AudioFlinger.cpp revision 2986460984580833161bdaabc7f17da1005a8961
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22//#define ATRACE_TAG ATRACE_TAG_AUDIO 23 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <binder/IPCThreadState.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41#include <cutils/compiler.h> 42 43#undef ADD_BATTERY_DATA 44 45#ifdef ADD_BATTERY_DATA 46#include <media/IMediaPlayerService.h> 47#include <media/IMediaDeathNotifier.h> 48#endif 49 50#include <private/media/AudioTrackShared.h> 51#include <private/media/AudioEffectShared.h> 52 53#include <system/audio.h> 54#include <hardware/audio.h> 55 56#include "AudioMixer.h" 57#include "AudioFlinger.h" 58#include "ServiceUtilities.h" 59 60#include <media/EffectsFactoryApi.h> 61#include <audio_effects/effect_visualizer.h> 62#include <audio_effects/effect_ns.h> 63#include <audio_effects/effect_aec.h> 64 65#include <audio_utils/primitives.h> 66 67#include <powermanager/PowerManager.h> 68 69// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 70#ifdef DEBUG_CPU_USAGE 71#include <cpustats/CentralTendencyStatistics.h> 72#include <cpustats/ThreadCpuUsage.h> 73#endif 74 75#include <common_time/cc_helper.h> 76#include <common_time/local_clock.h> 77 78#include "FastMixer.h" 79 80// NBAIO implementations 81#include "AudioStreamOutSink.h" 82#include "MonoPipe.h" 83#include "MonoPipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167// ---------------------------------------------------------------------------- 168 169#ifdef ADD_BATTERY_DATA 170// To collect the amplifier usage 171static void addBatteryData(uint32_t params) { 172 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 173 if (service == NULL) { 174 // it already logged 175 return; 176 } 177 178 service->addBatteryData(params); 179} 180#endif 181 182static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183{ 184 const hw_module_t *mod; 185 int rc; 186 187 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190 if (rc) { 191 goto out; 192 } 193 rc = audio_hw_device_open(mod, dev); 194 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 195 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196 if (rc) { 197 goto out; 198 } 199 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201 rc = BAD_VALUE; 202 goto out; 203 } 204 return 0; 205 206out: 207 *dev = NULL; 208 return rc; 209} 210 211// ---------------------------------------------------------------------------- 212 213AudioFlinger::AudioFlinger() 214 : BnAudioFlinger(), 215 mPrimaryHardwareDev(NULL), 216 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217 mMasterVolume(1.0f), 218 mMasterVolumeSupportLvl(MVS_NONE), 219 mMasterMute(false), 220 mNextUniqueId(1), 221 mMode(AUDIO_MODE_INVALID), 222 mBtNrecIsOff(false) 223{ 224} 225 226void AudioFlinger::onFirstRef() 227{ 228 int rc = 0; 229 230 Mutex::Autolock _l(mLock); 231 232 /* TODO: move all this work into an Init() function */ 233 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 235 uint32_t int_val; 236 if (1 == sscanf(val_str, "%u", &int_val)) { 237 mStandbyTimeInNsecs = milliseconds(int_val); 238 ALOGI("Using %u mSec as standby time.", int_val); 239 } else { 240 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241 ALOGI("Using default %u mSec as standby time.", 242 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243 } 244 } 245 246 mMode = AUDIO_MODE_NORMAL; 247 mMasterVolumeSW = 1.0; 248 mMasterVolume = 1.0; 249 mHardwareStatus = AUDIO_HW_IDLE; 250} 251 252AudioFlinger::~AudioFlinger() 253{ 254 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 478 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 479 if (mPlaybackThreads.keyAt(i) != output) { 480 // prevent same audio session on different output threads 481 uint32_t sessions = t->hasAudioSession(*sessionId); 482 if (sessions & PlaybackThread::TRACK_SESSION) { 483 ALOGE("createTrack() session ID %d already in use", *sessionId); 484 lStatus = BAD_VALUE; 485 goto Exit; 486 } 487 // check if an effect with same session ID is waiting for a track to be created 488 if (sessions & PlaybackThread::EFFECT_SESSION) { 489 effectThread = t.get(); 490 } 491 } 492 } 493 lSessionId = *sessionId; 494 } else { 495 // if no audio session id is provided, create one here 496 lSessionId = nextUniqueId(); 497 if (sessionId != NULL) { 498 *sessionId = lSessionId; 499 } 500 } 501 ALOGV("createTrack() lSessionId: %d", lSessionId); 502 503 track = thread->createTrack_l(client, streamType, sampleRate, format, 504 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 505 506 // move effect chain to this output thread if an effect on same session was waiting 507 // for a track to be created 508 if (lStatus == NO_ERROR && effectThread != NULL) { 509 Mutex::Autolock _dl(thread->mLock); 510 Mutex::Autolock _sl(effectThread->mLock); 511 moveEffectChain_l(lSessionId, effectThread, thread, true); 512 } 513 514 // Look for sync events awaiting for a session to be used. 515 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 516 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 517 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 518 if (lStatus == NO_ERROR) { 519 track->setSyncEvent(mPendingSyncEvents[i]); 520 } else { 521 mPendingSyncEvents[i]->cancel(); 522 } 523 mPendingSyncEvents.removeAt(i); 524 i--; 525 } 526 } 527 } 528 } 529 if (lStatus == NO_ERROR) { 530 trackHandle = new TrackHandle(track); 531 } else { 532 // remove local strong reference to Client before deleting the Track so that the Client 533 // destructor is called by the TrackBase destructor with mLock held 534 client.clear(); 535 track.clear(); 536 } 537 538Exit: 539 if (status != NULL) { 540 *status = lStatus; 541 } 542 return trackHandle; 543} 544 545uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 546{ 547 Mutex::Autolock _l(mLock); 548 PlaybackThread *thread = checkPlaybackThread_l(output); 549 if (thread == NULL) { 550 ALOGW("sampleRate() unknown thread %d", output); 551 return 0; 552 } 553 return thread->sampleRate(); 554} 555 556int AudioFlinger::channelCount(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("channelCount() unknown thread %d", output); 562 return 0; 563 } 564 return thread->channelCount(); 565} 566 567audio_format_t AudioFlinger::format(audio_io_handle_t output) const 568{ 569 Mutex::Autolock _l(mLock); 570 PlaybackThread *thread = checkPlaybackThread_l(output); 571 if (thread == NULL) { 572 ALOGW("format() unknown thread %d", output); 573 return AUDIO_FORMAT_INVALID; 574 } 575 return thread->format(); 576} 577 578size_t AudioFlinger::frameCount(audio_io_handle_t output) const 579{ 580 Mutex::Autolock _l(mLock); 581 PlaybackThread *thread = checkPlaybackThread_l(output); 582 if (thread == NULL) { 583 ALOGW("frameCount() unknown thread %d", output); 584 return 0; 585 } 586 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 587 // should examine all callers and fix them to handle smaller counts 588 return thread->frameCount(); 589} 590 591uint32_t AudioFlinger::latency(audio_io_handle_t output) const 592{ 593 Mutex::Autolock _l(mLock); 594 PlaybackThread *thread = checkPlaybackThread_l(output); 595 if (thread == NULL) { 596 ALOGW("latency() unknown thread %d", output); 597 return 0; 598 } 599 return thread->latency(); 600} 601 602status_t AudioFlinger::setMasterVolume(float value) 603{ 604 status_t ret = initCheck(); 605 if (ret != NO_ERROR) { 606 return ret; 607 } 608 609 // check calling permissions 610 if (!settingsAllowed()) { 611 return PERMISSION_DENIED; 612 } 613 614 float swmv = value; 615 616 Mutex::Autolock _l(mLock); 617 618 // when hw supports master volume, don't scale in sw mixer 619 if (MVS_NONE != mMasterVolumeSupportLvl) { 620 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 621 AutoMutex lock(mHardwareLock); 622 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 623 624 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 625 if (NULL != dev->set_master_volume) { 626 dev->set_master_volume(dev, value); 627 } 628 mHardwareStatus = AUDIO_HW_IDLE; 629 } 630 631 swmv = 1.0; 632 } 633 634 mMasterVolume = value; 635 mMasterVolumeSW = swmv; 636 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 637 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 638 639 return NO_ERROR; 640} 641 642status_t AudioFlinger::setMode(audio_mode_t mode) 643{ 644 status_t ret = initCheck(); 645 if (ret != NO_ERROR) { 646 return ret; 647 } 648 649 // check calling permissions 650 if (!settingsAllowed()) { 651 return PERMISSION_DENIED; 652 } 653 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 654 ALOGW("Illegal value: setMode(%d)", mode); 655 return BAD_VALUE; 656 } 657 658 { // scope for the lock 659 AutoMutex lock(mHardwareLock); 660 mHardwareStatus = AUDIO_HW_SET_MODE; 661 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 662 mHardwareStatus = AUDIO_HW_IDLE; 663 } 664 665 if (NO_ERROR == ret) { 666 Mutex::Autolock _l(mLock); 667 mMode = mode; 668 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 669 mPlaybackThreads.valueAt(i)->setMode(mode); 670 } 671 672 return ret; 673} 674 675status_t AudioFlinger::setMicMute(bool state) 676{ 677 status_t ret = initCheck(); 678 if (ret != NO_ERROR) { 679 return ret; 680 } 681 682 // check calling permissions 683 if (!settingsAllowed()) { 684 return PERMISSION_DENIED; 685 } 686 687 AutoMutex lock(mHardwareLock); 688 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 689 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return ret; 692} 693 694bool AudioFlinger::getMicMute() const 695{ 696 status_t ret = initCheck(); 697 if (ret != NO_ERROR) { 698 return false; 699 } 700 701 bool state = AUDIO_MODE_INVALID; 702 AutoMutex lock(mHardwareLock); 703 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 704 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 705 mHardwareStatus = AUDIO_HW_IDLE; 706 return state; 707} 708 709status_t AudioFlinger::setMasterMute(bool muted) 710{ 711 // check calling permissions 712 if (!settingsAllowed()) { 713 return PERMISSION_DENIED; 714 } 715 716 Mutex::Autolock _l(mLock); 717 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 718 mMasterMute = muted; 719 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 720 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 721 722 return NO_ERROR; 723} 724 725float AudioFlinger::masterVolume() const 726{ 727 Mutex::Autolock _l(mLock); 728 return masterVolume_l(); 729} 730 731float AudioFlinger::masterVolumeSW() const 732{ 733 Mutex::Autolock _l(mLock); 734 return masterVolumeSW_l(); 735} 736 737bool AudioFlinger::masterMute() const 738{ 739 Mutex::Autolock _l(mLock); 740 return masterMute_l(); 741} 742 743float AudioFlinger::masterVolume_l() const 744{ 745 if (MVS_FULL == mMasterVolumeSupportLvl) { 746 float ret_val; 747 AutoMutex lock(mHardwareLock); 748 749 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 750 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 751 (NULL != mPrimaryHardwareDev->get_master_volume), 752 "can't get master volume"); 753 754 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 755 mHardwareStatus = AUDIO_HW_IDLE; 756 return ret_val; 757 } 758 759 return mMasterVolume; 760} 761 762status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 763 audio_io_handle_t output) 764{ 765 // check calling permissions 766 if (!settingsAllowed()) { 767 return PERMISSION_DENIED; 768 } 769 770 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 771 ALOGE("setStreamVolume() invalid stream %d", stream); 772 return BAD_VALUE; 773 } 774 775 AutoMutex lock(mLock); 776 PlaybackThread *thread = NULL; 777 if (output) { 778 thread = checkPlaybackThread_l(output); 779 if (thread == NULL) { 780 return BAD_VALUE; 781 } 782 } 783 784 mStreamTypes[stream].volume = value; 785 786 if (thread == NULL) { 787 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 788 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 789 } 790 } else { 791 thread->setStreamVolume(stream, value); 792 } 793 794 return NO_ERROR; 795} 796 797status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 798{ 799 // check calling permissions 800 if (!settingsAllowed()) { 801 return PERMISSION_DENIED; 802 } 803 804 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 805 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 806 ALOGE("setStreamMute() invalid stream %d", stream); 807 return BAD_VALUE; 808 } 809 810 AutoMutex lock(mLock); 811 mStreamTypes[stream].mute = muted; 812 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 813 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 814 815 return NO_ERROR; 816} 817 818float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 819{ 820 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 821 return 0.0f; 822 } 823 824 AutoMutex lock(mLock); 825 float volume; 826 if (output) { 827 PlaybackThread *thread = checkPlaybackThread_l(output); 828 if (thread == NULL) { 829 return 0.0f; 830 } 831 volume = thread->streamVolume(stream); 832 } else { 833 volume = streamVolume_l(stream); 834 } 835 836 return volume; 837} 838 839bool AudioFlinger::streamMute(audio_stream_type_t stream) const 840{ 841 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 842 return true; 843 } 844 845 AutoMutex lock(mLock); 846 return streamMute_l(stream); 847} 848 849status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 850{ 851 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 852 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 853 // check calling permissions 854 if (!settingsAllowed()) { 855 return PERMISSION_DENIED; 856 } 857 858 // ioHandle == 0 means the parameters are global to the audio hardware interface 859 if (ioHandle == 0) { 860 Mutex::Autolock _l(mLock); 861 status_t final_result = NO_ERROR; 862 { 863 AutoMutex lock(mHardwareLock); 864 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 865 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 866 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 867 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 868 final_result = result ?: final_result; 869 } 870 mHardwareStatus = AUDIO_HW_IDLE; 871 } 872 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 873 AudioParameter param = AudioParameter(keyValuePairs); 874 String8 value; 875 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 876 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 877 if (mBtNrecIsOff != btNrecIsOff) { 878 for (size_t i = 0; i < mRecordThreads.size(); i++) { 879 sp<RecordThread> thread = mRecordThreads.valueAt(i); 880 RecordThread::RecordTrack *track = thread->track(); 881 if (track != NULL) { 882 audio_devices_t device = (audio_devices_t)( 883 thread->device() & AUDIO_DEVICE_IN_ALL); 884 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 885 thread->setEffectSuspended(FX_IID_AEC, 886 suspend, 887 track->sessionId()); 888 thread->setEffectSuspended(FX_IID_NS, 889 suspend, 890 track->sessionId()); 891 } 892 } 893 mBtNrecIsOff = btNrecIsOff; 894 } 895 } 896 return final_result; 897 } 898 899 // hold a strong ref on thread in case closeOutput() or closeInput() is called 900 // and the thread is exited once the lock is released 901 sp<ThreadBase> thread; 902 { 903 Mutex::Autolock _l(mLock); 904 thread = checkPlaybackThread_l(ioHandle); 905 if (thread == NULL) { 906 thread = checkRecordThread_l(ioHandle); 907 } else if (thread == primaryPlaybackThread_l()) { 908 // indicate output device change to all input threads for pre processing 909 AudioParameter param = AudioParameter(keyValuePairs); 910 int value; 911 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 912 (value != 0)) { 913 for (size_t i = 0; i < mRecordThreads.size(); i++) { 914 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 915 } 916 } 917 } 918 } 919 if (thread != 0) { 920 return thread->setParameters(keyValuePairs); 921 } 922 return BAD_VALUE; 923} 924 925String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 926{ 927// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 928// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 929 930 Mutex::Autolock _l(mLock); 931 932 if (ioHandle == 0) { 933 String8 out_s8; 934 935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 936 char *s; 937 { 938 AutoMutex lock(mHardwareLock); 939 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 940 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 941 s = dev->get_parameters(dev, keys.string()); 942 mHardwareStatus = AUDIO_HW_IDLE; 943 } 944 out_s8 += String8(s ? s : ""); 945 free(s); 946 } 947 return out_s8; 948 } 949 950 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 951 if (playbackThread != NULL) { 952 return playbackThread->getParameters(keys); 953 } 954 RecordThread *recordThread = checkRecordThread_l(ioHandle); 955 if (recordThread != NULL) { 956 return recordThread->getParameters(keys); 957 } 958 return String8(""); 959} 960 961size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 962{ 963 status_t ret = initCheck(); 964 if (ret != NO_ERROR) { 965 return 0; 966 } 967 968 AutoMutex lock(mHardwareLock); 969 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 970 struct audio_config config = { 971 sample_rate: sampleRate, 972 channel_mask: audio_channel_in_mask_from_count(channelCount), 973 format: format, 974 }; 975 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 976 mHardwareStatus = AUDIO_HW_IDLE; 977 return size; 978} 979 980unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 981{ 982 if (ioHandle == 0) { 983 return 0; 984 } 985 986 Mutex::Autolock _l(mLock); 987 988 RecordThread *recordThread = checkRecordThread_l(ioHandle); 989 if (recordThread != NULL) { 990 return recordThread->getInputFramesLost(); 991 } 992 return 0; 993} 994 995status_t AudioFlinger::setVoiceVolume(float value) 996{ 997 status_t ret = initCheck(); 998 if (ret != NO_ERROR) { 999 return ret; 1000 } 1001 1002 // check calling permissions 1003 if (!settingsAllowed()) { 1004 return PERMISSION_DENIED; 1005 } 1006 1007 AutoMutex lock(mHardwareLock); 1008 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1009 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1010 mHardwareStatus = AUDIO_HW_IDLE; 1011 1012 return ret; 1013} 1014 1015status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1016 audio_io_handle_t output) const 1017{ 1018 status_t status; 1019 1020 Mutex::Autolock _l(mLock); 1021 1022 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1023 if (playbackThread != NULL) { 1024 return playbackThread->getRenderPosition(halFrames, dspFrames); 1025 } 1026 1027 return BAD_VALUE; 1028} 1029 1030void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1031{ 1032 1033 Mutex::Autolock _l(mLock); 1034 1035 pid_t pid = IPCThreadState::self()->getCallingPid(); 1036 if (mNotificationClients.indexOfKey(pid) < 0) { 1037 sp<NotificationClient> notificationClient = new NotificationClient(this, 1038 client, 1039 pid); 1040 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1041 1042 mNotificationClients.add(pid, notificationClient); 1043 1044 sp<IBinder> binder = client->asBinder(); 1045 binder->linkToDeath(notificationClient); 1046 1047 // the config change is always sent from playback or record threads to avoid deadlock 1048 // with AudioSystem::gLock 1049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1050 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1051 } 1052 1053 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1054 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1055 } 1056 } 1057} 1058 1059void AudioFlinger::removeNotificationClient(pid_t pid) 1060{ 1061 Mutex::Autolock _l(mLock); 1062 1063 mNotificationClients.removeItem(pid); 1064 1065 ALOGV("%d died, releasing its sessions", pid); 1066 size_t num = mAudioSessionRefs.size(); 1067 bool removed = false; 1068 for (size_t i = 0; i< num; ) { 1069 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1070 ALOGV(" pid %d @ %d", ref->mPid, i); 1071 if (ref->mPid == pid) { 1072 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1073 mAudioSessionRefs.removeAt(i); 1074 delete ref; 1075 removed = true; 1076 num--; 1077 } else { 1078 i++; 1079 } 1080 } 1081 if (removed) { 1082 purgeStaleEffects_l(); 1083 } 1084} 1085 1086// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1087void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1088{ 1089 size_t size = mNotificationClients.size(); 1090 for (size_t i = 0; i < size; i++) { 1091 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1092 param2); 1093 } 1094} 1095 1096// removeClient_l() must be called with AudioFlinger::mLock held 1097void AudioFlinger::removeClient_l(pid_t pid) 1098{ 1099 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1100 mClients.removeItem(pid); 1101} 1102 1103 1104// ---------------------------------------------------------------------------- 1105 1106AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1107 uint32_t device, type_t type) 1108 : Thread(false), 1109 mType(type), 1110 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1111 // mChannelMask 1112 mChannelCount(0), 1113 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1114 mParamStatus(NO_ERROR), 1115 mStandby(false), mId(id), 1116 mDevice(device), 1117 mDeathRecipient(new PMDeathRecipient(this)) 1118{ 1119} 1120 1121AudioFlinger::ThreadBase::~ThreadBase() 1122{ 1123 mParamCond.broadcast(); 1124 // do not lock the mutex in destructor 1125 releaseWakeLock_l(); 1126 if (mPowerManager != 0) { 1127 sp<IBinder> binder = mPowerManager->asBinder(); 1128 binder->unlinkToDeath(mDeathRecipient); 1129 } 1130} 1131 1132void AudioFlinger::ThreadBase::exit() 1133{ 1134 ALOGV("ThreadBase::exit"); 1135 { 1136 // This lock prevents the following race in thread (uniprocessor for illustration): 1137 // if (!exitPending()) { 1138 // // context switch from here to exit() 1139 // // exit() calls requestExit(), what exitPending() observes 1140 // // exit() calls signal(), which is dropped since no waiters 1141 // // context switch back from exit() to here 1142 // mWaitWorkCV.wait(...); 1143 // // now thread is hung 1144 // } 1145 AutoMutex lock(mLock); 1146 requestExit(); 1147 mWaitWorkCV.signal(); 1148 } 1149 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1150 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1151 requestExitAndWait(); 1152} 1153 1154status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1155{ 1156 status_t status; 1157 1158 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1159 Mutex::Autolock _l(mLock); 1160 1161 mNewParameters.add(keyValuePairs); 1162 mWaitWorkCV.signal(); 1163 // wait condition with timeout in case the thread loop has exited 1164 // before the request could be processed 1165 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1166 status = mParamStatus; 1167 mWaitWorkCV.signal(); 1168 } else { 1169 status = TIMED_OUT; 1170 } 1171 return status; 1172} 1173 1174void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1175{ 1176 Mutex::Autolock _l(mLock); 1177 sendConfigEvent_l(event, param); 1178} 1179 1180// sendConfigEvent_l() must be called with ThreadBase::mLock held 1181void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1182{ 1183 ConfigEvent configEvent; 1184 configEvent.mEvent = event; 1185 configEvent.mParam = param; 1186 mConfigEvents.add(configEvent); 1187 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1188 mWaitWorkCV.signal(); 1189} 1190 1191void AudioFlinger::ThreadBase::processConfigEvents() 1192{ 1193 mLock.lock(); 1194 while (!mConfigEvents.isEmpty()) { 1195 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1196 ConfigEvent configEvent = mConfigEvents[0]; 1197 mConfigEvents.removeAt(0); 1198 // release mLock before locking AudioFlinger mLock: lock order is always 1199 // AudioFlinger then ThreadBase to avoid cross deadlock 1200 mLock.unlock(); 1201 mAudioFlinger->mLock.lock(); 1202 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1203 mAudioFlinger->mLock.unlock(); 1204 mLock.lock(); 1205 } 1206 mLock.unlock(); 1207} 1208 1209status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1210{ 1211 const size_t SIZE = 256; 1212 char buffer[SIZE]; 1213 String8 result; 1214 1215 bool locked = tryLock(mLock); 1216 if (!locked) { 1217 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1218 write(fd, buffer, strlen(buffer)); 1219 } 1220 1221 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1240 result.append(buffer); 1241 1242 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1243 result.append(buffer); 1244 result.append(" Index Command"); 1245 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1246 snprintf(buffer, SIZE, "\n %02d ", i); 1247 result.append(buffer); 1248 result.append(mNewParameters[i]); 1249 } 1250 1251 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1252 result.append(buffer); 1253 snprintf(buffer, SIZE, " Index event param\n"); 1254 result.append(buffer); 1255 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1256 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1257 result.append(buffer); 1258 } 1259 result.append("\n"); 1260 1261 write(fd, result.string(), result.size()); 1262 1263 if (locked) { 1264 mLock.unlock(); 1265 } 1266 return NO_ERROR; 1267} 1268 1269status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1270{ 1271 const size_t SIZE = 256; 1272 char buffer[SIZE]; 1273 String8 result; 1274 1275 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1276 write(fd, buffer, strlen(buffer)); 1277 1278 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1279 sp<EffectChain> chain = mEffectChains[i]; 1280 if (chain != 0) { 1281 chain->dump(fd, args); 1282 } 1283 } 1284 return NO_ERROR; 1285} 1286 1287void AudioFlinger::ThreadBase::acquireWakeLock() 1288{ 1289 Mutex::Autolock _l(mLock); 1290 acquireWakeLock_l(); 1291} 1292 1293void AudioFlinger::ThreadBase::acquireWakeLock_l() 1294{ 