AudioFlinger.cpp revision 44a957f06400a338e7af20b3d16c4c4ae22a673c
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//#define ATRACE_TAG ATRACE_TAG_AUDIO 23 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <binder/IPCThreadState.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41#include <cutils/compiler.h> 42 43#undef ADD_BATTERY_DATA 44 45#ifdef ADD_BATTERY_DATA 46#include <media/IMediaPlayerService.h> 47#include <media/IMediaDeathNotifier.h> 48#endif 49 50#include <private/media/AudioTrackShared.h> 51#include <private/media/AudioEffectShared.h> 52 53#include <system/audio.h> 54#include <hardware/audio.h> 55 56#include "AudioMixer.h" 57#include "AudioFlinger.h" 58#include "ServiceUtilities.h" 59 60#include <media/EffectsFactoryApi.h> 61#include <audio_effects/effect_visualizer.h> 62#include <audio_effects/effect_ns.h> 63#include <audio_effects/effect_aec.h> 64 65#include <audio_utils/primitives.h> 66 67#include <powermanager/PowerManager.h> 68 69// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 70#ifdef DEBUG_CPU_USAGE 71#include <cpustats/CentralTendencyStatistics.h> 72#include <cpustats/ThreadCpuUsage.h> 73#endif 74 75#include <common_time/cc_helper.h> 76#include <common_time/local_clock.h> 77 78#include "FastMixer.h" 79 80// NBAIO implementations 81#include "AudioStreamOutSink.h" 82#include "MonoPipe.h" 83#include "MonoPipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167// ---------------------------------------------------------------------------- 168 169#ifdef ADD_BATTERY_DATA 170// To collect the amplifier usage 171static void addBatteryData(uint32_t params) { 172 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 173 if (service == NULL) { 174 // it already logged 175 return; 176 } 177 178 service->addBatteryData(params); 179} 180#endif 181 182static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183{ 184 const hw_module_t *mod; 185 int rc; 186 187 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190 if (rc) { 191 goto out; 192 } 193 rc = audio_hw_device_open(mod, dev); 194 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 195 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196 if (rc) { 197 goto out; 198 } 199 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201 rc = BAD_VALUE; 202 goto out; 203 } 204 return 0; 205 206out: 207 *dev = NULL; 208 return rc; 209} 210 211// ---------------------------------------------------------------------------- 212 213AudioFlinger::AudioFlinger() 214 : BnAudioFlinger(), 215 mPrimaryHardwareDev(NULL), 216 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217 mMasterVolume(1.0f), 218 mMasterVolumeSupportLvl(MVS_NONE), 219 mMasterMute(false), 220 mNextUniqueId(1), 221 mMode(AUDIO_MODE_INVALID), 222 mBtNrecIsOff(false) 223{ 224} 225 226void AudioFlinger::onFirstRef() 227{ 228 int rc = 0; 229 230 Mutex::Autolock _l(mLock); 231 232 /* TODO: move all this work into an Init() function */ 233 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 235 uint32_t int_val; 236 if (1 == sscanf(val_str, "%u", &int_val)) { 237 mStandbyTimeInNsecs = milliseconds(int_val); 238 ALOGI("Using %u mSec as standby time.", int_val); 239 } else { 240 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241 ALOGI("Using default %u mSec as standby time.", 242 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243 } 244 } 245 246 mMode = AUDIO_MODE_NORMAL; 247 mMasterVolumeSW = 1.0; 248 mMasterVolume = 1.0; 249 mHardwareStatus = AUDIO_HW_IDLE; 250} 251 252AudioFlinger::~AudioFlinger() 253{ 254 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 478 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 479 if (mPlaybackThreads.keyAt(i) != output) { 480 // prevent same audio session on different output threads 481 uint32_t sessions = t->hasAudioSession(*sessionId); 482 if (sessions & PlaybackThread::TRACK_SESSION) { 483 ALOGE("createTrack() session ID %d already in use", *sessionId); 484 lStatus = BAD_VALUE; 485 goto Exit; 486 } 487 // check if an effect with same session ID is waiting for a track to be created 488 if (sessions & PlaybackThread::EFFECT_SESSION) { 489 effectThread = t.get(); 490 } 491 } 492 } 493 lSessionId = *sessionId; 494 } else { 495 // if no audio session id is provided, create one here 496 lSessionId = nextUniqueId(); 497 if (sessionId != NULL) { 498 *sessionId = lSessionId; 499 } 500 } 501 ALOGV("createTrack() lSessionId: %d", lSessionId); 502 503 track = thread->createTrack_l(client, streamType, sampleRate, format, 504 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 505 506 // move effect chain to this output thread if an effect on same session was waiting 507 // for a track to be created 508 if (lStatus == NO_ERROR && effectThread != NULL) { 509 Mutex::Autolock _dl(thread->mLock); 510 Mutex::Autolock _sl(effectThread->mLock); 511 moveEffectChain_l(lSessionId, effectThread, thread, true); 512 } 513 514 // Look for sync events awaiting for a session to be used. 515 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 516 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 517 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 518 if (lStatus == NO_ERROR) { 519 track->setSyncEvent(mPendingSyncEvents[i]); 520 } else { 521 mPendingSyncEvents[i]->cancel(); 522 } 523 mPendingSyncEvents.removeAt(i); 524 i--; 525 } 526 } 527 } 528 } 529 if (lStatus == NO_ERROR) { 530 trackHandle = new TrackHandle(track); 531 } else { 532 // remove local strong reference to Client before deleting the Track so that the Client 533 // destructor is called by the TrackBase destructor with mLock held 534 client.clear(); 535 track.clear(); 536 } 537 538Exit: 539 if (status != NULL) { 540 *status = lStatus; 541 } 542 return trackHandle; 543} 544 545uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 546{ 547 Mutex::Autolock _l(mLock); 548 PlaybackThread *thread = checkPlaybackThread_l(output); 549 if (thread == NULL) { 550 ALOGW("sampleRate() unknown thread %d", output); 551 return 0; 552 } 553 return thread->sampleRate(); 554} 555 556int AudioFlinger::channelCount(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("channelCount() unknown thread %d", output); 562 return 0; 563 } 564 return thread->channelCount(); 565} 566 567audio_format_t AudioFlinger::format(audio_io_handle_t output) const 568{ 569 Mutex::Autolock _l(mLock); 570 PlaybackThread *thread = checkPlaybackThread_l(output); 571 if (thread == NULL) { 572 ALOGW("format() unknown thread %d", output); 573 return AUDIO_FORMAT_INVALID; 574 } 575 return thread->format(); 576} 577 578size_t AudioFlinger::frameCount(audio_io_handle_t output) const 579{ 580 Mutex::Autolock _l(mLock); 581 PlaybackThread *thread = checkPlaybackThread_l(output); 582 if (thread == NULL) { 583 ALOGW("frameCount() unknown thread %d", output); 584 return 0; 585 } 586 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 587 // should examine all callers and fix them to handle smaller counts 588 return thread->frameCount(); 589} 590 591uint32_t AudioFlinger::latency(audio_io_handle_t output) const 592{ 593 Mutex::Autolock _l(mLock); 594 PlaybackThread *thread = checkPlaybackThread_l(output); 595 if (thread == NULL) { 596 ALOGW("latency() unknown thread %d", output); 597 return 0; 598 } 599 return thread->latency(); 600} 601 602status_t AudioFlinger::setMasterVolume(float value) 603{ 604 status_t ret = initCheck(); 605 if (ret != NO_ERROR) { 606 return ret; 607 } 608 609 // check calling permissions 610 if (!settingsAllowed()) { 611 return PERMISSION_DENIED; 612 } 613 614 float swmv = value; 615 616 Mutex::Autolock _l(mLock); 617 618 // when hw supports master volume, don't scale in sw mixer 619 if (MVS_NONE != mMasterVolumeSupportLvl) { 620 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 621 AutoMutex lock(mHardwareLock); 622 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 623 624 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 625 if (NULL != dev->set_master_volume) { 626 dev->set_master_volume(dev, value); 627 } 628 mHardwareStatus = AUDIO_HW_IDLE; 629 } 630 631 swmv = 1.0; 632 } 633 634 mMasterVolume = value; 635 mMasterVolumeSW = swmv; 636 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 637 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 638 639 return NO_ERROR; 640} 641 642status_t AudioFlinger::setMode(audio_mode_t mode) 643{ 644 status_t ret = initCheck(); 645 if (ret != NO_ERROR) { 646 return ret; 647 } 648 649 // check calling permissions 650 if (!settingsAllowed()) { 651 return PERMISSION_DENIED; 652 } 653 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 654 ALOGW("Illegal value: setMode(%d)", mode); 655 return BAD_VALUE; 656 } 657 658 { // scope for the lock 659 AutoMutex lock(mHardwareLock); 660 mHardwareStatus = AUDIO_HW_SET_MODE; 661 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 662 mHardwareStatus = AUDIO_HW_IDLE; 663 } 664 665 if (NO_ERROR == ret) { 666 Mutex::Autolock _l(mLock); 667 mMode = mode; 668 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 669 mPlaybackThreads.valueAt(i)->setMode(mode); 670 } 671 672 return ret; 673} 674 675status_t AudioFlinger::setMicMute(bool state) 676{ 677 status_t ret = initCheck(); 678 if (ret != NO_ERROR) { 679 return ret; 680 } 681 682 // check calling permissions 683 if (!settingsAllowed()) { 684 return PERMISSION_DENIED; 685 } 686 687 AutoMutex lock(mHardwareLock); 688 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 689 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return ret; 692} 693 694bool AudioFlinger::getMicMute() const 695{ 696 status_t ret = initCheck(); 697 if (ret != NO_ERROR) { 698 return false; 699 } 700 701 bool state = AUDIO_MODE_INVALID; 702 AutoMutex lock(mHardwareLock); 703 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 704 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 705 mHardwareStatus = AUDIO_HW_IDLE; 706 return state; 707} 708 709status_t AudioFlinger::setMasterMute(bool muted) 710{ 711 // check calling permissions 712 if (!settingsAllowed()) { 713 return PERMISSION_DENIED; 714 } 715 716 Mutex::Autolock _l(mLock); 717 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 718 mMasterMute = muted; 719 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 720 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 721 722 return NO_ERROR; 723} 724 725float AudioFlinger::masterVolume() const 726{ 727 Mutex::Autolock _l(mLock); 728 return masterVolume_l(); 729} 730 731float AudioFlinger::masterVolumeSW() const 732{ 733 Mutex::Autolock _l(mLock); 734 return masterVolumeSW_l(); 735} 736 737bool AudioFlinger::masterMute() const 738{ 739 Mutex::Autolock _l(mLock); 740 return masterMute_l(); 741} 742 743float AudioFlinger::masterVolume_l() const 744{ 745 if (MVS_FULL == mMasterVolumeSupportLvl) { 746 float ret_val; 747 AutoMutex lock(mHardwareLock); 748 749 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 750 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 751 (NULL != mPrimaryHardwareDev->get_master_volume), 752 "can't get master volume"); 753 754 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 755 mHardwareStatus = AUDIO_HW_IDLE; 756 return ret_val; 757 } 758 759 return mMasterVolume; 760} 761 762status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 763 audio_io_handle_t output) 764{ 765 // check calling permissions 766 if (!settingsAllowed()) { 767 return PERMISSION_DENIED; 768 } 769 770 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 771 ALOGE("setStreamVolume() invalid stream %d", stream); 772 return BAD_VALUE; 773 } 774 775 AutoMutex lock(mLock); 776 PlaybackThread *thread = NULL; 777 if (output) { 778 thread = checkPlaybackThread_l(output); 779 if (thread == NULL) { 780 return BAD_VALUE; 781 } 782 } 783 784 mStreamTypes[stream].volume = value; 785 786 if (thread == NULL) { 787 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 788 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 789 } 790 } else { 791 thread->setStreamVolume(stream, value); 792 } 793 794 return NO_ERROR; 795} 796 797status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 798{ 799 // check calling permissions 800 if (!settingsAllowed()) { 801 return PERMISSION_DENIED; 802 } 803 804 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 805 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 806 ALOGE("setStreamMute() invalid stream %d", stream); 807 return BAD_VALUE; 808 } 809 810 AutoMutex lock(mLock); 811 mStreamTypes[stream].mute = muted; 812 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 813 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 814 815 return NO_ERROR; 816} 817 818float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 819{ 820 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 821 return 0.0f; 822 } 823 824 AutoMutex lock(mLock); 825 float volume; 826 if (output) { 827 PlaybackThread *thread = checkPlaybackThread_l(output); 828 if (thread == NULL) { 829 return 0.0f; 830 } 831 volume = thread->streamVolume(stream); 832 } else { 833 volume = streamVolume_l(stream); 834 } 835 836 return volume; 837} 838 839bool AudioFlinger::streamMute(audio_stream_type_t stream) const 840{ 841 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 842 return true; 843 } 844 845 AutoMutex lock(mLock); 846 return streamMute_l(stream); 847} 848 849status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 850{ 851 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 852 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 853 // check calling permissions 854 if (!settingsAllowed()) { 855 return PERMISSION_DENIED; 856 } 857 858 // ioHandle == 0 means the parameters are global to the audio hardware interface 859 if (ioHandle == 0) { 860 Mutex::Autolock _l(mLock); 861 status_t final_result = NO_ERROR; 862 { 863 AutoMutex lock(mHardwareLock); 864 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 865 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 866 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 867 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 868 final_result = result ?: final_result; 869 } 870 mHardwareStatus = AUDIO_HW_IDLE; 871 } 872 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 873 AudioParameter param = AudioParameter(keyValuePairs); 874 String8 value; 875 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 876 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 877 if (mBtNrecIsOff != btNrecIsOff) { 878 for (size_t i = 0; i < mRecordThreads.size(); i++) { 879 sp<RecordThread> thread = mRecordThreads.valueAt(i); 880 RecordThread::RecordTrack *track = thread->track(); 881 if (track != NULL) { 882 audio_devices_t device = (audio_devices_t)( 883 thread->device() & AUDIO_DEVICE_IN_ALL); 884 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 885 thread->setEffectSuspended(FX_IID_AEC, 886 suspend, 887 track->sessionId()); 888 thread->setEffectSuspended(FX_IID_NS, 889 suspend, 890 track->sessionId()); 891 } 892 } 893 mBtNrecIsOff = btNrecIsOff; 894 } 895 } 896 return final_result; 897 } 898 899 // hold a strong ref on thread in case closeOutput() or closeInput() is called 900 // and the thread is exited once the lock is released 901 sp<ThreadBase> thread; 902 { 903 Mutex::Autolock _l(mLock); 904 thread = checkPlaybackThread_l(ioHandle); 905 if (thread == NULL) { 906 thread = checkRecordThread_l(ioHandle); 907 } else if (thread == primaryPlaybackThread_l()) { 908 // indicate output device change to all input threads for pre processing 909 AudioParameter param = AudioParameter(keyValuePairs); 910 int value; 911 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 912 (value != 0)) { 913 for (size_t i = 0; i < mRecordThreads.size(); i++) { 914 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 915 } 916 } 917 } 918 } 919 if (thread != 0) { 920 return thread->setParameters(keyValuePairs); 921 } 922 return BAD_VALUE; 923} 924 925String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 926{ 927// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 928// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 929 930 Mutex::Autolock _l(mLock); 931 932 if (ioHandle == 0) { 933 String8 out_s8; 934 935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 936 char *s; 937 { 938 AutoMutex lock(mHardwareLock); 939 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 940 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 941 s = dev->get_parameters(dev, keys.string()); 942 mHardwareStatus = AUDIO_HW_IDLE; 943 } 944 out_s8 += String8(s ? s : ""); 945 free(s); 946 } 947 return out_s8; 948 } 949 950 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 951 if (playbackThread != NULL) { 952 return playbackThread->getParameters(keys); 953 } 954 RecordThread *recordThread = checkRecordThread_l(ioHandle); 955 if (recordThread != NULL) { 956 return recordThread->getParameters(keys); 957 } 958 return String8(""); 959} 960 961size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 962{ 963 status_t ret = initCheck(); 964 if (ret != NO_ERROR) { 965 return 0; 966 } 967 968 AutoMutex lock(mHardwareLock); 969 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 970 struct audio_config config = { 971 sample_rate: sampleRate, 972 channel_mask: audio_channel_in_mask_from_count(channelCount), 973 format: format, 974 }; 975 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 976 mHardwareStatus = AUDIO_HW_IDLE; 977 return size; 978} 979 980unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 981{ 982 if (ioHandle == 0) { 983 return 0; 984 } 985 986 Mutex::Autolock _l(mLock); 987 988 RecordThread *recordThread = checkRecordThread_l(ioHandle); 989 if (recordThread != NULL) { 990 return recordThread->getInputFramesLost(); 991 } 992 return 0; 993} 994 995status_t AudioFlinger::setVoiceVolume(float value) 996{ 997 status_t ret = initCheck(); 998 if (ret != NO_ERROR) { 999 return ret; 1000 } 1001 1002 // check calling permissions 1003 if (!settingsAllowed()) { 1004 return PERMISSION_DENIED; 1005 } 1006 1007 AutoMutex lock(mHardwareLock); 1008 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1009 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1010 mHardwareStatus = AUDIO_HW_IDLE; 1011 1012 return ret; 1013} 1014 1015status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1016 audio_io_handle_t output) const 1017{ 1018 status_t status; 1019 1020 Mutex::Autolock _l(mLock); 1021 1022 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1023 if (playbackThread != NULL) { 1024 return playbackThread->getRenderPosition(halFrames, dspFrames); 1025 } 1026 1027 return BAD_VALUE; 1028} 1029 1030void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1031{ 1032 1033 Mutex::Autolock _l(mLock); 1034 1035 pid_t pid = IPCThreadState::self()->getCallingPid(); 1036 if (mNotificationClients.indexOfKey(pid) < 0) { 1037 sp<NotificationClient> notificationClient = new NotificationClient(this, 1038 client, 1039 pid); 1040 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1041 1042 mNotificationClients.add(pid, notificationClient); 1043 1044 sp<IBinder> binder = client->asBinder(); 1045 binder->linkToDeath(notificationClient); 1046 1047 // the config change is always sent from playback or record threads to avoid deadlock 1048 // with AudioSystem::gLock 1049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1050 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1051 } 1052 1053 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1054 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1055 } 1056 } 1057} 1058 1059void AudioFlinger::removeNotificationClient(pid_t pid) 1060{ 1061 Mutex::Autolock _l(mLock); 1062 1063 mNotificationClients.removeItem(pid); 1064 1065 ALOGV("%d died, releasing its sessions", pid); 1066 size_t num = mAudioSessionRefs.size(); 1067 bool removed = false; 1068 for (size_t i = 0; i< num; ) { 1069 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1070 ALOGV(" pid %d @ %d", ref->mPid, i); 1071 if (ref->mPid == pid) { 1072 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1073 mAudioSessionRefs.removeAt(i); 1074 delete ref; 1075 removed = true; 1076 num--; 1077 } else { 1078 i++; 1079 } 1080 } 1081 if (removed) { 1082 purgeStaleEffects_l(); 1083 } 1084} 1085 1086// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1087void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1088{ 1089 size_t size = mNotificationClients.size(); 1090 for (size_t i = 0; i < size; i++) { 1091 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1092 param2); 1093 } 1094} 1095 1096// removeClient_l() must be called with AudioFlinger::mLock held 1097void AudioFlinger::removeClient_l(pid_t pid) 1098{ 1099 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1100 mClients.removeItem(pid); 1101} 1102 1103 1104// ---------------------------------------------------------------------------- 1105 1106AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1107 uint32_t device, type_t type) 1108 : Thread(false), 1109 mType(type), 1110 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1111 // mChannelMask 1112 mChannelCount(0), 1113 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1114 mParamStatus(NO_ERROR), 1115 mStandby(false), mId(id), 1116 mDevice(device), 1117 mDeathRecipient(new PMDeathRecipient(this)) 1118{ 1119} 1120 1121AudioFlinger::ThreadBase::~ThreadBase() 1122{ 1123 mParamCond.broadcast(); 1124 // do not lock the mutex in destructor 1125 releaseWakeLock_l(); 1126 if (mPowerManager != 0) { 1127 sp<IBinder> binder = mPowerManager->asBinder(); 1128 binder->unlinkToDeath(mDeathRecipient); 1129 } 1130} 1131 1132void AudioFlinger::ThreadBase::exit() 1133{ 1134 ALOGV("ThreadBase::exit"); 1135 { 1136 // This lock prevents the following race in thread (uniprocessor for illustration): 1137 // if (!exitPending()) { 1138 // // context switch from here to exit() 1139 // // exit() calls requestExit(), what exitPending() observes 1140 // // exit() calls signal(), which is dropped since no waiters 1141 // // context switch back from exit() to here 1142 // mWaitWorkCV.wait(...); 1143 // // now thread is hung 1144 // } 1145 AutoMutex lock(mLock); 1146 requestExit(); 1147 mWaitWorkCV.signal(); 1148 } 1149 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1150 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1151 requestExitAndWait(); 1152} 1153 1154status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1155{ 1156 status_t status; 1157 1158 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1159 Mutex::Autolock _l(mLock); 1160 1161 mNewParameters.add(keyValuePairs); 1162 mWaitWorkCV.signal(); 1163 // wait condition with timeout in case the thread loop has exited 1164 // before the request could be processed 1165 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1166 status = mParamStatus; 1167 mWaitWorkCV.signal(); 1168 } else { 1169 status = TIMED_OUT; 1170 } 1171 return status; 1172} 1173 1174void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1175{ 1176 Mutex::Autolock _l(mLock); 1177 sendConfigEvent_l(event, param); 1178} 1179 1180// sendConfigEvent_l() must be called with ThreadBase::mLock held 1181void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1182{ 1183 ConfigEvent configEvent; 1184 configEvent.mEvent = event; 1185 configEvent.mParam = param; 1186 mConfigEvents.add(configEvent); 1187 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1188 mWaitWorkCV.signal(); 1189} 1190 1191void AudioFlinger::ThreadBase::processConfigEvents() 1192{ 1193 mLock.lock(); 1194 while (!mConfigEvents.isEmpty()) { 1195 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1196 ConfigEvent configEvent = mConfigEvents[0]; 1197 mConfigEvents.removeAt(0); 1198 // release mLock before locking AudioFlinger mLock: lock order is always 1199 // AudioFlinger then ThreadBase to avoid cross deadlock 1200 mLock.unlock(); 1201 mAudioFlinger->mLock.lock(); 1202 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1203 mAudioFlinger->mLock.unlock(); 1204 mLock.lock(); 1205 } 1206 mLock.unlock(); 1207} 1208 1209status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1210{ 1211 const size_t SIZE = 256; 1212 char buffer[SIZE]; 1213 String8 result; 1214 1215 bool locked = tryLock(mLock); 1216 if (!locked) { 1217 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1218 write(fd, buffer, strlen(buffer)); 1219 } 1220 1221 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1240 result.append(buffer); 1241 1242 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1243 result.append(buffer); 1244 result.append(" Index Command"); 1245 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1246 snprintf(buffer, SIZE, "\n %02d ", i); 1247 result.append(buffer); 1248 result.append(mNewParameters[i]); 1249 } 1250 1251 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1252 result.append(buffer); 1253 snprintf(buffer, SIZE, " Index event param\n"); 1254 result.append(buffer); 1255 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1256 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1257 result.append(buffer); 1258 } 1259 result.append("\n"); 1260 1261 write(fd, result.string(), result.size()); 1262 1263 if (locked) { 1264 mLock.unlock(); 1265 } 1266 return NO_ERROR; 1267} 1268 1269status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1270{ 1271 const size_t SIZE = 256; 1272 char buffer[SIZE]; 1273 String8 result; 1274 1275 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1276 write(fd, buffer, strlen(buffer)); 1277 1278 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1279 sp<EffectChain> chain = mEffectChains[i]; 1280 if (chain != 0) { 1281 chain->dump(fd, args); 1282 } 1283 } 1284 return NO_ERROR; 1285} 1286 1287void AudioFlinger::ThreadBase::acquireWakeLock() 1288{ 1289 Mutex::Autolock _l(mLock); 1290 acquireWakeLock_l(); 1291} 1292 1293void AudioFlinger::ThreadBase::acquireWakeLock_l() 1294{ 1295 if (mPowerManager == 0) { 1296 // use checkService() to avoid blocking if power service is not up yet 1297 sp<IBinder> binder = 1298 defaultServiceManager()->checkService(String16("power")); 1299 if (binder == 0) { 1300 ALOGW("Thread %s cannot connect to the power manager service", mName); 1301 } else { 1302 mPowerManager = interface_cast<IPowerManager>(binder); 1303 binder->linkToDeath(mDeathRecipient); 1304 } 1305 } 1306 if (mPowerManager != 0) { 1307 sp<IBinder> binder = new BBinder(); 1308 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1309 binder, 1310 String16(mName)); 1311 if (status == NO_ERROR) { 1312 mWakeLockToken = binder; 1313 } 1314 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1315 } 1316} 1317 1318void AudioFlinger::ThreadBase::releaseWakeLock() 1319{ 1320 Mutex::Autolock _l(mLock); 1321 releaseWakeLock_l(); 1322} 1323 1324void AudioFlinger::ThreadBase::releaseWakeLock_l() 1325{ 1326 if (mWakeLockToken != 0) { 1327 ALOGV("releaseWakeLock_l() %s", mName); 1328 if (mPowerManager != 0) { 1329 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1330 } 1331 mWakeLockToken.clear(); 1332 } 1333} 1334 1335void AudioFlinger::ThreadBase::clearPowerManager() 1336{ 1337 Mutex::Autolock _l(mLock); 1338 releaseWakeLock_l(); 1339 mPowerManager.clear(); 1340} 1341 1342void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1343{ 1344 sp<ThreadBase> thread = mThread.promote(); 1345 if (thread != 0) { 1346 thread->clearPowerManager(); 1347 } 1348 ALOGW("power manager service died !!!"); 1349} 1350 1351void AudioFlinger::ThreadBase::setEffectSuspended( 1352 const effect_uuid_t *type, bool suspend, int sessionId) 1353{ 1354 Mutex::Autolock _l(mLock); 1355 setEffectSuspended_l(type, suspend, sessionId); 1356} 1357 1358void AudioFlinger::ThreadBase::setEffectSuspended_l( 1359 const effect_uuid_t *type, bool suspend, int sessionId) 1360{ 1361 sp<EffectChain> chain = getEffectChain_l(sessionId); 1362 if (chain != 0) { 1363 if (type != NULL) { 1364 chain->setEffectSuspended_l(type, suspend); 1365 } else { 1366 chain->setEffectSuspendedAll_l(suspend); 1367 } 1368 } 1369 1370 updateSuspendedSessions_l(type, suspend, sessionId); 1371} 1372 1373void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1374{ 1375 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1376 if (index < 0) { 1377 return; 1378 } 1379 1380 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1381 mSuspendedSessions.editValueAt(index); 1382 1383 for (size_t i = 0; i < sessionEffects.size(); i++) { 1384 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1385 for (int j = 0; j < desc->mRefCount; j++) { 1386 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1387 chain->setEffectSuspendedAll_l(true); 1388 } else { 1389 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1390 desc->mType.timeLow); 1391 chain->setEffectSuspended_l(&desc->mType, true); 1392 } 1393 } 1394 } 1395} 1396 1397void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1398 bool suspend, 1399 int sessionId) 1400{ 1401 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1402 1403 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1404 1405 if (suspend) { 1406 if (index >= 0) { 1407 sessionEffects = mSuspendedSessions.editValueAt(index); 1408 } else { 1409 mSuspendedSessions.add(sessionId, sessionEffects); 1410 } 1411 } else { 1412 if (index < 0) { 1413 return; 1414 } 1415 sessionEffects = mSuspendedSessions.editValueAt(index); 1416 } 1417 1418 1419 int key = EffectChain::kKeyForSuspendAll; 1420 if (type != NULL) { 1421 key = type->timeLow; 1422 } 1423 index = sessionEffects.indexOfKey(key); 1424 1425 sp<SuspendedSessionDesc> desc; 1426 if (suspend) { 1427 if (index >= 0) { 1428 desc = sessionEffects.valueAt(index); 1429 } else { 1430 desc = new SuspendedSessionDesc(); 1431 if (type != NULL) { 1432 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1433 } 1434 sessionEffects.add(key, desc); 1435 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1436 } 1437 desc->mRefCount++; 1438 } else { 1439 if (index < 0) { 1440 return; 1441 } 1442 desc = sessionEffects.valueAt(index); 1443 if (--desc->mRefCount == 0) { 1444 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1445 sessionEffects.removeItemsAt(index); 1446 if (sessionEffects.isEmpty()) { 1447 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1448 sessionId); 1449 mSuspendedSessions.removeItem(sessionId); 1450 } 1451 } 1452 } 1453 if (!sessionEffects.isEmpty()) { 1454 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1455 } 1456} 1457 1458void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1459 bool enabled, 1460 int sessionId) 1461{ 1462 Mutex::Autolock _l(mLock); 1463 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1464} 1465 1466void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1467 bool enabled, 1468 int sessionId) 1469{ 1470 if (mType != RECORD) { 1471 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1472 // another session. This gives the priority to well behaved effect control panels 1473 // and applications not using global effects. 1474 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1475 // global effects 1476 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1477 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1478 } 1479 } 1480 1481 sp<EffectChain> chain = getEffectChain_l(sessionId); 1482 if (chain != 0) { 1483 chain->checkSuspendOnEffectEnabled(effect, enabled); 1484 } 1485} 1486 1487// ---------------------------------------------------------------------------- 1488 1489AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1490 AudioStreamOut* output, 1491 audio_io_handle_t id, 1492 uint32_t device, 1493 type_t type) 1494 : ThreadBase(audioFlinger, id, device, type), 1495 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1496 // Assumes constructor is called by AudioFlinger with it's mLock held, 1497 // but it would be safer to explicitly pass initial masterMute as parameter 1498 mMasterMute(audioFlinger->masterMute_l()), 1499 // mStreamTypes[] initialized in constructor body 1500 mOutput(output), 1501 // Assumes constructor is called by AudioFlinger with it's mLock held, 1502 // but it would be safer to explicitly pass initial masterVolume as parameter 1503 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1504 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1505 mMixerStatus(MIXER_IDLE), 1506 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1507 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1508 // index 0 is reserved for normal mixer's submix 1509 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1510{ 1511 snprintf(mName, kNameLength, "AudioOut_%X", id); 1512 1513 readOutputParameters(); 1514 1515 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1516 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1517 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1518 stream = (audio_stream_type_t) (stream + 1)) { 1519 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1520 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1521 } 1522 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1523 // because mAudioFlinger doesn't have one to copy from 1524} 1525 1526AudioFlinger::PlaybackThread::~PlaybackThread() 1527{ 1528 delete [] mMixBuffer; 1529} 1530 1531status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1532{ 1533 dumpInternals(fd, args); 1534 dumpTracks(fd, args); 1535 dumpEffectChains(fd, args); 1536 return NO_ERROR; 1537} 1538 1539status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1540{ 1541 const size_t SIZE = 256; 1542 char buffer[SIZE]; 1543 String8 result; 1544 1545 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1546 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1547 const stream_type_t *st = &mStreamTypes[i]; 1548 if (i > 0) { 1549 result.appendFormat(", "); 1550 } 1551 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1552 if (st->mute) { 1553 result.append("M"); 1554 } 1555 } 1556 result.append("\n"); 1557 write(fd, result.string(), result.length()); 1558 result.clear(); 1559 1560 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1561 result.append(buffer); 1562 Track::appendDumpHeader(result); 1563 for (size_t i = 0; i < mTracks.size(); ++i) { 1564 sp<Track> track = mTracks[i]; 1565 if (track != 0) { 1566 track->dump(buffer, SIZE); 1567 result.append(buffer); 1568 } 1569 } 1570 1571 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1572 result.append(buffer); 1573 Track::appendDumpHeader(result); 1574 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1575 sp<Track> track = mActiveTracks[i].promote(); 1576 if (track != 0) { 1577 track->dump(buffer, SIZE); 1578 result.append(buffer); 1579 } 1580 } 1581 write(fd, result.string(), result.size()); 1582 return NO_ERROR; 1583} 1584 1585status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1586{ 1587 const size_t SIZE = 256; 1588 char buffer[SIZE]; 1589 String8 result; 1590 1591 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1592 result.append(buffer); 1593 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1604 result.append(buffer); 1605 write(fd, result.string(), result.size()); 1606 1607 dumpBase(fd, args); 1608 1609 return NO_ERROR; 1610} 1611 1612// Thread virtuals 1613status_t AudioFlinger::PlaybackThread::readyToRun() 1614{ 1615 status_t status = initCheck(); 1616 if (status == NO_ERROR) { 1617 ALOGI("AudioFlinger's thread %p ready to run", this); 1618 } else { 1619 ALOGE("No working audio driver found."); 1620 } 1621 return status; 1622} 1623 1624void AudioFlinger::PlaybackThread::onFirstRef() 1625{ 1626 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1627} 1628 1629// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1630sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1631 const sp<AudioFlinger::Client>& client, 1632 audio_stream_type_t streamType, 1633 uint32_t sampleRate, 1634 audio_format_t format, 1635 uint32_t channelMask, 1636 int frameCount, 1637 const sp<IMemory>& sharedBuffer, 1638 int sessionId, 1639 IAudioFlinger::track_flags_t flags, 1640 pid_t tid, 1641 status_t *status) 1642{ 1643 sp<Track> track; 1644 status_t lStatus; 1645 1646 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1647 1648 // client expresses a preference for FAST, but we get the final say 1649 if (flags & IAudioFlinger::TRACK_FAST) { 1650 if ( 1651 // not timed 1652 (!isTimed) && 1653 // either of these use cases: 1654 ( 1655 // use case 1: shared buffer with any frame count 1656 ( 1657 (sharedBuffer != 0) 1658 ) || 1659 // use case 2: callback handler and frame count is default or at least as large as HAL 1660 ( 1661 (tid != -1) && 1662 ((frameCount == 0) || 1663 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1664 ) 1665 ) && 1666 // PCM data 1667 audio_is_linear_pcm(format) && 1668 // mono or stereo 1669 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1670 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1671#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1672 // hardware sample rate 1673 (sampleRate == mSampleRate) && 1674#endif 1675 // normal mixer has an associated fast mixer 1676 hasFastMixer() && 1677 // there are sufficient fast track slots available 1678 (mFastTrackAvailMask != 0) 1679 // FIXME test that MixerThread for this fast track has a capable output HAL 1680 // FIXME add a permission test also? 1681 ) { 1682 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1683 if (frameCount == 0) { 1684 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1685 } 1686 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1687 frameCount, mFrameCount); 1688 } else { 1689 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1690 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1691 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1692 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1693 audio_is_linear_pcm(format), 1694 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1695 flags &= ~IAudioFlinger::TRACK_FAST; 1696 // For compatibility with AudioTrack calculation, buffer depth is forced 1697 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1698 // This is probably too conservative, but legacy application code may depend on it. 1699 // If you change this calculation, also review the start threshold which is related. 