: %04X", vl); 2922 vl = MAX_GAIN_INT; 2923 } 2924 if (vr > MAX_GAIN_INT) { 2925 ALOGV("Track right volume out of range: %04X", vr); 2926 vr = MAX_GAIN_INT; 2927 } 2928 // now apply the master volume and stream type volume 2929 vl = (uint32_t)(v * vl) << 12; 2930 vr = (uint32_t)(v * vr) << 12; 2931 // assuming master volume and stream type volume each go up to 1.0, 2932 // vl and vr are now in 8.24 format 2933 2934 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2935 // send level comes from shared memory and so may be corrupt 2936 if (sendLevel > MAX_GAIN_INT) { 2937 ALOGV("Track send level out of range: %04X", sendLevel); 2938 sendLevel = MAX_GAIN_INT; 2939 } 2940 va = (uint32_t)(v * sendLevel); 2941 } 2942 // Delegate volume control to effect in track effect chain if needed 2943 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2944 // Do not ramp volume if volume is controlled by effect 2945 param = AudioMixer::VOLUME; 2946 track->mHasVolumeController = true; 2947 } else { 2948 // force no volume ramp when volume controller was just disabled or removed 2949 // from effect chain to avoid volume spike 2950 if (track->mHasVolumeController) { 2951 param = AudioMixer::VOLUME; 2952 } 2953 track->mHasVolumeController = false; 2954 } 2955 2956 // Convert volumes from 8.24 to 4.12 format 2957 // This additional clamping is needed in case chain->setVolume_l() overshot 2958 vl = (vl + (1 << 11)) >> 12; 2959 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2960 vr = (vr + (1 << 11)) >> 12; 2961 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2962 2963 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2964 2965 // XXX: these things DON'T need to be done each time 2966 mAudioMixer->setBufferProvider(name, track); 2967 mAudioMixer->enable(name); 2968 2969 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2970 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2971 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2972 mAudioMixer->setParameter( 2973 name, 2974 AudioMixer::TRACK, 2975 AudioMixer::FORMAT, (void *)track->format()); 2976 mAudioMixer->setParameter( 2977 name, 2978 AudioMixer::TRACK, 2979 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2980 mAudioMixer->setParameter( 2981 name, 2982 AudioMixer::RESAMPLE, 2983 AudioMixer::SAMPLE_RATE, 2984 (void *)(cblk->sampleRate)); 2985 mAudioMixer->setParameter( 2986 name, 2987 AudioMixer::TRACK, 2988 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2989 mAudioMixer->setParameter( 2990 name, 2991 AudioMixer::TRACK, 2992 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2993 2994 // reset retry count 2995 track->mRetryCount = kMaxTrackRetries; 2996 2997 // If one track is ready, set the mixer ready if: 2998 // - the mixer was not ready during previous round OR 2999 // - no other track is not ready 3000 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3001 mixerStatus != MIXER_TRACKS_ENABLED) { 3002 mixerStatus = MIXER_TRACKS_READY; 3003 } 3004 } else { 3005 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3006 if (track->isStopped()) { 3007 track->reset(); 3008 } 3009 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3010 track->isStopped() || track->isPaused()) { 3011 // We have consumed all the buffers of this track. 3012 // Remove it from the list of active tracks. 3013 // TODO: use actual buffer filling status instead of latency when available from 3014 // audio HAL 3015 size_t audioHALFrames = 3016 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3017 size_t framesWritten = 3018 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3019 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3020 tracksToRemove->add(track); 3021 } 3022 } else { 3023 // No buffers for this track. Give it a few chances to 3024 // fill a buffer, then remove it from active list. 3025 if (--(track->mRetryCount) <= 0) { 3026 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3027 tracksToRemove->add(track); 3028 // indicate to client process that the track was disabled because of underrun; 3029 // it will then automatically call start() when data is available 3030 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3031 // If one track is not ready, mark the mixer also not ready if: 3032 // - the mixer was ready during previous round OR 3033 // - no other track is ready 3034 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3035 mixerStatus != MIXER_TRACKS_READY) { 3036 mixerStatus = MIXER_TRACKS_ENABLED; 3037 } 3038 } 3039 mAudioMixer->disable(name); 3040 } 3041 3042 } // local variable scope to avoid goto warning 3043track_is_ready: ; 3044 3045 } 3046 3047 // Push the new FastMixer state if necessary 3048 if (didModify) { 3049 state->mFastTracksGen++; 3050 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3051 if (kUseFastMixer == FastMixer_Dynamic && 3052 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3053 state->mCommand = FastMixerState::COLD_IDLE; 3054 state->mColdFutexAddr = &mFastMixerFutex; 3055 state->mColdGen++; 3056 mFastMixerFutex = 0; 3057 if (kUseFastMixer == FastMixer_Dynamic) { 3058 mNormalSink = mOutputSink; 3059 } 3060 // If we go into cold idle, need to wait for acknowledgement 3061 // so that fast mixer stops doing I/O. 3062 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3063 } 3064 sq->end(); 3065 } 3066 if (sq != NULL) { 3067 sq->end(didModify); 3068 sq->push(block); 3069 } 3070 3071 // Now perform the deferred reset on fast tracks that have stopped 3072 while (resetMask != 0) { 3073 size_t i = __builtin_ctz(resetMask); 3074 ALOG_ASSERT(i < count); 3075 resetMask &= ~(1 << i); 3076 sp<Track> t = mActiveTracks[i].promote(); 3077 if (t == 0) continue; 3078 Track* track = t.get(); 3079 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3080 track->reset(); 3081 } 3082 3083 // remove all the tracks that need to be... 3084 count = tracksToRemove->size(); 3085 if (CC_UNLIKELY(count)) { 3086 for (size_t i=0 ; i<count ; i++) { 3087 const sp<Track>& track = tracksToRemove->itemAt(i); 3088 mActiveTracks.remove(track); 3089 if (track->mainBuffer() != mMixBuffer) { 3090 chain = getEffectChain_l(track->sessionId()); 3091 if (chain != 0) { 3092 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3093 chain->decActiveTrackCnt(); 3094 } 3095 } 3096 if (track->isTerminated()) { 3097 removeTrack_l(track); 3098 } 3099 } 3100 } 3101 3102 // mix buffer must be cleared if all tracks are connected to an 3103 // effect chain as in this case the mixer will not write to 3104 // mix buffer and track effects will accumulate into it 3105 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3106 // FIXME as a performance optimization, should remember previous zero status 3107 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3108 } 3109 3110 // if any fast tracks, then status is ready 3111 mMixerStatusIgnoringFastTracks = mixerStatus; 3112 if (fastTracks > 0) { 3113 mixerStatus = MIXER_TRACKS_READY; 3114 } 3115 return mixerStatus; 3116} 3117 3118/* 3119The derived values that are cached: 3120 - mixBufferSize from frame count * frame size 3121 - activeSleepTime from activeSleepTimeUs() 3122 - idleSleepTime from idleSleepTimeUs() 3123 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3124 - maxPeriod from frame count and sample rate (MIXER only) 3125 3126The parameters that affect these derived values are: 3127 - frame count 3128 - frame size 3129 - sample rate 3130 - device type: A2DP or not 3131 - device latency 3132 - format: PCM or not 3133 - active sleep time 3134 - idle sleep time 3135*/ 3136 3137void AudioFlinger::PlaybackThread::cacheParameters_l() 3138{ 3139 mixBufferSize = mNormalFrameCount * mFrameSize; 3140 activeSleepTime = activeSleepTimeUs(); 3141 idleSleepTime = idleSleepTimeUs(); 3142} 3143 3144void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3145{ 3146 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3147 this, streamType, mTracks.size()); 3148 Mutex::Autolock _l(mLock); 3149 3150 size_t size = mTracks.size(); 3151 for (size_t i = 0; i < size; i++) { 3152 sp<Track> t = mTracks[i]; 3153 if (t->streamType() == streamType) { 3154 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3155 t->mCblk->cv.signal(); 3156 } 3157 } 3158} 3159 3160// getTrackName_l() must be called with ThreadBase::mLock held 3161int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3162{ 3163 return mAudioMixer->getTrackName(channelMask); 3164} 3165 3166// deleteTrackName_l() must be called with ThreadBase::mLock held 3167void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3168{ 3169 ALOGV("remove track (%d) and delete from mixer", name); 3170 mAudioMixer->deleteTrackName(name); 3171} 3172 3173// checkForNewParameters_l() must be called with ThreadBase::mLock held 3174bool AudioFlinger::MixerThread::checkForNewParameters_l() 3175{ 3176 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3177 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3178 bool reconfig = false; 3179 3180 while (!mNewParameters.isEmpty()) { 3181 3182 if (mFastMixer != NULL) { 3183 FastMixerStateQueue *sq = mFastMixer->sq(); 3184 FastMixerState *state = sq->begin(); 3185 if (!(state->mCommand & FastMixerState::IDLE)) { 3186 previousCommand = state->mCommand; 3187 state->mCommand = FastMixerState::HOT_IDLE; 3188 sq->end(); 3189 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3190 } else { 3191 sq->end(false /*didModify*/); 3192 } 3193 } 3194 3195 status_t status = NO_ERROR; 3196 String8 keyValuePair = mNewParameters[0]; 3197 AudioParameter param = AudioParameter(keyValuePair); 3198 int value; 3199 3200 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3201 reconfig = true; 3202 } 3203 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3204 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3205 status = BAD_VALUE; 3206 } else { 3207 reconfig = true; 3208 } 3209 } 3210 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3211 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3212 status = BAD_VALUE; 3213 } else { 3214 reconfig = true; 3215 } 3216 } 3217 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3218 // do not accept frame count changes if tracks are open as the track buffer 3219 // size depends on frame count and correct behavior would not be guaranteed 3220 // if frame count is changed after track creation 3221 if (!mTracks.isEmpty()) { 3222 status = INVALID_OPERATION; 3223 } else { 3224 reconfig = true; 3225 } 3226 } 3227 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3228#ifdef ADD_BATTERY_DATA 3229 // when changing the audio output device, call addBatteryData to notify 3230 // the change 3231 if ((int)mDevice != value) { 3232 uint32_t params = 0; 3233 // check whether speaker is on 3234 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3235 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3236 } 3237 3238 int deviceWithoutSpeaker 3239 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3240 // check if any other device (except speaker) is on 3241 if (value & deviceWithoutSpeaker ) { 3242 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3243 } 3244 3245 if (params != 0) { 3246 addBatteryData(params); 3247 } 3248 } 3249#endif 3250 3251 // forward device change to effects that have requested to be 3252 // aware of attached audio device. 3253 mDevice = (uint32_t)value; 3254 for (size_t i = 0; i < mEffectChains.size(); i++) { 3255 mEffectChains[i]->setDevice_l(mDevice); 3256 } 3257 } 3258 3259 if (status == NO_ERROR) { 3260 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3261 keyValuePair.string()); 3262 if (!mStandby && status == INVALID_OPERATION) { 3263 mOutput->stream->common.standby(&mOutput->stream->common); 3264 mStandby = true; 3265 mBytesWritten = 0; 3266 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3267 keyValuePair.string()); 3268 } 3269 if (status == NO_ERROR && reconfig) { 3270 delete mAudioMixer; 3271 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3272 mAudioMixer = NULL; 3273 readOutputParameters(); 3274 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3275 for (size_t i = 0; i < mTracks.size() ; i++) { 3276 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3277 if (name < 0) break; 3278 mTracks[i]->mName = name; 3279 // limit track sample rate to 2 x new output sample rate 3280 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3281 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3282 } 3283 } 3284 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3285 } 3286 } 3287 3288 mNewParameters.removeAt(0); 3289 3290 mParamStatus = status; 3291 mParamCond.signal(); 3292 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3293 // already timed out waiting for the status and will never signal the condition. 3294 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3295 } 3296 3297 if (!(previousCommand & FastMixerState::IDLE)) { 3298 ALOG_ASSERT(mFastMixer != NULL); 3299 FastMixerStateQueue *sq = mFastMixer->sq(); 3300 FastMixerState *state = sq->begin(); 3301 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3302 state->mCommand = previousCommand; 3303 sq->end(); 3304 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3305 } 3306 3307 return reconfig; 3308} 3309 3310status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3311{ 3312 const size_t SIZE = 256; 3313 char buffer[SIZE]; 3314 String8 result; 3315 3316 PlaybackThread::dumpInternals(fd, args); 3317 3318 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3319 result.append(buffer); 3320 write(fd, result.string(), result.size()); 3321 3322 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3323 FastMixerDumpState copy = mFastMixerDumpState; 3324 copy.dump(fd); 3325 3326 return NO_ERROR; 3327} 3328 3329uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3330{ 3331 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3332} 3333 3334uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3335{ 3336 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3337} 3338 3339void AudioFlinger::MixerThread::cacheParameters_l() 3340{ 3341 PlaybackThread::cacheParameters_l(); 3342 3343 // FIXME: Relaxed timing because of a certain device that can't meet latency 3344 // Should be reduced to 2x after the vendor fixes the driver issue 3345 // increase threshold again due to low power audio mode. The way this warning 3346 // threshold is calculated and its usefulness should be reconsidered anyway. 3347 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3348} 3349 3350// ---------------------------------------------------------------------------- 3351AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3352 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3353 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3354 // mLeftVolFloat, mRightVolFloat 3355 // mLeftVolShort, mRightVolShort 3356{ 3357} 3358 3359AudioFlinger::DirectOutputThread::~DirectOutputThread() 3360{ 3361} 3362 3363AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3364 Vector< sp<Track> > *tracksToRemove 3365) 3366{ 3367 sp<Track> trackToRemove; 3368 3369 mixer_state mixerStatus = MIXER_IDLE; 3370 3371 // find out which tracks need to be processed 3372 if (mActiveTracks.size() != 0) { 3373 sp<Track> t = mActiveTracks[0].promote(); 3374 // The track died recently 3375 if (t == 0) return MIXER_IDLE; 3376 3377 Track* const track = t.get(); 3378 audio_track_cblk_t* cblk = track->cblk(); 3379 3380 // The first time a track is added we wait 3381 // for all its buffers to be filled before processing it 3382 if (cblk->framesReady() && track->isReady() && 3383 !track->isPaused() && !track->isTerminated()) 3384 { 3385 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3386 3387 if (track->mFillingUpStatus == Track::FS_FILLED) { 3388 track->mFillingUpStatus = Track::FS_ACTIVE; 3389 mLeftVolFloat = mRightVolFloat = 0; 3390 mLeftVolShort = mRightVolShort = 0; 3391 if (track->mState == TrackBase::RESUMING) { 3392 track->mState = TrackBase::ACTIVE; 3393 rampVolume = true; 3394 } 3395 } else if (cblk->server != 0) { 3396 // If the track is stopped before the first frame was mixed, 3397 // do not apply ramp 3398 rampVolume = true; 3399 } 3400 // compute volume for this track 3401 float left, right; 3402 if (track->isMuted() || mMasterMute || track->isPausing() || 3403 mStreamTypes[track->streamType()].mute) { 3404 left = right = 0; 3405 if (track->isPausing()) { 3406 track->setPaused(); 3407 } 3408 } else { 3409 float typeVolume = mStreamTypes[track->streamType()].volume; 3410 float v = mMasterVolume * typeVolume; 3411 uint32_t vlr = cblk->getVolumeLR(); 3412 float v_clamped = v * (vlr & 0xFFFF); 3413 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3414 left = v_clamped/MAX_GAIN; 3415 v_clamped = v * (vlr >> 16); 3416 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3417 right = v_clamped/MAX_GAIN; 3418 } 3419 3420 if (left != mLeftVolFloat || right != mRightVolFloat) { 3421 mLeftVolFloat = left; 3422 mRightVolFloat = right; 3423 3424 // If audio HAL implements volume control, 3425 // force software volume to nominal value 3426 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3427 left = 1.0f; 3428 right = 1.0f; 3429 } 3430 3431 // Convert volumes from float to 8.24 3432 uint32_t vl = (uint32_t)(left * (1 << 24)); 3433 uint32_t vr = (uint32_t)(right * (1 << 24)); 3434 3435 // Delegate volume control to effect in track effect chain if needed 3436 // only one effect chain can be present on DirectOutputThread, so if 3437 // there is one, the track is connected to it 3438 if (!mEffectChains.isEmpty()) { 3439 // Do not ramp volume if volume is controlled by effect 3440 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3441 rampVolume = false; 3442 } 3443 } 3444 3445 // Convert volumes from 8.24 to 4.12 format 3446 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3447 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3448 leftVol = (uint16_t)v_clamped; 3449 v_clamped = (vr + (1 << 11)) >> 12; 3450 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3451 rightVol = (uint16_t)v_clamped; 3452 } else { 3453 leftVol = mLeftVolShort; 3454 rightVol = mRightVolShort; 3455 rampVolume = false; 3456 } 3457 3458 // reset retry count 3459 track->mRetryCount = kMaxTrackRetriesDirect; 3460 mActiveTrack = t; 3461 mixerStatus = MIXER_TRACKS_READY; 3462 } else { 3463 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3464 if (track->isStopped()) { 3465 track->reset(); 3466 } 3467 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3468 // We have consumed all the buffers of this track. 3469 // Remove it from the list of active tracks. 3470 // TODO: implement behavior for compressed audio 3471 size_t audioHALFrames = 3472 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3473 size_t framesWritten = 3474 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3475 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3476 trackToRemove = track; 3477 } 3478 } else { 3479 // No buffers for this track. Give it a few chances to 3480 // fill a buffer, then remove it from active list. 3481 if (--(track->mRetryCount) <= 0) { 3482 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3483 trackToRemove = track; 3484 } else { 3485 mixerStatus = MIXER_TRACKS_ENABLED; 3486 } 3487 } 3488 } 3489 } 3490 3491 // FIXME merge this with similar code for removing multiple tracks 3492 // remove all the tracks that need to be... 3493 if (CC_UNLIKELY(trackToRemove != 0)) { 3494 tracksToRemove->add(trackToRemove); 3495 mActiveTracks.remove(trackToRemove); 3496 if (!mEffectChains.isEmpty()) { 3497 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3498 trackToRemove->sessionId()); 3499 mEffectChains[0]->decActiveTrackCnt(); 3500 } 3501 if (trackToRemove->isTerminated()) { 3502 removeTrack_l(trackToRemove); 3503 } 3504 } 3505 3506 return mixerStatus; 3507} 3508 3509void AudioFlinger::DirectOutputThread::threadLoop_mix() 3510{ 3511 AudioBufferProvider::Buffer buffer; 3512 size_t frameCount = mFrameCount; 3513 int8_t *curBuf = (int8_t *)mMixBuffer; 3514 // output audio to hardware 3515 while (frameCount) { 3516 buffer.frameCount = frameCount; 3517 mActiveTrack->getNextBuffer(&buffer); 3518 if (CC_UNLIKELY(buffer.raw == NULL)) { 3519 memset(curBuf, 0, frameCount * mFrameSize); 3520 break; 3521 } 3522 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3523 frameCount -= buffer.frameCount; 3524 curBuf += buffer.frameCount * mFrameSize; 3525 mActiveTrack->releaseBuffer(&buffer); 3526 } 3527 sleepTime = 0; 3528 standbyTime = systemTime() + standbyDelay; 3529 mActiveTrack.clear(); 3530 3531 // apply volume 3532 3533 // Do not apply volume on compressed audio 3534 if (!audio_is_linear_pcm(mFormat)) { 3535 return; 3536 } 3537 3538 // convert to signed 16 bit before volume calculation 3539 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3540 size_t count = mFrameCount * mChannelCount; 3541 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3542 int16_t *dst = mMixBuffer + count-1; 3543 while (count--) { 3544 *dst-- = (int16_t)(*src--^0x80) << 8; 3545 } 3546 } 3547 3548 frameCount = mFrameCount; 3549 int16_t *out = mMixBuffer; 3550 if (rampVolume) { 3551 if (mChannelCount == 1) { 3552 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3553 int32_t vlInc = d / (int32_t)frameCount; 3554 int32_t vl = ((int32_t)mLeftVolShort << 16); 3555 do { 3556 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3557 out++; 3558 vl += vlInc; 3559 } while (--frameCount); 3560 3561 } else { 3562 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3563 int32_t vlInc = d / (int32_t)frameCount; 3564 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3565 int32_t vrInc = d / (int32_t)frameCount; 3566 int32_t vl = ((int32_t)mLeftVolShort << 16); 3567 int32_t vr = ((int32_t)mRightVolShort << 16); 3568 do { 3569 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3570 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3571 out += 2; 3572 vl += vlInc; 3573 vr += vrInc; 3574 } while (--frameCount); 3575 } 3576 } else { 3577 if (mChannelCount == 1) { 3578 do { 3579 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3580 out++; 3581 } while (--frameCount); 3582 } else { 3583 do { 3584 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3585 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3586 out += 2; 3587 } while (--frameCount); 3588 } 3589 } 3590 3591 // convert back to unsigned 8 bit after volume calculation 3592 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3593 size_t count = mFrameCount * mChannelCount; 3594 int16_t *src = mMixBuffer; 3595 uint8_t *dst = (uint8_t *)mMixBuffer; 3596 while (count--) { 3597 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3598 } 3599 } 3600 3601 mLeftVolShort = leftVol; 3602 mRightVolShort = rightVol; 3603} 3604 3605void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3606{ 3607 if (sleepTime == 0) { 3608 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3609 sleepTime = activeSleepTime; 3610 } else { 3611 sleepTime = idleSleepTime; 3612 } 3613 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3614 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3615 sleepTime = 0; 3616 } 3617} 3618 3619// getTrackName_l() must be called with ThreadBase::mLock held 3620int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3621{ 3622 return 0; 3623} 3624 3625// deleteTrackName_l() must be called with ThreadBase::mLock held 3626void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3627{ 3628} 3629 3630// checkForNewParameters_l() must be called with ThreadBase::mLock held 3631bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3632{ 3633 bool reconfig = false; 3634 3635 while (!mNewParameters.isEmpty()) { 3636 status_t status = NO_ERROR; 3637 String8 keyValuePair = mNewParameters[0]; 3638 AudioParameter param = AudioParameter(keyValuePair); 3639 int value; 3640 3641 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3642 // do not accept frame count changes if tracks are open as the track buffer 3643 // size depends on frame count and correct behavior would not be garantied 3644 // if frame count is changed after track creation 3645 if (!mTracks.isEmpty()) { 3646 status = INVALID_OPERATION; 3647 } else { 3648 reconfig = true; 3649 } 3650 } 3651 if (status == NO_ERROR) { 3652 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3653 keyValuePair.string()); 3654 if (!mStandby && status == INVALID_OPERATION) { 3655 mOutput->stream->common.standby(&mOutput->stream->common); 3656 mStandby = true; 3657 mBytesWritten = 0; 3658 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3659 keyValuePair.string()); 3660 } 3661 if (status == NO_ERROR && reconfig) { 3662 readOutputParameters(); 3663 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3664 } 3665 } 3666 3667 mNewParameters.removeAt(0); 3668 3669 mParamStatus = status; 3670 mParamCond.signal(); 3671 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3672 // already timed out waiting for the status and will never signal the condition. 