1295 if (mPowerManager == 0) { 1296 // use checkService() to avoid blocking if power service is not up yet 1297 sp<IBinder> binder = 1298 defaultServiceManager()->checkService(String16("power")); 1299 if (binder == 0) { 1300 ALOGW("Thread %s cannot connect to the power manager service", mName); 1301 } else { 1302 mPowerManager = interface_cast<IPowerManager>(binder); 1303 binder->linkToDeath(mDeathRecipient); 1304 } 1305 } 1306 if (mPowerManager != 0) { 1307 sp<IBinder> binder = new BBinder(); 1308 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1309 binder, 1310 String16(mName)); 1311 if (status == NO_ERROR) { 1312 mWakeLockToken = binder; 1313 } 1314 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1315 } 1316} 1317 1318void AudioFlinger::ThreadBase::releaseWakeLock() 1319{ 1320 Mutex::Autolock _l(mLock); 1321 releaseWakeLock_l(); 1322} 1323 1324void AudioFlinger::ThreadBase::releaseWakeLock_l() 1325{ 1326 if (mWakeLockToken != 0) { 1327 ALOGV("releaseWakeLock_l() %s", mName); 1328 if (mPowerManager != 0) { 1329 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1330 } 1331 mWakeLockToken.clear(); 1332 } 1333} 1334 1335void AudioFlinger::ThreadBase::clearPowerManager() 1336{ 1337 Mutex::Autolock _l(mLock); 1338 releaseWakeLock_l(); 1339 mPowerManager.clear(); 1340} 1341 1342void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1343{ 1344 sp<ThreadBase> thread = mThread.promote(); 1345 if (thread != 0) { 1346 thread->clearPowerManager(); 1347 } 1348 ALOGW("power manager service died !!!"); 1349} 1350 1351void AudioFlinger::ThreadBase::setEffectSuspended( 1352 const effect_uuid_t *type, bool suspend, int sessionId) 1353{ 1354 Mutex::Autolock _l(mLock); 1355 setEffectSuspended_l(type, suspend, sessionId); 1356} 1357 1358void AudioFlinger::ThreadBase::setEffectSuspended_l( 1359 const effect_uuid_t *type, bool suspend, int sessionId) 1360{ 1361 sp<EffectChain> chain = getEffectChain_l(sessionId); 1362 if (chain != 0) { 1363 if (type != NULL) { 1364 chain->setEffectSuspended_l(type, suspend); 1365 } else { 1366 chain->setEffectSuspendedAll_l(suspend); 1367 } 1368 } 1369 1370 updateSuspendedSessions_l(type, suspend, sessionId); 1371} 1372 1373void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1374{ 1375 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1376 if (index < 0) { 1377 return; 1378 } 1379 1380 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1381 mSuspendedSessions.editValueAt(index); 1382 1383 for (size_t i = 0; i < sessionEffects.size(); i++) { 1384 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1385 for (int j = 0; j < desc->mRefCount; j++) { 1386 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1387 chain->setEffectSuspendedAll_l(true); 1388 } else { 1389 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1390 desc->mType.timeLow); 1391 chain->setEffectSuspended_l(&desc->mType, true); 1392 } 1393 } 1394 } 1395} 1396 1397void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1398 bool suspend, 1399 int sessionId) 1400{ 1401 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1402 1403 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1404 1405 if (suspend) { 1406 if (index >= 0) { 1407 sessionEffects = mSuspendedSessions.editValueAt(index); 1408 } else { 1409 mSuspendedSessions.add(sessionId, sessionEffects); 1410 } 1411 } else { 1412 if (index < 0) { 1413 return; 1414 } 1415 sessionEffects = mSuspendedSessions.editValueAt(index); 1416 } 1417 1418 1419 int key = EffectChain::kKeyForSuspendAll; 1420 if (type != NULL) { 1421 key = type->timeLow; 1422 } 1423 index = sessionEffects.indexOfKey(key); 1424 1425 sp<SuspendedSessionDesc> desc; 1426 if (suspend) { 1427 if (index >= 0) { 1428 desc = sessionEffects.valueAt(index); 1429 } else { 1430 desc = new SuspendedSessionDesc(); 1431 if (type != NULL) { 1432 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1433 } 1434 sessionEffects.add(key, desc); 1435 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1436 } 1437 desc->mRefCount++; 1438 } else { 1439 if (index < 0) { 1440 return; 1441 } 1442 desc = sessionEffects.valueAt(index); 1443 if (--desc->mRefCount == 0) { 1444 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1445 sessionEffects.removeItemsAt(index); 1446 if (sessionEffects.isEmpty()) { 1447 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1448 sessionId); 1449 mSuspendedSessions.removeItem(sessionId); 1450 } 1451 } 1452 } 1453 if (!sessionEffects.isEmpty()) { 1454 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1455 } 1456} 1457 1458void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1459 bool enabled, 1460 int sessionId) 1461{ 1462 Mutex::Autolock _l(mLock); 1463 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1464} 1465 1466void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1467 bool enabled, 1468 int sessionId) 1469{ 1470 if (mType != RECORD) { 1471 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1472 // another session. This gives the priority to well behaved effect control panels 1473 // and applications not using global effects. 1474 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1475 // global effects 1476 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1477 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1478 } 1479 } 1480 1481 sp<EffectChain> chain = getEffectChain_l(sessionId); 1482 if (chain != 0) { 1483 chain->checkSuspendOnEffectEnabled(effect, enabled); 1484 } 1485} 1486 1487// ---------------------------------------------------------------------------- 1488 1489AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1490 AudioStreamOut* output, 1491 audio_io_handle_t id, 1492 uint32_t device, 1493 type_t type) 1494 : ThreadBase(audioFlinger, id, device, type), 1495 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1496 // Assumes constructor is called by AudioFlinger with it's mLock held, 1497 // but it would be safer to explicitly pass initial masterMute as parameter 1498 mMasterMute(audioFlinger->masterMute_l()), 1499 // mStreamTypes[] initialized in constructor body 1500 mOutput(output), 1501 // Assumes constructor is called by AudioFlinger with it's mLock held, 1502 // but it would be safer to explicitly pass initial masterVolume as parameter 1503 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1504 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1505 mMixerStatus(MIXER_IDLE), 1506 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1507 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1508 // index 0 is reserved for normal mixer's submix 1509 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1510{ 1511 snprintf(mName, kNameLength, "AudioOut_%X", id); 1512 1513 readOutputParameters(); 1514 1515 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1516 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1517 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1518 stream = (audio_stream_type_t) (stream + 1)) { 1519 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1520 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1521 } 1522 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1523 // because mAudioFlinger doesn't have one to copy from 1524} 1525 1526AudioFlinger::PlaybackThread::~PlaybackThread() 1527{ 1528 delete [] mMixBuffer; 1529} 1530 1531status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1532{ 1533 dumpInternals(fd, args); 1534 dumpTracks(fd, args); 1535 dumpEffectChains(fd, args); 1536 return NO_ERROR; 1537} 1538 1539status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1540{ 1541 const size_t SIZE = 256; 1542 char buffer[SIZE]; 1543 String8 result; 1544 1545 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1546 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1547 const stream_type_t *st = &mStreamTypes[i]; 1548 if (i > 0) { 1549 result.appendFormat(", "); 1550 } 1551 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1552 if (st->mute) { 1553 result.append("M"); 1554 } 1555 } 1556 result.append("\n"); 1557 write(fd, result.string(), result.length()); 1558 result.clear(); 1559 1560 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1561 result.append(buffer); 1562 Track::appendDumpHeader(result); 1563 for (size_t i = 0; i < mTracks.size(); ++i) { 1564 sp<Track> track = mTracks[i]; 1565 if (track != 0) { 1566 track->dump(buffer, SIZE); 1567 result.append(buffer); 1568 } 1569 } 1570 1571 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1572 result.append(buffer); 1573 Track::appendDumpHeader(result); 1574 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1575 sp<Track> track = mActiveTracks[i].promote(); 1576 if (track != 0) { 1577 track->dump(buffer, SIZE); 1578 result.append(buffer); 1579 } 1580 } 1581 write(fd, result.string(), result.size()); 1582 return NO_ERROR; 1583} 1584 1585status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1586{ 1587 const size_t SIZE = 256; 1588 char buffer[SIZE]; 1589 String8 result; 1590 1591 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1592 result.append(buffer); 1593 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1604 result.append(buffer); 1605 write(fd, result.string(), result.size()); 1606 1607 dumpBase(fd, args); 1608 1609 return NO_ERROR; 1610} 1611 1612// Thread virtuals 1613status_t AudioFlinger::PlaybackThread::readyToRun() 1614{ 1615 status_t status = initCheck(); 1616 if (status == NO_ERROR) { 1617 ALOGI("AudioFlinger's thread %p ready to run", this); 1618 } else { 1619 ALOGE("No working audio driver found."); 1620 } 1621 return status; 1622} 1623 1624void AudioFlinger::PlaybackThread::onFirstRef() 1625{ 1626 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1627} 1628 1629// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1630sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1631 const sp<AudioFlinger::Client>& client, 1632 audio_stream_type_t streamType, 1633 uint32_t sampleRate, 1634 audio_format_t format, 1635 uint32_t channelMask, 1636 int frameCount, 1637 const sp<IMemory>& sharedBuffer, 1638 int sessionId, 1639 IAudioFlinger::track_flags_t flags, 1640 pid_t tid, 1641 status_t *status) 1642{ 1643 sp<Track> track; 1644 status_t lStatus; 1645 1646 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1647 1648 // client expresses a preference for FAST, but we get the final say 1649 if (flags & IAudioFlinger::TRACK_FAST) { 1650 if ( 1651 // not timed 1652 (!isTimed) && 1653 // either of these use cases: 1654 ( 1655 // use case 1: shared buffer with any frame count 1656 ( 1657 (sharedBuffer != 0) 1658 ) || 1659 // use case 2: callback handler and frame count is default or at least as large as HAL 1660 ( 1661 (tid != -1) && 1662 ((frameCount == 0) || 1663 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1664 ) 1665 ) && 1666 // PCM data 1667 audio_is_linear_pcm(format) && 1668 // mono or stereo 1669 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1670 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1671#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1672 // hardware sample rate 1673 (sampleRate == mSampleRate) && 1674#endif 1675 // normal mixer has an associated fast mixer 1676 hasFastMixer() && 1677 // there are sufficient fast track slots available 1678 (mFastTrackAvailMask != 0) 1679 // FIXME test that MixerThread for this fast track has a capable output HAL 1680 // FIXME add a permission test also? 1681 ) { 1682 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1683 if (frameCount == 0) { 1684 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1685 } 1686 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1687 frameCount, mFrameCount); 1688 } else { 1689 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1690 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1691 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1692 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1693 audio_is_linear_pcm(format), 1694 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1695 flags &= ~IAudioFlinger::TRACK_FAST; 1696 // For compatibility with AudioTrack calculation, buffer depth is forced 1697 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1698 // This is probably too conservative, but legacy application code may depend on it. 1699 // If you change this calculation, also review the start threshold which is related. 1700 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1701 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1702 if (minBufCount < 2) { 1703 minBufCount = 2; 1704 } 1705 int minFrameCount = mNormalFrameCount * minBufCount; 1706 if (frameCount < minFrameCount) { 1707 frameCount = minFrameCount; 1708 } 1709 } 1710 } 1711 1712 if (mType == DIRECT) { 1713 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1714 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1715 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1716 "for output %p with format %d", 1717 sampleRate, format, channelMask, mOutput, mFormat); 1718 lStatus = BAD_VALUE; 1719 goto Exit; 1720 } 1721 } 1722 } else { 1723 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1724 if (sampleRate > mSampleRate*2) { 1725 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1726 lStatus = BAD_VALUE; 1727 goto Exit; 1728 } 1729 } 1730 1731 lStatus = initCheck(); 1732 if (lStatus != NO_ERROR) { 1733 ALOGE("Audio driver not initialized."); 1734 goto Exit; 1735 } 1736 1737 { // scope for mLock 1738 Mutex::Autolock _l(mLock); 1739 1740 // all tracks in same audio session must share the same routing strategy otherwise 1741 // conflicts will happen when tracks are moved from one output to another by audio policy 1742 // manager 1743 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1744 for (size_t i = 0; i < mTracks.size(); ++i) { 1745 sp<Track> t = mTracks[i]; 1746 if (t != 0 && !t->isOutputTrack()) { 1747 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1748 if (sessionId == t->sessionId() && strategy != actual) { 1749 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1750 strategy, actual); 1751 lStatus = BAD_VALUE; 1752 goto Exit; 1753 } 1754 } 1755 } 1756 1757 if (!isTimed) { 1758 track = new Track(this, client, streamType, sampleRate, format, 1759 channelMask, frameCount, sharedBuffer, sessionId, flags); 1760 } else { 1761 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1762 channelMask, frameCount, sharedBuffer, sessionId); 1763 } 1764 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1765 lStatus = NO_MEMORY; 1766 goto Exit; 1767 } 1768 mTracks.add(track); 1769 1770 sp<EffectChain> chain = getEffectChain_l(sessionId); 1771 if (chain != 0) { 1772 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1773 track->setMainBuffer(chain->inBuffer()); 1774 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1775 chain->incTrackCnt(); 1776 } 1777 } 1778 1779#ifdef HAVE_REQUEST_PRIORITY 1780 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1781 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1782 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1783 // so ask activity manager to do this on our behalf 1784 int err = requestPriority(callingPid, tid, 1); 1785 if (err != 0) { 1786 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1787 1, callingPid, tid, err); 1788 } 1789 } 1790#endif 1791 1792 lStatus = NO_ERROR; 1793 1794Exit: 1795 if (status) { 1796 *status = lStatus; 1797 } 1798 return track; 1799} 1800 1801uint32_t AudioFlinger::PlaybackThread::latency() const 1802{ 1803 Mutex::Autolock _l(mLock); 1804 if (initCheck() == NO_ERROR) { 1805 return mOutput->stream->get_latency(mOutput->stream); 1806 } else { 1807 return 0; 1808 } 1809} 1810 1811void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1812{ 1813 Mutex::Autolock _l(mLock); 1814 mMasterVolume = value; 1815} 1816 1817void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1818{ 1819 Mutex::Autolock _l(mLock); 1820 setMasterMute_l(muted); 1821} 1822 1823void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1824{ 1825 Mutex::Autolock _l(mLock); 1826 mStreamTypes[stream].volume = value; 1827} 1828 1829void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1830{ 1831 Mutex::Autolock _l(mLock); 1832 mStreamTypes[stream].mute = muted; 1833} 1834 1835float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1836{ 1837 Mutex::Autolock _l(mLock); 1838 return mStreamTypes[stream].volume; 1839} 1840 1841// addTrack_l() must be called with ThreadBase::mLock held 1842status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1843{ 1844 status_t status = ALREADY_EXISTS; 1845 1846 // set retry count for buffer fill 1847 track->mRetryCount = kMaxTrackStartupRetries; 1848 if (mActiveTracks.indexOf(track) < 0) { 1849 // the track is newly added, make sure it fills up all its 1850 // buffers before playing. This is to ensure the client will 1851 // effectively get the latency it requested. 1852 track->mFillingUpStatus = Track::FS_FILLING; 1853 track->mResetDone = false; 1854 track->mPresentationCompleteFrames = 0; 1855 mActiveTracks.add(track); 1856 if (track->mainBuffer() != mMixBuffer) { 1857 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1858 if (chain != 0) { 1859 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1860 chain->incActiveTrackCnt(); 1861 } 1862 } 1863 1864 status = NO_ERROR; 1865 } 1866 1867 ALOGV("mWaitWorkCV.broadcast"); 1868 mWaitWorkCV.broadcast(); 1869 1870 return status; 1871} 1872 1873// destroyTrack_l() must be called with ThreadBase::mLock held 1874void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1875{ 1876 track->mState = TrackBase::TERMINATED; 1877 // active tracks are removed by threadLoop() 1878 if (mActiveTracks.indexOf(track) < 0) { 1879 removeTrack_l(track); 1880 } 1881} 1882 1883void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1884{ 1885 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1886 mTracks.remove(track); 1887 deleteTrackName_l(track->name()); 1888 // redundant as track is about to be destroyed, for dumpsys only 1889 track->mName = -1; 1890 if (track->isFastTrack()) { 1891 int index = track->mFastIndex; 1892 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1893 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1894 mFastTrackAvailMask |= 1 << index; 1895 // redundant as track is about to be destroyed, for dumpsys only 1896 track->mFastIndex = -1; 1897 } 1898 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1899 if (chain != 0) { 1900 chain->decTrackCnt(); 1901 } 1902} 1903 1904String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1905{ 1906 String8 out_s8 = String8(""); 1907 char *s; 1908 1909 Mutex::Autolock _l(mLock); 1910 if (initCheck() != NO_ERROR) { 1911 return out_s8; 1912 } 1913 1914 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1915 out_s8 = String8(s); 1916 free(s); 1917 return out_s8; 1918} 1919 1920// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1921void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1922 AudioSystem::OutputDescriptor desc; 1923 void *param2 = NULL; 1924 1925 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1926 1927 switch (event) { 1928 case AudioSystem::OUTPUT_OPENED: 1929 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1930 desc.channels = mChannelMask; 1931 desc.samplingRate = mSampleRate; 1932 desc.format = mFormat; 1933 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1934 desc.latency = latency(); 1935 param2 = &desc; 1936 break; 1937 1938 case AudioSystem::STREAM_CONFIG_CHANGED: 1939 param2 = ¶m; 1940 case AudioSystem::OUTPUT_CLOSED: 1941 default: 1942 break; 1943 } 1944 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1945} 1946 1947void AudioFlinger::PlaybackThread::readOutputParameters() 1948{ 1949 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1950 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1951 mChannelCount = (uint16_t)popcount(mChannelMask); 1952 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1953 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1954 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1955 if (mFrameCount & 15) { 1956 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1957 mFrameCount); 1958 } 1959 1960 // Calculate size of normal mix buffer relative to the HAL output buffer size 1961 double multiplier = 1.0; 1962 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1963 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1964 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1965 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1966 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1967 maxNormalFrameCount = maxNormalFrameCount & ~15; 1968 if (maxNormalFrameCount < minNormalFrameCount) { 1969 maxNormalFrameCount = minNormalFrameCount; 1970 } 1971 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1972 if (multiplier <= 1.0) { 1973 multiplier = 1.0; 1974 } else if (multiplier <= 2.0) { 1975 if (2 * mFrameCount <= maxNormalFrameCount) { 1976 multiplier = 2.0; 1977 } else { 1978 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1979 } 1980 } else { 1981 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1982 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1983 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 1984 // FIXME this rounding up should not be done if no HAL SRC 1985 uint32_t truncMult = (uint32_t) multiplier; 1986 if ((truncMult & 1)) { 1987 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1988 ++truncMult; 1989 } 1990 } 1991 multiplier = (double) truncMult; 1992 } 1993 } 1994 mNormalFrameCount = multiplier * mFrameCount; 1995 // round up to nearest 16 frames to satisfy AudioMixer 1996 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1997 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 1998 1999 // FIXME - Current mixer implementation only supports stereo output: Always 2000 // Allocate a stereo buffer even if HW output is mono. 2001 delete[] mMixBuffer; 2002 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2003 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2004 2005 // force reconfiguration of effect chains and engines to take new buffer size and audio 2006 // parameters into account 2007 // Note that mLock is not held when readOutputParameters() is called from the constructor 2008 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2009 // matter. 