1700 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1701 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1702 if (minBufCount < 2) { 1703 minBufCount = 2; 1704 } 1705 int minFrameCount = mNormalFrameCount * minBufCount; 1706 if (frameCount < minFrameCount) { 1707 frameCount = minFrameCount; 1708 } 1709 } 1710 } 1711 1712 if (mType == DIRECT) { 1713 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1714 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1715 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1716 "for output %p with format %d", 1717 sampleRate, format, channelMask, mOutput, mFormat); 1718 lStatus = BAD_VALUE; 1719 goto Exit; 1720 } 1721 } 1722 } else { 1723 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1724 if (sampleRate > mSampleRate*2) { 1725 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1726 lStatus = BAD_VALUE; 1727 goto Exit; 1728 } 1729 } 1730 1731 lStatus = initCheck(); 1732 if (lStatus != NO_ERROR) { 1733 ALOGE("Audio driver not initialized."); 1734 goto Exit; 1735 } 1736 1737 { // scope for mLock 1738 Mutex::Autolock _l(mLock); 1739 1740 // all tracks in same audio session must share the same routing strategy otherwise 1741 // conflicts will happen when tracks are moved from one output to another by audio policy 1742 // manager 1743 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1744 for (size_t i = 0; i < mTracks.size(); ++i) { 1745 sp<Track> t = mTracks[i]; 1746 if (t != 0 && !t->isOutputTrack()) { 1747 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1748 if (sessionId == t->sessionId() && strategy != actual) { 1749 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1750 strategy, actual); 1751 lStatus = BAD_VALUE; 1752 goto Exit; 1753 } 1754 } 1755 } 1756 1757 if (!isTimed) { 1758 track = new Track(this, client, streamType, sampleRate, format, 1759 channelMask, frameCount, sharedBuffer, sessionId, flags); 1760 } else { 1761 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1762 channelMask, frameCount, sharedBuffer, sessionId); 1763 } 1764 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1765 lStatus = NO_MEMORY; 1766 goto Exit; 1767 } 1768 mTracks.add(track); 1769 1770 sp<EffectChain> chain = getEffectChain_l(sessionId); 1771 if (chain != 0) { 1772 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1773 track->setMainBuffer(chain->inBuffer()); 1774 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1775 chain->incTrackCnt(); 1776 } 1777 } 1778 1779#ifdef HAVE_REQUEST_PRIORITY 1780 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1781 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1782 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1783 // so ask activity manager to do this on our behalf 1784 int err = requestPriority(callingPid, tid, 1); 1785 if (err != 0) { 1786 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1787 1, callingPid, tid, err); 1788 } 1789 } 1790#endif 1791 1792 lStatus = NO_ERROR; 1793 1794Exit: 1795 if (status) { 1796 *status = lStatus; 1797 } 1798 return track; 1799} 1800 1801uint32_t AudioFlinger::PlaybackThread::latency() const 1802{ 1803 Mutex::Autolock _l(mLock); 1804 if (initCheck() == NO_ERROR) { 1805 return mOutput->stream->get_latency(mOutput->stream); 1806 } else { 1807 return 0; 1808 } 1809} 1810 1811void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1812{ 1813 Mutex::Autolock _l(mLock); 1814 mMasterVolume = value; 1815} 1816 1817void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1818{ 1819 Mutex::Autolock _l(mLock); 1820 setMasterMute_l(muted); 1821} 1822 1823void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1824{ 1825 Mutex::Autolock _l(mLock); 1826 mStreamTypes[stream].volume = value; 1827} 1828 1829void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1830{ 1831 Mutex::Autolock _l(mLock); 1832 mStreamTypes[stream].mute = muted; 1833} 1834 1835float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1836{ 1837 Mutex::Autolock _l(mLock); 1838 return mStreamTypes[stream].volume; 1839} 1840 1841// addTrack_l() must be called with ThreadBase::mLock held 1842status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1843{ 1844 status_t status = ALREADY_EXISTS; 1845 1846 // set retry count for buffer fill 1847 track->mRetryCount = kMaxTrackStartupRetries; 1848 if (mActiveTracks.indexOf(track) < 0) { 1849 // the track is newly added, make sure it fills up all its 1850 // buffers before playing. This is to ensure the client will 1851 // effectively get the latency it requested. 1852 track->mFillingUpStatus = Track::FS_FILLING; 1853 track->mResetDone = false; 1854 track->mPresentationCompleteFrames = 0; 1855 mActiveTracks.add(track); 1856 if (track->mainBuffer() != mMixBuffer) { 1857 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1858 if (chain != 0) { 1859 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1860 chain->incActiveTrackCnt(); 1861 } 1862 } 1863 1864 status = NO_ERROR; 1865 } 1866 1867 ALOGV("mWaitWorkCV.broadcast"); 1868 mWaitWorkCV.broadcast(); 1869 1870 return status; 1871} 1872 1873// destroyTrack_l() must be called with ThreadBase::mLock held 1874void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1875{ 1876 track->mState = TrackBase::TERMINATED; 1877 // active tracks are removed by threadLoop() 1878 if (mActiveTracks.indexOf(track) < 0) { 1879 removeTrack_l(track); 1880 } 1881} 1882 1883void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1884{ 1885 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1886 mTracks.remove(track); 1887 deleteTrackName_l(track->name()); 1888 // redundant as track is about to be destroyed, for dumpsys only 1889 track->mName = -1; 1890 if (track->isFastTrack()) { 1891 int index = track->mFastIndex; 1892 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1893 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1894 mFastTrackAvailMask |= 1 << index; 1895 // redundant as track is about to be destroyed, for dumpsys only 1896 track->mFastIndex = -1; 1897 } 1898 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1899 if (chain != 0) { 1900 chain->decTrackCnt(); 1901 } 1902} 1903 1904String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1905{ 1906 String8 out_s8 = String8(""); 1907 char *s; 1908 1909 Mutex::Autolock _l(mLock); 1910 if (initCheck() != NO_ERROR) { 1911 return out_s8; 1912 } 1913 1914 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1915 out_s8 = String8(s); 1916 free(s); 1917 return out_s8; 1918} 1919 1920// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1921void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1922 AudioSystem::OutputDescriptor desc; 1923 void *param2 = NULL; 1924 1925 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1926 1927 switch (event) { 1928 case AudioSystem::OUTPUT_OPENED: 1929 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1930 desc.channels = mChannelMask; 1931 desc.samplingRate = mSampleRate; 1932 desc.format = mFormat; 1933 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1934 desc.latency = latency(); 1935 param2 = &desc; 1936 break; 1937 1938 case AudioSystem::STREAM_CONFIG_CHANGED: 1939 param2 = ¶m; 1940 case AudioSystem::OUTPUT_CLOSED: 1941 default: 1942 break; 1943 } 1944 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1945} 1946 1947void AudioFlinger::PlaybackThread::readOutputParameters() 1948{ 1949 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1950 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1951 mChannelCount = (uint16_t)popcount(mChannelMask); 1952 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1953 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1954 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1955 if (mFrameCount & 15) { 1956 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1957 mFrameCount); 1958 } 1959 1960 // Calculate size of normal mix buffer relative to the HAL output buffer size 1961 double multiplier = 1.0; 1962 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1963 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1964 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1965 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1966 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1967 maxNormalFrameCount = maxNormalFrameCount & ~15; 1968 if (maxNormalFrameCount < minNormalFrameCount) { 1969 maxNormalFrameCount = minNormalFrameCount; 1970 } 1971 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1972 if (multiplier <= 1.0) { 1973 multiplier = 1.0; 1974 } else if (multiplier <= 2.0) { 1975 if (2 * mFrameCount <= maxNormalFrameCount) { 1976 multiplier = 2.0; 1977 } else { 1978 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1979 } 1980 } else { 1981 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1982 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1983 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 1984 // FIXME this rounding up should not be done if no HAL SRC 1985 uint32_t truncMult = (uint32_t) multiplier; 1986 if ((truncMult & 1)) { 1987 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1988 ++truncMult; 1989 } 1990 } 1991 multiplier = (double) truncMult; 1992 } 1993 } 1994 mNormalFrameCount = multiplier * mFrameCount; 1995 // round up to nearest 16 frames to satisfy AudioMixer 1996 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1997 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 1998 1999 // FIXME - Current mixer implementation only supports stereo output: Always 2000 // Allocate a stereo buffer even if HW output is mono. 2001 delete[] mMixBuffer; 2002 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2003 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2004 2005 // force reconfiguration of effect chains and engines to take new buffer size and audio 2006 // parameters into account 2007 // Note that mLock is not held when readOutputParameters() is called from the constructor 2008 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2009 // matter. 2010 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2011 Vector< sp<EffectChain> > effectChains = mEffectChains; 2012 for (size_t i = 0; i < effectChains.size(); i ++) { 2013 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2014 } 2015} 2016 2017status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2018{ 2019 if (halFrames == NULL || dspFrames == NULL) { 2020 return BAD_VALUE; 2021 } 2022 Mutex::Autolock _l(mLock); 2023 if (initCheck() != NO_ERROR) { 2024 return INVALID_OPERATION; 2025 } 2026 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2027 2028 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2029} 2030 2031uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2032{ 2033 Mutex::Autolock _l(mLock); 2034 uint32_t result = 0; 2035 if (getEffectChain_l(sessionId) != 0) { 2036 result = EFFECT_SESSION; 2037 } 2038 2039 for (size_t i = 0; i < mTracks.size(); ++i) { 2040 sp<Track> track = mTracks[i]; 2041 if (sessionId == track->sessionId() && 2042 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2043 result |= TRACK_SESSION; 2044 break; 2045 } 2046 } 2047 2048 return result; 2049} 2050 2051uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2052{ 2053 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2054 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2055 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2056 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2057 } 2058 for (size_t i = 0; i < mTracks.size(); i++) { 2059 sp<Track> track = mTracks[i]; 2060 if (sessionId == track->sessionId() && 2061 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2062 return AudioSystem::getStrategyForStream(track->streamType()); 2063 } 2064 } 2065 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2066} 2067 2068 2069AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2070{ 2071 Mutex::Autolock _l(mLock); 2072 return mOutput; 2073} 2074 2075AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2076{ 2077 Mutex::Autolock _l(mLock); 2078 AudioStreamOut *output = mOutput; 2079 mOutput = NULL; 2080 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2081 // must push a NULL and wait for ack 2082 mOutputSink.clear(); 2083 mPipeSink.clear(); 2084 mNormalSink.clear(); 2085 return output; 2086} 2087 2088// this method must always be called either with ThreadBase mLock held or inside the thread loop 2089audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2090{ 2091 if (mOutput == NULL) { 2092 return NULL; 2093 } 2094 return &mOutput->stream->common; 2095} 2096 2097uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2098{ 2099 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2100 // decoding and transfer time. So sleeping for half of the latency would likely cause 2101 // underruns 2102 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2103 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2104 } else { 2105 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2106 } 2107} 2108 2109status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2110{ 2111 if (!isValidSyncEvent(event)) { 2112 return BAD_VALUE; 2113 } 2114 2115 Mutex::Autolock _l(mLock); 2116 2117 for (size_t i = 0; i < mTracks.size(); ++i) { 2118 sp<Track> track = mTracks[i]; 2119 if (event->triggerSession() == track->sessionId()) { 2120 track->setSyncEvent(event); 2121 return NO_ERROR; 2122 } 2123 } 2124 2125 return NAME_NOT_FOUND; 2126} 2127 2128bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2129{ 2130 switch (event->type()) { 2131 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2132 return true; 2133 default: 2134 break; 2135 } 2136 return false; 2137} 2138 2139void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2140{ 2141 size_t count = tracksToRemove.size(); 2142 if (CC_UNLIKELY(count)) { 2143 for (size_t i = 0 ; i < count ; i++) { 2144 const sp<Track>& track = tracksToRemove.itemAt(i); 2145 if ((track->sharedBuffer() != 0) && 2146 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2147 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2148 } 2149 } 2150 } 2151 2152} 2153 2154// ---------------------------------------------------------------------------- 2155 2156AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2157 audio_io_handle_t id, uint32_t device, type_t type) 2158 : PlaybackThread(audioFlinger, output, id, device, type), 2159 // mAudioMixer below 2160#ifdef SOAKER 2161 mSoaker(NULL), 2162#endif 2163 // mFastMixer below 2164 mFastMixerFutex(0) 2165 // mOutputSink below 2166 // mPipeSink below 2167 // mNormalSink below 2168{ 2169 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2170 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2171 "mFrameCount=%d, mNormalFrameCount=%d", 2172 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2173 mNormalFrameCount); 2174 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2175 2176 // FIXME - Current mixer implementation only supports stereo output 2177 if (mChannelCount == 1) { 2178 ALOGE("Invalid audio hardware channel count"); 2179 } 2180 2181 // create an NBAIO sink for the HAL output stream, and negotiate 2182 mOutputSink = new AudioStreamOutSink(output->stream); 2183 size_t numCounterOffers = 0; 2184 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2185 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2186 ALOG_ASSERT(index == 0); 2187 2188 // initialize fast mixer depending on configuration 2189 bool initFastMixer; 2190 switch (kUseFastMixer) { 2191 case FastMixer_Never: 2192 initFastMixer = false; 2193 break; 2194 case FastMixer_Always: 2195 initFastMixer = true; 2196 break; 2197 case FastMixer_Static: 2198 case FastMixer_Dynamic: 2199 initFastMixer = mFrameCount < mNormalFrameCount; 2200 break; 2201 } 2202 if (initFastMixer) { 2203 2204 // create a MonoPipe to connect our submix to FastMixer 2205 NBAIO_Format format = mOutputSink->format(); 2206 // frame count will be rounded up to a power of 2, so this formula should work well 2207 MonoPipe *monoPipe = new MonoPipe((mNormalFrameCount * 3) / 2, format, 2208 true /*writeCanBlock*/); 2209 const NBAIO_Format offers[1] = {format}; 2210 size_t numCounterOffers = 0; 2211 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2212 ALOG_ASSERT(index == 0); 2213 mPipeSink = monoPipe; 2214 2215#ifdef SOAKER 2216 // create a soaker as workaround for governor issues 2217 mSoaker = new Soaker(); 2218 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2219 mSoaker->run("Soaker", PRIORITY_LOWEST); 2220#endif 2221 2222 // create fast mixer and configure it initially with just one fast track for our submix 2223 mFastMixer = new FastMixer(); 2224 FastMixerStateQueue *sq = mFastMixer->sq(); 2225 FastMixerState *state = sq->begin(); 2226 FastTrack *fastTrack = &state->mFastTracks[0]; 2227 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2228 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2229 fastTrack->mVolumeProvider = NULL; 2230 fastTrack->mGeneration++; 2231 state->mFastTracksGen++; 2232 state->mTrackMask = 1; 2233 // fast mixer will use the HAL output sink 2234 state->mOutputSink = mOutputSink.get(); 2235 state->mOutputSinkGen++; 2236 state->mFrameCount = mFrameCount; 2237 state->mCommand = FastMixerState::COLD_IDLE; 2238 // already done in constructor initialization list 2239 //mFastMixerFutex = 0; 2240 state->mColdFutexAddr = &mFastMixerFutex; 2241 state->mColdGen++; 2242 state->mDumpState = &mFastMixerDumpState; 2243 sq->end(); 2244 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2245 2246 // start the fast mixer 2247 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2248#ifdef HAVE_REQUEST_PRIORITY 2249 pid_t tid = mFastMixer->getTid(); 2250 int err = requestPriority(getpid_cached, tid, 2); 2251 if (err != 0) { 2252 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2253 2, getpid_cached, tid, err); 2254 } 2255#endif 2256 2257 } else { 2258 mFastMixer = NULL; 2259 } 2260 2261 switch (kUseFastMixer) { 2262 case FastMixer_Never: 2263 case FastMixer_Dynamic: 2264 mNormalSink = mOutputSink; 2265 break; 2266 case FastMixer_Always: 2267 mNormalSink = mPipeSink; 2268 break; 2269 case FastMixer_Static: 2270 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2271 break; 2272 } 2273} 2274 2275AudioFlinger::MixerThread::~MixerThread() 2276{ 2277 if (mFastMixer != NULL) { 2278 FastMixerStateQueue *sq = mFastMixer->sq(); 2279 FastMixerState *state = sq->begin(); 2280 if (state->mCommand == FastMixerState::COLD_IDLE) { 2281 int32_t old = android_atomic_inc(&mFastMixerFutex); 2282 if (old == -1) { 2283 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2284 } 2285 } 2286 state->mCommand = FastMixerState::EXIT; 2287 sq->end(); 2288 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2289 mFastMixer->join(); 2290 // Though the fast mixer thread has exited, it's state queue is still valid. 2291 // We'll use that extract the final state which contains one remaining fast track 2292 // corresponding to our sub-mix. 2293 state = sq->begin(); 2294 ALOG_ASSERT(state->mTrackMask == 1); 2295 FastTrack *fastTrack = &state->mFastTracks[0]; 2296 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2297 delete fastTrack->mBufferProvider; 2298 sq->end(false /*didModify*/); 2299 delete mFastMixer; 2300#ifdef SOAKER 2301 if (mSoaker != NULL) { 2302 mSoaker->requestExitAndWait(); 2303 } 2304 delete mSoaker; 2305#endif 2306 } 2307 delete mAudioMixer; 2308} 2309 2310class CpuStats { 2311public: 2312 CpuStats(); 2313 void sample(const String8 &title); 2314#ifdef DEBUG_CPU_USAGE 2315private: 2316 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2317 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2318 2319 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2320 2321 int mCpuNum; // thread's current CPU number 2322 int mCpukHz; // frequency of thread's current CPU in kHz 2323#endif 2324}; 2325 2326CpuStats::CpuStats() 2327#ifdef DEBUG_CPU_USAGE 2328 : mCpuNum(-1), mCpukHz(-1) 2329#endif 2330{ 2331} 2332 2333void CpuStats::sample(const String8 &title) { 2334#ifdef DEBUG_CPU_USAGE 2335 // get current thread's delta CPU time in wall clock ns 2336 double wcNs; 2337 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2338 2339 // record sample for wall clock statistics 2340 if (valid) { 2341 mWcStats.sample(wcNs); 2342 } 2343 2344 // get the current CPU number 2345 int cpuNum = sched_getcpu(); 2346 2347 // get the current CPU frequency in kHz 2348 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2349 2350 // check if either CPU number or frequency changed 2351 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2352 mCpuNum = cpuNum; 2353 mCpukHz = cpukHz; 2354 // ignore sample for purposes of cycles 2355 valid = false; 2356 } 2357 2358 // if no change in CPU number or frequency, then record sample for cycle statistics 2359 if (valid && mCpukHz > 0) { 2360 double cycles = wcNs * cpukHz * 0.000001; 2361 mHzStats.sample(cycles); 2362 } 2363 2364 unsigned n = mWcStats.n(); 2365 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2366 if ((n & 127) == 1) { 2367 long long elapsed = mCpuUsage.elapsed(); 2368 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2369 double perLoop = elapsed / (double) n; 2370 double perLoop100 = perLoop * 0.01; 2371 double perLoop1k = perLoop * 0.001; 2372 double mean = mWcStats.mean(); 2373 double stddev = mWcStats.stddev(); 2374 double minimum = mWcStats.minimum(); 2375 double maximum = mWcStats.maximum(); 2376 double meanCycles = mHzStats.mean(); 2377 double stddevCycles = mHzStats.stddev(); 2378 double minCycles = mHzStats.minimum(); 2379 double maxCycles = mHzStats.maximum(); 2380 mCpuUsage.resetElapsed(); 2381 mWcStats.reset(); 2382 mHzStats.reset(); 2383 ALOGD("CPU usage for %s over past %.1f secs\n" 2384 " (%u mixer loops at %.1f mean ms per loop):\n" 2385 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2386 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2387 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2388 title.string(), 2389 elapsed * .000000001, n, perLoop * .000001, 2390 mean * .001, 2391 stddev * .001, 2392 minimum * .001, 2393 maximum * .001, 2394 mean / perLoop100, 2395 stddev / perLoop100, 2396 minimum / perLoop100, 2397 maximum / perLoop100, 2398 meanCycles / perLoop1k, 2399 stddevCycles / perLoop1k, 2400 minCycles / perLoop1k, 2401 maxCycles / perLoop1k); 2402 2403 } 2404 } 2405#endif 2406}; 2407 2408void AudioFlinger::PlaybackThread::checkSilentMode_l() 2409{ 2410 if (!mMasterMute) { 2411 char value[PROPERTY_VALUE_MAX]; 2412 if (property_get("ro.audio.silent", value, "0") > 0) { 2413 char *endptr; 2414 unsigned long ul = strtoul(value, &endptr, 0); 2415 if (*endptr == '\0' && ul != 0) { 2416 ALOGD("Silence is golden"); 2417 // The setprop command will not allow a property to be changed after 2418 // the first time it is set, so we don't have to worry about un-muting. 2419 setMasterMute_l(true); 2420 } 2421 } 2422 } 2423} 2424 2425bool AudioFlinger::PlaybackThread::threadLoop() 2426{ 2427 Vector< sp<Track> > tracksToRemove; 2428 2429 standbyTime = systemTime(); 2430 2431 // MIXER 2432 nsecs_t lastWarning = 0; 2433if (mType == MIXER) { 2434 longStandbyExit = false; 2435} 2436 2437 // DUPLICATING 2438 // FIXME could this be made local to while loop? 2439 writeFrames = 0; 2440 2441 cacheParameters_l(); 2442 sleepTime = idleSleepTime; 2443 2444if (mType == MIXER) { 2445 sleepTimeShift = 0; 2446} 2447 2448 CpuStats cpuStats; 2449 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2450 2451 acquireWakeLock(); 2452 2453 while (!exitPending()) 2454 { 2455 cpuStats.sample(myName); 2456 2457 Vector< sp<EffectChain> > effectChains; 2458 2459 processConfigEvents(); 2460 2461 { // scope for mLock 2462 2463 Mutex::Autolock _l(mLock); 2464 2465 if (checkForNewParameters_l()) { 2466 cacheParameters_l(); 2467 } 2468 2469 saveOutputTracks(); 2470 2471 // put audio hardware into standby after short delay 2472 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2473 mSuspended > 0)) { 2474 if (!mStandby) { 2475 2476 threadLoop_standby(); 2477 2478 mStandby = true; 2479 mBytesWritten = 0; 2480 } 2481 2482 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2483 // we're about to wait, flush the binder command buffer 2484 IPCThreadState::self()->flushCommands(); 2485 2486 clearOutputTracks(); 2487 2488 if (exitPending()) break; 2489 2490 releaseWakeLock_l(); 2491 // wait until we have something to do... 2492 ALOGV("%s going to sleep", myName.string()); 2493 mWaitWorkCV.wait(mLock); 2494 ALOGV("%s waking up", myName.string()); 2495 acquireWakeLock_l(); 2496 2497 mMixerStatus = MIXER_IDLE; 2498 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2499 2500 checkSilentMode_l(); 2501 2502 standbyTime = systemTime() + standbyDelay; 2503 sleepTime = idleSleepTime; 2504 if (mType == MIXER) { 2505 sleepTimeShift = 0; 2506 } 2507 2508 continue; 2509 } 2510 } 2511 2512 // mMixerStatusIgnoringFastTracks is also updated internally 2513 mMixerStatus = prepareTracks_l(&tracksToRemove); 2514 2515 // prevent any changes in effect chain list and in each effect chain 2516 // during mixing and effect process as the audio buffers could be deleted 2517 // or modified if an effect is created or deleted 2518 lockEffectChains_l(effectChains); 2519 } 2520 2521 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2522 threadLoop_mix(); 2523 } else { 2524 threadLoop_sleepTime(); 2525 } 2526 2527 if (mSuspended > 0) { 2528 sleepTime = suspendSleepTimeUs(); 2529 } 2530 2531 // only process effects if we're going to write 2532 if (sleepTime == 0) { 2533 for (size_t i = 0; i < effectChains.size(); i ++) { 2534 effectChains[i]->process_l(); 2535 } 2536 } 2537 2538 // enable changes in effect chain 2539 unlockEffectChains(effectChains); 2540 2541 // sleepTime == 0 means we must write to audio hardware 2542 if (sleepTime == 0) { 2543 2544 threadLoop_write(); 2545 2546if (mType == MIXER) { 2547 // write blocked detection 2548 nsecs_t now = systemTime(); 2549 nsecs_t delta = now - mLastWriteTime; 2550 if (!mStandby && delta > maxPeriod) { 2551 mNumDelayedWrites++; 2552 if ((now - lastWarning) > kWarningThrottleNs) { 2553 ScopedTrace st(ATRACE_TAG, "underrun"); 2554 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2555 ns2ms(delta), mNumDelayedWrites, this); 2556 lastWarning = now; 2557 } 2558 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2559 // a different threshold. Or completely removed for what it is worth anyway... 2560 if (mStandby) { 2561 longStandbyExit = true; 2562 } 2563 } 2564} 2565 2566 mStandby = false; 2567 } else { 2568 usleep(sleepTime); 2569 } 2570 2571 // Finally let go of removed track(s), without the lock held 2572 // since we can't guarantee the destructors won't acquire that 2573 // same lock. This will also mutate and push a new fast mixer state. 2574 threadLoop_removeTracks(tracksToRemove); 2575 tracksToRemove.clear(); 2576 2577 // FIXME I don't understand the need for this here; 2578 // it was in the original code but maybe the 2579 // assignment in saveOutputTracks() makes this unnecessary? 2580 clearOutputTracks(); 2581 2582 // Effect chains will be actually deleted here if they were removed from 2583 // mEffectChains list during mixing or effects processing 2584 effectChains.clear(); 2585 2586 // FIXME Note that the above .clear() is no longer necessary since effectChains 2587 // is now local to this block, but will keep it for now (at least until merge done). 2588 } 2589 2590if (mType == MIXER || mType == DIRECT) { 2591 // put output stream into standby mode 2592 if (!mStandby) { 2593 mOutput->stream->common.standby(&mOutput->stream->common); 2594 } 2595} 2596if (mType == DUPLICATING) { 2597 // for DuplicatingThread, standby mode is handled by the outputTracks 2598} 2599 2600 releaseWakeLock(); 2601 2602 ALOGV("Thread %p type %d exiting", this, mType); 2603 return false; 2604} 2605 2606void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2607{ 2608 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2609} 2610 2611void AudioFlinger::MixerThread::threadLoop_write() 2612{ 2613 // FIXME we should only do one push per cycle; confirm this is true 2614 // Start the fast mixer if it's not already running 2615 if (mFastMixer != NULL) { 2616 FastMixerStateQueue *sq = mFastMixer->sq(); 2617 FastMixerState *state = sq->begin(); 2618 if (state->mCommand != FastMixerState::MIX_WRITE && 2619 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2620 if (state->mCommand == FastMixerState::COLD_IDLE) { 2621 int32_t old = android_atomic_inc(&mFastMixerFutex); 2622 if (old == -1) { 2623 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2624 } 2625 } 2626 state->mCommand = FastMixerState::MIX_WRITE; 2627 sq->end(); 2628 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2629 if (kUseFastMixer == FastMixer_Dynamic) { 2630 mNormalSink = mPipeSink; 2631 } 2632 } else { 2633 sq->end(false /*didModify*/); 2634 } 2635 } 2636 PlaybackThread::threadLoop_write(); 2637} 2638 2639// shared by MIXER and DIRECT, overridden by DUPLICATING 2640void AudioFlinger::PlaybackThread::threadLoop_write() 2641{ 2642 // FIXME rewrite to reduce number of system calls 2643 mLastWriteTime = systemTime(); 2644 mInWrite = true; 2645 2646#define mBitShift 2 // FIXME 2647 size_t count = mixBufferSize >> mBitShift; 2648 Tracer::traceBegin(ATRACE_TAG, "write"); 2649 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2650 Tracer::traceEnd(ATRACE_TAG); 2651 if (framesWritten > 0) { 2652 size_t bytesWritten = framesWritten << mBitShift; 2653 mBytesWritten += bytesWritten; 2654 } 2655 2656 mNumWrites++; 2657 mInWrite = false; 2658} 2659 2660void AudioFlinger::MixerThread::threadLoop_standby() 2661{ 2662 // Idle the fast mixer if it's currently running 2663 if (mFastMixer != NULL) { 2664 FastMixerStateQueue *sq = mFastMixer->sq(); 2665 FastMixerState *state = sq->begin(); 2666 if (!(state->mCommand & FastMixerState::IDLE)) { 2667 state->mCommand = FastMixerState::COLD_IDLE; 2668 state->mColdFutexAddr = &mFastMixerFutex; 2669 state->mColdGen++; 2670 mFastMixerFutex = 0; 2671 sq->end(); 2672 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2673 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2674 if (kUseFastMixer == FastMixer_Dynamic) { 2675 mNormalSink = mOutputSink; 2676 } 2677 } else { 2678 sq->end(false /*didModify*/); 2679 } 2680 } 2681 PlaybackThread::threadLoop_standby(); 2682} 2683 2684// shared by MIXER and DIRECT, overridden by DUPLICATING 2685void AudioFlinger::PlaybackThread::threadLoop_standby() 2686{ 2687 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2688 mOutput->stream->common.standby(&mOutput->stream->common); 2689} 2690 2691void AudioFlinger::MixerThread::threadLoop_mix() 2692{ 2693 // obtain the presentation timestamp of the next output buffer 2694 int64_t pts; 2695 status_t status = INVALID_OPERATION; 2696 2697 if (NULL != mOutput->stream->get_next_write_timestamp) { 2698 status = mOutput->stream->get_next_write_timestamp( 2699 mOutput->stream, &pts); 2700 } 2701 2702 if (status != NO_ERROR) { 2703 pts = AudioBufferProvider::kInvalidPTS; 2704 } 2705 2706 // mix buffers... 2707 mAudioMixer->process(pts); 2708 // increase sleep time progressively when application underrun condition clears. 