3673 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3674 } 3675 return reconfig; 3676} 3677 3678uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3679{ 3680 uint32_t time; 3681 if (audio_is_linear_pcm(mFormat)) { 3682 time = PlaybackThread::activeSleepTimeUs(); 3683 } else { 3684 time = 10000; 3685 } 3686 return time; 3687} 3688 3689uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3690{ 3691 uint32_t time; 3692 if (audio_is_linear_pcm(mFormat)) { 3693 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3694 } else { 3695 time = 10000; 3696 } 3697 return time; 3698} 3699 3700uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3701{ 3702 uint32_t time; 3703 if (audio_is_linear_pcm(mFormat)) { 3704 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3705 } else { 3706 time = 10000; 3707 } 3708 return time; 3709} 3710 3711void AudioFlinger::DirectOutputThread::cacheParameters_l() 3712{ 3713 PlaybackThread::cacheParameters_l(); 3714 3715 // use shorter standby delay as on normal output to release 3716 // hardware resources as soon as possible 3717 standbyDelay = microseconds(activeSleepTime*2); 3718} 3719 3720// ---------------------------------------------------------------------------- 3721 3722AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3723 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3724 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3725 mWaitTimeMs(UINT_MAX) 3726{ 3727 addOutputTrack(mainThread); 3728} 3729 3730AudioFlinger::DuplicatingThread::~DuplicatingThread() 3731{ 3732 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3733 mOutputTracks[i]->destroy(); 3734 } 3735} 3736 3737void AudioFlinger::DuplicatingThread::threadLoop_mix() 3738{ 3739 // mix buffers... 3740 if (outputsReady(outputTracks)) { 3741 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3742 } else { 3743 memset(mMixBuffer, 0, mixBufferSize); 3744 } 3745 sleepTime = 0; 3746 writeFrames = mNormalFrameCount; 3747} 3748 3749void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3750{ 3751 if (sleepTime == 0) { 3752 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3753 sleepTime = activeSleepTime; 3754 } else { 3755 sleepTime = idleSleepTime; 3756 } 3757 } else if (mBytesWritten != 0) { 3758 // flush remaining overflow buffers in output tracks 3759 for (size_t i = 0; i < outputTracks.size(); i++) { 3760 if (outputTracks[i]->isActive()) { 3761 sleepTime = 0; 3762 writeFrames = 0; 3763 memset(mMixBuffer, 0, mixBufferSize); 3764 break; 3765 } 3766 } 3767 } 3768} 3769 3770void AudioFlinger::DuplicatingThread::threadLoop_write() 3771{ 3772 standbyTime = systemTime() + standbyDelay; 3773 for (size_t i = 0; i < outputTracks.size(); i++) { 3774 outputTracks[i]->write(mMixBuffer, writeFrames); 3775 } 3776 mBytesWritten += mixBufferSize; 3777} 3778 3779void AudioFlinger::DuplicatingThread::threadLoop_standby() 3780{ 3781 // DuplicatingThread implements standby by stopping all tracks 3782 for (size_t i = 0; i < outputTracks.size(); i++) { 3783 outputTracks[i]->stop(); 3784 } 3785} 3786 3787void AudioFlinger::DuplicatingThread::saveOutputTracks() 3788{ 3789 outputTracks = mOutputTracks; 3790} 3791 3792void AudioFlinger::DuplicatingThread::clearOutputTracks() 3793{ 3794 outputTracks.clear(); 3795} 3796 3797void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3798{ 3799 Mutex::Autolock _l(mLock); 3800 // FIXME explain this formula 3801 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3802 OutputTrack *outputTrack = new OutputTrack(thread, 3803 this, 3804 mSampleRate, 3805 mFormat, 3806 mChannelMask, 3807 frameCount); 3808 if (outputTrack->cblk() != NULL) { 3809 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3810 mOutputTracks.add(outputTrack); 3811 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3812 updateWaitTime_l(); 3813 } 3814} 3815 3816void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3817{ 3818 Mutex::Autolock _l(mLock); 3819 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3820 if (mOutputTracks[i]->thread() == thread) { 3821 mOutputTracks[i]->destroy(); 3822 mOutputTracks.removeAt(i); 3823 updateWaitTime_l(); 3824 return; 3825 } 3826 } 3827 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3828} 3829 3830// caller must hold mLock 3831void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3832{ 3833 mWaitTimeMs = UINT_MAX; 3834 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3835 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3836 if (strong != 0) { 3837 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3838 if (waitTimeMs < mWaitTimeMs) { 3839 mWaitTimeMs = waitTimeMs; 3840 } 3841 } 3842 } 3843} 3844 3845 3846bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3847{ 3848 for (size_t i = 0; i < outputTracks.size(); i++) { 3849 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3850 if (thread == 0) { 3851 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3852 return false; 3853 } 3854 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3855 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3856 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3857 return false; 3858 } 3859 } 3860 return true; 3861} 3862 3863uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3864{ 3865 return (mWaitTimeMs * 1000) / 2; 3866} 3867 3868void AudioFlinger::DuplicatingThread::cacheParameters_l() 3869{ 3870 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3871 updateWaitTime_l(); 3872 3873 MixerThread::cacheParameters_l(); 3874} 3875 3876// ---------------------------------------------------------------------------- 3877 3878// TrackBase constructor must be called with AudioFlinger::mLock held 3879AudioFlinger::ThreadBase::TrackBase::TrackBase( 3880 ThreadBase *thread, 3881 const sp<Client>& client, 3882 uint32_t sampleRate, 3883 audio_format_t format, 3884 uint32_t channelMask, 3885 int frameCount, 3886 const sp<IMemory>& sharedBuffer, 3887 int sessionId) 3888 : RefBase(), 3889 mThread(thread), 3890 mClient(client), 3891 mCblk(NULL), 3892 // mBuffer 3893 // mBufferEnd 3894 mFrameCount(0), 3895 mState(IDLE), 3896 mSampleRate(sampleRate), 3897 mFormat(format), 3898 mStepServerFailed(false), 3899 mSessionId(sessionId) 3900 // mChannelCount 3901 // mChannelMask 3902{ 3903 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3904 3905 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3906 size_t size = sizeof(audio_track_cblk_t); 3907 uint8_t channelCount = popcount(channelMask); 3908 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3909 if (sharedBuffer == 0) { 3910 size += bufferSize; 3911 } 3912 3913 if (client != NULL) { 3914 mCblkMemory = client->heap()->allocate(size); 3915 if (mCblkMemory != 0) { 3916 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3917 if (mCblk != NULL) { // construct the shared structure in-place. 3918 new(mCblk) audio_track_cblk_t(); 3919 // clear all buffers 3920 mCblk->frameCount = frameCount; 3921 mCblk->sampleRate = sampleRate; 3922// uncomment the following lines to quickly test 32-bit wraparound 3923// mCblk->user = 0xffff0000; 3924// mCblk->server = 0xffff0000; 3925// mCblk->userBase = 0xffff0000; 3926// mCblk->serverBase = 0xffff0000; 3927 mChannelCount = channelCount; 3928 mChannelMask = channelMask; 3929 if (sharedBuffer == 0) { 3930 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3931 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3932 // Force underrun condition to avoid false underrun callback until first data is 3933 // written to buffer (other flags are cleared) 3934 mCblk->flags = CBLK_UNDERRUN_ON; 3935 } else { 3936 mBuffer = sharedBuffer->pointer(); 3937 } 3938 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3939 } 3940 } else { 3941 ALOGE("not enough memory for AudioTrack size=%u", size); 3942 client->heap()->dump("AudioTrack"); 3943 return; 3944 } 3945 } else { 3946 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3947 // construct the shared structure in-place. 3948 new(mCblk) audio_track_cblk_t(); 3949 // clear all buffers 3950 mCblk->frameCount = frameCount; 3951 mCblk->sampleRate = sampleRate; 3952// uncomment the following lines to quickly test 32-bit wraparound 3953// mCblk->user = 0xffff0000; 3954// mCblk->server = 0xffff0000; 3955// mCblk->userBase = 0xffff0000; 3956// mCblk->serverBase = 0xffff0000; 3957 mChannelCount = channelCount; 3958 mChannelMask = channelMask; 3959 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3960 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3961 // Force underrun condition to avoid false underrun callback until first data is 3962 // written to buffer (other flags are cleared) 3963 mCblk->flags = CBLK_UNDERRUN_ON; 3964 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3965 } 3966} 3967 3968AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3969{ 3970 if (mCblk != NULL) { 3971 if (mClient == 0) { 3972 delete mCblk; 3973 } else { 3974 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3975 } 3976 } 3977 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3978 if (mClient != 0) { 3979 // Client destructor must run with AudioFlinger mutex locked 3980 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3981 // If the client's reference count drops to zero, the associated destructor 3982 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3983 // relying on the automatic clear() at end of scope. 3984 mClient.clear(); 3985 } 3986} 3987 3988// AudioBufferProvider interface 3989// getNextBuffer() = 0; 3990// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3991void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3992{ 3993 buffer->raw = NULL; 3994 mFrameCount = buffer->frameCount; 3995 // FIXME See note at getNextBuffer() 3996 (void) step(); // ignore return value of step() 3997 buffer->frameCount = 0; 3998} 3999 4000bool AudioFlinger::ThreadBase::TrackBase::step() { 4001 bool result; 4002 audio_track_cblk_t* cblk = this->cblk(); 4003 4004 result = cblk->stepServer(mFrameCount); 4005 if (!result) { 4006 ALOGV("stepServer failed acquiring cblk mutex"); 4007 mStepServerFailed = true; 4008 } 4009 return result; 4010} 4011 4012void AudioFlinger::ThreadBase::TrackBase::reset() { 4013 audio_track_cblk_t* cblk = this->cblk(); 4014 4015 cblk->user = 0; 4016 cblk->server = 0; 4017 cblk->userBase = 0; 4018 cblk->serverBase = 0; 4019 mStepServerFailed = false; 4020 ALOGV("TrackBase::reset"); 4021} 4022 4023int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4024 return (int)mCblk->sampleRate; 4025} 4026 4027void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4028 audio_track_cblk_t* cblk = this->cblk(); 4029 size_t frameSize = cblk->frameSize; 4030 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4031 int8_t *bufferEnd = bufferStart + frames * frameSize; 4032 4033 // Check validity of returned pointer in case the track control block would have been corrupted. 4034 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4035 "TrackBase::getBuffer buffer out of range:\n" 4036 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4037 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4038 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4039 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4040 4041 return bufferStart; 4042} 4043 4044status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4045{ 4046 mSyncEvents.add(event); 4047 return NO_ERROR; 4048} 4049 4050// ---------------------------------------------------------------------------- 4051 4052// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4053AudioFlinger::PlaybackThread::Track::Track( 4054 PlaybackThread *thread, 4055 const sp<Client>& client, 4056 audio_stream_type_t streamType, 4057 uint32_t sampleRate, 4058 audio_format_t format, 4059 uint32_t channelMask, 4060 int frameCount, 4061 const sp<IMemory>& sharedBuffer, 4062 int sessionId, 4063 IAudioFlinger::track_flags_t flags) 4064 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4065 mMute(false), 4066 mFillingUpStatus(FS_INVALID), 4067 // mRetryCount initialized later when needed 4068 mSharedBuffer(sharedBuffer), 4069 mStreamType(streamType), 4070 mName(-1), // see note below 4071 mMainBuffer(thread->mixBuffer()), 4072 mAuxBuffer(NULL), 4073 mAuxEffectId(0), mHasVolumeController(false), 4074 mPresentationCompleteFrames(0), 4075 mFlags(flags), 4076 mFastIndex(-1), 4077 mObservedUnderruns(0), 4078 mUnderrunCount(0), 4079 mCachedVolume(1.0) 4080{ 4081 if (mCblk != NULL) { 4082 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4083 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4084 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4085 if (flags & IAudioFlinger::TRACK_FAST) { 4086 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4087 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4088 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4089 ALOG_ASSERT(0 < i && i < FastMixerState::kMaxFastTracks); 4090 // FIXME This is too eager. We allocate a fast track index before the 4091 // fast track becomes active. Since fast tracks are a scarce resource, 4092 // this means we are potentially denying other more important fast tracks from 4093 // being created. It would be better to allocate the index dynamically. 4094 mFastIndex = i; 4095 // Read the initial underruns because this field is never cleared by the fast mixer 4096 mObservedUnderruns = thread->getFastTrackUnderruns(i) & ~1; 4097 thread->mFastTrackAvailMask &= ~(1 << i); 4098 } 4099 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4100 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4101 if (mName < 0) { 4102 ALOGE("no more track names available"); 4103 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4104 // then we leak a fast track index. Should swap these two sections, or better yet 4105 // only allocate a normal mixer name for normal tracks. 4106 } 4107 } 4108 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4109} 4110 4111AudioFlinger::PlaybackThread::Track::~Track() 4112{ 4113 ALOGV("PlaybackThread::Track destructor"); 4114 sp<ThreadBase> thread = mThread.promote(); 4115 if (thread != 0) { 4116 Mutex::Autolock _l(thread->mLock); 4117 mState = TERMINATED; 4118 } 4119} 4120 4121void AudioFlinger::PlaybackThread::Track::destroy() 4122{ 4123 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4124 // by removing it from mTracks vector, so there is a risk that this Tracks's 4125 // destructor is called. As the destructor needs to lock mLock, 4126 // we must acquire a strong reference on this Track before locking mLock 4127 // here so that the destructor is called only when exiting this function. 4128 // On the other hand, as long as Track::destroy() is only called by 4129 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4130 // this Track with its member mTrack. 4131 sp<Track> keep(this); 4132 { // scope for mLock 4133 sp<ThreadBase> thread = mThread.promote(); 4134 if (thread != 0) { 4135 if (!isOutputTrack()) { 4136 if (mState == ACTIVE || mState == RESUMING) { 4137 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4138 4139#ifdef ADD_BATTERY_DATA 4140 // to track the speaker usage 4141 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4142#endif 4143 } 4144 AudioSystem::releaseOutput(thread->id()); 4145 } 4146 Mutex::Autolock _l(thread->mLock); 4147 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4148 playbackThread->destroyTrack_l(this); 4149 } 4150 } 4151} 4152 4153/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4154{ 4155 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4156 " Server User Main buf Aux Buf Flags FastUnder\n"); 4157} 4158 4159void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4160{ 4161 uint32_t vlr = mCblk->getVolumeLR(); 4162 if (isFastTrack()) { 4163 sprintf(buffer, " F %2d", mFastIndex); 4164 } else { 4165 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4166 } 4167 track_state state = mState; 4168 char stateChar; 4169 switch (state) { 4170 case IDLE: 4171 stateChar = 'I'; 4172 break; 4173 case TERMINATED: 4174 stateChar = 'T'; 4175 break; 4176 case STOPPED: 4177 stateChar = 'S'; 4178 break; 4179 case RESUMING: 4180 stateChar = 'R'; 4181 break; 4182 case ACTIVE: 4183 stateChar = 'A'; 4184 break; 4185 case PAUSING: 4186 stateChar = 'p'; 4187 break; 4188 case PAUSED: 4189 stateChar = 'P'; 4190 break; 4191 default: 4192 stateChar = '?'; 4193 break; 4194 } 4195 bool nowInUnderrun = mObservedUnderruns & 1; 4196 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4197 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4198 (mClient == 0) ? getpid_cached : mClient->pid(), 4199 mStreamType, 4200 mFormat, 4201 mChannelMask, 4202 mSessionId, 4203 mFrameCount, 4204 mCblk->frameCount, 4205 stateChar, 4206 mMute, 4207 mFillingUpStatus, 4208 mCblk->sampleRate, 4209 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4210 20.0 * log10((vlr >> 16) / 4096.0), 4211 mCblk->server, 4212 mCblk->user, 4213 (int)mMainBuffer, 4214 (int)mAuxBuffer, 4215 mCblk->flags, 4216 mUnderrunCount, 4217 nowInUnderrun ? '*' : ' '); 4218} 4219 4220// AudioBufferProvider interface 4221status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4222 AudioBufferProvider::Buffer* buffer, int64_t pts) 4223{ 4224 audio_track_cblk_t* cblk = this->cblk(); 4225 uint32_t framesReady; 4226 uint32_t framesReq = buffer->frameCount; 4227 4228 // Check if last stepServer failed, try to step now 4229 if (mStepServerFailed) { 4230 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4231 // Since the fast mixer is higher priority than client callback thread, 4232 // it does not result in priority inversion for client. 4233 // But a non-blocking solution would be preferable to avoid 4234 // fast mixer being unable to tryLock(), and 4235 // to avoid the extra context switches if the client wakes up, 4236 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4237 if (!step()) goto getNextBuffer_exit; 4238 ALOGV("stepServer recovered"); 4239 mStepServerFailed = false; 4240 } 4241 4242 // FIXME Same as above 4243 framesReady = cblk->framesReady(); 4244 4245 if (CC_LIKELY(framesReady)) { 4246 uint32_t s = cblk->server; 4247 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4248 4249 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4250 if (framesReq > framesReady) { 4251 framesReq = framesReady; 4252 } 4253 if (framesReq > bufferEnd - s) { 4254 framesReq = bufferEnd - s; 4255 } 4256 4257 buffer->raw = getBuffer(s, framesReq); 4258 if (buffer->raw == NULL) goto getNextBuffer_exit; 4259 4260 buffer->frameCount = framesReq; 4261 return NO_ERROR; 4262 } 4263 4264getNextBuffer_exit: 4265 buffer->raw = NULL; 4266 buffer->frameCount = 0; 4267 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4268 return NOT_ENOUGH_DATA; 4269} 4270 4271// Note that framesReady() takes a mutex on the control block using tryLock(). 4272// This could result in priority inversion if framesReady() is called by the normal mixer, 4273// as the normal mixer thread runs at lower 4274// priority than the client's callback thread: there is a short window within framesReady() 4275// during which the normal mixer could be preempted, and the client callback would block. 4276// Another problem can occur if framesReady() is called by the fast mixer: 4277// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4278// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4279size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4280 return mCblk->framesReady(); 4281} 4282 4283// Don't call for fast tracks; the framesReady() could result in priority inversion 4284bool AudioFlinger::PlaybackThread::Track::isReady() const { 4285 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4286 4287 if (framesReady() >= mCblk->frameCount || 4288 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4289 mFillingUpStatus = FS_FILLED; 4290 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4291 return true; 4292 } 4293 return false; 4294} 4295 4296status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4297 int triggerSession) 4298{ 4299 status_t status = NO_ERROR; 4300 ALOGV("start(%d), calling pid %d session %d", 4301 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4302 4303 sp<ThreadBase> thread = mThread.promote(); 4304 if (thread != 0) { 4305 Mutex::Autolock _l(thread->mLock); 4306 track_state state = mState; 4307 // here the track could be either new, or restarted 4308 // in both cases "unstop" the track 4309 if (mState == PAUSED) { 4310 mState = TrackBase::RESUMING; 4311 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4312 } else { 4313 mState = TrackBase::ACTIVE; 4314 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4315 } 4316 4317 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4318 thread->mLock.unlock(); 4319 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4320 thread->mLock.lock(); 4321 4322#ifdef ADD_BATTERY_DATA 4323 // to track the speaker usage 4324 if (status == NO_ERROR) { 4325 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4326 } 4327#endif 4328 } 4329 if (status == NO_ERROR) { 4330 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4331 playbackThread->addTrack_l(this); 4332 } else { 4333 mState = state; 4334 } 4335 } else { 4336 status = BAD_VALUE; 4337 } 4338 return status; 4339} 4340 4341void AudioFlinger::PlaybackThread::Track::stop() 4342{ 4343 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4344 sp<ThreadBase> thread = mThread.promote(); 4345 if (thread != 0) { 4346 Mutex::Autolock _l(thread->mLock); 4347 track_state state = mState; 4348 if (mState > STOPPED) { 4349 mState = STOPPED; 4350 // If the track is not active (PAUSED and buffers full), flush buffers 4351 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4352 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4353 reset(); 4354 } 4355 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 4356 } 4357 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4358 thread->mLock.unlock(); 4359 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4360 thread->mLock.lock(); 4361 4362#ifdef ADD_BATTERY_DATA 4363 // to track the speaker usage 4364 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4365#endif 4366 } 4367 } 4368} 4369 4370void AudioFlinger::PlaybackThread::Track::pause() 4371{ 4372 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4373 sp<ThreadBase> thread = mThread.promote(); 4374 if (thread != 0) { 4375 Mutex::Autolock _l(thread->mLock); 4376 if (mState == ACTIVE || mState == RESUMING) { 4377 mState = PAUSING; 4378 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4379 if (!isOutputTrack()) { 4380 thread->mLock.unlock(); 4381 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4382 thread->mLock.lock(); 4383 4384#ifdef ADD_BATTERY_DATA 4385 // to track the speaker usage 4386 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4387#endif 4388 } 4389 } 4390 } 4391} 4392 4393void AudioFlinger::PlaybackThread::Track::flush() 4394{ 4395 ALOGV("flush(%d)", mName); 4396 sp<ThreadBase> thread = mThread.promote(); 4397 if (thread != 0) { 4398 Mutex::Autolock _l(thread->mLock); 4399 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 4400 return; 4401 } 4402 // No point remaining in PAUSED state after a flush => go to 4403 // STOPPED state 4404 mState = STOPPED; 4405 4406 // do not reset the track if it is still in the process of being stopped or paused. 4407 // this will be done by prepareTracks_l() when the track is stopped. 4408 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4409 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4410 reset(); 4411 } 4412 } 4413} 4414 4415void AudioFlinger::PlaybackThread::Track::reset() 4416{ 4417 // Do not reset twice to avoid discarding data written just after a flush and before 4418 // the audioflinger thread detects the track is stopped. 