2010 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2011 Vector< sp<EffectChain> > effectChains = mEffectChains; 2012 for (size_t i = 0; i < effectChains.size(); i ++) { 2013 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2014 } 2015} 2016 2017status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2018{ 2019 if (halFrames == NULL || dspFrames == NULL) { 2020 return BAD_VALUE; 2021 } 2022 Mutex::Autolock _l(mLock); 2023 if (initCheck() != NO_ERROR) { 2024 return INVALID_OPERATION; 2025 } 2026 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2027 2028 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2029} 2030 2031uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2032{ 2033 Mutex::Autolock _l(mLock); 2034 uint32_t result = 0; 2035 if (getEffectChain_l(sessionId) != 0) { 2036 result = EFFECT_SESSION; 2037 } 2038 2039 for (size_t i = 0; i < mTracks.size(); ++i) { 2040 sp<Track> track = mTracks[i]; 2041 if (sessionId == track->sessionId() && 2042 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2043 result |= TRACK_SESSION; 2044 break; 2045 } 2046 } 2047 2048 return result; 2049} 2050 2051uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2052{ 2053 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2054 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2055 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2056 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2057 } 2058 for (size_t i = 0; i < mTracks.size(); i++) { 2059 sp<Track> track = mTracks[i]; 2060 if (sessionId == track->sessionId() && 2061 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2062 return AudioSystem::getStrategyForStream(track->streamType()); 2063 } 2064 } 2065 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2066} 2067 2068 2069AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2070{ 2071 Mutex::Autolock _l(mLock); 2072 return mOutput; 2073} 2074 2075AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2076{ 2077 Mutex::Autolock _l(mLock); 2078 AudioStreamOut *output = mOutput; 2079 mOutput = NULL; 2080 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2081 // must push a NULL and wait for ack 2082 mOutputSink.clear(); 2083 mPipeSink.clear(); 2084 mNormalSink.clear(); 2085 return output; 2086} 2087 2088// this method must always be called either with ThreadBase mLock held or inside the thread loop 2089audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2090{ 2091 if (mOutput == NULL) { 2092 return NULL; 2093 } 2094 return &mOutput->stream->common; 2095} 2096 2097uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2098{ 2099 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2100 // decoding and transfer time. So sleeping for half of the latency would likely cause 2101 // underruns 2102 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2103 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2104 } else { 2105 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2106 } 2107} 2108 2109status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2110{ 2111 if (!isValidSyncEvent(event)) { 2112 return BAD_VALUE; 2113 } 2114 2115 Mutex::Autolock _l(mLock); 2116 2117 for (size_t i = 0; i < mTracks.size(); ++i) { 2118 sp<Track> track = mTracks[i]; 2119 if (event->triggerSession() == track->sessionId()) { 2120 track->setSyncEvent(event); 2121 return NO_ERROR; 2122 } 2123 } 2124 2125 return NAME_NOT_FOUND; 2126} 2127 2128bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2129{ 2130 switch (event->type()) { 2131 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2132 return true; 2133 default: 2134 break; 2135 } 2136 return false; 2137} 2138 2139// ---------------------------------------------------------------------------- 2140 2141AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2142 audio_io_handle_t id, uint32_t device, type_t type) 2143 : PlaybackThread(audioFlinger, output, id, device, type), 2144 // mAudioMixer below 2145#ifdef SOAKER 2146 mSoaker(NULL), 2147#endif 2148 // mFastMixer below 2149 mFastMixerFutex(0) 2150 // mOutputSink below 2151 // mPipeSink below 2152 // mNormalSink below 2153{ 2154 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2155 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2156 "mFrameCount=%d, mNormalFrameCount=%d", 2157 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2158 mNormalFrameCount); 2159 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2160 2161 // FIXME - Current mixer implementation only supports stereo output 2162 if (mChannelCount == 1) { 2163 ALOGE("Invalid audio hardware channel count"); 2164 } 2165 2166 // create an NBAIO sink for the HAL output stream, and negotiate 2167 mOutputSink = new AudioStreamOutSink(output->stream); 2168 size_t numCounterOffers = 0; 2169 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2170 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2171 ALOG_ASSERT(index == 0); 2172 2173 // initialize fast mixer depending on configuration 2174 bool initFastMixer; 2175 switch (kUseFastMixer) { 2176 case FastMixer_Never: 2177 initFastMixer = false; 2178 break; 2179 case FastMixer_Always: 2180 initFastMixer = true; 2181 break; 2182 case FastMixer_Static: 2183 case FastMixer_Dynamic: 2184 initFastMixer = mFrameCount < mNormalFrameCount; 2185 break; 2186 } 2187 if (initFastMixer) { 2188 2189 // create a MonoPipe to connect our submix to FastMixer 2190 NBAIO_Format format = mOutputSink->format(); 2191 // frame count will be rounded up to a power of 2, so this formula should work well 2192 MonoPipe *monoPipe = new MonoPipe((mNormalFrameCount * 3) / 2, format, 2193 true /*writeCanBlock*/); 2194 const NBAIO_Format offers[1] = {format}; 2195 size_t numCounterOffers = 0; 2196 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2197 ALOG_ASSERT(index == 0); 2198 mPipeSink = monoPipe; 2199 2200#ifdef SOAKER 2201 // create a soaker as workaround for governor issues 2202 mSoaker = new Soaker(); 2203 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2204 mSoaker->run("Soaker", PRIORITY_LOWEST); 2205#endif 2206 2207 // create fast mixer and configure it initially with just one fast track for our submix 2208 mFastMixer = new FastMixer(); 2209 FastMixerStateQueue *sq = mFastMixer->sq(); 2210 FastMixerState *state = sq->begin(); 2211 FastTrack *fastTrack = &state->mFastTracks[0]; 2212 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2213 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2214 fastTrack->mVolumeProvider = NULL; 2215 fastTrack->mGeneration++; 2216 state->mFastTracksGen++; 2217 state->mTrackMask = 1; 2218 // fast mixer will use the HAL output sink 2219 state->mOutputSink = mOutputSink.get(); 2220 state->mOutputSinkGen++; 2221 state->mFrameCount = mFrameCount; 2222 state->mCommand = FastMixerState::COLD_IDLE; 2223 // already done in constructor initialization list 2224 //mFastMixerFutex = 0; 2225 state->mColdFutexAddr = &mFastMixerFutex; 2226 state->mColdGen++; 2227 state->mDumpState = &mFastMixerDumpState; 2228 sq->end(); 2229 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2230 2231 // start the fast mixer 2232 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2233#ifdef HAVE_REQUEST_PRIORITY 2234 pid_t tid = mFastMixer->getTid(); 2235 int err = requestPriority(getpid_cached, tid, 2); 2236 if (err != 0) { 2237 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2238 2, getpid_cached, tid, err); 2239 } 2240#endif 2241 2242 } else { 2243 mFastMixer = NULL; 2244 } 2245 2246 switch (kUseFastMixer) { 2247 case FastMixer_Never: 2248 case FastMixer_Dynamic: 2249 mNormalSink = mOutputSink; 2250 break; 2251 case FastMixer_Always: 2252 mNormalSink = mPipeSink; 2253 break; 2254 case FastMixer_Static: 2255 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2256 break; 2257 } 2258} 2259 2260AudioFlinger::MixerThread::~MixerThread() 2261{ 2262 if (mFastMixer != NULL) { 2263 FastMixerStateQueue *sq = mFastMixer->sq(); 2264 FastMixerState *state = sq->begin(); 2265 if (state->mCommand == FastMixerState::COLD_IDLE) { 2266 int32_t old = android_atomic_inc(&mFastMixerFutex); 2267 if (old == -1) { 2268 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2269 } 2270 } 2271 state->mCommand = FastMixerState::EXIT; 2272 sq->end(); 2273 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2274 mFastMixer->join(); 2275 // Though the fast mixer thread has exited, it's state queue is still valid. 2276 // We'll use that extract the final state which contains one remaining fast track 2277 // corresponding to our sub-mix. 2278 state = sq->begin(); 2279 ALOG_ASSERT(state->mTrackMask == 1); 2280 FastTrack *fastTrack = &state->mFastTracks[0]; 2281 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2282 delete fastTrack->mBufferProvider; 2283 sq->end(false /*didModify*/); 2284 delete mFastMixer; 2285#ifdef SOAKER 2286 if (mSoaker != NULL) { 2287 mSoaker->requestExitAndWait(); 2288 } 2289 delete mSoaker; 2290#endif 2291 } 2292 delete mAudioMixer; 2293} 2294 2295class CpuStats { 2296public: 2297 CpuStats(); 2298 void sample(const String8 &title); 2299#ifdef DEBUG_CPU_USAGE 2300private: 2301 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2302 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2303 2304 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2305 2306 int mCpuNum; // thread's current CPU number 2307 int mCpukHz; // frequency of thread's current CPU in kHz 2308#endif 2309}; 2310 2311CpuStats::CpuStats() 2312#ifdef DEBUG_CPU_USAGE 2313 : mCpuNum(-1), mCpukHz(-1) 2314#endif 2315{ 2316} 2317 2318void CpuStats::sample(const String8 &title) { 2319#ifdef DEBUG_CPU_USAGE 2320 // get current thread's delta CPU time in wall clock ns 2321 double wcNs; 2322 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2323 2324 // record sample for wall clock statistics 2325 if (valid) { 2326 mWcStats.sample(wcNs); 2327 } 2328 2329 // get the current CPU number 2330 int cpuNum = sched_getcpu(); 2331 2332 // get the current CPU frequency in kHz 2333 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2334 2335 // check if either CPU number or frequency changed 2336 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2337 mCpuNum = cpuNum; 2338 mCpukHz = cpukHz; 2339 // ignore sample for purposes of cycles 2340 valid = false; 2341 } 2342 2343 // if no change in CPU number or frequency, then record sample for cycle statistics 2344 if (valid && mCpukHz > 0) { 2345 double cycles = wcNs * cpukHz * 0.000001; 2346 mHzStats.sample(cycles); 2347 } 2348 2349 unsigned n = mWcStats.n(); 2350 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2351 if ((n & 127) == 1) { 2352 long long elapsed = mCpuUsage.elapsed(); 2353 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2354 double perLoop = elapsed / (double) n; 2355 double perLoop100 = perLoop * 0.01; 2356 double perLoop1k = perLoop * 0.001; 2357 double mean = mWcStats.mean(); 2358 double stddev = mWcStats.stddev(); 2359 double minimum = mWcStats.minimum(); 2360 double maximum = mWcStats.maximum(); 2361 double meanCycles = mHzStats.mean(); 2362 double stddevCycles = mHzStats.stddev(); 2363 double minCycles = mHzStats.minimum(); 2364 double maxCycles = mHzStats.maximum(); 2365 mCpuUsage.resetElapsed(); 2366 mWcStats.reset(); 2367 mHzStats.reset(); 2368 ALOGD("CPU usage for %s over past %.1f secs\n" 2369 " (%u mixer loops at %.1f mean ms per loop):\n" 2370 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2371 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2372 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2373 title.string(), 2374 elapsed * .000000001, n, perLoop * .000001, 2375 mean * .001, 2376 stddev * .001, 2377 minimum * .001, 2378 maximum * .001, 2379 mean / perLoop100, 2380 stddev / perLoop100, 2381 minimum / perLoop100, 2382 maximum / perLoop100, 2383 meanCycles / perLoop1k, 2384 stddevCycles / perLoop1k, 2385 minCycles / perLoop1k, 2386 maxCycles / perLoop1k); 2387 2388 } 2389 } 2390#endif 2391}; 2392 2393void AudioFlinger::PlaybackThread::checkSilentMode_l() 2394{ 2395 if (!mMasterMute) { 2396 char value[PROPERTY_VALUE_MAX]; 2397 if (property_get("ro.audio.silent", value, "0") > 0) { 2398 char *endptr; 2399 unsigned long ul = strtoul(value, &endptr, 0); 2400 if (*endptr == '\0' && ul != 0) { 2401 ALOGD("Silence is golden"); 2402 // The setprop command will not allow a property to be changed after 2403 // the first time it is set, so we don't have to worry about un-muting. 2404 setMasterMute_l(true); 2405 } 2406 } 2407 } 2408} 2409 2410bool AudioFlinger::PlaybackThread::threadLoop() 2411{ 2412 Vector< sp<Track> > tracksToRemove; 2413 2414 standbyTime = systemTime(); 2415 2416 // MIXER 2417 nsecs_t lastWarning = 0; 2418if (mType == MIXER) { 2419 longStandbyExit = false; 2420} 2421 2422 // DUPLICATING 2423 // FIXME could this be made local to while loop? 2424 writeFrames = 0; 2425 2426 cacheParameters_l(); 2427 sleepTime = idleSleepTime; 2428 2429if (mType == MIXER) { 2430 sleepTimeShift = 0; 2431} 2432 2433 CpuStats cpuStats; 2434 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2435 2436 acquireWakeLock(); 2437 2438 while (!exitPending()) 2439 { 2440 cpuStats.sample(myName); 2441 2442 Vector< sp<EffectChain> > effectChains; 2443 2444 processConfigEvents(); 2445 2446 { // scope for mLock 2447 2448 Mutex::Autolock _l(mLock); 2449 2450 if (checkForNewParameters_l()) { 2451 cacheParameters_l(); 2452 } 2453 2454 saveOutputTracks(); 2455 2456 // put audio hardware into standby after short delay 2457 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2458 mSuspended > 0)) { 2459 if (!mStandby) { 2460 2461 threadLoop_standby(); 2462 2463 mStandby = true; 2464 mBytesWritten = 0; 2465 } 2466 2467 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2468 // we're about to wait, flush the binder command buffer 2469 IPCThreadState::self()->flushCommands(); 2470 2471 clearOutputTracks(); 2472 2473 if (exitPending()) break; 2474 2475 releaseWakeLock_l(); 2476 // wait until we have something to do... 2477 ALOGV("%s going to sleep", myName.string()); 2478 mWaitWorkCV.wait(mLock); 2479 ALOGV("%s waking up", myName.string()); 2480 acquireWakeLock_l(); 2481 2482 mMixerStatus = MIXER_IDLE; 2483 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2484 2485 checkSilentMode_l(); 2486 2487 standbyTime = systemTime() + standbyDelay; 2488 sleepTime = idleSleepTime; 2489 if (mType == MIXER) { 2490 sleepTimeShift = 0; 2491 } 2492 2493 continue; 2494 } 2495 } 2496 2497 // mMixerStatusIgnoringFastTracks is also updated internally 2498 mMixerStatus = prepareTracks_l(&tracksToRemove); 2499 2500 // prevent any changes in effect chain list and in each effect chain 2501 // during mixing and effect process as the audio buffers could be deleted 2502 // or modified if an effect is created or deleted 2503 lockEffectChains_l(effectChains); 2504 } 2505 2506 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2507 threadLoop_mix(); 2508 } else { 2509 threadLoop_sleepTime(); 2510 } 2511 2512 if (mSuspended > 0) { 2513 sleepTime = suspendSleepTimeUs(); 2514 } 2515 2516 // only process effects if we're going to write 2517 if (sleepTime == 0) { 2518 for (size_t i = 0; i < effectChains.size(); i ++) { 2519 effectChains[i]->process_l(); 2520 } 2521 } 2522 2523 // enable changes in effect chain 2524 unlockEffectChains(effectChains); 2525 2526 // sleepTime == 0 means we must write to audio hardware 2527 if (sleepTime == 0) { 2528 2529 threadLoop_write(); 2530 2531if (mType == MIXER) { 2532 // write blocked detection 2533 nsecs_t now = systemTime(); 2534 nsecs_t delta = now - mLastWriteTime; 2535 if (!mStandby && delta > maxPeriod) { 2536 mNumDelayedWrites++; 2537 if ((now - lastWarning) > kWarningThrottleNs) { 2538 ScopedTrace st(ATRACE_TAG, "underrun"); 2539 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2540 ns2ms(delta), mNumDelayedWrites, this); 2541 lastWarning = now; 2542 } 2543 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2544 // a different threshold. Or completely removed for what it is worth anyway... 2545 if (mStandby) { 2546 longStandbyExit = true; 2547 } 2548 } 2549} 2550 2551 mStandby = false; 2552 } else { 2553 usleep(sleepTime); 2554 } 2555 2556 // Finally let go of removed track(s), without the lock held 2557 // since we can't guarantee the destructors won't acquire that 2558 // same lock. This will also mutate and push a new fast mixer state. 2559 threadLoop_removeTracks(tracksToRemove); 2560 tracksToRemove.clear(); 2561 2562 // FIXME I don't understand the need for this here; 2563 // it was in the original code but maybe the 2564 // assignment in saveOutputTracks() makes this unnecessary? 2565 clearOutputTracks(); 2566 2567 // Effect chains will be actually deleted here if they were removed from 2568 // mEffectChains list during mixing or effects processing 2569 effectChains.clear(); 2570 2571 // FIXME Note that the above .clear() is no longer necessary since effectChains 2572 // is now local to this block, but will keep it for now (at least until merge done). 2573 } 2574 2575if (mType == MIXER || mType == DIRECT) { 2576 // put output stream into standby mode 2577 if (!mStandby) { 2578 mOutput->stream->common.standby(&mOutput->stream->common); 2579 } 2580} 2581if (mType == DUPLICATING) { 2582 // for DuplicatingThread, standby mode is handled by the outputTracks 2583} 2584 2585 releaseWakeLock(); 2586 2587 ALOGV("Thread %p type %d exiting", this, mType); 2588 return false; 2589} 2590 2591// returns (via tracksToRemove) a set of tracks to remove. 2592void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2593{ 2594 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2595} 2596 2597void AudioFlinger::MixerThread::threadLoop_write() 2598{ 2599 // FIXME we should only do one push per cycle; confirm this is true 2600 // Start the fast mixer if it's not already running 2601 if (mFastMixer != NULL) { 2602 FastMixerStateQueue *sq = mFastMixer->sq(); 2603 FastMixerState *state = sq->begin(); 2604 if (state->mCommand != FastMixerState::MIX_WRITE && 2605 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2606 if (state->mCommand == FastMixerState::COLD_IDLE) { 2607 int32_t old = android_atomic_inc(&mFastMixerFutex); 2608 if (old == -1) { 2609 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2610 } 2611 } 2612 state->mCommand = FastMixerState::MIX_WRITE; 2613 sq->end(); 2614 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2615 if (kUseFastMixer == FastMixer_Dynamic) { 2616 mNormalSink = mPipeSink; 2617 } 2618 } else { 2619 sq->end(false /*didModify*/); 2620 } 2621 } 2622 PlaybackThread::threadLoop_write(); 2623} 2624 2625// shared by MIXER and DIRECT, overridden by DUPLICATING 2626void AudioFlinger::PlaybackThread::threadLoop_write() 2627{ 2628 // FIXME rewrite to reduce number of system calls 2629 mLastWriteTime = systemTime(); 2630 mInWrite = true; 2631 2632#define mBitShift 2 // FIXME 2633 size_t count = mixBufferSize >> mBitShift; 2634 Tracer::traceBegin(ATRACE_TAG, "write"); 2635 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2636 Tracer::traceEnd(ATRACE_TAG); 2637 if (framesWritten > 0) { 2638 size_t bytesWritten = framesWritten << mBitShift; 2639 mBytesWritten += bytesWritten; 2640 } 2641 2642 mNumWrites++; 2643 mInWrite = false; 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_standby() 2647{ 2648 // Idle the fast mixer if it's currently running 2649 if (mFastMixer != NULL) { 2650 FastMixerStateQueue *sq = mFastMixer->sq(); 2651 FastMixerState *state = sq->begin(); 2652 if (!(state->mCommand & FastMixerState::IDLE)) { 2653 state->mCommand = FastMixerState::COLD_IDLE; 2654 state->mColdFutexAddr = &mFastMixerFutex; 2655 state->mColdGen++; 2656 mFastMixerFutex = 0; 2657 sq->end(); 2658 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2659 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2660 if (kUseFastMixer == FastMixer_Dynamic) { 2661 mNormalSink = mOutputSink; 2662 } 2663 } else { 2664 sq->end(false /*didModify*/); 2665 } 2666 } 2667 PlaybackThread::threadLoop_standby(); 2668} 2669 2670// shared by MIXER and DIRECT, overridden by DUPLICATING 2671void AudioFlinger::PlaybackThread::threadLoop_standby() 2672{ 2673 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2674 mOutput->stream->common.standby(&mOutput->stream->common); 2675} 2676 2677void AudioFlinger::MixerThread::threadLoop_mix() 2678{ 2679 // obtain the presentation timestamp of the next output buffer 2680 int64_t pts; 2681 status_t status = INVALID_OPERATION; 2682 2683 if (NULL != mOutput->stream->get_next_write_timestamp) { 2684 status = mOutput->stream->get_next_write_timestamp( 2685 mOutput->stream, &pts); 2686 } 2687 2688 if (status != NO_ERROR) { 2689 pts = AudioBufferProvider::kInvalidPTS; 2690 } 2691 2692 // mix buffers... 2693 mAudioMixer->process(pts); 2694 // increase sleep time progressively when application underrun condition clears. 2695 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2696 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2697 // such that we would underrun the audio HAL. 2698 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2699 sleepTimeShift--; 2700 } 2701 sleepTime = 0; 2702 standbyTime = systemTime() + standbyDelay; 2703 //TODO: delay standby when effects have a tail 2704} 2705 2706void AudioFlinger::MixerThread::threadLoop_sleepTime() 2707{ 2708 // If no tracks are ready, sleep once for the duration of an output 2709 // buffer size, then write 0s to the output 2710 if (sleepTime == 0) { 2711 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2712 sleepTime = activeSleepTime >> sleepTimeShift; 2713 if (sleepTime < kMinThreadSleepTimeUs) { 2714 sleepTime = kMinThreadSleepTimeUs; 2715 } 2716 // reduce sleep time in case of consecutive application underruns to avoid 2717 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2718 // duration we would end up writing less data than needed by the audio HAL if 2719 // the condition persists. 2720 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2721 sleepTimeShift++; 2722 } 2723 } else { 2724 sleepTime = idleSleepTime; 2725 } 2726 } else if (mBytesWritten != 0 || 2727 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2728 memset (mMixBuffer, 0, mixBufferSize); 2729 sleepTime = 0; 2730 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2731 } 2732 // TODO add standby time extension fct of effect tail 2733} 2734 2735// prepareTracks_l() must be called with ThreadBase::mLock held 2736AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2737 Vector< sp<Track> > *tracksToRemove) 2738{ 2739 2740 mixer_state mixerStatus = MIXER_IDLE; 2741 // find out which tracks need to be processed 2742 size_t count = mActiveTracks.size(); 2743 size_t mixedTracks = 0; 2744 size_t tracksWithEffect = 0; 2745 // counts only _active_ fast tracks 2746 size_t fastTracks = 0; 2747 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2748 2749 float masterVolume = mMasterVolume; 2750 bool masterMute = mMasterMute; 2751 2752 if (masterMute) { 2753 masterVolume = 0; 2754 } 2755 // Delegate master volume control to effect in output mix effect chain if needed 2756 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2757 if (chain != 0) { 2758 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2759 chain->setVolume_l(&v, &v); 2760 masterVolume = (float)((v + (1 << 23)) >> 24); 2761 chain.clear(); 2762 } 2763 2764 // prepare a new state to push 2765 FastMixerStateQueue *sq = NULL; 2766 FastMixerState *state = NULL; 2767 bool didModify = false; 2768 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2769 if (mFastMixer != NULL) { 2770 sq = mFastMixer->sq(); 2771 state = sq->begin(); 2772 } 2773 2774 for (size_t i=0 ; i<count ; i++) { 2775 sp<Track> t = mActiveTracks[i].promote(); 2776 if (t == 0) continue; 2777 2778 // this const just means the local variable doesn't change 2779 Track* const track = t.get(); 2780 2781 // process fast tracks 2782 if (track->isFastTrack()) { 2783 2784 // It's theoretically possible (though unlikely) for a fast track to be created 2785 // and then removed within the same normal mix cycle. This is not a problem, as 2786 // the track never becomes active so it's fast mixer slot is never touched. 2787 // The converse, of removing an (active) track and then creating a new track 2788 // at the identical fast mixer slot within the same normal mix cycle, 2789 // is impossible because the slot isn't marked available until the end of each cycle. 2790 int j = track->mFastIndex; 2791 FastTrack *fastTrack = &state->mFastTracks[j]; 2792 2793 // Determine whether the track is currently in underrun condition, 2794 // and whether it had a recent underrun. 2795 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2796 uint32_t recentFull = (underruns.mBitFields.mFull - 2797 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2798 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2799 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2800 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2801 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2802 uint32_t recentUnderruns = recentPartial + recentEmpty; 2803 track->mObservedUnderruns = underruns; 2804 // don't count underruns that occur while stopping or pausing 2805 // or stopped which can occur when flush() is called while active 2806 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2807 track->mUnderrunCount += recentUnderruns; 2808 } 2809 2810 // This is similar to the state machine for normal tracks, 2811 // with a few modifications for fast tracks. 2812 bool isActive = true; 2813 switch (track->mState) { 2814 case TrackBase::STOPPING_1: 2815 // track stays active in STOPPING_1 state until first underrun 2816 if (recentUnderruns > 0) { 2817 track->mState = TrackBase::STOPPING_2; 2818 } 2819 break; 2820 case TrackBase::PAUSING: 2821 // ramp down is not yet implemented 2822 track->setPaused(); 2823 break; 2824 case TrackBase::RESUMING: 2825 // ramp up is not yet implemented 2826 track->mState = TrackBase::ACTIVE; 2827 break; 2828 case TrackBase::ACTIVE: 2829 if (recentFull > 0 || recentPartial > 0) { 2830 // track has provided at least some frames recently: reset retry count 2831 track->mRetryCount = kMaxTrackRetries; 2832 } 2833 if (recentUnderruns == 0) { 2834 // no recent underruns: stay active 2835 break; 2836 } 2837 // there has recently been an underrun of some kind 2838 if (track->sharedBuffer() == 0) { 2839 // were any of the recent underruns "empty" (no frames available)? 2840 if (recentEmpty == 0) { 2841 // no, then ignore the partial underruns as they are allowed indefinitely 2842 break; 2843 } 2844 // there has recently been an "empty" underrun: decrement the retry counter 2845 if (--(track->mRetryCount) > 0) { 2846 break; 2847 } 2848 // indicate to client process that the track was disabled because of underrun; 2849 // it will then automatically call start() when data is available 2850 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2851 // remove from active list, but state remains ACTIVE [confusing but true] 2852 isActive = false; 2853 break; 2854 } 2855 // fall through 2856 case TrackBase::STOPPING_2: 2857 case TrackBase::PAUSED: 2858 case TrackBase::TERMINATED: 2859 case TrackBase::STOPPED: 2860 case TrackBase::FLUSHED: // flush() while active 2861 // Check for presentation complete if track is inactive 2862 // We have consumed all the buffers of this track. 2863 // This would be incomplete if we auto-paused on underrun 2864 { 2865 size_t audioHALFrames = 2866 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2867 size_t framesWritten = 2868 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2869 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2870 // track stays in active list until presentation is complete 2871 break; 2872 } 2873 } 2874 if (track->isStopping_2()) { 2875 track->mState = TrackBase::STOPPED; 2876 } 2877 if (track->isStopped()) { 2878 // Can't reset directly, as fast mixer is still polling this track 2879 // track->reset(); 2880 // So instead mark this track as needing to be reset after push with ack 2881 resetMask |= 1 << i; 2882 } 2883 isActive = false; 2884 break; 2885 case TrackBase::IDLE: 2886 default: 2887 LOG_FATAL("unexpected track state %d", track->mState); 2888 } 2889 2890 if (isActive) { 2891 // was it previously inactive? 2892 if (!(state->mTrackMask & (1 << j))) { 2893 ExtendedAudioBufferProvider *eabp = track; 2894 VolumeProvider *vp = track; 2895 fastTrack->mBufferProvider = eabp; 2896 fastTrack->mVolumeProvider = vp; 2897 fastTrack->mSampleRate = track->mSampleRate; 2898 fastTrack->mChannelMask = track->mChannelMask; 2899 fastTrack->mGeneration++; 2900 state->mTrackMask |= 1 << j; 2901 didModify = true; 2902 // no acknowledgement required for newly active tracks 2903 } 2904 // cache the combined master volume and stream type volume for fast mixer; this 2905 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2906 track->mCachedVolume = track->isMuted() ? 2907 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2908 ++fastTracks; 2909 } else { 2910 // was it previously active? 2911 if (state->mTrackMask & (1 << j)) { 2912 fastTrack->mBufferProvider = NULL; 2913 fastTrack->mGeneration++; 2914 state->mTrackMask &= ~(1 << j); 2915 didModify = true; 2916 // If any fast tracks were removed, we must wait for acknowledgement 2917 // because we're about to decrement the last sp<> on those tracks. 