AudioFlinger.cpp revision fbae5dae5187aca9d974cbe15ec818e9c6f56705
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.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 1583 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1584 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1585 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1586 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1587 1588 return NO_ERROR; 1589} 1590 1591status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1592{ 1593 const size_t SIZE = 256; 1594 char buffer[SIZE]; 1595 String8 result; 1596 1597 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1606 result.append(buffer); 1607 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1608 result.append(buffer); 1609 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1610 result.append(buffer); 1611 write(fd, result.string(), result.size()); 1612 1613 dumpBase(fd, args); 1614 1615 return NO_ERROR; 1616} 1617 1618// Thread virtuals 1619status_t AudioFlinger::PlaybackThread::readyToRun() 1620{ 1621 status_t status = initCheck(); 1622 if (status == NO_ERROR) { 1623 ALOGI("AudioFlinger's thread %p ready to run", this); 1624 } else { 1625 ALOGE("No working audio driver found."); 1626 } 1627 return status; 1628} 1629 1630void AudioFlinger::PlaybackThread::onFirstRef() 1631{ 1632 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1633} 1634 1635// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1636sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1637 const sp<AudioFlinger::Client>& client, 1638 audio_stream_type_t streamType, 1639 uint32_t sampleRate, 1640 audio_format_t format, 1641 uint32_t channelMask, 1642 int frameCount, 1643 const sp<IMemory>& sharedBuffer, 1644 int sessionId, 1645 IAudioFlinger::track_flags_t flags, 1646 pid_t tid, 1647 status_t *status) 1648{ 1649 sp<Track> track; 1650 status_t lStatus; 1651 1652 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1653 1654 // client expresses a preference for FAST, but we get the final say 1655 if (flags & IAudioFlinger::TRACK_FAST) { 1656 if ( 1657 // not timed 1658 (!isTimed) && 1659 // either of these use cases: 1660 ( 1661 // use case 1: shared buffer with any frame count 1662 ( 1663 (sharedBuffer != 0) 1664 ) || 1665 // use case 2: callback handler and frame count is default or at least as large as HAL 1666 ( 1667 (tid != -1) && 1668 ((frameCount == 0) || 1669 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1670 ) 1671 ) && 1672 // PCM data 1673 audio_is_linear_pcm(format) && 1674 // mono or stereo 1675 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1676 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1677#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1678 // hardware sample rate 1679 (sampleRate == mSampleRate) && 1680#endif 1681 // normal mixer has an associated fast mixer 1682 hasFastMixer() && 1683 // there are sufficient fast track slots available 1684 (mFastTrackAvailMask != 0) 1685 // FIXME test that MixerThread for this fast track has a capable output HAL 1686 // FIXME add a permission test also? 1687 ) { 1688 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1689 if (frameCount == 0) { 1690 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1691 } 1692 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1693 frameCount, mFrameCount); 1694 } else { 1695 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1696 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1697 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1698 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1699 audio_is_linear_pcm(format), 1700 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1701 flags &= ~IAudioFlinger::TRACK_FAST; 1702 // For compatibility with AudioTrack calculation, buffer depth is forced 1703 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1704 // This is probably too conservative, but legacy application code may depend on it. 1705 // If you change this calculation, also review the start threshold which is related. 1706 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1707 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1708 if (minBufCount < 2) { 1709 minBufCount = 2; 1710 } 1711 int minFrameCount = mNormalFrameCount * minBufCount; 1712 if (frameCount < minFrameCount) { 1713 frameCount = minFrameCount; 1714 } 1715 } 1716 } 1717 1718 if (mType == DIRECT) { 1719 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1720 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1721 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1722 "for output %p with format %d", 1723 sampleRate, format, channelMask, mOutput, mFormat); 1724 lStatus = BAD_VALUE; 1725 goto Exit; 1726 } 1727 } 1728 } else { 1729 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1730 if (sampleRate > mSampleRate*2) { 1731 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1732 lStatus = BAD_VALUE; 1733 goto Exit; 1734 } 1735 } 1736 1737 lStatus = initCheck(); 1738 if (lStatus != NO_ERROR) { 1739 ALOGE("Audio driver not initialized."); 1740 goto Exit; 1741 } 1742 1743 { // scope for mLock 1744 Mutex::Autolock _l(mLock); 1745 1746 // all tracks in same audio session must share the same routing strategy otherwise 1747 // conflicts will happen when tracks are moved from one output to another by audio policy 1748 // manager 1749 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1750 for (size_t i = 0; i < mTracks.size(); ++i) { 1751 sp<Track> t = mTracks[i]; 1752 if (t != 0 && !t->isOutputTrack()) { 1753 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1754 if (sessionId == t->sessionId() && strategy != actual) { 1755 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1756 strategy, actual); 1757 lStatus = BAD_VALUE; 1758 goto Exit; 1759 } 1760 } 1761 } 1762 1763 if (!isTimed) { 1764 track = new Track(this, client, streamType, sampleRate, format, 1765 channelMask, frameCount, sharedBuffer, sessionId, flags); 1766 } else { 1767 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1768 channelMask, frameCount, sharedBuffer, sessionId); 1769 } 1770 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1771 lStatus = NO_MEMORY; 1772 goto Exit; 1773 } 1774 mTracks.add(track); 1775 1776 sp<EffectChain> chain = getEffectChain_l(sessionId); 1777 if (chain != 0) { 1778 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1779 track->setMainBuffer(chain->inBuffer()); 1780 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1781 chain->incTrackCnt(); 1782 } 1783 } 1784 1785#ifdef HAVE_REQUEST_PRIORITY 1786 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1787 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1788 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1789 // so ask activity manager to do this on our behalf 1790 int err = requestPriority(callingPid, tid, 1); 1791 if (err != 0) { 1792 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1793 1, callingPid, tid, err); 1794 } 1795 } 1796#endif 1797 1798 lStatus = NO_ERROR; 1799 1800Exit: 1801 if (status) { 1802 *status = lStatus; 1803 } 1804 return track; 1805} 1806 1807uint32_t AudioFlinger::PlaybackThread::latency() const 1808{ 1809 Mutex::Autolock _l(mLock); 1810 if (initCheck() == NO_ERROR) { 1811 return mOutput->stream->get_latency(mOutput->stream); 1812 } else { 1813 return 0; 1814 } 1815} 1816 1817void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1818{ 1819 Mutex::Autolock _l(mLock); 1820 mMasterVolume = value; 1821} 1822 1823void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1824{ 1825 Mutex::Autolock _l(mLock); 1826 setMasterMute_l(muted); 1827} 1828 1829void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1830{ 1831 Mutex::Autolock _l(mLock); 1832 mStreamTypes[stream].volume = value; 1833} 1834 1835void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1836{ 1837 Mutex::Autolock _l(mLock); 1838 mStreamTypes[stream].mute = muted; 1839} 1840 1841float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1842{ 1843 Mutex::Autolock _l(mLock); 1844 return mStreamTypes[stream].volume; 1845} 1846 1847// addTrack_l() must be called with ThreadBase::mLock held 1848status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1849{ 1850 status_t status = ALREADY_EXISTS; 1851 1852 // set retry count for buffer fill 1853 track->mRetryCount = kMaxTrackStartupRetries; 1854 if (mActiveTracks.indexOf(track) < 0) { 1855 // the track is newly added, make sure it fills up all its 1856 // buffers before playing. This is to ensure the client will 1857 // effectively get the latency it requested. 1858 track->mFillingUpStatus = Track::FS_FILLING; 1859 track->mResetDone = false; 1860 track->mPresentationCompleteFrames = 0; 1861 mActiveTracks.add(track); 1862 if (track->mainBuffer() != mMixBuffer) { 1863 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1864 if (chain != 0) { 1865 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1866 chain->incActiveTrackCnt(); 1867 } 1868 } 1869 1870 status = NO_ERROR; 1871 } 1872 1873 ALOGV("mWaitWorkCV.broadcast"); 1874 mWaitWorkCV.broadcast(); 1875 1876 return status; 1877} 1878 1879// destroyTrack_l() must be called with ThreadBase::mLock held 1880void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1881{ 1882 track->mState = TrackBase::TERMINATED; 1883 // active tracks are removed by threadLoop() 1884 if (mActiveTracks.indexOf(track) < 0) { 1885 removeTrack_l(track); 1886 } 1887} 1888 1889void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1890{ 1891 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1892 mTracks.remove(track); 1893 deleteTrackName_l(track->name()); 1894 // redundant as track is about to be destroyed, for dumpsys only 1895 track->mName = -1; 1896 if (track->isFastTrack()) { 1897 int index = track->mFastIndex; 1898 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1899 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1900 mFastTrackAvailMask |= 1 << index; 1901 // redundant as track is about to be destroyed, for dumpsys only 1902 track->mFastIndex = -1; 1903 } 1904 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1905 if (chain != 0) { 1906 chain->decTrackCnt(); 1907 } 1908} 1909 1910String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1911{ 1912 String8 out_s8 = String8(""); 1913 char *s; 1914 1915 Mutex::Autolock _l(mLock); 1916 if (initCheck() != NO_ERROR) { 1917 return out_s8; 1918 } 1919 1920 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1921 out_s8 = String8(s); 1922 free(s); 1923 return out_s8; 1924} 1925 1926// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1927void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1928 AudioSystem::OutputDescriptor desc; 1929 void *param2 = NULL; 1930 1931 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1932 1933 switch (event) { 1934 case AudioSystem::OUTPUT_OPENED: 1935 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1936 desc.channels = mChannelMask; 1937 desc.samplingRate = mSampleRate; 1938 desc.format = mFormat; 1939 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1940 desc.latency = latency(); 1941 param2 = &desc; 1942 break; 1943 1944 case AudioSystem::STREAM_CONFIG_CHANGED: 1945 param2 = ¶m; 1946 case AudioSystem::OUTPUT_CLOSED: 1947 default: 1948 break; 1949 } 1950 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1951} 1952 1953void AudioFlinger::PlaybackThread::readOutputParameters() 1954{ 1955 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1956 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1957 mChannelCount = (uint16_t)popcount(mChannelMask); 1958 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1959 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1960 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1961 if (mFrameCount & 15) { 1962 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1963 mFrameCount); 1964 } 1965 1966 // Calculate size of normal mix buffer relative to the HAL output buffer size 1967 double multiplier = 1.0; 1968 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1969 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1970 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1971 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1972 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1973 maxNormalFrameCount = maxNormalFrameCount & ~15; 1974 if (maxNormalFrameCount < minNormalFrameCount) { 1975 maxNormalFrameCount = minNormalFrameCount; 1976 } 1977 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1978 if (multiplier <= 1.0) { 1979 multiplier = 1.0; 1980 } else if (multiplier <= 2.0) { 1981 if (2 * mFrameCount <= maxNormalFrameCount) { 1982 multiplier = 2.0; 1983 } else { 1984 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1985 } 1986 } else { 1987 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1988 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1989 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 1990 // FIXME this rounding up should not be done if no HAL SRC 1991 uint32_t truncMult = (uint32_t) multiplier; 1992 if ((truncMult & 1)) { 1993 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1994 ++truncMult; 1995 } 1996 } 1997 multiplier = (double) truncMult; 1998 } 1999 } 2000 mNormalFrameCount = multiplier * mFrameCount; 2001 // round up to nearest 16 frames to satisfy AudioMixer 2002 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2003 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2004 2005 // FIXME - Current mixer implementation only supports stereo output: Always 2006 // Allocate a stereo buffer even if HW output is mono. 2007 delete[] mMixBuffer; 2008 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2009 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2010 2011 // force reconfiguration of effect chains and engines to take new buffer size and audio 2012 // parameters into account 2013 // Note that mLock is not held when readOutputParameters() is called from the constructor 2014 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2015 // matter. 2016 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2017 Vector< sp<EffectChain> > effectChains = mEffectChains; 2018 for (size_t i = 0; i < effectChains.size(); i ++) { 2019 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2020 } 2021} 2022 2023status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2024{ 2025 if (halFrames == NULL || dspFrames == NULL) { 2026 return BAD_VALUE; 2027 } 2028 Mutex::Autolock _l(mLock); 2029 if (initCheck() != NO_ERROR) { 2030 return INVALID_OPERATION; 2031 } 2032 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2033 2034 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2035} 2036 2037uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2038{ 2039 Mutex::Autolock _l(mLock); 2040 uint32_t result = 0; 2041 if (getEffectChain_l(sessionId) != 0) { 2042 result = EFFECT_SESSION; 2043 } 2044 2045 for (size_t i = 0; i < mTracks.size(); ++i) { 2046 sp<Track> track = mTracks[i]; 2047 if (sessionId == track->sessionId() && 2048 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2049 result |= TRACK_SESSION; 2050 break; 2051 } 2052 } 2053 2054 return result; 2055} 2056 2057uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2058{ 2059 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2060 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2061 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2062 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2063 } 2064 for (size_t i = 0; i < mTracks.size(); i++) { 2065 sp<Track> track = mTracks[i]; 2066 if (sessionId == track->sessionId() && 2067 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2068 return AudioSystem::getStrategyForStream(track->streamType()); 2069 } 2070 } 2071 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2072} 2073 2074 2075AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2076{ 2077 Mutex::Autolock _l(mLock); 2078 return mOutput; 2079} 2080 2081AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2082{ 2083 Mutex::Autolock _l(mLock); 2084 AudioStreamOut *output = mOutput; 2085 mOutput = NULL; 2086 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2087 // must push a NULL and wait for ack 2088 mOutputSink.clear(); 2089 mPipeSink.clear(); 2090 mNormalSink.clear(); 2091 return output; 2092} 2093 2094// this method must always be called either with ThreadBase mLock held or inside the thread loop 2095audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2096{ 2097 if (mOutput == NULL) { 2098 return NULL; 2099 } 2100 return &mOutput->stream->common; 2101} 2102 2103uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2104{ 2105 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2106 // decoding and transfer time. So sleeping for half of the latency would likely cause 2107 // underruns 2108 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2109 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2110 } else { 2111 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2112 } 2113} 2114 2115status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2116{ 2117 if (!isValidSyncEvent(event)) { 2118 return BAD_VALUE; 2119 } 2120 2121 Mutex::Autolock _l(mLock); 2122 2123 for (size_t i = 0; i < mTracks.size(); ++i) { 2124 sp<Track> track = mTracks[i]; 2125 if (event->triggerSession() == track->sessionId()) { 2126 track->setSyncEvent(event); 2127 return NO_ERROR; 2128 } 2129 } 2130 2131 return NAME_NOT_FOUND; 2132} 2133 2134bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2135{ 2136 switch (event->type()) { 2137 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2138 return true; 2139 default: 2140 break; 2141 } 2142 return false; 2143} 2144 2145void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2146{ 2147 size_t count = tracksToRemove.size(); 2148 if (CC_UNLIKELY(count)) { 2149 for (size_t i = 0 ; i < count ; i++) { 2150 const sp<Track>& track = tracksToRemove.itemAt(i); 2151 if ((track->sharedBuffer() != 0) && 2152 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2153 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2154 } 2155 } 2156 } 2157 2158} 2159 2160// ---------------------------------------------------------------------------- 2161 2162AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2163 audio_io_handle_t id, uint32_t device, type_t type) 2164 : PlaybackThread(audioFlinger, output, id, device, type), 2165 // mAudioMixer below 2166#ifdef SOAKER 2167 mSoaker(NULL), 2168#endif 2169 // mFastMixer below 2170 mFastMixerFutex(0) 2171 // mOutputSink below 2172 // mPipeSink below 2173 // mNormalSink below 2174{ 2175 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2176 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2177 "mFrameCount=%d, mNormalFrameCount=%d", 2178 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2179 mNormalFrameCount); 2180 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2181 2182 // FIXME - Current mixer implementation only supports stereo output 2183 if (mChannelCount == 1) { 2184 ALOGE("Invalid audio hardware channel count"); 2185 } 2186 2187 // create an NBAIO sink for the HAL output stream, and negotiate 2188 mOutputSink = new AudioStreamOutSink(output->stream); 2189 size_t numCounterOffers = 0; 2190 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2191 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2192 ALOG_ASSERT(index == 0); 2193 2194 // initialize fast mixer depending on configuration 2195 bool initFastMixer; 2196 switch (kUseFastMixer) { 2197 case FastMixer_Never: 2198 initFastMixer = false; 2199 break; 2200 case FastMixer_Always: 2201 initFastMixer = true; 2202 break; 2203 case FastMixer_Static: 2204 case FastMixer_Dynamic: 2205 initFastMixer = mFrameCount < mNormalFrameCount; 2206 break; 2207 } 2208 if (initFastMixer) { 2209 2210 // create a MonoPipe to connect our submix to FastMixer 2211 NBAIO_Format format = mOutputSink->format(); 2212 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2213 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2214 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2215 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2216 const NBAIO_Format offers[1] = {format}; 2217 size_t numCounterOffers = 0; 2218 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2219 ALOG_ASSERT(index == 0); 2220 mPipeSink = monoPipe; 2221 2222#ifdef TEE_SINK_FRAMES 2223 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2224 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2225 numCounterOffers = 0; 2226 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2227 ALOG_ASSERT(index == 0); 2228 mTeeSink = teeSink; 2229 PipeReader *teeSource = new PipeReader(*teeSink); 2230 numCounterOffers = 0; 2231 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2232 ALOG_ASSERT(index == 0); 2233 mTeeSource = teeSource; 2234#endif 2235 2236#ifdef SOAKER 2237 // create a soaker as workaround for governor issues 2238 mSoaker = new Soaker(); 2239 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2240 mSoaker->run("Soaker", PRIORITY_LOWEST); 2241#endif 2242 2243 // create fast mixer and configure it initially with just one fast track for our submix 2244 mFastMixer = new FastMixer(); 2245 FastMixerStateQueue *sq = mFastMixer->sq(); 2246 FastMixerState *state = sq->begin(); 2247 FastTrack *fastTrack = &state->mFastTracks[0]; 2248 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2249 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2250 fastTrack->mVolumeProvider = NULL; 2251 fastTrack->mGeneration++; 2252 state->mFastTracksGen++; 2253 state->mTrackMask = 1; 2254 // fast mixer will use the HAL output sink 2255 state->mOutputSink = mOutputSink.get(); 2256 state->mOutputSinkGen++; 2257 state->mFrameCount = mFrameCount; 2258 state->mCommand = FastMixerState::COLD_IDLE; 2259 // already done in constructor initialization list 2260 //mFastMixerFutex = 0; 2261 state->mColdFutexAddr = &mFastMixerFutex; 2262 state->mColdGen++; 2263 state->mDumpState = &mFastMixerDumpState; 2264 state->mTeeSink = mTeeSink.get(); 2265 sq->end(); 2266 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2267 2268 // start the fast mixer 2269 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2270#ifdef HAVE_REQUEST_PRIORITY 2271 pid_t tid = mFastMixer->getTid(); 2272 int err = requestPriority(getpid_cached, tid, 2); 2273 if (err != 0) { 2274 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2275 2, getpid_cached, tid, err); 2276 } 2277#endif 2278 2279 } else { 2280 mFastMixer = NULL; 2281 } 2282 2283 switch (kUseFastMixer) { 2284 case FastMixer_Never: 2285 case FastMixer_Dynamic: 2286 mNormalSink = mOutputSink; 2287 break; 2288 case FastMixer_Always: 2289 mNormalSink = mPipeSink; 2290 break; 2291 case FastMixer_Static: 2292 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2293 break; 2294 } 2295} 2296 2297AudioFlinger::MixerThread::~MixerThread() 2298{ 2299 if (mFastMixer != NULL) { 2300 FastMixerStateQueue *sq = mFastMixer->sq(); 2301 FastMixerState *state = sq->begin(); 2302 if (state->mCommand == FastMixerState::COLD_IDLE) { 2303 int32_t old = android_atomic_inc(&mFastMixerFutex); 2304 if (old == -1) { 2305 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2306 } 2307 } 2308 state->mCommand = FastMixerState::EXIT; 2309 sq->end(); 2310 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2311 mFastMixer->join(); 2312 // Though the fast mixer thread has exited, it's state queue is still valid. 2313 // We'll use that extract the final state which contains one remaining fast track 2314 // corresponding to our sub-mix. 2315 state = sq->begin(); 2316 ALOG_ASSERT(state->mTrackMask == 1); 2317 FastTrack *fastTrack = &state->mFastTracks[0]; 2318 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2319 delete fastTrack->mBufferProvider; 2320 sq->end(false /*didModify*/); 2321 delete mFastMixer; 2322#ifdef SOAKER 2323 if (mSoaker != NULL) { 2324 mSoaker->requestExitAndWait(); 2325 } 2326 delete mSoaker; 2327#endif 2328 } 2329 delete mAudioMixer; 2330} 2331 2332class CpuStats { 2333public: 2334 CpuStats(); 2335 void sample(const String8 &title); 2336#ifdef DEBUG_CPU_USAGE 2337private: 2338 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2339 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2340 2341 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2342 2343 int mCpuNum; // thread's current CPU number 2344 int mCpukHz; // frequency of thread's current CPU in kHz 2345#endif 2346}; 2347 2348CpuStats::CpuStats() 2349#ifdef DEBUG_CPU_USAGE 2350 : mCpuNum(-1), mCpukHz(-1) 2351#endif 2352{ 2353} 2354 2355void CpuStats::sample(const String8 &title) { 2356#ifdef DEBUG_CPU_USAGE 2357 // get current thread's delta CPU time in wall clock ns 2358 double wcNs; 2359 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2360 2361 // record sample for wall clock statistics 2362 if (valid) { 2363 mWcStats.sample(wcNs); 2364 } 2365 2366 // get the current CPU number 2367 int cpuNum = sched_getcpu(); 2368 2369 // get the current CPU frequency in kHz 2370 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2371 2372 // check if either CPU number or frequency changed 2373 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2374 mCpuNum = cpuNum; 2375 mCpukHz = cpukHz; 2376 // ignore sample for purposes of cycles 2377 valid = false; 2378 } 2379 2380 // if no change in CPU number or frequency, then record sample for cycle statistics 2381 if (valid && mCpukHz > 0) { 2382 double cycles = wcNs * cpukHz * 0.000001; 2383 mHzStats.sample(cycles); 2384 } 2385 2386 unsigned n = mWcStats.n(); 2387 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2388 if ((n & 127) == 1) { 2389 long long elapsed = mCpuUsage.elapsed(); 2390 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2391 double perLoop = elapsed / (double) n; 2392 double perLoop100 = perLoop * 0.01; 2393 double perLoop1k = perLoop * 0.001; 2394 double mean = mWcStats.mean(); 2395 double stddev = mWcStats.stddev(); 2396 double minimum = mWcStats.minimum(); 2397 double maximum = mWcStats.maximum(); 2398 double meanCycles = mHzStats.mean(); 2399 double stddevCycles = mHzStats.stddev(); 2400 double minCycles = mHzStats.minimum(); 2401 double maxCycles = mHzStats.maximum(); 2402 mCpuUsage.resetElapsed(); 2403 mWcStats.reset(); 2404 mHzStats.reset(); 2405 ALOGD("CPU usage for %s over past %.1f secs\n" 2406 " (%u mixer loops at %.1f mean ms per loop):\n" 2407 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2408 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2409 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2410 title.string(), 2411 elapsed * .000000001, n, perLoop * .000001, 2412 mean * .001, 2413 stddev * .001, 2414 minimum * .001, 2415 maximum * .001, 2416 mean / perLoop100, 2417 stddev / perLoop100, 2418 minimum / perLoop100, 2419 maximum / perLoop100, 2420 meanCycles / perLoop1k, 2421 stddevCycles / perLoop1k, 2422 minCycles / perLoop1k, 2423 maxCycles / perLoop1k); 2424 2425 } 2426 } 2427#endif 2428}; 2429 2430void AudioFlinger::PlaybackThread::checkSilentMode_l() 2431{ 2432 if (!mMasterMute) { 2433 char value[PROPERTY_VALUE_MAX]; 2434 if (property_get("ro.audio.silent", value, "0") > 0) { 2435 char *endptr; 2436 unsigned long ul = strtoul(value, &endptr, 0); 2437 if (*endptr == '\0' && ul != 0) { 2438 ALOGD("Silence is golden"); 2439 // The setprop command will not allow a property to be changed after 2440 // the first time it is set, so we don't have to worry about un-muting. 2441 setMasterMute_l(true); 2442 } 2443 } 2444 } 2445} 2446 2447bool AudioFlinger::PlaybackThread::threadLoop() 2448{ 2449 Vector< sp<Track> > tracksToRemove; 2450 2451 standbyTime = systemTime(); 2452 2453 // MIXER 2454 nsecs_t lastWarning = 0; 2455if (mType == MIXER) { 2456 longStandbyExit = false; 2457} 2458 2459 // DUPLICATING 2460 // FIXME could this be made local to while loop? 2461 writeFrames = 0; 2462 2463 cacheParameters_l(); 2464 sleepTime = idleSleepTime; 2465 2466if (mType == MIXER) { 2467 sleepTimeShift = 0; 2468} 2469 2470 CpuStats cpuStats; 2471 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2472 2473 acquireWakeLock(); 2474 2475 while (!exitPending()) 2476 { 2477 cpuStats.sample(myName); 2478 2479 Vector< sp<EffectChain> > effectChains; 2480 2481 processConfigEvents(); 2482 2483 { // scope for mLock 2484 2485 Mutex::Autolock _l(mLock); 2486 2487 if (checkForNewParameters_l()) { 2488 cacheParameters_l(); 2489 } 2490 2491 saveOutputTracks(); 2492 2493 // put audio hardware into standby after short delay 2494 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2495 mSuspended > 0)) { 2496 if (!mStandby) { 2497 2498 threadLoop_standby(); 2499 2500 mStandby = true; 2501 mBytesWritten = 0; 2502 } 2503 2504 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2505 // we're about to wait, flush the binder command buffer 2506 IPCThreadState::self()->flushCommands(); 2507 2508 clearOutputTracks(); 2509 2510 if (exitPending()) break; 2511 2512 releaseWakeLock_l(); 2513 // wait until we have something to do... 2514 ALOGV("%s going to sleep", myName.string()); 2515 mWaitWorkCV.wait(mLock); 2516 ALOGV("%s waking up", myName.string()); 2517 acquireWakeLock_l(); 2518 2519 mMixerStatus = MIXER_IDLE; 2520 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2521 2522 checkSilentMode_l(); 2523 2524 standbyTime = systemTime() + standbyDelay; 2525 sleepTime = idleSleepTime; 2526 if (mType == MIXER) { 2527 sleepTimeShift = 0; 2528 } 2529 2530 continue; 2531 } 2532 } 2533 2534 // mMixerStatusIgnoringFastTracks is also updated internally 2535 mMixerStatus = prepareTracks_l(&tracksToRemove); 2536 2537 // prevent any changes in effect chain list and in each effect chain 2538 // during mixing and effect process as the audio buffers could be deleted 2539 // or modified if an effect is created or deleted 2540 lockEffectChains_l(effectChains); 2541 } 2542 2543 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2544 threadLoop_mix(); 2545 } else { 2546 threadLoop_sleepTime(); 2547 } 2548 2549 if (mSuspended > 0) { 2550 sleepTime = suspendSleepTimeUs(); 2551 } 2552 2553 // only process effects if we're going to write 2554 if (sleepTime == 0) { 2555 for (size_t i = 0; i < effectChains.size(); i ++) { 2556 effectChains[i]->process_l(); 2557 } 2558 } 2559 2560 // enable changes in effect chain 2561 unlockEffectChains(effectChains); 2562 2563 // sleepTime == 0 means we must write to audio hardware 2564 if (sleepTime == 0) { 2565 2566 threadLoop_write(); 2567 2568if (mType == MIXER) { 2569 // write blocked detection 2570 nsecs_t now = systemTime(); 2571 nsecs_t delta = now - mLastWriteTime; 2572 if (!mStandby && delta > maxPeriod) { 2573 mNumDelayedWrites++; 2574 if ((now - lastWarning) > kWarningThrottleNs) { 2575#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2576 ScopedTrace st(ATRACE_TAG, "underrun"); 2577#endif 2578 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2579 ns2ms(delta), mNumDelayedWrites, this); 2580 lastWarning = now; 2581 } 2582 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2583 // a different threshold. Or completely removed for what it is worth anyway... 