4419 if (!mResetDone) { 4420 TrackBase::reset(); 4421 // Force underrun condition to avoid false underrun callback until first data is 4422 // written to buffer 4423 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4424 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4425 mFillingUpStatus = FS_FILLING; 4426 mResetDone = true; 4427 mPresentationCompleteFrames = 0; 4428 } 4429} 4430 4431void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4432{ 4433 mMute = muted; 4434} 4435 4436status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4437{ 4438 status_t status = DEAD_OBJECT; 4439 sp<ThreadBase> thread = mThread.promote(); 4440 if (thread != 0) { 4441 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4442 status = playbackThread->attachAuxEffect(this, EffectId); 4443 } 4444 return status; 4445} 4446 4447void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4448{ 4449 mAuxEffectId = EffectId; 4450 mAuxBuffer = buffer; 4451} 4452 4453bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4454 size_t audioHalFrames) 4455{ 4456 // a track is considered presented when the total number of frames written to audio HAL 4457 // corresponds to the number of frames written when presentationComplete() is called for the 4458 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4459 if (mPresentationCompleteFrames == 0) { 4460 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4461 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4462 mPresentationCompleteFrames, audioHalFrames); 4463 } 4464 if (framesWritten >= mPresentationCompleteFrames) { 4465 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4466 mSessionId, framesWritten); 4467 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4468 mPresentationCompleteFrames = 0; 4469 return true; 4470 } 4471 return false; 4472} 4473 4474void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4475{ 4476 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4477 if (mSyncEvents[i]->type() == type) { 4478 mSyncEvents[i]->trigger(); 4479 mSyncEvents.removeAt(i); 4480 i--; 4481 } 4482 } 4483} 4484 4485// implement VolumeBufferProvider interface 4486 4487uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4488{ 4489 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4490 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4491 uint32_t vlr = mCblk->getVolumeLR(); 4492 uint32_t vl = vlr & 0xFFFF; 4493 uint32_t vr = vlr >> 16; 4494 // track volumes come from shared memory, so can't be trusted and must be clamped 4495 if (vl > MAX_GAIN_INT) { 4496 vl = MAX_GAIN_INT; 4497 } 4498 if (vr > MAX_GAIN_INT) { 4499 vr = MAX_GAIN_INT; 4500 } 4501 // now apply the cached master volume and stream type volume; 4502 // this is trusted but lacks any synchronization or barrier so may be stale 4503 float v = mCachedVolume; 4504 vl *= v; 4505 vr *= v; 4506 // re-combine into U4.16 4507 vlr = (vr << 16) | (vl & 0xFFFF); 4508 // FIXME look at mute, pause, and stop flags 4509 return vlr; 4510} 4511 4512// timed audio tracks 4513 4514sp<AudioFlinger::PlaybackThread::TimedTrack> 4515AudioFlinger::PlaybackThread::TimedTrack::create( 4516 PlaybackThread *thread, 4517 const sp<Client>& client, 4518 audio_stream_type_t streamType, 4519 uint32_t sampleRate, 4520 audio_format_t format, 4521 uint32_t channelMask, 4522 int frameCount, 4523 const sp<IMemory>& sharedBuffer, 4524 int sessionId) { 4525 if (!client->reserveTimedTrack()) 4526 return NULL; 4527 4528 return new TimedTrack( 4529 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4530 sharedBuffer, sessionId); 4531} 4532 4533AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4534 PlaybackThread *thread, 4535 const sp<Client>& client, 4536 audio_stream_type_t streamType, 4537 uint32_t sampleRate, 4538 audio_format_t format, 4539 uint32_t channelMask, 4540 int frameCount, 4541 const sp<IMemory>& sharedBuffer, 4542 int sessionId) 4543 : Track(thread, client, streamType, sampleRate, format, channelMask, 4544 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4545 mQueueHeadInFlight(false), 4546 mTrimQueueHeadOnRelease(false), 4547 mFramesPendingInQueue(0), 4548 mTimedSilenceBuffer(NULL), 4549 mTimedSilenceBufferSize(0), 4550 mTimedAudioOutputOnTime(false), 4551 mMediaTimeTransformValid(false) 4552{ 4553 LocalClock lc; 4554 mLocalTimeFreq = lc.getLocalFreq(); 4555 4556 mLocalTimeToSampleTransform.a_zero = 0; 4557 mLocalTimeToSampleTransform.b_zero = 0; 4558 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4559 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4560 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4561 &mLocalTimeToSampleTransform.a_to_b_denom); 4562 4563 mMediaTimeToSampleTransform.a_zero = 0; 4564 mMediaTimeToSampleTransform.b_zero = 0; 4565 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4566 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4567 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4568 &mMediaTimeToSampleTransform.a_to_b_denom); 4569} 4570 4571AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4572 mClient->releaseTimedTrack(); 4573 delete [] mTimedSilenceBuffer; 4574} 4575 4576status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4577 size_t size, sp<IMemory>* buffer) { 4578 4579 Mutex::Autolock _l(mTimedBufferQueueLock); 4580 4581 trimTimedBufferQueue_l(); 4582 4583 // lazily initialize the shared memory heap for timed buffers 4584 if (mTimedMemoryDealer == NULL) { 4585 const int kTimedBufferHeapSize = 512 << 10; 4586 4587 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4588 "AudioFlingerTimed"); 4589 if (mTimedMemoryDealer == NULL) 4590 return NO_MEMORY; 4591 } 4592 4593 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4594 if (newBuffer == NULL) { 4595 newBuffer = mTimedMemoryDealer->allocate(size); 4596 if (newBuffer == NULL) 4597 return NO_MEMORY; 4598 } 4599 4600 *buffer = newBuffer; 4601 return NO_ERROR; 4602} 4603 4604// caller must hold mTimedBufferQueueLock 4605void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4606 int64_t mediaTimeNow; 4607 { 4608 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4609 if (!mMediaTimeTransformValid) 4610 return; 4611 4612 int64_t targetTimeNow; 4613 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4614 ? mCCHelper.getCommonTime(&targetTimeNow) 4615 : mCCHelper.getLocalTime(&targetTimeNow); 4616 4617 if (OK != res) 4618 return; 4619 4620 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4621 &mediaTimeNow)) { 4622 return; 4623 } 4624 } 4625 4626 size_t trimEnd; 4627 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4628 int64_t bufEnd; 4629 4630 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4631 // We have a next buffer. Just use its PTS as the PTS of the frame 4632 // following the last frame in this buffer. If the stream is sparse 4633 // (ie, there are deliberate gaps left in the stream which should be 4634 // filled with silence by the TimedAudioTrack), then this can result 4635 // in one extra buffer being left un-trimmed when it could have 4636 // been. In general, this is not typical, and we would rather 4637 // optimized away the TS calculation below for the more common case 4638 // where PTSes are contiguous. 4639 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4640 } else { 4641 // We have no next buffer. Compute the PTS of the frame following 4642 // the last frame in this buffer by computing the duration of of 4643 // this frame in media time units and adding it to the PTS of the 4644 // buffer. 4645 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4646 / mCblk->frameSize; 4647 4648 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4649 &bufEnd)) { 4650 ALOGE("Failed to convert frame count of %lld to media time" 4651 " duration" " (scale factor %d/%u) in %s", 4652 frameCount, 4653 mMediaTimeToSampleTransform.a_to_b_numer, 4654 mMediaTimeToSampleTransform.a_to_b_denom, 4655 __PRETTY_FUNCTION__); 4656 break; 4657 } 4658 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4659 } 4660 4661 if (bufEnd > mediaTimeNow) 4662 break; 4663 4664 // Is the buffer we want to use in the middle of a mix operation right 4665 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4666 // from the mixer which should be coming back shortly. 4667 if (!trimEnd && mQueueHeadInFlight) { 4668 mTrimQueueHeadOnRelease = true; 4669 } 4670 } 4671 4672 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4673 if (trimStart < trimEnd) { 4674 // Update the bookkeeping for framesReady() 4675 for (size_t i = trimStart; i < trimEnd; ++i) { 4676 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4677 } 4678 4679 // Now actually remove the buffers from the queue. 4680 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4681 } 4682} 4683 4684void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4685 const char* logTag) { 4686 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4687 "%s called (reason \"%s\"), but timed buffer queue has no" 4688 " elements to trim.", __FUNCTION__, logTag); 4689 4690 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4691 mTimedBufferQueue.removeAt(0); 4692} 4693 4694void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4695 const TimedBuffer& buf, 4696 const char* logTag) { 4697 uint32_t bufBytes = buf.buffer()->size(); 4698 uint32_t consumedAlready = buf.position(); 4699 4700 ALOG_ASSERT(consumedAlready <= bufBytes, 4701 "Bad bookkeeping while updating frames pending. Timed buffer is" 4702 " only %u bytes long, but claims to have consumed %u" 4703 " bytes. (update reason: \"%s\")", 4704 bufBytes, consumedAlready, logTag); 4705 4706 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4707 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4708 "Bad bookkeeping while updating frames pending. Should have at" 4709 " least %u queued frames, but we think we have only %u. (update" 4710 " reason: \"%s\")", 4711 bufFrames, mFramesPendingInQueue, logTag); 4712 4713 mFramesPendingInQueue -= bufFrames; 4714} 4715 4716status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4717 const sp<IMemory>& buffer, int64_t pts) { 4718 4719 { 4720 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4721 if (!mMediaTimeTransformValid) 4722 return INVALID_OPERATION; 4723 } 4724 4725 Mutex::Autolock _l(mTimedBufferQueueLock); 4726 4727 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4728 mFramesPendingInQueue += bufFrames; 4729 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4730 4731 return NO_ERROR; 4732} 4733 4734status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4735 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4736 4737 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4738 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4739 target); 4740 4741 if (!(target == TimedAudioTrack::LOCAL_TIME || 4742 target == TimedAudioTrack::COMMON_TIME)) { 4743 return BAD_VALUE; 4744 } 4745 4746 Mutex::Autolock lock(mMediaTimeTransformLock); 4747 mMediaTimeTransform = xform; 4748 mMediaTimeTransformTarget = target; 4749 mMediaTimeTransformValid = true; 4750 4751 return NO_ERROR; 4752} 4753 4754#define min(a, b) ((a) < (b) ? (a) : (b)) 4755 4756// implementation of getNextBuffer for tracks whose buffers have timestamps 4757status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4758 AudioBufferProvider::Buffer* buffer, int64_t pts) 4759{ 4760 if (pts == AudioBufferProvider::kInvalidPTS) { 4761 buffer->raw = 0; 4762 buffer->frameCount = 0; 4763 mTimedAudioOutputOnTime = false; 4764 return INVALID_OPERATION; 4765 } 4766 4767 Mutex::Autolock _l(mTimedBufferQueueLock); 4768 4769 ALOG_ASSERT(!mQueueHeadInFlight, 4770 "getNextBuffer called without releaseBuffer!"); 4771 4772 while (true) { 4773 4774 // if we have no timed buffers, then fail 4775 if (mTimedBufferQueue.isEmpty()) { 4776 buffer->raw = 0; 4777 buffer->frameCount = 0; 4778 return NOT_ENOUGH_DATA; 4779 } 4780 4781 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4782 4783 // calculate the PTS of the head of the timed buffer queue expressed in 4784 // local time 4785 int64_t headLocalPTS; 4786 { 4787 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4788 4789 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4790 4791 if (mMediaTimeTransform.a_to_b_denom == 0) { 4792 // the transform represents a pause, so yield silence 4793 timedYieldSilence_l(buffer->frameCount, buffer); 4794 return NO_ERROR; 4795 } 4796 4797 int64_t transformedPTS; 4798 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4799 &transformedPTS)) { 4800 // the transform failed. this shouldn't happen, but if it does 4801 // then just drop this buffer 4802 ALOGW("timedGetNextBuffer transform failed"); 4803 buffer->raw = 0; 4804 buffer->frameCount = 0; 4805 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4806 return NO_ERROR; 4807 } 4808 4809 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4810 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4811 &headLocalPTS)) { 4812 buffer->raw = 0; 4813 buffer->frameCount = 0; 4814 return INVALID_OPERATION; 4815 } 4816 } else { 4817 headLocalPTS = transformedPTS; 4818 } 4819 } 4820 4821 // adjust the head buffer's PTS to reflect the portion of the head buffer 4822 // that has already been consumed 4823 int64_t effectivePTS = headLocalPTS + 4824 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4825 4826 // Calculate the delta in samples between the head of the input buffer 4827 // queue and the start of the next output buffer that will be written. 4828 // If the transformation fails because of over or underflow, it means 4829 // that the sample's position in the output stream is so far out of 4830 // whack that it should just be dropped. 4831 int64_t sampleDelta; 4832 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4833 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4834 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4835 " mix"); 4836 continue; 4837 } 4838 if (!mLocalTimeToSampleTransform.doForwardTransform( 4839 (effectivePTS - pts) << 32, &sampleDelta)) { 4840 ALOGV("*** too late during sample rate transform: dropped buffer"); 4841 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4842 continue; 4843 } 4844 4845 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4846 " sampleDelta=[%d.%08x]", 4847 head.pts(), head.position(), pts, 4848 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4849 + (sampleDelta >> 32)), 4850 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4851 4852 // if the delta between the ideal placement for the next input sample and 4853 // the current output position is within this threshold, then we will 4854 // concatenate the next input samples to the previous output 4855 const int64_t kSampleContinuityThreshold = 4856 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4857 4858 // if this is the first buffer of audio that we're emitting from this track 4859 // then it should be almost exactly on time. 4860 const int64_t kSampleStartupThreshold = 1LL << 32; 4861 4862 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4863 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4864 // the next input is close enough to being on time, so concatenate it 4865 // with the last output 4866 timedYieldSamples_l(buffer); 4867 4868 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4869 head.position(), buffer->frameCount); 4870 return NO_ERROR; 4871 } 4872 4873 // Looks like our output is not on time. Reset our on timed status. 4874 // Next time we mix samples from our input queue, then should be within 4875 // the StartupThreshold. 4876 mTimedAudioOutputOnTime = false; 4877 if (sampleDelta > 0) { 4878 // the gap between the current output position and the proper start of 4879 // the next input sample is too big, so fill it with silence 4880 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4881 4882 timedYieldSilence_l(framesUntilNextInput, buffer); 4883 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4884 return NO_ERROR; 4885 } else { 4886 // the next input sample is late 4887 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4888 size_t onTimeSamplePosition = 4889 head.position() + lateFrames * mCblk->frameSize; 4890 4891 if (onTimeSamplePosition > head.buffer()->size()) { 4892 // all the remaining samples in the head are too late, so 4893 // drop it and move on 4894 ALOGV("*** too late: dropped buffer"); 4895 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4896 continue; 4897 } else { 4898 // skip over the late samples 4899 head.setPosition(onTimeSamplePosition); 4900 4901 // yield the available samples 4902 timedYieldSamples_l(buffer); 4903 4904 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4905 return NO_ERROR; 4906 } 4907 } 4908 } 4909} 4910 4911// Yield samples from the timed buffer queue head up to the given output 4912// buffer's capacity. 4913// 4914// Caller must hold mTimedBufferQueueLock 4915void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4916 AudioBufferProvider::Buffer* buffer) { 4917 4918 const TimedBuffer& head = mTimedBufferQueue[0]; 4919 4920 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4921 head.position()); 4922 4923 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4924 mCblk->frameSize); 4925 size_t framesRequested = buffer->frameCount; 4926 buffer->frameCount = min(framesLeftInHead, framesRequested); 4927 4928 mQueueHeadInFlight = true; 4929 mTimedAudioOutputOnTime = true; 4930} 4931 4932// Yield samples of silence up to the given output buffer's capacity 4933// 4934// Caller must hold mTimedBufferQueueLock 4935void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4936 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4937 4938 // lazily allocate a buffer filled with silence 4939 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4940 delete [] mTimedSilenceBuffer; 4941 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4942 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4943 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4944 } 4945 4946 buffer->raw = mTimedSilenceBuffer; 4947 size_t framesRequested = buffer->frameCount; 4948 buffer->frameCount = min(numFrames, framesRequested); 4949 4950 mTimedAudioOutputOnTime = false; 4951} 4952 4953// AudioBufferProvider interface 4954void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4955 AudioBufferProvider::Buffer* buffer) { 4956 4957 Mutex::Autolock _l(mTimedBufferQueueLock); 4958 4959 // If the buffer which was just released is part of the buffer at the head 4960 // of the queue, be sure to update the amt of the buffer which has been 4961 // consumed. If the buffer being returned is not part of the head of the 4962 // queue, its either because the buffer is part of the silence buffer, or 4963 // because the head of the timed queue was trimmed after the mixer called 4964 // getNextBuffer but before the mixer called releaseBuffer. 4965 if (buffer->raw == mTimedSilenceBuffer) { 4966 ALOG_ASSERT(!mQueueHeadInFlight, 4967 "Queue head in flight during release of silence buffer!"); 4968 goto done; 4969 } 4970 4971 ALOG_ASSERT(mQueueHeadInFlight, 4972 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4973 " head in flight."); 4974 4975 if (mTimedBufferQueue.size()) { 4976 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4977 4978 void* start = head.buffer()->pointer(); 4979 void* end = reinterpret_cast<void*>( 4980 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4981 + head.buffer()->size()); 4982 4983 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4984 "released buffer not within the head of the timed buffer" 4985 " queue; qHead = [%p, %p], released buffer = %p", 4986 start, end, buffer->raw); 4987 4988 head.setPosition(head.position() + 4989 (buffer->frameCount * mCblk->frameSize)); 4990 mQueueHeadInFlight = false; 4991 4992 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4993 "Bad bookkeeping during releaseBuffer! Should have at" 4994 " least %u queued frames, but we think we have only %u", 4995 buffer->frameCount, mFramesPendingInQueue); 4996 4997 mFramesPendingInQueue -= buffer->frameCount; 4998 4999 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5000 || mTrimQueueHeadOnRelease) { 5001 trimTimedBufferQueueHead_l("releaseBuffer"); 5002 mTrimQueueHeadOnRelease = false; 5003 } 5004 } else { 5005 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5006 " buffers in the timed buffer queue"); 5007 } 5008 5009done: 5010 buffer->raw = 0; 5011 buffer->frameCount = 0; 5012} 5013 5014size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5015 Mutex::Autolock _l(mTimedBufferQueueLock); 5016 return mFramesPendingInQueue; 5017} 5018 5019AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5020 : mPTS(0), mPosition(0) {} 5021 5022AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5023 const sp<IMemory>& buffer, int64_t pts) 5024 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5025 5026// ---------------------------------------------------------------------------- 5027 5028// RecordTrack constructor must be called with AudioFlinger::mLock held 5029AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5030 RecordThread *thread, 5031 const sp<Client>& client, 5032 uint32_t sampleRate, 5033 audio_format_t format, 5034 uint32_t channelMask, 5035 int frameCount, 5036 int sessionId) 5037 : TrackBase(thread, client, sampleRate, format, 5038 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5039 mOverflow(false) 5040{ 5041 if (mCblk != NULL) { 5042 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5043 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5044 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5045 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5046 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5047 } else { 5048 mCblk->frameSize = sizeof(int8_t); 5049 } 5050 } 5051} 5052 5053AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5054{ 5055 sp<ThreadBase> thread = mThread.promote(); 5056 if (thread != 0) { 5057 AudioSystem::releaseInput(thread->id()); 5058 } 5059} 5060 5061// AudioBufferProvider interface 5062status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5063{ 5064 audio_track_cblk_t* cblk = this->cblk(); 5065 uint32_t framesAvail; 5066 uint32_t framesReq = buffer->frameCount; 5067 5068 // Check if last stepServer failed, try to step now 5069 if (mStepServerFailed) { 5070 if (!step()) goto getNextBuffer_exit; 5071 ALOGV("stepServer recovered"); 5072 mStepServerFailed = false; 5073 } 5074 5075 framesAvail = cblk->framesAvailable_l(); 5076 5077 if (CC_LIKELY(framesAvail)) { 5078 uint32_t s = cblk->server; 5079 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5080 5081 if (framesReq > framesAvail) { 5082 framesReq = framesAvail; 5083 } 5084 if (framesReq > bufferEnd - s) { 5085 framesReq = bufferEnd - s; 5086 } 5087 5088 buffer->raw = getBuffer(s, framesReq); 5089 if (buffer->raw == NULL) goto getNextBuffer_exit; 5090 5091 buffer->frameCount = framesReq; 5092 return NO_ERROR; 5093 } 5094 5095getNextBuffer_exit: 5096 buffer->raw = NULL; 5097 buffer->frameCount = 0; 5098 return NOT_ENOUGH_DATA; 5099} 5100 5101status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5102 int triggerSession) 5103{ 5104 sp<ThreadBase> thread = mThread.promote(); 5105 if (thread != 0) { 5106 RecordThread *recordThread = (RecordThread *)thread.get(); 5107 return recordThread->start(this, event, triggerSession); 5108 } else { 5109 return BAD_VALUE; 5110 } 5111} 5112 5113void AudioFlinger::RecordThread::RecordTrack::stop() 5114{ 5115 sp<ThreadBase> thread = mThread.promote(); 5116 if (thread != 0) { 5117 RecordThread *recordThread = (RecordThread *)thread.get(); 5118 recordThread->stop(this); 5119 TrackBase::reset(); 5120 // Force overrun condition to avoid false overrun callback until first data is 5121 // read from buffer 5122 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5123 } 5124} 5125 5126void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5127{ 5128 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5129 (mClient == 0) ? getpid_cached : mClient->pid(), 5130 mFormat, 5131 mChannelMask, 5132 mSessionId, 5133 mFrameCount, 5134 mState, 5135 mCblk->sampleRate, 5136 mCblk->server, 5137 mCblk->user); 5138} 5139 5140 5141// ---------------------------------------------------------------------------- 