2918 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2919 } else { 2920 LOG_FATAL("fast track %d should have been active", j); 2921 } 2922 tracksToRemove->add(track); 2923 // Avoids a misleading display in dumpsys 2924 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2925 } 2926 continue; 2927 } 2928 2929 { // local variable scope to avoid goto warning 2930 2931 audio_track_cblk_t* cblk = track->cblk(); 2932 2933 // The first time a track is added we wait 2934 // for all its buffers to be filled before processing it 2935 int name = track->name(); 2936 // make sure that we have enough frames to mix one full buffer. 2937 // enforce this condition only once to enable draining the buffer in case the client 2938 // app does not call stop() and relies on underrun to stop: 2939 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2940 // during last round 2941 uint32_t minFrames = 1; 2942 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2943 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2944 if (t->sampleRate() == (int)mSampleRate) { 2945 minFrames = mNormalFrameCount; 2946 } else { 2947 // +1 for rounding and +1 for additional sample needed for interpolation 2948 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2949 // add frames already consumed but not yet released by the resampler 2950 // because cblk->framesReady() will include these frames 2951 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2952 // the minimum track buffer size is normally twice the number of frames necessary 2953 // to fill one buffer and the resampler should not leave more than one buffer worth 2954 // of unreleased frames after each pass, but just in case... 2955 ALOG_ASSERT(minFrames <= cblk->frameCount); 2956 } 2957 } 2958 if ((track->framesReady() >= minFrames) && track->isReady() && 2959 !track->isPaused() && !track->isTerminated()) 2960 { 2961 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2962 2963 mixedTracks++; 2964 2965 // track->mainBuffer() != mMixBuffer means there is an effect chain 2966 // connected to the track 2967 chain.clear(); 2968 if (track->mainBuffer() != mMixBuffer) { 2969 chain = getEffectChain_l(track->sessionId()); 2970 // Delegate volume control to effect in track effect chain if needed 2971 if (chain != 0) { 2972 tracksWithEffect++; 2973 } else { 2974 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2975 name, track->sessionId()); 2976 } 2977 } 2978 2979 2980 int param = AudioMixer::VOLUME; 2981 if (track->mFillingUpStatus == Track::FS_FILLED) { 2982 // no ramp for the first volume setting 2983 track->mFillingUpStatus = Track::FS_ACTIVE; 2984 if (track->mState == TrackBase::RESUMING) { 2985 track->mState = TrackBase::ACTIVE; 2986 param = AudioMixer::RAMP_VOLUME; 2987 } 2988 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2989 } else if (cblk->server != 0) { 2990 // If the track is stopped before the first frame was mixed, 2991 // do not apply ramp 2992 param = AudioMixer::RAMP_VOLUME; 2993 } 2994 2995 // compute volume for this track 2996 uint32_t vl, vr, va; 2997 if (track->isMuted() || track->isPausing() || 2998 mStreamTypes[track->streamType()].mute) { 2999 vl = vr = va = 0; 3000 if (track->isPausing()) { 3001 track->setPaused(); 3002 } 3003 } else { 3004 3005 // read original volumes with volume control 3006 float typeVolume = mStreamTypes[track->streamType()].volume; 3007 float v = masterVolume * typeVolume; 3008 uint32_t vlr = cblk->getVolumeLR(); 3009 vl = vlr & 0xFFFF; 3010 vr = vlr >> 16; 3011 // track volumes come from shared memory, so can't be trusted and must be clamped 3012 if (vl > MAX_GAIN_INT) { 3013 ALOGV("Track left volume out of range: %04X", vl); 3014 vl = MAX_GAIN_INT; 3015 } 3016 if (vr > MAX_GAIN_INT) { 3017 ALOGV("Track right volume out of range: %04X", vr); 3018 vr = MAX_GAIN_INT; 3019 } 3020 // now apply the master volume and stream type volume 3021 vl = (uint32_t)(v * vl) << 12; 3022 vr = (uint32_t)(v * vr) << 12; 3023 // assuming master volume and stream type volume each go up to 1.0, 3024 // vl and vr are now in 8.24 format 3025 3026 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3027 // send level comes from shared memory and so may be corrupt 3028 if (sendLevel > MAX_GAIN_INT) { 3029 ALOGV("Track send level out of range: %04X", sendLevel); 3030 sendLevel = MAX_GAIN_INT; 3031 } 3032 va = (uint32_t)(v * sendLevel); 3033 } 3034 // Delegate volume control to effect in track effect chain if needed 3035 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3036 // Do not ramp volume if volume is controlled by effect 3037 param = AudioMixer::VOLUME; 3038 track->mHasVolumeController = true; 3039 } else { 3040 // force no volume ramp when volume controller was just disabled or removed 3041 // from effect chain to avoid volume spike 3042 if (track->mHasVolumeController) { 3043 param = AudioMixer::VOLUME; 3044 } 3045 track->mHasVolumeController = false; 3046 } 3047 3048 // Convert volumes from 8.24 to 4.12 format 3049 // This additional clamping is needed in case chain->setVolume_l() overshot 3050 vl = (vl + (1 << 11)) >> 12; 3051 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3052 vr = (vr + (1 << 11)) >> 12; 3053 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3054 3055 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3056 3057 // XXX: these things DON'T need to be done each time 3058 mAudioMixer->setBufferProvider(name, track); 3059 mAudioMixer->enable(name); 3060 3061 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3062 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3063 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3064 mAudioMixer->setParameter( 3065 name, 3066 AudioMixer::TRACK, 3067 AudioMixer::FORMAT, (void *)track->format()); 3068 mAudioMixer->setParameter( 3069 name, 3070 AudioMixer::TRACK, 3071 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3072 mAudioMixer->setParameter( 3073 name, 3074 AudioMixer::RESAMPLE, 3075 AudioMixer::SAMPLE_RATE, 3076 (void *)(cblk->sampleRate)); 3077 mAudioMixer->setParameter( 3078 name, 3079 AudioMixer::TRACK, 3080 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3081 mAudioMixer->setParameter( 3082 name, 3083 AudioMixer::TRACK, 3084 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3085 3086 // reset retry count 3087 track->mRetryCount = kMaxTrackRetries; 3088 3089 // If one track is ready, set the mixer ready if: 3090 // - the mixer was not ready during previous round OR 3091 // - no other track is not ready 3092 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3093 mixerStatus != MIXER_TRACKS_ENABLED) { 3094 mixerStatus = MIXER_TRACKS_READY; 3095 } 3096 } else { 3097 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3098 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3099 track->isStopped() || track->isPaused()) { 3100 // We have consumed all the buffers of this track. 3101 // Remove it from the list of active tracks. 3102 // TODO: use actual buffer filling status instead of latency when available from 3103 // audio HAL 3104 size_t audioHALFrames = 3105 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3106 size_t framesWritten = 3107 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3108 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3109 if (track->isStopped()) { 3110 track->reset(); 3111 } 3112 tracksToRemove->add(track); 3113 } 3114 } else { 3115 // No buffers for this track. Give it a few chances to 3116 // fill a buffer, then remove it from active list. 3117 if (--(track->mRetryCount) <= 0) { 3118 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3119 tracksToRemove->add(track); 3120 // indicate to client process that the track was disabled because of underrun; 3121 // it will then automatically call start() when data is available 3122 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3123 // If one track is not ready, mark the mixer also not ready if: 3124 // - the mixer was ready during previous round OR 3125 // - no other track is ready 3126 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3127 mixerStatus != MIXER_TRACKS_READY) { 3128 mixerStatus = MIXER_TRACKS_ENABLED; 3129 } 3130 } 3131 mAudioMixer->disable(name); 3132 } 3133 3134 } // local variable scope to avoid goto warning 3135track_is_ready: ; 3136 3137 } 3138 3139 // Push the new FastMixer state if necessary 3140 if (didModify) { 3141 state->mFastTracksGen++; 3142 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3143 if (kUseFastMixer == FastMixer_Dynamic && 3144 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3145 state->mCommand = FastMixerState::COLD_IDLE; 3146 state->mColdFutexAddr = &mFastMixerFutex; 3147 state->mColdGen++; 3148 mFastMixerFutex = 0; 3149 if (kUseFastMixer == FastMixer_Dynamic) { 3150 mNormalSink = mOutputSink; 3151 } 3152 // If we go into cold idle, need to wait for acknowledgement 3153 // so that fast mixer stops doing I/O. 3154 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3155 } 3156 sq->end(); 3157 } 3158 if (sq != NULL) { 3159 sq->end(didModify); 3160 sq->push(block); 3161 } 3162 3163 // Now perform the deferred reset on fast tracks that have stopped 3164 while (resetMask != 0) { 3165 size_t i = __builtin_ctz(resetMask); 3166 ALOG_ASSERT(i < count); 3167 resetMask &= ~(1 << i); 3168 sp<Track> t = mActiveTracks[i].promote(); 3169 if (t == 0) continue; 3170 Track* track = t.get(); 3171 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3172 track->reset(); 3173 } 3174 3175 // remove all the tracks that need to be... 3176 count = tracksToRemove->size(); 3177 if (CC_UNLIKELY(count)) { 3178 for (size_t i=0 ; i<count ; i++) { 3179 const sp<Track>& track = tracksToRemove->itemAt(i); 3180 mActiveTracks.remove(track); 3181 if (track->mainBuffer() != mMixBuffer) { 3182 chain = getEffectChain_l(track->sessionId()); 3183 if (chain != 0) { 3184 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3185 chain->decActiveTrackCnt(); 3186 } 3187 } 3188 if (track->isTerminated()) { 3189 removeTrack_l(track); 3190 } 3191 } 3192 } 3193 3194 // mix buffer must be cleared if all tracks are connected to an 3195 // effect chain as in this case the mixer will not write to 3196 // mix buffer and track effects will accumulate into it 3197 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3198 // FIXME as a performance optimization, should remember previous zero status 3199 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3200 } 3201 3202 // if any fast tracks, then status is ready 3203 mMixerStatusIgnoringFastTracks = mixerStatus; 3204 if (fastTracks > 0) { 3205 mixerStatus = MIXER_TRACKS_READY; 3206 } 3207 return mixerStatus; 3208} 3209 3210/* 3211The derived values that are cached: 3212 - mixBufferSize from frame count * frame size 3213 - activeSleepTime from activeSleepTimeUs() 3214 - idleSleepTime from idleSleepTimeUs() 3215 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3216 - maxPeriod from frame count and sample rate (MIXER only) 3217 3218The parameters that affect these derived values are: 3219 - frame count 3220 - frame size 3221 - sample rate 3222 - device type: A2DP or not 3223 - device latency 3224 - format: PCM or not 3225 - active sleep time 3226 - idle sleep time 3227*/ 3228 3229void AudioFlinger::PlaybackThread::cacheParameters_l() 3230{ 3231 mixBufferSize = mNormalFrameCount * mFrameSize; 3232 activeSleepTime = activeSleepTimeUs(); 3233 idleSleepTime = idleSleepTimeUs(); 3234} 3235 3236void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3237{ 3238 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3239 this, streamType, mTracks.size()); 3240 Mutex::Autolock _l(mLock); 3241 3242 size_t size = mTracks.size(); 3243 for (size_t i = 0; i < size; i++) { 3244 sp<Track> t = mTracks[i]; 3245 if (t->streamType() == streamType) { 3246 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3247 t->mCblk->cv.signal(); 3248 } 3249 } 3250} 3251 3252// getTrackName_l() must be called with ThreadBase::mLock held 3253int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3254{ 3255 return mAudioMixer->getTrackName(channelMask); 3256} 3257 3258// deleteTrackName_l() must be called with ThreadBase::mLock held 3259void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3260{ 3261 ALOGV("remove track (%d) and delete from mixer", name); 3262 mAudioMixer->deleteTrackName(name); 3263} 3264 3265// checkForNewParameters_l() must be called with ThreadBase::mLock held 3266bool AudioFlinger::MixerThread::checkForNewParameters_l() 3267{ 3268 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3269 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3270 bool reconfig = false; 3271 3272 while (!mNewParameters.isEmpty()) { 3273 3274 if (mFastMixer != NULL) { 3275 FastMixerStateQueue *sq = mFastMixer->sq(); 3276 FastMixerState *state = sq->begin(); 3277 if (!(state->mCommand & FastMixerState::IDLE)) { 3278 previousCommand = state->mCommand; 3279 state->mCommand = FastMixerState::HOT_IDLE; 3280 sq->end(); 3281 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3282 } else { 3283 sq->end(false /*didModify*/); 3284 } 3285 } 3286 3287 status_t status = NO_ERROR; 3288 String8 keyValuePair = mNewParameters[0]; 3289 AudioParameter param = AudioParameter(keyValuePair); 3290 int value; 3291 3292 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3293 reconfig = true; 3294 } 3295 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3296 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3297 status = BAD_VALUE; 3298 } else { 3299 reconfig = true; 3300 } 3301 } 3302 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3303 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3304 status = BAD_VALUE; 3305 } else { 3306 reconfig = true; 3307 } 3308 } 3309 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3310 // do not accept frame count changes if tracks are open as the track buffer 3311 // size depends on frame count and correct behavior would not be guaranteed 3312 // if frame count is changed after track creation 3313 if (!mTracks.isEmpty()) { 3314 status = INVALID_OPERATION; 3315 } else { 3316 reconfig = true; 3317 } 3318 } 3319 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3320#ifdef ADD_BATTERY_DATA 3321 // when changing the audio output device, call addBatteryData to notify 3322 // the change 3323 if ((int)mDevice != value) { 3324 uint32_t params = 0; 3325 // check whether speaker is on 3326 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3327 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3328 } 3329 3330 int deviceWithoutSpeaker 3331 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3332 // check if any other device (except speaker) is on 3333 if (value & deviceWithoutSpeaker ) { 3334 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3335 } 3336 3337 if (params != 0) { 3338 addBatteryData(params); 3339 } 3340 } 3341#endif 3342 3343 // forward device change to effects that have requested to be 3344 // aware of attached audio device. 3345 mDevice = (uint32_t)value; 3346 for (size_t i = 0; i < mEffectChains.size(); i++) { 3347 mEffectChains[i]->setDevice_l(mDevice); 3348 } 3349 } 3350 3351 if (status == NO_ERROR) { 3352 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3353 keyValuePair.string()); 3354 if (!mStandby && status == INVALID_OPERATION) { 3355 mOutput->stream->common.standby(&mOutput->stream->common); 3356 mStandby = true; 3357 mBytesWritten = 0; 3358 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3359 keyValuePair.string()); 3360 } 3361 if (status == NO_ERROR && reconfig) { 3362 delete mAudioMixer; 3363 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3364 mAudioMixer = NULL; 3365 readOutputParameters(); 3366 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3367 for (size_t i = 0; i < mTracks.size() ; i++) { 3368 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3369 if (name < 0) break; 3370 mTracks[i]->mName = name; 3371 // limit track sample rate to 2 x new output sample rate 3372 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3373 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3374 } 3375 } 3376 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3377 } 3378 } 3379 3380 mNewParameters.removeAt(0); 3381 3382 mParamStatus = status; 3383 mParamCond.signal(); 3384 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3385 // already timed out waiting for the status and will never signal the condition. 3386 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3387 } 3388 3389 if (!(previousCommand & FastMixerState::IDLE)) { 3390 ALOG_ASSERT(mFastMixer != NULL); 3391 FastMixerStateQueue *sq = mFastMixer->sq(); 3392 FastMixerState *state = sq->begin(); 3393 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3394 state->mCommand = previousCommand; 3395 sq->end(); 3396 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3397 } 3398 3399 return reconfig; 3400} 3401 3402status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3403{ 3404 const size_t SIZE = 256; 3405 char buffer[SIZE]; 3406 String8 result; 3407 3408 PlaybackThread::dumpInternals(fd, args); 3409 3410 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3411 result.append(buffer); 3412 write(fd, result.string(), result.size()); 3413 3414 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3415 FastMixerDumpState copy = mFastMixerDumpState; 3416 copy.dump(fd); 3417 3418 return NO_ERROR; 3419} 3420 3421uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3422{ 3423 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3424} 3425 3426uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3427{ 3428 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3429} 3430 3431void AudioFlinger::MixerThread::cacheParameters_l() 3432{ 3433 PlaybackThread::cacheParameters_l(); 3434 3435 // FIXME: Relaxed timing because of a certain device that can't meet latency 3436 // Should be reduced to 2x after the vendor fixes the driver issue 3437 // increase threshold again due to low power audio mode. The way this warning 3438 // threshold is calculated and its usefulness should be reconsidered anyway. 3439 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3440} 3441 3442// ---------------------------------------------------------------------------- 3443AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3444 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3445 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3446 // mLeftVolFloat, mRightVolFloat 3447 // mLeftVolShort, mRightVolShort 3448{ 3449} 3450 3451AudioFlinger::DirectOutputThread::~DirectOutputThread() 3452{ 3453} 3454 3455AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3456 Vector< sp<Track> > *tracksToRemove 3457) 3458{ 3459 sp<Track> trackToRemove; 3460 3461 mixer_state mixerStatus = MIXER_IDLE; 3462 3463 // find out which tracks need to be processed 3464 if (mActiveTracks.size() != 0) { 3465 sp<Track> t = mActiveTracks[0].promote(); 3466 // The track died recently 3467 if (t == 0) return MIXER_IDLE; 3468 3469 Track* const track = t.get(); 3470 audio_track_cblk_t* cblk = track->cblk(); 3471 3472 // The first time a track is added we wait 3473 // for all its buffers to be filled before processing it 3474 if (cblk->framesReady() && track->isReady() && 3475 !track->isPaused() && !track->isTerminated()) 3476 { 3477 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3478 3479 if (track->mFillingUpStatus == Track::FS_FILLED) { 3480 track->mFillingUpStatus = Track::FS_ACTIVE; 3481 mLeftVolFloat = mRightVolFloat = 0; 3482 mLeftVolShort = mRightVolShort = 0; 3483 if (track->mState == TrackBase::RESUMING) { 3484 track->mState = TrackBase::ACTIVE; 3485 rampVolume = true; 3486 } 3487 } else if (cblk->server != 0) { 3488 // If the track is stopped before the first frame was mixed, 3489 // do not apply ramp 3490 rampVolume = true; 3491 } 3492 // compute volume for this track 3493 float left, right; 3494 if (track->isMuted() || mMasterMute || track->isPausing() || 3495 mStreamTypes[track->streamType()].mute) { 3496 left = right = 0; 3497 if (track->isPausing()) { 3498 track->setPaused(); 3499 } 3500 } else { 3501 float typeVolume = mStreamTypes[track->streamType()].volume; 3502 float v = mMasterVolume * typeVolume; 3503 uint32_t vlr = cblk->getVolumeLR(); 3504 float v_clamped = v * (vlr & 0xFFFF); 3505 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3506 left = v_clamped/MAX_GAIN; 3507 v_clamped = v * (vlr >> 16); 3508 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3509 right = v_clamped/MAX_GAIN; 3510 } 3511 3512 if (left != mLeftVolFloat || right != mRightVolFloat) { 3513 mLeftVolFloat = left; 3514 mRightVolFloat = right; 3515 3516 // If audio HAL implements volume control, 3517 // force software volume to nominal value 3518 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3519 left = 1.0f; 3520 right = 1.0f; 3521 } 3522 3523 // Convert volumes from float to 8.24 3524 uint32_t vl = (uint32_t)(left * (1 << 24)); 3525 uint32_t vr = (uint32_t)(right * (1 << 24)); 3526 3527 // Delegate volume control to effect in track effect chain if needed 3528 // only one effect chain can be present on DirectOutputThread, so if 3529 // there is one, the track is connected to it 3530 if (!mEffectChains.isEmpty()) { 3531 // Do not ramp volume if volume is controlled by effect 3532 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3533 rampVolume = false; 3534 } 3535 } 3536 3537 // Convert volumes from 8.24 to 4.12 format 3538 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3539 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3540 leftVol = (uint16_t)v_clamped; 3541 v_clamped = (vr + (1 << 11)) >> 12; 3542 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3543 rightVol = (uint16_t)v_clamped; 3544 } else { 3545 leftVol = mLeftVolShort; 3546 rightVol = mRightVolShort; 3547 rampVolume = false; 3548 } 3549 3550 // reset retry count 3551 track->mRetryCount = kMaxTrackRetriesDirect; 3552 mActiveTrack = t; 3553 mixerStatus = MIXER_TRACKS_READY; 3554 } else { 3555 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3556 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3557 // We have consumed all the buffers of this track. 3558 // Remove it from the list of active tracks. 3559 // TODO: implement behavior for compressed audio 3560 size_t audioHALFrames = 3561 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3562 size_t framesWritten = 3563 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3564 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3565 if (track->isStopped()) { 3566 track->reset(); 3567 } 3568 trackToRemove = track; 3569 } 3570 } else { 3571 // No buffers for this track. Give it a few chances to 3572 // fill a buffer, then remove it from active list. 3573 if (--(track->mRetryCount) <= 0) { 3574 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3575 trackToRemove = track; 3576 } else { 3577 mixerStatus = MIXER_TRACKS_ENABLED; 3578 } 3579 } 3580 } 3581 } 3582 3583 // FIXME merge this with similar code for removing multiple tracks 3584 // remove all the tracks that need to be... 3585 if (CC_UNLIKELY(trackToRemove != 0)) { 3586 tracksToRemove->add(trackToRemove); 3587 mActiveTracks.remove(trackToRemove); 3588 if (!mEffectChains.isEmpty()) { 3589 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3590 trackToRemove->sessionId()); 3591 mEffectChains[0]->decActiveTrackCnt(); 3592 } 3593 if (trackToRemove->isTerminated()) { 3594 removeTrack_l(trackToRemove); 3595 } 3596 } 3597 3598 return mixerStatus; 3599} 3600 3601void AudioFlinger::DirectOutputThread::threadLoop_mix() 3602{ 3603 AudioBufferProvider::Buffer buffer; 3604 size_t frameCount = mFrameCount; 3605 int8_t *curBuf = (int8_t *)mMixBuffer; 3606 // output audio to hardware 3607 while (frameCount) { 3608 buffer.frameCount = frameCount; 3609 mActiveTrack->getNextBuffer(&buffer); 3610 if (CC_UNLIKELY(buffer.raw == NULL)) { 3611 memset(curBuf, 0, frameCount * mFrameSize); 3612 break; 3613 } 3614 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3615 frameCount -= buffer.frameCount; 3616 curBuf += buffer.frameCount * mFrameSize; 3617 mActiveTrack->releaseBuffer(&buffer); 3618 } 3619 sleepTime = 0; 3620 standbyTime = systemTime() + standbyDelay; 3621 mActiveTrack.clear(); 3622 3623 // apply volume 3624 3625 // Do not apply volume on compressed audio 3626 if (!audio_is_linear_pcm(mFormat)) { 3627 return; 3628 } 3629 3630 // convert to signed 16 bit before volume calculation 3631 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3632 size_t count = mFrameCount * mChannelCount; 3633 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3634 int16_t *dst = mMixBuffer + count-1; 3635 while (count--) { 3636 *dst-- = (int16_t)(*src--^0x80) << 8; 3637 } 3638 } 3639 3640 frameCount = mFrameCount; 3641 int16_t *out = mMixBuffer; 3642 if (rampVolume) { 3643 if (mChannelCount == 1) { 3644 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3645 int32_t vlInc = d / (int32_t)frameCount; 3646 int32_t vl = ((int32_t)mLeftVolShort << 16); 3647 do { 3648 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3649 out++; 3650 vl += vlInc; 3651 } while (--frameCount); 3652 3653 } else { 3654 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3655 int32_t vlInc = d / (int32_t)frameCount; 3656 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3657 int32_t vrInc = d / (int32_t)frameCount; 3658 int32_t vl = ((int32_t)mLeftVolShort << 16); 3659 int32_t vr = ((int32_t)mRightVolShort << 16); 3660 do { 3661 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3662 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3663 out += 2; 3664 vl += vlInc; 3665 vr += vrInc; 3666 } while (--frameCount); 3667 } 3668 } else { 3669 if (mChannelCount == 1) { 3670 do { 3671 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3672 out++; 3673 } while (--frameCount); 3674 } else { 3675 do { 3676 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3677 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3678 out += 2; 3679 } while (--frameCount); 3680 } 3681 } 3682 3683 // convert back to unsigned 8 bit after volume calculation 3684 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3685 size_t count = mFrameCount * mChannelCount; 3686 int16_t *src = mMixBuffer; 3687 uint8_t *dst = (uint8_t *)mMixBuffer; 3688 while (count--) { 3689 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3690 } 3691 } 3692 3693 mLeftVolShort = leftVol; 3694 mRightVolShort = rightVol; 3695} 3696 3697void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3698{ 3699 if (sleepTime == 0) { 3700 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3701 sleepTime = activeSleepTime; 3702 } else { 3703 sleepTime = idleSleepTime; 3704 } 3705 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3706 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3707 sleepTime = 0; 3708 } 3709} 3710 3711// getTrackName_l() must be called with ThreadBase::mLock held 3712int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3713{ 3714 return 0; 3715} 3716 3717// deleteTrackName_l() must be called with ThreadBase::mLock held 3718void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3719{ 3720} 3721 3722// checkForNewParameters_l() must be called with ThreadBase::mLock held 3723bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3724{ 3725 bool reconfig = false; 3726 3727 while (!mNewParameters.isEmpty()) { 3728 status_t status = NO_ERROR; 3729 String8 keyValuePair = mNewParameters[0]; 3730 AudioParameter param = AudioParameter(keyValuePair); 3731 int value; 3732 3733 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3734 // do not accept frame count changes if tracks are open as the track buffer 3735 // size depends on frame count and correct behavior would not be garantied 3736 // if frame count is changed after track creation 3737 if (!mTracks.isEmpty()) { 3738 status = INVALID_OPERATION; 3739 } else { 3740 reconfig = true; 3741 } 3742 } 3743 if (status == NO_ERROR) { 3744 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3745 keyValuePair.string()); 3746 if (!mStandby && status == INVALID_OPERATION) { 3747 mOutput->stream->common.standby(&mOutput->stream->common); 3748 mStandby = true; 3749 mBytesWritten = 0; 3750 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3751 keyValuePair.string()); 3752 } 3753 if (status == NO_ERROR && reconfig) { 3754 readOutputParameters(); 3755 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3756 } 3757 } 3758 3759 mNewParameters.removeAt(0); 3760 3761 mParamStatus = status; 3762 mParamCond.signal(); 3763 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3764 // already timed out waiting for the status and will never signal the condition. 