2584 if (mStandby) { 2585 longStandbyExit = true; 2586 } 2587 } 2588} 2589 2590 mStandby = false; 2591 } else { 2592 usleep(sleepTime); 2593 } 2594 2595 // Finally let go of removed track(s), without the lock held 2596 // since we can't guarantee the destructors won't acquire that 2597 // same lock. This will also mutate and push a new fast mixer state. 2598 threadLoop_removeTracks(tracksToRemove); 2599 tracksToRemove.clear(); 2600 2601 // FIXME I don't understand the need for this here; 2602 // it was in the original code but maybe the 2603 // assignment in saveOutputTracks() makes this unnecessary? 2604 clearOutputTracks(); 2605 2606 // Effect chains will be actually deleted here if they were removed from 2607 // mEffectChains list during mixing or effects processing 2608 effectChains.clear(); 2609 2610 // FIXME Note that the above .clear() is no longer necessary since effectChains 2611 // is now local to this block, but will keep it for now (at least until merge done). 2612 } 2613 2614if (mType == MIXER || mType == DIRECT) { 2615 // put output stream into standby mode 2616 if (!mStandby) { 2617 mOutput->stream->common.standby(&mOutput->stream->common); 2618 } 2619} 2620if (mType == DUPLICATING) { 2621 // for DuplicatingThread, standby mode is handled by the outputTracks 2622} 2623 2624 releaseWakeLock(); 2625 2626 ALOGV("Thread %p type %d exiting", this, mType); 2627 return false; 2628} 2629 2630void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2631{ 2632 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2633} 2634 2635void AudioFlinger::MixerThread::threadLoop_write() 2636{ 2637 // FIXME we should only do one push per cycle; confirm this is true 2638 // Start the fast mixer if it's not already running 2639 if (mFastMixer != NULL) { 2640 FastMixerStateQueue *sq = mFastMixer->sq(); 2641 FastMixerState *state = sq->begin(); 2642 if (state->mCommand != FastMixerState::MIX_WRITE && 2643 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2644 if (state->mCommand == FastMixerState::COLD_IDLE) { 2645 int32_t old = android_atomic_inc(&mFastMixerFutex); 2646 if (old == -1) { 2647 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2648 } 2649 } 2650 state->mCommand = FastMixerState::MIX_WRITE; 2651 sq->end(); 2652 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2653 if (kUseFastMixer == FastMixer_Dynamic) { 2654 mNormalSink = mPipeSink; 2655 } 2656 } else { 2657 sq->end(false /*didModify*/); 2658 } 2659 } 2660 PlaybackThread::threadLoop_write(); 2661} 2662 2663// shared by MIXER and DIRECT, overridden by DUPLICATING 2664void AudioFlinger::PlaybackThread::threadLoop_write() 2665{ 2666 // FIXME rewrite to reduce number of system calls 2667 mLastWriteTime = systemTime(); 2668 mInWrite = true; 2669 2670#define mBitShift 2 // FIXME 2671 size_t count = mixBufferSize >> mBitShift; 2672#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2673 Tracer::traceBegin(ATRACE_TAG, "write"); 2674#endif 2675 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2676#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2677 Tracer::traceEnd(ATRACE_TAG); 2678#endif 2679 if (framesWritten > 0) { 2680 size_t bytesWritten = framesWritten << mBitShift; 2681 mBytesWritten += bytesWritten; 2682 } 2683 2684 mNumWrites++; 2685 mInWrite = false; 2686} 2687 2688void AudioFlinger::MixerThread::threadLoop_standby() 2689{ 2690 // Idle the fast mixer if it's currently running 2691 if (mFastMixer != NULL) { 2692 FastMixerStateQueue *sq = mFastMixer->sq(); 2693 FastMixerState *state = sq->begin(); 2694 if (!(state->mCommand & FastMixerState::IDLE)) { 2695 state->mCommand = FastMixerState::COLD_IDLE; 2696 state->mColdFutexAddr = &mFastMixerFutex; 2697 state->mColdGen++; 2698 mFastMixerFutex = 0; 2699 sq->end(); 2700 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2701 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2702 if (kUseFastMixer == FastMixer_Dynamic) { 2703 mNormalSink = mOutputSink; 2704 } 2705 } else { 2706 sq->end(false /*didModify*/); 2707 } 2708 } 2709 PlaybackThread::threadLoop_standby(); 2710} 2711 2712// shared by MIXER and DIRECT, overridden by DUPLICATING 2713void AudioFlinger::PlaybackThread::threadLoop_standby() 2714{ 2715 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2716 mOutput->stream->common.standby(&mOutput->stream->common); 2717} 2718 2719void AudioFlinger::MixerThread::threadLoop_mix() 2720{ 2721 // obtain the presentation timestamp of the next output buffer 2722 int64_t pts; 2723 status_t status = INVALID_OPERATION; 2724 2725 if (NULL != mOutput->stream->get_next_write_timestamp) { 2726 status = mOutput->stream->get_next_write_timestamp( 2727 mOutput->stream, &pts); 2728 } 2729 2730 if (status != NO_ERROR) { 2731 pts = AudioBufferProvider::kInvalidPTS; 2732 } 2733 2734 // mix buffers... 2735 mAudioMixer->process(pts); 2736 // increase sleep time progressively when application underrun condition clears. 2737 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2738 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2739 // such that we would underrun the audio HAL. 2740 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2741 sleepTimeShift--; 2742 } 2743 sleepTime = 0; 2744 standbyTime = systemTime() + standbyDelay; 2745 //TODO: delay standby when effects have a tail 2746} 2747 2748void AudioFlinger::MixerThread::threadLoop_sleepTime() 2749{ 2750 // If no tracks are ready, sleep once for the duration of an output 2751 // buffer size, then write 0s to the output 2752 if (sleepTime == 0) { 2753 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2754 sleepTime = activeSleepTime >> sleepTimeShift; 2755 if (sleepTime < kMinThreadSleepTimeUs) { 2756 sleepTime = kMinThreadSleepTimeUs; 2757 } 2758 // reduce sleep time in case of consecutive application underruns to avoid 2759 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2760 // duration we would end up writing less data than needed by the audio HAL if 2761 // the condition persists. 2762 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2763 sleepTimeShift++; 2764 } 2765 } else { 2766 sleepTime = idleSleepTime; 2767 } 2768 } else if (mBytesWritten != 0 || 2769 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2770 memset (mMixBuffer, 0, mixBufferSize); 2771 sleepTime = 0; 2772 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2773 } 2774 // TODO add standby time extension fct of effect tail 2775} 2776 2777// prepareTracks_l() must be called with ThreadBase::mLock held 2778AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2779 Vector< sp<Track> > *tracksToRemove) 2780{ 2781 2782 mixer_state mixerStatus = MIXER_IDLE; 2783 // find out which tracks need to be processed 2784 size_t count = mActiveTracks.size(); 2785 size_t mixedTracks = 0; 2786 size_t tracksWithEffect = 0; 2787 // counts only _active_ fast tracks 2788 size_t fastTracks = 0; 2789 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2790 2791 float masterVolume = mMasterVolume; 2792 bool masterMute = mMasterMute; 2793 2794 if (masterMute) { 2795 masterVolume = 0; 2796 } 2797 // Delegate master volume control to effect in output mix effect chain if needed 2798 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2799 if (chain != 0) { 2800 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2801 chain->setVolume_l(&v, &v); 2802 masterVolume = (float)((v + (1 << 23)) >> 24); 2803 chain.clear(); 2804 } 2805 2806 // prepare a new state to push 2807 FastMixerStateQueue *sq = NULL; 2808 FastMixerState *state = NULL; 2809 bool didModify = false; 2810 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2811 if (mFastMixer != NULL) { 2812 sq = mFastMixer->sq(); 2813 state = sq->begin(); 2814 } 2815 2816 for (size_t i=0 ; i<count ; i++) { 2817 sp<Track> t = mActiveTracks[i].promote(); 2818 if (t == 0) continue; 2819 2820 // this const just means the local variable doesn't change 2821 Track* const track = t.get(); 2822 2823 // process fast tracks 2824 if (track->isFastTrack()) { 2825 2826 // It's theoretically possible (though unlikely) for a fast track to be created 2827 // and then removed within the same normal mix cycle. This is not a problem, as 2828 // the track never becomes active so it's fast mixer slot is never touched. 2829 // The converse, of removing an (active) track and then creating a new track 2830 // at the identical fast mixer slot within the same normal mix cycle, 2831 // is impossible because the slot isn't marked available until the end of each cycle. 2832 int j = track->mFastIndex; 2833 FastTrack *fastTrack = &state->mFastTracks[j]; 2834 2835 // Determine whether the track is currently in underrun condition, 2836 // and whether it had a recent underrun. 2837 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2838 uint32_t recentFull = (underruns.mBitFields.mFull - 2839 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2840 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2841 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2842 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2843 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2844 uint32_t recentUnderruns = recentPartial + recentEmpty; 2845 track->mObservedUnderruns = underruns; 2846 // don't count underruns that occur while stopping or pausing 2847 // or stopped which can occur when flush() is called while active 2848 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2849 track->mUnderrunCount += recentUnderruns; 2850 } 2851 2852 // This is similar to the state machine for normal tracks, 2853 // with a few modifications for fast tracks. 2854 bool isActive = true; 2855 switch (track->mState) { 2856 case TrackBase::STOPPING_1: 2857 // track stays active in STOPPING_1 state until first underrun 2858 if (recentUnderruns > 0) { 2859 track->mState = TrackBase::STOPPING_2; 2860 } 2861 break; 2862 case TrackBase::PAUSING: 2863 // ramp down is not yet implemented 2864 track->setPaused(); 2865 break; 2866 case TrackBase::RESUMING: 2867 // ramp up is not yet implemented 2868 track->mState = TrackBase::ACTIVE; 2869 break; 2870 case TrackBase::ACTIVE: 2871 if (recentFull > 0 || recentPartial > 0) { 2872 // track has provided at least some frames recently: reset retry count 2873 track->mRetryCount = kMaxTrackRetries; 2874 } 2875 if (recentUnderruns == 0) { 2876 // no recent underruns: stay active 2877 break; 2878 } 2879 // there has recently been an underrun of some kind 2880 if (track->sharedBuffer() == 0) { 2881 // were any of the recent underruns "empty" (no frames available)? 2882 if (recentEmpty == 0) { 2883 // no, then ignore the partial underruns as they are allowed indefinitely 2884 break; 2885 } 2886 // there has recently been an "empty" underrun: decrement the retry counter 2887 if (--(track->mRetryCount) > 0) { 2888 break; 2889 } 2890 // indicate to client process that the track was disabled because of underrun; 2891 // it will then automatically call start() when data is available 2892 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2893 // remove from active list, but state remains ACTIVE [confusing but true] 2894 isActive = false; 2895 break; 2896 } 2897 // fall through 2898 case TrackBase::STOPPING_2: 2899 case TrackBase::PAUSED: 2900 case TrackBase::TERMINATED: 2901 case TrackBase::STOPPED: 2902 case TrackBase::FLUSHED: // flush() while active 2903 // Check for presentation complete if track is inactive 2904 // We have consumed all the buffers of this track. 2905 // This would be incomplete if we auto-paused on underrun 2906 { 2907 size_t audioHALFrames = 2908 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2909 size_t framesWritten = 2910 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2911 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2912 // track stays in active list until presentation is complete 2913 break; 2914 } 2915 } 2916 if (track->isStopping_2()) { 2917 track->mState = TrackBase::STOPPED; 2918 } 2919 if (track->isStopped()) { 2920 // Can't reset directly, as fast mixer is still polling this track 2921 // track->reset(); 2922 // So instead mark this track as needing to be reset after push with ack 2923 resetMask |= 1 << i; 2924 } 2925 isActive = false; 2926 break; 2927 case TrackBase::IDLE: 2928 default: 2929 LOG_FATAL("unexpected track state %d", track->mState); 2930 } 2931 2932 if (isActive) { 2933 // was it previously inactive? 2934 if (!(state->mTrackMask & (1 << j))) { 2935 ExtendedAudioBufferProvider *eabp = track; 2936 VolumeProvider *vp = track; 2937 fastTrack->mBufferProvider = eabp; 2938 fastTrack->mVolumeProvider = vp; 2939 fastTrack->mSampleRate = track->mSampleRate; 2940 fastTrack->mChannelMask = track->mChannelMask; 2941 fastTrack->mGeneration++; 2942 state->mTrackMask |= 1 << j; 2943 didModify = true; 2944 // no acknowledgement required for newly active tracks 2945 } 2946 // cache the combined master volume and stream type volume for fast mixer; this 2947 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2948 track->mCachedVolume = track->isMuted() ? 2949 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2950 ++fastTracks; 2951 } else { 2952 // was it previously active? 2953 if (state->mTrackMask & (1 << j)) { 2954 fastTrack->mBufferProvider = NULL; 2955 fastTrack->mGeneration++; 2956 state->mTrackMask &= ~(1 << j); 2957 didModify = true; 2958 // If any fast tracks were removed, we must wait for acknowledgement 2959 // because we're about to decrement the last sp<> on those tracks. 2960 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2961 } else { 2962 LOG_FATAL("fast track %d should have been active", j); 2963 } 2964 tracksToRemove->add(track); 2965 // Avoids a misleading display in dumpsys 2966 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2967 } 2968 continue; 2969 } 2970 2971 { // local variable scope to avoid goto warning 2972 2973 audio_track_cblk_t* cblk = track->cblk(); 2974 2975 // The first time a track is added we wait 2976 // for all its buffers to be filled before processing it 2977 int name = track->name(); 2978 // make sure that we have enough frames to mix one full buffer. 2979 // enforce this condition only once to enable draining the buffer in case the client 2980 // app does not call stop() and relies on underrun to stop: 2981 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2982 // during last round 2983 uint32_t minFrames = 1; 2984 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2985 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2986 if (t->sampleRate() == (int)mSampleRate) { 2987 minFrames = mNormalFrameCount; 2988 } else { 2989 // +1 for rounding and +1 for additional sample needed for interpolation 2990 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2991 // add frames already consumed but not yet released by the resampler 2992 // because cblk->framesReady() will include these frames 2993 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2994 // the minimum track buffer size is normally twice the number of frames necessary 2995 // to fill one buffer and the resampler should not leave more than one buffer worth 2996 // of unreleased frames after each pass, but just in case... 2997 ALOG_ASSERT(minFrames <= cblk->frameCount); 2998 } 2999 } 3000 if ((track->framesReady() >= minFrames) && track->isReady() && 3001 !track->isPaused() && !track->isTerminated()) 3002 { 3003 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3004 3005 mixedTracks++; 3006 3007 // track->mainBuffer() != mMixBuffer means there is an effect chain 3008 // connected to the track 3009 chain.clear(); 3010 if (track->mainBuffer() != mMixBuffer) { 3011 chain = getEffectChain_l(track->sessionId()); 3012 // Delegate volume control to effect in track effect chain if needed 3013 if (chain != 0) { 3014 tracksWithEffect++; 3015 } else { 3016 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3017 name, track->sessionId()); 3018 } 3019 } 3020 3021 3022 int param = AudioMixer::VOLUME; 3023 if (track->mFillingUpStatus == Track::FS_FILLED) { 3024 // no ramp for the first volume setting 3025 track->mFillingUpStatus = Track::FS_ACTIVE; 3026 if (track->mState == TrackBase::RESUMING) { 3027 track->mState = TrackBase::ACTIVE; 3028 param = AudioMixer::RAMP_VOLUME; 3029 } 3030 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3031 } else if (cblk->server != 0) { 3032 // If the track is stopped before the first frame was mixed, 3033 // do not apply ramp 3034 param = AudioMixer::RAMP_VOLUME; 3035 } 3036 3037 // compute volume for this track 3038 uint32_t vl, vr, va; 3039 if (track->isMuted() || track->isPausing() || 3040 mStreamTypes[track->streamType()].mute) { 3041 vl = vr = va = 0; 3042 if (track->isPausing()) { 3043 track->setPaused(); 3044 } 3045 } else { 3046 3047 // read original volumes with volume control 3048 float typeVolume = mStreamTypes[track->streamType()].volume; 3049 float v = masterVolume * typeVolume; 3050 uint32_t vlr = cblk->getVolumeLR(); 3051 vl = vlr & 0xFFFF; 3052 vr = vlr >> 16; 3053 // track volumes come from shared memory, so can't be trusted and must be clamped 3054 if (vl > MAX_GAIN_INT) { 3055 ALOGV("Track left volume out of range: %04X", vl); 3056 vl = MAX_GAIN_INT; 3057 } 3058 if (vr > MAX_GAIN_INT) { 3059 ALOGV("Track right volume out of range: %04X", vr); 3060 vr = MAX_GAIN_INT; 3061 } 3062 // now apply the master volume and stream type volume 3063 vl = (uint32_t)(v * vl) << 12; 3064 vr = (uint32_t)(v * vr) << 12; 3065 // assuming master volume and stream type volume each go up to 1.0, 3066 // vl and vr are now in 8.24 format 3067 3068 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3069 // send level comes from shared memory and so may be corrupt 3070 if (sendLevel > MAX_GAIN_INT) { 3071 ALOGV("Track send level out of range: %04X", sendLevel); 3072 sendLevel = MAX_GAIN_INT; 3073 } 3074 va = (uint32_t)(v * sendLevel); 3075 } 3076 // Delegate volume control to effect in track effect chain if needed 3077 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3078 // Do not ramp volume if volume is controlled by effect 3079 param = AudioMixer::VOLUME; 3080 track->mHasVolumeController = true; 3081 } else { 3082 // force no volume ramp when volume controller was just disabled or removed 3083 // from effect chain to avoid volume spike 3084 if (track->mHasVolumeController) { 3085 param = AudioMixer::VOLUME; 3086 } 3087 track->mHasVolumeController = false; 3088 } 3089 3090 // Convert volumes from 8.24 to 4.12 format 3091 // This additional clamping is needed in case chain->setVolume_l() overshot 3092 vl = (vl + (1 << 11)) >> 12; 3093 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3094 vr = (vr + (1 << 11)) >> 12; 3095 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3096 3097 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3098 3099 // XXX: these things DON'T need to be done each time 3100 mAudioMixer->setBufferProvider(name, track); 3101 mAudioMixer->enable(name); 3102 3103 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3104 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3105 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3106 mAudioMixer->setParameter( 3107 name, 3108 AudioMixer::TRACK, 3109 AudioMixer::FORMAT, (void *)track->format()); 3110 mAudioMixer->setParameter( 3111 name, 3112 AudioMixer::TRACK, 3113 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3114 mAudioMixer->setParameter( 3115 name, 3116 AudioMixer::RESAMPLE, 3117 AudioMixer::SAMPLE_RATE, 3118 (void *)(cblk->sampleRate)); 3119 mAudioMixer->setParameter( 3120 name, 3121 AudioMixer::TRACK, 3122 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3123 mAudioMixer->setParameter( 3124 name, 3125 AudioMixer::TRACK, 3126 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3127 3128 // reset retry count 3129 track->mRetryCount = kMaxTrackRetries; 3130 3131 // If one track is ready, set the mixer ready if: 3132 // - the mixer was not ready during previous round OR 3133 // - no other track is not ready 3134 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3135 mixerStatus != MIXER_TRACKS_ENABLED) { 3136 mixerStatus = MIXER_TRACKS_READY; 3137 } 3138 } else { 3139 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3140 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3141 track->isStopped() || track->isPaused()) { 3142 // We have consumed all the buffers of this track. 3143 // Remove it from the list of active tracks. 3144 // TODO: use actual buffer filling status instead of latency when available from 3145 // audio HAL 3146 size_t audioHALFrames = 3147 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3148 size_t framesWritten = 3149 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3150 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3151 if (track->isStopped()) { 3152 track->reset(); 3153 } 3154 tracksToRemove->add(track); 3155 } 3156 } else { 3157 // No buffers for this track. Give it a few chances to 3158 // fill a buffer, then remove it from active list. 3159 if (--(track->mRetryCount) <= 0) { 3160 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3161 tracksToRemove->add(track); 3162 // indicate to client process that the track was disabled because of underrun; 3163 // it will then automatically call start() when data is available 3164 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3165 // If one track is not ready, mark the mixer also not ready if: 3166 // - the mixer was ready during previous round OR 3167 // - no other track is ready 3168 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3169 mixerStatus != MIXER_TRACKS_READY) { 3170 mixerStatus = MIXER_TRACKS_ENABLED; 3171 } 3172 } 3173 mAudioMixer->disable(name); 3174 } 3175 3176 } // local variable scope to avoid goto warning 3177track_is_ready: ; 3178 3179 } 3180 3181 // Push the new FastMixer state if necessary 3182 if (didModify) { 3183 state->mFastTracksGen++; 3184 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3185 if (kUseFastMixer == FastMixer_Dynamic && 3186 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3187 state->mCommand = FastMixerState::COLD_IDLE; 3188 state->mColdFutexAddr = &mFastMixerFutex; 3189 state->mColdGen++; 3190 mFastMixerFutex = 0; 3191 if (kUseFastMixer == FastMixer_Dynamic) { 3192 mNormalSink = mOutputSink; 3193 } 3194 // If we go into cold idle, need to wait for acknowledgement 3195 // so that fast mixer stops doing I/O. 3196 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3197 } 3198 sq->end(); 3199 } 3200 if (sq != NULL) { 3201 sq->end(didModify); 3202 sq->push(block); 3203 } 3204 3205 // Now perform the deferred reset on fast tracks that have stopped 3206 while (resetMask != 0) { 3207 size_t i = __builtin_ctz(resetMask); 3208 ALOG_ASSERT(i < count); 3209 resetMask &= ~(1 << i); 3210 sp<Track> t = mActiveTracks[i].promote(); 3211 if (t == 0) continue; 3212 Track* track = t.get(); 3213 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3214 track->reset(); 3215 } 3216 3217 // remove all the tracks that need to be... 3218 count = tracksToRemove->size(); 3219 if (CC_UNLIKELY(count)) { 3220 for (size_t i=0 ; i<count ; i++) { 3221 const sp<Track>& track = tracksToRemove->itemAt(i); 3222 mActiveTracks.remove(track); 3223 if (track->mainBuffer() != mMixBuffer) { 3224 chain = getEffectChain_l(track->sessionId()); 3225 if (chain != 0) { 3226 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3227 chain->decActiveTrackCnt(); 3228 } 3229 } 3230 if (track->isTerminated()) { 3231 removeTrack_l(track); 3232 } 3233 } 3234 } 3235 3236 // mix buffer must be cleared if all tracks are connected to an 3237 // effect chain as in this case the mixer will not write to 3238 // mix buffer and track effects will accumulate into it 3239 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3240 // FIXME as a performance optimization, should remember previous zero status 3241 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3242 } 3243 3244 // if any fast tracks, then status is ready 3245 mMixerStatusIgnoringFastTracks = mixerStatus; 3246 if (fastTracks > 0) { 3247 mixerStatus = MIXER_TRACKS_READY; 3248 } 3249 return mixerStatus; 3250} 3251 3252/* 3253The derived values that are cached: 3254 - mixBufferSize from frame count * frame size 3255 - activeSleepTime from activeSleepTimeUs() 3256 - idleSleepTime from idleSleepTimeUs() 3257 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3258 - maxPeriod from frame count and sample rate (MIXER only) 3259 3260The parameters that affect these derived values are: 3261 - frame count 3262 - frame size 3263 - sample rate 3264 - device type: A2DP or not 3265 - device latency 3266 - format: PCM or not 3267 - active sleep time 3268 - idle sleep time 3269*/ 3270 3271void AudioFlinger::PlaybackThread::cacheParameters_l() 3272{ 3273 mixBufferSize = mNormalFrameCount * mFrameSize; 3274 activeSleepTime = activeSleepTimeUs(); 3275 idleSleepTime = idleSleepTimeUs(); 3276} 3277 3278void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3279{ 3280 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3281 this, streamType, mTracks.size()); 3282 Mutex::Autolock _l(mLock); 3283 3284 size_t size = mTracks.size(); 3285 for (size_t i = 0; i < size; i++) { 3286 sp<Track> t = mTracks[i]; 3287 if (t->streamType() == streamType) { 3288 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3289 t->mCblk->cv.signal(); 3290 } 3291 } 3292} 3293 3294// getTrackName_l() must be called with ThreadBase::mLock held 3295int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3296{ 3297 return mAudioMixer->getTrackName(channelMask); 3298} 3299 3300// deleteTrackName_l() must be called with ThreadBase::mLock held 3301void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3302{ 3303 ALOGV("remove track (%d) and delete from mixer", name); 3304 mAudioMixer->deleteTrackName(name); 3305} 3306 3307// checkForNewParameters_l() must be called with ThreadBase::mLock held 3308bool AudioFlinger::MixerThread::checkForNewParameters_l() 3309{ 3310 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3311 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3312 bool reconfig = false; 3313 3314 while (!mNewParameters.isEmpty()) { 3315 3316 if (mFastMixer != NULL) { 3317 FastMixerStateQueue *sq = mFastMixer->sq(); 3318 FastMixerState *state = sq->begin(); 3319 if (!(state->mCommand & FastMixerState::IDLE)) { 3320 previousCommand = state->mCommand; 3321 state->mCommand = FastMixerState::HOT_IDLE; 3322 sq->end(); 3323 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3324 } else { 3325 sq->end(false /*didModify*/); 3326 } 3327 } 3328 3329 status_t status = NO_ERROR; 3330 String8 keyValuePair = mNewParameters[0]; 3331 AudioParameter param = AudioParameter(keyValuePair); 3332 int value; 3333 3334 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3335 reconfig = true; 3336 } 3337 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3338 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3339 status = BAD_VALUE; 3340 } else { 3341 reconfig = true; 3342 } 3343 } 3344 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3345 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3346 status = BAD_VALUE; 3347 } else { 3348 reconfig = true; 3349 } 3350 } 3351 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3352 // do not accept frame count changes if tracks are open as the track buffer 3353 // size depends on frame count and correct behavior would not be guaranteed 3354 // if frame count is changed after track creation 3355 if (!mTracks.isEmpty()) { 3356 status = INVALID_OPERATION; 3357 } else { 3358 reconfig = true; 3359 } 3360 } 3361 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3362#ifdef ADD_BATTERY_DATA 3363 // when changing the audio output device, call addBatteryData to notify 3364 // the change 3365 if ((int)mDevice != value) { 3366 uint32_t params = 0; 3367 // check whether speaker is on 3368 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3369 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3370 } 3371 3372 int deviceWithoutSpeaker 3373 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3374 // check if any other device (except speaker) is on 3375 if (value & deviceWithoutSpeaker ) { 3376 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3377 } 3378 3379 if (params != 0) { 3380 addBatteryData(params); 3381 } 3382 } 3383#endif 3384 3385 // forward device change to effects that have requested to be 3386 // aware of attached audio device. 3387 mDevice = (uint32_t)value; 3388 for (size_t i = 0; i < mEffectChains.size(); i++) { 3389 mEffectChains[i]->setDevice_l(mDevice); 3390 } 3391 } 3392 3393 if (status == NO_ERROR) { 3394 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3395 keyValuePair.string()); 3396 if (!mStandby && status == INVALID_OPERATION) { 3397 mOutput->stream->common.standby(&mOutput->stream->common); 3398 mStandby = true; 3399 mBytesWritten = 0; 3400 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3401 keyValuePair.string()); 3402 } 3403 if (status == NO_ERROR && reconfig) { 3404 delete mAudioMixer; 3405 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3406 mAudioMixer = NULL; 3407 readOutputParameters(); 3408 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3409 for (size_t i = 0; i < mTracks.size() ; i++) { 3410 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3411 if (name < 0) break; 3412 mTracks[i]->mName = name; 3413 // limit track sample rate to 2 x new output sample rate 3414 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3415 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3416 } 3417 } 3418 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3419 } 3420 } 3421 3422 mNewParameters.removeAt(0); 3423 3424 mParamStatus = status; 3425 mParamCond.signal(); 3426 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3427 // already timed out waiting for the status and will never signal the condition. 3428 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3429 } 3430 3431 if (!