2709 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2710 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2711 // such that we would underrun the audio HAL. 2712 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2713 sleepTimeShift--; 2714 } 2715 sleepTime = 0; 2716 standbyTime = systemTime() + standbyDelay; 2717 //TODO: delay standby when effects have a tail 2718} 2719 2720void AudioFlinger::MixerThread::threadLoop_sleepTime() 2721{ 2722 // If no tracks are ready, sleep once for the duration of an output 2723 // buffer size, then write 0s to the output 2724 if (sleepTime == 0) { 2725 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2726 sleepTime = activeSleepTime >> sleepTimeShift; 2727 if (sleepTime < kMinThreadSleepTimeUs) { 2728 sleepTime = kMinThreadSleepTimeUs; 2729 } 2730 // reduce sleep time in case of consecutive application underruns to avoid 2731 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2732 // duration we would end up writing less data than needed by the audio HAL if 2733 // the condition persists. 2734 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2735 sleepTimeShift++; 2736 } 2737 } else { 2738 sleepTime = idleSleepTime; 2739 } 2740 } else if (mBytesWritten != 0 || 2741 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2742 memset (mMixBuffer, 0, mixBufferSize); 2743 sleepTime = 0; 2744 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2745 } 2746 // TODO add standby time extension fct of effect tail 2747} 2748 2749// prepareTracks_l() must be called with ThreadBase::mLock held 2750AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2751 Vector< sp<Track> > *tracksToRemove) 2752{ 2753 2754 mixer_state mixerStatus = MIXER_IDLE; 2755 // find out which tracks need to be processed 2756 size_t count = mActiveTracks.size(); 2757 size_t mixedTracks = 0; 2758 size_t tracksWithEffect = 0; 2759 // counts only _active_ fast tracks 2760 size_t fastTracks = 0; 2761 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2762 2763 float masterVolume = mMasterVolume; 2764 bool masterMute = mMasterMute; 2765 2766 if (masterMute) { 2767 masterVolume = 0; 2768 } 2769 // Delegate master volume control to effect in output mix effect chain if needed 2770 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2771 if (chain != 0) { 2772 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2773 chain->setVolume_l(&v, &v); 2774 masterVolume = (float)((v + (1 << 23)) >> 24); 2775 chain.clear(); 2776 } 2777 2778 // prepare a new state to push 2779 FastMixerStateQueue *sq = NULL; 2780 FastMixerState *state = NULL; 2781 bool didModify = false; 2782 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2783 if (mFastMixer != NULL) { 2784 sq = mFastMixer->sq(); 2785 state = sq->begin(); 2786 } 2787 2788 for (size_t i=0 ; i<count ; i++) { 2789 sp<Track> t = mActiveTracks[i].promote(); 2790 if (t == 0) continue; 2791 2792 // this const just means the local variable doesn't change 2793 Track* const track = t.get(); 2794 2795 // process fast tracks 2796 if (track->isFastTrack()) { 2797 2798 // It's theoretically possible (though unlikely) for a fast track to be created 2799 // and then removed within the same normal mix cycle. This is not a problem, as 2800 // the track never becomes active so it's fast mixer slot is never touched. 2801 // The converse, of removing an (active) track and then creating a new track 2802 // at the identical fast mixer slot within the same normal mix cycle, 2803 // is impossible because the slot isn't marked available until the end of each cycle. 2804 int j = track->mFastIndex; 2805 FastTrack *fastTrack = &state->mFastTracks[j]; 2806 2807 // Determine whether the track is currently in underrun condition, 2808 // and whether it had a recent underrun. 2809 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2810 uint32_t recentFull = (underruns.mBitFields.mFull - 2811 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2812 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2813 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2814 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2815 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2816 uint32_t recentUnderruns = recentPartial + recentEmpty; 2817 track->mObservedUnderruns = underruns; 2818 // don't count underruns that occur while stopping or pausing 2819 // or stopped which can occur when flush() is called while active 2820 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2821 track->mUnderrunCount += recentUnderruns; 2822 } 2823 2824 // This is similar to the state machine for normal tracks, 2825 // with a few modifications for fast tracks. 2826 bool isActive = true; 2827 switch (track->mState) { 2828 case TrackBase::STOPPING_1: 2829 // track stays active in STOPPING_1 state until first underrun 2830 if (recentUnderruns > 0) { 2831 track->mState = TrackBase::STOPPING_2; 2832 } 2833 break; 2834 case TrackBase::PAUSING: 2835 // ramp down is not yet implemented 2836 track->setPaused(); 2837 break; 2838 case TrackBase::RESUMING: 2839 // ramp up is not yet implemented 2840 track->mState = TrackBase::ACTIVE; 2841 break; 2842 case TrackBase::ACTIVE: 2843 if (recentFull > 0 || recentPartial > 0) { 2844 // track has provided at least some frames recently: reset retry count 2845 track->mRetryCount = kMaxTrackRetries; 2846 } 2847 if (recentUnderruns == 0) { 2848 // no recent underruns: stay active 2849 break; 2850 } 2851 // there has recently been an underrun of some kind 2852 if (track->sharedBuffer() == 0) { 2853 // were any of the recent underruns "empty" (no frames available)? 2854 if (recentEmpty == 0) { 2855 // no, then ignore the partial underruns as they are allowed indefinitely 2856 break; 2857 } 2858 // there has recently been an "empty" underrun: decrement the retry counter 2859 if (--(track->mRetryCount) > 0) { 2860 break; 2861 } 2862 // indicate to client process that the track was disabled because of underrun; 2863 // it will then automatically call start() when data is available 2864 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2865 // remove from active list, but state remains ACTIVE [confusing but true] 2866 isActive = false; 2867 break; 2868 } 2869 // fall through 2870 case TrackBase::STOPPING_2: 2871 case TrackBase::PAUSED: 2872 case TrackBase::TERMINATED: 2873 case TrackBase::STOPPED: 2874 case TrackBase::FLUSHED: // flush() while active 2875 // Check for presentation complete if track is inactive 2876 // We have consumed all the buffers of this track. 2877 // This would be incomplete if we auto-paused on underrun 2878 { 2879 size_t audioHALFrames = 2880 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2881 size_t framesWritten = 2882 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2883 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2884 // track stays in active list until presentation is complete 2885 break; 2886 } 2887 } 2888 if (track->isStopping_2()) { 2889 track->mState = TrackBase::STOPPED; 2890 } 2891 if (track->isStopped()) { 2892 // Can't reset directly, as fast mixer is still polling this track 2893 // track->reset(); 2894 // So instead mark this track as needing to be reset after push with ack 2895 resetMask |= 1 << i; 2896 } 2897 isActive = false; 2898 break; 2899 case TrackBase::IDLE: 2900 default: 2901 LOG_FATAL("unexpected track state %d", track->mState); 2902 } 2903 2904 if (isActive) { 2905 // was it previously inactive? 2906 if (!(state->mTrackMask & (1 << j))) { 2907 ExtendedAudioBufferProvider *eabp = track; 2908 VolumeProvider *vp = track; 2909 fastTrack->mBufferProvider = eabp; 2910 fastTrack->mVolumeProvider = vp; 2911 fastTrack->mSampleRate = track->mSampleRate; 2912 fastTrack->mChannelMask = track->mChannelMask; 2913 fastTrack->mGeneration++; 2914 state->mTrackMask |= 1 << j; 2915 didModify = true; 2916 // no acknowledgement required for newly active tracks 2917 } 2918 // cache the combined master volume and stream type volume for fast mixer; this 2919 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2920 track->mCachedVolume = track->isMuted() ? 2921 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2922 ++fastTracks; 2923 } else { 2924 // was it previously active? 2925 if (state->mTrackMask & (1 << j)) { 2926 fastTrack->mBufferProvider = NULL; 2927 fastTrack->mGeneration++; 2928 state->mTrackMask &= ~(1 << j); 2929 didModify = true; 2930 // If any fast tracks were removed, we must wait for acknowledgement 2931 // because we're about to decrement the last sp<> on those tracks. 2932 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2933 } else { 2934 LOG_FATAL("fast track %d should have been active", j); 2935 } 2936 tracksToRemove->add(track); 2937 // Avoids a misleading display in dumpsys 2938 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2939 } 2940 continue; 2941 } 2942 2943 { // local variable scope to avoid goto warning 2944 2945 audio_track_cblk_t* cblk = track->cblk(); 2946 2947 // The first time a track is added we wait 2948 // for all its buffers to be filled before processing it 2949 int name = track->name(); 2950 // make sure that we have enough frames to mix one full buffer. 2951 // enforce this condition only once to enable draining the buffer in case the client 2952 // app does not call stop() and relies on underrun to stop: 2953 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2954 // during last round 2955 uint32_t minFrames = 1; 2956 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2957 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2958 if (t->sampleRate() == (int)mSampleRate) { 2959 minFrames = mNormalFrameCount; 2960 } else { 2961 // +1 for rounding and +1 for additional sample needed for interpolation 2962 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2963 // add frames already consumed but not yet released by the resampler 2964 // because cblk->framesReady() will include these frames 2965 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2966 // the minimum track buffer size is normally twice the number of frames necessary 2967 // to fill one buffer and the resampler should not leave more than one buffer worth 2968 // of unreleased frames after each pass, but just in case... 2969 ALOG_ASSERT(minFrames <= cblk->frameCount); 2970 } 2971 } 2972 if ((track->framesReady() >= minFrames) && track->isReady() && 2973 !track->isPaused() && !track->isTerminated()) 2974 { 2975 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2976 2977 mixedTracks++; 2978 2979 // track->mainBuffer() != mMixBuffer means there is an effect chain 2980 // connected to the track 2981 chain.clear(); 2982 if (track->mainBuffer() != mMixBuffer) { 2983 chain = getEffectChain_l(track->sessionId()); 2984 // Delegate volume control to effect in track effect chain if needed 2985 if (chain != 0) { 2986 tracksWithEffect++; 2987 } else { 2988 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2989 name, track->sessionId()); 2990 } 2991 } 2992 2993 2994 int param = AudioMixer::VOLUME; 2995 if (track->mFillingUpStatus == Track::FS_FILLED) { 2996 // no ramp for the first volume setting 2997 track->mFillingUpStatus = Track::FS_ACTIVE; 2998 if (track->mState == TrackBase::RESUMING) { 2999 track->mState = TrackBase::ACTIVE; 3000 param = AudioMixer::RAMP_VOLUME; 3001 } 3002 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3003 } else if (cblk->server != 0) { 3004 // If the track is stopped before the first frame was mixed, 3005 // do not apply ramp 3006 param = AudioMixer::RAMP_VOLUME; 3007 } 3008 3009 // compute volume for this track 3010 uint32_t vl, vr, va; 3011 if (track->isMuted() || track->isPausing() || 3012 mStreamTypes[track->streamType()].mute) { 3013 vl = vr = va = 0; 3014 if (track->isPausing()) { 3015 track->setPaused(); 3016 } 3017 } else { 3018 3019 // read original volumes with volume control 3020 float typeVolume = mStreamTypes[track->streamType()].volume; 3021 float v = masterVolume * typeVolume; 3022 uint32_t vlr = cblk->getVolumeLR(); 3023 vl = vlr & 0xFFFF; 3024 vr = vlr >> 16; 3025 // track volumes come from shared memory, so can't be trusted and must be clamped 3026 if (vl > MAX_GAIN_INT) { 3027 ALOGV("Track left volume out of range: %04X", vl); 3028 vl = MAX_GAIN_INT; 3029 } 3030 if (vr > MAX_GAIN_INT) { 3031 ALOGV("Track right volume out of range: %04X", vr); 3032 vr = MAX_GAIN_INT; 3033 } 3034 // now apply the master volume and stream type volume 3035 vl = (uint32_t)(v * vl) << 12; 3036 vr = (uint32_t)(v * vr) << 12; 3037 // assuming master volume and stream type volume each go up to 1.0, 3038 // vl and vr are now in 8.24 format 3039 3040 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3041 // send level comes from shared memory and so may be corrupt 3042 if (sendLevel > MAX_GAIN_INT) { 3043 ALOGV("Track send level out of range: %04X", sendLevel); 3044 sendLevel = MAX_GAIN_INT; 3045 } 3046 va = (uint32_t)(v * sendLevel); 3047 } 3048 // Delegate volume control to effect in track effect chain if needed 3049 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3050 // Do not ramp volume if volume is controlled by effect 3051 param = AudioMixer::VOLUME; 3052 track->mHasVolumeController = true; 3053 } else { 3054 // force no volume ramp when volume controller was just disabled or removed 3055 // from effect chain to avoid volume spike 3056 if (track->mHasVolumeController) { 3057 param = AudioMixer::VOLUME; 3058 } 3059 track->mHasVolumeController = false; 3060 } 3061 3062 // Convert volumes from 8.24 to 4.12 format 3063 // This additional clamping is needed in case chain->setVolume_l() overshot 3064 vl = (vl + (1 << 11)) >> 12; 3065 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3066 vr = (vr + (1 << 11)) >> 12; 3067 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3068 3069 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3070 3071 // XXX: these things DON'T need to be done each time 3072 mAudioMixer->setBufferProvider(name, track); 3073 mAudioMixer->enable(name); 3074 3075 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3076 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3077 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3078 mAudioMixer->setParameter( 3079 name, 3080 AudioMixer::TRACK, 3081 AudioMixer::FORMAT, (void *)track->format()); 3082 mAudioMixer->setParameter( 3083 name, 3084 AudioMixer::TRACK, 3085 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3086 mAudioMixer->setParameter( 3087 name, 3088 AudioMixer::RESAMPLE, 3089 AudioMixer::SAMPLE_RATE, 3090 (void *)(cblk->sampleRate)); 3091 mAudioMixer->setParameter( 3092 name, 3093 AudioMixer::TRACK, 3094 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3095 mAudioMixer->setParameter( 3096 name, 3097 AudioMixer::TRACK, 3098 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3099 3100 // reset retry count 3101 track->mRetryCount = kMaxTrackRetries; 3102 3103 // If one track is ready, set the mixer ready if: 3104 // - the mixer was not ready during previous round OR 3105 // - no other track is not ready 3106 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3107 mixerStatus != MIXER_TRACKS_ENABLED) { 3108 mixerStatus = MIXER_TRACKS_READY; 3109 } 3110 } else { 3111 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3112 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3113 track->isStopped() || track->isPaused()) { 3114 // We have consumed all the buffers of this track. 3115 // Remove it from the list of active tracks. 3116 // TODO: use actual buffer filling status instead of latency when available from 3117 // audio HAL 3118 size_t audioHALFrames = 3119 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3120 size_t framesWritten = 3121 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3122 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3123 if (track->isStopped()) { 3124 track->reset(); 3125 } 3126 tracksToRemove->add(track); 3127 } 3128 } else { 3129 // No buffers for this track. Give it a few chances to 3130 // fill a buffer, then remove it from active list. 3131 if (--(track->mRetryCount) <= 0) { 3132 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3133 tracksToRemove->add(track); 3134 // indicate to client process that the track was disabled because of underrun; 3135 // it will then automatically call start() when data is available 3136 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3137 // If one track is not ready, mark the mixer also not ready if: 3138 // - the mixer was ready during previous round OR 3139 // - no other track is ready 3140 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3141 mixerStatus != MIXER_TRACKS_READY) { 3142 mixerStatus = MIXER_TRACKS_ENABLED; 3143 } 3144 } 3145 mAudioMixer->disable(name); 3146 } 3147 3148 } // local variable scope to avoid goto warning 3149track_is_ready: ; 3150 3151 } 3152 3153 // Push the new FastMixer state if necessary 3154 if (didModify) { 3155 state->mFastTracksGen++; 3156 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3157 if (kUseFastMixer == FastMixer_Dynamic && 3158 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3159 state->mCommand = FastMixerState::COLD_IDLE; 3160 state->mColdFutexAddr = &mFastMixerFutex; 3161 state->mColdGen++; 3162 mFastMixerFutex = 0; 3163 if (kUseFastMixer == FastMixer_Dynamic) { 3164 mNormalSink = mOutputSink; 3165 } 3166 // If we go into cold idle, need to wait for acknowledgement 3167 // so that fast mixer stops doing I/O. 3168 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3169 } 3170 sq->end(); 3171 } 3172 if (sq != NULL) { 3173 sq->end(didModify); 3174 sq->push(block); 3175 } 3176 3177 // Now perform the deferred reset on fast tracks that have stopped 3178 while (resetMask != 0) { 3179 size_t i = __builtin_ctz(resetMask); 3180 ALOG_ASSERT(i < count); 3181 resetMask &= ~(1 << i); 3182 sp<Track> t = mActiveTracks[i].promote(); 3183 if (t == 0) continue; 3184 Track* track = t.get(); 3185 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3186 track->reset(); 3187 } 3188 3189 // remove all the tracks that need to be... 3190 count = tracksToRemove->size(); 3191 if (CC_UNLIKELY(count)) { 3192 for (size_t i=0 ; i<count ; i++) { 3193 const sp<Track>& track = tracksToRemove->itemAt(i); 3194 mActiveTracks.remove(track); 3195 if (track->mainBuffer() != mMixBuffer) { 3196 chain = getEffectChain_l(track->sessionId()); 3197 if (chain != 0) { 3198 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3199 chain->decActiveTrackCnt(); 3200 } 3201 } 3202 if (track->isTerminated()) { 3203 removeTrack_l(track); 3204 } 3205 } 3206 } 3207 3208 // mix buffer must be cleared if all tracks are connected to an 3209 // effect chain as in this case the mixer will not write to 3210 // mix buffer and track effects will accumulate into it 3211 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3212 // FIXME as a performance optimization, should remember previous zero status 3213 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3214 } 3215 3216 // if any fast tracks, then status is ready 3217 mMixerStatusIgnoringFastTracks = mixerStatus; 3218 if (fastTracks > 0) { 3219 mixerStatus = MIXER_TRACKS_READY; 3220 } 3221 return mixerStatus; 3222} 3223 3224/* 3225The derived values that are cached: 3226 - mixBufferSize from frame count * frame size 3227 - activeSleepTime from activeSleepTimeUs() 3228 - idleSleepTime from idleSleepTimeUs() 3229 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3230 - maxPeriod from frame count and sample rate (MIXER only) 3231 3232The parameters that affect these derived values are: 3233 - frame count 3234 - frame size 3235 - sample rate 3236 - device type: A2DP or not 3237 - device latency 3238 - format: PCM or not 3239 - active sleep time 3240 - idle sleep time 3241*/ 3242 3243void AudioFlinger::PlaybackThread::cacheParameters_l() 3244{ 3245 mixBufferSize = mNormalFrameCount * mFrameSize; 3246 activeSleepTime = activeSleepTimeUs(); 3247 idleSleepTime = idleSleepTimeUs(); 3248} 3249 3250void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3251{ 3252 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3253 this, streamType, mTracks.size()); 3254 Mutex::Autolock _l(mLock); 3255 3256 size_t size = mTracks.size(); 3257 for (size_t i = 0; i < size; i++) { 3258 sp<Track> t = mTracks[i]; 3259 if (t->streamType() == streamType) { 3260 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3261 t->mCblk->cv.signal(); 3262 } 3263 } 3264} 3265 3266// getTrackName_l() must be called with ThreadBase::mLock held 3267int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3268{ 3269 return mAudioMixer->getTrackName(channelMask); 3270} 3271 3272// deleteTrackName_l() must be called with ThreadBase::mLock held 3273void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3274{ 3275 ALOGV("remove track (%d) and delete from mixer", name); 3276 mAudioMixer->deleteTrackName(name); 3277} 3278 3279// checkForNewParameters_l() must be called with ThreadBase::mLock held 3280bool AudioFlinger::MixerThread::checkForNewParameters_l() 3281{ 3282 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3283 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3284 bool reconfig = false; 3285 3286 while (!mNewParameters.isEmpty()) { 3287 3288 if (mFastMixer != NULL) { 3289 FastMixerStateQueue *sq = mFastMixer->sq(); 3290 FastMixerState *state = sq->begin(); 3291 if (!(state->mCommand & FastMixerState::IDLE)) { 3292 previousCommand = state->mCommand; 3293 state->mCommand = FastMixerState::HOT_IDLE; 3294 sq->end(); 3295 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3296 } else { 3297 sq->end(false /*didModify*/); 3298 } 3299 } 3300 3301 status_t status = NO_ERROR; 3302 String8 keyValuePair = mNewParameters[0]; 3303 AudioParameter param = AudioParameter(keyValuePair); 3304 int value; 3305 3306 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3307 reconfig = true; 3308 } 3309 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3310 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3311 status = BAD_VALUE; 3312 } else { 3313 reconfig = true; 3314 } 3315 } 3316 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3317 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3318 status = BAD_VALUE; 3319 } else { 3320 reconfig = true; 3321 } 3322 } 3323 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3324 // do not accept frame count changes if tracks are open as the track buffer 3325 // size depends on frame count and correct behavior would not be guaranteed 3326 // if frame count is changed after track creation 3327 if (!mTracks.isEmpty()) { 3328 status = INVALID_OPERATION; 3329 } else { 3330 reconfig = true; 3331 } 3332 } 3333 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3334#ifdef ADD_BATTERY_DATA 3335 // when changing the audio output device, call addBatteryData to notify 3336 // the change 3337 if ((int)mDevice != value) { 3338 uint32_t params = 0; 3339 // check whether speaker is on 3340 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3341 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3342 } 3343 3344 int deviceWithoutSpeaker 3345 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3346 // check if any other device (except speaker) is on 3347 if (value & deviceWithoutSpeaker ) { 3348 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3349 } 3350 3351 if (params != 0) { 3352 addBatteryData(params); 3353 } 3354 } 3355#endif 3356 3357 // forward device change to effects that have requested to be 3358 // aware of attached audio device. 3359 mDevice = (uint32_t)value; 3360 for (size_t i = 0; i < mEffectChains.size(); i++) { 3361 mEffectChains[i]->setDevice_l(mDevice); 3362 } 3363 } 3364 3365 if (status == NO_ERROR) { 3366 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3367 keyValuePair.string()); 3368 if (!mStandby && status == INVALID_OPERATION) { 3369 mOutput->stream->common.standby(&mOutput->stream->common); 3370 mStandby = true; 3371 mBytesWritten = 0; 3372 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3373 keyValuePair.string()); 3374 } 3375 if (status == NO_ERROR && reconfig) { 3376 delete mAudioMixer; 3377 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3378 mAudioMixer = NULL; 3379 readOutputParameters(); 3380 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3381 for (size_t i = 0; i < mTracks.size() ; i++) { 3382 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3383 if (name < 0) break; 3384 mTracks[i]->mName = name; 3385 // limit track sample rate to 2 x new output sample rate 3386 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3387 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3388 } 3389 } 3390 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3391 } 3392 } 3393 3394 mNewParameters.removeAt(0); 3395 3396 mParamStatus = status; 3397 mParamCond.signal(); 3398 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3399 // already timed out waiting for the status and will never signal the condition. 3400 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3401 } 3402 3403 if (!(previousCommand & FastMixerState::IDLE)) { 3404 ALOG_ASSERT(mFastMixer != NULL); 3405 FastMixerStateQueue *sq = mFastMixer->sq(); 3406 FastMixerState *state = sq->begin(); 3407 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3408 state->mCommand = previousCommand; 3409 sq->end(); 3410 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3411 } 3412 3413 return reconfig; 3414} 3415 3416status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3417{ 3418 const size_t SIZE = 256; 3419 char buffer[SIZE]; 3420 String8 result; 3421 3422 PlaybackThread::dumpInternals(fd, args); 3423 3424 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3425 result.append(buffer); 3426 write(fd, result.string(), result.size()); 3427 3428 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3429 FastMixerDumpState copy = mFastMixerDumpState; 3430 copy.dump(fd); 3431 3432 return NO_ERROR; 3433} 3434 3435uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3436{ 3437 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3438} 3439 3440uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3441{ 3442 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3443} 3444 3445void AudioFlinger::MixerThread::cacheParameters_l() 3446{ 3447 PlaybackThread::cacheParameters_l(); 3448 3449 // FIXME: Relaxed timing because of a certain device that can't meet latency 3450 // Should be reduced to 2x after the vendor fixes the driver issue 3451 // increase threshold again due to low power audio mode. The way this warning 3452 // threshold is calculated and its usefulness should be reconsidered anyway. 3453 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3454} 3455 3456// ---------------------------------------------------------------------------- 3457AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3458 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3459 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3460 // mLeftVolFloat, mRightVolFloat 3461 // mLeftVolShort, mRightVolShort 3462{ 3463} 3464 3465AudioFlinger::DirectOutputThread::~DirectOutputThread() 3466{ 3467} 3468 3469AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3470 Vector< sp<Track> > *tracksToRemove 3471) 3472{ 3473 sp<Track> trackToRemove; 3474 3475 mixer_state mixerStatus = MIXER_IDLE; 3476 3477 // find out which tracks need to be processed 3478 if (mActiveTracks.size() != 0) { 3479 sp<Track> t = mActiveTracks[0].promote(); 3480 // The track died recently 3481 if (t == 0) return MIXER_IDLE; 3482 3483 Track* const track = t.get(); 3484 audio_track_cblk_t* cblk = track->cblk(); 3485 3486 // The first time a track is added we wait 3487 // for all its buffers to be filled before processing it 3488 if (cblk->framesReady() && track->isReady() && 3489 !track->isPaused() && !track->isTerminated()) 3490 { 3491 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3492 3493 if (track->mFillingUpStatus == Track::FS_FILLED) { 3494 track->mFillingUpStatus = Track::FS_ACTIVE; 3495 mLeftVolFloat = mRightVolFloat = 0; 3496 mLeftVolShort = mRightVolShort = 0; 3497 if (track->mState == TrackBase::RESUMING) { 3498 track->mState = TrackBase::ACTIVE; 3499 rampVolume = true; 3500 } 3501 } else if (cblk->server != 0) { 3502 // If the track is stopped before the first frame was mixed, 3503 // do not apply ramp 3504 rampVolume = true; 3505 } 3506 // compute volume for this track 3507 float left, right; 3508 if (track->isMuted() || mMasterMute || track->isPausing() || 3509 mStreamTypes[track->streamType()].mute) { 3510 left = right = 0; 3511 if (track->isPausing()) { 3512 track->setPaused(); 3513 } 3514 } else { 3515 float typeVolume = mStreamTypes[track->streamType()].volume; 3516 float v = mMasterVolume * typeVolume; 3517 uint32_t vlr = cblk->getVolumeLR(); 3518 float v_clamped = v * (vlr & 0xFFFF); 3519 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3520 left = v_clamped/MAX_GAIN; 3521 v_clamped = v * (vlr >> 16); 3522 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3523 right = v_clamped/MAX_GAIN; 3524 } 3525 3526 if (left != mLeftVolFloat || right != mRightVolFloat) { 3527 mLeftVolFloat = left; 3528 mRightVolFloat = right; 3529 3530 // If audio HAL implements volume control, 3531 // force software volume to nominal value 3532 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3533 left = 1.0f; 3534 right = 1.0f; 3535 } 3536 3537 // Convert volumes from float to 8.24 3538 uint32_t vl = (uint32_t)(left * (1 << 24)); 3539 uint32_t vr = (uint32_t)(right * (1 << 24)); 3540 3541 // Delegate volume control to effect in track effect chain if needed 3542 // only one effect chain can be present on DirectOutputThread, so if 3543 // there is one, the track is connected to it 3544 if (!mEffectChains.isEmpty()) { 3545 // Do not ramp volume if volume is controlled by effect 3546 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3547 rampVolume = false; 3548 } 3549 } 3550 3551 // Convert volumes from 8.24 to 4.12 format 3552 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3553 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3554 leftVol = (uint16_t)v_clamped; 3555 v_clamped = (vr + (1 << 11)) >> 12; 3556 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3557 rightVol = (uint16_t)v_clamped; 3558 } else { 3559 leftVol = mLeftVolShort; 3560 rightVol = mRightVolShort; 3561 rampVolume = false; 3562 } 3563 3564 // reset retry count 3565 track->mRetryCount = kMaxTrackRetriesDirect; 3566 mActiveTrack = t; 3567 mixerStatus = MIXER_TRACKS_READY; 3568 } else { 3569 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3570 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3571 // We have consumed all the buffers of this track. 3572 // Remove it from the list of active tracks. 3573 // TODO: implement behavior for compressed audio 3574 size_t audioHALFrames = 3575 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3576 size_t framesWritten = 3577 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3578 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3579 if (track->isStopped()) { 3580 track->reset(); 3581 } 3582 trackToRemove = track; 3583 } 3584 } else { 3585 // No buffers for this track. Give it a few chances to 3586 // fill a buffer, then remove it from active list. 3587 if (--(track->mRetryCount) <= 0) { 3588 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3589 trackToRemove = track; 3590 } else { 3591 mixerStatus = MIXER_TRACKS_ENABLED; 3592 } 3593 } 3594 } 3595 } 3596 3597 // FIXME merge this with similar code for removing multiple tracks 3598 // remove all the tracks that need to be... 3599 if (CC_UNLIKELY(trackToRemove != 0)) { 3600 tracksToRemove->add(trackToRemove); 3601 mActiveTracks.remove(trackToRemove); 3602 if (!mEffectChains.isEmpty()) { 3603 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3604 trackToRemove->sessionId()); 3605 mEffectChains[0]->decActiveTrackCnt(); 3606 } 3607 if (trackToRemove->isTerminated()) { 3608 removeTrack_l(trackToRemove); 3609 } 3610 } 3611 3612 return mixerStatus; 3613} 3614 3615void AudioFlinger::DirectOutputThread::threadLoop_mix() 3616{ 3617 AudioBufferProvider::Buffer buffer; 3618 size_t frameCount = mFrameCount; 3619 int8_t *curBuf = (int8_t *)mMixBuffer; 3620 // output audio to hardware 3621 while (frameCount) { 3622 buffer.frameCount = frameCount; 3623 mActiveTrack->getNextBuffer(&buffer); 3624 if (CC_UNLIKELY(buffer.raw == NULL)) { 3625 memset(curBuf, 0, frameCount * mFrameSize); 3626 break; 3627 } 3628 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3629 frameCount -= buffer.frameCount; 3630 curBuf += buffer.frameCount * mFrameSize; 3631 mActiveTrack->releaseBuffer(&buffer); 3632 } 3633 sleepTime = 0; 3634 standbyTime = systemTime() + standbyDelay; 3635 mActiveTrack.clear(); 3636 3637 // apply volume 3638 3639 // Do not apply volume on compressed audio 3640 if (!audio_is_linear_pcm(mFormat)) { 3641 return; 3642 } 3643 3644 // convert to signed 16 bit before volume calculation 3645 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3646 size_t count = mFrameCount * mChannelCount; 3647 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3648 int16_t *dst = mMixBuffer + count-1; 3649 while (count--) { 3650 *dst-- = (int16_t)(*src--^0x80) << 8; 3651 } 3652 } 3653 3654 frameCount = mFrameCount; 3655 int16_t *out = mMixBuffer; 3656 if (rampVolume) { 3657 if (mChannelCount == 1) { 3658 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3659 int32_t vlInc = d / (int32_t)frameCount; 3660 int32_t vl = ((int32_t)mLeftVolShort << 16); 3661 do { 3662 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3663 out++; 3664 vl += vlInc; 3665 } while (--frameCount); 3666 3667 } else { 3668 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3669 int32_t vlInc = d / (int32_t)frameCount; 3670 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3671 int32_t vrInc = d / (int32_t)frameCount; 3672 int32_t vl = ((int32_t)mLeftVolShort << 16); 3673 int32_t vr = ((int32_t)mRightVolShort << 16); 3674 do { 3675 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3676 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3677 out += 2; 3678 vl += vlInc; 3679 vr += vrInc; 3680 } while (--frameCount); 3681 } 3682 } else { 3683 if (mChannelCount == 1) { 3684 do { 3685 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3686 out++; 3687 } while (--frameCount); 3688 } else { 3689 do { 3690 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3691 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3692 out += 2; 3693 } while (--frameCount); 3694 } 3695 } 3696 3697 // convert back to unsigned 8 bit after volume calculation 3698 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3699 size_t count = mFrameCount * mChannelCount; 3700 int16_t *src = mMixBuffer; 3701 uint8_t *dst = (uint8_t *)mMixBuffer; 3702 while (count--) { 3703 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3704 } 3705 } 3706 3707 mLeftVolShort = leftVol; 3708 mRightVolShort = rightVol; 3709} 3710 3711void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3712{ 3713 if (sleepTime == 0) { 3714 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3715 sleepTime = activeSleepTime; 3716 } else { 3717 sleepTime = idleSleepTime; 3718 } 3719 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3720 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3721 sleepTime = 0; 3722 } 3723} 3724 3725// getTrackName_l() must be called with ThreadBase::mLock held 3726int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3727{ 3728 return 0; 3729} 3730 3731// deleteTrackName_l() must be called with ThreadBase::mLock held 3732void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3733{ 3734} 3735 3736// checkForNewParameters_l() must be called with ThreadBase::mLock held 3737bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3738{ 3739 bool reconfig = false; 3740 3741 while (!mNewParameters.isEmpty()) { 3742 status_t status = NO_ERROR; 3743 String8 keyValuePair = mNewParameters[0]; 3744 AudioParameter param = AudioParameter(keyValuePair); 3745 int value; 3746 3747 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3748 // do not accept frame count changes if tracks are open as the track buffer 3749 // size depends on frame count and correct behavior would not be garantied 3750 // if frame count is changed after track creation 3751 if (!mTracks.isEmpty()) { 3752 status = INVALID_OPERATION; 3753 } else { 3754 reconfig = true; 3755 } 3756 } 3757 if (status == NO_ERROR) { 3758 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3759 keyValuePair.string()); 3760 if (!mStandby && status == INVALID_OPERATION) { 3761 mOutput->stream->common.standby(&mOutput->stream->common); 3762 mStandby = true; 3763 mBytesWritten = 0; 3764 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3765 keyValuePair.string()); 3766 } 3767 if (status == NO_ERROR && reconfig) { 3768 readOutputParameters(); 3769 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3770 } 3771 } 3772 3773 mNewParameters.removeAt(0); 3774 3775 mParamStatus = status; 3776 mParamCond.signal(); 3777 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3778 // already timed out waiting for the status and will never signal the condition. 