3765 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3766 } 3767 return reconfig; 3768} 3769 3770uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3771{ 3772 uint32_t time; 3773 if (audio_is_linear_pcm(mFormat)) { 3774 time = PlaybackThread::activeSleepTimeUs(); 3775 } else { 3776 time = 10000; 3777 } 3778 return time; 3779} 3780 3781uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3782{ 3783 uint32_t time; 3784 if (audio_is_linear_pcm(mFormat)) { 3785 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3786 } else { 3787 time = 10000; 3788 } 3789 return time; 3790} 3791 3792uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3793{ 3794 uint32_t time; 3795 if (audio_is_linear_pcm(mFormat)) { 3796 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3797 } else { 3798 time = 10000; 3799 } 3800 return time; 3801} 3802 3803void AudioFlinger::DirectOutputThread::cacheParameters_l() 3804{ 3805 PlaybackThread::cacheParameters_l(); 3806 3807 // use shorter standby delay as on normal output to release 3808 // hardware resources as soon as possible 3809 standbyDelay = microseconds(activeSleepTime*2); 3810} 3811 3812// ---------------------------------------------------------------------------- 3813 3814AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3815 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3816 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3817 mWaitTimeMs(UINT_MAX) 3818{ 3819 addOutputTrack(mainThread); 3820} 3821 3822AudioFlinger::DuplicatingThread::~DuplicatingThread() 3823{ 3824 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3825 mOutputTracks[i]->destroy(); 3826 } 3827} 3828 3829void AudioFlinger::DuplicatingThread::threadLoop_mix() 3830{ 3831 // mix buffers... 3832 if (outputsReady(outputTracks)) { 3833 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3834 } else { 3835 memset(mMixBuffer, 0, mixBufferSize); 3836 } 3837 sleepTime = 0; 3838 writeFrames = mNormalFrameCount; 3839} 3840 3841void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3842{ 3843 if (sleepTime == 0) { 3844 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3845 sleepTime = activeSleepTime; 3846 } else { 3847 sleepTime = idleSleepTime; 3848 } 3849 } else if (mBytesWritten != 0) { 3850 // flush remaining overflow buffers in output tracks 3851 for (size_t i = 0; i < outputTracks.size(); i++) { 3852 if (outputTracks[i]->isActive()) { 3853 sleepTime = 0; 3854 writeFrames = 0; 3855 memset(mMixBuffer, 0, mixBufferSize); 3856 break; 3857 } 3858 } 3859 } 3860} 3861 3862void AudioFlinger::DuplicatingThread::threadLoop_write() 3863{ 3864 standbyTime = systemTime() + standbyDelay; 3865 for (size_t i = 0; i < outputTracks.size(); i++) { 3866 outputTracks[i]->write(mMixBuffer, writeFrames); 3867 } 3868 mBytesWritten += mixBufferSize; 3869} 3870 3871void AudioFlinger::DuplicatingThread::threadLoop_standby() 3872{ 3873 // DuplicatingThread implements standby by stopping all tracks 3874 for (size_t i = 0; i < outputTracks.size(); i++) { 3875 outputTracks[i]->stop(); 3876 } 3877} 3878 3879void AudioFlinger::DuplicatingThread::saveOutputTracks() 3880{ 3881 outputTracks = mOutputTracks; 3882} 3883 3884void AudioFlinger::DuplicatingThread::clearOutputTracks() 3885{ 3886 outputTracks.clear(); 3887} 3888 3889void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3890{ 3891 Mutex::Autolock _l(mLock); 3892 // FIXME explain this formula 3893 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3894 OutputTrack *outputTrack = new OutputTrack(thread, 3895 this, 3896 mSampleRate, 3897 mFormat, 3898 mChannelMask, 3899 frameCount); 3900 if (outputTrack->cblk() != NULL) { 3901 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3902 mOutputTracks.add(outputTrack); 3903 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3904 updateWaitTime_l(); 3905 } 3906} 3907 3908void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3909{ 3910 Mutex::Autolock _l(mLock); 3911 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3912 if (mOutputTracks[i]->thread() == thread) { 3913 mOutputTracks[i]->destroy(); 3914 mOutputTracks.removeAt(i); 3915 updateWaitTime_l(); 3916 return; 3917 } 3918 } 3919 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3920} 3921 3922// caller must hold mLock 3923void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3924{ 3925 mWaitTimeMs = UINT_MAX; 3926 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3927 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3928 if (strong != 0) { 3929 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3930 if (waitTimeMs < mWaitTimeMs) { 3931 mWaitTimeMs = waitTimeMs; 3932 } 3933 } 3934 } 3935} 3936 3937 3938bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3939{ 3940 for (size_t i = 0; i < outputTracks.size(); i++) { 3941 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3942 if (thread == 0) { 3943 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3944 return false; 3945 } 3946 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3947 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3948 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3949 return false; 3950 } 3951 } 3952 return true; 3953} 3954 3955uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3956{ 3957 return (mWaitTimeMs * 1000) / 2; 3958} 3959 3960void AudioFlinger::DuplicatingThread::cacheParameters_l() 3961{ 3962 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3963 updateWaitTime_l(); 3964 3965 MixerThread::cacheParameters_l(); 3966} 3967 3968// ---------------------------------------------------------------------------- 3969 3970// TrackBase constructor must be called with AudioFlinger::mLock held 3971AudioFlinger::ThreadBase::TrackBase::TrackBase( 3972 ThreadBase *thread, 3973 const sp<Client>& client, 3974 uint32_t sampleRate, 3975 audio_format_t format, 3976 uint32_t channelMask, 3977 int frameCount, 3978 const sp<IMemory>& sharedBuffer, 3979 int sessionId) 3980 : RefBase(), 3981 mThread(thread), 3982 mClient(client), 3983 mCblk(NULL), 3984 // mBuffer 3985 // mBufferEnd 3986 mFrameCount(0), 3987 mState(IDLE), 3988 mSampleRate(sampleRate), 3989 mFormat(format), 3990 mStepServerFailed(false), 3991 mSessionId(sessionId) 3992 // mChannelCount 3993 // mChannelMask 3994{ 3995 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3996 3997 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3998 size_t size = sizeof(audio_track_cblk_t); 3999 uint8_t channelCount = popcount(channelMask); 4000 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4001 if (sharedBuffer == 0) { 4002 size += bufferSize; 4003 } 4004 4005 if (client != NULL) { 4006 mCblkMemory = client->heap()->allocate(size); 4007 if (mCblkMemory != 0) { 4008 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4009 if (mCblk != NULL) { // construct the shared structure in-place. 4010 new(mCblk) audio_track_cblk_t(); 4011 // clear all buffers 4012 mCblk->frameCount = frameCount; 4013 mCblk->sampleRate = sampleRate; 4014// uncomment the following lines to quickly test 32-bit wraparound 4015// mCblk->user = 0xffff0000; 4016// mCblk->server = 0xffff0000; 4017// mCblk->userBase = 0xffff0000; 4018// mCblk->serverBase = 0xffff0000; 4019 mChannelCount = channelCount; 4020 mChannelMask = channelMask; 4021 if (sharedBuffer == 0) { 4022 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4023 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4024 // Force underrun condition to avoid false underrun callback until first data is 4025 // written to buffer (other flags are cleared) 4026 mCblk->flags = CBLK_UNDERRUN_ON; 4027 } else { 4028 mBuffer = sharedBuffer->pointer(); 4029 } 4030 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4031 } 4032 } else { 4033 ALOGE("not enough memory for AudioTrack size=%u", size); 4034 client->heap()->dump("AudioTrack"); 4035 return; 4036 } 4037 } else { 4038 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4039 // construct the shared structure in-place. 4040 new(mCblk) audio_track_cblk_t(); 4041 // clear all buffers 4042 mCblk->frameCount = frameCount; 4043 mCblk->sampleRate = sampleRate; 4044// uncomment the following lines to quickly test 32-bit wraparound 4045// mCblk->user = 0xffff0000; 4046// mCblk->server = 0xffff0000; 4047// mCblk->userBase = 0xffff0000; 4048// mCblk->serverBase = 0xffff0000; 4049 mChannelCount = channelCount; 4050 mChannelMask = channelMask; 4051 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4052 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4053 // Force underrun condition to avoid false underrun callback until first data is 4054 // written to buffer (other flags are cleared) 4055 mCblk->flags = CBLK_UNDERRUN_ON; 4056 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4057 } 4058} 4059 4060AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4061{ 4062 if (mCblk != NULL) { 4063 if (mClient == 0) { 4064 delete mCblk; 4065 } else { 4066 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4067 } 4068 } 4069 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4070 if (mClient != 0) { 4071 // Client destructor must run with AudioFlinger mutex locked 4072 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4073 // If the client's reference count drops to zero, the associated destructor 4074 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4075 // relying on the automatic clear() at end of scope. 4076 mClient.clear(); 4077 } 4078} 4079 4080// AudioBufferProvider interface 4081// getNextBuffer() = 0; 4082// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4083void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4084{ 4085 buffer->raw = NULL; 4086 mFrameCount = buffer->frameCount; 4087 // FIXME See note at getNextBuffer() 4088 (void) step(); // ignore return value of step() 4089 buffer->frameCount = 0; 4090} 4091 4092bool AudioFlinger::ThreadBase::TrackBase::step() { 4093 bool result; 4094 audio_track_cblk_t* cblk = this->cblk(); 4095 4096 result = cblk->stepServer(mFrameCount); 4097 if (!result) { 4098 ALOGV("stepServer failed acquiring cblk mutex"); 4099 mStepServerFailed = true; 4100 } 4101 return result; 4102} 4103 4104void AudioFlinger::ThreadBase::TrackBase::reset() { 4105 audio_track_cblk_t* cblk = this->cblk(); 4106 4107 cblk->user = 0; 4108 cblk->server = 0; 4109 cblk->userBase = 0; 4110 cblk->serverBase = 0; 4111 mStepServerFailed = false; 4112 ALOGV("TrackBase::reset"); 4113} 4114 4115int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4116 return (int)mCblk->sampleRate; 4117} 4118 4119void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4120 audio_track_cblk_t* cblk = this->cblk(); 4121 size_t frameSize = cblk->frameSize; 4122 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4123 int8_t *bufferEnd = bufferStart + frames * frameSize; 4124 4125 // Check validity of returned pointer in case the track control block would have been corrupted. 4126 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4127 "TrackBase::getBuffer buffer out of range:\n" 4128 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4129 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4130 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4131 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4132 4133 return bufferStart; 4134} 4135 4136status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4137{ 4138 mSyncEvents.add(event); 4139 return NO_ERROR; 4140} 4141 4142// ---------------------------------------------------------------------------- 4143 4144// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4145AudioFlinger::PlaybackThread::Track::Track( 4146 PlaybackThread *thread, 4147 const sp<Client>& client, 4148 audio_stream_type_t streamType, 4149 uint32_t sampleRate, 4150 audio_format_t format, 4151 uint32_t channelMask, 4152 int frameCount, 4153 const sp<IMemory>& sharedBuffer, 4154 int sessionId, 4155 IAudioFlinger::track_flags_t flags) 4156 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4157 mMute(false), 4158 mFillingUpStatus(FS_INVALID), 4159 // mRetryCount initialized later when needed 4160 mSharedBuffer(sharedBuffer), 4161 mStreamType(streamType), 4162 mName(-1), // see note below 4163 mMainBuffer(thread->mixBuffer()), 4164 mAuxBuffer(NULL), 4165 mAuxEffectId(0), mHasVolumeController(false), 4166 mPresentationCompleteFrames(0), 4167 mFlags(flags), 4168 mFastIndex(-1), 4169 mUnderrunCount(0), 4170 mCachedVolume(1.0) 4171{ 4172 if (mCblk != NULL) { 4173 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4174 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4175 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4176 if (flags & IAudioFlinger::TRACK_FAST) { 4177 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4178 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4179 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4180 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4181 // FIXME This is too eager. We allocate a fast track index before the 4182 // fast track becomes active. Since fast tracks are a scarce resource, 4183 // this means we are potentially denying other more important fast tracks from 4184 // being created. It would be better to allocate the index dynamically. 4185 mFastIndex = i; 4186 // Read the initial underruns because this field is never cleared by the fast mixer 4187 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4188 thread->mFastTrackAvailMask &= ~(1 << i); 4189 } 4190 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4191 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4192 if (mName < 0) { 4193 ALOGE("no more track names available"); 4194 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4195 // then we leak a fast track index. Should swap these two sections, or better yet 4196 // only allocate a normal mixer name for normal tracks. 4197 } 4198 } 4199 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4200} 4201 4202AudioFlinger::PlaybackThread::Track::~Track() 4203{ 4204 ALOGV("PlaybackThread::Track destructor"); 4205 sp<ThreadBase> thread = mThread.promote(); 4206 if (thread != 0) { 4207 Mutex::Autolock _l(thread->mLock); 4208 mState = TERMINATED; 4209 } 4210} 4211 4212void AudioFlinger::PlaybackThread::Track::destroy() 4213{ 4214 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4215 // by removing it from mTracks vector, so there is a risk that this Tracks's 4216 // destructor is called. As the destructor needs to lock mLock, 4217 // we must acquire a strong reference on this Track before locking mLock 4218 // here so that the destructor is called only when exiting this function. 4219 // On the other hand, as long as Track::destroy() is only called by 4220 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4221 // this Track with its member mTrack. 4222 sp<Track> keep(this); 4223 { // scope for mLock 4224 sp<ThreadBase> thread = mThread.promote(); 4225 if (thread != 0) { 4226 if (!isOutputTrack()) { 4227 if (mState == ACTIVE || mState == RESUMING) { 4228 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4229 4230#ifdef ADD_BATTERY_DATA 4231 // to track the speaker usage 4232 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4233#endif 4234 } 4235 AudioSystem::releaseOutput(thread->id()); 4236 } 4237 Mutex::Autolock _l(thread->mLock); 4238 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4239 playbackThread->destroyTrack_l(this); 4240 } 4241 } 4242} 4243 4244/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4245{ 4246 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4247 " Server User Main buf Aux Buf Flags FastUnder\n"); 4248} 4249 4250void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4251{ 4252 uint32_t vlr = mCblk->getVolumeLR(); 4253 if (isFastTrack()) { 4254 sprintf(buffer, " F %2d", mFastIndex); 4255 } else { 4256 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4257 } 4258 track_state state = mState; 4259 char stateChar; 4260 switch (state) { 4261 case IDLE: 4262 stateChar = 'I'; 4263 break; 4264 case TERMINATED: 4265 stateChar = 'T'; 4266 break; 4267 case STOPPING_1: 4268 stateChar = 's'; 4269 break; 4270 case STOPPING_2: 4271 stateChar = '5'; 4272 break; 4273 case STOPPED: 4274 stateChar = 'S'; 4275 break; 4276 case RESUMING: 4277 stateChar = 'R'; 4278 break; 4279 case ACTIVE: 4280 stateChar = 'A'; 4281 break; 4282 case PAUSING: 4283 stateChar = 'p'; 4284 break; 4285 case PAUSED: 4286 stateChar = 'P'; 4287 break; 4288 case FLUSHED: 4289 stateChar = 'F'; 4290 break; 4291 default: 4292 stateChar = '?'; 4293 break; 4294 } 4295 char nowInUnderrun; 4296 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4297 case UNDERRUN_FULL: 4298 nowInUnderrun = ' '; 4299 break; 4300 case UNDERRUN_PARTIAL: 4301 nowInUnderrun = '<'; 4302 break; 4303 case UNDERRUN_EMPTY: 4304 nowInUnderrun = '*'; 4305 break; 4306 default: 4307 nowInUnderrun = '?'; 4308 break; 4309 } 4310 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4311 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4312 (mClient == 0) ? getpid_cached : mClient->pid(), 4313 mStreamType, 4314 mFormat, 4315 mChannelMask, 4316 mSessionId, 4317 mFrameCount, 4318 mCblk->frameCount, 4319 stateChar, 4320 mMute, 4321 mFillingUpStatus, 4322 mCblk->sampleRate, 4323 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4324 20.0 * log10((vlr >> 16) / 4096.0), 4325 mCblk->server, 4326 mCblk->user, 4327 (int)mMainBuffer, 4328 (int)mAuxBuffer, 4329 mCblk->flags, 4330 mUnderrunCount, 4331 nowInUnderrun); 4332} 4333 4334// AudioBufferProvider interface 4335status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4336 AudioBufferProvider::Buffer* buffer, int64_t pts) 4337{ 4338 audio_track_cblk_t* cblk = this->cblk(); 4339 uint32_t framesReady; 4340 uint32_t framesReq = buffer->frameCount; 4341 4342 // Check if last stepServer failed, try to step now 4343 if (mStepServerFailed) { 4344 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4345 // Since the fast mixer is higher priority than client callback thread, 4346 // it does not result in priority inversion for client. 4347 // But a non-blocking solution would be preferable to avoid 4348 // fast mixer being unable to tryLock(), and 4349 // to avoid the extra context switches if the client wakes up, 4350 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4351 if (!step()) goto getNextBuffer_exit; 4352 ALOGV("stepServer recovered"); 4353 mStepServerFailed = false; 4354 } 4355 4356 // FIXME Same as above 4357 framesReady = cblk->framesReady(); 4358 4359 if (CC_LIKELY(framesReady)) { 4360 uint32_t s = cblk->server; 4361 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4362 4363 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4364 if (framesReq > framesReady) { 4365 framesReq = framesReady; 4366 } 4367 if (framesReq > bufferEnd - s) { 4368 framesReq = bufferEnd - s; 4369 } 4370 4371 buffer->raw = getBuffer(s, framesReq); 4372 if (buffer->raw == NULL) goto getNextBuffer_exit; 4373 4374 buffer->frameCount = framesReq; 4375 return NO_ERROR; 4376 } 4377 4378getNextBuffer_exit: 4379 buffer->raw = NULL; 4380 buffer->frameCount = 0; 4381 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4382 return NOT_ENOUGH_DATA; 4383} 4384 4385// Note that framesReady() takes a mutex on the control block using tryLock(). 4386// This could result in priority inversion if framesReady() is called by the normal mixer, 4387// as the normal mixer thread runs at lower 4388// priority than the client's callback thread: there is a short window within framesReady() 4389// during which the normal mixer could be preempted, and the client callback would block. 4390// Another problem can occur if framesReady() is called by the fast mixer: 4391// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4392// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4393size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4394 return mCblk->framesReady(); 4395} 4396 4397// Don't call for fast tracks; the framesReady() could result in priority inversion 4398bool AudioFlinger::PlaybackThread::Track::isReady() const { 4399 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4400 4401 if (framesReady() >= mCblk->frameCount || 4402 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4403 mFillingUpStatus = FS_FILLED; 4404 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4405 return true; 4406 } 4407 return false; 4408} 4409 4410status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4411 int triggerSession) 4412{ 4413 status_t status = NO_ERROR; 4414 ALOGV("start(%d), calling pid %d session %d", 4415 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4416 4417 sp<ThreadBase> thread = mThread.promote(); 4418 if (thread != 0) { 4419 Mutex::Autolock _l(thread->mLock); 4420 track_state state = mState; 4421 // here the track could be either new, or restarted 4422 // in both cases "unstop" the track 4423 if (mState == PAUSED) { 4424 mState = TrackBase::RESUMING; 4425 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4426 } else { 4427 mState = TrackBase::ACTIVE; 4428 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4429 } 4430 4431 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4432 thread->mLock.unlock(); 4433 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4434 thread->mLock.lock(); 4435 4436#ifdef ADD_BATTERY_DATA 4437 // to track the speaker usage 4438 if (status == NO_ERROR) { 4439 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4440 } 4441#endif 4442 } 4443 if (status == NO_ERROR) { 4444 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4445 playbackThread->addTrack_l(this); 4446 } else { 4447 mState = state; 4448 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4449 } 4450 } else { 4451 status = BAD_VALUE; 4452 } 4453 return status; 4454} 4455 4456void AudioFlinger::PlaybackThread::Track::stop() 4457{ 4458 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4459 sp<ThreadBase> thread = mThread.promote(); 4460 if (thread != 0) { 4461 Mutex::Autolock _l(thread->mLock); 4462 track_state state = mState; 4463 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4464 // If the track is not active (PAUSED and buffers full), flush buffers 4465 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4466 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4467 reset(); 4468 mState = STOPPED; 4469 } else if (!isFastTrack()) { 4470 mState = STOPPED; 4471 } else { 4472 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4473 // and then to STOPPED and reset() when presentation is complete 4474 mState = STOPPING_1; 4475 } 4476 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4477 } 4478 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4479 thread->mLock.unlock(); 4480 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4481 thread->mLock.lock(); 4482 4483#ifdef ADD_BATTERY_DATA 4484 // to track the speaker usage 4485 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4486#endif 4487 } 4488 } 4489} 4490 4491void AudioFlinger::PlaybackThread::Track::pause() 4492{ 4493 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4494 sp<ThreadBase> thread = mThread.promote(); 4495 if (thread != 0) { 4496 Mutex::Autolock _l(thread->mLock); 4497 if (mState == ACTIVE || mState == RESUMING) { 4498 mState = PAUSING; 4499 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4500 if (!isOutputTrack()) { 4501 thread->mLock.unlock(); 4502 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4503 thread->mLock.lock(); 4504 4505#ifdef ADD_BATTERY_DATA 4506 // to track the speaker usage 4507 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4508#endif 4509 } 4510 } 4511 } 4512} 4513 4514void AudioFlinger::PlaybackThread::Track::flush() 4515{ 4516 ALOGV("flush(%d)", mName); 4517 sp<ThreadBase> thread = mThread.promote(); 4518 if (thread != 0) { 4519 Mutex::Autolock _l(thread->mLock); 4520 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4521 mState != PAUSING) { 4522 return; 4523 } 4524 // No point remaining in PAUSED state after a flush => go to 4525 // FLUSHED state 4526 mState = FLUSHED; 4527 // do not reset the track if it is still in the process of being stopped or paused. 