(previousCommand & FastMixerState::IDLE)) { 3432 ALOG_ASSERT(mFastMixer != NULL); 3433 FastMixerStateQueue *sq = mFastMixer->sq(); 3434 FastMixerState *state = sq->begin(); 3435 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3436 state->mCommand = previousCommand; 3437 sq->end(); 3438 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3439 } 3440 3441 return reconfig; 3442} 3443 3444status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3445{ 3446 const size_t SIZE = 256; 3447 char buffer[SIZE]; 3448 String8 result; 3449 3450 PlaybackThread::dumpInternals(fd, args); 3451 3452 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3453 result.append(buffer); 3454 write(fd, result.string(), result.size()); 3455 3456 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3457 FastMixerDumpState copy = mFastMixerDumpState; 3458 copy.dump(fd); 3459 3460 // Write the tee output to a .wav file 3461 NBAIO_Source *teeSource = mTeeSource.get(); 3462 if (teeSource != NULL) { 3463 char teePath[64]; 3464 struct timeval tv; 3465 gettimeofday(&tv, NULL); 3466 struct tm tm; 3467 localtime_r(&tv.tv_sec, &tm); 3468 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3469 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3470 if (teeFd >= 0) { 3471 char wavHeader[44]; 3472 memcpy(wavHeader, 3473 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3474 sizeof(wavHeader)); 3475 NBAIO_Format format = teeSource->format(); 3476 unsigned channelCount = Format_channelCount(format); 3477 ALOG_ASSERT(channelCount <= FCC_2); 3478 unsigned sampleRate = Format_sampleRate(format); 3479 wavHeader[22] = channelCount; // number of channels 3480 wavHeader[24] = sampleRate; // sample rate 3481 wavHeader[25] = sampleRate >> 8; 3482 wavHeader[32] = channelCount * 2; // block alignment 3483 write(teeFd, wavHeader, sizeof(wavHeader)); 3484 size_t total = 0; 3485 bool firstRead = true; 3486 for (;;) { 3487#define TEE_SINK_READ 1024 3488 short buffer[TEE_SINK_READ * FCC_2]; 3489 size_t count = TEE_SINK_READ; 3490 ssize_t actual = teeSource->read(buffer, count); 3491 bool wasFirstRead = firstRead; 3492 firstRead = false; 3493 if (actual <= 0) { 3494 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3495 continue; 3496 } 3497 break; 3498 } 3499 ALOG_ASSERT(actual <= count); 3500 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3501 total += actual; 3502 } 3503 lseek(teeFd, (off_t) 4, SEEK_SET); 3504 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3505 write(teeFd, &temp, sizeof(temp)); 3506 lseek(teeFd, (off_t) 40, SEEK_SET); 3507 temp = total * channelCount * sizeof(short); 3508 write(teeFd, &temp, sizeof(temp)); 3509 close(teeFd); 3510 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3511 } else { 3512 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3513 } 3514 } 3515 3516 return NO_ERROR; 3517} 3518 3519uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3520{ 3521 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3522} 3523 3524uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3525{ 3526 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3527} 3528 3529void AudioFlinger::MixerThread::cacheParameters_l() 3530{ 3531 PlaybackThread::cacheParameters_l(); 3532 3533 // FIXME: Relaxed timing because of a certain device that can't meet latency 3534 // Should be reduced to 2x after the vendor fixes the driver issue 3535 // increase threshold again due to low power audio mode. The way this warning 3536 // threshold is calculated and its usefulness should be reconsidered anyway. 3537 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3538} 3539 3540// ---------------------------------------------------------------------------- 3541AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3542 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3543 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3544 // mLeftVolFloat, mRightVolFloat 3545 // mLeftVolShort, mRightVolShort 3546{ 3547} 3548 3549AudioFlinger::DirectOutputThread::~DirectOutputThread() 3550{ 3551} 3552 3553AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3554 Vector< sp<Track> > *tracksToRemove 3555) 3556{ 3557 sp<Track> trackToRemove; 3558 3559 mixer_state mixerStatus = MIXER_IDLE; 3560 3561 // find out which tracks need to be processed 3562 if (mActiveTracks.size() != 0) { 3563 sp<Track> t = mActiveTracks[0].promote(); 3564 // The track died recently 3565 if (t == 0) return MIXER_IDLE; 3566 3567 Track* const track = t.get(); 3568 audio_track_cblk_t* cblk = track->cblk(); 3569 3570 // The first time a track is added we wait 3571 // for all its buffers to be filled before processing it 3572 if (cblk->framesReady() && track->isReady() && 3573 !track->isPaused() && !track->isTerminated()) 3574 { 3575 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3576 3577 if (track->mFillingUpStatus == Track::FS_FILLED) { 3578 track->mFillingUpStatus = Track::FS_ACTIVE; 3579 mLeftVolFloat = mRightVolFloat = 0; 3580 mLeftVolShort = mRightVolShort = 0; 3581 if (track->mState == TrackBase::RESUMING) { 3582 track->mState = TrackBase::ACTIVE; 3583 rampVolume = true; 3584 } 3585 } else if (cblk->server != 0) { 3586 // If the track is stopped before the first frame was mixed, 3587 // do not apply ramp 3588 rampVolume = true; 3589 } 3590 // compute volume for this track 3591 float left, right; 3592 if (track->isMuted() || mMasterMute || track->isPausing() || 3593 mStreamTypes[track->streamType()].mute) { 3594 left = right = 0; 3595 if (track->isPausing()) { 3596 track->setPaused(); 3597 } 3598 } else { 3599 float typeVolume = mStreamTypes[track->streamType()].volume; 3600 float v = mMasterVolume * typeVolume; 3601 uint32_t vlr = cblk->getVolumeLR(); 3602 float v_clamped = v * (vlr & 0xFFFF); 3603 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3604 left = v_clamped/MAX_GAIN; 3605 v_clamped = v * (vlr >> 16); 3606 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3607 right = v_clamped/MAX_GAIN; 3608 } 3609 3610 if (left != mLeftVolFloat || right != mRightVolFloat) { 3611 mLeftVolFloat = left; 3612 mRightVolFloat = right; 3613 3614 // If audio HAL implements volume control, 3615 // force software volume to nominal value 3616 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3617 left = 1.0f; 3618 right = 1.0f; 3619 } 3620 3621 // Convert volumes from float to 8.24 3622 uint32_t vl = (uint32_t)(left * (1 << 24)); 3623 uint32_t vr = (uint32_t)(right * (1 << 24)); 3624 3625 // Delegate volume control to effect in track effect chain if needed 3626 // only one effect chain can be present on DirectOutputThread, so if 3627 // there is one, the track is connected to it 3628 if (!mEffectChains.isEmpty()) { 3629 // Do not ramp volume if volume is controlled by effect 3630 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3631 rampVolume = false; 3632 } 3633 } 3634 3635 // Convert volumes from 8.24 to 4.12 format 3636 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3637 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3638 leftVol = (uint16_t)v_clamped; 3639 v_clamped = (vr + (1 << 11)) >> 12; 3640 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3641 rightVol = (uint16_t)v_clamped; 3642 } else { 3643 leftVol = mLeftVolShort; 3644 rightVol = mRightVolShort; 3645 rampVolume = false; 3646 } 3647 3648 // reset retry count 3649 track->mRetryCount = kMaxTrackRetriesDirect; 3650 mActiveTrack = t; 3651 mixerStatus = MIXER_TRACKS_READY; 3652 } else { 3653 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3654 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3655 // We have consumed all the buffers of this track. 3656 // Remove it from the list of active tracks. 3657 // TODO: implement behavior for compressed audio 3658 size_t audioHALFrames = 3659 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3660 size_t framesWritten = 3661 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3662 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3663 if (track->isStopped()) { 3664 track->reset(); 3665 } 3666 trackToRemove = track; 3667 } 3668 } else { 3669 // No buffers for this track. Give it a few chances to 3670 // fill a buffer, then remove it from active list. 3671 if (--(track->mRetryCount) <= 0) { 3672 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3673 trackToRemove = track; 3674 } else { 3675 mixerStatus = MIXER_TRACKS_ENABLED; 3676 } 3677 } 3678 } 3679 } 3680 3681 // FIXME merge this with similar code for removing multiple tracks 3682 // remove all the tracks that need to be... 3683 if (CC_UNLIKELY(trackToRemove != 0)) { 3684 tracksToRemove->add(trackToRemove); 3685 mActiveTracks.remove(trackToRemove); 3686 if (!mEffectChains.isEmpty()) { 3687 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3688 trackToRemove->sessionId()); 3689 mEffectChains[0]->decActiveTrackCnt(); 3690 } 3691 if (trackToRemove->isTerminated()) { 3692 removeTrack_l(trackToRemove); 3693 } 3694 } 3695 3696 return mixerStatus; 3697} 3698 3699void AudioFlinger::DirectOutputThread::threadLoop_mix() 3700{ 3701 AudioBufferProvider::Buffer buffer; 3702 size_t frameCount = mFrameCount; 3703 int8_t *curBuf = (int8_t *)mMixBuffer; 3704 // output audio to hardware 3705 while (frameCount) { 3706 buffer.frameCount = frameCount; 3707 mActiveTrack->getNextBuffer(&buffer); 3708 if (CC_UNLIKELY(buffer.raw == NULL)) { 3709 memset(curBuf, 0, frameCount * mFrameSize); 3710 break; 3711 } 3712 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3713 frameCount -= buffer.frameCount; 3714 curBuf += buffer.frameCount * mFrameSize; 3715 mActiveTrack->releaseBuffer(&buffer); 3716 } 3717 sleepTime = 0; 3718 standbyTime = systemTime() + standbyDelay; 3719 mActiveTrack.clear(); 3720 3721 // apply volume 3722 3723 // Do not apply volume on compressed audio 3724 if (!audio_is_linear_pcm(mFormat)) { 3725 return; 3726 } 3727 3728 // convert to signed 16 bit before volume calculation 3729 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3730 size_t count = mFrameCount * mChannelCount; 3731 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3732 int16_t *dst = mMixBuffer + count-1; 3733 while (count--) { 3734 *dst-- = (int16_t)(*src--^0x80) << 8; 3735 } 3736 } 3737 3738 frameCount = mFrameCount; 3739 int16_t *out = mMixBuffer; 3740 if (rampVolume) { 3741 if (mChannelCount == 1) { 3742 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3743 int32_t vlInc = d / (int32_t)frameCount; 3744 int32_t vl = ((int32_t)mLeftVolShort << 16); 3745 do { 3746 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3747 out++; 3748 vl += vlInc; 3749 } while (--frameCount); 3750 3751 } else { 3752 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3753 int32_t vlInc = d / (int32_t)frameCount; 3754 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3755 int32_t vrInc = d / (int32_t)frameCount; 3756 int32_t vl = ((int32_t)mLeftVolShort << 16); 3757 int32_t vr = ((int32_t)mRightVolShort << 16); 3758 do { 3759 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3760 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3761 out += 2; 3762 vl += vlInc; 3763 vr += vrInc; 3764 } while (--frameCount); 3765 } 3766 } else { 3767 if (mChannelCount == 1) { 3768 do { 3769 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3770 out++; 3771 } while (--frameCount); 3772 } else { 3773 do { 3774 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3775 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3776 out += 2; 3777 } while (--frameCount); 3778 } 3779 } 3780 3781 // convert back to unsigned 8 bit after volume calculation 3782 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3783 size_t count = mFrameCount * mChannelCount; 3784 int16_t *src = mMixBuffer; 3785 uint8_t *dst = (uint8_t *)mMixBuffer; 3786 while (count--) { 3787 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3788 } 3789 } 3790 3791 mLeftVolShort = leftVol; 3792 mRightVolShort = rightVol; 3793} 3794 3795void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3796{ 3797 if (sleepTime == 0) { 3798 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3799 sleepTime = activeSleepTime; 3800 } else { 3801 sleepTime = idleSleepTime; 3802 } 3803 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3804 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3805 sleepTime = 0; 3806 } 3807} 3808 3809// getTrackName_l() must be called with ThreadBase::mLock held 3810int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3811{ 3812 return 0; 3813} 3814 3815// deleteTrackName_l() must be called with ThreadBase::mLock held 3816void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3817{ 3818} 3819 3820// checkForNewParameters_l() must be called with ThreadBase::mLock held 3821bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3822{ 3823 bool reconfig = false; 3824 3825 while (!mNewParameters.isEmpty()) { 3826 status_t status = NO_ERROR; 3827 String8 keyValuePair = mNewParameters[0]; 3828 AudioParameter param = AudioParameter(keyValuePair); 3829 int value; 3830 3831 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3832 // do not accept frame count changes if tracks are open as the track buffer 3833 // size depends on frame count and correct behavior would not be garantied 3834 // if frame count is changed after track creation 3835 if (!mTracks.isEmpty()) { 3836 status = INVALID_OPERATION; 3837 } else { 3838 reconfig = true; 3839 } 3840 } 3841 if (status == NO_ERROR) { 3842 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3843 keyValuePair.string()); 3844 if (!mStandby && status == INVALID_OPERATION) { 3845 mOutput->stream->common.standby(&mOutput->stream->common); 3846 mStandby = true; 3847 mBytesWritten = 0; 3848 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3849 keyValuePair.string()); 3850 } 3851 if (status == NO_ERROR && reconfig) { 3852 readOutputParameters(); 3853 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3854 } 3855 } 3856 3857 mNewParameters.removeAt(0); 3858 3859 mParamStatus = status; 3860 mParamCond.signal(); 3861 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3862 // already timed out waiting for the status and will never signal the condition. 3863 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3864 } 3865 return reconfig; 3866} 3867 3868uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3869{ 3870 uint32_t time; 3871 if (audio_is_linear_pcm(mFormat)) { 3872 time = PlaybackThread::activeSleepTimeUs(); 3873 } else { 3874 time = 10000; 3875 } 3876 return time; 3877} 3878 3879uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3880{ 3881 uint32_t time; 3882 if (audio_is_linear_pcm(mFormat)) { 3883 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3884 } else { 3885 time = 10000; 3886 } 3887 return time; 3888} 3889 3890uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3891{ 3892 uint32_t time; 3893 if (audio_is_linear_pcm(mFormat)) { 3894 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3895 } else { 3896 time = 10000; 3897 } 3898 return time; 3899} 3900 3901void AudioFlinger::DirectOutputThread::cacheParameters_l() 3902{ 3903 PlaybackThread::cacheParameters_l(); 3904 3905 // use shorter standby delay as on normal output to release 3906 // hardware resources as soon as possible 3907 standbyDelay = microseconds(activeSleepTime*2); 3908} 3909 3910// ---------------------------------------------------------------------------- 3911 3912AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3913 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3914 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3915 mWaitTimeMs(UINT_MAX) 3916{ 3917 addOutputTrack(mainThread); 3918} 3919 3920AudioFlinger::DuplicatingThread::~DuplicatingThread() 3921{ 3922 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3923 mOutputTracks[i]->destroy(); 3924 } 3925} 3926 3927void AudioFlinger::DuplicatingThread::threadLoop_mix() 3928{ 3929 // mix buffers... 3930 if (outputsReady(outputTracks)) { 3931 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3932 } else { 3933 memset(mMixBuffer, 0, mixBufferSize); 3934 } 3935 sleepTime = 0; 3936 writeFrames = mNormalFrameCount; 3937} 3938 3939void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3940{ 3941 if (sleepTime == 0) { 3942 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3943 sleepTime = activeSleepTime; 3944 } else { 3945 sleepTime = idleSleepTime; 3946 } 3947 } else if (mBytesWritten != 0) { 3948 // flush remaining overflow buffers in output tracks 3949 for (size_t i = 0; i < outputTracks.size(); i++) { 3950 if (outputTracks[i]->isActive()) { 3951 sleepTime = 0; 3952 writeFrames = 0; 3953 memset(mMixBuffer, 0, mixBufferSize); 3954 break; 3955 } 3956 } 3957 } 3958} 3959 3960void AudioFlinger::DuplicatingThread::threadLoop_write() 3961{ 3962 standbyTime = systemTime() + standbyDelay; 3963 for (size_t i = 0; i < outputTracks.size(); i++) { 3964 outputTracks[i]->write(mMixBuffer, writeFrames); 3965 } 3966 mBytesWritten += mixBufferSize; 3967} 3968 3969void AudioFlinger::DuplicatingThread::threadLoop_standby() 3970{ 3971 // DuplicatingThread implements standby by stopping all tracks 3972 for (size_t i = 0; i < outputTracks.size(); i++) { 3973 outputTracks[i]->stop(); 3974 } 3975} 3976 3977void AudioFlinger::DuplicatingThread::saveOutputTracks() 3978{ 3979 outputTracks = mOutputTracks; 3980} 3981 3982void AudioFlinger::DuplicatingThread::clearOutputTracks() 3983{ 3984 outputTracks.clear(); 3985} 3986 3987void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3988{ 3989 Mutex::Autolock _l(mLock); 3990 // FIXME explain this formula 3991 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3992 OutputTrack *outputTrack = new OutputTrack(thread, 3993 this, 3994 mSampleRate, 3995 mFormat, 3996 mChannelMask, 3997 frameCount); 3998 if (outputTrack->cblk() != NULL) { 3999 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4000 mOutputTracks.add(outputTrack); 4001 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4002 updateWaitTime_l(); 4003 } 4004} 4005 4006void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4007{ 4008 Mutex::Autolock _l(mLock); 4009 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4010 if (mOutputTracks[i]->thread() == thread) { 4011 mOutputTracks[i]->destroy(); 4012 mOutputTracks.removeAt(i); 4013 updateWaitTime_l(); 4014 return; 4015 } 4016 } 4017 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4018} 4019 4020// caller must hold mLock 4021void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4022{ 4023 mWaitTimeMs = UINT_MAX; 4024 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4025 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4026 if (strong != 0) { 4027 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4028 if (waitTimeMs < mWaitTimeMs) { 4029 mWaitTimeMs = waitTimeMs; 4030 } 4031 } 4032 } 4033} 4034 4035 4036bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4037{ 4038 for (size_t i = 0; i < outputTracks.size(); i++) { 4039 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4040 if (thread == 0) { 4041 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4042 return false; 4043 } 4044 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4045 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4046 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4047 return false; 4048 } 4049 } 4050 return true; 4051} 4052 4053uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4054{ 4055 return (mWaitTimeMs * 1000) / 2; 4056} 4057 4058void AudioFlinger::DuplicatingThread::cacheParameters_l() 4059{ 4060 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4061 updateWaitTime_l(); 4062 4063 MixerThread::cacheParameters_l(); 4064} 4065 4066// ---------------------------------------------------------------------------- 4067 4068// TrackBase constructor must be called with AudioFlinger::mLock held 4069AudioFlinger::ThreadBase::TrackBase::TrackBase( 4070 ThreadBase *thread, 4071 const sp<Client>& client, 4072 uint32_t sampleRate, 4073 audio_format_t format, 4074 uint32_t channelMask, 4075 int frameCount, 4076 const sp<IMemory>& sharedBuffer, 4077 int sessionId) 4078 : RefBase(), 4079 mThread(thread), 4080 mClient(client), 4081 mCblk(NULL), 4082 // mBuffer 4083 // mBufferEnd 4084 mFrameCount(0), 4085 mState(IDLE), 4086 mSampleRate(sampleRate), 4087 mFormat(format), 4088 mStepServerFailed(false), 4089 mSessionId(sessionId) 4090 // mChannelCount 4091 // mChannelMask 4092{ 4093 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4094 4095 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4096 size_t size = sizeof(audio_track_cblk_t); 4097 uint8_t channelCount = popcount(channelMask); 4098 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4099 if (sharedBuffer == 0) { 4100 size += bufferSize; 4101 } 4102 4103 if (client != NULL) { 4104 mCblkMemory = client->heap()->allocate(size); 4105 if (mCblkMemory != 0) { 4106 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4107 if (mCblk != NULL) { // construct the shared structure in-place. 4108 new(mCblk) audio_track_cblk_t(); 4109 // clear all buffers 4110 mCblk->frameCount = frameCount; 4111 mCblk->sampleRate = sampleRate; 4112// uncomment the following lines to quickly test 32-bit wraparound 4113// mCblk->user = 0xffff0000; 4114// mCblk->server = 0xffff0000; 4115// mCblk->userBase = 0xffff0000; 4116// mCblk->serverBase = 0xffff0000; 4117 mChannelCount = channelCount; 4118 mChannelMask = channelMask; 4119 if (sharedBuffer == 0) { 4120 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4121 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4122 // Force underrun condition to avoid false underrun callback until first data is 4123 // written to buffer (other flags are cleared) 4124 mCblk->flags = CBLK_UNDERRUN_ON; 4125 } else { 4126 mBuffer = sharedBuffer->pointer(); 4127 } 4128 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4129 } 4130 } else { 4131 ALOGE("not enough memory for AudioTrack size=%u", size); 4132 client->heap()->dump("AudioTrack"); 4133 return; 4134 } 4135 } else { 4136 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4137 // construct the shared structure in-place. 4138 new(mCblk) audio_track_cblk_t(); 4139 // clear all buffers 4140 mCblk->frameCount = frameCount; 4141 mCblk->sampleRate = sampleRate; 4142// uncomment the following lines to quickly test 32-bit wraparound 4143// mCblk->user = 0xffff0000; 4144// mCblk->server = 0xffff0000; 4145// mCblk->userBase = 0xffff0000; 4146// mCblk->serverBase = 0xffff0000; 4147 mChannelCount = channelCount; 4148 mChannelMask = channelMask; 4149 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4150 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4151 // Force underrun condition to avoid false underrun callback until first data is 4152 // written to buffer (other flags are cleared) 4153 mCblk->flags = CBLK_UNDERRUN_ON; 4154 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4155 } 4156} 4157 4158AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4159{ 4160 if (mCblk != NULL) { 4161 if (mClient == 0) { 4162 delete mCblk; 4163 } else { 4164 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4165 } 4166 } 4167 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4168 if (mClient != 0) { 4169 // Client destructor must run with AudioFlinger mutex locked 4170 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4171 // If the client's reference count drops to zero, the associated destructor 4172 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4173 // relying on the automatic clear() at end of scope. 4174 mClient.clear(); 4175 } 4176} 4177 4178// AudioBufferProvider interface 4179// getNextBuffer() = 0; 4180// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4181void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4182{ 4183 buffer->raw = NULL; 4184 mFrameCount = buffer->frameCount; 4185 // FIXME See note at getNextBuffer() 4186 (void) step(); // ignore return value of step() 4187 buffer->frameCount = 0; 4188} 4189 4190bool AudioFlinger::ThreadBase::TrackBase::step() { 4191 bool result; 4192 audio_track_cblk_t* cblk = this->cblk(); 4193 4194 result = cblk->stepServer(mFrameCount); 4195 if (!result) { 4196 ALOGV("stepServer failed acquiring cblk mutex"); 4197 mStepServerFailed = true; 4198 } 4199 return result; 4200} 4201 4202void AudioFlinger::ThreadBase::TrackBase::reset() { 4203 audio_track_cblk_t* cblk = this->cblk(); 4204 4205 cblk->user = 0; 4206 cblk->server = 0; 4207 cblk->userBase = 0; 4208 cblk->serverBase = 0; 4209 mStepServerFailed = false; 4210 ALOGV("TrackBase::reset"); 4211} 4212 4213int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4214 return (int)mCblk->sampleRate; 4215} 4216 4217void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4218 audio_track_cblk_t* cblk = this->cblk(); 4219 size_t frameSize = cblk->frameSize; 4220 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4221 int8_t *bufferEnd = bufferStart + frames * frameSize; 4222 4223 // Check validity of returned pointer in case the track control block would have been corrupted. 4224 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4225 "TrackBase::getBuffer buffer out of range:\n" 4226 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4227 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4228 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4229 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4230 4231 return bufferStart; 4232} 4233 4234status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4235{ 4236 mSyncEvents.add(event); 4237 return NO_ERROR; 4238} 4239 4240// ---------------------------------------------------------------------------- 4241 4242// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4243AudioFlinger::PlaybackThread::Track::Track( 4244 PlaybackThread *thread, 4245 const sp<Client>& client, 4246 audio_stream_type_t streamType, 4247 uint32_t sampleRate, 4248 audio_format_t format, 4249 uint32_t channelMask, 4250 int frameCount, 4251 const sp<IMemory>& sharedBuffer, 4252 int sessionId, 4253 IAudioFlinger::track_flags_t flags) 4254 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4255 mMute(false), 4256 mFillingUpStatus(FS_INVALID), 4257 // mRetryCount initialized later when needed 4258 mSharedBuffer(sharedBuffer), 4259 mStreamType(streamType), 4260 mName(-1), // see note below 4261 mMainBuffer(thread->mixBuffer()), 4262 mAuxBuffer(NULL), 4263 mAuxEffectId(0), mHasVolumeController(false), 4264 mPresentationCompleteFrames(0), 4265 mFlags(flags), 4266 mFastIndex(-1), 4267 mUnderrunCount(0), 4268 mCachedVolume(1.0) 4269{ 4270 if (mCblk != NULL) { 4271 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4272 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4273 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4274 if (flags & IAudioFlinger::TRACK_FAST) { 4275 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4276 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4277 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4278 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4279 // FIXME This is too eager. We allocate a fast track index before the 4280 // fast track becomes active. Since fast tracks are a scarce resource, 4281 // this means we are potentially denying other more important fast tracks from 4282 // being created. It would be better to allocate the index dynamically. 4283 mFastIndex = i; 4284 // Read the initial underruns because this field is never cleared by the fast mixer 4285 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4286 thread->mFastTrackAvailMask &= ~(1 << i); 4287 } 4288 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4289 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4290 if (mName < 0) { 4291 ALOGE("no more track names available"); 4292 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4293 // then we leak a fast track index. Should swap these two sections, or better yet 4294 // only allocate a normal mixer name for normal tracks. 4295 } 4296 } 4297 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4298} 4299 4300AudioFlinger::PlaybackThread::Track::~Track() 4301{ 4302 ALOGV("PlaybackThread::Track destructor"); 4303 sp<ThreadBase> thread = mThread.promote(); 4304 if (thread != 0) { 4305 Mutex::Autolock _l(thread->mLock); 4306 mState = TERMINATED; 4307 } 4308} 4309 4310void AudioFlinger::PlaybackThread::Track::destroy() 4311{ 4312 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4313 // by removing it from mTracks vector, so there is a risk that this Tracks's 4314 // destructor is called. As the destructor needs to lock mLock, 4315 // we must acquire a strong reference on this Track before locking mLock 4316 // here so that the destructor is called only when exiting this function. 4317 // On the other hand, as long as Track::destroy() is only called by 4318 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4319 // this Track with its member mTrack. 4320 sp<Track> keep(this); 4321 { // scope for mLock 4322 sp<ThreadBase> thread = mThread.promote(); 4323 if (thread != 0) { 4324 if (!isOutputTrack()) { 4325 if (mState == ACTIVE || mState == RESUMING) { 4326 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4327 4328#ifdef ADD_BATTERY_DATA 4329 // to track the speaker usage 4330 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4331#endif 4332 } 4333 AudioSystem::releaseOutput(thread->id()); 4334 } 4335 Mutex::Autolock _l(thread->mLock); 4336 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4337 playbackThread->destroyTrack_l(this); 4338 } 4339 } 4340} 4341 4342/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4343{ 4344 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4345 " Server User Main buf Aux Buf Flags FastUnder\n"); 4346} 4347 4348void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4349{ 4350 uint32_t vlr = mCblk->getVolumeLR(); 4351 if (isFastTrack()) { 4352 sprintf(buffer, " F %2d", mFastIndex); 4353 } else { 4354 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4355 } 4356 track_state state = mState; 4357 char stateChar; 4358 switch (state) { 4359 case IDLE: 4360 stateChar = 'I'; 4361 break; 4362 case TERMINATED: 4363 stateChar = 'T'; 4364 break; 4365 case STOPPING_1: 4366 stateChar = 's'; 4367 break; 4368 case STOPPING_2: 4369 stateChar = '5'; 4370 break; 4371 case STOPPED: 4372 stateChar = 'S'; 4373 break; 4374 case RESUMING: 4375 stateChar = 'R'; 4376 break; 4377 case ACTIVE: 4378 stateChar = 'A'; 4379 break; 4380 case PAUSING: 4381 stateChar = 'p'; 4382 break; 4383 case PAUSED: 4384 stateChar = 'P'; 4385 break; 4386 case FLUSHED: 4387 stateChar = 'F'; 4388 break; 4389 default: 4390 stateChar = '?'; 4391 break; 4392 } 4393 char nowInUnderrun; 4394 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4395 case UNDERRUN_FULL: 4396 nowInUnderrun = ' '; 4397 break; 4398 case UNDERRUN_PARTIAL: 4399 nowInUnderrun = '<'; 4400 break; 4401 case UNDERRUN_EMPTY: 4402 nowInUnderrun = '*'; 4403 break; 4404 default: 4405 nowInUnderrun = '?'; 4406 break; 4407 } 4408 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4409 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4410 (mClient == 0) ? getpid_cached : mClient->pid(), 4411 mStreamType, 4412 mFormat, 4413 mChannelMask, 4414 mSessionId, 4415 mFrameCount, 4416 mCblk->frameCount, 4417 stateChar, 4418 mMute, 4419 mFillingUpStatus, 4420 mCblk->sampleRate, 4421 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4422 20.0 * log10((vlr >> 16) / 4096.0), 4423 mCblk->server, 4424 mCblk->user, 4425 (int)mMainBuffer, 4426 (int)mAuxBuffer, 4427 mCblk->flags, 4428 mUnderrunCount, 4429 nowInUnderrun); 4430} 4431 4432// AudioBufferProvider interface 4433status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4434 AudioBufferProvider::Buffer* buffer, int64_t pts) 4435{ 4436 audio_track_cblk_t* cblk = this->cblk(); 4437 uint32_t framesReady; 4438 uint32_t framesReq = buffer->frameCount; 4439 4440 // Check if last stepServer failed, try to step now 4441 if (mStepServerFailed) { 4442 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4443 // Since the fast mixer is higher priority than client callback thread, 4444 // it does not result in priority inversion for client. 