5142 5143AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5144 PlaybackThread *playbackThread, 5145 DuplicatingThread *sourceThread, 5146 uint32_t sampleRate, 5147 audio_format_t format, 5148 uint32_t channelMask, 5149 int frameCount) 5150 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5151 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5152 mActive(false), mSourceThread(sourceThread) 5153{ 5154 5155 if (mCblk != NULL) { 5156 mCblk->flags |= CBLK_DIRECTION_OUT; 5157 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5158 mOutBuffer.frameCount = 0; 5159 playbackThread->mTracks.add(this); 5160 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5161 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5162 mCblk, mBuffer, mCblk->buffers, 5163 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5164 } else { 5165 ALOGW("Error creating output track on thread %p", playbackThread); 5166 } 5167} 5168 5169AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5170{ 5171 clearBufferQueue(); 5172} 5173 5174status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5175 int triggerSession) 5176{ 5177 status_t status = Track::start(event, triggerSession); 5178 if (status != NO_ERROR) { 5179 return status; 5180 } 5181 5182 mActive = true; 5183 mRetryCount = 127; 5184 return status; 5185} 5186 5187void AudioFlinger::PlaybackThread::OutputTrack::stop() 5188{ 5189 Track::stop(); 5190 clearBufferQueue(); 5191 mOutBuffer.frameCount = 0; 5192 mActive = false; 5193} 5194 5195bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5196{ 5197 Buffer *pInBuffer; 5198 Buffer inBuffer; 5199 uint32_t channelCount = mChannelCount; 5200 bool outputBufferFull = false; 5201 inBuffer.frameCount = frames; 5202 inBuffer.i16 = data; 5203 5204 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5205 5206 if (!mActive && frames != 0) { 5207 start(); 5208 sp<ThreadBase> thread = mThread.promote(); 5209 if (thread != 0) { 5210 MixerThread *mixerThread = (MixerThread *)thread.get(); 5211 if (mCblk->frameCount > frames){ 5212 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5213 uint32_t startFrames = (mCblk->frameCount - frames); 5214 pInBuffer = new Buffer; 5215 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5216 pInBuffer->frameCount = startFrames; 5217 pInBuffer->i16 = pInBuffer->mBuffer; 5218 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5219 mBufferQueue.add(pInBuffer); 5220 } else { 5221 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5222 } 5223 } 5224 } 5225 } 5226 5227 while (waitTimeLeftMs) { 5228 // First write pending buffers, then new data 5229 if (mBufferQueue.size()) { 5230 pInBuffer = mBufferQueue.itemAt(0); 5231 } else { 5232 pInBuffer = &inBuffer; 5233 } 5234 5235 if (pInBuffer->frameCount == 0) { 5236 break; 5237 } 5238 5239 if (mOutBuffer.frameCount == 0) { 5240 mOutBuffer.frameCount = pInBuffer->frameCount; 5241 nsecs_t startTime = systemTime(); 5242 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5243 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5244 outputBufferFull = true; 5245 break; 5246 } 5247 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5248 if (waitTimeLeftMs >= waitTimeMs) { 5249 waitTimeLeftMs -= waitTimeMs; 5250 } else { 5251 waitTimeLeftMs = 0; 5252 } 5253 } 5254 5255 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5256 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5257 mCblk->stepUser(outFrames); 5258 pInBuffer->frameCount -= outFrames; 5259 pInBuffer->i16 += outFrames * channelCount; 5260 mOutBuffer.frameCount -= outFrames; 5261 mOutBuffer.i16 += outFrames * channelCount; 5262 5263 if (pInBuffer->frameCount == 0) { 5264 if (mBufferQueue.size()) { 5265 mBufferQueue.removeAt(0); 5266 delete [] pInBuffer->mBuffer; 5267 delete pInBuffer; 5268 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5269 } else { 5270 break; 5271 } 5272 } 5273 } 5274 5275 // If we could not write all frames, allocate a buffer and queue it for next time. 5276 if (inBuffer.frameCount) { 5277 sp<ThreadBase> thread = mThread.promote(); 5278 if (thread != 0 && !thread->standby()) { 5279 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5280 pInBuffer = new Buffer; 5281 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5282 pInBuffer->frameCount = inBuffer.frameCount; 5283 pInBuffer->i16 = pInBuffer->mBuffer; 5284 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5285 mBufferQueue.add(pInBuffer); 5286 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5287 } else { 5288 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5289 } 5290 } 5291 } 5292 5293 // Calling write() with a 0 length buffer, means that no more data will be written: 5294 // If no more buffers are pending, fill output track buffer to make sure it is started 5295 // by output mixer. 5296 if (frames == 0 && mBufferQueue.size() == 0) { 5297 if (mCblk->user < mCblk->frameCount) { 5298 frames = mCblk->frameCount - mCblk->user; 5299 pInBuffer = new Buffer; 5300 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5301 pInBuffer->frameCount = frames; 5302 pInBuffer->i16 = pInBuffer->mBuffer; 5303 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5304 mBufferQueue.add(pInBuffer); 5305 } else if (mActive) { 5306 stop(); 5307 } 5308 } 5309 5310 return outputBufferFull; 5311} 5312 5313status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5314{ 5315 int active; 5316 status_t result; 5317 audio_track_cblk_t* cblk = mCblk; 5318 uint32_t framesReq = buffer->frameCount; 5319 5320// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5321 buffer->frameCount = 0; 5322 5323 uint32_t framesAvail = cblk->framesAvailable(); 5324 5325 5326 if (framesAvail == 0) { 5327 Mutex::Autolock _l(cblk->lock); 5328 goto start_loop_here; 5329 while (framesAvail == 0) { 5330 active = mActive; 5331 if (CC_UNLIKELY(!active)) { 5332 ALOGV("Not active and NO_MORE_BUFFERS"); 5333 return NO_MORE_BUFFERS; 5334 } 5335 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5336 if (result != NO_ERROR) { 5337 return NO_MORE_BUFFERS; 5338 } 5339 // read the server count again 5340 start_loop_here: 5341 framesAvail = cblk->framesAvailable_l(); 5342 } 5343 } 5344 5345// if (framesAvail < framesReq) { 5346// return NO_MORE_BUFFERS; 5347// } 5348 5349 if (framesReq > framesAvail) { 5350 framesReq = framesAvail; 5351 } 5352 5353 uint32_t u = cblk->user; 5354 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5355 5356 if (framesReq > bufferEnd - u) { 5357 framesReq = bufferEnd - u; 5358 } 5359 5360 buffer->frameCount = framesReq; 5361 buffer->raw = (void *)cblk->buffer(u); 5362 return NO_ERROR; 5363} 5364 5365 5366void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5367{ 5368 size_t size = mBufferQueue.size(); 5369 5370 for (size_t i = 0; i < size; i++) { 5371 Buffer *pBuffer = mBufferQueue.itemAt(i); 5372 delete [] pBuffer->mBuffer; 5373 delete pBuffer; 5374 } 5375 mBufferQueue.clear(); 5376} 5377 5378// ---------------------------------------------------------------------------- 5379 5380AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5381 : RefBase(), 5382 mAudioFlinger(audioFlinger), 5383 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5384 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5385 mPid(pid), 5386 mTimedTrackCount(0) 5387{ 5388 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5389} 5390 5391// Client destructor must be called with AudioFlinger::mLock held 5392AudioFlinger::Client::~Client() 5393{ 5394 mAudioFlinger->removeClient_l(mPid); 5395} 5396 5397sp<MemoryDealer> AudioFlinger::Client::heap() const 5398{ 5399 return mMemoryDealer; 5400} 5401 5402// Reserve one of the limited slots for a timed audio track associated 5403// with this client 5404bool AudioFlinger::Client::reserveTimedTrack() 5405{ 5406 const int kMaxTimedTracksPerClient = 4; 5407 5408 Mutex::Autolock _l(mTimedTrackLock); 5409 5410 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5411 ALOGW("can not create timed track - pid %d has exceeded the limit", 5412 mPid); 5413 return false; 5414 } 5415 5416 mTimedTrackCount++; 5417 return true; 5418} 5419 5420// Release a slot for a timed audio track 5421void AudioFlinger::Client::releaseTimedTrack() 5422{ 5423 Mutex::Autolock _l(mTimedTrackLock); 5424 mTimedTrackCount--; 5425} 5426 5427// ---------------------------------------------------------------------------- 5428 5429AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5430 const sp<IAudioFlingerClient>& client, 5431 pid_t pid) 5432 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5433{ 5434} 5435 5436AudioFlinger::NotificationClient::~NotificationClient() 5437{ 5438} 5439 5440void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5441{ 5442 sp<NotificationClient> keep(this); 5443 mAudioFlinger->removeNotificationClient(mPid); 5444} 5445 5446// ---------------------------------------------------------------------------- 5447 5448AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5449 : BnAudioTrack(), 5450 mTrack(track) 5451{ 5452} 5453 5454AudioFlinger::TrackHandle::~TrackHandle() { 5455 // just stop the track on deletion, associated resources 5456 // will be freed from the main thread once all pending buffers have 5457 // been played. Unless it's not in the active track list, in which 5458 // case we free everything now... 5459 mTrack->destroy(); 5460} 5461 5462sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5463 return mTrack->getCblk(); 5464} 5465 5466status_t AudioFlinger::TrackHandle::start() { 5467 return mTrack->start(); 5468} 5469 5470void AudioFlinger::TrackHandle::stop() { 5471 mTrack->stop(); 5472} 5473 5474void AudioFlinger::TrackHandle::flush() { 5475 mTrack->flush(); 5476} 5477 5478void AudioFlinger::TrackHandle::mute(bool e) { 5479 mTrack->mute(e); 5480} 5481 5482void AudioFlinger::TrackHandle::pause() { 5483 mTrack->pause(); 5484} 5485 5486status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5487{ 5488 return mTrack->attachAuxEffect(EffectId); 5489} 5490 5491status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5492 sp<IMemory>* buffer) { 5493 if (!mTrack->isTimedTrack()) 5494 return INVALID_OPERATION; 5495 5496 PlaybackThread::TimedTrack* tt = 5497 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5498 return tt->allocateTimedBuffer(size, buffer); 5499} 5500 5501status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5502 int64_t pts) { 5503 if (!mTrack->isTimedTrack()) 5504 return INVALID_OPERATION; 5505 5506 PlaybackThread::TimedTrack* tt = 5507 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5508 return tt->queueTimedBuffer(buffer, pts); 5509} 5510 5511status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5512 const LinearTransform& xform, int target) { 5513 5514 if (!mTrack->isTimedTrack()) 5515 return INVALID_OPERATION; 5516 5517 PlaybackThread::TimedTrack* tt = 5518 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5519 return tt->setMediaTimeTransform( 5520 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5521} 5522 5523status_t AudioFlinger::TrackHandle::onTransact( 5524 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5525{ 5526 return BnAudioTrack::onTransact(code, data, reply, flags); 5527} 5528 5529// ---------------------------------------------------------------------------- 5530 5531sp<IAudioRecord> AudioFlinger::openRecord( 5532 pid_t pid, 5533 audio_io_handle_t input, 5534 uint32_t sampleRate, 5535 audio_format_t format, 5536 uint32_t channelMask, 5537 int frameCount, 5538 IAudioFlinger::track_flags_t flags, 5539 int *sessionId, 5540 status_t *status) 5541{ 5542 sp<RecordThread::RecordTrack> recordTrack; 5543 sp<RecordHandle> recordHandle; 5544 sp<Client> client; 5545 status_t lStatus; 5546 RecordThread *thread; 5547 size_t inFrameCount; 5548 int lSessionId; 5549 5550 // check calling permissions 5551 if (!recordingAllowed()) { 5552 lStatus = PERMISSION_DENIED; 5553 goto Exit; 5554 } 5555 5556 // add client to list 5557 { // scope for mLock 5558 Mutex::Autolock _l(mLock); 5559 thread = checkRecordThread_l(input); 5560 if (thread == NULL) { 5561 lStatus = BAD_VALUE; 5562 goto Exit; 5563 } 5564 5565 client = registerPid_l(pid); 5566 5567 // If no audio session id is provided, create one here 5568 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5569 lSessionId = *sessionId; 5570 } else { 5571 lSessionId = nextUniqueId(); 5572 if (sessionId != NULL) { 5573 *sessionId = lSessionId; 5574 } 5575 } 5576 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5577 recordTrack = thread->createRecordTrack_l(client, 5578 sampleRate, 5579 format, 5580 channelMask, 5581 frameCount, 5582 lSessionId, 5583 &lStatus); 5584 } 5585 if (lStatus != NO_ERROR) { 5586 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5587 // destructor is called by the TrackBase destructor with mLock held 5588 client.clear(); 5589 recordTrack.clear(); 5590 goto Exit; 5591 } 5592 5593 // return to handle to client 5594 recordHandle = new RecordHandle(recordTrack); 5595 lStatus = NO_ERROR; 5596 5597Exit: 5598 if (status) { 5599 *status = lStatus; 5600 } 5601 return recordHandle; 5602} 5603 5604// ---------------------------------------------------------------------------- 5605 5606AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5607 : BnAudioRecord(), 5608 mRecordTrack(recordTrack) 5609{ 5610} 5611 5612AudioFlinger::RecordHandle::~RecordHandle() { 5613 stop(); 5614} 5615 5616sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5617 return mRecordTrack->getCblk(); 5618} 5619 5620status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5621 ALOGV("RecordHandle::start()"); 5622 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5623} 5624 5625void AudioFlinger::RecordHandle::stop() { 5626 ALOGV("RecordHandle::stop()"); 5627 mRecordTrack->stop(); 5628} 5629 5630status_t AudioFlinger::RecordHandle::onTransact( 5631 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5632{ 5633 return BnAudioRecord::onTransact(code, data, reply, flags); 5634} 5635 5636// ---------------------------------------------------------------------------- 5637 5638AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5639 AudioStreamIn *input, 5640 uint32_t sampleRate, 5641 uint32_t channels, 5642 audio_io_handle_t id, 5643 uint32_t device) : 5644 ThreadBase(audioFlinger, id, device, RECORD), 5645 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5646 // mRsmpInIndex and mInputBytes set by readInputParameters() 5647 mReqChannelCount(popcount(channels)), 5648 mReqSampleRate(sampleRate) 5649 // mBytesRead is only meaningful while active, and so is cleared in start() 5650 // (but might be better to also clear here for dump?) 5651{ 5652 snprintf(mName, kNameLength, "AudioIn_%X", id); 5653 5654 readInputParameters(); 5655} 5656 5657 5658AudioFlinger::RecordThread::~RecordThread() 5659{ 5660 delete[] mRsmpInBuffer; 5661 delete mResampler; 5662 delete[] mRsmpOutBuffer; 5663} 5664 5665void AudioFlinger::RecordThread::onFirstRef() 5666{ 5667 run(mName, PRIORITY_URGENT_AUDIO); 5668} 5669 5670status_t AudioFlinger::RecordThread::readyToRun() 5671{ 5672 status_t status = initCheck(); 5673 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5674 return status; 5675} 5676 5677bool AudioFlinger::RecordThread::threadLoop() 5678{ 5679 AudioBufferProvider::Buffer buffer; 5680 sp<RecordTrack> activeTrack; 5681 Vector< sp<EffectChain> > effectChains; 5682 5683 nsecs_t lastWarning = 0; 5684 5685 acquireWakeLock(); 5686 5687 // start recording 5688 while (!exitPending()) { 5689 5690 processConfigEvents(); 5691 5692 { // scope for mLock 5693 Mutex::Autolock _l(mLock); 5694 checkForNewParameters_l(); 5695 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5696 if (!mStandby) { 5697 mInput->stream->common.standby(&mInput->stream->common); 5698 mStandby = true; 5699 } 5700 5701 if (exitPending()) break; 5702 5703 releaseWakeLock_l(); 5704 ALOGV("RecordThread: loop stopping"); 5705 // go to sleep 5706 mWaitWorkCV.wait(mLock); 5707 ALOGV("RecordThread: loop starting"); 5708 acquireWakeLock_l(); 5709 continue; 5710 } 5711 if (mActiveTrack != 0) { 5712 if (mActiveTrack->mState == TrackBase::PAUSING) { 5713 if (!mStandby) { 5714 mInput->stream->common.standby(&mInput->stream->common); 5715 mStandby = true; 5716 } 5717 mActiveTrack.clear(); 5718 mStartStopCond.broadcast(); 5719 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5720 if (mReqChannelCount != mActiveTrack->channelCount()) { 5721 mActiveTrack.clear(); 5722 mStartStopCond.broadcast(); 5723 } else if (mBytesRead != 0) { 5724 // record start succeeds only if first read from audio input 5725 // succeeds 5726 if (mBytesRead > 0) { 5727 mActiveTrack->mState = TrackBase::ACTIVE; 5728 } else { 5729 mActiveTrack.clear(); 5730 } 5731 mStartStopCond.broadcast(); 5732 } 5733 mStandby = false; 5734 } 5735 } 5736 lockEffectChains_l(effectChains); 5737 } 5738 5739 if (mActiveTrack != 0) { 5740 if (mActiveTrack->mState != TrackBase::ACTIVE && 5741 mActiveTrack->mState != TrackBase::RESUMING) { 5742 unlockEffectChains(effectChains); 5743 usleep(kRecordThreadSleepUs); 5744 continue; 5745 } 5746 for (size_t i = 0; i < effectChains.size(); i ++) { 5747 effectChains[i]->process_l(); 5748 } 5749 5750 buffer.frameCount = mFrameCount; 5751 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5752 size_t framesOut = buffer.frameCount; 5753 if (mResampler == NULL) { 5754 // no resampling 5755 while (framesOut) { 5756 size_t framesIn = mFrameCount - mRsmpInIndex; 5757 if (framesIn) { 5758 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5759 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5760 if (framesIn > framesOut) 5761 framesIn = framesOut; 5762 mRsmpInIndex += framesIn; 5763 framesOut -= framesIn; 5764 if ((int)mChannelCount == mReqChannelCount || 5765 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5766 memcpy(dst, src, framesIn * mFrameSize); 5767 } else { 5768 int16_t *src16 = (int16_t *)src; 5769 int16_t *dst16 = (int16_t *)dst; 5770 if (mChannelCount == 1) { 5771 while (framesIn--) { 5772 *dst16++ = *src16; 5773 *dst16++ = *src16++; 5774 } 5775 } else { 5776 while (framesIn--) { 5777 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5778 src16 += 2; 5779 } 5780 } 5781 } 5782 } 5783 if (framesOut && mFrameCount == mRsmpInIndex) { 5784 if (framesOut == mFrameCount && 5785 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5786 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5787 framesOut = 0; 5788 } else { 5789 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5790 mRsmpInIndex = 0; 5791 } 5792 if (mBytesRead < 0) { 5793 ALOGE("Error reading audio input"); 5794 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5795 // Force input into standby so that it tries to 5796 // recover at next read attempt 5797 mInput->stream->common.standby(&mInput->stream->common); 5798 usleep(kRecordThreadSleepUs); 5799 } 5800 mRsmpInIndex = mFrameCount; 5801 framesOut = 0; 5802 buffer.frameCount = 0; 5803 } 5804 } 5805 } 5806 } else { 5807 // resampling 5808 5809 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5810 // alter output frame count as if we were expecting stereo samples 5811 if (mChannelCount == 1 && mReqChannelCount == 1) { 5812 framesOut >>= 1; 5813 } 5814 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5815 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5816 // are 32 bit aligned which should be always true. 5817 if (mChannelCount == 2 && mReqChannelCount == 1) { 5818 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5819 // the resampler always outputs stereo samples: do post stereo to mono conversion 5820 int16_t *src = (int16_t *)mRsmpOutBuffer; 5821 int16_t *dst = buffer.i16; 5822 while (framesOut--) { 5823 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5824 src += 2; 5825 } 5826 } else { 5827 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5828 } 5829 5830 } 5831 if (mFramestoDrop == 0) { 5832 mActiveTrack->releaseBuffer(&buffer); 5833 } else { 5834 if (mFramestoDrop > 0) { 5835 mFramestoDrop -= buffer.frameCount; 5836 if (mFramestoDrop < 0) { 5837 mFramestoDrop = 0; 5838 } 5839 } 5840 } 5841 mActiveTrack->overflow(); 5842 } 5843 // client isn't retrieving buffers fast enough 5844 else { 5845 if (!mActiveTrack->setOverflow()) { 5846 nsecs_t now = systemTime(); 5847 if ((now - lastWarning) > kWarningThrottleNs) { 5848 ALOGW("RecordThread: buffer overflow"); 5849 lastWarning = now; 5850 } 5851 } 5852 // Release the processor for a while before asking for a new buffer. 5853 // This will give the application more chance to read from the buffer and 5854 // clear the overflow. 5855 usleep(kRecordThreadSleepUs); 5856 } 5857 } 5858 // enable changes in effect chain 5859 unlockEffectChains(effectChains); 5860 effectChains.clear(); 5861 } 5862 5863 if (!mStandby) { 5864 mInput->stream->common.standby(&mInput->stream->common); 5865 } 5866 mActiveTrack.clear(); 5867 5868 mStartStopCond.broadcast(); 5869 5870 releaseWakeLock(); 5871 5872 ALOGV("RecordThread %p exiting", this); 5873 return false; 5874} 5875 5876 5877sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5878 const sp<AudioFlinger::Client>& client, 5879 uint32_t sampleRate, 5880 audio_format_t format, 5881 int channelMask, 5882 int frameCount, 5883 int sessionId, 5884 status_t *status) 5885{ 5886 sp<RecordTrack> track; 5887 status_t lStatus; 5888 5889 lStatus = initCheck(); 5890 if (lStatus != NO_ERROR) { 5891 ALOGE("Audio driver not initialized."); 5892 goto Exit; 5893 } 5894 5895 { // scope for mLock 5896 Mutex::Autolock _l(mLock); 5897 5898 track = new RecordTrack(this, client, sampleRate, 5899 format, channelMask, frameCount, sessionId); 5900 5901 if (track->getCblk() == 0) { 5902 lStatus = NO_MEMORY; 5903 goto Exit; 5904 } 5905 5906 mTrack = track.get(); 5907 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5908 bool suspend = audio_is_bluetooth_sco_device( 5909 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5910 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5911 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5912 } 5913 lStatus = NO_ERROR; 5914 5915Exit: 5916 if (status) { 5917 *status = lStatus; 5918 } 5919 return track; 5920} 5921 5922status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5923 AudioSystem::sync_event_t event, 5924 int triggerSession) 5925{ 5926 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5927 sp<ThreadBase> strongMe = this; 5928 status_t status = NO_ERROR; 5929 5930 if (event == AudioSystem::SYNC_EVENT_NONE) { 5931 mSyncStartEvent.clear(); 5932 mFramestoDrop = 0; 5933 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5934 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5935 triggerSession, 5936 recordTrack->sessionId(), 5937 syncStartEventCallback, 5938 this); 5939 mFramestoDrop = -1; 5940 } 5941 5942 { 5943 AutoMutex lock(mLock); 5944 if (mActiveTrack != 0) { 5945 if (recordTrack != mActiveTrack.get()) { 5946 status = -EBUSY; 5947 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5948 mActiveTrack->mState = TrackBase::ACTIVE; 5949 } 5950 return status; 5951 } 5952 5953 recordTrack->mState = TrackBase::IDLE; 5954 mActiveTrack = recordTrack; 5955 mLock.unlock(); 5956 status_t status = AudioSystem::startInput(mId); 5957 mLock.lock(); 5958 if (status != NO_ERROR) { 5959 mActiveTrack.clear(); 5960 clearSyncStartEvent(); 5961 return status; 5962 } 5963 mRsmpInIndex = mFrameCount; 5964 mBytesRead = 0; 5965 if (mResampler != NULL) { 5966 mResampler->reset(); 5967 } 5968 mActiveTrack->mState = TrackBase::RESUMING; 5969 // signal thread to start 5970 ALOGV("Signal record thread"); 5971 mWaitWorkCV.signal(); 5972 // do not wait for mStartStopCond if exiting 5973 if (exitPending()) { 5974 mActiveTrack.clear(); 5975 status = INVALID_OPERATION; 5976 goto startError; 5977 } 5978 mStartStopCond.wait(mLock); 5979 if (mActiveTrack == 0) { 5980 ALOGV("Record failed to start"); 5981 status = BAD_VALUE; 5982 goto startError; 5983 } 5984 ALOGV("Record started OK"); 5985 return status; 5986 } 5987startError: 5988 AudioSystem::stopInput(mId); 5989 clearSyncStartEvent(); 5990 return status; 5991} 5992 5993void AudioFlinger::RecordThread::clearSyncStartEvent() 5994{ 5995 if (mSyncStartEvent != 0) { 5996 mSyncStartEvent->cancel(); 5997 } 5998 mSyncStartEvent.clear(); 5999} 6000 6001void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6002{ 