4528 // this will be done by prepareTracks_l() when the track is stopped. 4529 // prepareTracks_l() will see mState == FLUSHED, then 4530 // remove from active track list, reset(), and trigger presentation complete 4531 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4532 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4533 reset(); 4534 } 4535 } 4536} 4537 4538void AudioFlinger::PlaybackThread::Track::reset() 4539{ 4540 // Do not reset twice to avoid discarding data written just after a flush and before 4541 // the audioflinger thread detects the track is stopped. 4542 if (!mResetDone) { 4543 TrackBase::reset(); 4544 // Force underrun condition to avoid false underrun callback until first data is 4545 // written to buffer 4546 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4547 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4548 mFillingUpStatus = FS_FILLING; 4549 mResetDone = true; 4550 if (mState == FLUSHED) { 4551 mState = IDLE; 4552 } 4553 } 4554} 4555 4556void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4557{ 4558 mMute = muted; 4559} 4560 4561status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4562{ 4563 status_t status = DEAD_OBJECT; 4564 sp<ThreadBase> thread = mThread.promote(); 4565 if (thread != 0) { 4566 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4567 status = playbackThread->attachAuxEffect(this, EffectId); 4568 } 4569 return status; 4570} 4571 4572void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4573{ 4574 mAuxEffectId = EffectId; 4575 mAuxBuffer = buffer; 4576} 4577 4578bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4579 size_t audioHalFrames) 4580{ 4581 // a track is considered presented when the total number of frames written to audio HAL 4582 // corresponds to the number of frames written when presentationComplete() is called for the 4583 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4584 if (mPresentationCompleteFrames == 0) { 4585 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4586 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4587 mPresentationCompleteFrames, audioHalFrames); 4588 } 4589 if (framesWritten >= mPresentationCompleteFrames) { 4590 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4591 mSessionId, framesWritten); 4592 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4593 return true; 4594 } 4595 return false; 4596} 4597 4598void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4599{ 4600 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4601 if (mSyncEvents[i]->type() == type) { 4602 mSyncEvents[i]->trigger(); 4603 mSyncEvents.removeAt(i); 4604 i--; 4605 } 4606 } 4607} 4608 4609// implement VolumeBufferProvider interface 4610 4611uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4612{ 4613 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4614 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4615 uint32_t vlr = mCblk->getVolumeLR(); 4616 uint32_t vl = vlr & 0xFFFF; 4617 uint32_t vr = vlr >> 16; 4618 // track volumes come from shared memory, so can't be trusted and must be clamped 4619 if (vl > MAX_GAIN_INT) { 4620 vl = MAX_GAIN_INT; 4621 } 4622 if (vr > MAX_GAIN_INT) { 4623 vr = MAX_GAIN_INT; 4624 } 4625 // now apply the cached master volume and stream type volume; 4626 // this is trusted but lacks any synchronization or barrier so may be stale 4627 float v = mCachedVolume; 4628 vl *= v; 4629 vr *= v; 4630 // re-combine into U4.16 4631 vlr = (vr << 16) | (vl & 0xFFFF); 4632 // FIXME look at mute, pause, and stop flags 4633 return vlr; 4634} 4635 4636status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4637{ 4638 if (mState == TERMINATED || mState == PAUSED || 4639 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4640 (mState == STOPPED)))) { 4641 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4642 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4643 event->cancel(); 4644 return INVALID_OPERATION; 4645 } 4646 TrackBase::setSyncEvent(event); 4647 return NO_ERROR; 4648} 4649 4650// timed audio tracks 4651 4652sp<AudioFlinger::PlaybackThread::TimedTrack> 4653AudioFlinger::PlaybackThread::TimedTrack::create( 4654 PlaybackThread *thread, 4655 const sp<Client>& client, 4656 audio_stream_type_t streamType, 4657 uint32_t sampleRate, 4658 audio_format_t format, 4659 uint32_t channelMask, 4660 int frameCount, 4661 const sp<IMemory>& sharedBuffer, 4662 int sessionId) { 4663 if (!client->reserveTimedTrack()) 4664 return NULL; 4665 4666 return new TimedTrack( 4667 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4668 sharedBuffer, sessionId); 4669} 4670 4671AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4672 PlaybackThread *thread, 4673 const sp<Client>& client, 4674 audio_stream_type_t streamType, 4675 uint32_t sampleRate, 4676 audio_format_t format, 4677 uint32_t channelMask, 4678 int frameCount, 4679 const sp<IMemory>& sharedBuffer, 4680 int sessionId) 4681 : Track(thread, client, streamType, sampleRate, format, channelMask, 4682 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4683 mQueueHeadInFlight(false), 4684 mTrimQueueHeadOnRelease(false), 4685 mFramesPendingInQueue(0), 4686 mTimedSilenceBuffer(NULL), 4687 mTimedSilenceBufferSize(0), 4688 mTimedAudioOutputOnTime(false), 4689 mMediaTimeTransformValid(false) 4690{ 4691 LocalClock lc; 4692 mLocalTimeFreq = lc.getLocalFreq(); 4693 4694 mLocalTimeToSampleTransform.a_zero = 0; 4695 mLocalTimeToSampleTransform.b_zero = 0; 4696 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4697 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4698 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4699 &mLocalTimeToSampleTransform.a_to_b_denom); 4700 4701 mMediaTimeToSampleTransform.a_zero = 0; 4702 mMediaTimeToSampleTransform.b_zero = 0; 4703 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4704 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4705 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4706 &mMediaTimeToSampleTransform.a_to_b_denom); 4707} 4708 4709AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4710 mClient->releaseTimedTrack(); 4711 delete [] mTimedSilenceBuffer; 4712} 4713 4714status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4715 size_t size, sp<IMemory>* buffer) { 4716 4717 Mutex::Autolock _l(mTimedBufferQueueLock); 4718 4719 trimTimedBufferQueue_l(); 4720 4721 // lazily initialize the shared memory heap for timed buffers 4722 if (mTimedMemoryDealer == NULL) { 4723 const int kTimedBufferHeapSize = 512 << 10; 4724 4725 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4726 "AudioFlingerTimed"); 4727 if (mTimedMemoryDealer == NULL) 4728 return NO_MEMORY; 4729 } 4730 4731 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4732 if (newBuffer == NULL) { 4733 newBuffer = mTimedMemoryDealer->allocate(size); 4734 if (newBuffer == NULL) 4735 return NO_MEMORY; 4736 } 4737 4738 *buffer = newBuffer; 4739 return NO_ERROR; 4740} 4741 4742// caller must hold mTimedBufferQueueLock 4743void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4744 int64_t mediaTimeNow; 4745 { 4746 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4747 if (!mMediaTimeTransformValid) 4748 return; 4749 4750 int64_t targetTimeNow; 4751 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4752 ? mCCHelper.getCommonTime(&targetTimeNow) 4753 : mCCHelper.getLocalTime(&targetTimeNow); 4754 4755 if (OK != res) 4756 return; 4757 4758 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4759 &mediaTimeNow)) { 4760 return; 4761 } 4762 } 4763 4764 size_t trimEnd; 4765 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4766 int64_t bufEnd; 4767 4768 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4769 // We have a next buffer. Just use its PTS as the PTS of the frame 4770 // following the last frame in this buffer. If the stream is sparse 4771 // (ie, there are deliberate gaps left in the stream which should be 4772 // filled with silence by the TimedAudioTrack), then this can result 4773 // in one extra buffer being left un-trimmed when it could have 4774 // been. In general, this is not typical, and we would rather 4775 // optimized away the TS calculation below for the more common case 4776 // where PTSes are contiguous. 4777 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4778 } else { 4779 // We have no next buffer. Compute the PTS of the frame following 4780 // the last frame in this buffer by computing the duration of of 4781 // this frame in media time units and adding it to the PTS of the 4782 // buffer. 4783 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4784 / mCblk->frameSize; 4785 4786 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4787 &bufEnd)) { 4788 ALOGE("Failed to convert frame count of %lld to media time" 4789 " duration" " (scale factor %d/%u) in %s", 4790 frameCount, 4791 mMediaTimeToSampleTransform.a_to_b_numer, 4792 mMediaTimeToSampleTransform.a_to_b_denom, 4793 __PRETTY_FUNCTION__); 4794 break; 4795 } 4796 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4797 } 4798 4799 if (bufEnd > mediaTimeNow) 4800 break; 4801 4802 // Is the buffer we want to use in the middle of a mix operation right 4803 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4804 // from the mixer which should be coming back shortly. 4805 if (!trimEnd && mQueueHeadInFlight) { 4806 mTrimQueueHeadOnRelease = true; 4807 } 4808 } 4809 4810 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4811 if (trimStart < trimEnd) { 4812 // Update the bookkeeping for framesReady() 4813 for (size_t i = trimStart; i < trimEnd; ++i) { 4814 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4815 } 4816 4817 // Now actually remove the buffers from the queue. 4818 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4819 } 4820} 4821 4822void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4823 const char* logTag) { 4824 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4825 "%s called (reason \"%s\"), but timed buffer queue has no" 4826 " elements to trim.", __FUNCTION__, logTag); 4827 4828 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4829 mTimedBufferQueue.removeAt(0); 4830} 4831 4832void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4833 const TimedBuffer& buf, 4834 const char* logTag) { 4835 uint32_t bufBytes = buf.buffer()->size(); 4836 uint32_t consumedAlready = buf.position(); 4837 4838 ALOG_ASSERT(consumedAlready <= bufBytes, 4839 "Bad bookkeeping while updating frames pending. Timed buffer is" 4840 " only %u bytes long, but claims to have consumed %u" 4841 " bytes. (update reason: \"%s\")", 4842 bufBytes, consumedAlready, logTag); 4843 4844 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4845 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4846 "Bad bookkeeping while updating frames pending. Should have at" 4847 " least %u queued frames, but we think we have only %u. (update" 4848 " reason: \"%s\")", 4849 bufFrames, mFramesPendingInQueue, logTag); 4850 4851 mFramesPendingInQueue -= bufFrames; 4852} 4853 4854status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4855 const sp<IMemory>& buffer, int64_t pts) { 4856 4857 { 4858 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4859 if (!mMediaTimeTransformValid) 4860 return INVALID_OPERATION; 4861 } 4862 4863 Mutex::Autolock _l(mTimedBufferQueueLock); 4864 4865 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4866 mFramesPendingInQueue += bufFrames; 4867 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4868 4869 return NO_ERROR; 4870} 4871 4872status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4873 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4874 4875 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4876 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4877 target); 4878 4879 if (!(target == TimedAudioTrack::LOCAL_TIME || 4880 target == TimedAudioTrack::COMMON_TIME)) { 4881 return BAD_VALUE; 4882 } 4883 4884 Mutex::Autolock lock(mMediaTimeTransformLock); 4885 mMediaTimeTransform = xform; 4886 mMediaTimeTransformTarget = target; 4887 mMediaTimeTransformValid = true; 4888 4889 return NO_ERROR; 4890} 4891 4892#define min(a, b) ((a) < (b) ? (a) : (b)) 4893 4894// implementation of getNextBuffer for tracks whose buffers have timestamps 4895status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4896 AudioBufferProvider::Buffer* buffer, int64_t pts) 4897{ 4898 if (pts == AudioBufferProvider::kInvalidPTS) { 4899 buffer->raw = 0; 4900 buffer->frameCount = 0; 4901 mTimedAudioOutputOnTime = false; 4902 return INVALID_OPERATION; 4903 } 4904 4905 Mutex::Autolock _l(mTimedBufferQueueLock); 4906 4907 ALOG_ASSERT(!mQueueHeadInFlight, 4908 "getNextBuffer called without releaseBuffer!"); 4909 4910 while (true) { 4911 4912 // if we have no timed buffers, then fail 4913 if (mTimedBufferQueue.isEmpty()) { 4914 buffer->raw = 0; 4915 buffer->frameCount = 0; 4916 return NOT_ENOUGH_DATA; 4917 } 4918 4919 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4920 4921 // calculate the PTS of the head of the timed buffer queue expressed in 4922 // local time 4923 int64_t headLocalPTS; 4924 { 4925 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4926 4927 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4928 4929 if (mMediaTimeTransform.a_to_b_denom == 0) { 4930 // the transform represents a pause, so yield silence 4931 timedYieldSilence_l(buffer->frameCount, buffer); 4932 return NO_ERROR; 4933 } 4934 4935 int64_t transformedPTS; 4936 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4937 &transformedPTS)) { 4938 // the transform failed. this shouldn't happen, but if it does 4939 // then just drop this buffer 4940 ALOGW("timedGetNextBuffer transform failed"); 4941 buffer->raw = 0; 4942 buffer->frameCount = 0; 4943 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4944 return NO_ERROR; 4945 } 4946 4947 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4948 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4949 &headLocalPTS)) { 4950 buffer->raw = 0; 4951 buffer->frameCount = 0; 4952 return INVALID_OPERATION; 4953 } 4954 } else { 4955 headLocalPTS = transformedPTS; 4956 } 4957 } 4958 4959 // adjust the head buffer's PTS to reflect the portion of the head buffer 4960 // that has already been consumed 4961 int64_t effectivePTS = headLocalPTS + 4962 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4963 4964 // Calculate the delta in samples between the head of the input buffer 4965 // queue and the start of the next output buffer that will be written. 4966 // If the transformation fails because of over or underflow, it means 4967 // that the sample's position in the output stream is so far out of 4968 // whack that it should just be dropped. 4969 int64_t sampleDelta; 4970 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4971 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4972 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4973 " mix"); 4974 continue; 4975 } 4976 if (!mLocalTimeToSampleTransform.doForwardTransform( 4977 (effectivePTS - pts) << 32, &sampleDelta)) { 4978 ALOGV("*** too late during sample rate transform: dropped buffer"); 4979 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4980 continue; 4981 } 4982 4983 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4984 " sampleDelta=[%d.%08x]", 4985 head.pts(), head.position(), pts, 4986 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4987 + (sampleDelta >> 32)), 4988 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4989 4990 // if the delta between the ideal placement for the next input sample and 4991 // the current output position is within this threshold, then we will 4992 // concatenate the next input samples to the previous output 4993 const int64_t kSampleContinuityThreshold = 4994 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4995 4996 // if this is the first buffer of audio that we're emitting from this track 4997 // then it should be almost exactly on time. 4998 const int64_t kSampleStartupThreshold = 1LL << 32; 4999 5000 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5001 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5002 // the next input is close enough to being on time, so concatenate it 5003 // with the last output 5004 timedYieldSamples_l(buffer); 5005 5006 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5007 head.position(), buffer->frameCount); 5008 return NO_ERROR; 5009 } 5010 5011 // Looks like our output is not on time. Reset our on timed status. 5012 // Next time we mix samples from our input queue, then should be within 5013 // the StartupThreshold. 5014 mTimedAudioOutputOnTime = false; 5015 if (sampleDelta > 0) { 5016 // the gap between the current output position and the proper start of 5017 // the next input sample is too big, so fill it with silence 5018 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5019 5020 timedYieldSilence_l(framesUntilNextInput, buffer); 5021 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5022 return NO_ERROR; 5023 } else { 5024 // the next input sample is late 5025 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5026 size_t onTimeSamplePosition = 5027 head.position() + lateFrames * mCblk->frameSize; 5028 5029 if (onTimeSamplePosition > head.buffer()->size()) { 5030 // all the remaining samples in the head are too late, so 5031 // drop it and move on 5032 ALOGV("*** too late: dropped buffer"); 5033 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5034 continue; 5035 } else { 5036 // skip over the late samples 5037 head.setPosition(onTimeSamplePosition); 5038 5039 // yield the available samples 5040 timedYieldSamples_l(buffer); 5041 5042 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5043 return NO_ERROR; 5044 } 5045 } 5046 } 5047} 5048 5049// Yield samples from the timed buffer queue head up to the given output 5050// buffer's capacity. 5051// 5052// Caller must hold mTimedBufferQueueLock 5053void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5054 AudioBufferProvider::Buffer* buffer) { 5055 5056 const TimedBuffer& head = mTimedBufferQueue[0]; 5057 5058 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5059 head.position()); 5060 5061 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5062 mCblk->frameSize); 5063 size_t framesRequested = buffer->frameCount; 5064 buffer->frameCount = min(framesLeftInHead, framesRequested); 5065 5066 mQueueHeadInFlight = true; 5067 mTimedAudioOutputOnTime = true; 5068} 5069 5070// Yield samples of silence up to the given output buffer's capacity 5071// 5072// Caller must hold mTimedBufferQueueLock 5073void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5074 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5075 5076 // lazily allocate a buffer filled with silence 5077 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5078 delete [] mTimedSilenceBuffer; 5079 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5080 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5081 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5082 } 5083 5084 buffer->raw = mTimedSilenceBuffer; 5085 size_t framesRequested = buffer->frameCount; 5086 buffer->frameCount = min(numFrames, framesRequested); 5087 5088 mTimedAudioOutputOnTime = false; 5089} 5090 5091// AudioBufferProvider interface 5092void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5093 AudioBufferProvider::Buffer* buffer) { 5094 5095 Mutex::Autolock _l(mTimedBufferQueueLock); 5096 5097 // If the buffer which was just released is part of the buffer at the head 5098 // of the queue, be sure to update the amt of the buffer which has been 5099 // consumed. If the buffer being returned is not part of the head of the 5100 // queue, its either because the buffer is part of the silence buffer, or 5101 // because the head of the timed queue was trimmed after the mixer called 5102 // getNextBuffer but before the mixer called releaseBuffer. 5103 if (buffer->raw == mTimedSilenceBuffer) { 5104 ALOG_ASSERT(!mQueueHeadInFlight, 5105 "Queue head in flight during release of silence buffer!"); 5106 goto done; 5107 } 5108 5109 ALOG_ASSERT(mQueueHeadInFlight, 5110 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5111 " head in flight."); 5112 5113 if (mTimedBufferQueue.size()) { 5114 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5115 5116 void* start = head.buffer()->pointer(); 5117 void* end = reinterpret_cast<void*>( 5118 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5119 + head.buffer()->size()); 5120 5121 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5122 "released buffer not within the head of the timed buffer" 5123 " queue; qHead = [%p, %p], released buffer = %p", 5124 start, end, buffer->raw); 5125 5126 head.setPosition(head.position() + 5127 (buffer->frameCount * mCblk->frameSize)); 5128 mQueueHeadInFlight = false; 5129 5130 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5131 "Bad bookkeeping during releaseBuffer! Should have at" 5132 " least %u queued frames, but we think we have only %u", 5133 buffer->frameCount, mFramesPendingInQueue); 5134 5135 mFramesPendingInQueue -= buffer->frameCount; 5136 5137 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5138 || mTrimQueueHeadOnRelease) { 5139 trimTimedBufferQueueHead_l("releaseBuffer"); 5140 mTrimQueueHeadOnRelease = false; 5141 } 5142 } else { 5143 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5144 " buffers in the timed buffer queue"); 5145 } 5146 5147done: 5148 buffer->raw = 0; 5149 buffer->frameCount = 0; 5150} 5151 5152size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5153 Mutex::Autolock _l(mTimedBufferQueueLock); 5154 return mFramesPendingInQueue; 5155} 5156 5157AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5158 : mPTS(0), mPosition(0) {} 5159 5160AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5161 const sp<IMemory>& buffer, int64_t pts) 5162 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5163 5164// ---------------------------------------------------------------------------- 5165 5166// RecordTrack constructor must be called with AudioFlinger::mLock held 5167AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5168 RecordThread *thread, 5169 const sp<Client>& client, 5170 uint32_t sampleRate, 5171 audio_format_t format, 5172 uint32_t channelMask, 5173 int frameCount, 5174 int sessionId) 5175 : TrackBase(thread, client, sampleRate, format, 5176 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5177 mOverflow(false) 5178{ 5179 if (mCblk != NULL) { 5180 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5181 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5182 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5183 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5184 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5185 } else { 5186 mCblk->frameSize = sizeof(int8_t); 5187 } 5188 } 5189} 5190 5191AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5192{ 5193 sp<ThreadBase> thread = mThread.promote(); 5194 if (thread != 0) { 5195 AudioSystem::releaseInput(thread->id()); 5196 } 5197} 5198 5199// AudioBufferProvider interface 5200status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5201{ 5202 audio_track_cblk_t* cblk = this->cblk(); 5203 uint32_t framesAvail; 5204 uint32_t framesReq = buffer->frameCount; 5205 5206 // Check if last stepServer failed, try to step now 5207 if (mStepServerFailed) { 5208 if (!step()) goto getNextBuffer_exit; 5209 ALOGV("stepServer recovered"); 5210 mStepServerFailed = false; 5211 } 5212 5213 framesAvail = cblk->framesAvailable_l(); 5214 5215 if (CC_LIKELY(framesAvail)) { 5216 uint32_t s = cblk->server; 5217 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5218 5219 if (framesReq > framesAvail) { 5220 framesReq = framesAvail; 5221 } 5222 if (framesReq > bufferEnd - s) { 5223 framesReq = bufferEnd - s; 5224 } 5225 5226 buffer->raw = getBuffer(s, framesReq); 5227 if (buffer->raw == NULL) goto getNextBuffer_exit; 5228 5229 buffer->frameCount = framesReq; 5230 return NO_ERROR; 5231 } 5232 5233getNextBuffer_exit: 5234 buffer->raw = NULL; 5235 buffer->frameCount = 0; 5236 return NOT_ENOUGH_DATA; 5237} 5238 5239status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5240 int triggerSession) 5241{ 5242 sp<ThreadBase> thread = mThread.promote(); 5243 if (thread != 0) { 5244 RecordThread *recordThread = (RecordThread *)thread.get(); 5245 return recordThread->start(this, event, triggerSession); 5246 } else { 5247 return BAD_VALUE; 5248 } 5249} 5250 5251void AudioFlinger::RecordThread::RecordTrack::stop() 5252{ 5253 sp<ThreadBase> thread = mThread.promote(); 5254 if (thread != 0) { 5255 RecordThread *recordThread = (RecordThread *)thread.get(); 5256 recordThread->stop(this); 5257 TrackBase::reset(); 5258 // Force overrun condition to avoid false overrun