4445 // But a non-blocking solution would be preferable to avoid 4446 // fast mixer being unable to tryLock(), and 4447 // to avoid the extra context switches if the client wakes up, 4448 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4449 if (!step()) goto getNextBuffer_exit; 4450 ALOGV("stepServer recovered"); 4451 mStepServerFailed = false; 4452 } 4453 4454 // FIXME Same as above 4455 framesReady = cblk->framesReady(); 4456 4457 if (CC_LIKELY(framesReady)) { 4458 uint32_t s = cblk->server; 4459 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4460 4461 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4462 if (framesReq > framesReady) { 4463 framesReq = framesReady; 4464 } 4465 if (framesReq > bufferEnd - s) { 4466 framesReq = bufferEnd - s; 4467 } 4468 4469 buffer->raw = getBuffer(s, framesReq); 4470 if (buffer->raw == NULL) goto getNextBuffer_exit; 4471 4472 buffer->frameCount = framesReq; 4473 return NO_ERROR; 4474 } 4475 4476getNextBuffer_exit: 4477 buffer->raw = NULL; 4478 buffer->frameCount = 0; 4479 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4480 return NOT_ENOUGH_DATA; 4481} 4482 4483// Note that framesReady() takes a mutex on the control block using tryLock(). 4484// This could result in priority inversion if framesReady() is called by the normal mixer, 4485// as the normal mixer thread runs at lower 4486// priority than the client's callback thread: there is a short window within framesReady() 4487// during which the normal mixer could be preempted, and the client callback would block. 4488// Another problem can occur if framesReady() is called by the fast mixer: 4489// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4490// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4491size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4492 return mCblk->framesReady(); 4493} 4494 4495// Don't call for fast tracks; the framesReady() could result in priority inversion 4496bool AudioFlinger::PlaybackThread::Track::isReady() const { 4497 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4498 4499 if (framesReady() >= mCblk->frameCount || 4500 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4501 mFillingUpStatus = FS_FILLED; 4502 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4503 return true; 4504 } 4505 return false; 4506} 4507 4508status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4509 int triggerSession) 4510{ 4511 status_t status = NO_ERROR; 4512 ALOGV("start(%d), calling pid %d session %d", 4513 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4514 4515 sp<ThreadBase> thread = mThread.promote(); 4516 if (thread != 0) { 4517 Mutex::Autolock _l(thread->mLock); 4518 track_state state = mState; 4519 // here the track could be either new, or restarted 4520 // in both cases "unstop" the track 4521 if (mState == PAUSED) { 4522 mState = TrackBase::RESUMING; 4523 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4524 } else { 4525 mState = TrackBase::ACTIVE; 4526 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4527 } 4528 4529 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4530 thread->mLock.unlock(); 4531 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4532 thread->mLock.lock(); 4533 4534#ifdef ADD_BATTERY_DATA 4535 // to track the speaker usage 4536 if (status == NO_ERROR) { 4537 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4538 } 4539#endif 4540 } 4541 if (status == NO_ERROR) { 4542 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4543 playbackThread->addTrack_l(this); 4544 } else { 4545 mState = state; 4546 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4547 } 4548 } else { 4549 status = BAD_VALUE; 4550 } 4551 return status; 4552} 4553 4554void AudioFlinger::PlaybackThread::Track::stop() 4555{ 4556 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4557 sp<ThreadBase> thread = mThread.promote(); 4558 if (thread != 0) { 4559 Mutex::Autolock _l(thread->mLock); 4560 track_state state = mState; 4561 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4562 // If the track is not active (PAUSED and buffers full), flush buffers 4563 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4564 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4565 reset(); 4566 mState = STOPPED; 4567 } else if (!isFastTrack()) { 4568 mState = STOPPED; 4569 } else { 4570 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4571 // and then to STOPPED and reset() when presentation is complete 4572 mState = STOPPING_1; 4573 } 4574 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4575 } 4576 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4577 thread->mLock.unlock(); 4578 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4579 thread->mLock.lock(); 4580 4581#ifdef ADD_BATTERY_DATA 4582 // to track the speaker usage 4583 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4584#endif 4585 } 4586 } 4587} 4588 4589void AudioFlinger::PlaybackThread::Track::pause() 4590{ 4591 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4592 sp<ThreadBase> thread = mThread.promote(); 4593 if (thread != 0) { 4594 Mutex::Autolock _l(thread->mLock); 4595 if (mState == ACTIVE || mState == RESUMING) { 4596 mState = PAUSING; 4597 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4598 if (!isOutputTrack()) { 4599 thread->mLock.unlock(); 4600 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4601 thread->mLock.lock(); 4602 4603#ifdef ADD_BATTERY_DATA 4604 // to track the speaker usage 4605 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4606#endif 4607 } 4608 } 4609 } 4610} 4611 4612void AudioFlinger::PlaybackThread::Track::flush() 4613{ 4614 ALOGV("flush(%d)", mName); 4615 sp<ThreadBase> thread = mThread.promote(); 4616 if (thread != 0) { 4617 Mutex::Autolock _l(thread->mLock); 4618 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4619 mState != PAUSING) { 4620 return; 4621 } 4622 // No point remaining in PAUSED state after a flush => go to 4623 // FLUSHED state 4624 mState = FLUSHED; 4625 // do not reset the track if it is still in the process of being stopped or paused. 4626 // this will be done by prepareTracks_l() when the track is stopped. 4627 // prepareTracks_l() will see mState == FLUSHED, then 4628 // remove from active track list, reset(), and trigger presentation complete 4629 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4630 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4631 reset(); 4632 } 4633 } 4634} 4635 4636void AudioFlinger::PlaybackThread::Track::reset() 4637{ 4638 // Do not reset twice to avoid discarding data written just after a flush and before 4639 // the audioflinger thread detects the track is stopped. 4640 if (!mResetDone) { 4641 TrackBase::reset(); 4642 // Force underrun condition to avoid false underrun callback until first data is 4643 // written to buffer 4644 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4645 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4646 mFillingUpStatus = FS_FILLING; 4647 mResetDone = true; 4648 if (mState == FLUSHED) { 4649 mState = IDLE; 4650 } 4651 } 4652} 4653 4654void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4655{ 4656 mMute = muted; 4657} 4658 4659status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4660{ 4661 status_t status = DEAD_OBJECT; 4662 sp<ThreadBase> thread = mThread.promote(); 4663 if (thread != 0) { 4664 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4665 status = playbackThread->attachAuxEffect(this, EffectId); 4666 } 4667 return status; 4668} 4669 4670void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4671{ 4672 mAuxEffectId = EffectId; 4673 mAuxBuffer = buffer; 4674} 4675 4676bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4677 size_t audioHalFrames) 4678{ 4679 // a track is considered presented when the total number of frames written to audio HAL 4680 // corresponds to the number of frames written when presentationComplete() is called for the 4681 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4682 if (mPresentationCompleteFrames == 0) { 4683 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4684 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4685 mPresentationCompleteFrames, audioHalFrames); 4686 } 4687 if (framesWritten >= mPresentationCompleteFrames) { 4688 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4689 mSessionId, framesWritten); 4690 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4691 return true; 4692 } 4693 return false; 4694} 4695 4696void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4697{ 4698 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4699 if (mSyncEvents[i]->type() == type) { 4700 mSyncEvents[i]->trigger(); 4701 mSyncEvents.removeAt(i); 4702 i--; 4703 } 4704 } 4705} 4706 4707// implement VolumeBufferProvider interface 4708 4709uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4710{ 4711 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4712 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4713 uint32_t vlr = mCblk->getVolumeLR(); 4714 uint32_t vl = vlr & 0xFFFF; 4715 uint32_t vr = vlr >> 16; 4716 // track volumes come from shared memory, so can't be trusted and must be clamped 4717 if (vl > MAX_GAIN_INT) { 4718 vl = MAX_GAIN_INT; 4719 } 4720 if (vr > MAX_GAIN_INT) { 4721 vr = MAX_GAIN_INT; 4722 } 4723 // now apply the cached master volume and stream type volume; 4724 // this is trusted but lacks any synchronization or barrier so may be stale 4725 float v = mCachedVolume; 4726 vl *= v; 4727 vr *= v; 4728 // re-combine into U4.16 4729 vlr = (vr << 16) | (vl & 0xFFFF); 4730 // FIXME look at mute, pause, and stop flags 4731 return vlr; 4732} 4733 4734status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4735{ 4736 if (mState == TERMINATED || mState == PAUSED || 4737 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4738 (mState == STOPPED)))) { 4739 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4740 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4741 event->cancel(); 4742 return INVALID_OPERATION; 4743 } 4744 TrackBase::setSyncEvent(event); 4745 return NO_ERROR; 4746} 4747 4748// timed audio tracks 4749 4750sp<AudioFlinger::PlaybackThread::TimedTrack> 4751AudioFlinger::PlaybackThread::TimedTrack::create( 4752 PlaybackThread *thread, 4753 const sp<Client>& client, 4754 audio_stream_type_t streamType, 4755 uint32_t sampleRate, 4756 audio_format_t format, 4757 uint32_t channelMask, 4758 int frameCount, 4759 const sp<IMemory>& sharedBuffer, 4760 int sessionId) { 4761 if (!client->reserveTimedTrack()) 4762 return NULL; 4763 4764 return new TimedTrack( 4765 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4766 sharedBuffer, sessionId); 4767} 4768 4769AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4770 PlaybackThread *thread, 4771 const sp<Client>& client, 4772 audio_stream_type_t streamType, 4773 uint32_t sampleRate, 4774 audio_format_t format, 4775 uint32_t channelMask, 4776 int frameCount, 4777 const sp<IMemory>& sharedBuffer, 4778 int sessionId) 4779 : Track(thread, client, streamType, sampleRate, format, channelMask, 4780 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4781 mQueueHeadInFlight(false), 4782 mTrimQueueHeadOnRelease(false), 4783 mFramesPendingInQueue(0), 4784 mTimedSilenceBuffer(NULL), 4785 mTimedSilenceBufferSize(0), 4786 mTimedAudioOutputOnTime(false), 4787 mMediaTimeTransformValid(false) 4788{ 4789 LocalClock lc; 4790 mLocalTimeFreq = lc.getLocalFreq(); 4791 4792 mLocalTimeToSampleTransform.a_zero = 0; 4793 mLocalTimeToSampleTransform.b_zero = 0; 4794 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4795 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4796 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4797 &mLocalTimeToSampleTransform.a_to_b_denom); 4798 4799 mMediaTimeToSampleTransform.a_zero = 0; 4800 mMediaTimeToSampleTransform.b_zero = 0; 4801 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4802 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4803 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4804 &mMediaTimeToSampleTransform.a_to_b_denom); 4805} 4806 4807AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4808 mClient->releaseTimedTrack(); 4809 delete [] mTimedSilenceBuffer; 4810} 4811 4812status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4813 size_t size, sp<IMemory>* buffer) { 4814 4815 Mutex::Autolock _l(mTimedBufferQueueLock); 4816 4817 trimTimedBufferQueue_l(); 4818 4819 // lazily initialize the shared memory heap for timed buffers 4820 if (mTimedMemoryDealer == NULL) { 4821 const int kTimedBufferHeapSize = 512 << 10; 4822 4823 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4824 "AudioFlingerTimed"); 4825 if (mTimedMemoryDealer == NULL) 4826 return NO_MEMORY; 4827 } 4828 4829 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4830 if (newBuffer == NULL) { 4831 newBuffer = mTimedMemoryDealer->allocate(size); 4832 if (newBuffer == NULL) 4833 return NO_MEMORY; 4834 } 4835 4836 *buffer = newBuffer; 4837 return NO_ERROR; 4838} 4839 4840// caller must hold mTimedBufferQueueLock 4841void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4842 int64_t mediaTimeNow; 4843 { 4844 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4845 if (!mMediaTimeTransformValid) 4846 return; 4847 4848 int64_t targetTimeNow; 4849 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4850 ? mCCHelper.getCommonTime(&targetTimeNow) 4851 : mCCHelper.getLocalTime(&targetTimeNow); 4852 4853 if (OK != res) 4854 return; 4855 4856 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4857 &mediaTimeNow)) { 4858 return; 4859 } 4860 } 4861 4862 size_t trimEnd; 4863 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4864 int64_t bufEnd; 4865 4866 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4867 // We have a next buffer. Just use its PTS as the PTS of the frame 4868 // following the last frame in this buffer. If the stream is sparse 4869 // (ie, there are deliberate gaps left in the stream which should be 4870 // filled with silence by the TimedAudioTrack), then this can result 4871 // in one extra buffer being left un-trimmed when it could have 4872 // been. In general, this is not typical, and we would rather 4873 // optimized away the TS calculation below for the more common case 4874 // where PTSes are contiguous. 4875 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4876 } else { 4877 // We have no next buffer. Compute the PTS of the frame following 4878 // the last frame in this buffer by computing the duration of of 4879 // this frame in media time units and adding it to the PTS of the 4880 // buffer. 4881 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4882 / mCblk->frameSize; 4883 4884 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4885 &bufEnd)) { 4886 ALOGE("Failed to convert frame count of %lld to media time" 4887 " duration" " (scale factor %d/%u) in %s", 4888 frameCount, 4889 mMediaTimeToSampleTransform.a_to_b_numer, 4890 mMediaTimeToSampleTransform.a_to_b_denom, 4891 __PRETTY_FUNCTION__); 4892 break; 4893 } 4894 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4895 } 4896 4897 if (bufEnd > mediaTimeNow) 4898 break; 4899 4900 // Is the buffer we want to use in the middle of a mix operation right 4901 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4902 // from the mixer which should be coming back shortly. 4903 if (!trimEnd && mQueueHeadInFlight) { 4904 mTrimQueueHeadOnRelease = true; 4905 } 4906 } 4907 4908 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4909 if (trimStart < trimEnd) { 4910 // Update the bookkeeping for framesReady() 4911 for (size_t i = trimStart; i < trimEnd; ++i) { 4912 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4913 } 4914 4915 // Now actually remove the buffers from the queue. 4916 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4917 } 4918} 4919 4920void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4921 const char* logTag) { 4922 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4923 "%s called (reason \"%s\"), but timed buffer queue has no" 4924 " elements to trim.", __FUNCTION__, logTag); 4925 4926 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4927 mTimedBufferQueue.removeAt(0); 4928} 4929 4930void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4931 const TimedBuffer& buf, 4932 const char* logTag) { 4933 uint32_t bufBytes = buf.buffer()->size(); 4934 uint32_t consumedAlready = buf.position(); 4935 4936 ALOG_ASSERT(consumedAlready <= bufBytes, 4937 "Bad bookkeeping while updating frames pending. Timed buffer is" 4938 " only %u bytes long, but claims to have consumed %u" 4939 " bytes. (update reason: \"%s\")", 4940 bufBytes, consumedAlready, logTag); 4941 4942 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4943 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4944 "Bad bookkeeping while updating frames pending. Should have at" 4945 " least %u queued frames, but we think we have only %u. (update" 4946 " reason: \"%s\")", 4947 bufFrames, mFramesPendingInQueue, logTag); 4948 4949 mFramesPendingInQueue -= bufFrames; 4950} 4951 4952status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4953 const sp<IMemory>& buffer, int64_t pts) { 4954 4955 { 4956 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4957 if (!mMediaTimeTransformValid) 4958 return INVALID_OPERATION; 4959 } 4960 4961 Mutex::Autolock _l(mTimedBufferQueueLock); 4962 4963 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4964 mFramesPendingInQueue += bufFrames; 4965 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4966 4967 return NO_ERROR; 4968} 4969 4970status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4971 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4972 4973 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4974 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4975 target); 4976 4977 if (!(target == TimedAudioTrack::LOCAL_TIME || 4978 target == TimedAudioTrack::COMMON_TIME)) { 4979 return BAD_VALUE; 4980 } 4981 4982 Mutex::Autolock lock(mMediaTimeTransformLock); 4983 mMediaTimeTransform = xform; 4984 mMediaTimeTransformTarget = target; 4985 mMediaTimeTransformValid = true; 4986 4987 return NO_ERROR; 4988} 4989 4990#define min(a, b) ((a) < (b) ? (a) : (b)) 4991 4992// implementation of getNextBuffer for tracks whose buffers have timestamps 4993status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4994 AudioBufferProvider::Buffer* buffer, int64_t pts) 4995{ 4996 if (pts == AudioBufferProvider::kInvalidPTS) { 4997 buffer->raw = 0; 4998 buffer->frameCount = 0; 4999 mTimedAudioOutputOnTime = false; 5000 return INVALID_OPERATION; 5001 } 5002 5003 Mutex::Autolock _l(mTimedBufferQueueLock); 5004 5005 ALOG_ASSERT(!mQueueHeadInFlight, 5006 "getNextBuffer called without releaseBuffer!"); 5007 5008 while (true) { 5009 5010 // if we have no timed buffers, then fail 5011 if (mTimedBufferQueue.isEmpty()) { 5012 buffer->raw = 0; 5013 buffer->frameCount = 0; 5014 return NOT_ENOUGH_DATA; 5015 } 5016 5017 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5018 5019 // calculate the PTS of the head of the timed buffer queue expressed in 5020 // local time 5021 int64_t headLocalPTS; 5022 { 5023 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5024 5025 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5026 5027 if (mMediaTimeTransform.a_to_b_denom == 0) { 5028 // the transform represents a pause, so yield silence 5029 timedYieldSilence_l(buffer->frameCount, buffer); 5030 return NO_ERROR; 5031 } 5032 5033 int64_t transformedPTS; 5034 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5035 &transformedPTS)) { 5036 // the transform failed. this shouldn't happen, but if it does 5037 // then just drop this buffer 5038 ALOGW("timedGetNextBuffer transform failed"); 5039 buffer->raw = 0; 5040 buffer->frameCount = 0; 5041 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5042 return NO_ERROR; 5043 } 5044 5045 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5046 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5047 &headLocalPTS)) { 5048 buffer->raw = 0; 5049 buffer->frameCount = 0; 5050 return INVALID_OPERATION; 5051 } 5052 } else { 5053 headLocalPTS = transformedPTS; 5054 } 5055 } 5056 5057 // adjust the head buffer's PTS to reflect the portion of the head buffer 5058 // that has already been consumed 5059 int64_t effectivePTS = headLocalPTS + 5060 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5061 5062 // Calculate the delta in samples between the head of the input buffer 5063 // queue and the start of the next output buffer that will be written. 5064 // If the transformation fails because of over or underflow, it means 5065 // that the sample's position in the output stream is so far out of 5066 // whack that it should just be dropped. 5067 int64_t sampleDelta; 5068 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5069 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5070 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5071 " mix"); 5072 continue; 5073 } 5074 if (!mLocalTimeToSampleTransform.doForwardTransform( 5075 (effectivePTS - pts) << 32, &sampleDelta)) { 5076 ALOGV("*** too late during sample rate transform: dropped buffer"); 5077 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5078 continue; 5079 } 5080 5081 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5082 " sampleDelta=[%d.%08x]", 5083 head.pts(), head.position(), pts, 5084 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5085 + (sampleDelta >> 32)), 5086 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5087 5088 // if the delta between the ideal placement for the next input sample and 5089 // the current output position is within this threshold, then we will 5090 // concatenate the next input samples to the previous output 5091 const int64_t kSampleContinuityThreshold = 5092 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5093 5094 // if this is the first buffer of audio that we're emitting from this track 5095 // then it should be almost exactly on time. 5096 const int64_t kSampleStartupThreshold = 1LL << 32; 5097 5098 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5099 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5100 // the next input is close enough to being on time, so concatenate it 5101 // with the last output 5102 timedYieldSamples_l(buffer); 5103 5104 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5105 head.position(), buffer->frameCount); 5106 return NO_ERROR; 5107 } 5108 5109 // Looks like our output is not on time. Reset our on timed status. 5110 // Next time we mix samples from our input queue, then should be within 5111 // the StartupThreshold. 5112 mTimedAudioOutputOnTime = false; 5113 if (sampleDelta > 0) { 5114 // the gap between the current output position and the proper start of 5115 // the next input sample is too big, so fill it with silence 5116 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5117 5118 timedYieldSilence_l(framesUntilNextInput, buffer); 5119 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5120 return NO_ERROR; 5121 } else { 5122 // the next input sample is late 5123 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5124 size_t onTimeSamplePosition = 5125 head.position() + lateFrames * mCblk->frameSize; 5126 5127 if (onTimeSamplePosition > head.buffer()->size()) { 5128 // all the remaining samples in the head are too late, so 5129 // drop it and move on 5130 ALOGV("*** too late: dropped buffer"); 5131 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5132 continue; 5133 } else { 5134 // skip over the late samples 5135 head.setPosition(onTimeSamplePosition); 5136 5137 // yield the available samples 5138 timedYieldSamples_l(buffer); 5139 5140 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5141 return NO_ERROR; 5142 } 5143 } 5144 } 5145} 5146 5147// Yield samples from the timed buffer queue head up to the given output 5148// buffer's capacity. 5149// 5150// Caller must hold mTimedBufferQueueLock 5151void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5152 AudioBufferProvider::Buffer* buffer) { 5153 5154 const TimedBuffer& head = mTimedBufferQueue[0]; 5155 5156 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5157 head.position()); 5158 5159 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5160 mCblk->frameSize); 5161 size_t framesRequested = buffer->frameCount; 5162 buffer->frameCount = min(framesLeftInHead, framesRequested); 5163 5164 mQueueHeadInFlight = true; 5165 mTimedAudioOutputOnTime = true; 5166} 5167 5168// Yield samples of silence up to the given output buffer's capacity 5169// 5170// Caller must hold mTimedBufferQueueLock 5171void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5172 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5173 5174 // lazily allocate a buffer filled with silence 5175 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5176 delete [] mTimedSilenceBuffer; 5177 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5178 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5179 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5180 } 5181 5182 buffer->raw = mTimedSilenceBuffer; 5183 size_t framesRequested = buffer->frameCount; 5184 buffer->frameCount = min(numFrames, framesRequested); 5185 5186 mTimedAudioOutputOnTime = false; 5187} 5188 5189// AudioBufferProvider interface 5190void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5191 AudioBufferProvider::Buffer* buffer) { 5192 5193 Mutex::Autolock _l(mTimedBufferQueueLock); 5194 5195 // If the buffer which was just released is part of the buffer at the head 5196 // of the queue, be sure to update the amt of the buffer which has been 5197 // consumed. If the buffer being returned is not part of the head of the 5198 // queue, its either because the buffer is part of the silence buffer, or 5199 // because the head of the timed queue was trimmed after the mixer called 5200 // getNextBuffer but before the mixer called releaseBuffer. 