6003 sp<SyncEvent> strongEvent = event.promote(); 6004 6005 if (strongEvent != 0) { 6006 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6007 me->handleSyncStartEvent(strongEvent); 6008 } 6009} 6010 6011void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6012{ 6013 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 6014 mActiveTrack.get(), 6015 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 6016 event->listenerSession()); 6017 6018 if (mActiveTrack != 0 && 6019 event == mSyncStartEvent) { 6020 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6021 // from audio HAL 6022 mFramestoDrop = mFrameCount * 2; 6023 mSyncStartEvent.clear(); 6024 } 6025} 6026 6027void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6028 ALOGV("RecordThread::stop"); 6029 sp<ThreadBase> strongMe = this; 6030 { 6031 AutoMutex lock(mLock); 6032 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6033 mActiveTrack->mState = TrackBase::PAUSING; 6034 // do not wait for mStartStopCond if exiting 6035 if (exitPending()) { 6036 return; 6037 } 6038 mStartStopCond.wait(mLock); 6039 // if we have been restarted, recordTrack == mActiveTrack.get() here 6040 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6041 mLock.unlock(); 6042 AudioSystem::stopInput(mId); 6043 mLock.lock(); 6044 ALOGV("Record stopped OK"); 6045 } 6046 } 6047 } 6048} 6049 6050bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6051{ 6052 return false; 6053} 6054 6055status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6056{ 6057 if (!isValidSyncEvent(event)) { 6058 return BAD_VALUE; 6059 } 6060 6061 Mutex::Autolock _l(mLock); 6062 6063 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6064 mTrack->setSyncEvent(event); 6065 return NO_ERROR; 6066 } 6067 return NAME_NOT_FOUND; 6068} 6069 6070status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6071{ 6072 const size_t SIZE = 256; 6073 char buffer[SIZE]; 6074 String8 result; 6075 6076 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6077 result.append(buffer); 6078 6079 if (mActiveTrack != 0) { 6080 result.append("Active Track:\n"); 6081 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6082 mActiveTrack->dump(buffer, SIZE); 6083 result.append(buffer); 6084 6085 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6086 result.append(buffer); 6087 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6088 result.append(buffer); 6089 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6090 result.append(buffer); 6091 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6092 result.append(buffer); 6093 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6094 result.append(buffer); 6095 6096 6097 } else { 6098 result.append("No record client\n"); 6099 } 6100 write(fd, result.string(), result.size()); 6101 6102 dumpBase(fd, args); 6103 dumpEffectChains(fd, args); 6104 6105 return NO_ERROR; 6106} 6107 6108// AudioBufferProvider interface 6109status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6110{ 6111 size_t framesReq = buffer->frameCount; 6112 size_t framesReady = mFrameCount - mRsmpInIndex; 6113 int channelCount; 6114 6115 if (framesReady == 0) { 6116 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6117 if (mBytesRead < 0) { 6118 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6119 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6120 // Force input into standby so that it tries to 6121 // recover at next read attempt 6122 mInput->stream->common.standby(&mInput->stream->common); 6123 usleep(kRecordThreadSleepUs); 6124 } 6125 buffer->raw = NULL; 6126 buffer->frameCount = 0; 6127 return NOT_ENOUGH_DATA; 6128 } 6129 mRsmpInIndex = 0; 6130 framesReady = mFrameCount; 6131 } 6132 6133 if (framesReq > framesReady) { 6134 framesReq = framesReady; 6135 } 6136 6137 if (mChannelCount == 1 && mReqChannelCount == 2) { 6138 channelCount = 1; 6139 } else { 6140 channelCount = 2; 6141 } 6142 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6143 buffer->frameCount = framesReq; 6144 return NO_ERROR; 6145} 6146 6147// AudioBufferProvider interface 6148void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6149{ 6150 mRsmpInIndex += buffer->frameCount; 6151 buffer->frameCount = 0; 6152} 6153 6154bool AudioFlinger::RecordThread::checkForNewParameters_l() 6155{ 6156 bool reconfig = false; 6157 6158 while (!mNewParameters.isEmpty()) { 6159 status_t status = NO_ERROR; 6160 String8 keyValuePair = mNewParameters[0]; 6161 AudioParameter param = AudioParameter(keyValuePair); 6162 int value; 6163 audio_format_t reqFormat = mFormat; 6164 int reqSamplingRate = mReqSampleRate; 6165 int reqChannelCount = mReqChannelCount; 6166 6167 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6168 reqSamplingRate = value; 6169 reconfig = true; 6170 } 6171 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6172 reqFormat = (audio_format_t) value; 6173 reconfig = true; 6174 } 6175 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6176 reqChannelCount = popcount(value); 6177 reconfig = true; 6178 } 6179 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6180 // do not accept frame count changes if tracks are open as the track buffer 6181 // size depends on frame count and correct behavior would not be guaranteed 6182 // if frame count is changed after track creation 6183 if (mActiveTrack != 0) { 6184 status = INVALID_OPERATION; 6185 } else { 6186 reconfig = true; 6187 } 6188 } 6189 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6190 // forward device change to effects that have requested to be 6191 // aware of attached audio device. 6192 for (size_t i = 0; i < mEffectChains.size(); i++) { 6193 mEffectChains[i]->setDevice_l(value); 6194 } 6195 // store input device and output device but do not forward output device to audio HAL. 6196 // Note that status is ignored by the caller for output device 6197 // (see AudioFlinger::setParameters() 6198 if (value & AUDIO_DEVICE_OUT_ALL) { 6199 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6200 status = BAD_VALUE; 6201 } else { 6202 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6203 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6204 if (mTrack != NULL) { 6205 bool suspend = audio_is_bluetooth_sco_device( 6206 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6207 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6208 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6209 } 6210 } 6211 mDevice |= (uint32_t)value; 6212 } 6213 if (status == NO_ERROR) { 6214 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6215 if (status == INVALID_OPERATION) { 6216 mInput->stream->common.standby(&mInput->stream->common); 6217 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6218 keyValuePair.string()); 6219 } 6220 if (reconfig) { 6221 if (status == BAD_VALUE && 6222 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6223 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6224 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6225 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6226 (reqChannelCount <= FCC_2)) { 6227 status = NO_ERROR; 6228 } 6229 if (status == NO_ERROR) { 6230 readInputParameters(); 6231 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6232 } 6233 } 6234 } 6235 6236 mNewParameters.removeAt(0); 6237 6238 mParamStatus = status; 6239 mParamCond.signal(); 6240 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6241 // already timed out waiting for the status and will never signal the condition. 6242 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6243 } 6244 return reconfig; 6245} 6246 6247String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6248{ 6249 char *s; 6250 String8 out_s8 = String8(); 6251 6252 Mutex::Autolock _l(mLock); 6253 if (initCheck() != NO_ERROR) { 6254 return out_s8; 6255 } 6256 6257 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6258 out_s8 = String8(s); 6259 free(s); 6260 return out_s8; 6261} 6262 6263void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6264 AudioSystem::OutputDescriptor desc; 6265 void *param2 = NULL; 6266 6267 switch (event) { 6268 case AudioSystem::INPUT_OPENED: 6269 case AudioSystem::INPUT_CONFIG_CHANGED: 6270 desc.channels = mChannelMask; 6271 desc.samplingRate = mSampleRate; 6272 desc.format = mFormat; 6273 desc.frameCount = mFrameCount; 6274 desc.latency = 0; 6275 param2 = &desc; 6276 break; 6277 6278 case AudioSystem::INPUT_CLOSED: 6279 default: 6280 break; 6281 } 6282 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6283} 6284 6285void AudioFlinger::RecordThread::readInputParameters() 6286{ 6287 delete mRsmpInBuffer; 6288 // mRsmpInBuffer is always assigned a new[] below 6289 delete mRsmpOutBuffer; 6290 mRsmpOutBuffer = NULL; 6291 delete mResampler; 6292 mResampler = NULL; 6293 6294 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6295 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6296 mChannelCount = (uint16_t)popcount(mChannelMask); 6297 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6298 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6299 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6300 mFrameCount = mInputBytes / mFrameSize; 6301 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6302 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6303 6304 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6305 { 6306 int channelCount; 6307 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6308 // stereo to mono post process as the resampler always outputs stereo. 6309 if (mChannelCount == 1 && mReqChannelCount == 2) { 6310 channelCount = 1; 6311 } else { 6312 channelCount = 2; 6313 } 6314 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6315 mResampler->setSampleRate(mSampleRate); 6316 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6317 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6318 6319 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6320 if (mChannelCount == 1 && mReqChannelCount == 1) { 6321 mFrameCount >>= 1; 6322 } 6323 6324 } 6325 mRsmpInIndex = mFrameCount; 6326} 6327 6328unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6329{ 6330 Mutex::Autolock _l(mLock); 6331 if (initCheck() != NO_ERROR) { 6332 return 0; 6333 } 6334 6335 return mInput->stream->get_input_frames_lost(mInput->stream); 6336} 6337 6338uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6339{ 6340 Mutex::Autolock _l(mLock); 6341 uint32_t result = 0; 6342 if (getEffectChain_l(sessionId) != 0) { 6343 result = EFFECT_SESSION; 6344 } 6345 6346 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6347 result |= TRACK_SESSION; 6348 } 6349 6350 return result; 6351} 6352 6353AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6354{ 6355 Mutex::Autolock _l(mLock); 6356 return mTrack; 6357} 6358 6359AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6360{ 6361 Mutex::Autolock _l(mLock); 6362 return mInput; 6363} 6364 6365AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6366{ 6367 Mutex::Autolock _l(mLock); 6368 AudioStreamIn *input = mInput; 6369 mInput = NULL; 6370 return input; 6371} 6372 6373// this method must always be called either with ThreadBase mLock held or inside the thread loop 6374audio_stream_t* AudioFlinger::RecordThread::stream() const 6375{ 6376 if (mInput == NULL) { 6377 return NULL; 6378 } 6379 return &mInput->stream->common; 6380} 6381 6382 6383// ---------------------------------------------------------------------------- 6384 6385audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6386{ 6387 if (!settingsAllowed()) { 6388 return 0; 6389 } 6390 Mutex::Autolock _l(mLock); 6391 return loadHwModule_l(name); 6392} 6393 6394// loadHwModule_l() must be called with AudioFlinger::mLock held 6395audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6396{ 6397 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6398 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6399 ALOGW("loadHwModule() module %s already loaded", name); 6400 return mAudioHwDevs.keyAt(i); 6401 } 6402 } 6403 6404 audio_hw_device_t *dev; 6405 6406 int rc = load_audio_interface(name, &dev); 6407 if (rc) { 6408 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6409 return 0; 6410 } 6411 6412 mHardwareStatus = AUDIO_HW_INIT; 6413 rc = dev->init_check(dev); 6414 mHardwareStatus = AUDIO_HW_IDLE; 6415 if (rc) { 6416 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6417 return 0; 6418 } 6419 6420 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6421 (NULL != dev->set_master_volume)) { 6422 AutoMutex lock(mHardwareLock); 6423 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6424 dev->set_master_volume(dev, mMasterVolume); 6425 mHardwareStatus = AUDIO_HW_IDLE; 6426 } 6427 6428 audio_module_handle_t handle = nextUniqueId(); 6429 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6430 6431 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6432 name, dev->common.module->name, dev->common.module->id, handle); 6433 6434 return handle; 6435 6436} 6437 6438audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6439 audio_devices_t *pDevices, 6440 uint32_t *pSamplingRate, 6441 audio_format_t *pFormat, 6442 audio_channel_mask_t *pChannelMask, 6443 uint32_t *pLatencyMs, 6444 audio_output_flags_t flags) 6445{ 6446 status_t status; 6447 PlaybackThread *thread = NULL; 6448 struct audio_config config = { 6449 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6450 channel_mask: pChannelMask ? *pChannelMask : 0, 6451 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6452 }; 6453 audio_stream_out_t *outStream = NULL; 6454 audio_hw_device_t *outHwDev; 6455 6456 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6457 module, 6458 (pDevices != NULL) ? (int)*pDevices : 0, 6459 config.sample_rate, 6460 config.format, 6461 config.channel_mask, 6462 flags); 6463 6464 if (pDevices == NULL || *pDevices == 0) { 6465 return 0; 6466 } 6467 6468 Mutex::Autolock _l(mLock); 6469 6470 outHwDev = findSuitableHwDev_l(module, *pDevices); 6471 if (outHwDev == NULL) 6472 return 0; 6473 6474 audio_io_handle_t id = nextUniqueId(); 6475 6476 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6477 6478 status = outHwDev->open_output_stream(outHwDev, 6479 id, 6480 *pDevices, 6481 (audio_output_flags_t)flags, 6482 &config, 6483 &outStream); 6484 6485 mHardwareStatus = AUDIO_HW_IDLE; 6486 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6487 outStream, 6488 config.sample_rate, 6489 config.format, 6490 config.channel_mask, 6491 status); 6492 6493 if (status == NO_ERROR && outStream != NULL) { 6494 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6495 6496 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6497 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6498 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6499 thread = new DirectOutputThread(this, output, id, *pDevices); 6500 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6501 } else { 6502 thread = new MixerThread(this, output, id, *pDevices); 6503 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6504 } 6505 mPlaybackThreads.add(id, thread); 6506 6507 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6508 if (pFormat != NULL) *pFormat = config.format; 6509 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6510 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6511 6512 // notify client processes of the new output creation 6513 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6514 6515 // the first primary output opened designates the primary hw device 6516 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6517 ALOGI("Using module %d has the primary audio interface", module); 6518 mPrimaryHardwareDev = outHwDev; 6519 6520 AutoMutex lock(mHardwareLock); 6521 mHardwareStatus = AUDIO_HW_SET_MODE; 6522 outHwDev->set_mode(outHwDev, mMode); 6523 6524 // Determine the level of master volume support the primary audio HAL has, 6525 // and set the initial master volume at the same time. 6526 float initialVolume = 1.0; 6527 mMasterVolumeSupportLvl = MVS_NONE; 6528 6529 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6530 if ((NULL != outHwDev->get_master_volume) && 6531 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6532 mMasterVolumeSupportLvl = MVS_FULL; 6533 } else { 6534 mMasterVolumeSupportLvl = MVS_SETONLY; 6535 initialVolume = 1.0; 6536 } 6537 6538 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6539 if ((NULL == outHwDev->set_master_volume) || 6540 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6541 mMasterVolumeSupportLvl = MVS_NONE; 6542 } 6543 // now that we have a primary device, initialize master volume on other devices 6544 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6545 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6546 6547 if ((dev != mPrimaryHardwareDev) && 6548 (NULL != dev->set_master_volume)) { 6549 dev->set_master_volume(dev, initialVolume); 6550 } 6551 } 6552 mHardwareStatus = AUDIO_HW_IDLE; 6553 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6554 ? initialVolume 6555 : 1.0; 6556 mMasterVolume = initialVolume; 6557 } 6558 return id; 6559 } 6560 6561 return 0; 6562} 6563 6564audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6565 audio_io_handle_t output2) 6566{ 6567 Mutex::Autolock _l(mLock); 6568 MixerThread *thread1 = checkMixerThread_l(output1); 6569 MixerThread *thread2 = checkMixerThread_l(output2); 6570 6571 if (thread1 == NULL || thread2 == NULL) { 6572 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6573 return 0; 6574 } 6575 6576 audio_io_handle_t id = nextUniqueId(); 6577 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6578 thread->addOutputTrack(thread2); 6579 mPlaybackThreads.add(id, thread); 6580 // notify client processes of the new output creation 6581 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6582 return id; 6583} 6584 6585status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6586{ 6587 // keep strong reference on the playback thread so that 6588 // it is not destroyed while exit() is executed 6589 sp<PlaybackThread> thread; 6590 { 6591 Mutex::Autolock _l(mLock); 6592 thread = checkPlaybackThread_l(output); 6593 if (thread == NULL) { 6594 return BAD_VALUE; 6595 } 6596 6597 ALOGV("closeOutput() %d", output); 6598 6599 if (thread->type() == ThreadBase::MIXER) { 6600 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6601 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6602 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6603 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6604 } 6605 } 6606 } 6607 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6608 mPlaybackThreads.removeItem(output); 6609 } 6610 thread->exit(); 6611 // The thread entity (active unit of execution) is no longer running here, 6612 // but the ThreadBase container still exists. 6613 6614 if (thread->type() != ThreadBase::DUPLICATING) { 6615 AudioStreamOut *out = thread->clearOutput(); 6616 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6617 // from now on thread->mOutput is NULL 6618 out->hwDev->close_output_stream(out->hwDev, out->stream); 6619 delete out; 6620 } 6621 return NO_ERROR; 6622} 6623 6624status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6625{ 6626 Mutex::Autolock _l(mLock); 6627 PlaybackThread *thread = checkPlaybackThread_l(output); 6628 6629 if (thread == NULL) { 6630 return BAD_VALUE; 6631 } 6632 6633 ALOGV("suspendOutput() %d", output); 6634 thread->suspend(); 6635 6636 return NO_ERROR; 6637} 6638 6639status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6640{ 6641 Mutex::Autolock _l(mLock); 6642 PlaybackThread *thread = checkPlaybackThread_l(output); 6643 6644 if (thread == NULL) { 6645 return BAD_VALUE; 6646 } 6647 6648 ALOGV("restoreOutput() %d", output); 6649 6650 thread->restore(); 6651 6652 return NO_ERROR; 6653} 6654 6655audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6656 audio_devices_t *pDevices, 6657 uint32_t *pSamplingRate, 6658 audio_format_t *pFormat, 6659 uint32_t *pChannelMask) 6660{ 6661 status_t status; 6662 RecordThread *thread = NULL; 6663 struct audio_config config = { 6664 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6665 channel_mask: pChannelMask ? *pChannelMask : 0, 6666 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6667 }; 6668 uint32_t reqSamplingRate = config.sample_rate; 6669 audio_format_t reqFormat = config.format; 6670 audio_channel_mask_t reqChannels = config.channel_mask; 6671 audio_stream_in_t *inStream = NULL; 6672 audio_hw_device_t *inHwDev; 6673 6674 if (pDevices == NULL || *pDevices == 0) { 6675 return 0; 6676 } 6677 6678 Mutex::Autolock _l(mLock); 6679 6680 inHwDev = findSuitableHwDev_l(module, *pDevices); 6681 if (inHwDev == NULL) 6682 return 0; 6683 6684 audio_io_handle_t id = nextUniqueId(); 6685 6686 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6687 &inStream); 6688 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6689 inStream, 6690 config.sample_rate, 6691 config.format, 6692 config.channel_mask, 6693 status); 6694 6695 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6696 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6697 // or stereo to mono conversions on 16 bit PCM inputs. 6698 if (status == BAD_VALUE && 6699 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6700 (config.sample_rate <= 2 * reqSamplingRate) && 6701 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6702 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6703 inStream = NULL; 6704 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6705 } 6706 6707 if (status == NO_ERROR && inStream != NULL) { 6708 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6709 6710 // Start record thread 6711 // RecorThread require both input and output device indication to forward to audio 6712 // pre processing modules 6713 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6714 thread = new RecordThread(this, 6715 input, 6716 reqSamplingRate, 6717 reqChannels, 6718 id, 6719 device); 6720 mRecordThreads.add(id, thread); 6721 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6722 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6723 if (pFormat != NULL) *pFormat = config.format; 6724 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6725 6726 input->stream->common.standby(&input->stream->common); 6727 6728 // notify client processes of the new input creation 6729 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6730 return id; 6731 } 6732 6733 return 0; 6734} 6735 6736status_t AudioFlinger::closeInput(audio_io_handle_t input) 6737{ 6738 // keep strong reference on the record thread so that 6739 // it is not destroyed while exit() is executed 6740 sp<RecordThread> thread; 6741 { 6742 Mutex::Autolock _l(mLock); 6743 thread = checkRecordThread_l(input); 6744 if (thread == NULL) { 6745 return BAD_VALUE; 6746 } 6747 6748 ALOGV("closeInput() %d", input); 6749 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6750 mRecordThreads.removeItem(input); 6751 } 6752 thread->exit(); 6753 // The thread entity (active unit of execution) is no longer running here, 6754 // but the ThreadBase container still exists. 