3779 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3780 } 3781 return reconfig; 3782} 3783 3784uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3785{ 3786 uint32_t time; 3787 if (audio_is_linear_pcm(mFormat)) { 3788 time = PlaybackThread::activeSleepTimeUs(); 3789 } else { 3790 time = 10000; 3791 } 3792 return time; 3793} 3794 3795uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3796{ 3797 uint32_t time; 3798 if (audio_is_linear_pcm(mFormat)) { 3799 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3800 } else { 3801 time = 10000; 3802 } 3803 return time; 3804} 3805 3806uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3807{ 3808 uint32_t time; 3809 if (audio_is_linear_pcm(mFormat)) { 3810 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3811 } else { 3812 time = 10000; 3813 } 3814 return time; 3815} 3816 3817void AudioFlinger::DirectOutputThread::cacheParameters_l() 3818{ 3819 PlaybackThread::cacheParameters_l(); 3820 3821 // use shorter standby delay as on normal output to release 3822 // hardware resources as soon as possible 3823 standbyDelay = microseconds(activeSleepTime*2); 3824} 3825 3826// ---------------------------------------------------------------------------- 3827 3828AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3829 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3830 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3831 mWaitTimeMs(UINT_MAX) 3832{ 3833 addOutputTrack(mainThread); 3834} 3835 3836AudioFlinger::DuplicatingThread::~DuplicatingThread() 3837{ 3838 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3839 mOutputTracks[i]->destroy(); 3840 } 3841} 3842 3843void AudioFlinger::DuplicatingThread::threadLoop_mix() 3844{ 3845 // mix buffers... 3846 if (outputsReady(outputTracks)) { 3847 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3848 } else { 3849 memset(mMixBuffer, 0, mixBufferSize); 3850 } 3851 sleepTime = 0; 3852 writeFrames = mNormalFrameCount; 3853} 3854 3855void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3856{ 3857 if (sleepTime == 0) { 3858 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3859 sleepTime = activeSleepTime; 3860 } else { 3861 sleepTime = idleSleepTime; 3862 } 3863 } else if (mBytesWritten != 0) { 3864 // flush remaining overflow buffers in output tracks 3865 for (size_t i = 0; i < outputTracks.size(); i++) { 3866 if (outputTracks[i]->isActive()) { 3867 sleepTime = 0; 3868 writeFrames = 0; 3869 memset(mMixBuffer, 0, mixBufferSize); 3870 break; 3871 } 3872 } 3873 } 3874} 3875 3876void AudioFlinger::DuplicatingThread::threadLoop_write() 3877{ 3878 standbyTime = systemTime() + standbyDelay; 3879 for (size_t i = 0; i < outputTracks.size(); i++) { 3880 outputTracks[i]->write(mMixBuffer, writeFrames); 3881 } 3882 mBytesWritten += mixBufferSize; 3883} 3884 3885void AudioFlinger::DuplicatingThread::threadLoop_standby() 3886{ 3887 // DuplicatingThread implements standby by stopping all tracks 3888 for (size_t i = 0; i < outputTracks.size(); i++) { 3889 outputTracks[i]->stop(); 3890 } 3891} 3892 3893void AudioFlinger::DuplicatingThread::saveOutputTracks() 3894{ 3895 outputTracks = mOutputTracks; 3896} 3897 3898void AudioFlinger::DuplicatingThread::clearOutputTracks() 3899{ 3900 outputTracks.clear(); 3901} 3902 3903void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3904{ 3905 Mutex::Autolock _l(mLock); 3906 // FIXME explain this formula 3907 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3908 OutputTrack *outputTrack = new OutputTrack(thread, 3909 this, 3910 mSampleRate, 3911 mFormat, 3912 mChannelMask, 3913 frameCount); 3914 if (outputTrack->cblk() != NULL) { 3915 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3916 mOutputTracks.add(outputTrack); 3917 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3918 updateWaitTime_l(); 3919 } 3920} 3921 3922void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3923{ 3924 Mutex::Autolock _l(mLock); 3925 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3926 if (mOutputTracks[i]->thread() == thread) { 3927 mOutputTracks[i]->destroy(); 3928 mOutputTracks.removeAt(i); 3929 updateWaitTime_l(); 3930 return; 3931 } 3932 } 3933 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3934} 3935 3936// caller must hold mLock 3937void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3938{ 3939 mWaitTimeMs = UINT_MAX; 3940 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3941 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3942 if (strong != 0) { 3943 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3944 if (waitTimeMs < mWaitTimeMs) { 3945 mWaitTimeMs = waitTimeMs; 3946 } 3947 } 3948 } 3949} 3950 3951 3952bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3953{ 3954 for (size_t i = 0; i < outputTracks.size(); i++) { 3955 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3956 if (thread == 0) { 3957 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3958 return false; 3959 } 3960 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3961 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3962 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3963 return false; 3964 } 3965 } 3966 return true; 3967} 3968 3969uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3970{ 3971 return (mWaitTimeMs * 1000) / 2; 3972} 3973 3974void AudioFlinger::DuplicatingThread::cacheParameters_l() 3975{ 3976 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3977 updateWaitTime_l(); 3978 3979 MixerThread::cacheParameters_l(); 3980} 3981 3982// ---------------------------------------------------------------------------- 3983 3984// TrackBase constructor must be called with AudioFlinger::mLock held 3985AudioFlinger::ThreadBase::TrackBase::TrackBase( 3986 ThreadBase *thread, 3987 const sp<Client>& client, 3988 uint32_t sampleRate, 3989 audio_format_t format, 3990 uint32_t channelMask, 3991 int frameCount, 3992 const sp<IMemory>& sharedBuffer, 3993 int sessionId) 3994 : RefBase(), 3995 mThread(thread), 3996 mClient(client), 3997 mCblk(NULL), 3998 // mBuffer 3999 // mBufferEnd 4000 mFrameCount(0), 4001 mState(IDLE), 4002 mSampleRate(sampleRate), 4003 mFormat(format), 4004 mStepServerFailed(false), 4005 mSessionId(sessionId) 4006 // mChannelCount 4007 // mChannelMask 4008{ 4009 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4010 4011 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4012 size_t size = sizeof(audio_track_cblk_t); 4013 uint8_t channelCount = popcount(channelMask); 4014 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4015 if (sharedBuffer == 0) { 4016 size += bufferSize; 4017 } 4018 4019 if (client != NULL) { 4020 mCblkMemory = client->heap()->allocate(size); 4021 if (mCblkMemory != 0) { 4022 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4023 if (mCblk != NULL) { // construct the shared structure in-place. 4024 new(mCblk) audio_track_cblk_t(); 4025 // clear all buffers 4026 mCblk->frameCount = frameCount; 4027 mCblk->sampleRate = sampleRate; 4028// uncomment the following lines to quickly test 32-bit wraparound 4029// mCblk->user = 0xffff0000; 4030// mCblk->server = 0xffff0000; 4031// mCblk->userBase = 0xffff0000; 4032// mCblk->serverBase = 0xffff0000; 4033 mChannelCount = channelCount; 4034 mChannelMask = channelMask; 4035 if (sharedBuffer == 0) { 4036 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4037 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4038 // Force underrun condition to avoid false underrun callback until first data is 4039 // written to buffer (other flags are cleared) 4040 mCblk->flags = CBLK_UNDERRUN_ON; 4041 } else { 4042 mBuffer = sharedBuffer->pointer(); 4043 } 4044 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4045 } 4046 } else { 4047 ALOGE("not enough memory for AudioTrack size=%u", size); 4048 client->heap()->dump("AudioTrack"); 4049 return; 4050 } 4051 } else { 4052 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4053 // construct the shared structure in-place. 4054 new(mCblk) audio_track_cblk_t(); 4055 // clear all buffers 4056 mCblk->frameCount = frameCount; 4057 mCblk->sampleRate = sampleRate; 4058// uncomment the following lines to quickly test 32-bit wraparound 4059// mCblk->user = 0xffff0000; 4060// mCblk->server = 0xffff0000; 4061// mCblk->userBase = 0xffff0000; 4062// mCblk->serverBase = 0xffff0000; 4063 mChannelCount = channelCount; 4064 mChannelMask = channelMask; 4065 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4066 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4067 // Force underrun condition to avoid false underrun callback until first data is 4068 // written to buffer (other flags are cleared) 4069 mCblk->flags = CBLK_UNDERRUN_ON; 4070 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4071 } 4072} 4073 4074AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4075{ 4076 if (mCblk != NULL) { 4077 if (mClient == 0) { 4078 delete mCblk; 4079 } else { 4080 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4081 } 4082 } 4083 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4084 if (mClient != 0) { 4085 // Client destructor must run with AudioFlinger mutex locked 4086 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4087 // If the client's reference count drops to zero, the associated destructor 4088 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4089 // relying on the automatic clear() at end of scope. 4090 mClient.clear(); 4091 } 4092} 4093 4094// AudioBufferProvider interface 4095// getNextBuffer() = 0; 4096// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4097void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4098{ 4099 buffer->raw = NULL; 4100 mFrameCount = buffer->frameCount; 4101 // FIXME See note at getNextBuffer() 4102 (void) step(); // ignore return value of step() 4103 buffer->frameCount = 0; 4104} 4105 4106bool AudioFlinger::ThreadBase::TrackBase::step() { 4107 bool result; 4108 audio_track_cblk_t* cblk = this->cblk(); 4109 4110 result = cblk->stepServer(mFrameCount); 4111 if (!result) { 4112 ALOGV("stepServer failed acquiring cblk mutex"); 4113 mStepServerFailed = true; 4114 } 4115 return result; 4116} 4117 4118void AudioFlinger::ThreadBase::TrackBase::reset() { 4119 audio_track_cblk_t* cblk = this->cblk(); 4120 4121 cblk->user = 0; 4122 cblk->server = 0; 4123 cblk->userBase = 0; 4124 cblk->serverBase = 0; 4125 mStepServerFailed = false; 4126 ALOGV("TrackBase::reset"); 4127} 4128 4129int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4130 return (int)mCblk->sampleRate; 4131} 4132 4133void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4134 audio_track_cblk_t* cblk = this->cblk(); 4135 size_t frameSize = cblk->frameSize; 4136 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4137 int8_t *bufferEnd = bufferStart + frames * frameSize; 4138 4139 // Check validity of returned pointer in case the track control block would have been corrupted. 4140 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4141 "TrackBase::getBuffer buffer out of range:\n" 4142 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4143 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4144 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4145 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4146 4147 return bufferStart; 4148} 4149 4150status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4151{ 4152 mSyncEvents.add(event); 4153 return NO_ERROR; 4154} 4155 4156// ---------------------------------------------------------------------------- 4157 4158// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4159AudioFlinger::PlaybackThread::Track::Track( 4160 PlaybackThread *thread, 4161 const sp<Client>& client, 4162 audio_stream_type_t streamType, 4163 uint32_t sampleRate, 4164 audio_format_t format, 4165 uint32_t channelMask, 4166 int frameCount, 4167 const sp<IMemory>& sharedBuffer, 4168 int sessionId, 4169 IAudioFlinger::track_flags_t flags) 4170 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4171 mMute(false), 4172 mFillingUpStatus(FS_INVALID), 4173 // mRetryCount initialized later when needed 4174 mSharedBuffer(sharedBuffer), 4175 mStreamType(streamType), 4176 mName(-1), // see note below 4177 mMainBuffer(thread->mixBuffer()), 4178 mAuxBuffer(NULL), 4179 mAuxEffectId(0), mHasVolumeController(false), 4180 mPresentationCompleteFrames(0), 4181 mFlags(flags), 4182 mFastIndex(-1), 4183 mUnderrunCount(0), 4184 mCachedVolume(1.0) 4185{ 4186 if (mCblk != NULL) { 4187 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4188 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4189 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4190 if (flags & IAudioFlinger::TRACK_FAST) { 4191 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4192 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4193 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4194 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4195 // FIXME This is too eager. We allocate a fast track index before the 4196 // fast track becomes active. Since fast tracks are a scarce resource, 4197 // this means we are potentially denying other more important fast tracks from 4198 // being created. It would be better to allocate the index dynamically. 4199 mFastIndex = i; 4200 // Read the initial underruns because this field is never cleared by the fast mixer 4201 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4202 thread->mFastTrackAvailMask &= ~(1 << i); 4203 } 4204 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4205 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4206 if (mName < 0) { 4207 ALOGE("no more track names available"); 4208 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4209 // then we leak a fast track index. Should swap these two sections, or better yet 4210 // only allocate a normal mixer name for normal tracks. 4211 } 4212 } 4213 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4214} 4215 4216AudioFlinger::PlaybackThread::Track::~Track() 4217{ 4218 ALOGV("PlaybackThread::Track destructor"); 4219 sp<ThreadBase> thread = mThread.promote(); 4220 if (thread != 0) { 4221 Mutex::Autolock _l(thread->mLock); 4222 mState = TERMINATED; 4223 } 4224} 4225 4226void AudioFlinger::PlaybackThread::Track::destroy() 4227{ 4228 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4229 // by removing it from mTracks vector, so there is a risk that this Tracks's 4230 // destructor is called. As the destructor needs to lock mLock, 4231 // we must acquire a strong reference on this Track before locking mLock 4232 // here so that the destructor is called only when exiting this function. 4233 // On the other hand, as long as Track::destroy() is only called by 4234 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4235 // this Track with its member mTrack. 4236 sp<Track> keep(this); 4237 { // scope for mLock 4238 sp<ThreadBase> thread = mThread.promote(); 4239 if (thread != 0) { 4240 if (!isOutputTrack()) { 4241 if (mState == ACTIVE || mState == RESUMING) { 4242 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4243 4244#ifdef ADD_BATTERY_DATA 4245 // to track the speaker usage 4246 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4247#endif 4248 } 4249 AudioSystem::releaseOutput(thread->id()); 4250 } 4251 Mutex::Autolock _l(thread->mLock); 4252 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4253 playbackThread->destroyTrack_l(this); 4254 } 4255 } 4256} 4257 4258/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4259{ 4260 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4261 " Server User Main buf Aux Buf Flags FastUnder\n"); 4262} 4263 4264void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4265{ 4266 uint32_t vlr = mCblk->getVolumeLR(); 4267 if (isFastTrack()) { 4268 sprintf(buffer, " F %2d", mFastIndex); 4269 } else { 4270 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4271 } 4272 track_state state = mState; 4273 char stateChar; 4274 switch (state) { 4275 case IDLE: 4276 stateChar = 'I'; 4277 break; 4278 case TERMINATED: 4279 stateChar = 'T'; 4280 break; 4281 case STOPPING_1: 4282 stateChar = 's'; 4283 break; 4284 case STOPPING_2: 4285 stateChar = '5'; 4286 break; 4287 case STOPPED: 4288 stateChar = 'S'; 4289 break; 4290 case RESUMING: 4291 stateChar = 'R'; 4292 break; 4293 case ACTIVE: 4294 stateChar = 'A'; 4295 break; 4296 case PAUSING: 4297 stateChar = 'p'; 4298 break; 4299 case PAUSED: 4300 stateChar = 'P'; 4301 break; 4302 case FLUSHED: 4303 stateChar = 'F'; 4304 break; 4305 default: 4306 stateChar = '?'; 4307 break; 4308 } 4309 char nowInUnderrun; 4310 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4311 case UNDERRUN_FULL: 4312 nowInUnderrun = ' '; 4313 break; 4314 case UNDERRUN_PARTIAL: 4315 nowInUnderrun = '<'; 4316 break; 4317 case UNDERRUN_EMPTY: 4318 nowInUnderrun = '*'; 4319 break; 4320 default: 4321 nowInUnderrun = '?'; 4322 break; 4323 } 4324 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4325 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4326 (mClient == 0) ? getpid_cached : mClient->pid(), 4327 mStreamType, 4328 mFormat, 4329 mChannelMask, 4330 mSessionId, 4331 mFrameCount, 4332 mCblk->frameCount, 4333 stateChar, 4334 mMute, 4335 mFillingUpStatus, 4336 mCblk->sampleRate, 4337 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4338 20.0 * log10((vlr >> 16) / 4096.0), 4339 mCblk->server, 4340 mCblk->user, 4341 (int)mMainBuffer, 4342 (int)mAuxBuffer, 4343 mCblk->flags, 4344 mUnderrunCount, 4345 nowInUnderrun); 4346} 4347 4348// AudioBufferProvider interface 4349status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4350 AudioBufferProvider::Buffer* buffer, int64_t pts) 4351{ 4352 audio_track_cblk_t* cblk = this->cblk(); 4353 uint32_t framesReady; 4354 uint32_t framesReq = buffer->frameCount; 4355 4356 // Check if last stepServer failed, try to step now 4357 if (mStepServerFailed) { 4358 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4359 // Since the fast mixer is higher priority than client callback thread, 4360 // it does not result in priority inversion for client. 4361 // But a non-blocking solution would be preferable to avoid 4362 // fast mixer being unable to tryLock(), and 4363 // to avoid the extra context switches if the client wakes up, 4364 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4365 if (!step()) goto getNextBuffer_exit; 4366 ALOGV("stepServer recovered"); 4367 mStepServerFailed = false; 4368 } 4369 4370 // FIXME Same as above 4371 framesReady = cblk->framesReady(); 4372 4373 if (CC_LIKELY(framesReady)) { 4374 uint32_t s = cblk->server; 4375 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4376 4377 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4378 if (framesReq > framesReady) { 4379 framesReq = framesReady; 4380 } 4381 if (framesReq > bufferEnd - s) { 4382 framesReq = bufferEnd - s; 4383 } 4384 4385 buffer->raw = getBuffer(s, framesReq); 4386 if (buffer->raw == NULL) goto getNextBuffer_exit; 4387 4388 buffer->frameCount = framesReq; 4389 return NO_ERROR; 4390 } 4391 4392getNextBuffer_exit: 4393 buffer->raw = NULL; 4394 buffer->frameCount = 0; 4395 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4396 return NOT_ENOUGH_DATA; 4397} 4398 4399// Note that framesReady() takes a mutex on the control block using tryLock(). 4400// This could result in priority inversion if framesReady() is called by the normal mixer, 4401// as the normal mixer thread runs at lower 4402// priority than the client's callback thread: there is a short window within framesReady() 4403// during which the normal mixer could be preempted, and the client callback would block. 4404// Another problem can occur if framesReady() is called by the fast mixer: 4405// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4406// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4407size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4408 return mCblk->framesReady(); 4409} 4410 4411// Don't call for fast tracks; the framesReady() could result in priority inversion 4412bool AudioFlinger::PlaybackThread::Track::isReady() const { 4413 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4414 4415 if (framesReady() >= mCblk->frameCount || 4416 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4417 mFillingUpStatus = FS_FILLED; 4418 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4419 return true; 4420 } 4421 return false; 4422} 4423 4424status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4425 int triggerSession) 4426{ 4427 status_t status = NO_ERROR; 4428 ALOGV("start(%d), calling pid %d session %d", 4429 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4430 4431 sp<ThreadBase> thread = mThread.promote(); 4432 if (thread != 0) { 4433 Mutex::Autolock _l(thread->mLock); 4434 track_state state = mState; 4435 // here the track could be either new, or restarted 4436 // in both cases "unstop" the track 4437 if (mState == PAUSED) { 4438 mState = TrackBase::RESUMING; 4439 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4440 } else { 4441 mState = TrackBase::ACTIVE; 4442 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4443 } 4444 4445 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4446 thread->mLock.unlock(); 4447 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4448 thread->mLock.lock(); 4449 4450#ifdef ADD_BATTERY_DATA 4451 // to track the speaker usage 4452 if (status == NO_ERROR) { 4453 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4454 } 4455#endif 4456 } 4457 if (status == NO_ERROR) { 4458 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4459 playbackThread->addTrack_l(this); 4460 } else { 4461 mState = state; 4462 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4463 } 4464 } else { 4465 status = BAD_VALUE; 4466 } 4467 return status; 4468} 4469 4470void AudioFlinger::PlaybackThread::Track::stop() 4471{ 4472 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4473 sp<ThreadBase> thread = mThread.promote(); 4474 if (thread != 0) { 4475 Mutex::Autolock _l(thread->mLock); 4476 track_state state = mState; 4477 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4478 // If the track is not active (PAUSED and buffers full), flush buffers 4479 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4480 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4481 reset(); 4482 mState = STOPPED; 4483 } else if (!isFastTrack()) { 4484 mState = STOPPED; 4485 } else { 4486 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4487 // and then to STOPPED and reset() when presentation is complete 4488 mState = STOPPING_1; 4489 } 4490 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4491 } 4492 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4493 thread->mLock.unlock(); 4494 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4495 thread->mLock.lock(); 4496 4497#ifdef ADD_BATTERY_DATA 4498 // to track the speaker usage 4499 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4500#endif 4501 } 4502 } 4503} 4504 4505void AudioFlinger::PlaybackThread::Track::pause() 4506{ 4507 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4508 sp<ThreadBase> thread = mThread.promote(); 4509 if (thread != 0) { 4510 Mutex::Autolock _l(thread->mLock); 4511 if (mState == ACTIVE || mState == RESUMING) { 4512 mState = PAUSING; 4513 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4514 if (!isOutputTrack()) { 4515 thread->mLock.unlock(); 4516 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4517 thread->mLock.lock(); 4518 4519#ifdef ADD_BATTERY_DATA 4520 // to track the speaker usage 4521 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4522#endif 4523 } 4524 } 4525 } 4526} 4527 4528void AudioFlinger::PlaybackThread::Track::flush() 4529{ 4530 ALOGV("flush(%d)", mName); 4531 sp<ThreadBase> thread = mThread.promote(); 4532 if (thread != 0) { 4533 Mutex::Autolock _l(thread->mLock); 4534 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4535 mState != PAUSING) { 4536 return; 4537 } 4538 // No point remaining in PAUSED state after a flush => go to 4539 // FLUSHED state 4540 mState = FLUSHED; 4541 // do not reset the track if it is still in the process of being stopped or paused. 4542 // this will be done by prepareTracks_l() when the track is stopped. 4543 // prepareTracks_l() will see mState == FLUSHED, then 4544 // remove from active track list, reset(), and trigger presentation complete 4545 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4546 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4547 reset(); 4548 } 4549 } 4550} 4551 4552void AudioFlinger::PlaybackThread::Track::reset() 4553{ 4554 // Do not reset twice to avoid discarding data written just after a flush and before 4555 // the audioflinger thread detects the track is stopped. 4556 if (!mResetDone) { 4557 TrackBase::reset(); 4558 // Force underrun condition to avoid false underrun callback until first data is 4559 // written to buffer 4560 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4561 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4562 mFillingUpStatus = FS_FILLING; 4563 mResetDone = true; 4564 if (mState == FLUSHED) { 4565 mState = IDLE; 4566 } 4567 } 4568} 4569 4570void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4571{ 4572 mMute = muted; 4573} 4574 4575status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4576{ 4577 status_t status = DEAD_OBJECT; 4578 sp<ThreadBase> thread = mThread.promote(); 4579 if (thread != 0) { 4580 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4581 status = playbackThread->attachAuxEffect(this, EffectId); 4582 } 4583 return status; 4584} 4585 4586void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4587{ 4588 mAuxEffectId = EffectId; 4589 mAuxBuffer = buffer; 4590} 4591 4592bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4593 size_t audioHalFrames) 4594{ 4595 // a track is considered presented when the total number of frames written to audio HAL 4596 // corresponds to the number of frames written when presentationComplete() is called for the 4597 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4598 if (mPresentationCompleteFrames == 0) { 4599 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4600 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4601 mPresentationCompleteFrames, audioHalFrames); 4602 } 4603 if (framesWritten >= mPresentationCompleteFrames) { 4604 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4605 mSessionId, framesWritten); 4606 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4607 return true; 4608 } 4609 return false; 4610} 4611 4612void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4613{ 4614 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4615 if (mSyncEvents[i]->type() == type) { 4616 mSyncEvents[i]->trigger(); 4617 mSyncEvents.removeAt(i); 4618 i--; 4619 } 4620 } 4621} 4622 4623// implement VolumeBufferProvider interface 4624 4625uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4626{ 4627 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4628 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4629 uint32_t vlr = mCblk->getVolumeLR(); 4630 uint32_t vl = vlr & 0xFFFF; 4631 uint32_t vr = vlr >> 16; 4632 // track volumes come from shared memory, so can't be trusted and must be clamped 4633 if (vl > MAX_GAIN_INT) { 4634 vl = MAX_GAIN_INT; 4635 } 4636 if (vr > MAX_GAIN_INT) { 4637 vr = MAX_GAIN_INT; 4638 } 4639 // now apply the cached master volume and stream type volume; 4640 // this is trusted but lacks any synchronization or barrier so may be stale 4641 float v = mCachedVolume; 4642 vl *= v; 4643 vr *= v; 4644 // re-combine into U4.16 4645 vlr = (vr << 16) | (vl & 0xFFFF); 4646 // FIXME look at mute, pause, and stop flags 4647 return vlr; 4648} 4649 4650status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4651{ 4652 if (mState == TERMINATED || mState == PAUSED || 4653 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4654 (mState == STOPPED)))) { 4655 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4656 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4657 event->cancel(); 4658 return INVALID_OPERATION; 4659 } 4660 TrackBase::setSyncEvent(event); 4661 return NO_ERROR; 4662} 4663 4664// timed audio tracks 4665 4666sp<AudioFlinger::PlaybackThread::TimedTrack> 4667AudioFlinger::PlaybackThread::TimedTrack::create( 4668 PlaybackThread *thread, 4669 const sp<Client>& client, 4670 audio_stream_type_t streamType, 4671 uint32_t sampleRate, 4672 audio_format_t format, 4673 uint32_t channelMask, 4674 int frameCount, 4675 const sp<IMemory>& sharedBuffer, 4676 int sessionId) { 4677 if (!client->reserveTimedTrack()) 4678 return NULL; 4679 4680 return new TimedTrack( 4681 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4682 sharedBuffer, sessionId); 4683} 4684 4685AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4686 PlaybackThread *thread, 4687 const sp<Client>& client, 4688 audio_stream_type_t streamType, 4689 uint32_t sampleRate, 4690 audio_format_t format, 4691 uint32_t channelMask, 4692 int frameCount, 4693 const sp<IMemory>& sharedBuffer, 4694 int sessionId) 4695 : Track(thread, client, streamType, sampleRate, format, channelMask, 4696 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4697 mQueueHeadInFlight(false), 4698 mTrimQueueHeadOnRelease(false), 4699 mFramesPendingInQueue(0), 4700 mTimedSilenceBuffer(NULL), 4701 mTimedSilenceBufferSize(0), 4702 mTimedAudioOutputOnTime(false), 4703 mMediaTimeTransformValid(false) 4704{ 4705 LocalClock lc; 4706 mLocalTimeFreq = lc.getLocalFreq(); 4707 4708 mLocalTimeToSampleTransform.a_zero = 0; 4709 mLocalTimeToSampleTransform.b_zero = 0; 4710 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4711 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4712 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4713 &mLocalTimeToSampleTransform.a_to_b_denom); 4714 4715 mMediaTimeToSampleTransform.a_zero = 0; 4716 mMediaTimeToSampleTransform.b_zero = 0; 4717 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4718 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4719 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4720 &mMediaTimeToSampleTransform.a_to_b_denom); 4721} 4722 4723AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4724 mClient->releaseTimedTrack(); 4725 delete [] mTimedSilenceBuffer; 4726} 4727 4728status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4729 size_t size, sp<IMemory>* buffer) { 4730 4731 Mutex::Autolock _l(mTimedBufferQueueLock); 4732 4733 trimTimedBufferQueue_l(); 4734 4735 // lazily initialize the shared memory heap for timed buffers 4736 if (mTimedMemoryDealer == NULL) { 4737 const int kTimedBufferHeapSize = 512 << 10; 4738 4739 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4740 "AudioFlingerTimed"); 4741 if (mTimedMemoryDealer == NULL) 4742 return NO_MEMORY; 4743 } 4744 4745 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4746 if (newBuffer == NULL) { 4747 newBuffer = mTimedMemoryDealer->allocate(size); 4748 if (newBuffer == NULL) 4749 return NO_MEMORY; 4750 } 4751 4752 *buffer = newBuffer; 4753 return NO_ERROR; 4754} 4755 4756// caller must hold mTimedBufferQueueLock 4757void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4758 int64_t mediaTimeNow; 4759 { 4760 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4761 if (!mMediaTimeTransformValid) 4762 return; 4763 4764 int64_t targetTimeNow; 4765 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4766 ? mCCHelper.getCommonTime(&targetTimeNow) 4767 : mCCHelper.getLocalTime(&targetTimeNow); 4768 4769 if (OK != res) 4770 return; 4771 4772 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4773 &mediaTimeNow)) { 4774 return; 4775 } 4776 } 4777 4778 size_t trimEnd; 4779 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4780 int64_t bufEnd; 4781 4782 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4783 // We have a next buffer. Just use its PTS as the PTS of the frame 4784 // following the last frame in this buffer. If the stream is sparse 4785 // (ie, there are deliberate gaps left in the stream which should be 4786 // filled with silence by the TimedAudioTrack), then this can result 4787 // in one extra buffer being left un-trimmed when it could have 4788 // been. In general, this is not typical, and we would rather 4789 // optimized away the TS calculation below for the more common case 4790 // where PTSes are contiguous. 4791 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4792 } else { 4793 // We have no next buffer. Compute the PTS of the frame following 4794 // the last frame in this buffer by computing the duration of of 4795 // this frame in media time units and adding it to the PTS of the 4796 // buffer. 