5201 if (buffer->raw == mTimedSilenceBuffer) { 5202 ALOG_ASSERT(!mQueueHeadInFlight, 5203 "Queue head in flight during release of silence buffer!"); 5204 goto done; 5205 } 5206 5207 ALOG_ASSERT(mQueueHeadInFlight, 5208 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5209 " head in flight."); 5210 5211 if (mTimedBufferQueue.size()) { 5212 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5213 5214 void* start = head.buffer()->pointer(); 5215 void* end = reinterpret_cast<void*>( 5216 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5217 + head.buffer()->size()); 5218 5219 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5220 "released buffer not within the head of the timed buffer" 5221 " queue; qHead = [%p, %p], released buffer = %p", 5222 start, end, buffer->raw); 5223 5224 head.setPosition(head.position() + 5225 (buffer->frameCount * mCblk->frameSize)); 5226 mQueueHeadInFlight = false; 5227 5228 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5229 "Bad bookkeeping during releaseBuffer! Should have at" 5230 " least %u queued frames, but we think we have only %u", 5231 buffer->frameCount, mFramesPendingInQueue); 5232 5233 mFramesPendingInQueue -= buffer->frameCount; 5234 5235 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5236 || mTrimQueueHeadOnRelease) { 5237 trimTimedBufferQueueHead_l("releaseBuffer"); 5238 mTrimQueueHeadOnRelease = false; 5239 } 5240 } else { 5241 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5242 " buffers in the timed buffer queue"); 5243 } 5244 5245done: 5246 buffer->raw = 0; 5247 buffer->frameCount = 0; 5248} 5249 5250size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5251 Mutex::Autolock _l(mTimedBufferQueueLock); 5252 return mFramesPendingInQueue; 5253} 5254 5255AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5256 : mPTS(0), mPosition(0) {} 5257 5258AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5259 const sp<IMemory>& buffer, int64_t pts) 5260 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5261 5262// ---------------------------------------------------------------------------- 5263 5264// RecordTrack constructor must be called with AudioFlinger::mLock held 5265AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5266 RecordThread *thread, 5267 const sp<Client>& client, 5268 uint32_t sampleRate, 5269 audio_format_t format, 5270 uint32_t channelMask, 5271 int frameCount, 5272 int sessionId) 5273 : TrackBase(thread, client, sampleRate, format, 5274 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5275 mOverflow(false) 5276{ 5277 if (mCblk != NULL) { 5278 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5279 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5280 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5281 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5282 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5283 } else { 5284 mCblk->frameSize = sizeof(int8_t); 5285 } 5286 } 5287} 5288 5289AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5290{ 5291 sp<ThreadBase> thread = mThread.promote(); 5292 if (thread != 0) { 5293 AudioSystem::releaseInput(thread->id()); 5294 } 5295} 5296 5297// AudioBufferProvider interface 5298status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5299{ 5300 audio_track_cblk_t* cblk = this->cblk(); 5301 uint32_t framesAvail; 5302 uint32_t framesReq = buffer->frameCount; 5303 5304 // Check if last stepServer failed, try to step now 5305 if (mStepServerFailed) { 5306 if (!step()) goto getNextBuffer_exit; 5307 ALOGV("stepServer recovered"); 5308 mStepServerFailed = false; 5309 } 5310 5311 framesAvail = cblk->framesAvailable_l(); 5312 5313 if (CC_LIKELY(framesAvail)) { 5314 uint32_t s = cblk->server; 5315 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5316 5317 if (framesReq > framesAvail) { 5318 framesReq = framesAvail; 5319 } 5320 if (framesReq > bufferEnd - s) { 5321 framesReq = bufferEnd - s; 5322 } 5323 5324 buffer->raw = getBuffer(s, framesReq); 5325 if (buffer->raw == NULL) goto getNextBuffer_exit; 5326 5327 buffer->frameCount = framesReq; 5328 return NO_ERROR; 5329 } 5330 5331getNextBuffer_exit: 5332 buffer->raw = NULL; 5333 buffer->frameCount = 0; 5334 return NOT_ENOUGH_DATA; 5335} 5336 5337status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5338 int triggerSession) 5339{ 5340 sp<ThreadBase> thread = mThread.promote(); 5341 if (thread != 0) { 5342 RecordThread *recordThread = (RecordThread *)thread.get(); 5343 return recordThread->start(this, event, triggerSession); 5344 } else { 5345 return BAD_VALUE; 5346 } 5347} 5348 5349void AudioFlinger::RecordThread::RecordTrack::stop() 5350{ 5351 sp<ThreadBase> thread = mThread.promote(); 5352 if (thread != 0) { 5353 RecordThread *recordThread = (RecordThread *)thread.get(); 5354 recordThread->stop(this); 5355 TrackBase::reset(); 5356 // Force overrun condition to avoid false overrun callback until first data is 5357 // read from buffer 5358 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5359 } 5360} 5361 5362void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5363{ 5364 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5365 (mClient == 0) ? getpid_cached : mClient->pid(), 5366 mFormat, 5367 mChannelMask, 5368 mSessionId, 5369 mFrameCount, 5370 mState, 5371 mCblk->sampleRate, 5372 mCblk->server, 5373 mCblk->user); 5374} 5375 5376 5377// ---------------------------------------------------------------------------- 5378 5379AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5380 PlaybackThread *playbackThread, 5381 DuplicatingThread *sourceThread, 5382 uint32_t sampleRate, 5383 audio_format_t format, 5384 uint32_t channelMask, 5385 int frameCount) 5386 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5387 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5388 mActive(false), mSourceThread(sourceThread) 5389{ 5390 5391 if (mCblk != NULL) { 5392 mCblk->flags |= CBLK_DIRECTION_OUT; 5393 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5394 mOutBuffer.frameCount = 0; 5395 playbackThread->mTracks.add(this); 5396 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5397 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5398 mCblk, mBuffer, mCblk->buffers, 5399 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5400 } else { 5401 ALOGW("Error creating output track on thread %p", playbackThread); 5402 } 5403} 5404 5405AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5406{ 5407 clearBufferQueue(); 5408} 5409 5410status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5411 int triggerSession) 5412{ 5413 status_t status = Track::start(event, triggerSession); 5414 if (status != NO_ERROR) { 5415 return status; 5416 } 5417 5418 mActive = true; 5419 mRetryCount = 127; 5420 return status; 5421} 5422 5423void AudioFlinger::PlaybackThread::OutputTrack::stop() 5424{ 5425 Track::stop(); 5426 clearBufferQueue(); 5427 mOutBuffer.frameCount = 0; 5428 mActive = false; 5429} 5430 5431bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5432{ 5433 Buffer *pInBuffer; 5434 Buffer inBuffer; 5435 uint32_t channelCount = mChannelCount; 5436 bool outputBufferFull = false; 5437 inBuffer.frameCount = frames; 5438 inBuffer.i16 = data; 5439 5440 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5441 5442 if (!mActive && frames != 0) { 5443 start(); 5444 sp<ThreadBase> thread = mThread.promote(); 5445 if (thread != 0) { 5446 MixerThread *mixerThread = (MixerThread *)thread.get(); 5447 if (mCblk->frameCount > frames){ 5448 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5449 uint32_t startFrames = (mCblk->frameCount - frames); 5450 pInBuffer = new Buffer; 5451 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5452 pInBuffer->frameCount = startFrames; 5453 pInBuffer->i16 = pInBuffer->mBuffer; 5454 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5455 mBufferQueue.add(pInBuffer); 5456 } else { 5457 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5458 } 5459 } 5460 } 5461 } 5462 5463 while (waitTimeLeftMs) { 5464 // First write pending buffers, then new data 5465 if (mBufferQueue.size()) { 5466 pInBuffer = mBufferQueue.itemAt(0); 5467 } else { 5468 pInBuffer = &inBuffer; 5469 } 5470 5471 if (pInBuffer->frameCount == 0) { 5472 break; 5473 } 5474 5475 if (mOutBuffer.frameCount == 0) { 5476 mOutBuffer.frameCount = pInBuffer->frameCount; 5477 nsecs_t startTime = systemTime(); 5478 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5479 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5480 outputBufferFull = true; 5481 break; 5482 } 5483 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5484 if (waitTimeLeftMs >= waitTimeMs) { 5485 waitTimeLeftMs -= waitTimeMs; 5486 } else { 5487 waitTimeLeftMs = 0; 5488 } 5489 } 5490 5491 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5492 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5493 mCblk->stepUser(outFrames); 5494 pInBuffer->frameCount -= outFrames; 5495 pInBuffer->i16 += outFrames * channelCount; 5496 mOutBuffer.frameCount -= outFrames; 5497 mOutBuffer.i16 += outFrames * channelCount; 5498 5499 if (pInBuffer->frameCount == 0) { 5500 if (mBufferQueue.size()) { 5501 mBufferQueue.removeAt(0); 5502 delete [] pInBuffer->mBuffer; 5503 delete pInBuffer; 5504 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5505 } else { 5506 break; 5507 } 5508 } 5509 } 5510 5511 // If we could not write all frames, allocate a buffer and queue it for next time. 5512 if (inBuffer.frameCount) { 5513 sp<ThreadBase> thread = mThread.promote(); 5514 if (thread != 0 && !thread->standby()) { 5515 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5516 pInBuffer = new Buffer; 5517 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5518 pInBuffer->frameCount = inBuffer.frameCount; 5519 pInBuffer->i16 = pInBuffer->mBuffer; 5520 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5521 mBufferQueue.add(pInBuffer); 5522 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5523 } else { 5524 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5525 } 5526 } 5527 } 5528 5529 // Calling write() with a 0 length buffer, means that no more data will be written: 5530 // If no more buffers are pending, fill output track buffer to make sure it is started 5531 // by output mixer. 5532 if (frames == 0 && mBufferQueue.size() == 0) { 5533 if (mCblk->user < mCblk->frameCount) { 5534 frames = mCblk->frameCount - mCblk->user; 5535 pInBuffer = new Buffer; 5536 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5537 pInBuffer->frameCount = frames; 5538 pInBuffer->i16 = pInBuffer->mBuffer; 5539 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5540 mBufferQueue.add(pInBuffer); 5541 } else if (mActive) { 5542 stop(); 5543 } 5544 } 5545 5546 return outputBufferFull; 5547} 5548 5549status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5550{ 5551 int active; 5552 status_t result; 5553 audio_track_cblk_t* cblk = mCblk; 5554 uint32_t framesReq = buffer->frameCount; 5555 5556// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5557 buffer->frameCount = 0; 5558 5559 uint32_t framesAvail = cblk->framesAvailable(); 5560 5561 5562 if (framesAvail == 0) { 5563 Mutex::Autolock _l(cblk->lock); 5564 goto start_loop_here; 5565 while (framesAvail == 0) { 5566 active = mActive; 5567 if (CC_UNLIKELY(!active)) { 5568 ALOGV("Not active and NO_MORE_BUFFERS"); 5569 return NO_MORE_BUFFERS; 5570 } 5571 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5572 if (result != NO_ERROR) { 5573 return NO_MORE_BUFFERS; 5574 } 5575 // read the server count again 5576 start_loop_here: 5577 framesAvail = cblk->framesAvailable_l(); 5578 } 5579 } 5580 5581// if (framesAvail < framesReq) { 5582// return NO_MORE_BUFFERS; 5583// } 5584 5585 if (framesReq > framesAvail) { 5586 framesReq = framesAvail; 5587 } 5588 5589 uint32_t u = cblk->user; 5590 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5591 5592 if (framesReq > bufferEnd - u) { 5593 framesReq = bufferEnd - u; 5594 } 5595 5596 buffer->frameCount = framesReq; 5597 buffer->raw = (void *)cblk->buffer(u); 5598 return NO_ERROR; 5599} 5600 5601 5602void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5603{ 5604 size_t size = mBufferQueue.size(); 5605 5606 for (size_t i = 0; i < size; i++) { 5607 Buffer *pBuffer = mBufferQueue.itemAt(i); 5608 delete [] pBuffer->mBuffer; 5609 delete pBuffer; 5610 } 5611 mBufferQueue.clear(); 5612} 5613 5614// ---------------------------------------------------------------------------- 5615 5616AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5617 : RefBase(), 5618 mAudioFlinger(audioFlinger), 5619 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5620 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5621 mPid(pid), 5622 mTimedTrackCount(0) 5623{ 5624 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5625} 5626 5627// Client destructor must be called with AudioFlinger::mLock held 5628AudioFlinger::Client::~Client() 5629{ 5630 mAudioFlinger->removeClient_l(mPid); 5631} 5632 5633sp<MemoryDealer> AudioFlinger::Client::heap() const 5634{ 5635 return mMemoryDealer; 5636} 5637 5638// Reserve one of the limited slots for a timed audio track associated 5639// with this client 5640bool AudioFlinger::Client::reserveTimedTrack() 5641{ 5642 const int kMaxTimedTracksPerClient = 4; 5643 5644 Mutex::Autolock _l(mTimedTrackLock); 5645 5646 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5647 ALOGW("can not create timed track - pid %d has exceeded the limit", 5648 mPid); 5649 return false; 5650 } 5651 5652 mTimedTrackCount++; 5653 return true; 5654} 5655 5656// Release a slot for a timed audio track 5657void AudioFlinger::Client::releaseTimedTrack() 5658{ 5659 Mutex::Autolock _l(mTimedTrackLock); 5660 mTimedTrackCount--; 5661} 5662 5663// ---------------------------------------------------------------------------- 5664 5665AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5666 const sp<IAudioFlingerClient>& client, 5667 pid_t pid) 5668 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5669{ 5670} 5671 5672AudioFlinger::NotificationClient::~NotificationClient() 5673{ 5674} 5675 5676void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5677{ 5678 sp<NotificationClient> keep(this); 5679 mAudioFlinger->removeNotificationClient(mPid); 5680} 5681 5682// ---------------------------------------------------------------------------- 5683 5684AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5685 : BnAudioTrack(), 5686 mTrack(track) 5687{ 5688} 5689 5690AudioFlinger::TrackHandle::~TrackHandle() { 5691 // just stop the track on deletion, associated resources 5692 // will be freed from the main thread once all pending buffers have 5693 // been played. Unless it's not in the active track list, in which 5694 // case we free everything now... 5695 mTrack->destroy(); 5696} 5697 5698sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5699 return mTrack->getCblk(); 5700} 5701 5702status_t AudioFlinger::TrackHandle::start() { 5703 return mTrack->start(); 5704} 5705 5706void AudioFlinger::TrackHandle::stop() { 5707 mTrack->stop(); 5708} 5709 5710void AudioFlinger::TrackHandle::flush() { 5711 mTrack->flush(); 5712} 5713 5714void AudioFlinger::TrackHandle::mute(bool e) { 5715 mTrack->mute(e); 5716} 5717 5718void AudioFlinger::TrackHandle::pause() { 5719 mTrack->pause(); 5720} 5721 5722status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5723{ 5724 return mTrack->attachAuxEffect(EffectId); 5725} 5726 5727status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5728 sp<IMemory>* buffer) { 5729 if (!mTrack->isTimedTrack()) 5730 return INVALID_OPERATION; 5731 5732 PlaybackThread::TimedTrack* tt = 5733 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5734 return tt->allocateTimedBuffer(size, buffer); 5735} 5736 5737status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5738 int64_t pts) { 5739 if (!mTrack->isTimedTrack()) 5740 return INVALID_OPERATION; 5741 5742 PlaybackThread::TimedTrack* tt = 5743 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5744 return tt->queueTimedBuffer(buffer, pts); 5745} 5746 5747status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5748 const LinearTransform& xform, int target) { 5749 5750 if (!mTrack->isTimedTrack()) 5751 return INVALID_OPERATION; 5752 5753 PlaybackThread::TimedTrack* tt = 5754 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5755 return tt->setMediaTimeTransform( 5756 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5757} 5758 5759status_t AudioFlinger::TrackHandle::onTransact( 5760 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5761{ 5762 return BnAudioTrack::onTransact(code, data, reply, flags); 5763} 5764 5765// ---------------------------------------------------------------------------- 5766 5767sp<IAudioRecord> AudioFlinger::openRecord( 5768 pid_t pid, 5769 audio_io_handle_t input, 5770 uint32_t sampleRate, 5771 audio_format_t format, 5772 uint32_t channelMask, 5773 int frameCount, 5774 IAudioFlinger::track_flags_t flags, 5775 int *sessionId, 5776 status_t *status) 5777{ 5778 sp<RecordThread::RecordTrack> recordTrack; 5779 sp<RecordHandle> recordHandle; 5780 sp<Client> client; 5781 status_t lStatus; 5782 RecordThread *thread; 5783 size_t inFrameCount; 5784 int lSessionId; 5785 5786 // check calling permissions 5787 if (!recordingAllowed()) { 5788 lStatus = PERMISSION_DENIED; 5789 goto Exit; 5790 } 5791 5792 // add client to list 5793 { // scope for mLock 5794 Mutex::Autolock _l(mLock); 5795 thread = checkRecordThread_l(input); 5796 if (thread == NULL) { 5797 lStatus = BAD_VALUE; 5798 goto Exit; 5799 } 5800 5801 client = registerPid_l(pid); 5802 5803 // If no audio session id is provided, create one here 5804 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5805 lSessionId = *sessionId; 5806 } else { 5807 lSessionId = nextUniqueId(); 5808 if (sessionId != NULL) { 5809 *sessionId = lSessionId; 5810 } 5811 } 5812 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5813 recordTrack = thread->createRecordTrack_l(client, 5814 sampleRate, 5815 format, 5816 channelMask, 5817 frameCount, 5818 lSessionId, 5819 &lStatus); 5820 } 5821 if (lStatus != NO_ERROR) { 5822 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5823 // destructor is called by the TrackBase destructor with mLock held 5824 client.clear(); 5825 recordTrack.clear(); 5826 goto Exit; 5827 } 5828 5829 // return to handle to client 5830 recordHandle = new RecordHandle(recordTrack); 5831 lStatus = NO_ERROR; 5832 5833Exit: 5834 if (status) { 5835 *status = lStatus; 5836 } 5837 return recordHandle; 5838} 5839 5840// ---------------------------------------------------------------------------- 5841 5842AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5843 : BnAudioRecord(), 5844 mRecordTrack(recordTrack) 5845{ 5846} 5847 5848AudioFlinger::RecordHandle::~RecordHandle() { 5849 stop(); 5850} 5851 5852sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5853 return mRecordTrack->getCblk(); 5854} 5855 5856status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5857 ALOGV("RecordHandle::start()"); 5858 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5859} 5860 5861void AudioFlinger::RecordHandle::stop() { 5862 ALOGV("RecordHandle::stop()"); 5863 mRecordTrack->stop(); 5864} 5865 5866status_t AudioFlinger::RecordHandle::onTransact( 5867 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5868{ 5869 return BnAudioRecord::onTransact(code, data, reply, flags); 5870} 5871 5872// ---------------------------------------------------------------------------- 5873 5874AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5875 AudioStreamIn *input, 5876 uint32_t sampleRate, 5877 uint32_t channels, 5878 audio_io_handle_t id, 5879 uint32_t device) : 5880 ThreadBase(audioFlinger, id, device, RECORD), 5881 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5882 // mRsmpInIndex and mInputBytes set by readInputParameters() 5883 mReqChannelCount(popcount(channels)), 5884 mReqSampleRate(sampleRate) 5885 // mBytesRead is only meaningful while active, and so is cleared in start() 5886 // (but might be better to also clear here for dump?) 5887{ 5888 snprintf(mName, kNameLength, "AudioIn_%X", id); 5889 5890 readInputParameters(); 5891} 5892 5893 5894AudioFlinger::RecordThread::~RecordThread() 5895{ 5896 delete[] mRsmpInBuffer; 5897 delete mResampler; 5898 delete[] mRsmpOutBuffer; 5899} 5900 5901void AudioFlinger::RecordThread::onFirstRef() 5902{ 5903 run(mName, PRIORITY_URGENT_AUDIO); 5904} 5905 5906status_t AudioFlinger::RecordThread::readyToRun() 5907{ 5908 status_t status = initCheck(); 5909 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5910 return status; 5911} 5912 5913bool AudioFlinger::RecordThread::threadLoop() 5914{ 5915 AudioBufferProvider::Buffer buffer; 5916 sp<RecordTrack> activeTrack; 5917 Vector< sp<EffectChain> > effectChains; 5918 5919 nsecs_t lastWarning = 0; 5920 5921 acquireWakeLock(); 5922 5923 // start recording 5924 while (!exitPending()) { 5925 5926 processConfigEvents(); 5927 5928 { // scope for mLock 5929 Mutex::Autolock _l(mLock); 5930 checkForNewParameters_l(); 5931 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5932 if (!mStandby) { 5933 mInput->stream->common.standby(&mInput->stream->common); 5934 mStandby = true; 5935 } 5936 5937 if (exitPending()) break; 5938 5939 releaseWakeLock_l(); 5940 ALOGV("RecordThread: loop stopping"); 5941 // go to sleep 5942 mWaitWorkCV.wait(mLock); 5943 ALOGV("RecordThread: loop starting"); 5944 acquireWakeLock_l(); 5945 continue; 5946 } 5947 if (mActiveTrack != 0) { 5948 if (mActiveTrack->mState == TrackBase::PAUSING) { 5949 if (!mStandby) { 5950 mInput->stream->common.standby(&mInput->stream->common); 5951 mStandby = true; 5952 } 5953 mActiveTrack.clear(); 5954 mStartStopCond.broadcast(); 5955 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5956 if (mReqChannelCount != mActiveTrack->channelCount()) { 5957 mActiveTrack.clear(); 5958 mStartStopCond.broadcast(); 5959 } else if (mBytesRead != 0) { 5960 // record start succeeds only if first read from audio input 5961 // succeeds 5962 if (mBytesRead > 0) { 5963 mActiveTrack->mState = TrackBase::ACTIVE; 5964 } else { 5965 mActiveTrack.clear(); 5966 } 5967 mStartStopCond.broadcast(); 5968 } 5969 mStandby = false; 5970 } 5971 } 5972 lockEffectChains_l(effectChains); 5973 } 5974 5975 if (mActiveTrack != 0) { 5976 if (mActiveTrack->mState != TrackBase::ACTIVE && 5977 mActiveTrack->mState != TrackBase::RESUMING) { 5978 unlockEffectChains(effectChains); 5979 usleep(kRecordThreadSleepUs); 5980 continue; 5981 } 5982 for (size_t i = 0; i < effectChains.size(); i ++) { 5983 effectChains[i]->process_l(); 5984 } 5985 5986 buffer.frameCount = mFrameCount; 5987 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5988 size_t framesOut = buffer.frameCount; 5989 if (mResampler == NULL) { 5990 // no resampling 5991 while (framesOut) { 5992 size_t framesIn = mFrameCount - mRsmpInIndex; 5993 if (framesIn) { 5994 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5995 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5996 if (framesIn > framesOut) 5997 framesIn = framesOut; 5998 mRsmpInIndex += framesIn; 5999 framesOut -= framesIn; 6000 if ((int)mChannelCount == mReqChannelCount || 6001 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6002 memcpy(dst, src, framesIn * mFrameSize); 6003 } else { 6004 int16_t *src16 = (int16_t *)src; 6005 int16_t *dst16 = (int16_t *)dst; 6006 if (mChannelCount == 1) { 6007 while (framesIn--) { 6008 *dst16++ = *src16; 6009 *dst16++ = *src16++; 6010 } 6011 } else { 6012 while (framesIn--) { 6013 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6014 src16 += 2; 6015 } 6016 } 6017 } 6018 } 6019 if (framesOut && mFrameCount == mRsmpInIndex) { 6020 if (framesOut == mFrameCount && 6021 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6022 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6023 framesOut = 0; 6024 } else { 6025 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6026 mRsmpInIndex = 0; 6027 } 6028 if (mBytesRead < 0) { 6029 ALOGE("Error reading audio input"); 6030 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6031 // Force input into standby so that it tries to 6032 // recover at next read attempt 6033 mInput->stream->common.standby(&mInput->stream->common); 6034 usleep(kRecordThreadSleepUs); 6035 } 6036 mRsmpInIndex = mFrameCount; 6037 framesOut = 0; 6038 buffer.frameCount = 0; 6039 } 6040 } 6041 } 6042 } else { 6043 // resampling 6044 6045 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6046 // alter output frame count as if we were expecting stereo samples 6047 if (mChannelCount == 1 && mReqChannelCount == 1) { 6048 framesOut >>= 1; 6049 } 6050 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6051 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6052 // are 32 bit aligned which should be always true. 6053 if (mChannelCount == 2 && mReqChannelCount == 1) { 6054 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6055 // the resampler always outputs stereo samples: do post stereo to mono conversion 6056 int16_t *src = (int16_t *)mRsmpOutBuffer; 6057 int16_t *dst = buffer.i16; 6058 while (framesOut--) { 6059 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6060 src += 2; 6061 } 6062 } else { 6063 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6064 } 6065 6066 } 6067 if (mFramestoDrop == 0) { 6068 mActiveTrack->releaseBuffer(&buffer); 6069 } else { 6070 if (mFramestoDrop > 0) { 6071 mFramestoDrop -= buffer.frameCount; 6072 if (mFramestoDrop <= 0) { 6073 clearSyncStartEvent(); 6074 } 6075 } else { 6076 mFramestoDrop += buffer.frameCount; 6077 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6078 mSyncStartEvent->isCancelled()) { 6079 ALOGW("Synced record %s, session %d, trigger session %d", 6080 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6081 mActiveTrack->sessionId(), 6082 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6083 clearSyncStartEvent(); 6084 } 6085 } 6086 } 6087 mActiveTrack->overflow(); 6088 } 6089 // client isn't retrieving buffers fast enough 6090 else { 6091 if (!mActiveTrack->setOverflow()) { 6092 nsecs_t now = systemTime(); 6093 if ((now - lastWarning) > kWarningThrottleNs) { 6094 ALOGW("RecordThread: buffer overflow"); 6095 lastWarning = now; 6096 } 6097 } 6098 // Release the processor for a while before asking for a new buffer. 6099 // This will give the application more chance to read from the buffer and 6100 // clear the overflow. 6101 usleep(kRecordThreadSleepUs); 6102 } 6103 } 6104 // enable changes in effect chain 6105 unlockEffectChains(effectChains); 6106 effectChains.clear(); 6107 } 6108 6109 if (!mStandby) { 6110 mInput->stream->common.standby(&mInput->stream->common); 6111 } 6112 mActiveTrack.clear(); 6113 6114 mStartStopCond.broadcast(); 6115 6116 releaseWakeLock(); 6117 6118 ALOGV("RecordThread %p exiting", this); 6119 return false; 6120} 6121 6122 6123sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6124 const sp<AudioFlinger::Client>& client, 6125 uint32_t sampleRate, 6126 audio_format_t format, 6127 int channelMask, 6128 int frameCount, 6129 int sessionId, 6130 status_t *status) 6131{ 6132 sp<RecordTrack> track; 6133 status_t lStatus; 6134 6135 lStatus = initCheck(); 6136 if (lStatus != NO_ERROR) { 6137 ALOGE("Audio driver not initialized."); 6138 goto Exit; 6139 } 6140 6141 { // scope for mLock 6142 Mutex::Autolock _l(mLock); 6143 6144 track = new RecordTrack(this, client, sampleRate, 6145 format, channelMask, frameCount, sessionId); 6146 6147 if (track->getCblk() == 0) { 6148 lStatus = NO_MEMORY; 6149 goto Exit; 6150 } 6151 6152 mTrack = track.get(); 6153 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6154 bool suspend = audio_is_bluetooth_sco_device( 6155 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6156 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6157 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6158 } 6159 lStatus = NO_ERROR; 6160 6161Exit: 6162 if (status) { 6163 *status = lStatus; 6164 } 6165 return track; 6166} 6167 6168status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6169 AudioSystem::sync_event_t event, 6170 int triggerSession) 6171{ 6172 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6173 sp<ThreadBase> strongMe = this; 6174 status_t status = NO_ERROR; 6175 6176 if (event == AudioSystem::SYNC_EVENT_NONE) { 6177 clearSyncStartEvent(); 6178 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6179 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6180 triggerSession, 6181 recordTrack->sessionId(), 6182 syncStartEventCallback, 6183 this); 6184 // Sync event can be cancelled by the trigger session if the track is not in a 6185 // compatible state in which case we start record immediately 6186 if (mSyncStartEvent->isCancelled()) { 6187 clearSyncStartEvent(); 6188 } else { 6189 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6190 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6191 } 6192 } 6193 6194 { 6195 AutoMutex lock(mLock); 6196 if (mActiveTrack != 0) { 6197 if (recordTrack != mActiveTrack.get()) { 6198 status = -EBUSY; 6199 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6200 mActiveTrack->mState = TrackBase::ACTIVE; 6201 } 6202 return status; 6203 } 6204 6205 recordTrack->mState = TrackBase::IDLE; 6206 mActiveTrack = recordTrack; 6207 mLock.unlock(); 6208 status_t status = AudioSystem::startInput(mId); 6209 mLock.lock(); 6210 if (status != NO_ERROR) { 6211 mActiveTrack.clear(); 6212 clearSyncStartEvent(); 6213 return status; 6214 } 6215 mRsmpInIndex = mFrameCount; 6216 mBytesRead = 0; 6217 if (mResampler != NULL) { 6218 mResampler->reset(); 6219 } 6220 mActiveTrack->mState = TrackBase::RESUMING; 6221 // signal thread to start 6222 ALOGV("Signal record thread"); 6223 mWaitWorkCV.signal(); 6224 // do not wait for mStartStopCond if exiting 6225 if (exitPending()) { 6226 mActiveTrack.clear(); 6227 status = INVALID_OPERATION; 6228 goto startError; 6229 } 6230 mStartStopCond.wait(mLock); 6231 if (mActiveTrack == 0) { 6232 ALOGV("Record failed to start"); 6233 status = BAD_VALUE; 6234 goto startError; 6235 } 6236 ALOGV("Record started OK"); 6237 return status; 6238 } 6239startError: 6240 AudioSystem::stopInput(mId); 6241 clearSyncStartEvent(); 6242 return status; 6243} 6244 6245void AudioFlinger::RecordThread::clearSyncStartEvent() 6246{ 6247 if (mSyncStartEvent != 0) { 6248 mSyncStartEvent->cancel(); 6249 } 6250 mSyncStartEvent.clear(); 6251 mFramestoDrop = 0; 6252} 6253 6254void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6255{ 6256 sp<SyncEvent> strongEvent = event.promote(); 6257 6258 if (strongEvent != 0) { 6259 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6260 me->handleSyncStartEvent(strongEvent); 6261 } 6262} 6263 6264void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6265{ 6266 if (event == mSyncStartEvent) { 6267 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6268 // from audio HAL 6269 mFramestoDrop = mFrameCount * 2; 6270 } 6271} 6272 6273void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6274 ALOGV("RecordThread::stop"); 6275 sp<ThreadBase> strongMe = this; 6276 { 6277 AutoMutex lock(mLock); 6278 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6279 mActiveTrack->mState = TrackBase::PAUSING; 6280 // do not wait for mStartStopCond if exiting 6281 if (exitPending()) { 6282 return; 6283 } 6284 mStartStopCond.wait(mLock); 6285 // if we have been restarted, recordTrack == mActiveTrack.get() here 6286 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6287 mLock.unlock(); 6288 AudioSystem::stopInput(mId); 6289 mLock.lock(); 6290 ALOGV("Record stopped OK"); 6291 } 6292 } 6293 } 6294} 6295 6296bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6297{ 6298 return false; 6299} 6300 6301status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6302{ 6303 if (!isValidSyncEvent(event)) { 6304 return BAD_VALUE; 6305 } 6306 6307 Mutex::Autolock _l(mLock); 6308 6309 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6310 mTrack->setSyncEvent(event); 6311 return NO_ERROR; 6312 } 6313 return NAME_NOT_FOUND; 6314} 6315 6316status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6317{ 6318 const size_t SIZE = 256; 6319 char buffer[SIZE]; 6320 String8 result; 6321 6322 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6323 result.append(buffer); 6324 6325 if (mActiveTrack != 0) { 6326 result.append("Active Track:\n"); 6327 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6328 mActiveTrack->dump(buffer, SIZE); 6329 result.append(buffer); 6330 6331 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6332 result.append(buffer); 6333 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6334 result.append(buffer); 6335 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6336 result.append(buffer); 6337 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6338 result.append(buffer); 6339 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6340 result.append(buffer); 6341 6342 6343 } else { 6344 result.append("No record client\n"); 6345 } 6346 write(fd, result.string(), result.size()); 6347 6348 dumpBase(fd, args); 6349 dumpEffectChains(fd, args); 6350 6351 return NO_ERROR; 6352} 6353 6354// AudioBufferProvider interface 6355status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6356{ 6357 size_t framesReq = buffer->frameCount; 6358 size_t framesReady = mFrameCount - mRsmpInIndex; 6359 int channelCount; 6360 6361 if (framesReady == 0) { 6362 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6363 if (mBytesRead < 0) { 6364 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6365 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6366 // Force input into standby so that it tries to 6367 // recover at next read attempt 6368 mInput->stream->common.standby(&mInput->stream->common); 6369 usleep(kRecordThreadSleepUs); 6370 } 6371 buffer->raw = NULL; 6372 buffer->frameCount = 0; 6373 return NOT_ENOUGH_DATA; 6374 } 6375 mRsmpInIndex = 0; 6376 framesReady = mFrameCount; 6377 } 6378 6379 if (framesReq > framesReady) { 6380 framesReq = framesReady; 6381 } 6382 6383 if (mChannelCount == 1 && mReqChannelCount == 2) { 6384 channelCount = 1; 6385 } else { 6386 channelCount = 2; 6387 } 6388 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6389 buffer->frameCount = framesReq; 6390 return NO_ERROR; 6391} 6392 6393// AudioBufferProvider interface 6394void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6395{ 6396 mRsmpInIndex += buffer->frameCount; 6397 buffer->frameCount = 0; 6398} 6399 6400bool AudioFlinger::RecordThread::checkForNewParameters_l() 6401{ 6402 bool reconfig = false; 6403 6404 while (!mNewParameters.isEmpty()) { 6405 status_t status = NO_ERROR; 6406 String8 keyValuePair = mNewParameters[0]; 6407 AudioParameter param = AudioParameter(keyValuePair); 6408 int value; 6409 audio_format_t reqFormat = mFormat; 6410 int reqSamplingRate = mReqSampleRate; 6411 int reqChannelCount = mReqChannelCount; 6412 6413 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6414 reqSamplingRate = value; 6415 reconfig = true; 6416 } 6417 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6418 reqFormat = (audio_format_t) value; 6419 reconfig = true; 6420 } 6421 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6422 reqChannelCount = popcount(value); 6423 reconfig = true; 6424 } 6425 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6426 // do not accept frame count changes if tracks are open as the track buffer 6427 // size depends on frame count and correct behavior would not be guaranteed 6428 // if frame count is changed after track creation 6429 if (mActiveTrack != 0) { 6430 status = INVALID_OPERATION; 6431 } else { 6432 reconfig = true; 6433 } 6434 } 6435 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6436 // forward device change to effects that have requested to be 6437 // aware of attached audio device. 