6755 6756 AudioStreamIn *in = thread->clearInput(); 6757 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6758 // from now on thread->mInput is NULL 6759 in->hwDev->close_input_stream(in->hwDev, in->stream); 6760 delete in; 6761 6762 return NO_ERROR; 6763} 6764 6765status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6766{ 6767 Mutex::Autolock _l(mLock); 6768 MixerThread *dstThread = checkMixerThread_l(output); 6769 if (dstThread == NULL) { 6770 ALOGW("setStreamOutput() bad output id %d", output); 6771 return BAD_VALUE; 6772 } 6773 6774 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6775 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6776 6777 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6778 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6779 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6780 MixerThread *srcThread = (MixerThread *)thread; 6781 srcThread->invalidateTracks(stream); 6782 } 6783 } 6784 6785 return NO_ERROR; 6786} 6787 6788 6789int AudioFlinger::newAudioSessionId() 6790{ 6791 return nextUniqueId(); 6792} 6793 6794void AudioFlinger::acquireAudioSessionId(int audioSession) 6795{ 6796 Mutex::Autolock _l(mLock); 6797 pid_t caller = IPCThreadState::self()->getCallingPid(); 6798 ALOGV("acquiring %d from %d", audioSession, caller); 6799 size_t num = mAudioSessionRefs.size(); 6800 for (size_t i = 0; i< num; i++) { 6801 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6802 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6803 ref->mCnt++; 6804 ALOGV(" incremented refcount to %d", ref->mCnt); 6805 return; 6806 } 6807 } 6808 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6809 ALOGV(" added new entry for %d", audioSession); 6810} 6811 6812void AudioFlinger::releaseAudioSessionId(int audioSession) 6813{ 6814 Mutex::Autolock _l(mLock); 6815 pid_t caller = IPCThreadState::self()->getCallingPid(); 6816 ALOGV("releasing %d from %d", audioSession, caller); 6817 size_t num = mAudioSessionRefs.size(); 6818 for (size_t i = 0; i< num; i++) { 6819 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6820 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6821 ref->mCnt--; 6822 ALOGV(" decremented refcount to %d", ref->mCnt); 6823 if (ref->mCnt == 0) { 6824 mAudioSessionRefs.removeAt(i); 6825 delete ref; 6826 purgeStaleEffects_l(); 6827 } 6828 return; 6829 } 6830 } 6831 ALOGW("session id %d not found for pid %d", audioSession, caller); 6832} 6833 6834void AudioFlinger::purgeStaleEffects_l() { 6835 6836 ALOGV("purging stale effects"); 6837 6838 Vector< sp<EffectChain> > chains; 6839 6840 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6841 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6842 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6843 sp<EffectChain> ec = t->mEffectChains[j]; 6844 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6845 chains.push(ec); 6846 } 6847 } 6848 } 6849 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6850 sp<RecordThread> t = mRecordThreads.valueAt(i); 6851 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6852 sp<EffectChain> ec = t->mEffectChains[j]; 6853 chains.push(ec); 6854 } 6855 } 6856 6857 for (size_t i = 0; i < chains.size(); i++) { 6858 sp<EffectChain> ec = chains[i]; 6859 int sessionid = ec->sessionId(); 6860 sp<ThreadBase> t = ec->mThread.promote(); 6861 if (t == 0) { 6862 continue; 6863 } 6864 size_t numsessionrefs = mAudioSessionRefs.size(); 6865 bool found = false; 6866 for (size_t k = 0; k < numsessionrefs; k++) { 6867 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6868 if (ref->mSessionid == sessionid) { 6869 ALOGV(" session %d still exists for %d with %d refs", 6870 sessionid, ref->mPid, ref->mCnt); 6871 found = true; 6872 break; 6873 } 6874 } 6875 if (!found) { 6876 // remove all effects from the chain 6877 while (ec->mEffects.size()) { 6878 sp<EffectModule> effect = ec->mEffects[0]; 6879 effect->unPin(); 6880 Mutex::Autolock _l (t->mLock); 6881 t->removeEffect_l(effect); 6882 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6883 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6884 if (handle != 0) { 6885 handle->mEffect.clear(); 6886 if (handle->mHasControl && handle->mEnabled) { 6887 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6888 } 6889 } 6890 } 6891 AudioSystem::unregisterEffect(effect->id()); 6892 } 6893 } 6894 } 6895 return; 6896} 6897 6898// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6899AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6900{ 6901 return mPlaybackThreads.valueFor(output).get(); 6902} 6903 6904// checkMixerThread_l() must be called with AudioFlinger::mLock held 6905AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6906{ 6907 PlaybackThread *thread = checkPlaybackThread_l(output); 6908 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6909} 6910 6911// checkRecordThread_l() must be called with AudioFlinger::mLock held 6912AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6913{ 6914 return mRecordThreads.valueFor(input).get(); 6915} 6916 6917uint32_t AudioFlinger::nextUniqueId() 6918{ 6919 return android_atomic_inc(&mNextUniqueId); 6920} 6921 6922AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6923{ 6924 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6925 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6926 AudioStreamOut *output = thread->getOutput(); 6927 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6928 return thread; 6929 } 6930 } 6931 return NULL; 6932} 6933 6934uint32_t AudioFlinger::primaryOutputDevice_l() const 6935{ 6936 PlaybackThread *thread = primaryPlaybackThread_l(); 6937 6938 if (thread == NULL) { 6939 return 0; 6940 } 6941 6942 return thread->device(); 6943} 6944 6945sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6946 int triggerSession, 6947 int listenerSession, 6948 sync_event_callback_t callBack, 6949 void *cookie) 6950{ 6951 Mutex::Autolock _l(mLock); 6952 6953 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6954 status_t playStatus = NAME_NOT_FOUND; 6955 status_t recStatus = NAME_NOT_FOUND; 6956 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6957 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6958 if (playStatus == NO_ERROR) { 6959 return event; 6960 } 6961 } 6962 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6963 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6964 if (recStatus == NO_ERROR) { 6965 return event; 6966 } 6967 } 6968 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6969 mPendingSyncEvents.add(event); 6970 } else { 6971 ALOGV("createSyncEvent() invalid event %d", event->type()); 6972 event.clear(); 6973 } 6974 return event; 6975} 6976 6977// ---------------------------------------------------------------------------- 6978// Effect management 6979// ---------------------------------------------------------------------------- 6980 6981 6982status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6983{ 6984 Mutex::Autolock _l(mLock); 6985 return EffectQueryNumberEffects(numEffects); 6986} 6987 6988status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6989{ 6990 Mutex::Autolock _l(mLock); 6991 return EffectQueryEffect(index, descriptor); 6992} 6993 6994status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6995 effect_descriptor_t *descriptor) const 6996{ 6997 Mutex::Autolock _l(mLock); 6998 return EffectGetDescriptor(pUuid, descriptor); 6999} 7000 7001 7002sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7003 effect_descriptor_t *pDesc, 7004 const sp<IEffectClient>& effectClient, 7005 int32_t priority, 7006 audio_io_handle_t io, 7007 int sessionId, 7008 status_t *status, 7009 int *id, 7010 int *enabled) 7011{ 7012 status_t lStatus = NO_ERROR; 7013 sp<EffectHandle> handle; 7014 effect_descriptor_t desc; 7015 7016 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7017 pid, effectClient.get(), priority, sessionId, io); 7018 7019 if (pDesc == NULL) { 7020 lStatus = BAD_VALUE; 7021 goto Exit; 7022 } 7023 7024 // check audio settings permission for global effects 7025 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7026 lStatus = PERMISSION_DENIED; 7027 goto Exit; 7028 } 7029 7030 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7031 // that can only be created by audio policy manager (running in same process) 7032 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7033 lStatus = PERMISSION_DENIED; 7034 goto Exit; 7035 } 7036 7037 if (io == 0) { 7038 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7039 // output must be specified by AudioPolicyManager when using session 7040 // AUDIO_SESSION_OUTPUT_STAGE 7041 lStatus = BAD_VALUE; 7042 goto Exit; 7043 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7044 // if the output returned by getOutputForEffect() is removed before we lock the 7045 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7046 // and we will exit safely 7047 io = AudioSystem::getOutputForEffect(&desc); 7048 } 7049 } 7050 7051 { 7052 Mutex::Autolock _l(mLock); 7053 7054 7055 if (!EffectIsNullUuid(&pDesc->uuid)) { 7056 // if uuid is specified, request effect descriptor 7057 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7058 if (lStatus < 0) { 7059 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7060 goto Exit; 7061 } 7062 } else { 7063 // if uuid is not specified, look for an available implementation 7064 // of the required type in effect factory 7065 if (EffectIsNullUuid(&pDesc->type)) { 7066 ALOGW("createEffect() no effect type"); 7067 lStatus = BAD_VALUE; 7068 goto Exit; 7069 } 7070 uint32_t numEffects = 0; 7071 effect_descriptor_t d; 7072 d.flags = 0; // prevent compiler warning 7073 bool found = false; 7074 7075 lStatus = EffectQueryNumberEffects(&numEffects); 7076 if (lStatus < 0) { 7077 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7078 goto Exit; 7079 } 7080 for (uint32_t i = 0; i < numEffects; i++) { 7081 lStatus = EffectQueryEffect(i, &desc); 7082 if (lStatus < 0) { 7083 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7084 continue; 7085 } 7086 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7087 // If matching type found save effect descriptor. If the session is 7088 // 0 and the effect is not auxiliary, continue enumeration in case 7089 // an auxiliary version of this effect type is available 7090 found = true; 7091 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7092 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7093 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7094 break; 7095 } 7096 } 7097 } 7098 if (!found) { 7099 lStatus = BAD_VALUE; 7100 ALOGW("createEffect() effect not found"); 7101 goto Exit; 7102 } 7103 // For same effect type, chose auxiliary version over insert version if 7104 // connect to output mix (Compliance to OpenSL ES) 7105 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7106 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7107 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7108 } 7109 } 7110 7111 // Do not allow auxiliary effects on a session different from 0 (output mix) 7112 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7113 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7114 lStatus = INVALID_OPERATION; 7115 goto Exit; 7116 } 7117 7118 // check recording permission for visualizer 7119 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7120 !recordingAllowed()) { 7121 lStatus = PERMISSION_DENIED; 7122 goto Exit; 7123 } 7124 7125 // return effect descriptor 7126 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7127 7128 // If output is not specified try to find a matching audio session ID in one of the 7129 // output threads. 7130 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7131 // because of code checking output when entering the function. 7132 // Note: io is never 0 when creating an effect on an input 7133 if (io == 0) { 7134 // look for the thread where the specified audio session is present 7135 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7136 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7137 io = mPlaybackThreads.keyAt(i); 7138 break; 7139 } 7140 } 7141 if (io == 0) { 7142 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7143 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7144 io = mRecordThreads.keyAt(i); 7145 break; 7146 } 7147 } 7148 } 7149 // If no output thread contains the requested session ID, default to 7150 // first output. The effect chain will be moved to the correct output 7151 // thread when a track with the same session ID is created 7152 if (io == 0 && mPlaybackThreads.size()) { 7153 io = mPlaybackThreads.keyAt(0); 7154 } 7155 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7156 } 7157 ThreadBase *thread = checkRecordThread_l(io); 7158 if (thread == NULL) { 7159 thread = checkPlaybackThread_l(io); 7160 if (thread == NULL) { 7161 ALOGE("createEffect() unknown output thread"); 7162 lStatus = BAD_VALUE; 7163 goto Exit; 7164 } 7165 } 7166 7167 sp<Client> client = registerPid_l(pid); 7168 7169 // create effect on selected output thread 7170 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7171 &desc, enabled, &lStatus); 7172 if (handle != 0 && id != NULL) { 7173 *id = handle->id(); 7174 } 7175 } 7176 7177Exit: 7178 if (status != NULL) { 7179 *status = lStatus; 7180 } 7181 return handle; 7182} 7183 7184status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7185 audio_io_handle_t dstOutput) 7186{ 7187 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7188 sessionId, srcOutput, dstOutput); 7189 Mutex::Autolock _l(mLock); 7190 if (srcOutput == dstOutput) { 7191 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7192 return NO_ERROR; 7193 } 7194 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7195 if (srcThread == NULL) { 7196 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7197 return BAD_VALUE; 7198 } 7199 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7200 if (dstThread == NULL) { 7201 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7202 return BAD_VALUE; 7203 } 7204 7205 Mutex::Autolock _dl(dstThread->mLock); 7206 Mutex::Autolock _sl(srcThread->mLock); 7207 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7208 7209 return NO_ERROR; 7210} 7211 7212// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7213status_t AudioFlinger::moveEffectChain_l(int sessionId, 7214 AudioFlinger::PlaybackThread *srcThread, 7215 AudioFlinger::PlaybackThread *dstThread, 7216 bool reRegister) 7217{ 7218 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7219 sessionId, srcThread, dstThread); 7220 7221 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7222 if (chain == 0) { 7223 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7224 sessionId, srcThread); 7225 return INVALID_OPERATION; 7226 } 7227 7228 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7229 // so that a new chain is created with correct parameters when first effect is added. This is 7230 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7231 // removed. 7232 srcThread->removeEffectChain_l(chain); 7233 7234 // transfer all effects one by one so that new effect chain is created on new thread with 7235 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7236 audio_io_handle_t dstOutput = dstThread->id(); 7237 sp<EffectChain> dstChain; 7238 uint32_t strategy = 0; // prevent compiler warning 7239 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7240 while (effect != 0) { 7241 srcThread->removeEffect_l(effect); 7242 dstThread->addEffect_l(effect); 7243 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7244 if (effect->state() == EffectModule::ACTIVE || 7245 effect->state() == EffectModule::STOPPING) { 7246 effect->start(); 7247 } 7248 // if the move request is not received from audio policy manager, the effect must be 7249 // re-registered with the new strategy and output 7250 if (dstChain == 0) { 7251 dstChain = effect->chain().promote(); 7252 if (dstChain == 0) { 7253 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7254 srcThread->addEffect_l(effect); 7255 return NO_INIT; 7256 } 7257 strategy = dstChain->strategy(); 7258 } 7259 if (reRegister) { 7260 AudioSystem::unregisterEffect(effect->id()); 7261 AudioSystem::registerEffect(&effect->desc(), 7262 dstOutput, 7263 strategy, 7264 sessionId, 7265 effect->id()); 7266 } 7267 effect = chain->getEffectFromId_l(0); 7268 } 7269 7270 return NO_ERROR; 7271} 7272 7273 7274// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7275sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7276 const sp<AudioFlinger::Client>& client, 7277 const sp<IEffectClient>& effectClient, 7278 int32_t priority, 7279 int sessionId, 7280 effect_descriptor_t *desc, 7281 int *enabled, 7282 status_t *status 7283 ) 7284{ 7285 sp<EffectModule> effect; 7286 sp<EffectHandle> handle; 7287 status_t lStatus; 7288 sp<EffectChain> chain; 7289 bool chainCreated = false; 7290 bool effectCreated = false; 7291 bool effectRegistered = false; 7292 7293 lStatus = initCheck(); 7294 if (lStatus != NO_ERROR) { 7295 ALOGW("createEffect_l() Audio driver not initialized."); 7296 goto Exit; 7297 } 7298 7299 // Do not allow effects with session ID 0 on direct output or duplicating threads 7300 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7301 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7302 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7303 desc->name, sessionId); 7304 lStatus = BAD_VALUE; 7305 goto Exit; 7306 } 7307 // Only Pre processor effects are allowed on input threads and only on input threads 7308 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7309 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7310 desc->name, desc->flags, mType); 7311 lStatus = BAD_VALUE; 7312 goto Exit; 7313 } 7314 7315 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7316 7317 { // scope for mLock 7318 Mutex::Autolock _l(mLock); 7319 7320 // check for existing effect chain with the requested audio session 7321 chain = getEffectChain_l(sessionId); 7322 if (chain == 0) { 7323 // create a new chain for this session 7324 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7325 chain = new EffectChain(this, sessionId); 7326 addEffectChain_l(chain); 7327 chain->setStrategy(getStrategyForSession_l(sessionId)); 7328 chainCreated = true; 7329 } else { 7330 effect = chain->getEffectFromDesc_l(desc); 7331 } 7332 7333 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7334 7335 if (effect == 0) { 7336 int id = mAudioFlinger->nextUniqueId(); 7337 // Check CPU and memory usage 7338 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7339 if (lStatus != NO_ERROR) { 7340 goto Exit; 7341 } 7342 effectRegistered = true; 7343 // create a new effect module if none present in the chain 7344 effect = new EffectModule(this, chain, desc, id, sessionId); 7345 lStatus = effect->status(); 7346 if (lStatus != NO_ERROR) { 7347 goto Exit; 7348 } 7349 lStatus = chain->addEffect_l(effect); 7350 if (lStatus != NO_ERROR) { 7351 goto Exit; 7352 } 7353 effectCreated = true; 7354 7355 effect->setDevice(mDevice); 7356 effect->setMode(mAudioFlinger->getMode()); 7357 } 7358 // create effect handle and connect it to effect module 7359 handle = new EffectHandle(effect, client, effectClient, priority); 7360 lStatus = effect->addHandle(handle); 7361 if (enabled != NULL) { 7362 *enabled = (int)effect->isEnabled(); 7363 } 7364 } 7365 7366Exit: 7367 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7368 Mutex::Autolock _l(mLock); 7369 if (effectCreated) { 7370 chain->removeEffect_l(effect); 7371 } 7372 if (effectRegistered) { 7373 AudioSystem::unregisterEffect(effect->id()); 7374 } 7375 if (chainCreated) { 7376 removeEffectChain_l(chain); 7377 } 7378 handle.clear(); 7379 } 7380 7381 if (status != NULL) { 7382 *status = lStatus; 7383 } 7384 return handle; 7385} 7386 7387sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7388{ 7389 sp<EffectChain> chain = getEffectChain_l(sessionId); 7390 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7391} 7392 7393// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7394// PlaybackThread::mLock held 7395status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7396{ 7397 // check for existing effect chain with the requested audio session 7398 int sessionId = effect->sessionId(); 7399 sp<EffectChain> chain = getEffectChain_l(sessionId); 7400 bool chainCreated = false; 7401 7402 if (chain == 0) { 7403 // create a new chain for this session 7404 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7405 chain = new EffectChain(this, sessionId); 7406 addEffectChain_l(chain); 7407 chain->setStrategy(getStrategyForSession_l(sessionId)); 7408 chainCreated = true; 7409 } 7410 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7411 7412 if (chain->getEffectFromId_l(effect->id()) != 0) { 7413 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7414 this, effect->desc().name, chain.get()); 7415 return BAD_VALUE; 7416 } 7417 7418 status_t status = chain->addEffect_l(effect); 7419 if (status != NO_ERROR) { 7420 if (chainCreated) { 7421 removeEffectChain_l(chain); 7422 } 7423 return status; 7424 } 7425 7426 effect->setDevice(mDevice); 7427 effect->setMode(mAudioFlinger->getMode()); 7428 return NO_ERROR; 7429} 7430 7431void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7432 7433 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7434 effect_descriptor_t desc = effect->desc(); 7435 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7436 detachAuxEffect_l(effect->id()); 7437 } 7438 7439 sp<EffectChain> chain = effect->chain().promote(); 7440 if (chain != 0) { 7441 // remove effect chain if removing last effect 7442 if (chain->removeEffect_l(effect) == 0) { 7443 removeEffectChain_l(chain); 7444 } 7445 } else { 7446 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7447 } 7448} 7449 7450void AudioFlinger::ThreadBase::lockEffectChains_l( 7451 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7452{ 7453 effectChains = mEffectChains; 7454 for (size_t i = 0; i < mEffectChains.size(); i++) { 7455 mEffectChains[i]->lock(); 7456 } 7457} 7458 7459void AudioFlinger::ThreadBase::unlockEffectChains( 7460 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7461{ 7462 for (size_t i = 0; i < effectChains.size(); i++) { 7463 effectChains[i]->unlock(); 7464 } 7465} 7466 7467sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7468{ 7469 Mutex::Autolock _l(mLock); 7470 return getEffectChain_l(sessionId); 7471} 7472 7473sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7474{ 7475 size_t size = mEffectChains.size(); 7476 for (size_t i = 0; i < size; i++) { 7477 if (mEffectChains[i]->sessionId() == sessionId) { 7478 return mEffectChains[i]; 7479 } 7480 } 7481 return 0; 7482} 7483 7484void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7485{ 7486 Mutex::Autolock _l(mLock); 7487 size_t size = mEffectChains.size(); 7488 for (size_t i = 0; i < size; i++) { 7489 mEffectChains[i]->setMode_l(mode); 7490 } 7491} 7492 7493void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7494 const wp<EffectHandle>& handle, 7495 bool unpinIfLast) { 7496 7497 Mutex::Autolock _l(mLock); 7498 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7499 // delete the effect module if removing last handle on it 7500 if (effect->removeHandle(handle) == 0) { 7501 if (!effect->isPinned() || unpinIfLast) { 7502 removeEffect_l(effect); 7503 AudioSystem::unregisterEffect(effect->id()); 7504 } 7505 } 7506} 7507 7508status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7509{ 7510 int session = chain->sessionId(); 7511 int16_t *buffer = mMixBuffer; 7512 bool ownsBuffer = false; 7513 7514 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7515 if (session > 0) { 7516 // Only one effect chain can be present in direct output thread and it uses 7517 // the mix buffer as input 7518 if (mType != DIRECT) { 7519 size_t numSamples = mNormalFrameCount * mChannelCount; 7520 buffer = new int16_t[numSamples]; 7521 memset(buffer, 0, numSamples * sizeof(int16_t)); 7522 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7523 ownsBuffer = true; 7524 } 7525 7526 // Attach all tracks with same session ID to this chain. 