4797 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4798 / mCblk->frameSize; 4799 4800 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4801 &bufEnd)) { 4802 ALOGE("Failed to convert frame count of %lld to media time" 4803 " duration" " (scale factor %d/%u) in %s", 4804 frameCount, 4805 mMediaTimeToSampleTransform.a_to_b_numer, 4806 mMediaTimeToSampleTransform.a_to_b_denom, 4807 __PRETTY_FUNCTION__); 4808 break; 4809 } 4810 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4811 } 4812 4813 if (bufEnd > mediaTimeNow) 4814 break; 4815 4816 // Is the buffer we want to use in the middle of a mix operation right 4817 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4818 // from the mixer which should be coming back shortly. 4819 if (!trimEnd && mQueueHeadInFlight) { 4820 mTrimQueueHeadOnRelease = true; 4821 } 4822 } 4823 4824 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4825 if (trimStart < trimEnd) { 4826 // Update the bookkeeping for framesReady() 4827 for (size_t i = trimStart; i < trimEnd; ++i) { 4828 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4829 } 4830 4831 // Now actually remove the buffers from the queue. 4832 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4833 } 4834} 4835 4836void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4837 const char* logTag) { 4838 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4839 "%s called (reason \"%s\"), but timed buffer queue has no" 4840 " elements to trim.", __FUNCTION__, logTag); 4841 4842 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4843 mTimedBufferQueue.removeAt(0); 4844} 4845 4846void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4847 const TimedBuffer& buf, 4848 const char* logTag) { 4849 uint32_t bufBytes = buf.buffer()->size(); 4850 uint32_t consumedAlready = buf.position(); 4851 4852 ALOG_ASSERT(consumedAlready <= bufBytes, 4853 "Bad bookkeeping while updating frames pending. Timed buffer is" 4854 " only %u bytes long, but claims to have consumed %u" 4855 " bytes. (update reason: \"%s\")", 4856 bufBytes, consumedAlready, logTag); 4857 4858 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4859 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4860 "Bad bookkeeping while updating frames pending. Should have at" 4861 " least %u queued frames, but we think we have only %u. (update" 4862 " reason: \"%s\")", 4863 bufFrames, mFramesPendingInQueue, logTag); 4864 4865 mFramesPendingInQueue -= bufFrames; 4866} 4867 4868status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4869 const sp<IMemory>& buffer, int64_t pts) { 4870 4871 { 4872 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4873 if (!mMediaTimeTransformValid) 4874 return INVALID_OPERATION; 4875 } 4876 4877 Mutex::Autolock _l(mTimedBufferQueueLock); 4878 4879 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4880 mFramesPendingInQueue += bufFrames; 4881 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4882 4883 return NO_ERROR; 4884} 4885 4886status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4887 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4888 4889 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4890 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4891 target); 4892 4893 if (!(target == TimedAudioTrack::LOCAL_TIME || 4894 target == TimedAudioTrack::COMMON_TIME)) { 4895 return BAD_VALUE; 4896 } 4897 4898 Mutex::Autolock lock(mMediaTimeTransformLock); 4899 mMediaTimeTransform = xform; 4900 mMediaTimeTransformTarget = target; 4901 mMediaTimeTransformValid = true; 4902 4903 return NO_ERROR; 4904} 4905 4906#define min(a, b) ((a) < (b) ? (a) : (b)) 4907 4908// implementation of getNextBuffer for tracks whose buffers have timestamps 4909status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4910 AudioBufferProvider::Buffer* buffer, int64_t pts) 4911{ 4912 if (pts == AudioBufferProvider::kInvalidPTS) { 4913 buffer->raw = 0; 4914 buffer->frameCount = 0; 4915 mTimedAudioOutputOnTime = false; 4916 return INVALID_OPERATION; 4917 } 4918 4919 Mutex::Autolock _l(mTimedBufferQueueLock); 4920 4921 ALOG_ASSERT(!mQueueHeadInFlight, 4922 "getNextBuffer called without releaseBuffer!"); 4923 4924 while (true) { 4925 4926 // if we have no timed buffers, then fail 4927 if (mTimedBufferQueue.isEmpty()) { 4928 buffer->raw = 0; 4929 buffer->frameCount = 0; 4930 return NOT_ENOUGH_DATA; 4931 } 4932 4933 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4934 4935 // calculate the PTS of the head of the timed buffer queue expressed in 4936 // local time 4937 int64_t headLocalPTS; 4938 { 4939 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4940 4941 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4942 4943 if (mMediaTimeTransform.a_to_b_denom == 0) { 4944 // the transform represents a pause, so yield silence 4945 timedYieldSilence_l(buffer->frameCount, buffer); 4946 return NO_ERROR; 4947 } 4948 4949 int64_t transformedPTS; 4950 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4951 &transformedPTS)) { 4952 // the transform failed. this shouldn't happen, but if it does 4953 // then just drop this buffer 4954 ALOGW("timedGetNextBuffer transform failed"); 4955 buffer->raw = 0; 4956 buffer->frameCount = 0; 4957 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4958 return NO_ERROR; 4959 } 4960 4961 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4962 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4963 &headLocalPTS)) { 4964 buffer->raw = 0; 4965 buffer->frameCount = 0; 4966 return INVALID_OPERATION; 4967 } 4968 } else { 4969 headLocalPTS = transformedPTS; 4970 } 4971 } 4972 4973 // adjust the head buffer's PTS to reflect the portion of the head buffer 4974 // that has already been consumed 4975 int64_t effectivePTS = headLocalPTS + 4976 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4977 4978 // Calculate the delta in samples between the head of the input buffer 4979 // queue and the start of the next output buffer that will be written. 4980 // If the transformation fails because of over or underflow, it means 4981 // that the sample's position in the output stream is so far out of 4982 // whack that it should just be dropped. 4983 int64_t sampleDelta; 4984 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4985 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4986 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4987 " mix"); 4988 continue; 4989 } 4990 if (!mLocalTimeToSampleTransform.doForwardTransform( 4991 (effectivePTS - pts) << 32, &sampleDelta)) { 4992 ALOGV("*** too late during sample rate transform: dropped buffer"); 4993 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4994 continue; 4995 } 4996 4997 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4998 " sampleDelta=[%d.%08x]", 4999 head.pts(), head.position(), pts, 5000 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5001 + (sampleDelta >> 32)), 5002 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5003 5004 // if the delta between the ideal placement for the next input sample and 5005 // the current output position is within this threshold, then we will 5006 // concatenate the next input samples to the previous output 5007 const int64_t kSampleContinuityThreshold = 5008 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5009 5010 // if this is the first buffer of audio that we're emitting from this track 5011 // then it should be almost exactly on time. 5012 const int64_t kSampleStartupThreshold = 1LL << 32; 5013 5014 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5015 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5016 // the next input is close enough to being on time, so concatenate it 5017 // with the last output 5018 timedYieldSamples_l(buffer); 5019 5020 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5021 head.position(), buffer->frameCount); 5022 return NO_ERROR; 5023 } 5024 5025 // Looks like our output is not on time. Reset our on timed status. 5026 // Next time we mix samples from our input queue, then should be within 5027 // the StartupThreshold. 5028 mTimedAudioOutputOnTime = false; 5029 if (sampleDelta > 0) { 5030 // the gap between the current output position and the proper start of 5031 // the next input sample is too big, so fill it with silence 5032 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5033 5034 timedYieldSilence_l(framesUntilNextInput, buffer); 5035 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5036 return NO_ERROR; 5037 } else { 5038 // the next input sample is late 5039 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5040 size_t onTimeSamplePosition = 5041 head.position() + lateFrames * mCblk->frameSize; 5042 5043 if (onTimeSamplePosition > head.buffer()->size()) { 5044 // all the remaining samples in the head are too late, so 5045 // drop it and move on 5046 ALOGV("*** too late: dropped buffer"); 5047 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5048 continue; 5049 } else { 5050 // skip over the late samples 5051 head.setPosition(onTimeSamplePosition); 5052 5053 // yield the available samples 5054 timedYieldSamples_l(buffer); 5055 5056 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5057 return NO_ERROR; 5058 } 5059 } 5060 } 5061} 5062 5063// Yield samples from the timed buffer queue head up to the given output 5064// buffer's capacity. 5065// 5066// Caller must hold mTimedBufferQueueLock 5067void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5068 AudioBufferProvider::Buffer* buffer) { 5069 5070 const TimedBuffer& head = mTimedBufferQueue[0]; 5071 5072 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5073 head.position()); 5074 5075 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5076 mCblk->frameSize); 5077 size_t framesRequested = buffer->frameCount; 5078 buffer->frameCount = min(framesLeftInHead, framesRequested); 5079 5080 mQueueHeadInFlight = true; 5081 mTimedAudioOutputOnTime = true; 5082} 5083 5084// Yield samples of silence up to the given output buffer's capacity 5085// 5086// Caller must hold mTimedBufferQueueLock 5087void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5088 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5089 5090 // lazily allocate a buffer filled with silence 5091 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5092 delete [] mTimedSilenceBuffer; 5093 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5094 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5095 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5096 } 5097 5098 buffer->raw = mTimedSilenceBuffer; 5099 size_t framesRequested = buffer->frameCount; 5100 buffer->frameCount = min(numFrames, framesRequested); 5101 5102 mTimedAudioOutputOnTime = false; 5103} 5104 5105// AudioBufferProvider interface 5106void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5107 AudioBufferProvider::Buffer* buffer) { 5108 5109 Mutex::Autolock _l(mTimedBufferQueueLock); 5110 5111 // If the buffer which was just released is part of the buffer at the head 5112 // of the queue, be sure to update the amt of the buffer which has been 5113 // consumed. If the buffer being returned is not part of the head of the 5114 // queue, its either because the buffer is part of the silence buffer, or 5115 // because the head of the timed queue was trimmed after the mixer called 5116 // getNextBuffer but before the mixer called releaseBuffer. 5117 if (buffer->raw == mTimedSilenceBuffer) { 5118 ALOG_ASSERT(!mQueueHeadInFlight, 5119 "Queue head in flight during release of silence buffer!"); 5120 goto done; 5121 } 5122 5123 ALOG_ASSERT(mQueueHeadInFlight, 5124 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5125 " head in flight."); 5126 5127 if (mTimedBufferQueue.size()) { 5128 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5129 5130 void* start = head.buffer()->pointer(); 5131 void* end = reinterpret_cast<void*>( 5132 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5133 + head.buffer()->size()); 5134 5135 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5136 "released buffer not within the head of the timed buffer" 5137 " queue; qHead = [%p, %p], released buffer = %p", 5138 start, end, buffer->raw); 5139 5140 head.setPosition(head.position() + 5141 (buffer->frameCount * mCblk->frameSize)); 5142 mQueueHeadInFlight = false; 5143 5144 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5145 "Bad bookkeeping during releaseBuffer! Should have at" 5146 " least %u queued frames, but we think we have only %u", 5147 buffer->frameCount, mFramesPendingInQueue); 5148 5149 mFramesPendingInQueue -= buffer->frameCount; 5150 5151 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5152 || mTrimQueueHeadOnRelease) { 5153 trimTimedBufferQueueHead_l("releaseBuffer"); 5154 mTrimQueueHeadOnRelease = false; 5155 } 5156 } else { 5157 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5158 " buffers in the timed buffer queue"); 5159 } 5160 5161done: 5162 buffer->raw = 0; 5163 buffer->frameCount = 0; 5164} 5165 5166size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5167 Mutex::Autolock _l(mTimedBufferQueueLock); 5168 return mFramesPendingInQueue; 5169} 5170 5171AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5172 : mPTS(0), mPosition(0) {} 5173 5174AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5175 const sp<IMemory>& buffer, int64_t pts) 5176 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5177 5178// ---------------------------------------------------------------------------- 5179 5180// RecordTrack constructor must be called with AudioFlinger::mLock held 5181AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5182 RecordThread *thread, 5183 const sp<Client>& client, 5184 uint32_t sampleRate, 5185 audio_format_t format, 5186 uint32_t channelMask, 5187 int frameCount, 5188 int sessionId) 5189 : TrackBase(thread, client, sampleRate, format, 5190 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5191 mOverflow(false) 5192{ 5193 if (mCblk != NULL) { 5194 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5195 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5196 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5197 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5198 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5199 } else { 5200 mCblk->frameSize = sizeof(int8_t); 5201 } 5202 } 5203} 5204 5205AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5206{ 5207 sp<ThreadBase> thread = mThread.promote(); 5208 if (thread != 0) { 5209 AudioSystem::releaseInput(thread->id()); 5210 } 5211} 5212 5213// AudioBufferProvider interface 5214status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5215{ 5216 audio_track_cblk_t* cblk = this->cblk(); 5217 uint32_t framesAvail; 5218 uint32_t framesReq = buffer->frameCount; 5219 5220 // Check if last stepServer failed, try to step now 5221 if (mStepServerFailed) { 5222 if (!step()) goto getNextBuffer_exit; 5223 ALOGV("stepServer recovered"); 5224 mStepServerFailed = false; 5225 } 5226 5227 framesAvail = cblk->framesAvailable_l(); 5228 5229 if (CC_LIKELY(framesAvail)) { 5230 uint32_t s = cblk->server; 5231 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5232 5233 if (framesReq > framesAvail) { 5234 framesReq = framesAvail; 5235 } 5236 if (framesReq > bufferEnd - s) { 5237 framesReq = bufferEnd - s; 5238 } 5239 5240 buffer->raw = getBuffer(s, framesReq); 5241 if (buffer->raw == NULL) goto getNextBuffer_exit; 5242 5243 buffer->frameCount = framesReq; 5244 return NO_ERROR; 5245 } 5246 5247getNextBuffer_exit: 5248 buffer->raw = NULL; 5249 buffer->frameCount = 0; 5250 return NOT_ENOUGH_DATA; 5251} 5252 5253status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5254 int triggerSession) 5255{ 5256 sp<ThreadBase> thread = mThread.promote(); 5257 if (thread != 0) { 5258 RecordThread *recordThread = (RecordThread *)thread.get(); 5259 return recordThread->start(this, event, triggerSession); 5260 } else { 5261 return BAD_VALUE; 5262 } 5263} 5264 5265void AudioFlinger::RecordThread::RecordTrack::stop() 5266{ 5267 sp<ThreadBase> thread = mThread.promote(); 5268 if (thread != 0) { 5269 RecordThread *recordThread = (RecordThread *)thread.get(); 5270 recordThread->stop(this); 5271 TrackBase::reset(); 5272 // Force overrun condition to avoid false overrun callback until first data is 5273 // read from buffer 5274 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5275 } 5276} 5277 5278void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5279{ 5280 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5281 (mClient == 0) ? getpid_cached : mClient->pid(), 5282 mFormat, 5283 mChannelMask, 5284 mSessionId, 5285 mFrameCount, 5286 mState, 5287 mCblk->sampleRate, 5288 mCblk->server, 5289 mCblk->user); 5290} 5291 5292 5293// ---------------------------------------------------------------------------- 5294 5295AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5296 PlaybackThread *playbackThread, 5297 DuplicatingThread *sourceThread, 5298 uint32_t sampleRate, 5299 audio_format_t format, 5300 uint32_t channelMask, 5301 int frameCount) 5302 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5303 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5304 mActive(false), mSourceThread(sourceThread) 5305{ 5306 5307 if (mCblk != NULL) { 5308 mCblk->flags |= CBLK_DIRECTION_OUT; 5309 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5310 mOutBuffer.frameCount = 0; 5311 playbackThread->mTracks.add(this); 5312 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5313 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5314 mCblk, mBuffer, mCblk->buffers, 5315 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5316 } else { 5317 ALOGW("Error creating output track on thread %p", playbackThread); 5318 } 5319} 5320 5321AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5322{ 5323 clearBufferQueue(); 5324} 5325 5326status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5327 int triggerSession) 5328{ 5329 status_t status = Track::start(event, triggerSession); 5330 if (status != NO_ERROR) { 5331 return status; 5332 } 5333 5334 mActive = true; 5335 mRetryCount = 127; 5336 return status; 5337} 5338 5339void AudioFlinger::PlaybackThread::OutputTrack::stop() 5340{ 5341 Track::stop(); 5342 clearBufferQueue(); 5343 mOutBuffer.frameCount = 0; 5344 mActive = false; 5345} 5346 5347bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5348{ 5349 Buffer *pInBuffer; 5350 Buffer inBuffer; 5351 uint32_t channelCount = mChannelCount; 5352 bool outputBufferFull = false; 5353 inBuffer.frameCount = frames; 5354 inBuffer.i16 = data; 5355 5356 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5357 5358 if (!mActive && frames != 0) { 5359 start(); 5360 sp<ThreadBase> thread = mThread.promote(); 5361 if (thread != 0) { 5362 MixerThread *mixerThread = (MixerThread *)thread.get(); 5363 if (mCblk->frameCount > frames){ 5364 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5365 uint32_t startFrames = (mCblk->frameCount - frames); 5366 pInBuffer = new Buffer; 5367 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5368 pInBuffer->frameCount = startFrames; 5369 pInBuffer->i16 = pInBuffer->mBuffer; 5370 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5371 mBufferQueue.add(pInBuffer); 5372 } else { 5373 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5374 } 5375 } 5376 } 5377 } 5378 5379 while (waitTimeLeftMs) { 5380 // First write pending buffers, then new data 5381 if (mBufferQueue.size()) { 5382 pInBuffer = mBufferQueue.itemAt(0); 5383 } else { 5384 pInBuffer = &inBuffer; 5385 } 5386 5387 if (pInBuffer->frameCount == 0) { 5388 break; 5389 } 5390 5391 if (mOutBuffer.frameCount == 0) { 5392 mOutBuffer.frameCount = pInBuffer->frameCount; 5393 nsecs_t startTime = systemTime(); 5394 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5395 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5396 outputBufferFull = true; 5397 break; 5398 } 5399 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5400 if (waitTimeLeftMs >= waitTimeMs) { 5401 waitTimeLeftMs -= waitTimeMs; 5402 } else { 5403 waitTimeLeftMs = 0; 5404 } 5405 } 5406 5407 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5408 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5409 mCblk->stepUser(outFrames); 5410 pInBuffer->frameCount -= outFrames; 5411 pInBuffer->i16 += outFrames * channelCount; 5412 mOutBuffer.frameCount -= outFrames; 5413 mOutBuffer.i16 += outFrames * channelCount; 5414 5415 if (pInBuffer->frameCount == 0) { 5416 if (mBufferQueue.size()) { 5417 mBufferQueue.removeAt(0); 5418 delete [] pInBuffer->mBuffer; 5419 delete pInBuffer; 5420 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5421 } else { 5422 break; 5423 } 5424 } 5425 } 5426 5427 // If we could not write all frames, allocate a buffer and queue it for next time. 5428 if (inBuffer.frameCount) { 5429 sp<ThreadBase> thread = mThread.promote(); 5430 if (thread != 0 && !thread->standby()) { 5431 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5432 pInBuffer = new Buffer; 5433 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5434 pInBuffer->frameCount = inBuffer.frameCount; 5435 pInBuffer->i16 = pInBuffer->mBuffer; 5436 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5437 mBufferQueue.add(pInBuffer); 5438 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5439 } else { 5440 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5441 } 5442 } 5443 } 5444 5445 // Calling write() with a 0 length buffer, means that no more data will be written: 5446 // If no more buffers are pending, fill output track buffer to make sure it is started 5447 // by output mixer. 5448 if (frames == 0 && mBufferQueue.size() == 0) { 5449 if (mCblk->user < mCblk->frameCount) { 5450 frames = mCblk->frameCount - mCblk->user; 5451 pInBuffer = new Buffer; 5452 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5453 pInBuffer->frameCount = frames; 5454 pInBuffer->i16 = pInBuffer->mBuffer; 5455 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5456 mBufferQueue.add(pInBuffer); 5457 } else if (mActive) { 5458 stop(); 5459 } 5460 } 5461 5462 return outputBufferFull; 5463} 5464 5465status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5466{ 5467 int active; 5468 status_t result; 5469 audio_track_cblk_t* cblk = mCblk; 5470 uint32_t framesReq = buffer->frameCount; 5471 5472// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5473 buffer->frameCount = 0; 5474 5475 uint32_t framesAvail = cblk->framesAvailable(); 5476 5477 5478 if (framesAvail == 0) { 5479 Mutex::Autolock _l(cblk->lock); 5480 goto start_loop_here; 5481 while (framesAvail == 0) { 5482 active = mActive; 5483 if (CC_UNLIKELY(!active)) { 5484 ALOGV("Not active and NO_MORE_BUFFERS"); 5485 return NO_MORE_BUFFERS; 5486 } 5487 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5488 if (result != NO_ERROR) { 5489 return NO_MORE_BUFFERS; 5490 } 5491 // read the server count again 5492 start_loop_here: 5493 framesAvail = cblk->framesAvailable_l(); 5494 } 5495 } 5496 5497// if (framesAvail < framesReq) { 5498// return NO_MORE_BUFFERS; 5499// } 5500 5501 if (framesReq > framesAvail) { 5502 framesReq = framesAvail; 5503 } 5504 5505 uint32_t u = cblk->user; 5506 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5507 5508 if (framesReq > bufferEnd - u) { 5509 framesReq = bufferEnd - u; 5510 } 5511 5512 buffer->frameCount = framesReq; 5513 buffer->raw = (void *)cblk->buffer(u); 5514 return NO_ERROR; 5515} 5516 5517 5518void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5519{ 5520 size_t size = mBufferQueue.size(); 5521 5522 for (size_t i = 0; i < size; i++) { 5523 Buffer *pBuffer = mBufferQueue.itemAt(i); 5524 delete [] pBuffer->mBuffer; 5525 delete pBuffer; 5526 } 5527 mBufferQueue.clear(); 5528} 5529 5530// ---------------------------------------------------------------------------- 5531 5532AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5533 : RefBase(), 5534 mAudioFlinger(audioFlinger), 5535 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5536 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5537 mPid(pid), 5538 mTimedTrackCount(0) 5539{ 5540 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5541} 5542 5543// Client destructor must be called with AudioFlinger::mLock held 5544AudioFlinger::Client::~Client() 5545{ 5546 mAudioFlinger->removeClient_l(mPid); 5547} 5548 5549sp<MemoryDealer> AudioFlinger::Client::heap() const 5550{ 5551 return mMemoryDealer; 5552} 5553 5554// Reserve one of the limited slots for a timed audio track associated 5555// with this client 5556bool AudioFlinger::Client::reserveTimedTrack() 5557{ 5558 const int kMaxTimedTracksPerClient = 4; 5559 5560 Mutex::Autolock _l(mTimedTrackLock); 5561 5562 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5563 ALOGW("can not create timed track - pid %d has exceeded the limit", 5564 mPid); 5565 return false; 5566 } 5567 5568 mTimedTrackCount++; 5569 return true; 5570} 5571 5572// Release a slot for a timed audio track 5573void AudioFlinger::Client::releaseTimedTrack() 5574{ 5575 Mutex::Autolock _l(mTimedTrackLock); 5576 mTimedTrackCount--; 5577} 5578 5579// ---------------------------------------------------------------------------- 5580 5581AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5582 const sp<IAudioFlingerClient>& client, 5583 pid_t pid) 5584 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5585{ 5586} 5587 5588AudioFlinger::NotificationClient::~NotificationClient() 5589{ 5590} 5591 5592void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5593{ 5594 sp<NotificationClient> keep(this); 5595 mAudioFlinger->removeNotificationClient(mPid); 5596} 5597 5598// ---------------------------------------------------------------------------- 5599 5600AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5601 : BnAudioTrack(), 5602 mTrack(track) 5603{ 5604} 5605 5606AudioFlinger::TrackHandle::~TrackHandle() { 5607 // just stop the track on deletion, associated resources 5608 // will be freed from the main thread once all pending buffers have 5609 // been played. Unless it's not in the active track list, in which 5610 // case we free everything now... 5611 mTrack->destroy(); 5612} 5613 5614sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5615 return mTrack->getCblk(); 5616} 5617 5618status_t AudioFlinger::TrackHandle::start() { 5619 return mTrack->start(); 5620} 5621 5622void AudioFlinger::TrackHandle::stop() { 5623 mTrack->stop(); 5624} 5625 5626void AudioFlinger::TrackHandle::flush() { 5627 mTrack->flush(); 5628} 5629 5630void AudioFlinger::TrackHandle::mute(bool e) { 5631 mTrack->mute(e); 5632} 5633 5634void AudioFlinger::TrackHandle::pause() { 5635 mTrack->pause(); 5636} 5637 5638status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5639{ 5640 return mTrack->attachAuxEffect(EffectId); 5641} 5642 5643status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5644 sp<IMemory>* buffer) { 5645 if (!mTrack->isTimedTrack()) 5646 return INVALID_OPERATION; 5647 5648 PlaybackThread::TimedTrack* tt = 5649 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5650 return tt->allocateTimedBuffer(size, buffer); 5651} 5652 5653status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5654 int64_t pts) { 5655 if (!mTrack->isTimedTrack()) 5656 return INVALID_OPERATION; 5657 5658 PlaybackThread::TimedTrack* tt = 5659 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5660 return tt->queueTimedBuffer(buffer, pts); 5661} 5662 5663status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5664 const LinearTransform& xform, int target) { 5665 5666 if (!mTrack->isTimedTrack()) 5667 return INVALID_OPERATION; 5668 5669 PlaybackThread::TimedTrack* tt = 5670 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5671 return tt->setMediaTimeTransform( 5672 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5673} 5674 5675status_t AudioFlinger::TrackHandle::onTransact( 5676 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5677{ 5678 return BnAudioTrack::onTransact(code, data, reply, flags); 5679} 5680 5681// ---------------------------------------------------------------------------- 5682 5683sp<IAudioRecord> AudioFlinger::openRecord( 5684 pid_t pid, 5685 audio_io_handle_t input, 5686 uint32_t sampleRate, 5687 audio_format_t format, 5688 uint32_t channelMask, 5689 int frameCount, 5690 IAudioFlinger::track_flags_t flags, 5691 int *sessionId, 5692 status_t *status) 5693{ 5694 sp<RecordThread::RecordTrack> recordTrack; 5695 sp<RecordHandle> recordHandle; 5696 sp<Client> client; 5697 status_t lStatus; 5698 RecordThread *thread; 5699 size_t inFrameCount; 5700 int lSessionId; 5701 5702 // check calling permissions 5703 if (!recordingAllowed()) { 5704 lStatus = PERMISSION_DENIED; 5705 goto Exit; 5706 } 5707 5708 // add client to list 5709 { // scope for mLock 5710 Mutex::Autolock _l(mLock); 5711 thread = checkRecordThread_l(input); 5712 if (thread == NULL) { 5713 lStatus = BAD_VALUE; 5714 goto Exit; 5715 } 5716 5717 client = registerPid_l(pid); 5718 5719 // If no audio session id is provided, create one here 5720 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5721 lSessionId = *sessionId; 5722 } else { 5723 lSessionId = nextUniqueId(); 5724 if (sessionId != NULL) { 5725 *sessionId = lSessionId; 5726 } 5727 } 5728 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5729 recordTrack = thread->createRecordTrack_l(client, 5730 sampleRate, 5731 format, 5732 channelMask, 5733 frameCount, 5734 lSessionId, 5735 &lStatus); 5736 } 5737 if (lStatus != NO_ERROR) { 5738 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5739 // destructor is called by the TrackBase destructor with mLock held 5740 client.clear(); 5741 recordTrack.clear(); 5742 goto Exit; 5743 } 5744 5745 // return to handle to client 5746 recordHandle = new RecordHandle(recordTrack); 5747 lStatus = NO_ERROR; 5748 5749Exit: 5750 if (status) { 5751 *status = lStatus; 5752 } 5753 return recordHandle; 5754} 5755 5756// ---------------------------------------------------------------------------- 5757 5758AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5759 : BnAudioRecord(), 5760 mRecordTrack(recordTrack) 5761{ 5762} 5763 5764AudioFlinger::RecordHandle::~RecordHandle() { 5765 stop(); 5766} 5767 5768sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5769 return mRecordTrack->getCblk(); 5770} 5771 5772status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5773 ALOGV("RecordHandle::start()"); 5774 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5775} 5776 5777void AudioFlinger::RecordHandle::stop() { 5778 ALOGV("RecordHandle::stop()"); 5779 mRecordTrack->stop(); 5780} 5781 5782status_t AudioFlinger::RecordHandle::onTransact( 5783 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5784{ 5785 return BnAudioRecord::onTransact(code, data, reply, flags); 5786} 5787 5788// ---------------------------------------------------------------------------- 5789 5790AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5791 AudioStreamIn *input, 5792 uint32_t sampleRate, 5793 uint32_t channels, 5794 audio_io_handle_t id, 5795 uint32_t device) : 5796 ThreadBase(audioFlinger, id, device, RECORD), 5797 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5798 // mRsmpInIndex and mInputBytes set by readInputParameters() 5799 mReqChannelCount(popcount(channels)), 5800 mReqSampleRate(sampleRate) 5801 // mBytesRead is only meaningful while active, and so is cleared in start() 5802 // (but might be better to also clear here for dump?) 5803{ 5804 snprintf(mName, kNameLength, "AudioIn_%X", id); 5805 5806 readInputParameters(); 5807} 5808 5809 5810AudioFlinger::RecordThread::~RecordThread() 5811{ 5812 delete[] mRsmpInBuffer; 5813 delete mResampler; 5814 delete[] mRsmpOutBuffer; 5815} 5816 5817void AudioFlinger::RecordThread::onFirstRef() 5818{ 5819 run(mName, PRIORITY_URGENT_AUDIO); 5820} 5821 5822status_t AudioFlinger::RecordThread::readyToRun() 5823{ 5824 status_t status = initCheck(); 5825 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5826 return status; 5827} 5828 5829bool AudioFlinger::RecordThread::threadLoop() 5830{ 5831 AudioBufferProvider::Buffer buffer; 5832 sp<RecordTrack> activeTrack; 5833 Vector< sp<EffectChain> > effectChains; 5834 5835 nsecs_t lastWarning = 0; 5836 5837 acquireWakeLock(); 5838 5839 // start recording 5840 while (!exitPending()) { 5841 5842 processConfigEvents(); 5843 5844 { // scope for mLock 5845 Mutex::Autolock _l(mLock); 5846 checkForNewParameters_l(); 5847 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5848 if (!mStandby) { 5849 mInput->stream->common.standby(&mInput->stream->common); 5850 mStandby = true; 5851 } 5852 5853 if (exitPending()) break; 5854 5855 releaseWakeLock_l(); 5856 ALOGV("RecordThread: loop stopping"); 5857 // go to sleep 5858 mWaitWorkCV.wait(mLock); 5859 ALOGV("RecordThread: loop starting"); 5860 acquireWakeLock_l(); 5861 continue; 5862 } 5863 if (mActiveTrack != 0) { 5864 if (mActiveTrack->mState == TrackBase::PAUSING) { 5865 if (!mStandby) { 5866 mInput->stream->common.standby(&mInput->stream->common); 5867 mStandby = true; 5868 } 5869 mActiveTrack.clear(); 5870 mStartStopCond.broadcast(); 5871 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5872 if (mReqChannelCount != mActiveTrack->channelCount()) { 5873 mActiveTrack.clear(); 5874 mStartStopCond.broadcast(); 5875 } else if (mBytesRead != 0) { 5876 // record start succeeds only if first read from audio input 5877 // succeeds 5878 if (mBytesRead > 0) { 5879 mActiveTrack->mState = TrackBase::ACTIVE; 5880 } else { 5881 mActiveTrack.clear(); 5882 } 5883 mStartStopCond.broadcast(); 5884 } 5885 mStandby = false; 5886 } 5887 } 5888 lockEffectChains_l(effectChains); 5889 } 5890 5891 if (mActiveTrack != 0) { 5892 if (mActiveTrack->mState != TrackBase::ACTIVE && 5893 mActiveTrack->mState != TrackBase::RESUMING) { 5894 unlockEffectChains(effectChains); 5895 usleep(kRecordThreadSleepUs); 5896 continue; 5897 } 5898 for (size_t i = 0; i < effectChains.size(); i ++) { 5899 effectChains[i]->process_l(); 5900 } 5901 5902 buffer.frameCount = mFrameCount; 5903 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5904 size_t framesOut = buffer.frameCount; 5905 if (mResampler == NULL) { 5906 // no resampling 5907 while (framesOut) { 5908 size_t framesIn = mFrameCount - mRsmpInIndex; 5909 if (framesIn) { 5910 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5911 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5912 if (framesIn > framesOut) 5913 framesIn = framesOut; 5914 mRsmpInIndex += framesIn; 5915 framesOut -= framesIn; 5916 if ((int)mChannelCount == mReqChannelCount || 5917 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5918 memcpy(dst, src, framesIn * mFrameSize); 5919 } else { 5920 int16_t *src16 = (int16_t *)src; 5921 int16_t *dst16 = (int16_t *)dst; 5922 if (mChannelCount == 1) { 5923 while (framesIn--) { 5924 *dst16++ = *src16; 5925 *dst16++ = *src16++; 5926 } 5927 } else { 5928 while (framesIn--) { 5929 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5930 src16 += 2; 5931 } 5932 } 5933 } 5934 } 5935 if (framesOut && mFrameCount == mRsmpInIndex) { 5936 if (framesOut == mFrameCount && 5937 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5938 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5939 framesOut = 0; 5940 } else { 5941 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5942 mRsmpInIndex = 0; 5943 } 5944 if (mBytesRead < 0) { 5945 ALOGE("Error reading audio input"); 5946 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5947 // Force input into standby so that it tries to 5948 // recover at next read attempt 5949 mInput->stream->common.standby(&mInput->stream->common); 5950 usleep(kRecordThreadSleepUs); 5951 } 5952 mRsmpInIndex = mFrameCount; 5953 framesOut = 0; 5954 buffer.frameCount = 0; 5955 } 5956 } 5957 } 5958 } else { 5959 // resampling 5960 5961 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5962 // alter output frame count as if we were expecting stereo samples 5963 if (mChannelCount == 1 && mReqChannelCount == 1) { 5964 framesOut >>= 1; 5965 } 5966 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5967 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5968 // are 32 bit aligned which should be always true. 5969 if (mChannelCount == 2 && mReqChannelCount == 1) { 5970 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5971 // the resampler always outputs stereo samples: do post stereo to mono conversion 5972 int16_t *src = (int16_t *)mRsmpOutBuffer; 5973 int16_t *dst = buffer.i16; 5974 while (framesOut--) { 5975 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5976 src += 2; 5977 } 5978 } else { 5979 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5980 } 5981 5982 } 5983 if (mFramestoDrop == 0) { 5984 mActiveTrack->releaseBuffer(&buffer); 5985 } else { 5986 if (mFramestoDrop > 0) { 5987 mFramestoDrop -= buffer.frameCount; 5988 if (mFramestoDrop <= 0) { 5989 clearSyncStartEvent(); 5990 } 5991 } else { 5992 mFramestoDrop += buffer.frameCount; 5993 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 5994 mSyncStartEvent->isCancelled()) { 5995 ALOGW("Synced record %s, session %d, trigger session %d", 5996 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 5997 mActiveTrack->sessionId(), 5998 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 5999 clearSyncStartEvent(); 6000 } 6001 } 6002 } 6003 mActiveTrack->overflow(); 6004 } 6005 // client isn't retrieving buffers fast enough 6006 else { 6007 if (!mActiveTrack->setOverflow()) { 6008 nsecs_t now = systemTime(); 6009 if ((now - lastWarning) > kWarningThrottleNs) { 6010 ALOGW("RecordThread: buffer overflow"); 6011 lastWarning = now; 6012 } 6013 } 6014 // Release the processor for a while before asking for a new buffer. 