6438 for (size_t i = 0; i < mEffectChains.size(); i++) { 6439 mEffectChains[i]->setDevice_l(value); 6440 } 6441 // store input device and output device but do not forward output device to audio HAL. 6442 // Note that status is ignored by the caller for output device 6443 // (see AudioFlinger::setParameters() 6444 if (value & AUDIO_DEVICE_OUT_ALL) { 6445 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6446 status = BAD_VALUE; 6447 } else { 6448 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6449 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6450 if (mTrack != NULL) { 6451 bool suspend = audio_is_bluetooth_sco_device( 6452 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6453 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6454 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6455 } 6456 } 6457 mDevice |= (uint32_t)value; 6458 } 6459 if (status == NO_ERROR) { 6460 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6461 if (status == INVALID_OPERATION) { 6462 mInput->stream->common.standby(&mInput->stream->common); 6463 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6464 keyValuePair.string()); 6465 } 6466 if (reconfig) { 6467 if (status == BAD_VALUE && 6468 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6469 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6470 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6471 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6472 (reqChannelCount <= FCC_2)) { 6473 status = NO_ERROR; 6474 } 6475 if (status == NO_ERROR) { 6476 readInputParameters(); 6477 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6478 } 6479 } 6480 } 6481 6482 mNewParameters.removeAt(0); 6483 6484 mParamStatus = status; 6485 mParamCond.signal(); 6486 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6487 // already timed out waiting for the status and will never signal the condition. 6488 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6489 } 6490 return reconfig; 6491} 6492 6493String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6494{ 6495 char *s; 6496 String8 out_s8 = String8(); 6497 6498 Mutex::Autolock _l(mLock); 6499 if (initCheck() != NO_ERROR) { 6500 return out_s8; 6501 } 6502 6503 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6504 out_s8 = String8(s); 6505 free(s); 6506 return out_s8; 6507} 6508 6509void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6510 AudioSystem::OutputDescriptor desc; 6511 void *param2 = NULL; 6512 6513 switch (event) { 6514 case AudioSystem::INPUT_OPENED: 6515 case AudioSystem::INPUT_CONFIG_CHANGED: 6516 desc.channels = mChannelMask; 6517 desc.samplingRate = mSampleRate; 6518 desc.format = mFormat; 6519 desc.frameCount = mFrameCount; 6520 desc.latency = 0; 6521 param2 = &desc; 6522 break; 6523 6524 case AudioSystem::INPUT_CLOSED: 6525 default: 6526 break; 6527 } 6528 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6529} 6530 6531void AudioFlinger::RecordThread::readInputParameters() 6532{ 6533 delete mRsmpInBuffer; 6534 // mRsmpInBuffer is always assigned a new[] below 6535 delete mRsmpOutBuffer; 6536 mRsmpOutBuffer = NULL; 6537 delete mResampler; 6538 mResampler = NULL; 6539 6540 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6541 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6542 mChannelCount = (uint16_t)popcount(mChannelMask); 6543 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6544 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6545 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6546 mFrameCount = mInputBytes / mFrameSize; 6547 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6548 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6549 6550 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6551 { 6552 int channelCount; 6553 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6554 // stereo to mono post process as the resampler always outputs stereo. 6555 if (mChannelCount == 1 && mReqChannelCount == 2) { 6556 channelCount = 1; 6557 } else { 6558 channelCount = 2; 6559 } 6560 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6561 mResampler->setSampleRate(mSampleRate); 6562 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6563 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6564 6565 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6566 if (mChannelCount == 1 && mReqChannelCount == 1) { 6567 mFrameCount >>= 1; 6568 } 6569 6570 } 6571 mRsmpInIndex = mFrameCount; 6572} 6573 6574unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6575{ 6576 Mutex::Autolock _l(mLock); 6577 if (initCheck() != NO_ERROR) { 6578 return 0; 6579 } 6580 6581 return mInput->stream->get_input_frames_lost(mInput->stream); 6582} 6583 6584uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6585{ 6586 Mutex::Autolock _l(mLock); 6587 uint32_t result = 0; 6588 if (getEffectChain_l(sessionId) != 0) { 6589 result = EFFECT_SESSION; 6590 } 6591 6592 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6593 result |= TRACK_SESSION; 6594 } 6595 6596 return result; 6597} 6598 6599AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6600{ 6601 Mutex::Autolock _l(mLock); 6602 return mTrack; 6603} 6604 6605AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6606{ 6607 Mutex::Autolock _l(mLock); 6608 return mInput; 6609} 6610 6611AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6612{ 6613 Mutex::Autolock _l(mLock); 6614 AudioStreamIn *input = mInput; 6615 mInput = NULL; 6616 return input; 6617} 6618 6619// this method must always be called either with ThreadBase mLock held or inside the thread loop 6620audio_stream_t* AudioFlinger::RecordThread::stream() const 6621{ 6622 if (mInput == NULL) { 6623 return NULL; 6624 } 6625 return &mInput->stream->common; 6626} 6627 6628 6629// ---------------------------------------------------------------------------- 6630 6631audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6632{ 6633 if (!settingsAllowed()) { 6634 return 0; 6635 } 6636 Mutex::Autolock _l(mLock); 6637 return loadHwModule_l(name); 6638} 6639 6640// loadHwModule_l() must be called with AudioFlinger::mLock held 6641audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6642{ 6643 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6644 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6645 ALOGW("loadHwModule() module %s already loaded", name); 6646 return mAudioHwDevs.keyAt(i); 6647 } 6648 } 6649 6650 audio_hw_device_t *dev; 6651 6652 int rc = load_audio_interface(name, &dev); 6653 if (rc) { 6654 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6655 return 0; 6656 } 6657 6658 mHardwareStatus = AUDIO_HW_INIT; 6659 rc = dev->init_check(dev); 6660 mHardwareStatus = AUDIO_HW_IDLE; 6661 if (rc) { 6662 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6663 return 0; 6664 } 6665 6666 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6667 (NULL != dev->set_master_volume)) { 6668 AutoMutex lock(mHardwareLock); 6669 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6670 dev->set_master_volume(dev, mMasterVolume); 6671 mHardwareStatus = AUDIO_HW_IDLE; 6672 } 6673 6674 audio_module_handle_t handle = nextUniqueId(); 6675 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6676 6677 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6678 name, dev->common.module->name, dev->common.module->id, handle); 6679 6680 return handle; 6681 6682} 6683 6684audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6685 audio_devices_t *pDevices, 6686 uint32_t *pSamplingRate, 6687 audio_format_t *pFormat, 6688 audio_channel_mask_t *pChannelMask, 6689 uint32_t *pLatencyMs, 6690 audio_output_flags_t flags) 6691{ 6692 status_t status; 6693 PlaybackThread *thread = NULL; 6694 struct audio_config config = { 6695 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6696 channel_mask: pChannelMask ? *pChannelMask : 0, 6697 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6698 }; 6699 audio_stream_out_t *outStream = NULL; 6700 audio_hw_device_t *outHwDev; 6701 6702 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6703 module, 6704 (pDevices != NULL) ? (int)*pDevices : 0, 6705 config.sample_rate, 6706 config.format, 6707 config.channel_mask, 6708 flags); 6709 6710 if (pDevices == NULL || *pDevices == 0) { 6711 return 0; 6712 } 6713 6714 Mutex::Autolock _l(mLock); 6715 6716 outHwDev = findSuitableHwDev_l(module, *pDevices); 6717 if (outHwDev == NULL) 6718 return 0; 6719 6720 audio_io_handle_t id = nextUniqueId(); 6721 6722 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6723 6724 status = outHwDev->open_output_stream(outHwDev, 6725 id, 6726 *pDevices, 6727 (audio_output_flags_t)flags, 6728 &config, 6729 &outStream); 6730 6731 mHardwareStatus = AUDIO_HW_IDLE; 6732 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6733 outStream, 6734 config.sample_rate, 6735 config.format, 6736 config.channel_mask, 6737 status); 6738 6739 if (status == NO_ERROR && outStream != NULL) { 6740 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6741 6742 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6743 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6744 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6745 thread = new DirectOutputThread(this, output, id, *pDevices); 6746 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6747 } else { 6748 thread = new MixerThread(this, output, id, *pDevices); 6749 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6750 } 6751 mPlaybackThreads.add(id, thread); 6752 6753 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6754 if (pFormat != NULL) *pFormat = config.format; 6755 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6756 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6757 6758 // notify client processes of the new output creation 6759 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6760 6761 // the first primary output opened designates the primary hw device 6762 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6763 ALOGI("Using module %d has the primary audio interface", module); 6764 mPrimaryHardwareDev = outHwDev; 6765 6766 AutoMutex lock(mHardwareLock); 6767 mHardwareStatus = AUDIO_HW_SET_MODE; 6768 outHwDev->set_mode(outHwDev, mMode); 6769 6770 // Determine the level of master volume support the primary audio HAL has, 6771 // and set the initial master volume at the same time. 6772 float initialVolume = 1.0; 6773 mMasterVolumeSupportLvl = MVS_NONE; 6774 6775 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6776 if ((NULL != outHwDev->get_master_volume) && 6777 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6778 mMasterVolumeSupportLvl = MVS_FULL; 6779 } else { 6780 mMasterVolumeSupportLvl = MVS_SETONLY; 6781 initialVolume = 1.0; 6782 } 6783 6784 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6785 if ((NULL == outHwDev->set_master_volume) || 6786 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6787 mMasterVolumeSupportLvl = MVS_NONE; 6788 } 6789 // now that we have a primary device, initialize master volume on other devices 6790 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6791 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6792 6793 if ((dev != mPrimaryHardwareDev) && 6794 (NULL != dev->set_master_volume)) { 6795 dev->set_master_volume(dev, initialVolume); 6796 } 6797 } 6798 mHardwareStatus = AUDIO_HW_IDLE; 6799 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6800 ? initialVolume 6801 : 1.0; 6802 mMasterVolume = initialVolume; 6803 } 6804 return id; 6805 } 6806 6807 return 0; 6808} 6809 6810audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6811 audio_io_handle_t output2) 6812{ 6813 Mutex::Autolock _l(mLock); 6814 MixerThread *thread1 = checkMixerThread_l(output1); 6815 MixerThread *thread2 = checkMixerThread_l(output2); 6816 6817 if (thread1 == NULL || thread2 == NULL) { 6818 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6819 return 0; 6820 } 6821 6822 audio_io_handle_t id = nextUniqueId(); 6823 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6824 thread->addOutputTrack(thread2); 6825 mPlaybackThreads.add(id, thread); 6826 // notify client processes of the new output creation 6827 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6828 return id; 6829} 6830 6831status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6832{ 6833 // keep strong reference on the playback thread so that 6834 // it is not destroyed while exit() is executed 6835 sp<PlaybackThread> thread; 6836 { 6837 Mutex::Autolock _l(mLock); 6838 thread = checkPlaybackThread_l(output); 6839 if (thread == NULL) { 6840 return BAD_VALUE; 6841 } 6842 6843 ALOGV("closeOutput() %d", output); 6844 6845 if (thread->type() == ThreadBase::MIXER) { 6846 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6847 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6848 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6849 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6850 } 6851 } 6852 } 6853 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6854 mPlaybackThreads.removeItem(output); 6855 } 6856 thread->exit(); 6857 // The thread entity (active unit of execution) is no longer running here, 6858 // but the ThreadBase container still exists. 6859 6860 if (thread->type() != ThreadBase::DUPLICATING) { 6861 AudioStreamOut *out = thread->clearOutput(); 6862 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6863 // from now on thread->mOutput is NULL 6864 out->hwDev->close_output_stream(out->hwDev, out->stream); 6865 delete out; 6866 } 6867 return NO_ERROR; 6868} 6869 6870status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6871{ 6872 Mutex::Autolock _l(mLock); 6873 PlaybackThread *thread = checkPlaybackThread_l(output); 6874 6875 if (thread == NULL) { 6876 return BAD_VALUE; 6877 } 6878 6879 ALOGV("suspendOutput() %d", output); 6880 thread->suspend(); 6881 6882 return NO_ERROR; 6883} 6884 6885status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6886{ 6887 Mutex::Autolock _l(mLock); 6888 PlaybackThread *thread = checkPlaybackThread_l(output); 6889 6890 if (thread == NULL) { 6891 return BAD_VALUE; 6892 } 6893 6894 ALOGV("restoreOutput() %d", output); 6895 6896 thread->restore(); 6897 6898 return NO_ERROR; 6899} 6900 6901audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6902 audio_devices_t *pDevices, 6903 uint32_t *pSamplingRate, 6904 audio_format_t *pFormat, 6905 uint32_t *pChannelMask) 6906{ 6907 status_t status; 6908 RecordThread *thread = NULL; 6909 struct audio_config config = { 6910 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6911 channel_mask: pChannelMask ? *pChannelMask : 0, 6912 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6913 }; 6914 uint32_t reqSamplingRate = config.sample_rate; 6915 audio_format_t reqFormat = config.format; 6916 audio_channel_mask_t reqChannels = config.channel_mask; 6917 audio_stream_in_t *inStream = NULL; 6918 audio_hw_device_t *inHwDev; 6919 6920 if (pDevices == NULL || *pDevices == 0) { 6921 return 0; 6922 } 6923 6924 Mutex::Autolock _l(mLock); 6925 6926 inHwDev = findSuitableHwDev_l(module, *pDevices); 6927 if (inHwDev == NULL) 6928 return 0; 6929 6930 audio_io_handle_t id = nextUniqueId(); 6931 6932 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6933 &inStream); 6934 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6935 inStream, 6936 config.sample_rate, 6937 config.format, 6938 config.channel_mask, 6939 status); 6940 6941 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6942 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6943 // or stereo to mono conversions on 16 bit PCM inputs. 6944 if (status == BAD_VALUE && 6945 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6946 (config.sample_rate <= 2 * reqSamplingRate) && 6947 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6948 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6949 inStream = NULL; 6950 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6951 } 6952 6953 if (status == NO_ERROR && inStream != NULL) { 6954 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6955 6956 // Start record thread 6957 // RecorThread require both input and output device indication to forward to audio 6958 // pre processing modules 6959 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6960 thread = new RecordThread(this, 6961 input, 6962 reqSamplingRate, 6963 reqChannels, 6964 id, 6965 device); 6966 mRecordThreads.add(id, thread); 6967 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6968 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6969 if (pFormat != NULL) *pFormat = config.format; 6970 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6971 6972 input->stream->common.standby(&input->stream->common); 6973 6974 // notify client processes of the new input creation 6975 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6976 return id; 6977 } 6978 6979 return 0; 6980} 6981 6982status_t AudioFlinger::closeInput(audio_io_handle_t input) 6983{ 6984 // keep strong reference on the record thread so that 6985 // it is not destroyed while exit() is executed 6986 sp<RecordThread> thread; 6987 { 6988 Mutex::Autolock _l(mLock); 6989 thread = checkRecordThread_l(input); 6990 if (thread == NULL) { 6991 return BAD_VALUE; 6992 } 6993 6994 ALOGV("closeInput() %d", input); 6995 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6996 mRecordThreads.removeItem(input); 6997 } 6998 thread->exit(); 6999 // The thread entity (active unit of execution) is no longer running here, 7000 // but the ThreadBase container still exists. 7001 7002 AudioStreamIn *in = thread->clearInput(); 7003 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7004 // from now on thread->mInput is NULL 7005 in->hwDev->close_input_stream(in->hwDev, in->stream); 7006 delete in; 7007 7008 return NO_ERROR; 7009} 7010 7011status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7012{ 7013 Mutex::Autolock _l(mLock); 7014 MixerThread *dstThread = checkMixerThread_l(output); 7015 if (dstThread == NULL) { 7016 ALOGW("setStreamOutput() bad output id %d", output); 7017 return BAD_VALUE; 7018 } 7019 7020 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7021 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7022 7023 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7024 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7025 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7026 MixerThread *srcThread = (MixerThread *)thread; 7027 srcThread->invalidateTracks(stream); 7028 } 7029 } 7030 7031 return NO_ERROR; 7032} 7033 7034 7035int AudioFlinger::newAudioSessionId() 7036{ 7037 return nextUniqueId(); 7038} 7039 7040void AudioFlinger::acquireAudioSessionId(int audioSession) 7041{ 7042 Mutex::Autolock _l(mLock); 7043 pid_t caller = IPCThreadState::self()->getCallingPid(); 7044 ALOGV("acquiring %d from %d", audioSession, caller); 7045 size_t num = mAudioSessionRefs.size(); 7046 for (size_t i = 0; i< num; i++) { 7047 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7048 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7049 ref->mCnt++; 7050 ALOGV(" incremented refcount to %d", ref->mCnt); 7051 return; 7052 } 7053 } 7054 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7055 ALOGV(" added new entry for %d", audioSession); 7056} 7057 7058void AudioFlinger::releaseAudioSessionId(int audioSession) 7059{ 7060 Mutex::Autolock _l(mLock); 7061 pid_t caller = IPCThreadState::self()->getCallingPid(); 7062 ALOGV("releasing %d from %d", audioSession, caller); 7063 size_t num = mAudioSessionRefs.size(); 7064 for (size_t i = 0; i< num; i++) { 7065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7066 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7067 ref->mCnt--; 7068 ALOGV(" decremented refcount to %d", ref->mCnt); 7069 if (ref->mCnt == 0) { 7070 mAudioSessionRefs.removeAt(i); 7071 delete ref; 7072 purgeStaleEffects_l(); 7073 } 7074 return; 7075 } 7076 } 7077 ALOGW("session id %d not found for pid %d", audioSession, caller); 7078} 7079 7080void AudioFlinger::purgeStaleEffects_l() { 7081 7082 ALOGV("purging stale effects"); 7083 7084 Vector< sp<EffectChain> > chains; 7085 7086 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7087 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7088 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7089 sp<EffectChain> ec = t->mEffectChains[j]; 7090 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7091 chains.push(ec); 7092 } 7093 } 7094 } 7095 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7096 sp<RecordThread> t = mRecordThreads.valueAt(i); 7097 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7098 sp<EffectChain> ec = t->mEffectChains[j]; 7099 chains.push(ec); 7100 } 7101 } 7102 7103 for (size_t i = 0; i < chains.size(); i++) { 7104 sp<EffectChain> ec = chains[i]; 7105 int sessionid = ec->sessionId(); 7106 sp<ThreadBase> t = ec->mThread.promote(); 7107 if (t == 0) { 7108 continue; 7109 } 7110 size_t numsessionrefs = mAudioSessionRefs.size(); 7111 bool found = false; 7112 for (size_t k = 0; k < numsessionrefs; k++) { 7113 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7114 if (ref->mSessionid == sessionid) { 7115 ALOGV(" session %d still exists for %d with %d refs", 7116 sessionid, ref->mPid, ref->mCnt); 7117 found = true; 7118 break; 7119 } 7120 } 7121 if (!found) { 7122 // remove all effects from the chain 7123 while (ec->mEffects.size()) { 7124 sp<EffectModule> effect = ec->mEffects[0]; 7125 effect->unPin(); 7126 Mutex::Autolock _l (t->mLock); 7127 t->removeEffect_l(effect); 7128 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7129 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7130 if (handle != 0) { 7131 handle->mEffect.clear(); 7132 if (handle->mHasControl && handle->mEnabled) { 7133 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7134 } 7135 } 7136 } 7137 AudioSystem::unregisterEffect(effect->id()); 7138 } 7139 } 7140 } 7141 return; 7142} 7143 7144// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7145AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7146{ 7147 return mPlaybackThreads.valueFor(output).get(); 7148} 7149 7150// checkMixerThread_l() must be called with AudioFlinger::mLock held 7151AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7152{ 7153 PlaybackThread *thread = checkPlaybackThread_l(output); 7154 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7155} 7156 7157// checkRecordThread_l() must be called with AudioFlinger::mLock held 7158AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7159{ 7160 return mRecordThreads.valueFor(input).get(); 7161} 7162 7163uint32_t AudioFlinger::nextUniqueId() 7164{ 7165 return android_atomic_inc(&mNextUniqueId); 7166} 7167 7168AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7169{ 7170 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7171 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7172 AudioStreamOut *output = thread->getOutput(); 7173 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7174 return thread; 7175 } 7176 } 7177 return NULL; 7178} 7179 7180uint32_t AudioFlinger::primaryOutputDevice_l() const 7181{ 7182 PlaybackThread *thread = primaryPlaybackThread_l(); 7183 7184 if (thread == NULL) { 7185 return 0; 7186 } 7187 7188 return thread->device(); 7189} 7190 7191sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7192 int triggerSession, 7193 int listenerSession, 7194 sync_event_callback_t callBack, 7195 void *cookie) 7196{ 7197 Mutex::Autolock _l(mLock); 7198 7199 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7200 status_t playStatus = NAME_NOT_FOUND; 7201 status_t recStatus = NAME_NOT_FOUND; 7202 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7203 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7204 if (playStatus == NO_ERROR) { 7205 return event; 7206 } 7207 } 7208 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7209 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7210 if (recStatus == NO_ERROR) { 7211 return event; 7212 } 7213 } 7214 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7215 mPendingSyncEvents.add(event); 7216 } else { 7217 ALOGV("createSyncEvent() invalid event %d", event->type()); 7218 event.clear(); 7219 } 7220 return event; 7221} 7222 7223// ---------------------------------------------------------------------------- 7224// Effect management 7225// ---------------------------------------------------------------------------- 7226 7227 7228status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7229{ 7230 Mutex::Autolock _l(mLock); 7231 return EffectQueryNumberEffects(numEffects); 7232} 7233 7234status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7235{ 7236 Mutex::Autolock _l(mLock); 7237 return EffectQueryEffect(index, descriptor); 7238} 7239 7240status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7241 effect_descriptor_t *descriptor) const 7242{ 7243 Mutex::Autolock _l(mLock); 7244 return EffectGetDescriptor(pUuid, descriptor); 7245} 7246 7247 7248sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7249 effect_descriptor_t *pDesc, 7250 const sp<IEffectClient>& effectClient, 7251 int32_t priority, 7252 audio_io_handle_t io, 7253 int sessionId, 7254 status_t *status, 7255 int *id, 7256 int *enabled) 7257{ 7258 status_t lStatus = NO_ERROR; 7259 sp<EffectHandle> handle; 7260 effect_descriptor_t desc; 7261 7262 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7263 pid, effectClient.get(), priority, sessionId, io); 7264 7265 if (pDesc == NULL) { 7266 lStatus = BAD_VALUE; 7267 goto Exit; 7268 } 7269 7270 // check audio settings permission for global effects 7271 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7272 lStatus = PERMISSION_DENIED; 7273 goto Exit; 7274 } 7275 7276 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7277 // that can only be created by audio policy manager (running in same process) 7278 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7279 lStatus = PERMISSION_DENIED; 7280 goto Exit; 7281 } 7282 7283 if (io == 0) { 7284 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7285 // output must be specified by AudioPolicyManager when using session 7286 // AUDIO_SESSION_OUTPUT_STAGE 7287 lStatus = BAD_VALUE; 7288 goto Exit; 7289 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7290 // if the output returned by getOutputForEffect() is removed before we lock the 7291 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7292 // and we will exit safely 7293 io = AudioSystem::getOutputForEffect(&desc); 7294 } 7295 } 7296 7297 { 7298 Mutex::Autolock _l(mLock); 7299 7300 7301 if (!EffectIsNullUuid(&pDesc->uuid)) { 7302 // if uuid is specified, request effect descriptor 7303 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7304 if (lStatus < 0) { 7305 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7306 goto Exit; 7307 } 7308 } else { 7309 // if uuid is not specified, look for an available implementation 7310 // of the required type in effect factory 7311 if (EffectIsNullUuid(&pDesc->type)) { 7312 ALOGW("createEffect() no effect type"); 7313 lStatus = BAD_VALUE; 7314 goto Exit; 7315 } 7316 uint32_t numEffects = 0; 7317 effect_descriptor_t d; 7318 d.flags = 0; // prevent compiler warning 7319 bool found = false; 7320 7321 lStatus = EffectQueryNumberEffects(&numEffects); 7322 if (lStatus < 0) { 7323 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7324 goto Exit; 7325 } 7326 for (uint32_t i = 0; i < numEffects; i++) { 7327 lStatus = EffectQueryEffect(i, &desc); 7328 if (lStatus < 0) { 7329 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7330 continue; 7331 } 7332 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7333 // If matching type found save effect descriptor. If the session is 7334 // 0 and the effect is not auxiliary, continue enumeration in case 7335 // an auxiliary version of this effect type is available 7336 found = true; 7337 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7338 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7339 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7340 break; 7341 } 7342 } 7343 } 7344 if (!found) { 7345 lStatus = BAD_VALUE; 7346 ALOGW("createEffect() effect not found"); 7347 goto Exit; 7348 } 7349 // For same effect type, chose auxiliary version over insert version if 7350 // connect to output mix (Compliance to OpenSL ES) 7351 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7352 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7353 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7354 } 7355 } 7356 7357 // Do not allow auxiliary effects on a session different from 0 (output mix) 7358 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7359 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7360 lStatus = INVALID_OPERATION; 7361 goto Exit; 7362 } 7363 7364 // check recording permission for visualizer 7365 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7366 !recordingAllowed()) { 7367 lStatus = PERMISSION_DENIED; 7368 goto Exit; 7369 } 7370 7371 // return effect descriptor 7372 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7373 7374 // If output is not specified try to find a matching audio session ID in one of the 7375 // output threads. 7376 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7377 // because of code checking output when entering the function. 