callback until first data is 5259 // read from buffer 5260 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5261 } 5262} 5263 5264void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5265{ 5266 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5267 (mClient == 0) ? getpid_cached : mClient->pid(), 5268 mFormat, 5269 mChannelMask, 5270 mSessionId, 5271 mFrameCount, 5272 mState, 5273 mCblk->sampleRate, 5274 mCblk->server, 5275 mCblk->user); 5276} 5277 5278 5279// ---------------------------------------------------------------------------- 5280 5281AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5282 PlaybackThread *playbackThread, 5283 DuplicatingThread *sourceThread, 5284 uint32_t sampleRate, 5285 audio_format_t format, 5286 uint32_t channelMask, 5287 int frameCount) 5288 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5289 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5290 mActive(false), mSourceThread(sourceThread) 5291{ 5292 5293 if (mCblk != NULL) { 5294 mCblk->flags |= CBLK_DIRECTION_OUT; 5295 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5296 mOutBuffer.frameCount = 0; 5297 playbackThread->mTracks.add(this); 5298 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5299 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5300 mCblk, mBuffer, mCblk->buffers, 5301 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5302 } else { 5303 ALOGW("Error creating output track on thread %p", playbackThread); 5304 } 5305} 5306 5307AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5308{ 5309 clearBufferQueue(); 5310} 5311 5312status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5313 int triggerSession) 5314{ 5315 status_t status = Track::start(event, triggerSession); 5316 if (status != NO_ERROR) { 5317 return status; 5318 } 5319 5320 mActive = true; 5321 mRetryCount = 127; 5322 return status; 5323} 5324 5325void AudioFlinger::PlaybackThread::OutputTrack::stop() 5326{ 5327 Track::stop(); 5328 clearBufferQueue(); 5329 mOutBuffer.frameCount = 0; 5330 mActive = false; 5331} 5332 5333bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5334{ 5335 Buffer *pInBuffer; 5336 Buffer inBuffer; 5337 uint32_t channelCount = mChannelCount; 5338 bool outputBufferFull = false; 5339 inBuffer.frameCount = frames; 5340 inBuffer.i16 = data; 5341 5342 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5343 5344 if (!mActive && frames != 0) { 5345 start(); 5346 sp<ThreadBase> thread = mThread.promote(); 5347 if (thread != 0) { 5348 MixerThread *mixerThread = (MixerThread *)thread.get(); 5349 if (mCblk->frameCount > frames){ 5350 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5351 uint32_t startFrames = (mCblk->frameCount - frames); 5352 pInBuffer = new Buffer; 5353 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5354 pInBuffer->frameCount = startFrames; 5355 pInBuffer->i16 = pInBuffer->mBuffer; 5356 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5357 mBufferQueue.add(pInBuffer); 5358 } else { 5359 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5360 } 5361 } 5362 } 5363 } 5364 5365 while (waitTimeLeftMs) { 5366 // First write pending buffers, then new data 5367 if (mBufferQueue.size()) { 5368 pInBuffer = mBufferQueue.itemAt(0); 5369 } else { 5370 pInBuffer = &inBuffer; 5371 } 5372 5373 if (pInBuffer->frameCount == 0) { 5374 break; 5375 } 5376 5377 if (mOutBuffer.frameCount == 0) { 5378 mOutBuffer.frameCount = pInBuffer->frameCount; 5379 nsecs_t startTime = systemTime(); 5380 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5381 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5382 outputBufferFull = true; 5383 break; 5384 } 5385 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5386 if (waitTimeLeftMs >= waitTimeMs) { 5387 waitTimeLeftMs -= waitTimeMs; 5388 } else { 5389 waitTimeLeftMs = 0; 5390 } 5391 } 5392 5393 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5394 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5395 mCblk->stepUser(outFrames); 5396 pInBuffer->frameCount -= outFrames; 5397 pInBuffer->i16 += outFrames * channelCount; 5398 mOutBuffer.frameCount -= outFrames; 5399 mOutBuffer.i16 += outFrames * channelCount; 5400 5401 if (pInBuffer->frameCount == 0) { 5402 if (mBufferQueue.size()) { 5403 mBufferQueue.removeAt(0); 5404 delete [] pInBuffer->mBuffer; 5405 delete pInBuffer; 5406 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5407 } else { 5408 break; 5409 } 5410 } 5411 } 5412 5413 // If we could not write all frames, allocate a buffer and queue it for next time. 5414 if (inBuffer.frameCount) { 5415 sp<ThreadBase> thread = mThread.promote(); 5416 if (thread != 0 && !thread->standby()) { 5417 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5418 pInBuffer = new Buffer; 5419 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5420 pInBuffer->frameCount = inBuffer.frameCount; 5421 pInBuffer->i16 = pInBuffer->mBuffer; 5422 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5423 mBufferQueue.add(pInBuffer); 5424 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5425 } else { 5426 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5427 } 5428 } 5429 } 5430 5431 // Calling write() with a 0 length buffer, means that no more data will be written: 5432 // If no more buffers are pending, fill output track buffer to make sure it is started 5433 // by output mixer. 5434 if (frames == 0 && mBufferQueue.size() == 0) { 5435 if (mCblk->user < mCblk->frameCount) { 5436 frames = mCblk->frameCount - mCblk->user; 5437 pInBuffer = new Buffer; 5438 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5439 pInBuffer->frameCount = frames; 5440 pInBuffer->i16 = pInBuffer->mBuffer; 5441 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5442 mBufferQueue.add(pInBuffer); 5443 } else if (mActive) { 5444 stop(); 5445 } 5446 } 5447 5448 return outputBufferFull; 5449} 5450 5451status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5452{ 5453 int active; 5454 status_t result; 5455 audio_track_cblk_t* cblk = mCblk; 5456 uint32_t framesReq = buffer->frameCount; 5457 5458// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5459 buffer->frameCount = 0; 5460 5461 uint32_t framesAvail = cblk->framesAvailable(); 5462 5463 5464 if (framesAvail == 0) { 5465 Mutex::Autolock _l(cblk->lock); 5466 goto start_loop_here; 5467 while (framesAvail == 0) { 5468 active = mActive; 5469 if (CC_UNLIKELY(!active)) { 5470 ALOGV("Not active and NO_MORE_BUFFERS"); 5471 return NO_MORE_BUFFERS; 5472 } 5473 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5474 if (result != NO_ERROR) { 5475 return NO_MORE_BUFFERS; 5476 } 5477 // read the server count again 5478 start_loop_here: 5479 framesAvail = cblk->framesAvailable_l(); 5480 } 5481 } 5482 5483// if (framesAvail < framesReq) { 5484// return NO_MORE_BUFFERS; 5485// } 5486 5487 if (framesReq > framesAvail) { 5488 framesReq = framesAvail; 5489 } 5490 5491 uint32_t u = cblk->user; 5492 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5493 5494 if (framesReq > bufferEnd - u) { 5495 framesReq = bufferEnd - u; 5496 } 5497 5498 buffer->frameCount = framesReq; 5499 buffer->raw = (void *)cblk->buffer(u); 5500 return NO_ERROR; 5501} 5502 5503 5504void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5505{ 5506 size_t size = mBufferQueue.size(); 5507 5508 for (size_t i = 0; i < size; i++) { 5509 Buffer *pBuffer = mBufferQueue.itemAt(i); 5510 delete [] pBuffer->mBuffer; 5511 delete pBuffer; 5512 } 5513 mBufferQueue.clear(); 5514} 5515 5516// ---------------------------------------------------------------------------- 5517 5518AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5519 : RefBase(), 5520 mAudioFlinger(audioFlinger), 5521 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5522 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5523 mPid(pid), 5524 mTimedTrackCount(0) 5525{ 5526 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5527} 5528 5529// Client destructor must be called with AudioFlinger::mLock held 5530AudioFlinger::Client::~Client() 5531{ 5532 mAudioFlinger->removeClient_l(mPid); 5533} 5534 5535sp<MemoryDealer> AudioFlinger::Client::heap() const 5536{ 5537 return mMemoryDealer; 5538} 5539 5540// Reserve one of the limited slots for a timed audio track associated 5541// with this client 5542bool AudioFlinger::Client::reserveTimedTrack() 5543{ 5544 const int kMaxTimedTracksPerClient = 4; 5545 5546 Mutex::Autolock _l(mTimedTrackLock); 5547 5548 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5549 ALOGW("can not create timed track - pid %d has exceeded the limit", 5550 mPid); 5551 return false; 5552 } 5553 5554 mTimedTrackCount++; 5555 return true; 5556} 5557 5558// Release a slot for a timed audio track 5559void AudioFlinger::Client::releaseTimedTrack() 5560{ 5561 Mutex::Autolock _l(mTimedTrackLock); 5562 mTimedTrackCount--; 5563} 5564 5565// ---------------------------------------------------------------------------- 5566 5567AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5568 const sp<IAudioFlingerClient>& client, 5569 pid_t pid) 5570 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5571{ 5572} 5573 5574AudioFlinger::NotificationClient::~NotificationClient() 5575{ 5576} 5577 5578void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5579{ 5580 sp<NotificationClient> keep(this); 5581 mAudioFlinger->removeNotificationClient(mPid); 5582} 5583 5584// ---------------------------------------------------------------------------- 5585 5586AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5587 : BnAudioTrack(), 5588 mTrack(track) 5589{ 5590} 5591 5592AudioFlinger::TrackHandle::~TrackHandle() { 5593 // just stop the track on deletion, associated resources 5594 // will be freed from the main thread once all pending buffers have 5595 // been played. Unless it's not in the active track list, in which 5596 // case we free everything now... 5597 mTrack->destroy(); 5598} 5599 5600sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5601 return mTrack->getCblk(); 5602} 5603 5604status_t AudioFlinger::TrackHandle::start() { 5605 return mTrack->start(); 5606} 5607 5608void AudioFlinger::TrackHandle::stop() { 5609 mTrack->stop(); 5610} 5611 5612void AudioFlinger::TrackHandle::flush() { 5613 mTrack->flush(); 5614} 5615 5616void AudioFlinger::TrackHandle::mute(bool e) { 5617 mTrack->mute(e); 5618} 5619 5620void AudioFlinger::TrackHandle::pause() { 5621 mTrack->pause(); 5622} 5623 5624status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5625{ 5626 return mTrack->attachAuxEffect(EffectId); 5627} 5628 5629status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5630 sp<IMemory>* buffer) { 5631 if (!mTrack->isTimedTrack()) 5632 return INVALID_OPERATION; 5633 5634 PlaybackThread::TimedTrack* tt = 5635 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5636 return tt->allocateTimedBuffer(size, buffer); 5637} 5638 5639status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5640 int64_t pts) { 5641 if (!mTrack->isTimedTrack()) 5642 return INVALID_OPERATION; 5643 5644 PlaybackThread::TimedTrack* tt = 5645 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5646 return tt->queueTimedBuffer(buffer, pts); 5647} 5648 5649status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5650 const LinearTransform& xform, int target) { 5651 5652 if (!mTrack->isTimedTrack()) 5653 return INVALID_OPERATION; 5654 5655 PlaybackThread::TimedTrack* tt = 5656 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5657 return tt->setMediaTimeTransform( 5658 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5659} 5660 5661status_t AudioFlinger::TrackHandle::onTransact( 5662 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5663{ 5664 return BnAudioTrack::onTransact(code, data, reply, flags); 5665} 5666 5667// ---------------------------------------------------------------------------- 5668 5669sp<IAudioRecord> AudioFlinger::openRecord( 5670 pid_t pid, 5671 audio_io_handle_t input, 5672 uint32_t sampleRate, 5673 audio_format_t format, 5674 uint32_t channelMask, 5675 int frameCount, 5676 IAudioFlinger::track_flags_t flags, 5677 int *sessionId, 5678 status_t *status) 5679{ 5680 sp<RecordThread::RecordTrack> recordTrack; 5681 sp<RecordHandle> recordHandle; 5682 sp<Client> client; 5683 status_t lStatus; 5684 RecordThread *thread; 5685 size_t inFrameCount; 5686 int lSessionId; 5687 5688 // check calling permissions 5689 if (!recordingAllowed()) { 5690 lStatus = PERMISSION_DENIED; 5691 goto Exit; 5692 } 5693 5694 // add client to list 5695 { // scope for mLock 5696 Mutex::Autolock _l(mLock); 5697 thread = checkRecordThread_l(input); 5698 if (thread == NULL) { 5699 lStatus = BAD_VALUE; 5700 goto Exit; 5701 } 5702 5703 client = registerPid_l(pid); 5704 5705 // If no audio session id is provided, create one here 5706 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5707 lSessionId = *sessionId; 5708 } else { 5709 lSessionId = nextUniqueId(); 5710 if (sessionId != NULL) { 5711 *sessionId = lSessionId; 5712 } 5713 } 5714 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5715 recordTrack = thread->createRecordTrack_l(client, 5716 sampleRate, 5717 format, 5718 channelMask, 5719 frameCount, 5720 lSessionId, 5721 &lStatus); 5722 } 5723 if (lStatus != NO_ERROR) { 5724 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5725 // destructor is called by the TrackBase destructor with mLock held 5726 client.clear(); 5727 recordTrack.clear(); 5728 goto Exit; 5729 } 5730 5731 // return to handle to client 5732 recordHandle = new RecordHandle(recordTrack); 5733 lStatus = NO_ERROR; 5734 5735Exit: 5736 if (status) { 5737 *status = lStatus; 5738 } 5739 return recordHandle; 5740} 5741 5742// ---------------------------------------------------------------------------- 5743 5744AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5745 : BnAudioRecord(), 5746 mRecordTrack(recordTrack) 5747{ 5748} 5749 5750AudioFlinger::RecordHandle::~RecordHandle() { 5751 stop(); 5752} 5753 5754sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5755 return mRecordTrack->getCblk(); 5756} 5757 5758status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5759 ALOGV("RecordHandle::start()"); 5760 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5761} 5762 5763void AudioFlinger::RecordHandle::stop() { 5764 ALOGV("RecordHandle::stop()"); 5765 mRecordTrack->stop(); 5766} 5767 5768status_t AudioFlinger::RecordHandle::onTransact( 5769 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5770{ 5771 return BnAudioRecord::onTransact(code, data, reply, flags); 5772} 5773 5774// ---------------------------------------------------------------------------- 5775 5776AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5777 AudioStreamIn *input, 5778 uint32_t sampleRate, 5779 uint32_t channels, 5780 audio_io_handle_t id, 5781 uint32_t device) : 5782 ThreadBase(audioFlinger, id, device, RECORD), 5783 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5784 // mRsmpInIndex and mInputBytes set by readInputParameters() 5785 mReqChannelCount(popcount(channels)), 5786 mReqSampleRate(sampleRate) 5787 // mBytesRead is only meaningful while active, and so is cleared in start() 5788 // (but might be better to also clear here for dump?) 5789{ 5790 snprintf(mName, kNameLength, "AudioIn_%X", id); 5791 5792 readInputParameters(); 5793} 5794 5795 5796AudioFlinger::RecordThread::~RecordThread() 5797{ 5798 delete[] mRsmpInBuffer; 5799 delete mResampler; 5800 delete[] mRsmpOutBuffer; 5801} 5802 5803void AudioFlinger::RecordThread::onFirstRef() 5804{ 5805 run(mName, PRIORITY_URGENT_AUDIO); 5806} 5807 5808status_t AudioFlinger::RecordThread::readyToRun() 5809{ 5810 status_t status = initCheck(); 5811 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5812 return status; 5813} 5814 5815bool AudioFlinger::RecordThread::threadLoop() 5816{ 5817 AudioBufferProvider::Buffer buffer; 5818 sp<RecordTrack> activeTrack; 5819 Vector< sp<EffectChain> > effectChains; 5820 5821 nsecs_t lastWarning = 0; 5822 5823 acquireWakeLock(); 5824 5825 // start recording 5826 while (!exitPending()) { 5827 5828 processConfigEvents(); 5829 5830 { // scope for mLock 5831 Mutex::Autolock _l(mLock); 5832 checkForNewParameters_l(); 5833 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5834 if (!mStandby) { 5835 mInput->stream->common.standby(&mInput->stream->common); 5836 mStandby = true; 5837 } 5838 5839 if (exitPending()) break; 5840 5841 releaseWakeLock_l(); 5842 ALOGV("RecordThread: loop stopping"); 5843 // go to sleep 5844 mWaitWorkCV.wait(mLock); 5845 ALOGV("RecordThread: loop starting"); 5846 acquireWakeLock_l(); 5847 continue; 5848 } 5849 if (mActiveTrack != 0) { 5850 if (mActiveTrack->mState == TrackBase::PAUSING) { 5851 if (!mStandby) { 5852 mInput->stream->common.standby(&mInput->stream->common); 5853 mStandby = true; 5854 } 5855 mActiveTrack.clear(); 5856 mStartStopCond.broadcast(); 5857 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5858 if (mReqChannelCount != mActiveTrack->channelCount()) { 5859 mActiveTrack.clear(); 5860 mStartStopCond.broadcast(); 5861 } else if (mBytesRead != 0) { 5862 // record start succeeds only if first read from audio input 5863 // succeeds 5864 if (mBytesRead > 0) { 5865 mActiveTrack->mState = TrackBase::ACTIVE; 5866 } else { 5867 mActiveTrack.clear(); 5868 } 5869 mStartStopCond.broadcast(); 5870 } 5871 mStandby = false; 5872 } 5873 } 5874 lockEffectChains_l(effectChains); 5875 } 5876 5877 if (mActiveTrack != 0) { 5878 if (mActiveTrack->mState != TrackBase::ACTIVE && 5879 mActiveTrack->mState != TrackBase::RESUMING) { 5880 unlockEffectChains(effectChains); 5881 usleep(kRecordThreadSleepUs); 5882 continue; 5883 } 5884 for (size_t i = 0; i < effectChains.size(); i ++) { 5885 effectChains[i]->process_l(); 5886 } 5887 5888 buffer.frameCount = mFrameCount; 5889 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5890 size_t framesOut = buffer.frameCount; 5891 if (mResampler == NULL) { 5892 // no resampling 5893 while (framesOut) { 5894 size_t framesIn = mFrameCount - mRsmpInIndex; 5895 if (framesIn) { 5896 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5897 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5898 if (framesIn > framesOut) 5899 framesIn = framesOut; 5900 mRsmpInIndex += framesIn; 5901 framesOut -= framesIn; 5902 if ((int)mChannelCount == mReqChannelCount || 5903 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5904 memcpy(dst, src, framesIn * mFrameSize); 5905 } else { 5906 int16_t *src16 = (int16_t *)src; 5907 int16_t *dst16 = (int16_t *)dst; 5908 if (mChannelCount == 1) { 5909 while (framesIn--) { 5910 *dst16++ = *src16; 5911 *dst16++ = *src16++; 5912 } 5913 } else { 5914 while (framesIn--) { 5915 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5916 src16 += 2; 5917 } 5918 } 5919 } 5920 } 5921 if (framesOut && mFrameCount == mRsmpInIndex) { 5922 if (framesOut == mFrameCount && 5923 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5924 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5925 framesOut = 0; 5926 } else { 5927 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5928 mRsmpInIndex = 0; 5929 } 5930 if (mBytesRead < 0) { 5931 ALOGE("Error reading audio input"); 5932 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5933 // Force input into standby so that it tries to 5934 // recover at next read attempt 5935 mInput->stream->common.standby(&mInput->stream->common); 5936 usleep(kRecordThreadSleepUs); 5937 } 5938 mRsmpInIndex = mFrameCount; 5939 framesOut = 0; 5940 buffer.frameCount = 0; 5941 } 5942 } 5943 } 5944 } else { 5945 // resampling 5946 5947 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5948 // alter output frame count as if we were expecting stereo samples 5949 if (mChannelCount == 1 && mReqChannelCount == 1) { 5950 framesOut >>= 1; 5951 } 5952 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5953 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5954 // are 32 bit aligned which should be always true. 5955 if (mChannelCount == 2 && mReqChannelCount == 1) { 5956 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5957 // the resampler always outputs stereo samples: do post stereo to mono conversion 5958 int16_t *src = (int16_t *)mRsmpOutBuffer; 5959 int16_t *dst = buffer.i16; 5960 while (framesOut--) { 5961 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5962 src += 2; 5963 } 5964 } else { 5965 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5966 } 5967 5968 } 5969 if (mFramestoDrop == 0) { 5970 mActiveTrack->releaseBuffer(&buffer); 5971 } else { 5972 if (mFramestoDrop > 0) { 5973 mFramestoDrop -= buffer.frameCount; 5974 if (mFramestoDrop <= 0) { 5975 clearSyncStartEvent(); 5976 } 5977 } else { 5978 mFramestoDrop += buffer.frameCount; 5979 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 5980 mSyncStartEvent->isCancelled()) { 5981 ALOGW("Synced record %s, session %d, trigger session %d", 5982 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 5983 mActiveTrack->sessionId(), 5984 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 5985 clearSyncStartEvent(); 5986 } 5987 } 5988 } 5989 mActiveTrack->overflow(); 5990 } 5991 // client isn't retrieving buffers fast enough 5992 else { 5993 if (!mActiveTrack->setOverflow()) { 5994 nsecs_t now = systemTime(); 5995 if ((now - lastWarning) > kWarningThrottleNs) { 5996 ALOGW("RecordThread: buffer overflow"); 5997 lastWarning = now; 5998 } 5999 } 6000 // Release the processor for a while before asking for a new buffer. 6001 // This will give the application more chance to read from the buffer and 6002 // clear the overflow. 6003 usleep(kRecordThreadSleepUs); 6004 } 6005 } 6006 // enable changes in effect chain 6007 unlockEffectChains(effectChains); 6008 effectChains.clear(); 6009 } 6010 6011 if (!mStandby) { 6012 mInput->stream->common.standby(&mInput->stream->common); 6013 } 6014 mActiveTrack.clear(); 6015 6016 mStartStopCond.broadcast(); 6017 6018 releaseWakeLock(); 6019 6020 ALOGV("RecordThread %p exiting", this); 6021 return false; 6022} 6023 6024 6025sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6026 const sp<AudioFlinger::Client>& client, 6027 uint32_t sampleRate, 6028 audio_format_t format, 6029 int channelMask, 6030 int frameCount, 6031 int sessionId, 6032 status_t *status) 6033{ 6034 sp<RecordTrack> track; 6035 status_t lStatus; 6036 6037 lStatus = initCheck(); 6038 if (lStatus != NO_ERROR) { 6039 ALOGE("Audio driver not initialized."); 6040 goto Exit; 6041 } 6042 6043 { // scope for mLock 6044 Mutex::Autolock _l(mLock); 6045 6046 track = new RecordTrack(this, client, sampleRate, 6047 format, channelMask, frameCount, sessionId); 6048 6049 if (track->getCblk() == 0) { 6050 lStatus = NO_MEMORY; 6051 goto Exit; 6052 } 6053 6054 mTrack = track.get(); 6055 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6056 bool suspend = audio_is_bluetooth_sco_device( 6057 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6058 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6059 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6060 } 6061 lStatus = NO_ERROR; 6062 6063Exit: 6064 if (status) { 6065 *status = lStatus; 6066 } 6067 return track; 6068} 6069 6070status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6071 AudioSystem::sync_event_t event, 6072 int triggerSession) 6073{ 6074 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6075 sp<ThreadBase> strongMe = this; 6076 status_t status = NO_ERROR; 6077 6078 if (event == AudioSystem::SYNC_EVENT_NONE) { 6079 clearSyncStartEvent(); 6080 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6081 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6082 triggerSession, 6083 recordTrack->sessionId(), 6084 syncStartEventCallback, 6085 this); 6086 // Sync event can be cancelled by the trigger session if the track is not in a 6087 // compatible state in which case we start record immediately 6088 if (mSyncStartEvent->isCancelled()) { 6089 clearSyncStartEvent(); 6090 } else { 6091 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6092 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6093 } 6094 } 6095 6096 { 6097 AutoMutex lock(mLock); 6098 if (mActiveTrack != 0) { 6099 if (recordTrack != mActiveTrack.get()) { 6100 status = -EBUSY; 6101 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6102 mActiveTrack->mState = TrackBase::ACTIVE; 6103 } 6104 return status; 6105 } 6106 6107 recordTrack->mState = TrackBase::IDLE; 6108 mActiveTrack = recordTrack; 6109 mLock.unlock(); 6110 status_t status = AudioSystem::startInput(mId); 6111 mLock.lock(); 6112 if (status != NO_ERROR) { 6113 mActiveTrack.clear(); 6114 clearSyncStartEvent(); 6115 return status; 6116 } 6117 mRsmpInIndex = mFrameCount; 6118 mBytesRead = 0; 6119 if (mResampler != NULL) { 6120 mResampler->reset(); 6121 } 6122 mActiveTrack->mState = TrackBase::RESUMING; 6123 // signal thread to start 6124 ALOGV("Signal record thread"); 6125 mWaitWorkCV.signal(); 6126 // do not wait for mStartStopCond if exiting 6127 if (exitPending()) { 6128 mActiveTrack.clear(); 6129 status = INVALID_OPERATION; 6130 goto startError; 6131 } 6132 mStartStopCond.wait(mLock); 6133 if (mActiveTrack == 0) { 6134 ALOGV("Record failed to start"); 6135 status = BAD_VALUE; 6136 goto startError; 6137 } 6138 ALOGV("Record started OK"); 6139 return status; 6140 } 6141startError: 6142 AudioSystem::stopInput(mId); 6143 clearSyncStartEvent(); 6144 return status; 6145} 6146 6147void AudioFlinger::RecordThread::clearSyncStartEvent() 6148{ 6149 if (mSyncStartEvent != 0) { 6150 mSyncStartEvent->cancel(); 6151 } 6152 mSyncStartEvent.clear(); 6153 mFramestoDrop = 0; 6154} 6155 6156void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6157{ 6158 sp<SyncEvent> strongEvent = event.promote(); 6159 6160 if (strongEvent != 0) { 6161 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6162 me->handleSyncStartEvent(strongEvent); 6163 } 