6015 // This will give the application more chance to read from the buffer and 6016 // clear the overflow. 6017 usleep(kRecordThreadSleepUs); 6018 } 6019 } 6020 // enable changes in effect chain 6021 unlockEffectChains(effectChains); 6022 effectChains.clear(); 6023 } 6024 6025 if (!mStandby) { 6026 mInput->stream->common.standby(&mInput->stream->common); 6027 } 6028 mActiveTrack.clear(); 6029 6030 mStartStopCond.broadcast(); 6031 6032 releaseWakeLock(); 6033 6034 ALOGV("RecordThread %p exiting", this); 6035 return false; 6036} 6037 6038 6039sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6040 const sp<AudioFlinger::Client>& client, 6041 uint32_t sampleRate, 6042 audio_format_t format, 6043 int channelMask, 6044 int frameCount, 6045 int sessionId, 6046 status_t *status) 6047{ 6048 sp<RecordTrack> track; 6049 status_t lStatus; 6050 6051 lStatus = initCheck(); 6052 if (lStatus != NO_ERROR) { 6053 ALOGE("Audio driver not initialized."); 6054 goto Exit; 6055 } 6056 6057 { // scope for mLock 6058 Mutex::Autolock _l(mLock); 6059 6060 track = new RecordTrack(this, client, sampleRate, 6061 format, channelMask, frameCount, sessionId); 6062 6063 if (track->getCblk() == 0) { 6064 lStatus = NO_MEMORY; 6065 goto Exit; 6066 } 6067 6068 mTrack = track.get(); 6069 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6070 bool suspend = audio_is_bluetooth_sco_device( 6071 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6072 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6073 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6074 } 6075 lStatus = NO_ERROR; 6076 6077Exit: 6078 if (status) { 6079 *status = lStatus; 6080 } 6081 return track; 6082} 6083 6084status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6085 AudioSystem::sync_event_t event, 6086 int triggerSession) 6087{ 6088 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6089 sp<ThreadBase> strongMe = this; 6090 status_t status = NO_ERROR; 6091 6092 if (event == AudioSystem::SYNC_EVENT_NONE) { 6093 clearSyncStartEvent(); 6094 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6095 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6096 triggerSession, 6097 recordTrack->sessionId(), 6098 syncStartEventCallback, 6099 this); 6100 // Sync event can be cancelled by the trigger session if the track is not in a 6101 // compatible state in which case we start record immediately 6102 if (mSyncStartEvent->isCancelled()) { 6103 clearSyncStartEvent(); 6104 } else { 6105 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6106 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6107 } 6108 } 6109 6110 { 6111 AutoMutex lock(mLock); 6112 if (mActiveTrack != 0) { 6113 if (recordTrack != mActiveTrack.get()) { 6114 status = -EBUSY; 6115 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6116 mActiveTrack->mState = TrackBase::ACTIVE; 6117 } 6118 return status; 6119 } 6120 6121 recordTrack->mState = TrackBase::IDLE; 6122 mActiveTrack = recordTrack; 6123 mLock.unlock(); 6124 status_t status = AudioSystem::startInput(mId); 6125 mLock.lock(); 6126 if (status != NO_ERROR) { 6127 mActiveTrack.clear(); 6128 clearSyncStartEvent(); 6129 return status; 6130 } 6131 mRsmpInIndex = mFrameCount; 6132 mBytesRead = 0; 6133 if (mResampler != NULL) { 6134 mResampler->reset(); 6135 } 6136 mActiveTrack->mState = TrackBase::RESUMING; 6137 // signal thread to start 6138 ALOGV("Signal record thread"); 6139 mWaitWorkCV.signal(); 6140 // do not wait for mStartStopCond if exiting 6141 if (exitPending()) { 6142 mActiveTrack.clear(); 6143 status = INVALID_OPERATION; 6144 goto startError; 6145 } 6146 mStartStopCond.wait(mLock); 6147 if (mActiveTrack == 0) { 6148 ALOGV("Record failed to start"); 6149 status = BAD_VALUE; 6150 goto startError; 6151 } 6152 ALOGV("Record started OK"); 6153 return status; 6154 } 6155startError: 6156 AudioSystem::stopInput(mId); 6157 clearSyncStartEvent(); 6158 return status; 6159} 6160 6161void AudioFlinger::RecordThread::clearSyncStartEvent() 6162{ 6163 if (mSyncStartEvent != 0) { 6164 mSyncStartEvent->cancel(); 6165 } 6166 mSyncStartEvent.clear(); 6167 mFramestoDrop = 0; 6168} 6169 6170void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6171{ 6172 sp<SyncEvent> strongEvent = event.promote(); 6173 6174 if (strongEvent != 0) { 6175 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6176 me->handleSyncStartEvent(strongEvent); 6177 } 6178} 6179 6180void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6181{ 6182 if (event == mSyncStartEvent) { 6183 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6184 // from audio HAL 6185 mFramestoDrop = mFrameCount * 2; 6186 } 6187} 6188 6189void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6190 ALOGV("RecordThread::stop"); 6191 sp<ThreadBase> strongMe = this; 6192 { 6193 AutoMutex lock(mLock); 6194 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6195 mActiveTrack->mState = TrackBase::PAUSING; 6196 // do not wait for mStartStopCond if exiting 6197 if (exitPending()) { 6198 return; 6199 } 6200 mStartStopCond.wait(mLock); 6201 // if we have been restarted, recordTrack == mActiveTrack.get() here 6202 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6203 mLock.unlock(); 6204 AudioSystem::stopInput(mId); 6205 mLock.lock(); 6206 ALOGV("Record stopped OK"); 6207 } 6208 } 6209 } 6210} 6211 6212bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6213{ 6214 return false; 6215} 6216 6217status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6218{ 6219 if (!isValidSyncEvent(event)) { 6220 return BAD_VALUE; 6221 } 6222 6223 Mutex::Autolock _l(mLock); 6224 6225 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6226 mTrack->setSyncEvent(event); 6227 return NO_ERROR; 6228 } 6229 return NAME_NOT_FOUND; 6230} 6231 6232status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6233{ 6234 const size_t SIZE = 256; 6235 char buffer[SIZE]; 6236 String8 result; 6237 6238 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6239 result.append(buffer); 6240 6241 if (mActiveTrack != 0) { 6242 result.append("Active Track:\n"); 6243 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6244 mActiveTrack->dump(buffer, SIZE); 6245 result.append(buffer); 6246 6247 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6248 result.append(buffer); 6249 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6250 result.append(buffer); 6251 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6252 result.append(buffer); 6253 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6254 result.append(buffer); 6255 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6256 result.append(buffer); 6257 6258 6259 } else { 6260 result.append("No record client\n"); 6261 } 6262 write(fd, result.string(), result.size()); 6263 6264 dumpBase(fd, args); 6265 dumpEffectChains(fd, args); 6266 6267 return NO_ERROR; 6268} 6269 6270// AudioBufferProvider interface 6271status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6272{ 6273 size_t framesReq = buffer->frameCount; 6274 size_t framesReady = mFrameCount - mRsmpInIndex; 6275 int channelCount; 6276 6277 if (framesReady == 0) { 6278 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6279 if (mBytesRead < 0) { 6280 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6281 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6282 // Force input into standby so that it tries to 6283 // recover at next read attempt 6284 mInput->stream->common.standby(&mInput->stream->common); 6285 usleep(kRecordThreadSleepUs); 6286 } 6287 buffer->raw = NULL; 6288 buffer->frameCount = 0; 6289 return NOT_ENOUGH_DATA; 6290 } 6291 mRsmpInIndex = 0; 6292 framesReady = mFrameCount; 6293 } 6294 6295 if (framesReq > framesReady) { 6296 framesReq = framesReady; 6297 } 6298 6299 if (mChannelCount == 1 && mReqChannelCount == 2) { 6300 channelCount = 1; 6301 } else { 6302 channelCount = 2; 6303 } 6304 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6305 buffer->frameCount = framesReq; 6306 return NO_ERROR; 6307} 6308 6309// AudioBufferProvider interface 6310void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6311{ 6312 mRsmpInIndex += buffer->frameCount; 6313 buffer->frameCount = 0; 6314} 6315 6316bool AudioFlinger::RecordThread::checkForNewParameters_l() 6317{ 6318 bool reconfig = false; 6319 6320 while (!mNewParameters.isEmpty()) { 6321 status_t status = NO_ERROR; 6322 String8 keyValuePair = mNewParameters[0]; 6323 AudioParameter param = AudioParameter(keyValuePair); 6324 int value; 6325 audio_format_t reqFormat = mFormat; 6326 int reqSamplingRate = mReqSampleRate; 6327 int reqChannelCount = mReqChannelCount; 6328 6329 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6330 reqSamplingRate = value; 6331 reconfig = true; 6332 } 6333 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6334 reqFormat = (audio_format_t) value; 6335 reconfig = true; 6336 } 6337 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6338 reqChannelCount = popcount(value); 6339 reconfig = true; 6340 } 6341 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6342 // do not accept frame count changes if tracks are open as the track buffer 6343 // size depends on frame count and correct behavior would not be guaranteed 6344 // if frame count is changed after track creation 6345 if (mActiveTrack != 0) { 6346 status = INVALID_OPERATION; 6347 } else { 6348 reconfig = true; 6349 } 6350 } 6351 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6352 // forward device change to effects that have requested to be 6353 // aware of attached audio device. 6354 for (size_t i = 0; i < mEffectChains.size(); i++) { 6355 mEffectChains[i]->setDevice_l(value); 6356 } 6357 // store input device and output device but do not forward output device to audio HAL. 6358 // Note that status is ignored by the caller for output device 6359 // (see AudioFlinger::setParameters() 6360 if (value & AUDIO_DEVICE_OUT_ALL) { 6361 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6362 status = BAD_VALUE; 6363 } else { 6364 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6365 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6366 if (mTrack != NULL) { 6367 bool suspend = audio_is_bluetooth_sco_device( 6368 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6369 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6370 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6371 } 6372 } 6373 mDevice |= (uint32_t)value; 6374 } 6375 if (status == NO_ERROR) { 6376 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6377 if (status == INVALID_OPERATION) { 6378 mInput->stream->common.standby(&mInput->stream->common); 6379 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6380 keyValuePair.string()); 6381 } 6382 if (reconfig) { 6383 if (status == BAD_VALUE && 6384 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6385 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6386 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6387 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6388 (reqChannelCount <= FCC_2)) { 6389 status = NO_ERROR; 6390 } 6391 if (status == NO_ERROR) { 6392 readInputParameters(); 6393 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6394 } 6395 } 6396 } 6397 6398 mNewParameters.removeAt(0); 6399 6400 mParamStatus = status; 6401 mParamCond.signal(); 6402 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6403 // already timed out waiting for the status and will never signal the condition. 6404 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6405 } 6406 return reconfig; 6407} 6408 6409String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6410{ 6411 char *s; 6412 String8 out_s8 = String8(); 6413 6414 Mutex::Autolock _l(mLock); 6415 if (initCheck() != NO_ERROR) { 6416 return out_s8; 6417 } 6418 6419 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6420 out_s8 = String8(s); 6421 free(s); 6422 return out_s8; 6423} 6424 6425void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6426 AudioSystem::OutputDescriptor desc; 6427 void *param2 = NULL; 6428 6429 switch (event) { 6430 case AudioSystem::INPUT_OPENED: 6431 case AudioSystem::INPUT_CONFIG_CHANGED: 6432 desc.channels = mChannelMask; 6433 desc.samplingRate = mSampleRate; 6434 desc.format = mFormat; 6435 desc.frameCount = mFrameCount; 6436 desc.latency = 0; 6437 param2 = &desc; 6438 break; 6439 6440 case AudioSystem::INPUT_CLOSED: 6441 default: 6442 break; 6443 } 6444 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6445} 6446 6447void AudioFlinger::RecordThread::readInputParameters() 6448{ 6449 delete mRsmpInBuffer; 6450 // mRsmpInBuffer is always assigned a new[] below 6451 delete mRsmpOutBuffer; 6452 mRsmpOutBuffer = NULL; 6453 delete mResampler; 6454 mResampler = NULL; 6455 6456 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6457 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6458 mChannelCount = (uint16_t)popcount(mChannelMask); 6459 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6460 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6461 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6462 mFrameCount = mInputBytes / mFrameSize; 6463 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6464 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6465 6466 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6467 { 6468 int channelCount; 6469 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6470 // stereo to mono post process as the resampler always outputs stereo. 6471 if (mChannelCount == 1 && mReqChannelCount == 2) { 6472 channelCount = 1; 6473 } else { 6474 channelCount = 2; 6475 } 6476 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6477 mResampler->setSampleRate(mSampleRate); 6478 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6479 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6480 6481 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6482 if (mChannelCount == 1 && mReqChannelCount == 1) { 6483 mFrameCount >>= 1; 6484 } 6485 6486 } 6487 mRsmpInIndex = mFrameCount; 6488} 6489 6490unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6491{ 6492 Mutex::Autolock _l(mLock); 6493 if (initCheck() != NO_ERROR) { 6494 return 0; 6495 } 6496 6497 return mInput->stream->get_input_frames_lost(mInput->stream); 6498} 6499 6500uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6501{ 6502 Mutex::Autolock _l(mLock); 6503 uint32_t result = 0; 6504 if (getEffectChain_l(sessionId) != 0) { 6505 result = EFFECT_SESSION; 6506 } 6507 6508 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6509 result |= TRACK_SESSION; 6510 } 6511 6512 return result; 6513} 6514 6515AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6516{ 6517 Mutex::Autolock _l(mLock); 6518 return mTrack; 6519} 6520 6521AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6522{ 6523 Mutex::Autolock _l(mLock); 6524 return mInput; 6525} 6526 6527AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6528{ 6529 Mutex::Autolock _l(mLock); 6530 AudioStreamIn *input = mInput; 6531 mInput = NULL; 6532 return input; 6533} 6534 6535// this method must always be called either with ThreadBase mLock held or inside the thread loop 6536audio_stream_t* AudioFlinger::RecordThread::stream() const 6537{ 6538 if (mInput == NULL) { 6539 return NULL; 6540 } 6541 return &mInput->stream->common; 6542} 6543 6544 6545// ---------------------------------------------------------------------------- 6546 6547audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6548{ 6549 if (!settingsAllowed()) { 6550 return 0; 6551 } 6552 Mutex::Autolock _l(mLock); 6553 return loadHwModule_l(name); 6554} 6555 6556// loadHwModule_l() must be called with AudioFlinger::mLock held 6557audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6558{ 6559 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6560 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6561 ALOGW("loadHwModule() module %s already loaded", name); 6562 return mAudioHwDevs.keyAt(i); 6563 } 6564 } 6565 6566 audio_hw_device_t *dev; 6567 6568 int rc = load_audio_interface(name, &dev); 6569 if (rc) { 6570 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6571 return 0; 6572 } 6573 6574 mHardwareStatus = AUDIO_HW_INIT; 6575 rc = dev->init_check(dev); 6576 mHardwareStatus = AUDIO_HW_IDLE; 6577 if (rc) { 6578 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6579 return 0; 6580 } 6581 6582 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6583 (NULL != dev->set_master_volume)) { 6584 AutoMutex lock(mHardwareLock); 6585 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6586 dev->set_master_volume(dev, mMasterVolume); 6587 mHardwareStatus = AUDIO_HW_IDLE; 6588 } 6589 6590 audio_module_handle_t handle = nextUniqueId(); 6591 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6592 6593 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6594 name, dev->common.module->name, dev->common.module->id, handle); 6595 6596 return handle; 6597 6598} 6599 6600audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6601 audio_devices_t *pDevices, 6602 uint32_t *pSamplingRate, 6603 audio_format_t *pFormat, 6604 audio_channel_mask_t *pChannelMask, 6605 uint32_t *pLatencyMs, 6606 audio_output_flags_t flags) 6607{ 6608 status_t status; 6609 PlaybackThread *thread = NULL; 6610 struct audio_config config = { 6611 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6612 channel_mask: pChannelMask ? *pChannelMask : 0, 6613 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6614 }; 6615 audio_stream_out_t *outStream = NULL; 6616 audio_hw_device_t *outHwDev; 6617 6618 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6619 module, 6620 (pDevices != NULL) ? (int)*pDevices : 0, 6621 config.sample_rate, 6622 config.format, 6623 config.channel_mask, 6624 flags); 6625 6626 if (pDevices == NULL || *pDevices == 0) { 6627 return 0; 6628 } 6629 6630 Mutex::Autolock _l(mLock); 6631 6632 outHwDev = findSuitableHwDev_l(module, *pDevices); 6633 if (outHwDev == NULL) 6634 return 0; 6635 6636 audio_io_handle_t id = nextUniqueId(); 6637 6638 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6639 6640 status = outHwDev->open_output_stream(outHwDev, 6641 id, 6642 *pDevices, 6643 (audio_output_flags_t)flags, 6644 &config, 6645 &outStream); 6646 6647 mHardwareStatus = AUDIO_HW_IDLE; 6648 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6649 outStream, 6650 config.sample_rate, 6651 config.format, 6652 config.channel_mask, 6653 status); 6654 6655 if (status == NO_ERROR && outStream != NULL) { 6656 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6657 6658 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6659 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6660 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6661 thread = new DirectOutputThread(this, output, id, *pDevices); 6662 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6663 } else { 6664 thread = new MixerThread(this, output, id, *pDevices); 6665 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6666 } 6667 mPlaybackThreads.add(id, thread); 6668 6669 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6670 if (pFormat != NULL) *pFormat = config.format; 6671 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6672 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6673 6674 // notify client processes of the new output creation 6675 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6676 6677 // the first primary output opened designates the primary hw device 6678 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6679 ALOGI("Using module %d has the primary audio interface", module); 6680 mPrimaryHardwareDev = outHwDev; 6681 6682 AutoMutex lock(mHardwareLock); 6683 mHardwareStatus = AUDIO_HW_SET_MODE; 6684 outHwDev->set_mode(outHwDev, mMode); 6685 6686 // Determine the level of master volume support the primary audio HAL has, 6687 // and set the initial master volume at the same time. 6688 float initialVolume = 1.0; 6689 mMasterVolumeSupportLvl = MVS_NONE; 6690 6691 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6692 if ((NULL != outHwDev->get_master_volume) && 6693 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6694 mMasterVolumeSupportLvl = MVS_FULL; 6695 } else { 6696 mMasterVolumeSupportLvl = MVS_SETONLY; 6697 initialVolume = 1.0; 6698 } 6699 6700 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6701 if ((NULL == outHwDev->set_master_volume) || 6702 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6703 mMasterVolumeSupportLvl = MVS_NONE; 6704 } 6705 // now that we have a primary device, initialize master volume on other devices 6706 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6707 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6708 6709 if ((dev != mPrimaryHardwareDev) && 6710 (NULL != dev->set_master_volume)) { 6711 dev->set_master_volume(dev, initialVolume); 6712 } 6713 } 6714 mHardwareStatus = AUDIO_HW_IDLE; 6715 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6716 ? initialVolume 6717 : 1.0; 6718 mMasterVolume = initialVolume; 6719 } 6720 return id; 6721 } 6722 6723 return 0; 6724} 6725 6726audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6727 audio_io_handle_t output2) 6728{ 6729 Mutex::Autolock _l(mLock); 6730 MixerThread *thread1 = checkMixerThread_l(output1); 6731 MixerThread *thread2 = checkMixerThread_l(output2); 6732 6733 if (thread1 == NULL || thread2 == NULL) { 6734 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6735 return 0; 6736 } 6737 6738 audio_io_handle_t id = nextUniqueId(); 6739 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6740 thread->addOutputTrack(thread2); 6741 mPlaybackThreads.add(id, thread); 6742 // notify client processes of the new output creation 6743 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6744 return id; 6745} 6746 6747status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6748{ 6749 // keep strong reference on the playback thread so that 6750 // it is not destroyed while exit() is executed 6751 sp<PlaybackThread> thread; 6752 { 6753 Mutex::Autolock _l(mLock); 6754 thread = checkPlaybackThread_l(output); 6755 if (thread == NULL) { 6756 return BAD_VALUE; 6757 } 6758 6759 ALOGV("closeOutput() %d", output); 6760 6761 if (thread->type() == ThreadBase::MIXER) { 6762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6763 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6764 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6765 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6766 } 6767 } 6768 } 6769 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6770 mPlaybackThreads.removeItem(output); 6771 } 6772 thread->exit(); 6773 // The thread entity (active unit of execution) is no longer running here, 6774 // but the ThreadBase container still exists. 6775 6776 if (thread->type() != ThreadBase::DUPLICATING) { 6777 AudioStreamOut *out = thread->clearOutput(); 6778 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6779 // from now on thread->mOutput is NULL 6780 out->hwDev->close_output_stream(out->hwDev, out->stream); 6781 delete out; 6782 } 6783 return NO_ERROR; 6784} 6785 6786status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6787{ 6788 Mutex::Autolock _l(mLock); 6789 PlaybackThread *thread = checkPlaybackThread_l(output); 6790 6791 if (thread == NULL) { 6792 return BAD_VALUE; 6793 } 6794 6795 ALOGV("suspendOutput() %d", output); 6796 thread->suspend(); 6797 6798 return NO_ERROR; 6799} 6800 6801status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6802{ 6803 Mutex::Autolock _l(mLock); 6804 PlaybackThread *thread = checkPlaybackThread_l(output); 6805 6806 if (thread == NULL) { 6807 return BAD_VALUE; 6808 } 6809 6810 ALOGV("restoreOutput() %d", output); 6811 6812 thread->restore(); 6813 6814 return NO_ERROR; 6815} 6816 6817audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6818 audio_devices_t *pDevices, 6819 uint32_t *pSamplingRate, 6820 audio_format_t *pFormat, 6821 uint32_t *pChannelMask) 6822{ 6823 status_t status; 6824 RecordThread *thread = NULL; 6825 struct audio_config config = { 6826 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6827 channel_mask: pChannelMask ? *pChannelMask : 0, 6828 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6829 }; 6830 uint32_t reqSamplingRate = config.sample_rate; 6831 audio_format_t reqFormat = config.format; 6832 audio_channel_mask_t reqChannels = config.channel_mask; 6833 audio_stream_in_t *inStream = NULL; 6834 audio_hw_device_t *inHwDev; 6835 6836 if (pDevices == NULL || *pDevices == 0) { 6837 return 0; 6838 } 6839 6840 Mutex::Autolock _l(mLock); 6841 6842 inHwDev = findSuitableHwDev_l(module, *pDevices); 6843 if (inHwDev == NULL) 6844 return 0; 6845 6846 audio_io_handle_t id = nextUniqueId(); 6847 6848 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6849 &inStream); 6850 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6851 inStream, 6852 config.sample_rate, 6853 config.format, 6854 config.channel_mask, 6855 status); 6856 6857 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6858 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6859 // or stereo to mono conversions on 16 bit PCM inputs. 6860 if (status == BAD_VALUE && 6861 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6862 (config.sample_rate <= 2 * reqSamplingRate) && 6863 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6864 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6865 inStream = NULL; 6866 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6867 } 6868 6869 if (status == NO_ERROR && inStream != NULL) { 6870 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6871 6872 // Start record thread 6873 // RecorThread require both input and output device indication to forward to audio 6874 // pre processing modules 6875 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6876 thread = new RecordThread(this, 6877 input, 6878 reqSamplingRate, 6879 reqChannels, 6880 id, 6881 device); 6882 mRecordThreads.add(id, thread); 6883 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6884 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6885 if (pFormat != NULL) *pFormat = config.format; 6886 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6887 6888 input->stream->common.standby(&input->stream->common); 6889 6890 // notify client processes of the new input creation 6891 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6892 return id; 6893 } 6894 6895 return 0; 6896} 6897 6898status_t AudioFlinger::closeInput(audio_io_handle_t input) 6899{ 6900 // keep strong reference on the record thread so that 6901 // it is not destroyed while exit() is executed 6902 sp<RecordThread> thread; 6903 { 6904 Mutex::Autolock _l(mLock); 6905 thread = checkRecordThread_l(input); 6906 if (thread == NULL) { 6907 return BAD_VALUE; 6908 } 6909 6910 ALOGV("closeInput() %d", input); 6911 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6912 mRecordThreads.removeItem(input); 6913 } 6914 thread->exit(); 6915 // The thread entity (active unit of execution) is no longer running here, 6916 // but the ThreadBase container still exists. 