7378 // Note: io is never 0 when creating an effect on an input 7379 if (io == 0) { 7380 // look for the thread where the specified audio session is present 7381 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7382 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7383 io = mPlaybackThreads.keyAt(i); 7384 break; 7385 } 7386 } 7387 if (io == 0) { 7388 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7389 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7390 io = mRecordThreads.keyAt(i); 7391 break; 7392 } 7393 } 7394 } 7395 // If no output thread contains the requested session ID, default to 7396 // first output. The effect chain will be moved to the correct output 7397 // thread when a track with the same session ID is created 7398 if (io == 0 && mPlaybackThreads.size()) { 7399 io = mPlaybackThreads.keyAt(0); 7400 } 7401 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7402 } 7403 ThreadBase *thread = checkRecordThread_l(io); 7404 if (thread == NULL) { 7405 thread = checkPlaybackThread_l(io); 7406 if (thread == NULL) { 7407 ALOGE("createEffect() unknown output thread"); 7408 lStatus = BAD_VALUE; 7409 goto Exit; 7410 } 7411 } 7412 7413 sp<Client> client = registerPid_l(pid); 7414 7415 // create effect on selected output thread 7416 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7417 &desc, enabled, &lStatus); 7418 if (handle != 0 && id != NULL) { 7419 *id = handle->id(); 7420 } 7421 } 7422 7423Exit: 7424 if (status != NULL) { 7425 *status = lStatus; 7426 } 7427 return handle; 7428} 7429 7430status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7431 audio_io_handle_t dstOutput) 7432{ 7433 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7434 sessionId, srcOutput, dstOutput); 7435 Mutex::Autolock _l(mLock); 7436 if (srcOutput == dstOutput) { 7437 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7438 return NO_ERROR; 7439 } 7440 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7441 if (srcThread == NULL) { 7442 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7443 return BAD_VALUE; 7444 } 7445 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7446 if (dstThread == NULL) { 7447 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7448 return BAD_VALUE; 7449 } 7450 7451 Mutex::Autolock _dl(dstThread->mLock); 7452 Mutex::Autolock _sl(srcThread->mLock); 7453 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7454 7455 return NO_ERROR; 7456} 7457 7458// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7459status_t AudioFlinger::moveEffectChain_l(int sessionId, 7460 AudioFlinger::PlaybackThread *srcThread, 7461 AudioFlinger::PlaybackThread *dstThread, 7462 bool reRegister) 7463{ 7464 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7465 sessionId, srcThread, dstThread); 7466 7467 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7468 if (chain == 0) { 7469 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7470 sessionId, srcThread); 7471 return INVALID_OPERATION; 7472 } 7473 7474 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7475 // so that a new chain is created with correct parameters when first effect is added. This is 7476 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7477 // removed. 7478 srcThread->removeEffectChain_l(chain); 7479 7480 // transfer all effects one by one so that new effect chain is created on new thread with 7481 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7482 audio_io_handle_t dstOutput = dstThread->id(); 7483 sp<EffectChain> dstChain; 7484 uint32_t strategy = 0; // prevent compiler warning 7485 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7486 while (effect != 0) { 7487 srcThread->removeEffect_l(effect); 7488 dstThread->addEffect_l(effect); 7489 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7490 if (effect->state() == EffectModule::ACTIVE || 7491 effect->state() == EffectModule::STOPPING) { 7492 effect->start(); 7493 } 7494 // if the move request is not received from audio policy manager, the effect must be 7495 // re-registered with the new strategy and output 7496 if (dstChain == 0) { 7497 dstChain = effect->chain().promote(); 7498 if (dstChain == 0) { 7499 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7500 srcThread->addEffect_l(effect); 7501 return NO_INIT; 7502 } 7503 strategy = dstChain->strategy(); 7504 } 7505 if (reRegister) { 7506 AudioSystem::unregisterEffect(effect->id()); 7507 AudioSystem::registerEffect(&effect->desc(), 7508 dstOutput, 7509 strategy, 7510 sessionId, 7511 effect->id()); 7512 } 7513 effect = chain->getEffectFromId_l(0); 7514 } 7515 7516 return NO_ERROR; 7517} 7518 7519 7520// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7521sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7522 const sp<AudioFlinger::Client>& client, 7523 const sp<IEffectClient>& effectClient, 7524 int32_t priority, 7525 int sessionId, 7526 effect_descriptor_t *desc, 7527 int *enabled, 7528 status_t *status 7529 ) 7530{ 7531 sp<EffectModule> effect; 7532 sp<EffectHandle> handle; 7533 status_t lStatus; 7534 sp<EffectChain> chain; 7535 bool chainCreated = false; 7536 bool effectCreated = false; 7537 bool effectRegistered = false; 7538 7539 lStatus = initCheck(); 7540 if (lStatus != NO_ERROR) { 7541 ALOGW("createEffect_l() Audio driver not initialized."); 7542 goto Exit; 7543 } 7544 7545 // Do not allow effects with session ID 0 on direct output or duplicating threads 7546 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7547 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7548 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7549 desc->name, sessionId); 7550 lStatus = BAD_VALUE; 7551 goto Exit; 7552 } 7553 // Only Pre processor effects are allowed on input threads and only on input threads 7554 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7555 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7556 desc->name, desc->flags, mType); 7557 lStatus = BAD_VALUE; 7558 goto Exit; 7559 } 7560 7561 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7562 7563 { // scope for mLock 7564 Mutex::Autolock _l(mLock); 7565 7566 // check for existing effect chain with the requested audio session 7567 chain = getEffectChain_l(sessionId); 7568 if (chain == 0) { 7569 // create a new chain for this session 7570 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7571 chain = new EffectChain(this, sessionId); 7572 addEffectChain_l(chain); 7573 chain->setStrategy(getStrategyForSession_l(sessionId)); 7574 chainCreated = true; 7575 } else { 7576 effect = chain->getEffectFromDesc_l(desc); 7577 } 7578 7579 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7580 7581 if (effect == 0) { 7582 int id = mAudioFlinger->nextUniqueId(); 7583 // Check CPU and memory usage 7584 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7585 if (lStatus != NO_ERROR) { 7586 goto Exit; 7587 } 7588 effectRegistered = true; 7589 // create a new effect module if none present in the chain 7590 effect = new EffectModule(this, chain, desc, id, sessionId); 7591 lStatus = effect->status(); 7592 if (lStatus != NO_ERROR) { 7593 goto Exit; 7594 } 7595 lStatus = chain->addEffect_l(effect); 7596 if (lStatus != NO_ERROR) { 7597 goto Exit; 7598 } 7599 effectCreated = true; 7600 7601 effect->setDevice(mDevice); 7602 effect->setMode(mAudioFlinger->getMode()); 7603 } 7604 // create effect handle and connect it to effect module 7605 handle = new EffectHandle(effect, client, effectClient, priority); 7606 lStatus = effect->addHandle(handle); 7607 if (enabled != NULL) { 7608 *enabled = (int)effect->isEnabled(); 7609 } 7610 } 7611 7612Exit: 7613 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7614 Mutex::Autolock _l(mLock); 7615 if (effectCreated) { 7616 chain->removeEffect_l(effect); 7617 } 7618 if (effectRegistered) { 7619 AudioSystem::unregisterEffect(effect->id()); 7620 } 7621 if (chainCreated) { 7622 removeEffectChain_l(chain); 7623 } 7624 handle.clear(); 7625 } 7626 7627 if (status != NULL) { 7628 *status = lStatus; 7629 } 7630 return handle; 7631} 7632 7633sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7634{ 7635 sp<EffectChain> chain = getEffectChain_l(sessionId); 7636 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7637} 7638 7639// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7640// PlaybackThread::mLock held 7641status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7642{ 7643 // check for existing effect chain with the requested audio session 7644 int sessionId = effect->sessionId(); 7645 sp<EffectChain> chain = getEffectChain_l(sessionId); 7646 bool chainCreated = false; 7647 7648 if (chain == 0) { 7649 // create a new chain for this session 7650 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7651 chain = new EffectChain(this, sessionId); 7652 addEffectChain_l(chain); 7653 chain->setStrategy(getStrategyForSession_l(sessionId)); 7654 chainCreated = true; 7655 } 7656 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7657 7658 if (chain->getEffectFromId_l(effect->id()) != 0) { 7659 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7660 this, effect->desc().name, chain.get()); 7661 return BAD_VALUE; 7662 } 7663 7664 status_t status = chain->addEffect_l(effect); 7665 if (status != NO_ERROR) { 7666 if (chainCreated) { 7667 removeEffectChain_l(chain); 7668 } 7669 return status; 7670 } 7671 7672 effect->setDevice(mDevice); 7673 effect->setMode(mAudioFlinger->getMode()); 7674 return NO_ERROR; 7675} 7676 7677void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7678 7679 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7680 effect_descriptor_t desc = effect->desc(); 7681 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7682 detachAuxEffect_l(effect->id()); 7683 } 7684 7685 sp<EffectChain> chain = effect->chain().promote(); 7686 if (chain != 0) { 7687 // remove effect chain if removing last effect 7688 if (chain->removeEffect_l(effect) == 0) { 7689 removeEffectChain_l(chain); 7690 } 7691 } else { 7692 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7693 } 7694} 7695 7696void AudioFlinger::ThreadBase::lockEffectChains_l( 7697 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7698{ 7699 effectChains = mEffectChains; 7700 for (size_t i = 0; i < mEffectChains.size(); i++) { 7701 mEffectChains[i]->lock(); 7702 } 7703} 7704 7705void AudioFlinger::ThreadBase::unlockEffectChains( 7706 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7707{ 7708 for (size_t i = 0; i < effectChains.size(); i++) { 7709 effectChains[i]->unlock(); 7710 } 7711} 7712 7713sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7714{ 7715 Mutex::Autolock _l(mLock); 7716 return getEffectChain_l(sessionId); 7717} 7718 7719sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7720{ 7721 size_t size = mEffectChains.size(); 7722 for (size_t i = 0; i < size; i++) { 7723 if (mEffectChains[i]->sessionId() == sessionId) { 7724 return mEffectChains[i]; 7725 } 7726 } 7727 return 0; 7728} 7729 7730void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7731{ 7732 Mutex::Autolock _l(mLock); 7733 size_t size = mEffectChains.size(); 7734 for (size_t i = 0; i < size; i++) { 7735 mEffectChains[i]->setMode_l(mode); 7736 } 7737} 7738 7739void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7740 const wp<EffectHandle>& handle, 7741 bool unpinIfLast) { 7742 7743 Mutex::Autolock _l(mLock); 7744 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7745 // delete the effect module if removing last handle on it 7746 if (effect->removeHandle(handle) == 0) { 7747 if (!effect->isPinned() || unpinIfLast) { 7748 removeEffect_l(effect); 7749 AudioSystem::unregisterEffect(effect->id()); 7750 } 7751 } 7752} 7753 7754status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7755{ 7756 int session = chain->sessionId(); 7757 int16_t *buffer = mMixBuffer; 7758 bool ownsBuffer = false; 7759 7760 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7761 if (session > 0) { 7762 // Only one effect chain can be present in direct output thread and it uses 7763 // the mix buffer as input 7764 if (mType != DIRECT) { 7765 size_t numSamples = mNormalFrameCount * mChannelCount; 7766 buffer = new int16_t[numSamples]; 7767 memset(buffer, 0, numSamples * sizeof(int16_t)); 7768 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7769 ownsBuffer = true; 7770 } 7771 7772 // Attach all tracks with same session ID to this chain. 7773 for (size_t i = 0; i < mTracks.size(); ++i) { 7774 sp<Track> track = mTracks[i]; 7775 if (session == track->sessionId()) { 7776 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7777 track->setMainBuffer(buffer); 7778 chain->incTrackCnt(); 7779 } 7780 } 7781 7782 // indicate all active tracks in the chain 7783 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7784 sp<Track> track = mActiveTracks[i].promote(); 7785 if (track == 0) continue; 7786 if (session == track->sessionId()) { 7787 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7788 chain->incActiveTrackCnt(); 7789 } 7790 } 7791 } 7792 7793 chain->setInBuffer(buffer, ownsBuffer); 7794 chain->setOutBuffer(mMixBuffer); 7795 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7796 // chains list in order to be processed last as it contains output stage effects 7797 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7798 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7799 // after track specific effects and before output stage 7800 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7801 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7802 // Effect chain for other sessions are inserted at beginning of effect 7803 // chains list to be processed before output mix effects. Relative order between other 7804 // sessions is not important 7805 size_t size = mEffectChains.size(); 7806 size_t i = 0; 7807 for (i = 0; i < size; i++) { 7808 if (mEffectChains[i]->sessionId() < session) break; 7809 } 7810 mEffectChains.insertAt(chain, i); 7811 checkSuspendOnAddEffectChain_l(chain); 7812 7813 return NO_ERROR; 7814} 7815 7816size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7817{ 7818 int session = chain->sessionId(); 7819 7820 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7821 7822 for (size_t i = 0; i < mEffectChains.size(); i++) { 7823 if (chain == mEffectChains[i]) { 7824 mEffectChains.removeAt(i); 7825 // detach all active tracks from the chain 7826 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7827 sp<Track> track = mActiveTracks[i].promote(); 7828 if (track == 0) continue; 7829 if (session == track->sessionId()) { 7830 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7831 chain.get(), session); 7832 chain->decActiveTrackCnt(); 7833 } 7834 } 7835 7836 // detach all tracks with same session ID from this chain 7837 for (size_t i = 0; i < mTracks.size(); ++i) { 7838 sp<Track> track = mTracks[i]; 7839 if (session == track->sessionId()) { 7840 track->setMainBuffer(mMixBuffer); 7841 chain->decTrackCnt(); 7842 } 7843 } 7844 break; 7845 } 7846 } 7847 return mEffectChains.size(); 7848} 7849 7850status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7851 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7852{ 7853 Mutex::Autolock _l(mLock); 7854 return attachAuxEffect_l(track, EffectId); 7855} 7856 7857status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7858 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7859{ 7860 status_t status = NO_ERROR; 7861 7862 if (EffectId == 0) { 7863 track->setAuxBuffer(0, NULL); 7864 } else { 7865 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7866 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7867 if (effect != 0) { 7868 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7869 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7870 } else { 7871 status = INVALID_OPERATION; 7872 } 7873 } else { 7874 status = BAD_VALUE; 7875 } 7876 } 7877 return status; 7878} 7879 7880void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7881{ 7882 for (size_t i = 0; i < mTracks.size(); ++i) { 7883 sp<Track> track = mTracks[i]; 7884 if (track->auxEffectId() == effectId) { 7885 attachAuxEffect_l(track, 0); 7886 } 7887 } 7888} 7889 7890status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7891{ 7892 // only one chain per input thread 7893 if (mEffectChains.size() != 0) { 7894 return INVALID_OPERATION; 7895 } 7896 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7897 7898 chain->setInBuffer(NULL); 7899 chain->setOutBuffer(NULL); 7900 7901 checkSuspendOnAddEffectChain_l(chain); 7902 7903 mEffectChains.add(chain); 7904 7905 return NO_ERROR; 7906} 7907 7908size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7909{ 7910 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7911 ALOGW_IF(mEffectChains.size() != 1, 7912 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7913 chain.get(), mEffectChains.size(), this); 7914 if (mEffectChains.size() == 1) { 7915 mEffectChains.removeAt(0); 7916 } 7917 return 0; 7918} 7919 7920// ---------------------------------------------------------------------------- 7921// EffectModule implementation 7922// ---------------------------------------------------------------------------- 7923 7924#undef LOG_TAG 7925#define LOG_TAG "AudioFlinger::EffectModule" 7926 7927AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7928 const wp<AudioFlinger::EffectChain>& chain, 7929 effect_descriptor_t *desc, 7930 int id, 7931 int sessionId) 7932 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7933 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7934{ 7935 ALOGV("Constructor %p", this); 7936 int lStatus; 7937 if (thread == NULL) { 7938 return; 7939 } 7940 7941 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7942 7943 // create effect engine from effect factory 7944 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7945 7946 if (mStatus != NO_ERROR) { 7947 return; 7948 } 7949 lStatus = init(); 7950 if (lStatus < 0) { 7951 mStatus = lStatus; 7952 goto Error; 7953 } 7954 7955 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7956 mPinned = true; 7957 } 7958 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7959 return; 7960Error: 7961 EffectRelease(mEffectInterface); 7962 mEffectInterface = NULL; 7963 ALOGV("Constructor Error %d", mStatus); 7964} 7965 7966AudioFlinger::EffectModule::~EffectModule() 7967{ 7968 ALOGV("Destructor %p", this); 7969 if (mEffectInterface != NULL) { 7970 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7971 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7972 sp<ThreadBase> thread = mThread.promote(); 7973 if (thread != 0) { 7974 audio_stream_t *stream = thread->stream(); 7975 if (stream != NULL) { 7976 stream->remove_audio_effect(stream, mEffectInterface); 7977 } 7978 } 7979 } 7980 // release effect engine 7981 EffectRelease(mEffectInterface); 7982 } 7983} 7984 7985status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7986{ 7987 status_t status; 7988 7989 Mutex::Autolock _l(mLock); 7990 int priority = handle->priority(); 7991 size_t size = mHandles.size(); 7992 sp<EffectHandle> h; 7993 size_t i; 7994 for (i = 0; i < size; i++) { 7995 h = mHandles[i].promote(); 7996 if (h == 0) continue; 7997 if (h->priority() <= priority) break; 7998 } 7999 // if inserted in first place, move effect control from previous owner to this handle 8000 if (i == 0) { 8001 bool enabled = false; 8002 if (h != 0) { 8003 enabled = h->enabled(); 8004 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8005 } 8006 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8007 status = NO_ERROR; 8008 } else { 8009 status = ALREADY_EXISTS; 8010 } 8011 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8012 mHandles.insertAt(handle, i); 8013 return status; 8014} 8015 8016size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8017{ 8018 Mutex::Autolock _l(mLock); 8019 size_t size = mHandles.size(); 8020 size_t i; 8021 for (i = 0; i < size; i++) { 8022 if (mHandles[i] == handle) break; 8023 } 8024 if (i == size) { 8025 return size; 8026 } 8027 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8028 8029 bool enabled = false; 8030 EffectHandle *hdl = handle.unsafe_get(); 8031 if (hdl != NULL) { 8032 ALOGV("removeHandle() unsafe_get OK"); 8033 enabled = hdl->enabled(); 8034 } 8035 mHandles.removeAt(i); 8036 size = mHandles.size(); 8037 // if removed from first place, move effect control from this handle to next in line 8038 if (i == 0 && size != 0) { 8039 sp<EffectHandle> h = mHandles[0].promote(); 8040 if (h != 0) { 8041 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8042 } 8043 } 8044 8045 // Prevent calls to process() and other functions on effect interface from now on. 8046 // The effect engine will be released by the destructor when the last strong reference on 8047 // this object is released which can happen after next process is called. 8048 if (size == 0 && !mPinned) { 8049 mState = DESTROYED; 8050 } 8051 8052 return size; 8053} 8054 8055sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8056{ 8057 Mutex::Autolock _l(mLock); 8058 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8059} 8060 8061void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8062{ 8063 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8064 // keep a strong reference on this EffectModule to avoid calling the 8065 // destructor before we exit 8066 sp<EffectModule> keep(this); 8067 { 8068 sp<ThreadBase> thread = mThread.promote(); 8069 if (thread != 0) { 8070 thread->disconnectEffect(keep, handle, unpinIfLast); 8071 } 8072 } 8073} 8074 8075void AudioFlinger::EffectModule::updateState() { 8076 Mutex::Autolock _l(mLock); 8077 8078 switch (mState) { 8079 case RESTART: 8080 reset_l(); 8081 // FALL THROUGH 8082 8083 case STARTING: 8084 // clear auxiliary effect input buffer for next accumulation 8085 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8086 memset(mConfig.inputCfg.buffer.raw, 8087 0, 8088 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8089 } 8090 start_l(); 8091 mState = ACTIVE; 8092 break; 8093 case STOPPING: 8094 stop_l(); 8095 mDisableWaitCnt = mMaxDisableWaitCnt; 8096 mState = STOPPED; 8097 break; 8098 case STOPPED: 8099 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8100 // turn off sequence. 8101 if (--mDisableWaitCnt == 0) { 8102 reset_l(); 8103 mState = IDLE; 8104 } 8105 break; 8106 default: //IDLE , ACTIVE, DESTROYED 8107 break; 8108 } 8109} 8110 8111void AudioFlinger::EffectModule::process() 8112{ 8113 Mutex::Autolock _l(mLock); 8114 8115 if (mState == DESTROYED || mEffectInterface == NULL || 8116 mConfig.inputCfg.buffer.raw == NULL || 8117 mConfig.outputCfg.buffer.raw == NULL) { 8118 return; 8119 } 8120 8121 if (isProcessEnabled()) { 8122 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8123 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8124 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8125 mConfig.inputCfg.buffer.s32, 8126 mConfig.inputCfg.buffer.frameCount/2); 8127 } 8128 8129 // do the actual processing in the effect engine 8130 int ret = (*mEffectInterface)->process(mEffectInterface, 8131 &mConfig.inputCfg.buffer, 8132 &mConfig.outputCfg.buffer); 8133 8134 // force transition to IDLE state when engine is ready 8135 if (mState == STOPPED && ret == -ENODATA) { 8136 mDisableWaitCnt = 1; 8137 } 8138 8139 // clear auxiliary effect input buffer for next accumulation 8140 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8141 memset(mConfig.inputCfg.buffer.raw, 0, 8142 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8143 } 8144 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8145 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8146 // If an insert effect is idle and input buffer is different from output buffer, 8147 // accumulate input onto output 8148 sp<EffectChain> chain = mChain.promote(); 8149 if (chain != 0 && chain->activeTrackCnt() != 0) { 8150 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8151 int16_t *in = mConfig.inputCfg.buffer.s16; 8152 int16_t *out = mConfig.outputCfg.buffer.s16; 8153 for (size_t i = 0; i < frameCnt; i++) { 8154 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8155 } 8156 } 8157 } 8158} 8159 8160void AudioFlinger::EffectModule::reset_l() 8161{ 8162 if (mEffectInterface == NULL) { 8163 return; 8164 } 8165 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8166} 8167 8168status_t AudioFlinger::EffectModule::configure() 8169{ 8170 uint32_t channels; 8171 if (mEffectInterface == NULL) { 8172 return NO_INIT; 8173 } 8174 8175 sp<ThreadBase> thread = mThread.promote(); 8176 if (thread == 0) { 8177 return DEAD_OBJECT; 8178 } 8179 8180 // TODO: handle configuration of effects replacing track process 8181 if (thread->channelCount() == 1) { 8182 channels = AUDIO_CHANNEL_OUT_MONO; 8183 } else { 8184 channels = AUDIO_CHANNEL_OUT_STEREO; 8185 } 8186 8187 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8188 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8189 } else { 8190 mConfig.inputCfg.channels = channels; 8191 } 8192 mConfig.outputCfg.channels = channels; 8193 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8194 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8195 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8196 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8197 mConfig.inputCfg.bufferProvider.cookie = NULL; 8198 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8199 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8200 mConfig.outputCfg.bufferProvider.cookie = NULL; 8201 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8202 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8203 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8204 // Insert effect: 8205 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8206 // always overwrites output buffer: input buffer == output buffer 8207 // - in other sessions: 8208 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8209 // other effect: overwrites output buffer: input buffer == output buffer 8210 // Auxiliary effect: 8211 // accumulates in output buffer: input buffer != output buffer 8212 // Therefore: accumulate <=> input buffer != output buffer 8213 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8214 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8215 } else { 8216 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8217 } 8218 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8219 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8220 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8221 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8222 8223 ALOGV("configure() %p thread %p buffer %p framecount %d", 8224 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8225 8226 status_t cmdStatus; 8227 uint32_t size = sizeof(int); 8228 status_t status = (*mEffectInterface)->command(mEffectInterface, 8229 EFFECT_CMD_SET_CONFIG, 8230 sizeof(effect_config_t), 8231 &mConfig, 8232 &size, 8233 &cmdStatus); 8234 if (status == 0) { 8235 status = cmdStatus; 8236 } 8237 8238 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8239 (1000 * mConfig.outputCfg.buffer.frameCount); 8240 8241 return status; 8242} 8243 8244status_t AudioFlinger::EffectModule::init() 8245{ 8246 Mutex::Autolock _l(mLock); 8247 if (mEffectInterface == NULL) { 8248 return NO_INIT; 8249 } 8250 status_t cmdStatus; 8251 uint32_t size = sizeof(status_t); 8252 status_t status = (*mEffectInterface)->command(mEffectInterface, 8253 EFFECT_CMD_INIT, 8254 0, 8255 NULL, 8256 &size, 8257 &cmdStatus); 8258 if (status == 0) { 8259 status = cmdStatus; 8260 } 8261 return status; 8262} 8263 8264status_t AudioFlinger::EffectModule::start() 8265{ 8266 Mutex::Autolock _l(mLock); 8267 return start_l(); 8268} 8269 8270status_t AudioFlinger::EffectModule::start_l() 8271{ 8272 if (mEffectInterface == NULL) { 8273 return NO_INIT; 8274 } 8275 status_t cmdStatus; 8276 uint32_t size = sizeof(status_t); 8277 status_t status = (*mEffectInterface)->command(mEffectInterface, 8278 EFFECT_CMD_ENABLE, 8279 0, 8280 NULL, 8281 &size, 8282 &cmdStatus); 8283 if (status == 0) { 8284 status = cmdStatus; 8285 } 8286 if (status == 0 && 8287 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8288 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8289 sp<ThreadBase> thread = mThread.promote(); 8290 if (thread != 0) { 8291 audio_stream_t *stream = thread->stream(); 8292 if (stream != NULL) { 8293 stream->add_audio_effect(stream, mEffectInterface); 8294 } 8295 } 8296 } 8297 return status; 8298} 8299 8300status_t AudioFlinger::EffectModule::stop() 8301{ 8302 Mutex::Autolock _l(mLock); 8303 return stop_l(); 8304} 8305 8306status_t AudioFlinger::EffectModule::stop_l() 8307{ 8308 if (mEffectInterface == NULL) { 8309 return NO_INIT; 8310 } 8311 status_t cmdStatus; 8312 uint32_t size = sizeof(status_t); 8313 status_t status = (*mEffectInterface)->command(mEffectInterface, 8314 EFFECT_CMD_DISABLE, 8315 0, 8316 NULL, 8317 &size, 8318 &cmdStatus); 8319 if (status == 0) { 8320 status = cmdStatus; 8321 } 8322 if (status == 0 && 8323 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8324 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8325 sp<ThreadBase> thread = mThread.promote(); 8326 if (thread != 0) { 8327 audio_stream_t *stream = thread->stream(); 8328 if (stream != NULL) { 8329 stream->remove_audio_effect(stream, mEffectInterface); 8330 } 8331 } 8332 } 8333 return status; 8334} 8335 8336status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8337 uint32_t cmdSize, 8338 void *pCmdData, 8339 uint32_t *replySize, 8340 void *pReplyData) 8341{ 8342 Mutex::Autolock _l(mLock); 8343// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8344 8345 if (mState == DESTROYED || mEffectInterface == NULL) { 8346 return NO_INIT; 8347 } 8348 status_t status = (*mEffectInterface)->command(mEffectInterface, 8349 cmdCode, 8350 cmdSize, 8351 pCmdData, 8352 replySize, 8353 pReplyData); 8354 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8355 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8356 for (size_t i = 1; i < mHandles.size(); i++) { 8357 sp<EffectHandle> h = mHandles[i].promote(); 8358 if (h != 0) { 8359 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8360 } 8361 } 8362 } 8363 return status; 8364} 8365 8366status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8367{ 8368 8369 Mutex::Autolock _l(mLock); 8370 ALOGV("setEnabled %p enabled %d", this, enabled); 8371 8372 if (enabled != isEnabled()) { 8373 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8374 if (enabled && status != NO_ERROR) { 8375 return status; 8376 } 8377 8378 switch (mState) { 8379 // going from disabled to enabled 8380 case IDLE: 8381 mState = STARTING; 8382 break; 8383 case STOPPED: 8384 mState = RESTART; 8385 break; 8386 case STOPPING: 8387 mState = ACTIVE; 8388 break; 8389 8390 // going from enabled to disabled 8391 case RESTART: 8392 mState = STOPPED; 8393 break; 8394 case STARTING: 8395 mState = IDLE; 8396 break; 8397 case ACTIVE: 8398 mState = STOPPING; 8399 break; 8400 case DESTROYED: 8401 return NO_ERROR; // simply ignore as we are being destroyed 8402 } 8403 for (size_t i = 1; i < mHandles.size(); i++) { 8404 sp<EffectHandle> h = mHandles[i].promote(); 8405 if (h != 0) { 8406 h->setEnabled(enabled); 8407 } 8408 } 8409 } 8410 return NO_ERROR; 8411} 8412 8413bool AudioFlinger::EffectModule::isEnabled() const 8414{ 8415 switch (mState) { 8416 case RESTART: 8417 case STARTING: 8418 case ACTIVE: 8419 return true; 8420 case IDLE: 8421 case STOPPING: 8422 case STOPPED: 8423 case DESTROYED: 8424 default: 8425 return false; 8426 } 8427} 8428 8429bool AudioFlinger::EffectModule::isProcessEnabled() const 8430{ 8431 switch (mState) { 8432 case RESTART: 8433 case ACTIVE: 8434 case STOPPING: 8435 case STOPPED: 8436 return true; 8437 case IDLE: 8438 case STARTING: 8439 case