7527 for (size_t i = 0; i < mTracks.size(); ++i) { 7528 sp<Track> track = mTracks[i]; 7529 if (session == track->sessionId()) { 7530 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7531 track->setMainBuffer(buffer); 7532 chain->incTrackCnt(); 7533 } 7534 } 7535 7536 // indicate all active tracks in the chain 7537 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7538 sp<Track> track = mActiveTracks[i].promote(); 7539 if (track == 0) continue; 7540 if (session == track->sessionId()) { 7541 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7542 chain->incActiveTrackCnt(); 7543 } 7544 } 7545 } 7546 7547 chain->setInBuffer(buffer, ownsBuffer); 7548 chain->setOutBuffer(mMixBuffer); 7549 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7550 // chains list in order to be processed last as it contains output stage effects 7551 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7552 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7553 // after track specific effects and before output stage 7554 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7555 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7556 // Effect chain for other sessions are inserted at beginning of effect 7557 // chains list to be processed before output mix effects. Relative order between other 7558 // sessions is not important 7559 size_t size = mEffectChains.size(); 7560 size_t i = 0; 7561 for (i = 0; i < size; i++) { 7562 if (mEffectChains[i]->sessionId() < session) break; 7563 } 7564 mEffectChains.insertAt(chain, i); 7565 checkSuspendOnAddEffectChain_l(chain); 7566 7567 return NO_ERROR; 7568} 7569 7570size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7571{ 7572 int session = chain->sessionId(); 7573 7574 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7575 7576 for (size_t i = 0; i < mEffectChains.size(); i++) { 7577 if (chain == mEffectChains[i]) { 7578 mEffectChains.removeAt(i); 7579 // detach all active tracks from the chain 7580 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7581 sp<Track> track = mActiveTracks[i].promote(); 7582 if (track == 0) continue; 7583 if (session == track->sessionId()) { 7584 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7585 chain.get(), session); 7586 chain->decActiveTrackCnt(); 7587 } 7588 } 7589 7590 // detach all tracks with same session ID from this chain 7591 for (size_t i = 0; i < mTracks.size(); ++i) { 7592 sp<Track> track = mTracks[i]; 7593 if (session == track->sessionId()) { 7594 track->setMainBuffer(mMixBuffer); 7595 chain->decTrackCnt(); 7596 } 7597 } 7598 break; 7599 } 7600 } 7601 return mEffectChains.size(); 7602} 7603 7604status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7605 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7606{ 7607 Mutex::Autolock _l(mLock); 7608 return attachAuxEffect_l(track, EffectId); 7609} 7610 7611status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7612 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7613{ 7614 status_t status = NO_ERROR; 7615 7616 if (EffectId == 0) { 7617 track->setAuxBuffer(0, NULL); 7618 } else { 7619 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7620 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7621 if (effect != 0) { 7622 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7623 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7624 } else { 7625 status = INVALID_OPERATION; 7626 } 7627 } else { 7628 status = BAD_VALUE; 7629 } 7630 } 7631 return status; 7632} 7633 7634void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7635{ 7636 for (size_t i = 0; i < mTracks.size(); ++i) { 7637 sp<Track> track = mTracks[i]; 7638 if (track->auxEffectId() == effectId) { 7639 attachAuxEffect_l(track, 0); 7640 } 7641 } 7642} 7643 7644status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7645{ 7646 // only one chain per input thread 7647 if (mEffectChains.size() != 0) { 7648 return INVALID_OPERATION; 7649 } 7650 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7651 7652 chain->setInBuffer(NULL); 7653 chain->setOutBuffer(NULL); 7654 7655 checkSuspendOnAddEffectChain_l(chain); 7656 7657 mEffectChains.add(chain); 7658 7659 return NO_ERROR; 7660} 7661 7662size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7663{ 7664 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7665 ALOGW_IF(mEffectChains.size() != 1, 7666 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7667 chain.get(), mEffectChains.size(), this); 7668 if (mEffectChains.size() == 1) { 7669 mEffectChains.removeAt(0); 7670 } 7671 return 0; 7672} 7673 7674// ---------------------------------------------------------------------------- 7675// EffectModule implementation 7676// ---------------------------------------------------------------------------- 7677 7678#undef LOG_TAG 7679#define LOG_TAG "AudioFlinger::EffectModule" 7680 7681AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7682 const wp<AudioFlinger::EffectChain>& chain, 7683 effect_descriptor_t *desc, 7684 int id, 7685 int sessionId) 7686 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7687 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7688{ 7689 ALOGV("Constructor %p", this); 7690 int lStatus; 7691 if (thread == NULL) { 7692 return; 7693 } 7694 7695 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7696 7697 // create effect engine from effect factory 7698 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7699 7700 if (mStatus != NO_ERROR) { 7701 return; 7702 } 7703 lStatus = init(); 7704 if (lStatus < 0) { 7705 mStatus = lStatus; 7706 goto Error; 7707 } 7708 7709 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7710 mPinned = true; 7711 } 7712 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7713 return; 7714Error: 7715 EffectRelease(mEffectInterface); 7716 mEffectInterface = NULL; 7717 ALOGV("Constructor Error %d", mStatus); 7718} 7719 7720AudioFlinger::EffectModule::~EffectModule() 7721{ 7722 ALOGV("Destructor %p", this); 7723 if (mEffectInterface != NULL) { 7724 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7725 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7726 sp<ThreadBase> thread = mThread.promote(); 7727 if (thread != 0) { 7728 audio_stream_t *stream = thread->stream(); 7729 if (stream != NULL) { 7730 stream->remove_audio_effect(stream, mEffectInterface); 7731 } 7732 } 7733 } 7734 // release effect engine 7735 EffectRelease(mEffectInterface); 7736 } 7737} 7738 7739status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7740{ 7741 status_t status; 7742 7743 Mutex::Autolock _l(mLock); 7744 int priority = handle->priority(); 7745 size_t size = mHandles.size(); 7746 sp<EffectHandle> h; 7747 size_t i; 7748 for (i = 0; i < size; i++) { 7749 h = mHandles[i].promote(); 7750 if (h == 0) continue; 7751 if (h->priority() <= priority) break; 7752 } 7753 // if inserted in first place, move effect control from previous owner to this handle 7754 if (i == 0) { 7755 bool enabled = false; 7756 if (h != 0) { 7757 enabled = h->enabled(); 7758 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7759 } 7760 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7761 status = NO_ERROR; 7762 } else { 7763 status = ALREADY_EXISTS; 7764 } 7765 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7766 mHandles.insertAt(handle, i); 7767 return status; 7768} 7769 7770size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7771{ 7772 Mutex::Autolock _l(mLock); 7773 size_t size = mHandles.size(); 7774 size_t i; 7775 for (i = 0; i < size; i++) { 7776 if (mHandles[i] == handle) break; 7777 } 7778 if (i == size) { 7779 return size; 7780 } 7781 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7782 7783 bool enabled = false; 7784 EffectHandle *hdl = handle.unsafe_get(); 7785 if (hdl != NULL) { 7786 ALOGV("removeHandle() unsafe_get OK"); 7787 enabled = hdl->enabled(); 7788 } 7789 mHandles.removeAt(i); 7790 size = mHandles.size(); 7791 // if removed from first place, move effect control from this handle to next in line 7792 if (i == 0 && size != 0) { 7793 sp<EffectHandle> h = mHandles[0].promote(); 7794 if (h != 0) { 7795 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7796 } 7797 } 7798 7799 // Prevent calls to process() and other functions on effect interface from now on. 7800 // The effect engine will be released by the destructor when the last strong reference on 7801 // this object is released which can happen after next process is called. 7802 if (size == 0 && !mPinned) { 7803 mState = DESTROYED; 7804 } 7805 7806 return size; 7807} 7808 7809sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7810{ 7811 Mutex::Autolock _l(mLock); 7812 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7813} 7814 7815void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7816{ 7817 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7818 // keep a strong reference on this EffectModule to avoid calling the 7819 // destructor before we exit 7820 sp<EffectModule> keep(this); 7821 { 7822 sp<ThreadBase> thread = mThread.promote(); 7823 if (thread != 0) { 7824 thread->disconnectEffect(keep, handle, unpinIfLast); 7825 } 7826 } 7827} 7828 7829void AudioFlinger::EffectModule::updateState() { 7830 Mutex::Autolock _l(mLock); 7831 7832 switch (mState) { 7833 case RESTART: 7834 reset_l(); 7835 // FALL THROUGH 7836 7837 case STARTING: 7838 // clear auxiliary effect input buffer for next accumulation 7839 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7840 memset(mConfig.inputCfg.buffer.raw, 7841 0, 7842 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7843 } 7844 start_l(); 7845 mState = ACTIVE; 7846 break; 7847 case STOPPING: 7848 stop_l(); 7849 mDisableWaitCnt = mMaxDisableWaitCnt; 7850 mState = STOPPED; 7851 break; 7852 case STOPPED: 7853 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7854 // turn off sequence. 7855 if (--mDisableWaitCnt == 0) { 7856 reset_l(); 7857 mState = IDLE; 7858 } 7859 break; 7860 default: //IDLE , ACTIVE, DESTROYED 7861 break; 7862 } 7863} 7864 7865void AudioFlinger::EffectModule::process() 7866{ 7867 Mutex::Autolock _l(mLock); 7868 7869 if (mState == DESTROYED || mEffectInterface == NULL || 7870 mConfig.inputCfg.buffer.raw == NULL || 7871 mConfig.outputCfg.buffer.raw == NULL) { 7872 return; 7873 } 7874 7875 if (isProcessEnabled()) { 7876 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7877 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7878 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7879 mConfig.inputCfg.buffer.s32, 7880 mConfig.inputCfg.buffer.frameCount/2); 7881 } 7882 7883 // do the actual processing in the effect engine 7884 int ret = (*mEffectInterface)->process(mEffectInterface, 7885 &mConfig.inputCfg.buffer, 7886 &mConfig.outputCfg.buffer); 7887 7888 // force transition to IDLE state when engine is ready 7889 if (mState == STOPPED && ret == -ENODATA) { 7890 mDisableWaitCnt = 1; 7891 } 7892 7893 // clear auxiliary effect input buffer for next accumulation 7894 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7895 memset(mConfig.inputCfg.buffer.raw, 0, 7896 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7897 } 7898 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7899 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7900 // If an insert effect is idle and input buffer is different from output buffer, 7901 // accumulate input onto output 7902 sp<EffectChain> chain = mChain.promote(); 7903 if (chain != 0 && chain->activeTrackCnt() != 0) { 7904 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7905 int16_t *in = mConfig.inputCfg.buffer.s16; 7906 int16_t *out = mConfig.outputCfg.buffer.s16; 7907 for (size_t i = 0; i < frameCnt; i++) { 7908 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7909 } 7910 } 7911 } 7912} 7913 7914void AudioFlinger::EffectModule::reset_l() 7915{ 7916 if (mEffectInterface == NULL) { 7917 return; 7918 } 7919 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7920} 7921 7922status_t AudioFlinger::EffectModule::configure() 7923{ 7924 uint32_t channels; 7925 if (mEffectInterface == NULL) { 7926 return NO_INIT; 7927 } 7928 7929 sp<ThreadBase> thread = mThread.promote(); 7930 if (thread == 0) { 7931 return DEAD_OBJECT; 7932 } 7933 7934 // TODO: handle configuration of effects replacing track process 7935 if (thread->channelCount() == 1) { 7936 channels = AUDIO_CHANNEL_OUT_MONO; 7937 } else { 7938 channels = AUDIO_CHANNEL_OUT_STEREO; 7939 } 7940 7941 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7942 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7943 } else { 7944 mConfig.inputCfg.channels = channels; 7945 } 7946 mConfig.outputCfg.channels = channels; 7947 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7948 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7949 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7950 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7951 mConfig.inputCfg.bufferProvider.cookie = NULL; 7952 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7953 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7954 mConfig.outputCfg.bufferProvider.cookie = NULL; 7955 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7956 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7957 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7958 // Insert effect: 7959 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7960 // always overwrites output buffer: input buffer == output buffer 7961 // - in other sessions: 7962 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7963 // other effect: overwrites output buffer: input buffer == output buffer 7964 // Auxiliary effect: 7965 // accumulates in output buffer: input buffer != output buffer 7966 // Therefore: accumulate <=> input buffer != output buffer 7967 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7968 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7969 } else { 7970 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7971 } 7972 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7973 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7974 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7975 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7976 7977 ALOGV("configure() %p thread %p buffer %p framecount %d", 7978 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7979 7980 status_t cmdStatus; 7981 uint32_t size = sizeof(int); 7982 status_t status = (*mEffectInterface)->command(mEffectInterface, 7983 EFFECT_CMD_SET_CONFIG, 7984 sizeof(effect_config_t), 7985 &mConfig, 7986 &size, 7987 &cmdStatus); 7988 if (status == 0) { 7989 status = cmdStatus; 7990 } 7991 7992 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7993 (1000 * mConfig.outputCfg.buffer.frameCount); 7994 7995 return status; 7996} 7997 7998status_t AudioFlinger::EffectModule::init() 7999{ 8000 Mutex::Autolock _l(mLock); 8001 if (mEffectInterface == NULL) { 8002 return NO_INIT; 8003 } 8004 status_t cmdStatus; 8005 uint32_t size = sizeof(status_t); 8006 status_t status = (*mEffectInterface)->command(mEffectInterface, 8007 EFFECT_CMD_INIT, 8008 0, 8009 NULL, 8010 &size, 8011 &cmdStatus); 8012 if (status == 0) { 8013 status = cmdStatus; 8014 } 8015 return status; 8016} 8017 8018status_t AudioFlinger::EffectModule::start() 8019{ 8020 Mutex::Autolock _l(mLock); 8021 return start_l(); 8022} 8023 8024status_t AudioFlinger::EffectModule::start_l() 8025{ 8026 if (mEffectInterface == NULL) { 8027 return NO_INIT; 8028 } 8029 status_t cmdStatus; 8030 uint32_t size = sizeof(status_t); 8031 status_t status = (*mEffectInterface)->command(mEffectInterface, 8032 EFFECT_CMD_ENABLE, 8033 0, 8034 NULL, 8035 &size, 8036 &cmdStatus); 8037 if (status == 0) { 8038 status = cmdStatus; 8039 } 8040 if (status == 0 && 8041 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8042 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8043 sp<ThreadBase> thread = mThread.promote(); 8044 if (thread != 0) { 8045 audio_stream_t *stream = thread->stream(); 8046 if (stream != NULL) { 8047 stream->add_audio_effect(stream, mEffectInterface); 8048 } 8049 } 8050 } 8051 return status; 8052} 8053 8054status_t AudioFlinger::EffectModule::stop() 8055{ 8056 Mutex::Autolock _l(mLock); 8057 return stop_l(); 8058} 8059 8060status_t AudioFlinger::EffectModule::stop_l() 8061{ 8062 if (mEffectInterface == NULL) { 8063 return NO_INIT; 8064 } 8065 status_t cmdStatus; 8066 uint32_t size = sizeof(status_t); 8067 status_t status = (*mEffectInterface)->command(mEffectInterface, 8068 EFFECT_CMD_DISABLE, 8069 0, 8070 NULL, 8071 &size, 8072 &cmdStatus); 8073 if (status == 0) { 8074 status = cmdStatus; 8075 } 8076 if (status == 0 && 8077 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8078 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8079 sp<ThreadBase> thread = mThread.promote(); 8080 if (thread != 0) { 8081 audio_stream_t *stream = thread->stream(); 8082 if (stream != NULL) { 8083 stream->remove_audio_effect(stream, mEffectInterface); 8084 } 8085 } 8086 } 8087 return status; 8088} 8089 8090status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8091 uint32_t cmdSize, 8092 void *pCmdData, 8093 uint32_t *replySize, 8094 void *pReplyData) 8095{ 8096 Mutex::Autolock _l(mLock); 8097// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8098 8099 if (mState == DESTROYED || mEffectInterface == NULL) { 8100 return NO_INIT; 8101 } 8102 status_t status = (*mEffectInterface)->command(mEffectInterface, 8103 cmdCode, 8104 cmdSize, 8105 pCmdData, 8106 replySize, 8107 pReplyData); 8108 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8109 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8110 for (size_t i = 1; i < mHandles.size(); i++) { 8111 sp<EffectHandle> h = mHandles[i].promote(); 8112 if (h != 0) { 8113 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8114 } 8115 } 8116 } 8117 return status; 8118} 8119 8120status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8121{ 8122 8123 Mutex::Autolock _l(mLock); 8124 ALOGV("setEnabled %p enabled %d", this, enabled); 8125 8126 if (enabled != isEnabled()) { 8127 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8128 if (enabled && status != NO_ERROR) { 8129 return status; 8130 } 8131 8132 switch (mState) { 8133 // going from disabled to enabled 8134 case IDLE: 8135 mState = STARTING; 8136 break; 8137 case STOPPED: 8138 mState = RESTART; 8139 break; 8140 case STOPPING: 8141 mState = ACTIVE; 8142 break; 8143 8144 // going from enabled to disabled 8145 case RESTART: 8146 mState = STOPPED; 8147 break; 8148 case STARTING: 8149 mState = IDLE; 8150 break; 8151 case ACTIVE: 8152 mState = STOPPING; 8153 break; 8154 case DESTROYED: 8155 return NO_ERROR; // simply ignore as we are being destroyed 8156 } 8157 for (size_t i = 1; i < mHandles.size(); i++) { 8158 sp<EffectHandle> h = mHandles[i].promote(); 8159 if (h != 0) { 8160 h->setEnabled(enabled); 8161 } 8162 } 8163 } 8164 return NO_ERROR; 8165} 8166 8167bool AudioFlinger::EffectModule::isEnabled() const 8168{ 8169 switch (mState) { 8170 case RESTART: 8171 case STARTING: 8172 case ACTIVE: 8173 return true; 8174 case IDLE: 8175 case STOPPING: 8176 case STOPPED: 8177 case DESTROYED: 8178 default: 8179 return false; 8180 } 8181} 8182 8183bool AudioFlinger::EffectModule::isProcessEnabled() const 8184{ 8185 switch (mState) { 8186 case RESTART: 8187 case ACTIVE: 8188 case STOPPING: 8189 case STOPPED: 8190 return true; 8191 case IDLE: 8192 case STARTING: 8193 case DESTROYED: 8194 default: 8195 return false; 8196 } 8197} 8198 8199status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8200{ 8201 Mutex::Autolock _l(mLock); 8202 status_t status = NO_ERROR; 8203 8204 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8205 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8206 if (isProcessEnabled() && 8207 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8208 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8209 status_t cmdStatus; 8210 uint32_t volume[2]; 8211 uint32_t *pVolume = NULL; 8212 uint32_t size = sizeof(volume); 8213 volume[0] = *left; 8214 volume[1] = *right; 8215 if (controller) { 8216 pVolume = volume; 8217 } 8218 status = (*mEffectInterface)->command(mEffectInterface, 8219 EFFECT_CMD_SET_VOLUME, 8220 size, 8221 volume, 8222 &size, 8223 pVolume); 8224 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8225 *left = volume[0]; 8226 *right = volume[1]; 8227 } 8228 } 8229 return status; 8230} 8231 8232status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8233{ 8234 Mutex::Autolock _l(mLock); 8235 status_t status = NO_ERROR; 8236 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8237 // audio pre processing modules on RecordThread can receive both output and 8238 // input device indication in the same call 8239 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8240 if (dev) { 8241 status_t cmdStatus; 8242 uint32_t size = sizeof(status_t); 8243 8244 status = (*mEffectInterface)->command(mEffectInterface, 8245 EFFECT_CMD_SET_DEVICE, 8246 sizeof(uint32_t), 8247 &dev, 8248 &size, 8249 &cmdStatus); 8250 if (status == NO_ERROR) { 8251 status = cmdStatus; 8252 } 8253 } 8254 dev = device & AUDIO_DEVICE_IN_ALL; 8255 if (dev) { 8256 status_t cmdStatus; 8257 uint32_t size = sizeof(status_t); 8258 8259 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8260 EFFECT_CMD_SET_INPUT_DEVICE, 8261 sizeof(uint32_t), 8262 &dev, 8263 &size, 8264 &cmdStatus); 8265 if (status2 == NO_ERROR) { 8266 status2 = cmdStatus; 8267 } 8268 if (status == NO_ERROR) { 8269 status = status2; 8270 } 8271 } 8272 } 8273 return status; 8274} 8275 8276status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8277{ 8278 Mutex::Autolock _l(mLock); 8279 status_t status = NO_ERROR; 8280 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8281 status_t cmdStatus; 8282 uint32_t size = sizeof(status_t); 8283 status = (*mEffectInterface)->command(mEffectInterface, 8284 EFFECT_CMD_SET_AUDIO_MODE, 8285 sizeof(audio_mode_t), 8286 &mode, 8287 &size, 8288 &cmdStatus); 8289 if (status == NO_ERROR) { 8290 status = cmdStatus; 8291 } 8292 } 8293 return status; 8294} 8295 8296void AudioFlinger::EffectModule::setSuspended(bool suspended) 8297{ 8298 Mutex::Autolock _l(mLock); 8299 mSuspended = suspended; 8300} 8301 8302bool AudioFlinger::EffectModule::suspended() const 8303{ 8304 Mutex::Autolock _l(mLock); 8305 return mSuspended; 8306} 8307 8308status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8309{ 8310 const size_t SIZE = 256; 8311 char buffer[SIZE]; 8312 String8 result; 8313 8314 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8315 result.append(buffer); 8316 8317 bool locked = tryLock(mLock); 8318 // failed to lock - AudioFlinger is probably deadlocked 8319 if (!locked) { 8320 result.append("\t\tCould not lock Fx mutex:\n"); 8321 } 8322 8323 result.append("\t\tSession Status State Engine:\n"); 8324 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8325 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8326 result.append(buffer); 8327 8328 result.append("\t\tDescriptor:\n"); 8329 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8330 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8331 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8332 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8333 result.append(buffer); 8334 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8335 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8336 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8337 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8338 result.append(buffer); 8339 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8340 mDescriptor.apiVersion, 8341 mDescriptor.flags); 8342 result.append(buffer); 8343 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8344 mDescriptor.name); 8345 result.append(buffer); 8346 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8347 mDescriptor.implementor); 8348 result.append(buffer); 8349 8350 result.append("\t\t- Input configuration:\n"); 8351 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8352 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8353 (uint32_t)mConfig.inputCfg.buffer.raw, 8354 mConfig.inputCfg.buffer.frameCount, 8355 mConfig.inputCfg.samplingRate, 8356 mConfig.inputCfg.channels, 8357 mConfig.inputCfg.format); 8358 result.append(buffer); 8359 8360 result.append("\t\t- Output configuration:\n"); 8361 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8362 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8363 (uint32_t)mConfig.outputCfg.buffer.raw, 8364 mConfig.outputCfg.buffer.frameCount, 8365 mConfig.outputCfg.samplingRate, 8366 mConfig.outputCfg.channels, 8367 mConfig.outputCfg.format); 8368 result.append(buffer); 8369 8370 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8371 result.append(buffer); 8372 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8373 for (size_t i = 0; i < mHandles.size(); ++i) { 8374 sp<EffectHandle> handle = mHandles[i].promote(); 8375 if (handle != 0) { 8376 handle->dump(buffer, SIZE); 8377 result.append(buffer); 8378 } 8379 } 8380 8381 result.append("\n"); 8382 8383 write(fd, result.string(), result.length()); 8384 8385 if (locked) { 8386 mLock.unlock(); 8387 } 8388 8389 return NO_ERROR; 8390} 8391 8392// ---------------------------------------------------------------------------- 8393// EffectHandle implementation 8394// ---------------------------------------------------------------------------- 8395 8396#undef LOG_TAG 8397#define LOG_TAG "AudioFlinger::EffectHandle" 8398 8399AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8400 const sp<AudioFlinger::Client>& client, 8401 const sp<IEffectClient>& effectClient, 8402 int32_t priority) 8403 : BnEffect(), 8404 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8405 mPriority(priority), mHasControl(false), mEnabled(false) 8406{ 8407 ALOGV("constructor %p", this); 8408 8409 if (client == 0) { 8410 return; 8411 } 8412 