6917 6918 AudioStreamIn *in = thread->clearInput(); 6919 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6920 // from now on thread->mInput is NULL 6921 in->hwDev->close_input_stream(in->hwDev, in->stream); 6922 delete in; 6923 6924 return NO_ERROR; 6925} 6926 6927status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6928{ 6929 Mutex::Autolock _l(mLock); 6930 MixerThread *dstThread = checkMixerThread_l(output); 6931 if (dstThread == NULL) { 6932 ALOGW("setStreamOutput() bad output id %d", output); 6933 return BAD_VALUE; 6934 } 6935 6936 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6937 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6938 6939 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6940 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6941 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6942 MixerThread *srcThread = (MixerThread *)thread; 6943 srcThread->invalidateTracks(stream); 6944 } 6945 } 6946 6947 return NO_ERROR; 6948} 6949 6950 6951int AudioFlinger::newAudioSessionId() 6952{ 6953 return nextUniqueId(); 6954} 6955 6956void AudioFlinger::acquireAudioSessionId(int audioSession) 6957{ 6958 Mutex::Autolock _l(mLock); 6959 pid_t caller = IPCThreadState::self()->getCallingPid(); 6960 ALOGV("acquiring %d from %d", audioSession, caller); 6961 size_t num = mAudioSessionRefs.size(); 6962 for (size_t i = 0; i< num; i++) { 6963 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6964 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6965 ref->mCnt++; 6966 ALOGV(" incremented refcount to %d", ref->mCnt); 6967 return; 6968 } 6969 } 6970 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6971 ALOGV(" added new entry for %d", audioSession); 6972} 6973 6974void AudioFlinger::releaseAudioSessionId(int audioSession) 6975{ 6976 Mutex::Autolock _l(mLock); 6977 pid_t caller = IPCThreadState::self()->getCallingPid(); 6978 ALOGV("releasing %d from %d", audioSession, caller); 6979 size_t num = mAudioSessionRefs.size(); 6980 for (size_t i = 0; i< num; i++) { 6981 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6982 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6983 ref->mCnt--; 6984 ALOGV(" decremented refcount to %d", ref->mCnt); 6985 if (ref->mCnt == 0) { 6986 mAudioSessionRefs.removeAt(i); 6987 delete ref; 6988 purgeStaleEffects_l(); 6989 } 6990 return; 6991 } 6992 } 6993 ALOGW("session id %d not found for pid %d", audioSession, caller); 6994} 6995 6996void AudioFlinger::purgeStaleEffects_l() { 6997 6998 ALOGV("purging stale effects"); 6999 7000 Vector< sp<EffectChain> > chains; 7001 7002 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7003 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7004 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7005 sp<EffectChain> ec = t->mEffectChains[j]; 7006 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7007 chains.push(ec); 7008 } 7009 } 7010 } 7011 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7012 sp<RecordThread> t = mRecordThreads.valueAt(i); 7013 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7014 sp<EffectChain> ec = t->mEffectChains[j]; 7015 chains.push(ec); 7016 } 7017 } 7018 7019 for (size_t i = 0; i < chains.size(); i++) { 7020 sp<EffectChain> ec = chains[i]; 7021 int sessionid = ec->sessionId(); 7022 sp<ThreadBase> t = ec->mThread.promote(); 7023 if (t == 0) { 7024 continue; 7025 } 7026 size_t numsessionrefs = mAudioSessionRefs.size(); 7027 bool found = false; 7028 for (size_t k = 0; k < numsessionrefs; k++) { 7029 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7030 if (ref->mSessionid == sessionid) { 7031 ALOGV(" session %d still exists for %d with %d refs", 7032 sessionid, ref->mPid, ref->mCnt); 7033 found = true; 7034 break; 7035 } 7036 } 7037 if (!found) { 7038 // remove all effects from the chain 7039 while (ec->mEffects.size()) { 7040 sp<EffectModule> effect = ec->mEffects[0]; 7041 effect->unPin(); 7042 Mutex::Autolock _l (t->mLock); 7043 t->removeEffect_l(effect); 7044 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7045 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7046 if (handle != 0) { 7047 handle->mEffect.clear(); 7048 if (handle->mHasControl && handle->mEnabled) { 7049 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7050 } 7051 } 7052 } 7053 AudioSystem::unregisterEffect(effect->id()); 7054 } 7055 } 7056 } 7057 return; 7058} 7059 7060// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7061AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7062{ 7063 return mPlaybackThreads.valueFor(output).get(); 7064} 7065 7066// checkMixerThread_l() must be called with AudioFlinger::mLock held 7067AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7068{ 7069 PlaybackThread *thread = checkPlaybackThread_l(output); 7070 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7071} 7072 7073// checkRecordThread_l() must be called with AudioFlinger::mLock held 7074AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7075{ 7076 return mRecordThreads.valueFor(input).get(); 7077} 7078 7079uint32_t AudioFlinger::nextUniqueId() 7080{ 7081 return android_atomic_inc(&mNextUniqueId); 7082} 7083 7084AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7085{ 7086 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7087 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7088 AudioStreamOut *output = thread->getOutput(); 7089 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7090 return thread; 7091 } 7092 } 7093 return NULL; 7094} 7095 7096uint32_t AudioFlinger::primaryOutputDevice_l() const 7097{ 7098 PlaybackThread *thread = primaryPlaybackThread_l(); 7099 7100 if (thread == NULL) { 7101 return 0; 7102 } 7103 7104 return thread->device(); 7105} 7106 7107sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7108 int triggerSession, 7109 int listenerSession, 7110 sync_event_callback_t callBack, 7111 void *cookie) 7112{ 7113 Mutex::Autolock _l(mLock); 7114 7115 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7116 status_t playStatus = NAME_NOT_FOUND; 7117 status_t recStatus = NAME_NOT_FOUND; 7118 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7119 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7120 if (playStatus == NO_ERROR) { 7121 return event; 7122 } 7123 } 7124 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7125 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7126 if (recStatus == NO_ERROR) { 7127 return event; 7128 } 7129 } 7130 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7131 mPendingSyncEvents.add(event); 7132 } else { 7133 ALOGV("createSyncEvent() invalid event %d", event->type()); 7134 event.clear(); 7135 } 7136 return event; 7137} 7138 7139// ---------------------------------------------------------------------------- 7140// Effect management 7141// ---------------------------------------------------------------------------- 7142 7143 7144status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7145{ 7146 Mutex::Autolock _l(mLock); 7147 return EffectQueryNumberEffects(numEffects); 7148} 7149 7150status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7151{ 7152 Mutex::Autolock _l(mLock); 7153 return EffectQueryEffect(index, descriptor); 7154} 7155 7156status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7157 effect_descriptor_t *descriptor) const 7158{ 7159 Mutex::Autolock _l(mLock); 7160 return EffectGetDescriptor(pUuid, descriptor); 7161} 7162 7163 7164sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7165 effect_descriptor_t *pDesc, 7166 const sp<IEffectClient>& effectClient, 7167 int32_t priority, 7168 audio_io_handle_t io, 7169 int sessionId, 7170 status_t *status, 7171 int *id, 7172 int *enabled) 7173{ 7174 status_t lStatus = NO_ERROR; 7175 sp<EffectHandle> handle; 7176 effect_descriptor_t desc; 7177 7178 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7179 pid, effectClient.get(), priority, sessionId, io); 7180 7181 if (pDesc == NULL) { 7182 lStatus = BAD_VALUE; 7183 goto Exit; 7184 } 7185 7186 // check audio settings permission for global effects 7187 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7188 lStatus = PERMISSION_DENIED; 7189 goto Exit; 7190 } 7191 7192 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7193 // that can only be created by audio policy manager (running in same process) 7194 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7195 lStatus = PERMISSION_DENIED; 7196 goto Exit; 7197 } 7198 7199 if (io == 0) { 7200 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7201 // output must be specified by AudioPolicyManager when using session 7202 // AUDIO_SESSION_OUTPUT_STAGE 7203 lStatus = BAD_VALUE; 7204 goto Exit; 7205 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7206 // if the output returned by getOutputForEffect() is removed before we lock the 7207 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7208 // and we will exit safely 7209 io = AudioSystem::getOutputForEffect(&desc); 7210 } 7211 } 7212 7213 { 7214 Mutex::Autolock _l(mLock); 7215 7216 7217 if (!EffectIsNullUuid(&pDesc->uuid)) { 7218 // if uuid is specified, request effect descriptor 7219 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7220 if (lStatus < 0) { 7221 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7222 goto Exit; 7223 } 7224 } else { 7225 // if uuid is not specified, look for an available implementation 7226 // of the required type in effect factory 7227 if (EffectIsNullUuid(&pDesc->type)) { 7228 ALOGW("createEffect() no effect type"); 7229 lStatus = BAD_VALUE; 7230 goto Exit; 7231 } 7232 uint32_t numEffects = 0; 7233 effect_descriptor_t d; 7234 d.flags = 0; // prevent compiler warning 7235 bool found = false; 7236 7237 lStatus = EffectQueryNumberEffects(&numEffects); 7238 if (lStatus < 0) { 7239 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7240 goto Exit; 7241 } 7242 for (uint32_t i = 0; i < numEffects; i++) { 7243 lStatus = EffectQueryEffect(i, &desc); 7244 if (lStatus < 0) { 7245 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7246 continue; 7247 } 7248 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7249 // If matching type found save effect descriptor. If the session is 7250 // 0 and the effect is not auxiliary, continue enumeration in case 7251 // an auxiliary version of this effect type is available 7252 found = true; 7253 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7254 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7255 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7256 break; 7257 } 7258 } 7259 } 7260 if (!found) { 7261 lStatus = BAD_VALUE; 7262 ALOGW("createEffect() effect not found"); 7263 goto Exit; 7264 } 7265 // For same effect type, chose auxiliary version over insert version if 7266 // connect to output mix (Compliance to OpenSL ES) 7267 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7268 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7269 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7270 } 7271 } 7272 7273 // Do not allow auxiliary effects on a session different from 0 (output mix) 7274 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7275 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7276 lStatus = INVALID_OPERATION; 7277 goto Exit; 7278 } 7279 7280 // check recording permission for visualizer 7281 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7282 !recordingAllowed()) { 7283 lStatus = PERMISSION_DENIED; 7284 goto Exit; 7285 } 7286 7287 // return effect descriptor 7288 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7289 7290 // If output is not specified try to find a matching audio session ID in one of the 7291 // output threads. 7292 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7293 // because of code checking output when entering the function. 7294 // Note: io is never 0 when creating an effect on an input 7295 if (io == 0) { 7296 // look for the thread where the specified audio session is present 7297 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7298 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7299 io = mPlaybackThreads.keyAt(i); 7300 break; 7301 } 7302 } 7303 if (io == 0) { 7304 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7305 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7306 io = mRecordThreads.keyAt(i); 7307 break; 7308 } 7309 } 7310 } 7311 // If no output thread contains the requested session ID, default to 7312 // first output. The effect chain will be moved to the correct output 7313 // thread when a track with the same session ID is created 7314 if (io == 0 && mPlaybackThreads.size()) { 7315 io = mPlaybackThreads.keyAt(0); 7316 } 7317 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7318 } 7319 ThreadBase *thread = checkRecordThread_l(io); 7320 if (thread == NULL) { 7321 thread = checkPlaybackThread_l(io); 7322 if (thread == NULL) { 7323 ALOGE("createEffect() unknown output thread"); 7324 lStatus = BAD_VALUE; 7325 goto Exit; 7326 } 7327 } 7328 7329 sp<Client> client = registerPid_l(pid); 7330 7331 // create effect on selected output thread 7332 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7333 &desc, enabled, &lStatus); 7334 if (handle != 0 && id != NULL) { 7335 *id = handle->id(); 7336 } 7337 } 7338 7339Exit: 7340 if (status != NULL) { 7341 *status = lStatus; 7342 } 7343 return handle; 7344} 7345 7346status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7347 audio_io_handle_t dstOutput) 7348{ 7349 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7350 sessionId, srcOutput, dstOutput); 7351 Mutex::Autolock _l(mLock); 7352 if (srcOutput == dstOutput) { 7353 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7354 return NO_ERROR; 7355 } 7356 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7357 if (srcThread == NULL) { 7358 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7359 return BAD_VALUE; 7360 } 7361 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7362 if (dstThread == NULL) { 7363 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7364 return BAD_VALUE; 7365 } 7366 7367 Mutex::Autolock _dl(dstThread->mLock); 7368 Mutex::Autolock _sl(srcThread->mLock); 7369 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7370 7371 return NO_ERROR; 7372} 7373 7374// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7375status_t AudioFlinger::moveEffectChain_l(int sessionId, 7376 AudioFlinger::PlaybackThread *srcThread, 7377 AudioFlinger::PlaybackThread *dstThread, 7378 bool reRegister) 7379{ 7380 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7381 sessionId, srcThread, dstThread); 7382 7383 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7384 if (chain == 0) { 7385 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7386 sessionId, srcThread); 7387 return INVALID_OPERATION; 7388 } 7389 7390 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7391 // so that a new chain is created with correct parameters when first effect is added. This is 7392 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7393 // removed. 7394 srcThread->removeEffectChain_l(chain); 7395 7396 // transfer all effects one by one so that new effect chain is created on new thread with 7397 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7398 audio_io_handle_t dstOutput = dstThread->id(); 7399 sp<EffectChain> dstChain; 7400 uint32_t strategy = 0; // prevent compiler warning 7401 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7402 while (effect != 0) { 7403 srcThread->removeEffect_l(effect); 7404 dstThread->addEffect_l(effect); 7405 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7406 if (effect->state() == EffectModule::ACTIVE || 7407 effect->state() == EffectModule::STOPPING) { 7408 effect->start(); 7409 } 7410 // if the move request is not received from audio policy manager, the effect must be 7411 // re-registered with the new strategy and output 7412 if (dstChain == 0) { 7413 dstChain = effect->chain().promote(); 7414 if (dstChain == 0) { 7415 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7416 srcThread->addEffect_l(effect); 7417 return NO_INIT; 7418 } 7419 strategy = dstChain->strategy(); 7420 } 7421 if (reRegister) { 7422 AudioSystem::unregisterEffect(effect->id()); 7423 AudioSystem::registerEffect(&effect->desc(), 7424 dstOutput, 7425 strategy, 7426 sessionId, 7427 effect->id()); 7428 } 7429 effect = chain->getEffectFromId_l(0); 7430 } 7431 7432 return NO_ERROR; 7433} 7434 7435 7436// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7437sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7438 const sp<AudioFlinger::Client>& client, 7439 const sp<IEffectClient>& effectClient, 7440 int32_t priority, 7441 int sessionId, 7442 effect_descriptor_t *desc, 7443 int *enabled, 7444 status_t *status 7445 ) 7446{ 7447 sp<EffectModule> effect; 7448 sp<EffectHandle> handle; 7449 status_t lStatus; 7450 sp<EffectChain> chain; 7451 bool chainCreated = false; 7452 bool effectCreated = false; 7453 bool effectRegistered = false; 7454 7455 lStatus = initCheck(); 7456 if (lStatus != NO_ERROR) { 7457 ALOGW("createEffect_l() Audio driver not initialized."); 7458 goto Exit; 7459 } 7460 7461 // Do not allow effects with session ID 0 on direct output or duplicating threads 7462 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7463 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7464 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7465 desc->name, sessionId); 7466 lStatus = BAD_VALUE; 7467 goto Exit; 7468 } 7469 // Only Pre processor effects are allowed on input threads and only on input threads 7470 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7471 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7472 desc->name, desc->flags, mType); 7473 lStatus = BAD_VALUE; 7474 goto Exit; 7475 } 7476 7477 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7478 7479 { // scope for mLock 7480 Mutex::Autolock _l(mLock); 7481 7482 // check for existing effect chain with the requested audio session 7483 chain = getEffectChain_l(sessionId); 7484 if (chain == 0) { 7485 // create a new chain for this session 7486 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7487 chain = new EffectChain(this, sessionId); 7488 addEffectChain_l(chain); 7489 chain->setStrategy(getStrategyForSession_l(sessionId)); 7490 chainCreated = true; 7491 } else { 7492 effect = chain->getEffectFromDesc_l(desc); 7493 } 7494 7495 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7496 7497 if (effect == 0) { 7498 int id = mAudioFlinger->nextUniqueId(); 7499 // Check CPU and memory usage 7500 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7501 if (lStatus != NO_ERROR) { 7502 goto Exit; 7503 } 7504 effectRegistered = true; 7505 // create a new effect module if none present in the chain 7506 effect = new EffectModule(this, chain, desc, id, sessionId); 7507 lStatus = effect->status(); 7508 if (lStatus != NO_ERROR) { 7509 goto Exit; 7510 } 7511 lStatus = chain->addEffect_l(effect); 7512 if (lStatus != NO_ERROR) { 7513 goto Exit; 7514 } 7515 effectCreated = true; 7516 7517 effect->setDevice(mDevice); 7518 effect->setMode(mAudioFlinger->getMode()); 7519 } 7520 // create effect handle and connect it to effect module 7521 handle = new EffectHandle(effect, client, effectClient, priority); 7522 lStatus = effect->addHandle(handle); 7523 if (enabled != NULL) { 7524 *enabled = (int)effect->isEnabled(); 7525 } 7526 } 7527 7528Exit: 7529 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7530 Mutex::Autolock _l(mLock); 7531 if (effectCreated) { 7532 chain->removeEffect_l(effect); 7533 } 7534 if (effectRegistered) { 7535 AudioSystem::unregisterEffect(effect->id()); 7536 } 7537 if (chainCreated) { 7538 removeEffectChain_l(chain); 7539 } 7540 handle.clear(); 7541 } 7542 7543 if (status != NULL) { 7544 *status = lStatus; 7545 } 7546 return handle; 7547} 7548 7549sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7550{ 7551 sp<EffectChain> chain = getEffectChain_l(sessionId); 7552 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7553} 7554 7555// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7556// PlaybackThread::mLock held 7557status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7558{ 7559 // check for existing effect chain with the requested audio session 7560 int sessionId = effect->sessionId(); 7561 sp<EffectChain> chain = getEffectChain_l(sessionId); 7562 bool chainCreated = false; 7563 7564 if (chain == 0) { 7565 // create a new chain for this session 7566 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7567 chain = new EffectChain(this, sessionId); 7568 addEffectChain_l(chain); 7569 chain->setStrategy(getStrategyForSession_l(sessionId)); 7570 chainCreated = true; 7571 } 7572 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7573 7574 if (chain->getEffectFromId_l(effect->id()) != 0) { 7575 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7576 this, effect->desc().name, chain.get()); 7577 return BAD_VALUE; 7578 } 7579 7580 status_t status = chain->addEffect_l(effect); 7581 if (status != NO_ERROR) { 7582 if (chainCreated) { 7583 removeEffectChain_l(chain); 7584 } 7585 return status; 7586 } 7587 7588 effect->setDevice(mDevice); 7589 effect->setMode(mAudioFlinger->getMode()); 7590 return NO_ERROR; 7591} 7592 7593void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7594 7595 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7596 effect_descriptor_t desc = effect->desc(); 7597 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7598 detachAuxEffect_l(effect->id()); 7599 } 7600 7601 sp<EffectChain> chain = effect->chain().promote(); 7602 if (chain != 0) { 7603 // remove effect chain if removing last effect 7604 if (chain->removeEffect_l(effect) == 0) { 7605 removeEffectChain_l(chain); 7606 } 7607 } else { 7608 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7609 } 7610} 7611 7612void AudioFlinger::ThreadBase::lockEffectChains_l( 7613 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7614{ 7615 effectChains = mEffectChains; 7616 for (size_t i = 0; i < mEffectChains.size(); i++) { 7617 mEffectChains[i]->lock(); 7618 } 7619} 7620 7621void AudioFlinger::ThreadBase::unlockEffectChains( 7622 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7623{ 7624 for (size_t i = 0; i < effectChains.size(); i++) { 7625 effectChains[i]->unlock(); 7626 } 7627} 7628 7629sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7630{ 7631 Mutex::Autolock _l(mLock); 7632 return getEffectChain_l(sessionId); 7633} 7634 7635sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7636{ 7637 size_t size = mEffectChains.size(); 7638 for (size_t i = 0; i < size; i++) { 7639 if (mEffectChains[i]->sessionId() == sessionId) { 7640 return mEffectChains[i]; 7641 } 7642 } 7643 return 0; 7644} 7645 7646void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7647{ 7648 Mutex::Autolock _l(mLock); 7649 size_t size = mEffectChains.size(); 7650 for (size_t i = 0; i < size; i++) { 7651 mEffectChains[i]->setMode_l(mode); 7652 } 7653} 7654 7655void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7656 const wp<EffectHandle>& handle, 7657 bool unpinIfLast) { 7658 7659 Mutex::Autolock _l(mLock); 7660 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7661 // delete the effect module if removing last handle on it 7662 if (effect->removeHandle(handle) == 0) { 7663 if (!effect->isPinned() || unpinIfLast) { 7664 removeEffect_l(effect); 7665 AudioSystem::unregisterEffect(effect->id()); 7666 } 7667 } 7668} 7669 7670status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7671{ 7672 int session = chain->sessionId(); 7673 int16_t *buffer = mMixBuffer; 7674 bool ownsBuffer = false; 7675 7676 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7677 if (session > 0) { 7678 // Only one effect chain can be present in direct output thread and it uses 7679 // the mix buffer as input 7680 if (mType != DIRECT) { 7681 size_t numSamples = mNormalFrameCount * mChannelCount; 7682 buffer = new int16_t[numSamples]; 7683 memset(buffer, 0, numSamples * sizeof(int16_t)); 7684 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7685 ownsBuffer = true; 7686 } 7687 7688 // Attach all tracks with same session ID to this chain. 7689 for (size_t i = 0; i < mTracks.size(); ++i) { 7690 sp<Track> track = mTracks[i]; 7691 if (session == track->sessionId()) { 7692 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7693 track->setMainBuffer(buffer); 7694 chain->incTrackCnt(); 7695 } 7696 } 7697 7698 // indicate all active tracks in the chain 7699 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7700 sp<Track> track = mActiveTracks[i].promote(); 7701 if (track == 0) continue; 7702 if (session == track->sessionId()) { 7703 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7704 chain->incActiveTrackCnt(); 7705 } 7706 } 7707 } 7708 7709 chain->setInBuffer(buffer, ownsBuffer); 7710 chain->setOutBuffer(mMixBuffer); 7711 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7712 // chains list in order to be processed last as it contains output stage effects 7713 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7714 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7715 // after track specific effects and before output stage 7716 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7717 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7718 // Effect chain for other sessions are inserted at beginning of effect 7719 // chains list to be processed before output mix effects. Relative order between other 7720 // sessions is not important 7721 size_t size = mEffectChains.size(); 7722 size_t i = 0; 7723 for (i = 0; i < size; i++) { 7724 if (mEffectChains[i]->sessionId() < session) break; 7725 } 7726 mEffectChains.insertAt(chain, i); 7727 checkSuspendOnAddEffectChain_l(chain); 7728 7729 return NO_ERROR; 7730} 7731 7732size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7733{ 7734 int session = chain->sessionId(); 7735 7736 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7737 7738 for (size_t i = 0; i < mEffectChains.size(); i++) { 7739 if (chain == mEffectChains[i]) { 7740 mEffectChains.removeAt(i); 7741 // detach all active tracks from the chain 7742 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7743 sp<Track> track = mActiveTracks[i].promote(); 7744 if (track == 0) continue; 7745 if (session == track->sessionId()) { 7746 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7747 chain.get(), session); 7748 chain->decActiveTrackCnt(); 7749 } 7750 } 7751 7752 // detach all tracks with same session ID from this chain 7753 for (size_t i = 0; i < mTracks.size(); ++i) { 7754 sp<Track> track = mTracks[i]; 7755 if (session == track->sessionId()) { 7756 track->setMainBuffer(mMixBuffer); 7757 chain->decTrackCnt(); 7758 } 7759 } 7760 break; 7761 } 7762 } 7763 return mEffectChains.size(); 7764} 7765 7766status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7767 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7768{ 7769 Mutex::Autolock _l(mLock); 7770 return attachAuxEffect_l(track, EffectId); 7771} 7772 7773status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7774 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7775{ 7776 status_t status = NO_ERROR; 7777 7778 if (EffectId == 0) { 7779 track->setAuxBuffer(0, NULL); 7780 } else { 7781 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7782 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7783 if (effect != 0) { 7784 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7785 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7786 } else { 7787 status = INVALID_OPERATION; 7788 } 7789 } else { 7790 status = BAD_VALUE; 7791 } 7792 } 7793 return status; 7794} 7795 7796void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7797{ 7798 for (size_t i = 0; i < mTracks.size(); ++i) { 7799 sp<Track> track = mTracks[i]; 7800 if (track->auxEffectId() == effectId) { 7801 attachAuxEffect_l(track, 0); 7802 } 7803 } 7804} 7805 7806status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7807{ 7808 // only one chain per input thread 7809 if (mEffectChains.size() != 0) { 7810 return INVALID_OPERATION; 7811 } 7812 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7813 7814 chain->setInBuffer(NULL); 7815 chain->setOutBuffer(NULL); 7816 7817 checkSuspendOnAddEffectChain_l(chain); 7818 7819 mEffectChains.add(chain); 7820 7821 return NO_ERROR; 7822} 7823 7824size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7825{ 7826 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7827 ALOGW_IF(mEffectChains.size() != 1, 7828 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7829 chain.get(), mEffectChains.size(), this); 7830 if (mEffectChains.size() == 1) { 7831 mEffectChains.removeAt(0); 7832 } 7833 return 0; 7834} 7835 7836// ---------------------------------------------------------------------------- 7837// EffectModule implementation 7838// ---------------------------------------------------------------------------- 7839 7840#undef LOG_TAG 7841#define LOG_TAG "AudioFlinger::EffectModule" 7842 7843AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7844 const wp<AudioFlinger::EffectChain>& chain, 7845 effect_descriptor_t *desc, 7846 int id, 7847 int sessionId) 7848 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7849 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7850{ 7851 ALOGV("Constructor %p", this); 7852 int lStatus; 7853 if (thread == NULL) { 7854 return; 7855 } 7856 7857 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7858 7859 // create effect engine from effect factory 7860 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7861 7862 if (mStatus != NO_ERROR) { 7863 return; 7864 } 7865 lStatus = init(); 7866 if (lStatus < 0) { 7867 mStatus = lStatus; 7868 goto Error; 7869 } 7870 7871 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7872 mPinned = true; 7873 } 7874 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7875 return; 7876Error: 7877 EffectRelease(mEffectInterface); 7878 mEffectInterface = NULL; 7879 ALOGV("Constructor Error %d", mStatus); 7880} 7881 7882AudioFlinger::EffectModule::~EffectModule() 7883{ 7884 ALOGV("Destructor %p", this); 7885 if (mEffectInterface != NULL) { 7886 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7887 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7888 sp<ThreadBase> thread = mThread.promote(); 7889 if (thread != 0) { 7890 audio_stream_t *stream = thread->stream(); 7891 if (stream != NULL) { 7892 stream->remove_audio_effect(stream, mEffectInterface); 7893 } 7894 } 7895 } 7896 // release effect engine 7897 EffectRelease(mEffectInterface); 7898 } 7899} 7900 7901status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7902{ 7903 status_t status; 7904 7905 Mutex::Autolock _l(mLock); 7906 int priority = handle->priority(); 7907 size_t size = mHandles.size(); 7908 sp<EffectHandle> h; 7909 size_t i; 7910 for (i = 0; i < size; i++) { 7911 h = mHandles[i].promote(); 7912 if (h == 0) continue; 7913 if (h->priority() <= priority) break; 7914 } 7915 // if inserted in first place, move effect control from previous owner to this handle 7916 if (i == 0) { 7917 bool enabled = false; 7918 if (h != 0) { 7919 enabled = h->enabled(); 7920 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7921 } 7922 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7923 status = NO_ERROR; 7924 } else { 7925 status = ALREADY_EXISTS; 7926 } 7927 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7928 mHandles.insertAt(handle, i); 7929 return status; 7930} 7931 7932size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7933{ 7934 Mutex::Autolock _l(mLock); 7935 size_t size = mHandles.size(); 7936 size_t i; 7937 for (i = 0; i < size; i++) { 7938 if (mHandles[i] == handle) break; 7939 } 7940 if (i == size) { 7941 return size; 7942 } 7943 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7944 7945 bool enabled = false; 7946 EffectHandle *hdl = handle.unsafe_get(); 7947 if (hdl != NULL) { 7948 ALOGV("removeHandle() unsafe_get OK"); 7949 enabled = hdl->enabled(); 7950 } 7951 mHandles.removeAt(i); 7952 size = mHandles.size(); 7953 // if removed from first place, move effect control from this handle to next in line 7954 if (i == 0 && size != 0) { 7955 sp<EffectHandle> h = mHandles[0].promote(); 7956 if (h != 0) { 7957 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7958 } 7959 } 7960 7961 // Prevent calls to process() and other functions on effect interface from now on. 7962 // The effect engine will be released by the destructor when the last strong reference on 7963 // this object is released which can happen after next process is called. 7964 if (size == 0 && !mPinned) { 7965 mState = DESTROYED; 7966 } 7967 7968 return size; 7969} 7970 7971sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7972{ 7973 Mutex::Autolock _l(mLock); 7974 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7975} 7976 7977void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7978{ 7979 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7980 // keep a strong reference on this EffectModule to avoid calling the 7981 // destructor before we exit 7982 sp<EffectModule> keep(this); 7983 { 7984 sp<ThreadBase> thread = mThread.promote(); 7985 if (thread != 0) { 7986 thread->disconnectEffect(keep, handle, unpinIfLast); 7987 } 7988 } 7989} 7990 7991void AudioFlinger::EffectModule::updateState() { 7992 Mutex::Autolock _l(mLock); 7993 7994 switch (mState) { 7995 case RESTART: 7996 reset_l(); 7997 // FALL THROUGH 7998 7999 case STARTING: 8000 // clear auxiliary effect input buffer for next accumulation 8001 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8002 memset(mConfig.inputCfg.buffer.raw, 8003 0, 8004 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8005 } 8006 start_l(); 8007 mState = ACTIVE; 8008 break; 8009 case STOPPING: 8010 stop_l(); 8011 mDisableWaitCnt = mMaxDisableWaitCnt; 8012 mState = STOPPED; 8013 break; 8014 case STOPPED: 8015 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8016 // turn off sequence. 8017 if (--mDisableWaitCnt == 0) { 8018 reset_l(); 8019 mState = IDLE; 8020 } 8021 break; 8022 default: //IDLE , ACTIVE, DESTROYED 8023 break; 8024 } 8025} 8026 8027void AudioFlinger::EffectModule::process() 8028{ 8029 Mutex::Autolock _l(mLock); 8030 8031 if (mState == DESTROYED || mEffectInterface == NULL || 8032 mConfig.inputCfg.buffer.raw == NULL || 8033 mConfig.outputCfg.buffer.raw == NULL) { 8034 return; 8035 } 8036 8037 if (isProcessEnabled()) { 8038 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8039 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8040 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8041 mConfig.inputCfg.buffer.s32, 8042 mConfig.inputCfg.buffer.frameCount/2); 8043 } 8044 8045 // do the actual processing in the effect engine 8046 int ret = (*mEffectInterface)->process(mEffectInterface, 8047 &mConfig.inputCfg.buffer, 8048 &mConfig.outputCfg.buffer); 8049 8050 // force transition to IDLE state when engine is ready 8051 if (mState == STOPPED && ret == -ENODATA) { 8052 mDisableWaitCnt = 1; 8053 } 8054 8055 // clear auxiliary effect input buffer for next accumulation 8056 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8057 memset(mConfig.inputCfg.buffer.raw, 0, 8058 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8059 } 8060 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8061 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8062 // If an insert effect is idle and input buffer is different from output buffer, 8063 // accumulate input onto output 8064 sp<EffectChain> chain = mChain.promote(); 8065 if (chain != 0 && chain->activeTrackCnt() != 0) { 8066 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8067 int16_t *in = mConfig.inputCfg.buffer.s16; 8068 int16_t *out = mConfig.outputCfg.buffer.s16; 8069 for (size_t i = 0; i < frameCnt; i++) { 8070 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8071 } 8072 } 8073 } 8074} 8075 8076void AudioFlinger::EffectModule::reset_l() 8077{ 8078 if (mEffectInterface == NULL) { 8079 return; 8080 } 8081 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8082} 8083 8084status_t AudioFlinger::EffectModule::configure() 8085{ 8086 uint32_t channels; 8087 if (mEffectInterface == NULL) { 8088 return NO_INIT; 8089 } 8090 8091 sp<ThreadBase> thread = mThread.promote(); 8092 if (thread == 0) { 8093 return DEAD_OBJECT; 8094 } 8095 8096 // TODO: handle configuration of effects replacing track process 8097 if (thread->channelCount() == 1) { 8098 channels = AUDIO_CHANNEL_OUT_MONO; 8099 } else { 8100 channels = AUDIO_CHANNEL_OUT_STEREO; 8101 } 8102 8103 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8104 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8105 } else { 8106 mConfig.inputCfg.channels = channels; 8107 } 8108 mConfig.outputCfg.channels = channels; 8109 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8110 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8111 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8112 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8113 mConfig.inputCfg.bufferProvider.cookie = NULL; 8114 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8115 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8116 mConfig.outputCfg.bufferProvider.cookie = NULL; 8117 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8118 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8119 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8120 // Insert effect: 8121 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8122 // always overwrites output buffer: input buffer == output buffer 8123 // - in other sessions: 8124 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8125 // other effect: overwrites output buffer: input buffer == output buffer 8126 // Auxiliary effect: 8127 // accumulates in output buffer: input buffer != output buffer 8128 // Therefore: accumulate <=> input buffer != output buffer 8129 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8130 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8131 } else { 8132 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8133 } 8134 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8135 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8136 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8137 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8138 8139 ALOGV("configure() %p thread %p buffer %p framecount %d", 8140 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8141 8142 status_t cmdStatus; 8143 uint32_t size = sizeof(int); 8144 status_t status = (*mEffectInterface)->command(mEffectInterface, 8145 EFFECT_CMD_SET_CONFIG, 8146 sizeof(effect_config_t), 8147 &mConfig, 8148 &size, 8149 &cmdStatus); 8150 if (status == 0) { 8151 status = cmdStatus; 8152 } 8153 8154 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8155 (1000 * mConfig.outputCfg.buffer.frameCount); 8156 8157 return status; 8158} 8159 8160status_t AudioFlinger::EffectModule::init() 8161{ 8162 Mutex::Autolock _l(mLock); 8163 if (mEffectInterface == NULL) { 8164 return NO_INIT; 8165 } 8166 status_t cmdStatus; 8167 uint32_t size = sizeof(status_t); 8168 status_t status = (*mEffectInterface)->command(mEffectInterface, 8169 EFFECT_CMD_INIT, 8170 0, 8171 NULL, 8172 &size, 8173 &cmdStatus); 8174 if (status == 0) { 8175 status = cmdStatus; 8176 } 8177 return status; 8178} 8179 8180status_t AudioFlinger::EffectModule::start() 8181{ 8182 Mutex::Autolock _l(mLock); 8183 return start_l(); 8184} 8185 8186status_t AudioFlinger::EffectModule::start_l() 8187{ 8188 if (mEffectInterface == NULL) { 8189 return NO_INIT; 8190 } 8191 status_t cmdStatus; 8192 uint32_t size = sizeof(status_t); 