DESTROYED: 8440 default: 8441 return false; 8442 } 8443} 8444 8445status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8446{ 8447 Mutex::Autolock _l(mLock); 8448 status_t status = NO_ERROR; 8449 8450 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8451 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8452 if (isProcessEnabled() && 8453 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8454 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8455 status_t cmdStatus; 8456 uint32_t volume[2]; 8457 uint32_t *pVolume = NULL; 8458 uint32_t size = sizeof(volume); 8459 volume[0] = *left; 8460 volume[1] = *right; 8461 if (controller) { 8462 pVolume = volume; 8463 } 8464 status = (*mEffectInterface)->command(mEffectInterface, 8465 EFFECT_CMD_SET_VOLUME, 8466 size, 8467 volume, 8468 &size, 8469 pVolume); 8470 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8471 *left = volume[0]; 8472 *right = volume[1]; 8473 } 8474 } 8475 return status; 8476} 8477 8478status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8479{ 8480 Mutex::Autolock _l(mLock); 8481 status_t status = NO_ERROR; 8482 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8483 // audio pre processing modules on RecordThread can receive both output and 8484 // input device indication in the same call 8485 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8486 if (dev) { 8487 status_t cmdStatus; 8488 uint32_t size = sizeof(status_t); 8489 8490 status = (*mEffectInterface)->command(mEffectInterface, 8491 EFFECT_CMD_SET_DEVICE, 8492 sizeof(uint32_t), 8493 &dev, 8494 &size, 8495 &cmdStatus); 8496 if (status == NO_ERROR) { 8497 status = cmdStatus; 8498 } 8499 } 8500 dev = device & AUDIO_DEVICE_IN_ALL; 8501 if (dev) { 8502 status_t cmdStatus; 8503 uint32_t size = sizeof(status_t); 8504 8505 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8506 EFFECT_CMD_SET_INPUT_DEVICE, 8507 sizeof(uint32_t), 8508 &dev, 8509 &size, 8510 &cmdStatus); 8511 if (status2 == NO_ERROR) { 8512 status2 = cmdStatus; 8513 } 8514 if (status == NO_ERROR) { 8515 status = status2; 8516 } 8517 } 8518 } 8519 return status; 8520} 8521 8522status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8523{ 8524 Mutex::Autolock _l(mLock); 8525 status_t status = NO_ERROR; 8526 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8527 status_t cmdStatus; 8528 uint32_t size = sizeof(status_t); 8529 status = (*mEffectInterface)->command(mEffectInterface, 8530 EFFECT_CMD_SET_AUDIO_MODE, 8531 sizeof(audio_mode_t), 8532 &mode, 8533 &size, 8534 &cmdStatus); 8535 if (status == NO_ERROR) { 8536 status = cmdStatus; 8537 } 8538 } 8539 return status; 8540} 8541 8542void AudioFlinger::EffectModule::setSuspended(bool suspended) 8543{ 8544 Mutex::Autolock _l(mLock); 8545 mSuspended = suspended; 8546} 8547 8548bool AudioFlinger::EffectModule::suspended() const 8549{ 8550 Mutex::Autolock _l(mLock); 8551 return mSuspended; 8552} 8553 8554status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8555{ 8556 const size_t SIZE = 256; 8557 char buffer[SIZE]; 8558 String8 result; 8559 8560 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8561 result.append(buffer); 8562 8563 bool locked = tryLock(mLock); 8564 // failed to lock - AudioFlinger is probably deadlocked 8565 if (!locked) { 8566 result.append("\t\tCould not lock Fx mutex:\n"); 8567 } 8568 8569 result.append("\t\tSession Status State Engine:\n"); 8570 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8571 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8572 result.append(buffer); 8573 8574 result.append("\t\tDescriptor:\n"); 8575 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8576 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8577 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8578 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8579 result.append(buffer); 8580 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8581 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8582 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8583 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8584 result.append(buffer); 8585 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8586 mDescriptor.apiVersion, 8587 mDescriptor.flags); 8588 result.append(buffer); 8589 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8590 mDescriptor.name); 8591 result.append(buffer); 8592 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8593 mDescriptor.implementor); 8594 result.append(buffer); 8595 8596 result.append("\t\t- Input configuration:\n"); 8597 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8598 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8599 (uint32_t)mConfig.inputCfg.buffer.raw, 8600 mConfig.inputCfg.buffer.frameCount, 8601 mConfig.inputCfg.samplingRate, 8602 mConfig.inputCfg.channels, 8603 mConfig.inputCfg.format); 8604 result.append(buffer); 8605 8606 result.append("\t\t- Output configuration:\n"); 8607 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8608 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8609 (uint32_t)mConfig.outputCfg.buffer.raw, 8610 mConfig.outputCfg.buffer.frameCount, 8611 mConfig.outputCfg.samplingRate, 8612 mConfig.outputCfg.channels, 8613 mConfig.outputCfg.format); 8614 result.append(buffer); 8615 8616 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8617 result.append(buffer); 8618 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8619 for (size_t i = 0; i < mHandles.size(); ++i) { 8620 sp<EffectHandle> handle = mHandles[i].promote(); 8621 if (handle != 0) { 8622 handle->dump(buffer, SIZE); 8623 result.append(buffer); 8624 } 8625 } 8626 8627 result.append("\n"); 8628 8629 write(fd, result.string(), result.length()); 8630 8631 if (locked) { 8632 mLock.unlock(); 8633 } 8634 8635 return NO_ERROR; 8636} 8637 8638// ---------------------------------------------------------------------------- 8639// EffectHandle implementation 8640// ---------------------------------------------------------------------------- 8641 8642#undef LOG_TAG 8643#define LOG_TAG "AudioFlinger::EffectHandle" 8644 8645AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8646 const sp<AudioFlinger::Client>& client, 8647 const sp<IEffectClient>& effectClient, 8648 int32_t priority) 8649 : BnEffect(), 8650 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8651 mPriority(priority), mHasControl(false), mEnabled(false) 8652{ 8653 ALOGV("constructor %p", this); 8654 8655 if (client == 0) { 8656 return; 8657 } 8658 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8659 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8660 if (mCblkMemory != 0) { 8661 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8662 8663 if (mCblk != NULL) { 8664 new(mCblk) effect_param_cblk_t(); 8665 mBuffer = (uint8_t *)mCblk + bufOffset; 8666 } 8667 } else { 8668 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8669 return; 8670 } 8671} 8672 8673AudioFlinger::EffectHandle::~EffectHandle() 8674{ 8675 ALOGV("Destructor %p", this); 8676 disconnect(false); 8677 ALOGV("Destructor DONE %p", this); 8678} 8679 8680status_t AudioFlinger::EffectHandle::enable() 8681{ 8682 ALOGV("enable %p", this); 8683 if (!mHasControl) return INVALID_OPERATION; 8684 if (mEffect == 0) return DEAD_OBJECT; 8685 8686 if (mEnabled) { 8687 return NO_ERROR; 8688 } 8689 8690 mEnabled = true; 8691 8692 sp<ThreadBase> thread = mEffect->thread().promote(); 8693 if (thread != 0) { 8694 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8695 } 8696 8697 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8698 if (mEffect->suspended()) { 8699 return NO_ERROR; 8700 } 8701 8702 status_t status = mEffect->setEnabled(true); 8703 if (status != NO_ERROR) { 8704 if (thread != 0) { 8705 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8706 } 8707 mEnabled = false; 8708 } 8709 return status; 8710} 8711 8712status_t AudioFlinger::EffectHandle::disable() 8713{ 8714 ALOGV("disable %p", this); 8715 if (!mHasControl) return INVALID_OPERATION; 8716 if (mEffect == 0) return DEAD_OBJECT; 8717 8718 if (!mEnabled) { 8719 return NO_ERROR; 8720 } 8721 mEnabled = false; 8722 8723 if (mEffect->suspended()) { 8724 return NO_ERROR; 8725 } 8726 8727 status_t status = mEffect->setEnabled(false); 8728 8729 sp<ThreadBase> thread = mEffect->thread().promote(); 8730 if (thread != 0) { 8731 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8732 } 8733 8734 return status; 8735} 8736 8737void AudioFlinger::EffectHandle::disconnect() 8738{ 8739 disconnect(true); 8740} 8741 8742void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8743{ 8744 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8745 if (mEffect == 0) { 8746 return; 8747 } 8748 mEffect->disconnect(this, unpinIfLast); 8749 8750 if (mHasControl && mEnabled) { 8751 sp<ThreadBase> thread = mEffect->thread().promote(); 8752 if (thread != 0) { 8753 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8754 } 8755 } 8756 8757 // release sp on module => module destructor can be called now 8758 mEffect.clear(); 8759 if (mClient != 0) { 8760 if (mCblk != NULL) { 8761 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8762 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8763 } 8764 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8765 // Client destructor must run with AudioFlinger mutex locked 8766 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8767 mClient.clear(); 8768 } 8769} 8770 8771status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8772 uint32_t cmdSize, 8773 void *pCmdData, 8774 uint32_t *replySize, 8775 void *pReplyData) 8776{ 8777// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8778// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8779 8780 // only get parameter command is permitted for applications not controlling the effect 8781 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8782 return INVALID_OPERATION; 8783 } 8784 if (mEffect == 0) return DEAD_OBJECT; 8785 if (mClient == 0) return INVALID_OPERATION; 8786 8787 // handle commands that are not forwarded transparently to effect engine 8788 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8789 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8790 // no risk to block the whole media server process or mixer threads is we are stuck here 8791 Mutex::Autolock _l(mCblk->lock); 8792 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8793 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8794 mCblk->serverIndex = 0; 8795 mCblk->clientIndex = 0; 8796 return BAD_VALUE; 8797 } 8798 status_t status = NO_ERROR; 8799 while (mCblk->serverIndex < mCblk->clientIndex) { 8800 int reply; 8801 uint32_t rsize = sizeof(int); 8802 int *p = (int *)(mBuffer + mCblk->serverIndex); 8803 int size = *p++; 8804 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8805 ALOGW("command(): invalid parameter block size"); 8806 break; 8807 } 8808 effect_param_t *param = (effect_param_t *)p; 8809 if (param->psize == 0 || param->vsize == 0) { 8810 ALOGW("command(): null parameter or value size"); 8811 mCblk->serverIndex += size; 8812 continue; 8813 } 8814 uint32_t psize = sizeof(effect_param_t) + 8815 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8816 param->vsize; 8817 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8818 psize, 8819 p, 8820 &rsize, 8821 &reply); 8822 // stop at first error encountered 8823 if (ret != NO_ERROR) { 8824 status = ret; 8825 *(int *)pReplyData = reply; 8826 break; 8827 } else if (reply != NO_ERROR) { 8828 *(int *)pReplyData = reply; 8829 break; 8830 } 8831 mCblk->serverIndex += size; 8832 } 8833 mCblk->serverIndex = 0; 8834 mCblk->clientIndex = 0; 8835 return status; 8836 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8837 *(int *)pReplyData = NO_ERROR; 8838 return enable(); 8839 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8840 *(int *)pReplyData = NO_ERROR; 8841 return disable(); 8842 } 8843 8844 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8845} 8846 8847void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8848{ 8849 ALOGV("setControl %p control %d", this, hasControl); 8850 8851 mHasControl = hasControl; 8852 mEnabled = enabled; 8853 8854 if (signal && mEffectClient != 0) { 8855 mEffectClient->controlStatusChanged(hasControl); 8856 } 8857} 8858 8859void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8860 uint32_t cmdSize, 8861 void *pCmdData, 8862 uint32_t replySize, 8863 void *pReplyData) 8864{ 8865 if (mEffectClient != 0) { 8866 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8867 } 8868} 8869 8870 8871 8872void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8873{ 8874 if (mEffectClient != 0) { 8875 mEffectClient->enableStatusChanged(enabled); 8876 } 8877} 8878 8879status_t AudioFlinger::EffectHandle::onTransact( 8880 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8881{ 8882 return BnEffect::onTransact(code, data, reply, flags); 8883} 8884 8885 8886void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8887{ 8888 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8889 8890 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8891 (mClient == 0) ? getpid_cached : mClient->pid(), 8892 mPriority, 8893 mHasControl, 8894 !locked, 8895 mCblk ? mCblk->clientIndex : 0, 8896 mCblk ? mCblk->serverIndex : 0 8897 ); 8898 8899 if (locked) { 8900 mCblk->lock.unlock(); 8901 } 8902} 8903 8904#undef LOG_TAG 8905#define LOG_TAG "AudioFlinger::EffectChain" 8906 8907AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8908 int sessionId) 8909 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8910 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8911 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8912{ 8913 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8914 if (thread == NULL) { 8915 return; 8916 } 8917 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8918 thread->frameCount(); 8919} 8920 8921AudioFlinger::EffectChain::~EffectChain() 8922{ 8923 if (mOwnInBuffer) { 8924 delete mInBuffer; 8925 } 8926 8927} 8928 8929// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8930sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8931{ 8932 size_t size = mEffects.size(); 8933 8934 for (size_t i = 0; i < size; i++) { 8935 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8936 return mEffects[i]; 8937 } 8938 } 8939 return 0; 8940} 8941 8942// getEffectFromId_l() must be called with ThreadBase::mLock held 8943sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8944{ 8945 size_t size = mEffects.size(); 8946 8947 for (size_t i = 0; i < size; i++) { 8948 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8949 if (id == 0 || mEffects[i]->id() == id) { 8950 return mEffects[i]; 8951 } 8952 } 8953 return 0; 8954} 8955 8956// getEffectFromType_l() must be called with ThreadBase::mLock held 8957sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8958 const effect_uuid_t *type) 8959{ 8960 size_t size = mEffects.size(); 8961 8962 for (size_t i = 0; i < size; i++) { 8963 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8964 return mEffects[i]; 8965 } 8966 } 8967 return 0; 8968} 8969 8970// Must be called with EffectChain::mLock locked 8971void AudioFlinger::EffectChain::process_l() 8972{ 8973 sp<ThreadBase> thread = mThread.promote(); 8974 if (thread == 0) { 8975 ALOGW("process_l(): cannot promote mixer thread"); 8976 return; 8977 } 8978 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8979 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8980 // always process effects unless no more tracks are on the session and the effect tail 8981 // has been rendered 8982 bool doProcess = true; 8983 if (!isGlobalSession) { 8984 bool tracksOnSession = (trackCnt() != 0); 8985 8986 if (!tracksOnSession && mTailBufferCount == 0) { 8987 doProcess = false; 8988 } 8989 8990 if (activeTrackCnt() == 0) { 8991 // if no track is active and the effect tail has not been rendered, 8992 // the input buffer must be cleared here as the mixer process will not do it 8993 if (tracksOnSession || mTailBufferCount > 0) { 8994 size_t numSamples = thread->frameCount() * thread->channelCount(); 8995 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8996 if (mTailBufferCount > 0) { 8997 mTailBufferCount--; 8998 } 8999 } 9000 } 9001 } 9002 9003 size_t size = mEffects.size(); 9004 if (doProcess) { 9005 for (size_t i = 0; i < size; i++) { 9006 mEffects[i]->process(); 9007 } 9008 } 9009 for (size_t i = 0; i < size; i++) { 9010 mEffects[i]->updateState(); 9011 } 9012} 9013 9014// addEffect_l() must be called with PlaybackThread::mLock held 9015status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9016{ 9017 effect_descriptor_t desc = effect->desc(); 9018 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9019 9020 Mutex::Autolock _l(mLock); 9021 effect->setChain(this); 9022 sp<ThreadBase> thread = mThread.promote(); 9023 if (thread == 0) { 9024 return NO_INIT; 9025 } 9026 effect->setThread(thread); 9027 9028 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9029 // Auxiliary effects are inserted at the beginning of mEffects vector as 9030 // they are processed first and accumulated in chain input buffer 9031 mEffects.insertAt(effect, 0); 9032 9033 // the input buffer for auxiliary effect contains mono samples in 9034 // 32 bit format. This is to avoid saturation in AudoMixer 9035 // accumulation stage. Saturation is done in EffectModule::process() before 9036 // calling the process in effect engine 9037 size_t numSamples = thread->frameCount(); 9038 int32_t *buffer = new int32_t[numSamples]; 9039 memset(buffer, 0, numSamples * sizeof(int32_t)); 9040 effect->setInBuffer((int16_t *)buffer); 9041 // auxiliary effects output samples to chain input buffer for further processing 9042 // by insert effects 9043 effect->setOutBuffer(mInBuffer); 9044 } else { 9045 // Insert effects are inserted at the end of mEffects vector as they are processed 9046 // after track and auxiliary effects. 9047 // Insert effect order as a function of indicated preference: 9048 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9049 // another effect is present 9050 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9051 // last effect claiming first position 9052 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9053 // first effect claiming last position 9054 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9055 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9056 // already present 9057 9058 size_t size = mEffects.size(); 9059 size_t idx_insert = size; 9060 ssize_t idx_insert_first = -1; 9061 ssize_t idx_insert_last = -1; 9062 9063 for (size_t i = 0; i < size; i++) { 9064 effect_descriptor_t d = mEffects[i]->desc(); 9065 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9066 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9067 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9068 // check invalid effect chaining combinations 9069 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9070 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9071 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9072 return INVALID_OPERATION; 9073 } 9074 // remember position of first insert effect and by default 9075 // select this as insert position for new effect 9076 if (idx_insert == size) { 9077 idx_insert = i; 9078 } 9079 // remember position of last insert effect claiming 9080 // first position 9081 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9082 idx_insert_first = i; 9083 } 9084 // remember position of first insert effect claiming 9085 // last position 9086 if (iPref == EFFECT_FLAG_INSERT_LAST && 9087 idx_insert_last == -1) { 9088 idx_insert_last = i; 9089 } 9090 } 9091 } 9092 9093 // modify idx_insert from first position if needed 9094 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9095 if (idx_insert_last != -1) { 9096 idx_insert = idx_insert_last; 9097 } else { 9098 idx_insert = size; 9099 } 9100 } else { 9101 if (idx_insert_first != -1) { 9102 idx_insert = idx_insert_first + 1; 9103 } 9104 } 9105 9106 // always read samples from chain input buffer 9107 effect->setInBuffer(mInBuffer); 9108 9109 // if last effect in the chain, output samples to chain 9110 // output buffer, otherwise to chain input buffer 9111 if (idx_insert == size) { 9112 if (idx_insert != 0) { 9113 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9114 mEffects[idx_insert-1]->configure(); 9115 } 9116 effect->setOutBuffer(mOutBuffer); 9117 } else { 9118 effect->setOutBuffer(mInBuffer); 9119 } 9120 mEffects.insertAt(effect, idx_insert); 9121 9122 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9123 } 9124 effect->configure(); 9125 return NO_ERROR; 9126} 9127 9128// removeEffect_l() must be called with PlaybackThread::mLock held 9129size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9130{ 9131 Mutex::Autolock _l(mLock); 9132 size_t size = mEffects.size(); 9133 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9134 9135 for (size_t i = 0; i < size; i++) { 9136 if (effect == mEffects[i]) { 9137 // calling stop here will remove pre-processing effect from the audio HAL. 9138 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9139 // the middle of a read from audio HAL 9140 if (mEffects[i]->state() == EffectModule::ACTIVE || 9141 mEffects[i]->state() == EffectModule::STOPPING) { 9142 mEffects[i]->stop(); 9143 } 9144 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9145 delete[] effect->inBuffer(); 9146 } else { 9147 if (i == size - 1 && i != 0) { 9148 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9149 mEffects[i - 1]->configure(); 9150 } 9151 } 9152 mEffects.removeAt(i); 9153 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9154 break; 9155 } 9156 } 9157 9158 return mEffects.size(); 9159} 9160 9161// setDevice_l() must be called with PlaybackThread::mLock held 9162void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9163{ 9164 size_t size = mEffects.size(); 9165 for (size_t i = 0; i < size; i++) { 9166 mEffects[i]->setDevice(device); 9167 } 9168} 9169 9170// setMode_l() must be called with PlaybackThread::mLock held 9171void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9172{ 9173 size_t size = mEffects.size(); 9174 for (size_t i = 0; i < size; i++) { 9175 mEffects[i]->setMode(mode); 9176 } 9177} 9178 9179// setVolume_l() must be called with PlaybackThread::mLock held 9180bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9181{ 9182 uint32_t newLeft = *left; 9183 uint32_t newRight = *right; 9184 bool hasControl = false; 9185 int ctrlIdx = -1; 9186 size_t size = mEffects.size(); 9187 9188 // first update volume controller 9189 for (size_t i = size; i > 0; i--) { 9190 if (mEffects[i - 1]->isProcessEnabled() && 9191 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9192 ctrlIdx = i - 1; 9193 hasControl = true; 9194 break; 9195 } 9196 } 9197 9198 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9199 if (hasControl) { 9200 *left = mNewLeftVolume; 9201 *right = mNewRightVolume; 9202 } 9203 return hasControl; 9204 } 9205 9206 mVolumeCtrlIdx = ctrlIdx; 9207 mLeftVolume = newLeft; 9208 mRightVolume = newRight; 9209 9210 // second get volume update from volume controller 9211 if (ctrlIdx >= 0) { 9212 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9213 mNewLeftVolume = newLeft; 9214 mNewRightVolume = newRight; 9215 } 9216 // then indicate volume to all other effects in chain. 9217 // Pass altered volume to effects before volume controller 9218 // and requested volume to effects after controller 9219 uint32_t lVol = newLeft; 9220 uint32_t rVol = newRight; 9221 9222 for (size_t i = 0; i < size; i++) { 9223 if ((int)i == ctrlIdx) continue; 9224 // this also works for ctrlIdx == -1 when there is no volume controller 9225 if ((int)i > ctrlIdx) { 9226 lVol = *left; 9227 rVol = *right; 9228 } 9229 mEffects[i]->setVolume(&lVol, &rVol, false); 9230 } 9231 *left = newLeft; 9232 *right = newRight; 9233 9234 return hasControl; 9235} 9236 9237status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9238{ 9239 const size_t SIZE = 256; 9240 char buffer[SIZE]; 9241 String8 result; 9242 9243 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9244 result.append(buffer); 9245 9246 bool locked = tryLock(mLock); 9247 // failed to lock - AudioFlinger is probably deadlocked 9248 if (!locked) { 9249 result.append("\tCould not lock mutex:\n"); 9250 } 9251 9252 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9253 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9254 mEffects.size(), 9255 (uint32_t)mInBuffer, 9256 (uint32_t)mOutBuffer, 9257 mActiveTrackCnt); 9258 result.append(buffer); 9259 write(fd, result.string(), result.size()); 9260 9261 for (size_t i = 0; i < mEffects.size(); ++i) { 9262 sp<EffectModule> effect = mEffects[i]; 9263 if (effect != 0) { 9264 effect->dump(fd, args); 9265 } 9266 } 9267 9268 if (locked) { 9269 mLock.unlock(); 9270 } 9271 9272 return NO_ERROR; 9273} 9274 9275// must be called with ThreadBase::mLock held 9276void AudioFlinger::EffectChain::setEffectSuspended_l( 9277 const effect_uuid_t *type, bool suspend) 9278{ 9279 sp<SuspendedEffectDesc> desc; 9280 // use effect type UUID timelow as key as there is no real risk of identical 9281 // timeLow fields among effect type UUIDs. 9282 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9283 if (suspend) { 9284 if (index >= 0) { 9285 desc = mSuspendedEffects.valueAt(index); 9286 } else { 9287 desc = new SuspendedEffectDesc(); 9288 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9289 mSuspendedEffects.add(type->timeLow, desc); 9290 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9291 } 9292 if (desc->mRefCount++ == 0) { 9293 sp<EffectModule> effect = getEffectIfEnabled(type); 9294 if (effect != 0) { 9295 desc->mEffect = effect; 9296 effect->setSuspended(true); 9297 effect->setEnabled(false); 9298 } 9299 } 9300 } else { 9301 if (index < 0) { 9302 return; 9303 } 9304 desc = mSuspendedEffects.valueAt(index); 9305 if (desc->mRefCount <= 0) { 9306 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9307 desc->mRefCount = 1; 9308 } 9309 if (--desc->mRefCount == 0) { 9310 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9311 if (desc->mEffect != 0) { 9312 sp<EffectModule> effect = desc->mEffect.promote(); 9313 if (effect != 0) { 9314 effect->setSuspended(false); 9315 sp<EffectHandle> handle = effect->controlHandle(); 9316 if (handle != 0) { 9317 effect->setEnabled(handle->enabled()); 9318 } 9319 } 9320 desc->mEffect.clear(); 9321 } 9322 mSuspendedEffects.removeItemsAt(index); 9323 } 9324 } 9325} 9326 9327// must be called with ThreadBase::mLock held 9328void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9329{ 9330 sp<SuspendedEffectDesc> desc; 9331 9332 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9333 if (suspend) { 9334 if (index >= 0) { 9335 desc = mSuspendedEffects.valueAt(index); 9336 } else { 9337 desc = new SuspendedEffectDesc(); 9338 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9339 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9340 } 9341 if (desc->mRefCount++ == 0) { 9342 Vector< sp<EffectModule> > effects; 9343 getSuspendEligibleEffects(effects); 9344 for (size_t i = 0; i < effects.size(); i++) { 9345 setEffectSuspended_l(&effects[i]->desc().type, true); 9346 } 9347 } 9348 } else { 9349 if (index < 0) { 9350 return; 9351 } 9352 desc = mSuspendedEffects.valueAt(index); 9353 if (desc->mRefCount <= 0) { 9354 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9355 desc->mRefCount = 1; 9356 } 9357 if (--desc->mRefCount == 0) { 9358 Vector<const effect_uuid_t *> types; 9359 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9360 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9361 continue; 9362 } 9363 types.add(&mSuspendedEffects.valueAt(i)->mType); 9364 } 9365 for (size_t i = 0; i < types.size(); i++) { 9366 setEffectSuspended_l(types[i], false); 9367 } 9368 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9369 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9370 } 9371 } 9372} 9373 9374 9375// The volume effect is used for automated tests only 9376#ifndef OPENSL_ES_H_ 9377static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9378 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9379const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9380#endif //OPENSL_ES_H_ 9381 9382bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9383{ 9384 // auxiliary effects and visualizer are never suspended on output mix 9385 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9386 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9387 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9388 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9389 return false; 9390 } 9391 return true; 9392} 9393 9394void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9395{ 9396 effects.clear(); 9397 for (size_t i = 0; i < mEffects.size(); i++) { 9398 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9399 effects.add(mEffects[i]); 9400 } 9401 } 9402} 9403 9404sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9405 const effect_uuid_t *type) 9406{ 9407 sp<EffectModule> effect = getEffectFromType_l(type); 9408 return effect != 0 && effect->isEnabled() ? effect : 0; 9409} 9410 9411void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9412 bool enabled) 9413{ 9414 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9415 if (enabled) { 9416 if (index < 0) { 9417 // if the effect is not suspend check if all effects are suspended 9418 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9419 if (index < 0) { 9420 return; 9421 } 9422 if (!isEffectEligibleForSuspend(effect->desc())) { 9423 return; 9424 } 9425 setEffectSuspended_l(&effect->desc().type, enabled); 9426 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9427 if (index < 0) { 9428 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9429 return; 9430 } 9431 } 9432 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9433 effect->desc().type.timeLow); 9434 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9435 // if effect is requested to suspended but was not yet enabled, supend it now. 9436 if (desc->mEffect == 0) { 9437 desc->mEffect = effect; 9438 effect->setEnabled(false); 9439 effect->setSuspended(true); 9440 } 9441 } else { 9442 if (index < 0) { 9443 return; 9444 } 9445 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9446 effect->desc().type.timeLow); 9447 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9448 desc->mEffect.clear(); 9449 effect->setSuspended(false); 9450 } 9451} 9452 9453#undef LOG_TAG 9454#define LOG_TAG "AudioFlinger" 9455 9456// ---------------------------------------------------------------------------- 9457 9458status_t AudioFlinger::onTransact( 9459 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9460{ 9461 return BnAudioFlinger::onTransact(code, data, reply, flags); 9462} 9463 9464}; // namespace android 9465