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8413 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8414 if (mCblkMemory != 0) { 8415 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8416 8417 if (mCblk != NULL) { 8418 new(mCblk) effect_param_cblk_t(); 8419 mBuffer = (uint8_t *)mCblk + bufOffset; 8420 } 8421 } else { 8422 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8423 return; 8424 } 8425} 8426 8427AudioFlinger::EffectHandle::~EffectHandle() 8428{ 8429 ALOGV("Destructor %p", this); 8430 disconnect(false); 8431 ALOGV("Destructor DONE %p", this); 8432} 8433 8434status_t AudioFlinger::EffectHandle::enable() 8435{ 8436 ALOGV("enable %p", this); 8437 if (!mHasControl) return INVALID_OPERATION; 8438 if (mEffect == 0) return DEAD_OBJECT; 8439 8440 if (mEnabled) { 8441 return NO_ERROR; 8442 } 8443 8444 mEnabled = true; 8445 8446 sp<ThreadBase> thread = mEffect->thread().promote(); 8447 if (thread != 0) { 8448 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8449 } 8450 8451 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8452 if (mEffect->suspended()) { 8453 return NO_ERROR; 8454 } 8455 8456 status_t status = mEffect->setEnabled(true); 8457 if (status != NO_ERROR) { 8458 if (thread != 0) { 8459 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8460 } 8461 mEnabled = false; 8462 } 8463 return status; 8464} 8465 8466status_t AudioFlinger::EffectHandle::disable() 8467{ 8468 ALOGV("disable %p", this); 8469 if (!mHasControl) return INVALID_OPERATION; 8470 if (mEffect == 0) return DEAD_OBJECT; 8471 8472 if (!mEnabled) { 8473 return NO_ERROR; 8474 } 8475 mEnabled = false; 8476 8477 if (mEffect->suspended()) { 8478 return NO_ERROR; 8479 } 8480 8481 status_t status = mEffect->setEnabled(false); 8482 8483 sp<ThreadBase> thread = mEffect->thread().promote(); 8484 if (thread != 0) { 8485 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8486 } 8487 8488 return status; 8489} 8490 8491void AudioFlinger::EffectHandle::disconnect() 8492{ 8493 disconnect(true); 8494} 8495 8496void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8497{ 8498 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8499 if (mEffect == 0) { 8500 return; 8501 } 8502 mEffect->disconnect(this, unpinIfLast); 8503 8504 if (mHasControl && mEnabled) { 8505 sp<ThreadBase> thread = mEffect->thread().promote(); 8506 if (thread != 0) { 8507 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8508 } 8509 } 8510 8511 // release sp on module => module destructor can be called now 8512 mEffect.clear(); 8513 if (mClient != 0) { 8514 if (mCblk != NULL) { 8515 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8516 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8517 } 8518 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8519 // Client destructor must run with AudioFlinger mutex locked 8520 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8521 mClient.clear(); 8522 } 8523} 8524 8525status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8526 uint32_t cmdSize, 8527 void *pCmdData, 8528 uint32_t *replySize, 8529 void *pReplyData) 8530{ 8531// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8532// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8533 8534 // only get parameter command is permitted for applications not controlling the effect 8535 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8536 return INVALID_OPERATION; 8537 } 8538 if (mEffect == 0) return DEAD_OBJECT; 8539 if (mClient == 0) return INVALID_OPERATION; 8540 8541 // handle commands that are not forwarded transparently to effect engine 8542 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8543 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8544 // no risk to block the whole media server process or mixer threads is we are stuck here 8545 Mutex::Autolock _l(mCblk->lock); 8546 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8547 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8548 mCblk->serverIndex = 0; 8549 mCblk->clientIndex = 0; 8550 return BAD_VALUE; 8551 } 8552 status_t status = NO_ERROR; 8553 while (mCblk->serverIndex < mCblk->clientIndex) { 8554 int reply; 8555 uint32_t rsize = sizeof(int); 8556 int *p = (int *)(mBuffer + mCblk->serverIndex); 8557 int size = *p++; 8558 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8559 ALOGW("command(): invalid parameter block size"); 8560 break; 8561 } 8562 effect_param_t *param = (effect_param_t *)p; 8563 if (param->psize == 0 || param->vsize == 0) { 8564 ALOGW("command(): null parameter or value size"); 8565 mCblk->serverIndex += size; 8566 continue; 8567 } 8568 uint32_t psize = sizeof(effect_param_t) + 8569 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8570 param->vsize; 8571 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8572 psize, 8573 p, 8574 &rsize, 8575 &reply); 8576 // stop at first error encountered 8577 if (ret != NO_ERROR) { 8578 status = ret; 8579 *(int *)pReplyData = reply; 8580 break; 8581 } else if (reply != NO_ERROR) { 8582 *(int *)pReplyData = reply; 8583 break; 8584 } 8585 mCblk->serverIndex += size; 8586 } 8587 mCblk->serverIndex = 0; 8588 mCblk->clientIndex = 0; 8589 return status; 8590 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8591 *(int *)pReplyData = NO_ERROR; 8592 return enable(); 8593 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8594 *(int *)pReplyData = NO_ERROR; 8595 return disable(); 8596 } 8597 8598 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8599} 8600 8601void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8602{ 8603 ALOGV("setControl %p control %d", this, hasControl); 8604 8605 mHasControl = hasControl; 8606 mEnabled = enabled; 8607 8608 if (signal && mEffectClient != 0) { 8609 mEffectClient->controlStatusChanged(hasControl); 8610 } 8611} 8612 8613void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8614 uint32_t cmdSize, 8615 void *pCmdData, 8616 uint32_t replySize, 8617 void *pReplyData) 8618{ 8619 if (mEffectClient != 0) { 8620 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8621 } 8622} 8623 8624 8625 8626void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8627{ 8628 if (mEffectClient != 0) { 8629 mEffectClient->enableStatusChanged(enabled); 8630 } 8631} 8632 8633status_t AudioFlinger::EffectHandle::onTransact( 8634 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8635{ 8636 return BnEffect::onTransact(code, data, reply, flags); 8637} 8638 8639 8640void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8641{ 8642 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8643 8644 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8645 (mClient == 0) ? getpid_cached : mClient->pid(), 8646 mPriority, 8647 mHasControl, 8648 !locked, 8649 mCblk ? mCblk->clientIndex : 0, 8650 mCblk ? mCblk->serverIndex : 0 8651 ); 8652 8653 if (locked) { 8654 mCblk->lock.unlock(); 8655 } 8656} 8657 8658#undef LOG_TAG 8659#define LOG_TAG "AudioFlinger::EffectChain" 8660 8661AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8662 int sessionId) 8663 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8664 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8665 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8666{ 8667 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8668 if (thread == NULL) { 8669 return; 8670 } 8671 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8672 thread->frameCount(); 8673} 8674 8675AudioFlinger::EffectChain::~EffectChain() 8676{ 8677 if (mOwnInBuffer) { 8678 delete mInBuffer; 8679 } 8680 8681} 8682 8683// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8684sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8685{ 8686 size_t size = mEffects.size(); 8687 8688 for (size_t i = 0; i < size; i++) { 8689 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8690 return mEffects[i]; 8691 } 8692 } 8693 return 0; 8694} 8695 8696// getEffectFromId_l() must be called with ThreadBase::mLock held 8697sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8698{ 8699 size_t size = mEffects.size(); 8700 8701 for (size_t i = 0; i < size; i++) { 8702 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8703 if (id == 0 || mEffects[i]->id() == id) { 8704 return mEffects[i]; 8705 } 8706 } 8707 return 0; 8708} 8709 8710// getEffectFromType_l() must be called with ThreadBase::mLock held 8711sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8712 const effect_uuid_t *type) 8713{ 8714 size_t size = mEffects.size(); 8715 8716 for (size_t i = 0; i < size; i++) { 8717 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8718 return mEffects[i]; 8719 } 8720 } 8721 return 0; 8722} 8723 8724// Must be called with EffectChain::mLock locked 8725void AudioFlinger::EffectChain::process_l() 8726{ 8727 sp<ThreadBase> thread = mThread.promote(); 8728 if (thread == 0) { 8729 ALOGW("process_l(): cannot promote mixer thread"); 8730 return; 8731 } 8732 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8733 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8734 // always process effects unless no more tracks are on the session and the effect tail 8735 // has been rendered 8736 bool doProcess = true; 8737 if (!isGlobalSession) { 8738 bool tracksOnSession = (trackCnt() != 0); 8739 8740 if (!tracksOnSession && mTailBufferCount == 0) { 8741 doProcess = false; 8742 } 8743 8744 if (activeTrackCnt() == 0) { 8745 // if no track is active and the effect tail has not been rendered, 8746 // the input buffer must be cleared here as the mixer process will not do it 8747 if (tracksOnSession || mTailBufferCount > 0) { 8748 size_t numSamples = thread->frameCount() * thread->channelCount(); 8749 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8750 if (mTailBufferCount > 0) { 8751 mTailBufferCount--; 8752 } 8753 } 8754 } 8755 } 8756 8757 size_t size = mEffects.size(); 8758 if (doProcess) { 8759 for (size_t i = 0; i < size; i++) { 8760 mEffects[i]->process(); 8761 } 8762 } 8763 for (size_t i = 0; i < size; i++) { 8764 mEffects[i]->updateState(); 8765 } 8766} 8767 8768// addEffect_l() must be called with PlaybackThread::mLock held 8769status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8770{ 8771 effect_descriptor_t desc = effect->desc(); 8772 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8773 8774 Mutex::Autolock _l(mLock); 8775 effect->setChain(this); 8776 sp<ThreadBase> thread = mThread.promote(); 8777 if (thread == 0) { 8778 return NO_INIT; 8779 } 8780 effect->setThread(thread); 8781 8782 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8783 // Auxiliary effects are inserted at the beginning of mEffects vector as 8784 // they are processed first and accumulated in chain input buffer 8785 mEffects.insertAt(effect, 0); 8786 8787 // the input buffer for auxiliary effect contains mono samples in 8788 // 32 bit format. This is to avoid saturation in AudoMixer 8789 // accumulation stage. Saturation is done in EffectModule::process() before 8790 // calling the process in effect engine 8791 size_t numSamples = thread->frameCount(); 8792 int32_t *buffer = new int32_t[numSamples]; 8793 memset(buffer, 0, numSamples * sizeof(int32_t)); 8794 effect->setInBuffer((int16_t *)buffer); 8795 // auxiliary effects output samples to chain input buffer for further processing 8796 // by insert effects 8797 effect->setOutBuffer(mInBuffer); 8798 } else { 8799 // Insert effects are inserted at the end of mEffects vector as they are processed 8800 // after track and auxiliary effects. 8801 // Insert effect order as a function of indicated preference: 8802 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8803 // another effect is present 8804 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8805 // last effect claiming first position 8806 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8807 // first effect claiming last position 8808 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8809 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8810 // already present 8811 8812 size_t size = mEffects.size(); 8813 size_t idx_insert = size; 8814 ssize_t idx_insert_first = -1; 8815 ssize_t idx_insert_last = -1; 8816 8817 for (size_t i = 0; i < size; i++) { 8818 effect_descriptor_t d = mEffects[i]->desc(); 8819 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8820 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8821 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8822 // check invalid effect chaining combinations 8823 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8824 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8825 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8826 return INVALID_OPERATION; 8827 } 8828 // remember position of first insert effect and by default 8829 // select this as insert position for new effect 8830 if (idx_insert == size) { 8831 idx_insert = i; 8832 } 8833 // remember position of last insert effect claiming 8834 // first position 8835 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8836 idx_insert_first = i; 8837 } 8838 // remember position of first insert effect claiming 8839 // last position 8840 if (iPref == EFFECT_FLAG_INSERT_LAST && 8841 idx_insert_last == -1) { 8842 idx_insert_last = i; 8843 } 8844 } 8845 } 8846 8847 // modify idx_insert from first position if needed 8848 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8849 if (idx_insert_last != -1) { 8850 idx_insert = idx_insert_last; 8851 } else { 8852 idx_insert = size; 8853 } 8854 } else { 8855 if (idx_insert_first != -1) { 8856 idx_insert = idx_insert_first + 1; 8857 } 8858 } 8859 8860 // always read samples from chain input buffer 8861 effect->setInBuffer(mInBuffer); 8862 8863 // if last effect in the chain, output samples to chain 8864 // output buffer, otherwise to chain input buffer 8865 if (idx_insert == size) { 8866 if (idx_insert != 0) { 8867 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8868 mEffects[idx_insert-1]->configure(); 8869 } 8870 effect->setOutBuffer(mOutBuffer); 8871 } else { 8872 effect->setOutBuffer(mInBuffer); 8873 } 8874 mEffects.insertAt(effect, idx_insert); 8875 8876 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8877 } 8878 effect->configure(); 8879 return NO_ERROR; 8880} 8881 8882// removeEffect_l() must be called with PlaybackThread::mLock held 8883size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8884{ 8885 Mutex::Autolock _l(mLock); 8886 size_t size = mEffects.size(); 8887 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8888 8889 for (size_t i = 0; i < size; i++) { 8890 if (effect == mEffects[i]) { 8891 // calling stop here will remove pre-processing effect from the audio HAL. 8892 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8893 // the middle of a read from audio HAL 8894 if (mEffects[i]->state() == EffectModule::ACTIVE || 8895 mEffects[i]->state() == EffectModule::STOPPING) { 8896 mEffects[i]->stop(); 8897 } 8898 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8899 delete[] effect->inBuffer(); 8900 } else { 8901 if (i == size - 1 && i != 0) { 8902 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8903 mEffects[i - 1]->configure(); 8904 } 8905 } 8906 mEffects.removeAt(i); 8907 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8908 break; 8909 } 8910 } 8911 8912 return mEffects.size(); 8913} 8914 8915// setDevice_l() must be called with PlaybackThread::mLock held 8916void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8917{ 8918 size_t size = mEffects.size(); 8919 for (size_t i = 0; i < size; i++) { 8920 mEffects[i]->setDevice(device); 8921 } 8922} 8923 8924// setMode_l() must be called with PlaybackThread::mLock held 8925void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8926{ 8927 size_t size = mEffects.size(); 8928 for (size_t i = 0; i < size; i++) { 8929 mEffects[i]->setMode(mode); 8930 } 8931} 8932 8933// setVolume_l() must be called with PlaybackThread::mLock held 8934bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8935{ 8936 uint32_t newLeft = *left; 8937 uint32_t newRight = *right; 8938 bool hasControl = false; 8939 int ctrlIdx = -1; 8940 size_t size = mEffects.size(); 8941 8942 // first update volume controller 8943 for (size_t i = size; i > 0; i--) { 8944 if (mEffects[i - 1]->isProcessEnabled() && 8945 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8946 ctrlIdx = i - 1; 8947 hasControl = true; 8948 break; 8949 } 8950 } 8951 8952 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8953 if (hasControl) { 8954 *left = mNewLeftVolume; 8955 *right = mNewRightVolume; 8956 } 8957 return hasControl; 8958 } 8959 8960 mVolumeCtrlIdx = ctrlIdx; 8961 mLeftVolume = newLeft; 8962 mRightVolume = newRight; 8963 8964 // second get volume update from volume controller 8965 if (ctrlIdx >= 0) { 8966 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8967 mNewLeftVolume = newLeft; 8968 mNewRightVolume = newRight; 8969 } 8970 // then indicate volume to all other effects in chain. 8971 // Pass altered volume to effects before volume controller 8972 // and requested volume to effects after controller 8973 uint32_t lVol = newLeft; 8974 uint32_t rVol = newRight; 8975 8976 for (size_t i = 0; i < size; i++) { 8977 if ((int)i == ctrlIdx) continue; 8978 // this also works for ctrlIdx == -1 when there is no volume controller 8979 if ((int)i > ctrlIdx) { 8980 lVol = *left; 8981 rVol = *right; 8982 } 8983 mEffects[i]->setVolume(&lVol, &rVol, false); 8984 } 8985 *left = newLeft; 8986 *right = newRight; 8987 8988 return hasControl; 8989} 8990 8991status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8992{ 8993 const size_t SIZE = 256; 8994 char buffer[SIZE]; 8995 String8 result; 8996 8997 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8998 result.append(buffer); 8999 9000 bool locked = tryLock(mLock); 9001 // failed to lock - AudioFlinger is probably deadlocked 9002 if (!locked) { 9003 result.append("\tCould not lock mutex:\n"); 9004 } 9005 9006 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9007 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9008 mEffects.size(), 9009 (uint32_t)mInBuffer, 9010 (uint32_t)mOutBuffer, 9011 mActiveTrackCnt); 9012 result.append(buffer); 9013 write(fd, result.string(), result.size()); 9014 9015 for (size_t i = 0; i < mEffects.size(); ++i) { 9016 sp<EffectModule> effect = mEffects[i]; 9017 if (effect != 0) { 9018 effect->dump(fd, args); 9019 } 9020 } 9021 9022 if (locked) { 9023 mLock.unlock(); 9024 } 9025 9026 return NO_ERROR; 9027} 9028 9029// must be called with ThreadBase::mLock held 9030void AudioFlinger::EffectChain::setEffectSuspended_l( 9031 const effect_uuid_t *type, bool suspend) 9032{ 9033 sp<SuspendedEffectDesc> desc; 9034 // use effect type UUID timelow as key as there is no real risk of identical 9035 // timeLow fields among effect type UUIDs. 9036 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9037 if (suspend) { 9038 if (index >= 0) { 9039 desc = mSuspendedEffects.valueAt(index); 9040 } else { 9041 desc = new SuspendedEffectDesc(); 9042 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9043 mSuspendedEffects.add(type->timeLow, desc); 9044 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9045 } 9046 if (desc->mRefCount++ == 0) { 9047 sp<EffectModule> effect = getEffectIfEnabled(type); 9048 if (effect != 0) { 9049 desc->mEffect = effect; 9050 effect->setSuspended(true); 9051 effect->setEnabled(false); 9052 } 9053 } 9054 } else { 9055 if (index < 0) { 9056 return; 9057 } 9058 desc = mSuspendedEffects.valueAt(index); 9059 if (desc->mRefCount <= 0) { 9060 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9061 desc->mRefCount = 1; 9062 } 9063 if (--desc->mRefCount == 0) { 9064 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9065 if (desc->mEffect != 0) { 9066 sp<EffectModule> effect = desc->mEffect.promote(); 9067 if (effect != 0) { 9068 effect->setSuspended(false); 9069 sp<EffectHandle> handle = effect->controlHandle(); 9070 if (handle != 0) { 9071 effect->setEnabled(handle->enabled()); 9072 } 9073 } 9074 desc->mEffect.clear(); 9075 } 9076 mSuspendedEffects.removeItemsAt(index); 9077 } 9078 } 9079} 9080 9081// must be called with ThreadBase::mLock held 9082void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9083{ 9084 sp<SuspendedEffectDesc> desc; 9085 9086 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9087 if (suspend) { 9088 if (index >= 0) { 9089 desc = mSuspendedEffects.valueAt(index); 9090 } else { 9091 desc = new SuspendedEffectDesc(); 9092 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9093 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9094 } 9095 if (desc->mRefCount++ == 0) { 9096 Vector< sp<EffectModule> > effects; 9097 getSuspendEligibleEffects(effects); 9098 for (size_t i = 0; i < effects.size(); i++) { 9099 setEffectSuspended_l(&effects[i]->desc().type, true); 9100 } 9101 } 9102 } else { 9103 if (index < 0) { 9104 return; 9105 } 9106 desc = mSuspendedEffects.valueAt(index); 9107 if (desc->mRefCount <= 0) { 9108 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9109 desc->mRefCount = 1; 9110 } 9111 if (--desc->mRefCount == 0) { 9112 Vector<const effect_uuid_t *> types; 9113 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9114 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9115 continue; 9116 } 9117 types.add(&mSuspendedEffects.valueAt(i)->mType); 9118 } 9119 for (size_t i = 0; i < types.size(); i++) { 9120 setEffectSuspended_l(types[i], false); 9121 } 9122 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9123 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9124 } 9125 } 9126} 9127 9128 9129// The volume effect is used for automated tests only 9130#ifndef OPENSL_ES_H_ 9131static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9132 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9133const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9134#endif //OPENSL_ES_H_ 9135 9136bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9137{ 9138 // auxiliary effects and visualizer are never suspended on output mix 9139 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9140 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9141 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9142 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9143 return false; 9144 } 9145 return true; 9146} 9147 9148void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9149{ 9150 effects.clear(); 9151 for (size_t i = 0; i < mEffects.size(); i++) { 9152 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9153 effects.add(mEffects[i]); 9154 } 9155 } 9156} 9157 9158sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9159 const effect_uuid_t *type) 9160{ 9161 sp<EffectModule> effect = getEffectFromType_l(type); 9162 return effect != 0 && effect->isEnabled() ? effect : 0; 9163} 9164 9165void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9166 bool enabled) 9167{ 9168 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9169 if (enabled) { 9170 if (index < 0) { 9171 // if the effect is not suspend check if all effects are suspended 9172 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9173 if (index < 0) { 9174 return; 9175 } 9176 if (!isEffectEligibleForSuspend(effect->desc())) { 9177 return; 9178 } 9179 setEffectSuspended_l(&effect->desc().type, enabled); 9180 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9181 if (index < 0) { 9182 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9183 return; 9184 } 9185 } 9186 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9187 effect->desc().type.timeLow); 9188 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9189 // if effect is requested to suspended but was not yet enabled, supend it now. 9190 if (desc->mEffect == 0) { 9191 desc->mEffect = effect; 9192 effect->setEnabled(false); 9193 effect->setSuspended(true); 9194 } 9195 } else { 9196 if (index < 0) { 9197 return; 9198 } 9199 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9200 effect->desc().type.timeLow); 9201 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9202 desc->mEffect.clear(); 9203 effect->setSuspended(false); 9204 } 9205} 9206 9207#undef LOG_TAG 9208#define LOG_TAG "AudioFlinger" 9209 9210// ---------------------------------------------------------------------------- 9211 9212status_t AudioFlinger::onTransact( 9213 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9214{ 9215 return BnAudioFlinger::onTransact(code, data, reply, flags); 9216} 9217 9218}; // namespace android 9219