8193 status_t status = (*mEffectInterface)->command(mEffectInterface, 8194 EFFECT_CMD_ENABLE, 8195 0, 8196 NULL, 8197 &size, 8198 &cmdStatus); 8199 if (status == 0) { 8200 status = cmdStatus; 8201 } 8202 if (status == 0 && 8203 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8204 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8205 sp<ThreadBase> thread = mThread.promote(); 8206 if (thread != 0) { 8207 audio_stream_t *stream = thread->stream(); 8208 if (stream != NULL) { 8209 stream->add_audio_effect(stream, mEffectInterface); 8210 } 8211 } 8212 } 8213 return status; 8214} 8215 8216status_t AudioFlinger::EffectModule::stop() 8217{ 8218 Mutex::Autolock _l(mLock); 8219 return stop_l(); 8220} 8221 8222status_t AudioFlinger::EffectModule::stop_l() 8223{ 8224 if (mEffectInterface == NULL) { 8225 return NO_INIT; 8226 } 8227 status_t cmdStatus; 8228 uint32_t size = sizeof(status_t); 8229 status_t status = (*mEffectInterface)->command(mEffectInterface, 8230 EFFECT_CMD_DISABLE, 8231 0, 8232 NULL, 8233 &size, 8234 &cmdStatus); 8235 if (status == 0) { 8236 status = cmdStatus; 8237 } 8238 if (status == 0 && 8239 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8240 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8241 sp<ThreadBase> thread = mThread.promote(); 8242 if (thread != 0) { 8243 audio_stream_t *stream = thread->stream(); 8244 if (stream != NULL) { 8245 stream->remove_audio_effect(stream, mEffectInterface); 8246 } 8247 } 8248 } 8249 return status; 8250} 8251 8252status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8253 uint32_t cmdSize, 8254 void *pCmdData, 8255 uint32_t *replySize, 8256 void *pReplyData) 8257{ 8258 Mutex::Autolock _l(mLock); 8259// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8260 8261 if (mState == DESTROYED || mEffectInterface == NULL) { 8262 return NO_INIT; 8263 } 8264 status_t status = (*mEffectInterface)->command(mEffectInterface, 8265 cmdCode, 8266 cmdSize, 8267 pCmdData, 8268 replySize, 8269 pReplyData); 8270 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8271 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8272 for (size_t i = 1; i < mHandles.size(); i++) { 8273 sp<EffectHandle> h = mHandles[i].promote(); 8274 if (h != 0) { 8275 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8276 } 8277 } 8278 } 8279 return status; 8280} 8281 8282status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8283{ 8284 8285 Mutex::Autolock _l(mLock); 8286 ALOGV("setEnabled %p enabled %d", this, enabled); 8287 8288 if (enabled != isEnabled()) { 8289 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8290 if (enabled && status != NO_ERROR) { 8291 return status; 8292 } 8293 8294 switch (mState) { 8295 // going from disabled to enabled 8296 case IDLE: 8297 mState = STARTING; 8298 break; 8299 case STOPPED: 8300 mState = RESTART; 8301 break; 8302 case STOPPING: 8303 mState = ACTIVE; 8304 break; 8305 8306 // going from enabled to disabled 8307 case RESTART: 8308 mState = STOPPED; 8309 break; 8310 case STARTING: 8311 mState = IDLE; 8312 break; 8313 case ACTIVE: 8314 mState = STOPPING; 8315 break; 8316 case DESTROYED: 8317 return NO_ERROR; // simply ignore as we are being destroyed 8318 } 8319 for (size_t i = 1; i < mHandles.size(); i++) { 8320 sp<EffectHandle> h = mHandles[i].promote(); 8321 if (h != 0) { 8322 h->setEnabled(enabled); 8323 } 8324 } 8325 } 8326 return NO_ERROR; 8327} 8328 8329bool AudioFlinger::EffectModule::isEnabled() const 8330{ 8331 switch (mState) { 8332 case RESTART: 8333 case STARTING: 8334 case ACTIVE: 8335 return true; 8336 case IDLE: 8337 case STOPPING: 8338 case STOPPED: 8339 case DESTROYED: 8340 default: 8341 return false; 8342 } 8343} 8344 8345bool AudioFlinger::EffectModule::isProcessEnabled() const 8346{ 8347 switch (mState) { 8348 case RESTART: 8349 case ACTIVE: 8350 case STOPPING: 8351 case STOPPED: 8352 return true; 8353 case IDLE: 8354 case STARTING: 8355 case DESTROYED: 8356 default: 8357 return false; 8358 } 8359} 8360 8361status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8362{ 8363 Mutex::Autolock _l(mLock); 8364 status_t status = NO_ERROR; 8365 8366 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8367 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8368 if (isProcessEnabled() && 8369 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8370 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8371 status_t cmdStatus; 8372 uint32_t volume[2]; 8373 uint32_t *pVolume = NULL; 8374 uint32_t size = sizeof(volume); 8375 volume[0] = *left; 8376 volume[1] = *right; 8377 if (controller) { 8378 pVolume = volume; 8379 } 8380 status = (*mEffectInterface)->command(mEffectInterface, 8381 EFFECT_CMD_SET_VOLUME, 8382 size, 8383 volume, 8384 &size, 8385 pVolume); 8386 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8387 *left = volume[0]; 8388 *right = volume[1]; 8389 } 8390 } 8391 return status; 8392} 8393 8394status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8395{ 8396 Mutex::Autolock _l(mLock); 8397 status_t status = NO_ERROR; 8398 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8399 // audio pre processing modules on RecordThread can receive both output and 8400 // input device indication in the same call 8401 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8402 if (dev) { 8403 status_t cmdStatus; 8404 uint32_t size = sizeof(status_t); 8405 8406 status = (*mEffectInterface)->command(mEffectInterface, 8407 EFFECT_CMD_SET_DEVICE, 8408 sizeof(uint32_t), 8409 &dev, 8410 &size, 8411 &cmdStatus); 8412 if (status == NO_ERROR) { 8413 status = cmdStatus; 8414 } 8415 } 8416 dev = device & AUDIO_DEVICE_IN_ALL; 8417 if (dev) { 8418 status_t cmdStatus; 8419 uint32_t size = sizeof(status_t); 8420 8421 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8422 EFFECT_CMD_SET_INPUT_DEVICE, 8423 sizeof(uint32_t), 8424 &dev, 8425 &size, 8426 &cmdStatus); 8427 if (status2 == NO_ERROR) { 8428 status2 = cmdStatus; 8429 } 8430 if (status == NO_ERROR) { 8431 status = status2; 8432 } 8433 } 8434 } 8435 return status; 8436} 8437 8438status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8439{ 8440 Mutex::Autolock _l(mLock); 8441 status_t status = NO_ERROR; 8442 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8443 status_t cmdStatus; 8444 uint32_t size = sizeof(status_t); 8445 status = (*mEffectInterface)->command(mEffectInterface, 8446 EFFECT_CMD_SET_AUDIO_MODE, 8447 sizeof(audio_mode_t), 8448 &mode, 8449 &size, 8450 &cmdStatus); 8451 if (status == NO_ERROR) { 8452 status = cmdStatus; 8453 } 8454 } 8455 return status; 8456} 8457 8458void AudioFlinger::EffectModule::setSuspended(bool suspended) 8459{ 8460 Mutex::Autolock _l(mLock); 8461 mSuspended = suspended; 8462} 8463 8464bool AudioFlinger::EffectModule::suspended() const 8465{ 8466 Mutex::Autolock _l(mLock); 8467 return mSuspended; 8468} 8469 8470status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8471{ 8472 const size_t SIZE = 256; 8473 char buffer[SIZE]; 8474 String8 result; 8475 8476 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8477 result.append(buffer); 8478 8479 bool locked = tryLock(mLock); 8480 // failed to lock - AudioFlinger is probably deadlocked 8481 if (!locked) { 8482 result.append("\t\tCould not lock Fx mutex:\n"); 8483 } 8484 8485 result.append("\t\tSession Status State Engine:\n"); 8486 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8487 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8488 result.append(buffer); 8489 8490 result.append("\t\tDescriptor:\n"); 8491 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8492 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8493 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8494 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8495 result.append(buffer); 8496 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8497 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8498 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8499 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8500 result.append(buffer); 8501 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8502 mDescriptor.apiVersion, 8503 mDescriptor.flags); 8504 result.append(buffer); 8505 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8506 mDescriptor.name); 8507 result.append(buffer); 8508 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8509 mDescriptor.implementor); 8510 result.append(buffer); 8511 8512 result.append("\t\t- Input configuration:\n"); 8513 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8514 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8515 (uint32_t)mConfig.inputCfg.buffer.raw, 8516 mConfig.inputCfg.buffer.frameCount, 8517 mConfig.inputCfg.samplingRate, 8518 mConfig.inputCfg.channels, 8519 mConfig.inputCfg.format); 8520 result.append(buffer); 8521 8522 result.append("\t\t- Output configuration:\n"); 8523 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8524 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8525 (uint32_t)mConfig.outputCfg.buffer.raw, 8526 mConfig.outputCfg.buffer.frameCount, 8527 mConfig.outputCfg.samplingRate, 8528 mConfig.outputCfg.channels, 8529 mConfig.outputCfg.format); 8530 result.append(buffer); 8531 8532 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8533 result.append(buffer); 8534 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8535 for (size_t i = 0; i < mHandles.size(); ++i) { 8536 sp<EffectHandle> handle = mHandles[i].promote(); 8537 if (handle != 0) { 8538 handle->dump(buffer, SIZE); 8539 result.append(buffer); 8540 } 8541 } 8542 8543 result.append("\n"); 8544 8545 write(fd, result.string(), result.length()); 8546 8547 if (locked) { 8548 mLock.unlock(); 8549 } 8550 8551 return NO_ERROR; 8552} 8553 8554// ---------------------------------------------------------------------------- 8555// EffectHandle implementation 8556// ---------------------------------------------------------------------------- 8557 8558#undef LOG_TAG 8559#define LOG_TAG "AudioFlinger::EffectHandle" 8560 8561AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8562 const sp<AudioFlinger::Client>& client, 8563 const sp<IEffectClient>& effectClient, 8564 int32_t priority) 8565 : BnEffect(), 8566 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8567 mPriority(priority), mHasControl(false), mEnabled(false) 8568{ 8569 ALOGV("constructor %p", this); 8570 8571 if (client == 0) { 8572 return; 8573 } 8574 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8575 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8576 if (mCblkMemory != 0) { 8577 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8578 8579 if (mCblk != NULL) { 8580 new(mCblk) effect_param_cblk_t(); 8581 mBuffer = (uint8_t *)mCblk + bufOffset; 8582 } 8583 } else { 8584 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8585 return; 8586 } 8587} 8588 8589AudioFlinger::EffectHandle::~EffectHandle() 8590{ 8591 ALOGV("Destructor %p", this); 8592 disconnect(false); 8593 ALOGV("Destructor DONE %p", this); 8594} 8595 8596status_t AudioFlinger::EffectHandle::enable() 8597{ 8598 ALOGV("enable %p", this); 8599 if (!mHasControl) return INVALID_OPERATION; 8600 if (mEffect == 0) return DEAD_OBJECT; 8601 8602 if (mEnabled) { 8603 return NO_ERROR; 8604 } 8605 8606 mEnabled = true; 8607 8608 sp<ThreadBase> thread = mEffect->thread().promote(); 8609 if (thread != 0) { 8610 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8611 } 8612 8613 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8614 if (mEffect->suspended()) { 8615 return NO_ERROR; 8616 } 8617 8618 status_t status = mEffect->setEnabled(true); 8619 if (status != NO_ERROR) { 8620 if (thread != 0) { 8621 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8622 } 8623 mEnabled = false; 8624 } 8625 return status; 8626} 8627 8628status_t AudioFlinger::EffectHandle::disable() 8629{ 8630 ALOGV("disable %p", this); 8631 if (!mHasControl) return INVALID_OPERATION; 8632 if (mEffect == 0) return DEAD_OBJECT; 8633 8634 if (!mEnabled) { 8635 return NO_ERROR; 8636 } 8637 mEnabled = false; 8638 8639 if (mEffect->suspended()) { 8640 return NO_ERROR; 8641 } 8642 8643 status_t status = mEffect->setEnabled(false); 8644 8645 sp<ThreadBase> thread = mEffect->thread().promote(); 8646 if (thread != 0) { 8647 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8648 } 8649 8650 return status; 8651} 8652 8653void AudioFlinger::EffectHandle::disconnect() 8654{ 8655 disconnect(true); 8656} 8657 8658void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8659{ 8660 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8661 if (mEffect == 0) { 8662 return; 8663 } 8664 mEffect->disconnect(this, unpinIfLast); 8665 8666 if (mHasControl && mEnabled) { 8667 sp<ThreadBase> thread = mEffect->thread().promote(); 8668 if (thread != 0) { 8669 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8670 } 8671 } 8672 8673 // release sp on module => module destructor can be called now 8674 mEffect.clear(); 8675 if (mClient != 0) { 8676 if (mCblk != NULL) { 8677 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8678 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8679 } 8680 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8681 // Client destructor must run with AudioFlinger mutex locked 8682 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8683 mClient.clear(); 8684 } 8685} 8686 8687status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8688 uint32_t cmdSize, 8689 void *pCmdData, 8690 uint32_t *replySize, 8691 void *pReplyData) 8692{ 8693// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8694// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8695 8696 // only get parameter command is permitted for applications not controlling the effect 8697 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8698 return INVALID_OPERATION; 8699 } 8700 if (mEffect == 0) return DEAD_OBJECT; 8701 if (mClient == 0) return INVALID_OPERATION; 8702 8703 // handle commands that are not forwarded transparently to effect engine 8704 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8705 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8706 // no risk to block the whole media server process or mixer threads is we are stuck here 8707 Mutex::Autolock _l(mCblk->lock); 8708 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8709 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8710 mCblk->serverIndex = 0; 8711 mCblk->clientIndex = 0; 8712 return BAD_VALUE; 8713 } 8714 status_t status = NO_ERROR; 8715 while (mCblk->serverIndex < mCblk->clientIndex) { 8716 int reply; 8717 uint32_t rsize = sizeof(int); 8718 int *p = (int *)(mBuffer + mCblk->serverIndex); 8719 int size = *p++; 8720 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8721 ALOGW("command(): invalid parameter block size"); 8722 break; 8723 } 8724 effect_param_t *param = (effect_param_t *)p; 8725 if (param->psize == 0 || param->vsize == 0) { 8726 ALOGW("command(): null parameter or value size"); 8727 mCblk->serverIndex += size; 8728 continue; 8729 } 8730 uint32_t psize = sizeof(effect_param_t) + 8731 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8732 param->vsize; 8733 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8734 psize, 8735 p, 8736 &rsize, 8737 &reply); 8738 // stop at first error encountered 8739 if (ret != NO_ERROR) { 8740 status = ret; 8741 *(int *)pReplyData = reply; 8742 break; 8743 } else if (reply != NO_ERROR) { 8744 *(int *)pReplyData = reply; 8745 break; 8746 } 8747 mCblk->serverIndex += size; 8748 } 8749 mCblk->serverIndex = 0; 8750 mCblk->clientIndex = 0; 8751 return status; 8752 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8753 *(int *)pReplyData = NO_ERROR; 8754 return enable(); 8755 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8756 *(int *)pReplyData = NO_ERROR; 8757 return disable(); 8758 } 8759 8760 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8761} 8762 8763void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8764{ 8765 ALOGV("setControl %p control %d", this, hasControl); 8766 8767 mHasControl = hasControl; 8768 mEnabled = enabled; 8769 8770 if (signal && mEffectClient != 0) { 8771 mEffectClient->controlStatusChanged(hasControl); 8772 } 8773} 8774 8775void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8776 uint32_t cmdSize, 8777 void *pCmdData, 8778 uint32_t replySize, 8779 void *pReplyData) 8780{ 8781 if (mEffectClient != 0) { 8782 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8783 } 8784} 8785 8786 8787 8788void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8789{ 8790 if (mEffectClient != 0) { 8791 mEffectClient->enableStatusChanged(enabled); 8792 } 8793} 8794 8795status_t AudioFlinger::EffectHandle::onTransact( 8796 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8797{ 8798 return BnEffect::onTransact(code, data, reply, flags); 8799} 8800 8801 8802void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8803{ 8804 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8805 8806 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8807 (mClient == 0) ? getpid_cached : mClient->pid(), 8808 mPriority, 8809 mHasControl, 8810 !locked, 8811 mCblk ? mCblk->clientIndex : 0, 8812 mCblk ? mCblk->serverIndex : 0 8813 ); 8814 8815 if (locked) { 8816 mCblk->lock.unlock(); 8817 } 8818} 8819 8820#undef LOG_TAG 8821#define LOG_TAG "AudioFlinger::EffectChain" 8822 8823AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8824 int sessionId) 8825 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8826 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8827 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8828{ 8829 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8830 if (thread == NULL) { 8831 return; 8832 } 8833 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8834 thread->frameCount(); 8835} 8836 8837AudioFlinger::EffectChain::~EffectChain() 8838{ 8839 if (mOwnInBuffer) { 8840 delete mInBuffer; 8841 } 8842 8843} 8844 8845// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8846sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8847{ 8848 size_t size = mEffects.size(); 8849 8850 for (size_t i = 0; i < size; i++) { 8851 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8852 return mEffects[i]; 8853 } 8854 } 8855 return 0; 8856} 8857 8858// getEffectFromId_l() must be called with ThreadBase::mLock held 8859sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8860{ 8861 size_t size = mEffects.size(); 8862 8863 for (size_t i = 0; i < size; i++) { 8864 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8865 if (id == 0 || mEffects[i]->id() == id) { 8866 return mEffects[i]; 8867 } 8868 } 8869 return 0; 8870} 8871 8872// getEffectFromType_l() must be called with ThreadBase::mLock held 8873sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8874 const effect_uuid_t *type) 8875{ 8876 size_t size = mEffects.size(); 8877 8878 for (size_t i = 0; i < size; i++) { 8879 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8880 return mEffects[i]; 8881 } 8882 } 8883 return 0; 8884} 8885 8886// Must be called with EffectChain::mLock locked 8887void AudioFlinger::EffectChain::process_l() 8888{ 8889 sp<ThreadBase> thread = mThread.promote(); 8890 if (thread == 0) { 8891 ALOGW("process_l(): cannot promote mixer thread"); 8892 return; 8893 } 8894 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8895 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8896 // always process effects unless no more tracks are on the session and the effect tail 8897 // has been rendered 8898 bool doProcess = true; 8899 if (!isGlobalSession) { 8900 bool tracksOnSession = (trackCnt() != 0); 8901 8902 if (!tracksOnSession && mTailBufferCount == 0) { 8903 doProcess = false; 8904 } 8905 8906 if (activeTrackCnt() == 0) { 8907 // if no track is active and the effect tail has not been rendered, 8908 // the input buffer must be cleared here as the mixer process will not do it 8909 if (tracksOnSession || mTailBufferCount > 0) { 8910 size_t numSamples = thread->frameCount() * thread->channelCount(); 8911 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8912 if (mTailBufferCount > 0) { 8913 mTailBufferCount--; 8914 } 8915 } 8916 } 8917 } 8918 8919 size_t size = mEffects.size(); 8920 if (doProcess) { 8921 for (size_t i = 0; i < size; i++) { 8922 mEffects[i]->process(); 8923 } 8924 } 8925 for (size_t i = 0; i < size; i++) { 8926 mEffects[i]->updateState(); 8927 } 8928} 8929 8930// addEffect_l() must be called with PlaybackThread::mLock held 8931status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8932{ 8933 effect_descriptor_t desc = effect->desc(); 8934 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8935 8936 Mutex::Autolock _l(mLock); 8937 effect->setChain(this); 8938 sp<ThreadBase> thread = mThread.promote(); 8939 if (thread == 0) { 8940 return NO_INIT; 8941 } 8942 effect->setThread(thread); 8943 8944 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8945 // Auxiliary effects are inserted at the beginning of mEffects vector as 8946 // they are processed first and accumulated in chain input buffer 8947 mEffects.insertAt(effect, 0); 8948 8949 // the input buffer for auxiliary effect contains mono samples in 8950 // 32 bit format. This is to avoid saturation in AudoMixer 8951 // accumulation stage. Saturation is done in EffectModule::process() before 8952 // calling the process in effect engine 8953 size_t numSamples = thread->frameCount(); 8954 int32_t *buffer = new int32_t[numSamples]; 8955 memset(buffer, 0, numSamples * sizeof(int32_t)); 8956 effect->setInBuffer((int16_t *)buffer); 8957 // auxiliary effects output samples to chain input buffer for further processing 8958 // by insert effects 8959 effect->setOutBuffer(mInBuffer); 8960 } else { 8961 // Insert effects are inserted at the end of mEffects vector as they are processed 8962 // after track and auxiliary effects. 8963 // Insert effect order as a function of indicated preference: 8964 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8965 // another effect is present 8966 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8967 // last effect claiming first position 8968 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8969 // first effect claiming last position 8970 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8971 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8972 // already present 8973 8974 size_t size = mEffects.size(); 8975 size_t idx_insert = size; 8976 ssize_t idx_insert_first = -1; 8977 ssize_t idx_insert_last = -1; 8978 8979 for (size_t i = 0; i < size; i++) { 8980 effect_descriptor_t d = mEffects[i]->desc(); 8981 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8982 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8983 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8984 // check invalid effect chaining combinations 8985 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8986 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8987 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8988 return INVALID_OPERATION; 8989 } 8990 // remember position of first insert effect and by default 8991 // select this as insert position for new effect 8992 if (idx_insert == size) { 8993 idx_insert = i; 8994 } 8995 // remember position of last insert effect claiming 8996 // first position 8997 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8998 idx_insert_first = i; 8999 } 9000 // remember position of first insert effect claiming 9001 // last position 9002 if (iPref == EFFECT_FLAG_INSERT_LAST && 9003 idx_insert_last == -1) { 9004 idx_insert_last = i; 9005 } 9006 } 9007 } 9008 9009 // modify idx_insert from first position if needed 9010 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9011 if (idx_insert_last != -1) { 9012 idx_insert = idx_insert_last; 9013 } else { 9014 idx_insert = size; 9015 } 9016 } else { 9017 if (idx_insert_first != -1) { 9018 idx_insert = idx_insert_first + 1; 9019 } 9020 } 9021 9022 // always read samples from chain input buffer 9023 effect->setInBuffer(mInBuffer); 9024 9025 // if last effect in the chain, output samples to chain 9026 // output buffer, otherwise to chain input buffer 9027 if (idx_insert == size) { 9028 if (idx_insert != 0) { 9029 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9030 mEffects[idx_insert-1]->configure(); 9031 } 9032 effect->setOutBuffer(mOutBuffer); 9033 } else { 9034 effect->setOutBuffer(mInBuffer); 9035 } 9036 mEffects.insertAt(effect, idx_insert); 9037 9038 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9039 } 9040 effect->configure(); 9041 return NO_ERROR; 9042} 9043 9044// removeEffect_l() must be called with PlaybackThread::mLock held 9045size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9046{ 9047 Mutex::Autolock _l(mLock); 9048 size_t size = mEffects.size(); 9049 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9050 9051 for (size_t i = 0; i < size; i++) { 9052 if (effect == mEffects[i]) { 9053 // calling stop here will remove pre-processing effect from the audio HAL. 9054 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9055 // the middle of a read from audio HAL 9056 if (mEffects[i]->state() == EffectModule::ACTIVE || 9057 mEffects[i]->state() == EffectModule::STOPPING) { 9058 mEffects[i]->stop(); 9059 } 9060 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9061 delete[] effect->inBuffer(); 9062 } else { 9063 if (i == size - 1 && i != 0) { 9064 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9065 mEffects[i - 1]->configure(); 9066 } 9067 } 9068 mEffects.removeAt(i); 9069 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9070 break; 9071 } 9072 } 9073 9074 return mEffects.size(); 9075} 9076 9077// setDevice_l() must be called with PlaybackThread::mLock held 9078void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9079{ 9080 size_t size = mEffects.size(); 9081 for (size_t i = 0; i < size; i++) { 9082 mEffects[i]->setDevice(device); 9083 } 9084} 9085 9086// setMode_l() must be called with PlaybackThread::mLock held 9087void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9088{ 9089 size_t size = mEffects.size(); 9090 for (size_t i = 0; i < size; i++) { 9091 mEffects[i]->setMode(mode); 9092 } 9093} 9094 9095// setVolume_l() must be called with PlaybackThread::mLock held 9096bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9097{ 9098 uint32_t newLeft = *left; 9099 uint32_t newRight = *right; 9100 bool hasControl = false; 9101 int ctrlIdx = -1; 9102 size_t size = mEffects.size(); 9103 9104 // first update volume controller 9105 for (size_t i = size; i > 0; i--) { 9106 if (mEffects[i - 1]->isProcessEnabled() && 9107 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9108 ctrlIdx = i - 1; 9109 hasControl = true; 9110 break; 9111 } 9112 } 9113 9114 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9115 if (hasControl) { 9116 *left = mNewLeftVolume; 9117 *right = mNewRightVolume; 9118 } 9119 return hasControl; 9120 } 9121 9122 mVolumeCtrlIdx = ctrlIdx; 9123 mLeftVolume = newLeft; 9124 mRightVolume = newRight; 9125 9126 // second get volume update from volume controller 9127 if (ctrlIdx >= 0) { 9128 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9129 mNewLeftVolume = newLeft; 9130 mNewRightVolume = newRight; 9131 } 9132 // then indicate volume to all other effects in chain. 9133 // Pass altered volume to effects before volume controller 9134 // and requested volume to effects after controller 9135 uint32_t lVol = newLeft; 9136 uint32_t rVol = newRight; 9137 9138 for (size_t i = 0; i < size; i++) { 9139 if ((int)i == ctrlIdx) continue; 9140 // this also works for ctrlIdx == -1 when there is no volume controller 9141 if ((int)i > ctrlIdx) { 9142 lVol = *left; 9143 rVol = *right; 9144 } 9145 mEffects[i]->setVolume(&lVol, &rVol, false); 9146 } 9147 *left = newLeft; 9148 *right = newRight; 9149 9150 return hasControl; 9151} 9152 9153status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9154{ 9155 const size_t SIZE = 256; 9156 char buffer[SIZE]; 9157 String8 result; 9158 9159 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9160 result.append(buffer); 9161 9162 bool locked = tryLock(mLock); 9163 // failed to lock - AudioFlinger is probably deadlocked 9164 if (!locked) { 9165 result.append("\tCould not lock mutex:\n"); 9166 } 9167 9168 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9169 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9170 mEffects.size(), 9171 (uint32_t)mInBuffer, 9172 (uint32_t)mOutBuffer, 9173 mActiveTrackCnt); 9174 result.append(buffer); 9175 write(fd, result.string(), result.size()); 9176 9177 for (size_t i = 0; i < mEffects.size(); ++i) { 9178 sp<EffectModule> effect = mEffects[i]; 9179 if (effect != 0) { 9180 effect->dump(fd, args); 9181 } 9182 } 9183 9184 if (locked) { 9185 mLock.unlock(); 9186 } 9187 9188 return NO_ERROR; 9189} 9190 9191// must be called with ThreadBase::mLock held 9192void AudioFlinger::EffectChain::setEffectSuspended_l( 9193 const effect_uuid_t *type, bool suspend) 9194{ 9195 sp<SuspendedEffectDesc> desc; 9196 // use effect type UUID timelow as key as there is no real risk of identical 9197 // timeLow fields among effect type UUIDs. 9198 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9199 if (suspend) { 9200 if (index >= 0) { 9201 desc = mSuspendedEffects.valueAt(index); 9202 } else { 9203 desc = new SuspendedEffectDesc(); 9204 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9205 mSuspendedEffects.add(type->timeLow, desc); 9206 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9207 } 9208 if (desc->mRefCount++ == 0) { 9209 sp<EffectModule> effect = getEffectIfEnabled(type); 9210 if (effect != 0) { 9211 desc->mEffect = effect; 9212 effect->setSuspended(true); 9213 effect->setEnabled(false); 9214 } 9215 } 9216 } else { 9217 if (index < 0) { 9218 return; 9219 } 9220 desc = mSuspendedEffects.valueAt(index); 9221 if (desc->mRefCount <= 0) { 9222 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9223 desc->mRefCount = 1; 9224 } 9225 if (--desc->mRefCount == 0) { 9226 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9227 if (desc->mEffect != 0) { 9228 sp<EffectModule> effect = desc->mEffect.promote(); 9229 if (effect != 0) { 9230 effect->setSuspended(false); 9231 sp<EffectHandle> handle = effect->controlHandle(); 9232 if (handle != 0) { 9233 effect->setEnabled(handle->enabled()); 9234 } 9235 } 9236 desc->mEffect.clear(); 9237 } 9238 mSuspendedEffects.removeItemsAt(index); 9239 } 9240 } 9241} 9242 9243// must be called with ThreadBase::mLock held 9244void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9245{ 9246 sp<SuspendedEffectDesc> desc; 9247 9248 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9249 if (suspend) { 9250 if (index >= 0) { 9251 desc = mSuspendedEffects.valueAt(index); 9252 } else { 9253 desc = new SuspendedEffectDesc(); 9254 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9255 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9256 } 9257 if (desc->mRefCount++ == 0) { 9258 Vector< sp<EffectModule> > effects; 9259 getSuspendEligibleEffects(effects); 9260 for (size_t i = 0; i < effects.size(); i++) { 9261 setEffectSuspended_l(&effects[i]->desc().type, true); 9262 } 9263 } 9264 } else { 9265 if (index < 0) { 9266 return; 9267 } 9268 desc = mSuspendedEffects.valueAt(index); 9269 if (desc->mRefCount <= 0) { 9270 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9271 desc->mRefCount = 1; 9272 } 9273 if (--desc->mRefCount == 0) { 9274 Vector<const effect_uuid_t *> types; 9275 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9276 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9277 continue; 9278 } 9279 types.add(&mSuspendedEffects.valueAt(i)->mType); 9280 } 9281 for (size_t i = 0; i < types.size(); i++) { 9282 setEffectSuspended_l(types[i], false); 9283 } 9284 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9285 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9286 } 9287 } 9288} 9289 9290 9291// The volume effect is used for automated tests only 9292#ifndef OPENSL_ES_H_ 9293static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9294 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9295const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9296#endif //OPENSL_ES_H_ 9297 9298bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9299{ 9300 // auxiliary effects and visualizer are never suspended on output mix 9301 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9302 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9303 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9304 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9305 return false; 9306 } 9307 return true; 9308} 9309 9310void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9311{ 9312 effects.clear(); 9313 for (size_t i = 0; i < mEffects.size(); i++) { 9314 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9315 effects.add(mEffects[i]); 9316 } 9317 } 9318} 9319 9320sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9321 const effect_uuid_t *type) 9322{ 9323 sp<EffectModule> effect = getEffectFromType_l(type); 9324 return effect != 0 && effect->isEnabled() ? effect : 0; 9325} 9326 9327void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9328 bool enabled) 9329{ 9330 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9331 if (enabled) { 9332 if (index < 0) { 9333 // if the effect is not suspend check if all effects are suspended 9334 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9335 if (index < 0) { 9336 return; 9337 } 9338 if (!isEffectEligibleForSuspend(effect->desc())) { 9339 return; 9340 } 9341 setEffectSuspended_l(&effect->desc().type, enabled); 9342 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9343 if (index < 0) { 9344 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9345 return; 9346 } 9347 } 9348 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9349 effect->desc().type.timeLow); 9350 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9351 // if effect is requested to suspended but was not yet enabled, supend it now. 9352 if (desc->mEffect == 0) { 9353 desc->mEffect = effect; 9354 effect->setEnabled(false); 9355 effect->setSuspended(true); 9356 } 9357 } else { 9358 if (index < 0) { 9359 return; 9360 } 9361 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9362 effect->desc().type.timeLow); 9363 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9364 desc->mEffect.clear(); 9365 effect->setSuspended(false); 9366 } 9367} 9368 9369#undef LOG_TAG 9370#define LOG_TAG "AudioFlinger" 9371 9372// ---------------------------------------------------------------------------- 9373 9374status_t AudioFlinger::onTransact( 9375 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9376{ 9377 return BnAudioFlinger::onTransact(code, data, reply, flags); 9378} 9379 9380}; // namespace android 9381