6164} 6165 6166void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6167{ 6168 if (event == mSyncStartEvent) { 6169 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6170 // from audio HAL 6171 mFramestoDrop = mFrameCount * 2; 6172 } 6173} 6174 6175void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6176 ALOGV("RecordThread::stop"); 6177 sp<ThreadBase> strongMe = this; 6178 { 6179 AutoMutex lock(mLock); 6180 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6181 mActiveTrack->mState = TrackBase::PAUSING; 6182 // do not wait for mStartStopCond if exiting 6183 if (exitPending()) { 6184 return; 6185 } 6186 mStartStopCond.wait(mLock); 6187 // if we have been restarted, recordTrack == mActiveTrack.get() here 6188 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6189 mLock.unlock(); 6190 AudioSystem::stopInput(mId); 6191 mLock.lock(); 6192 ALOGV("Record stopped OK"); 6193 } 6194 } 6195 } 6196} 6197 6198bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6199{ 6200 return false; 6201} 6202 6203status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6204{ 6205 if (!isValidSyncEvent(event)) { 6206 return BAD_VALUE; 6207 } 6208 6209 Mutex::Autolock _l(mLock); 6210 6211 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6212 mTrack->setSyncEvent(event); 6213 return NO_ERROR; 6214 } 6215 return NAME_NOT_FOUND; 6216} 6217 6218status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6219{ 6220 const size_t SIZE = 256; 6221 char buffer[SIZE]; 6222 String8 result; 6223 6224 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6225 result.append(buffer); 6226 6227 if (mActiveTrack != 0) { 6228 result.append("Active Track:\n"); 6229 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6230 mActiveTrack->dump(buffer, SIZE); 6231 result.append(buffer); 6232 6233 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6234 result.append(buffer); 6235 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6236 result.append(buffer); 6237 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6238 result.append(buffer); 6239 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6240 result.append(buffer); 6241 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6242 result.append(buffer); 6243 6244 6245 } else { 6246 result.append("No record client\n"); 6247 } 6248 write(fd, result.string(), result.size()); 6249 6250 dumpBase(fd, args); 6251 dumpEffectChains(fd, args); 6252 6253 return NO_ERROR; 6254} 6255 6256// AudioBufferProvider interface 6257status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6258{ 6259 size_t framesReq = buffer->frameCount; 6260 size_t framesReady = mFrameCount - mRsmpInIndex; 6261 int channelCount; 6262 6263 if (framesReady == 0) { 6264 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6265 if (mBytesRead < 0) { 6266 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6267 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6268 // Force input into standby so that it tries to 6269 // recover at next read attempt 6270 mInput->stream->common.standby(&mInput->stream->common); 6271 usleep(kRecordThreadSleepUs); 6272 } 6273 buffer->raw = NULL; 6274 buffer->frameCount = 0; 6275 return NOT_ENOUGH_DATA; 6276 } 6277 mRsmpInIndex = 0; 6278 framesReady = mFrameCount; 6279 } 6280 6281 if (framesReq > framesReady) { 6282 framesReq = framesReady; 6283 } 6284 6285 if (mChannelCount == 1 && mReqChannelCount == 2) { 6286 channelCount = 1; 6287 } else { 6288 channelCount = 2; 6289 } 6290 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6291 buffer->frameCount = framesReq; 6292 return NO_ERROR; 6293} 6294 6295// AudioBufferProvider interface 6296void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6297{ 6298 mRsmpInIndex += buffer->frameCount; 6299 buffer->frameCount = 0; 6300} 6301 6302bool AudioFlinger::RecordThread::checkForNewParameters_l() 6303{ 6304 bool reconfig = false; 6305 6306 while (!mNewParameters.isEmpty()) { 6307 status_t status = NO_ERROR; 6308 String8 keyValuePair = mNewParameters[0]; 6309 AudioParameter param = AudioParameter(keyValuePair); 6310 int value; 6311 audio_format_t reqFormat = mFormat; 6312 int reqSamplingRate = mReqSampleRate; 6313 int reqChannelCount = mReqChannelCount; 6314 6315 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6316 reqSamplingRate = value; 6317 reconfig = true; 6318 } 6319 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6320 reqFormat = (audio_format_t) value; 6321 reconfig = true; 6322 } 6323 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6324 reqChannelCount = popcount(value); 6325 reconfig = true; 6326 } 6327 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6328 // do not accept frame count changes if tracks are open as the track buffer 6329 // size depends on frame count and correct behavior would not be guaranteed 6330 // if frame count is changed after track creation 6331 if (mActiveTrack != 0) { 6332 status = INVALID_OPERATION; 6333 } else { 6334 reconfig = true; 6335 } 6336 } 6337 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6338 // forward device change to effects that have requested to be 6339 // aware of attached audio device. 6340 for (size_t i = 0; i < mEffectChains.size(); i++) { 6341 mEffectChains[i]->setDevice_l(value); 6342 } 6343 // store input device and output device but do not forward output device to audio HAL. 6344 // Note that status is ignored by the caller for output device 6345 // (see AudioFlinger::setParameters() 6346 if (value & AUDIO_DEVICE_OUT_ALL) { 6347 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6348 status = BAD_VALUE; 6349 } else { 6350 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6351 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6352 if (mTrack != NULL) { 6353 bool suspend = audio_is_bluetooth_sco_device( 6354 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6355 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6356 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6357 } 6358 } 6359 mDevice |= (uint32_t)value; 6360 } 6361 if (status == NO_ERROR) { 6362 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6363 if (status == INVALID_OPERATION) { 6364 mInput->stream->common.standby(&mInput->stream->common); 6365 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6366 keyValuePair.string()); 6367 } 6368 if (reconfig) { 6369 if (status == BAD_VALUE && 6370 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6371 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6372 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6373 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6374 (reqChannelCount <= FCC_2)) { 6375 status = NO_ERROR; 6376 } 6377 if (status == NO_ERROR) { 6378 readInputParameters(); 6379 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6380 } 6381 } 6382 } 6383 6384 mNewParameters.removeAt(0); 6385 6386 mParamStatus = status; 6387 mParamCond.signal(); 6388 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6389 // already timed out waiting for the status and will never signal the condition. 6390 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6391 } 6392 return reconfig; 6393} 6394 6395String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6396{ 6397 char *s; 6398 String8 out_s8 = String8(); 6399 6400 Mutex::Autolock _l(mLock); 6401 if (initCheck() != NO_ERROR) { 6402 return out_s8; 6403 } 6404 6405 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6406 out_s8 = String8(s); 6407 free(s); 6408 return out_s8; 6409} 6410 6411void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6412 AudioSystem::OutputDescriptor desc; 6413 void *param2 = NULL; 6414 6415 switch (event) { 6416 case AudioSystem::INPUT_OPENED: 6417 case AudioSystem::INPUT_CONFIG_CHANGED: 6418 desc.channels = mChannelMask; 6419 desc.samplingRate = mSampleRate; 6420 desc.format = mFormat; 6421 desc.frameCount = mFrameCount; 6422 desc.latency = 0; 6423 param2 = &desc; 6424 break; 6425 6426 case AudioSystem::INPUT_CLOSED: 6427 default: 6428 break; 6429 } 6430 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6431} 6432 6433void AudioFlinger::RecordThread::readInputParameters() 6434{ 6435 delete mRsmpInBuffer; 6436 // mRsmpInBuffer is always assigned a new[] below 6437 delete mRsmpOutBuffer; 6438 mRsmpOutBuffer = NULL; 6439 delete mResampler; 6440 mResampler = NULL; 6441 6442 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6443 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6444 mChannelCount = (uint16_t)popcount(mChannelMask); 6445 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6446 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6447 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6448 mFrameCount = mInputBytes / mFrameSize; 6449 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6450 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6451 6452 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6453 { 6454 int channelCount; 6455 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6456 // stereo to mono post process as the resampler always outputs stereo. 6457 if (mChannelCount == 1 && mReqChannelCount == 2) { 6458 channelCount = 1; 6459 } else { 6460 channelCount = 2; 6461 } 6462 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6463 mResampler->setSampleRate(mSampleRate); 6464 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6465 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6466 6467 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6468 if (mChannelCount == 1 && mReqChannelCount == 1) { 6469 mFrameCount >>= 1; 6470 } 6471 6472 } 6473 mRsmpInIndex = mFrameCount; 6474} 6475 6476unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6477{ 6478 Mutex::Autolock _l(mLock); 6479 if (initCheck() != NO_ERROR) { 6480 return 0; 6481 } 6482 6483 return mInput->stream->get_input_frames_lost(mInput->stream); 6484} 6485 6486uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6487{ 6488 Mutex::Autolock _l(mLock); 6489 uint32_t result = 0; 6490 if (getEffectChain_l(sessionId) != 0) { 6491 result = EFFECT_SESSION; 6492 } 6493 6494 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6495 result |= TRACK_SESSION; 6496 } 6497 6498 return result; 6499} 6500 6501AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6502{ 6503 Mutex::Autolock _l(mLock); 6504 return mTrack; 6505} 6506 6507AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6508{ 6509 Mutex::Autolock _l(mLock); 6510 return mInput; 6511} 6512 6513AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6514{ 6515 Mutex::Autolock _l(mLock); 6516 AudioStreamIn *input = mInput; 6517 mInput = NULL; 6518 return input; 6519} 6520 6521// this method must always be called either with ThreadBase mLock held or inside the thread loop 6522audio_stream_t* AudioFlinger::RecordThread::stream() const 6523{ 6524 if (mInput == NULL) { 6525 return NULL; 6526 } 6527 return &mInput->stream->common; 6528} 6529 6530 6531// ---------------------------------------------------------------------------- 6532 6533audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6534{ 6535 if (!settingsAllowed()) { 6536 return 0; 6537 } 6538 Mutex::Autolock _l(mLock); 6539 return loadHwModule_l(name); 6540} 6541 6542// loadHwModule_l() must be called with AudioFlinger::mLock held 6543audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6544{ 6545 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6546 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6547 ALOGW("loadHwModule() module %s already loaded", name); 6548 return mAudioHwDevs.keyAt(i); 6549 } 6550 } 6551 6552 audio_hw_device_t *dev; 6553 6554 int rc = load_audio_interface(name, &dev); 6555 if (rc) { 6556 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6557 return 0; 6558 } 6559 6560 mHardwareStatus = AUDIO_HW_INIT; 6561 rc = dev->init_check(dev); 6562 mHardwareStatus = AUDIO_HW_IDLE; 6563 if (rc) { 6564 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6565 return 0; 6566 } 6567 6568 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6569 (NULL != dev->set_master_volume)) { 6570 AutoMutex lock(mHardwareLock); 6571 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6572 dev->set_master_volume(dev, mMasterVolume); 6573 mHardwareStatus = AUDIO_HW_IDLE; 6574 } 6575 6576 audio_module_handle_t handle = nextUniqueId(); 6577 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6578 6579 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6580 name, dev->common.module->name, dev->common.module->id, handle); 6581 6582 return handle; 6583 6584} 6585 6586audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6587 audio_devices_t *pDevices, 6588 uint32_t *pSamplingRate, 6589 audio_format_t *pFormat, 6590 audio_channel_mask_t *pChannelMask, 6591 uint32_t *pLatencyMs, 6592 audio_output_flags_t flags) 6593{ 6594 status_t status; 6595 PlaybackThread *thread = NULL; 6596 struct audio_config config = { 6597 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6598 channel_mask: pChannelMask ? *pChannelMask : 0, 6599 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6600 }; 6601 audio_stream_out_t *outStream = NULL; 6602 audio_hw_device_t *outHwDev; 6603 6604 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6605 module, 6606 (pDevices != NULL) ? (int)*pDevices : 0, 6607 config.sample_rate, 6608 config.format, 6609 config.channel_mask, 6610 flags); 6611 6612 if (pDevices == NULL || *pDevices == 0) { 6613 return 0; 6614 } 6615 6616 Mutex::Autolock _l(mLock); 6617 6618 outHwDev = findSuitableHwDev_l(module, *pDevices); 6619 if (outHwDev == NULL) 6620 return 0; 6621 6622 audio_io_handle_t id = nextUniqueId(); 6623 6624 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6625 6626 status = outHwDev->open_output_stream(outHwDev, 6627 id, 6628 *pDevices, 6629 (audio_output_flags_t)flags, 6630 &config, 6631 &outStream); 6632 6633 mHardwareStatus = AUDIO_HW_IDLE; 6634 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6635 outStream, 6636 config.sample_rate, 6637 config.format, 6638 config.channel_mask, 6639 status); 6640 6641 if (status == NO_ERROR && outStream != NULL) { 6642 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6643 6644 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6645 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6646 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6647 thread = new DirectOutputThread(this, output, id, *pDevices); 6648 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6649 } else { 6650 thread = new MixerThread(this, output, id, *pDevices); 6651 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6652 } 6653 mPlaybackThreads.add(id, thread); 6654 6655 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6656 if (pFormat != NULL) *pFormat = config.format; 6657 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6658 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6659 6660 // notify client processes of the new output creation 6661 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6662 6663 // the first primary output opened designates the primary hw device 6664 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6665 ALOGI("Using module %d has the primary audio interface", module); 6666 mPrimaryHardwareDev = outHwDev; 6667 6668 AutoMutex lock(mHardwareLock); 6669 mHardwareStatus = AUDIO_HW_SET_MODE; 6670 outHwDev->set_mode(outHwDev, mMode); 6671 6672 // Determine the level of master volume support the primary audio HAL has, 6673 // and set the initial master volume at the same time. 6674 float initialVolume = 1.0; 6675 mMasterVolumeSupportLvl = MVS_NONE; 6676 6677 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6678 if ((NULL != outHwDev->get_master_volume) && 6679 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6680 mMasterVolumeSupportLvl = MVS_FULL; 6681 } else { 6682 mMasterVolumeSupportLvl = MVS_SETONLY; 6683 initialVolume = 1.0; 6684 } 6685 6686 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6687 if ((NULL == outHwDev->set_master_volume) || 6688 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6689 mMasterVolumeSupportLvl = MVS_NONE; 6690 } 6691 // now that we have a primary device, initialize master volume on other devices 6692 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6693 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6694 6695 if ((dev != mPrimaryHardwareDev) && 6696 (NULL != dev->set_master_volume)) { 6697 dev->set_master_volume(dev, initialVolume); 6698 } 6699 } 6700 mHardwareStatus = AUDIO_HW_IDLE; 6701 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6702 ? initialVolume 6703 : 1.0; 6704 mMasterVolume = initialVolume; 6705 } 6706 return id; 6707 } 6708 6709 return 0; 6710} 6711 6712audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6713 audio_io_handle_t output2) 6714{ 6715 Mutex::Autolock _l(mLock); 6716 MixerThread *thread1 = checkMixerThread_l(output1); 6717 MixerThread *thread2 = checkMixerThread_l(output2); 6718 6719 if (thread1 == NULL || thread2 == NULL) { 6720 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6721 return 0; 6722 } 6723 6724 audio_io_handle_t id = nextUniqueId(); 6725 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6726 thread->addOutputTrack(thread2); 6727 mPlaybackThreads.add(id, thread); 6728 // notify client processes of the new output creation 6729 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6730 return id; 6731} 6732 6733status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6734{ 6735 // keep strong reference on the playback thread so that 6736 // it is not destroyed while exit() is executed 6737 sp<PlaybackThread> thread; 6738 { 6739 Mutex::Autolock _l(mLock); 6740 thread = checkPlaybackThread_l(output); 6741 if (thread == NULL) { 6742 return BAD_VALUE; 6743 } 6744 6745 ALOGV("closeOutput() %d", output); 6746 6747 if (thread->type() == ThreadBase::MIXER) { 6748 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6749 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6750 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6751 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6752 } 6753 } 6754 } 6755 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6756 mPlaybackThreads.removeItem(output); 6757 } 6758 thread->exit(); 6759 // The thread entity (active unit of execution) is no longer running here, 6760 // but the ThreadBase container still exists. 6761 6762 if (thread->type() != ThreadBase::DUPLICATING) { 6763 AudioStreamOut *out = thread->clearOutput(); 6764 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6765 // from now on thread->mOutput is NULL 6766 out->hwDev->close_output_stream(out->hwDev, out->stream); 6767 delete out; 6768 } 6769 return NO_ERROR; 6770} 6771 6772status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6773{ 6774 Mutex::Autolock _l(mLock); 6775 PlaybackThread *thread = checkPlaybackThread_l(output); 6776 6777 if (thread == NULL) { 6778 return BAD_VALUE; 6779 } 6780 6781 ALOGV("suspendOutput() %d", output); 6782 thread->suspend(); 6783 6784 return NO_ERROR; 6785} 6786 6787status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6788{ 6789 Mutex::Autolock _l(mLock); 6790 PlaybackThread *thread = checkPlaybackThread_l(output); 6791 6792 if (thread == NULL) { 6793 return BAD_VALUE; 6794 } 6795 6796 ALOGV("restoreOutput() %d", output); 6797 6798 thread->restore(); 6799 6800 return NO_ERROR; 6801} 6802 6803audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6804 audio_devices_t *pDevices, 6805 uint32_t *pSamplingRate, 6806 audio_format_t *pFormat, 6807 uint32_t *pChannelMask) 6808{ 6809 status_t status; 6810 RecordThread *thread = NULL; 6811 struct audio_config config = { 6812 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6813 channel_mask: pChannelMask ? *pChannelMask : 0, 6814 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6815 }; 6816 uint32_t reqSamplingRate = config.sample_rate; 6817 audio_format_t reqFormat = config.format; 6818 audio_channel_mask_t reqChannels = config.channel_mask; 6819 audio_stream_in_t *inStream = NULL; 6820 audio_hw_device_t *inHwDev; 6821 6822 if (pDevices == NULL || *pDevices == 0) { 6823 return 0; 6824 } 6825 6826 Mutex::Autolock _l(mLock); 6827 6828 inHwDev = findSuitableHwDev_l(module, *pDevices); 6829 if (inHwDev == NULL) 6830 return 0; 6831 6832 audio_io_handle_t id = nextUniqueId(); 6833 6834 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6835 &inStream); 6836 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6837 inStream, 6838 config.sample_rate, 6839 config.format, 6840 config.channel_mask, 6841 status); 6842 6843 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6844 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6845 // or stereo to mono conversions on 16 bit PCM inputs. 6846 if (status == BAD_VALUE && 6847 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6848 (config.sample_rate <= 2 * reqSamplingRate) && 6849 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6850 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6851 inStream = NULL; 6852 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6853 } 6854 6855 if (status == NO_ERROR && inStream != NULL) { 6856 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6857 6858 // Start record thread 6859 // RecorThread require both input and output device indication to forward to audio 6860 // pre processing modules 6861 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6862 thread = new RecordThread(this, 6863 input, 6864 reqSamplingRate, 6865 reqChannels, 6866 id, 6867 device); 6868 mRecordThreads.add(id, thread); 6869 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6870 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6871 if (pFormat != NULL) *pFormat = config.format; 6872 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6873 6874 input->stream->common.standby(&input->stream->common); 6875 6876 // notify client processes of the new input creation 6877 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6878 return id; 6879 } 6880 6881 return 0; 6882} 6883 6884status_t AudioFlinger::closeInput(audio_io_handle_t input) 6885{ 6886 // keep strong reference on the record thread so that 6887 // it is not destroyed while exit() is executed 6888 sp<RecordThread> thread; 6889 { 6890 Mutex::Autolock _l(mLock); 6891 thread = checkRecordThread_l(input); 6892 if (thread == NULL) { 6893 return BAD_VALUE; 6894 } 6895 6896 ALOGV("closeInput() %d", input); 6897 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6898 mRecordThreads.removeItem(input); 6899 } 6900 thread->exit(); 6901 // The thread entity (active unit of execution) is no longer running here, 6902 // but the ThreadBase container still exists. 6903 6904 AudioStreamIn *in = thread->clearInput(); 6905 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6906 // from now on thread->mInput is NULL 6907 in->hwDev->close_input_stream(in->hwDev, in->stream); 6908 delete in; 6909 6910 return NO_ERROR; 6911} 6912 6913status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6914{ 6915 Mutex::Autolock _l(mLock); 6916 MixerThread *dstThread = checkMixerThread_l(output); 6917 if (dstThread == NULL) { 6918 ALOGW("setStreamOutput() bad output id %d", output); 6919 return BAD_VALUE; 6920 } 6921 6922 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6923 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6924 6925 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6926 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6927 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6928 MixerThread *srcThread = (MixerThread *)thread; 6929 srcThread->invalidateTracks(stream); 6930 } 6931 } 6932 6933 return NO_ERROR; 6934} 6935 6936 6937int AudioFlinger::newAudioSessionId() 6938{ 6939 return nextUniqueId(); 6940} 6941 6942void AudioFlinger::acquireAudioSessionId(int audioSession) 6943{ 6944 Mutex::Autolock _l(mLock); 6945 pid_t caller = IPCThreadState::self()->getCallingPid(); 6946 ALOGV("acquiring %d from %d", audioSession, caller); 6947 size_t num = mAudioSessionRefs.size(); 6948 for (size_t i = 0; i< num; i++) { 6949 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6950 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6951 ref->mCnt++; 6952 ALOGV(" incremented refcount to %d", ref->mCnt); 6953 return; 6954 } 6955 } 6956 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6957 ALOGV(" added new entry for %d", audioSession); 6958} 6959 6960void AudioFlinger::releaseAudioSessionId(int audioSession) 6961{ 6962 Mutex::Autolock _l(mLock); 6963 pid_t caller = IPCThreadState::self()->getCallingPid(); 6964 ALOGV("releasing %d from %d", audioSession, caller); 6965 size_t num = mAudioSessionRefs.size(); 6966 for (size_t i = 0; i< num; i++) { 6967 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6968 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6969 ref->mCnt--; 6970 ALOGV(" decremented refcount to %d", ref->mCnt); 6971 if (ref->mCnt == 0) { 6972 mAudioSessionRefs.removeAt(i); 6973 delete ref; 6974 purgeStaleEffects_l(); 6975 } 6976 return; 6977 } 6978 } 6979 ALOGW("session id %d not found for pid %d", audioSession, caller); 6980} 6981 6982void AudioFlinger::purgeStaleEffects_l() { 6983 6984 ALOGV("purging stale effects"); 6985 6986 Vector< sp<EffectChain> > chains; 6987 6988 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6989 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6990 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6991 sp<EffectChain> ec = t->mEffectChains[j]; 6992 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6993 chains.push(ec); 6994 } 6995 } 6996 } 6997 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6998 sp<RecordThread> t = mRecordThreads.valueAt(i); 6999 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7000 sp<EffectChain> ec = t->mEffectChains[j]; 7001 chains.push(ec); 7002 } 7003 } 7004 7005 for (size_t i = 0; i < chains.size(); i++) { 7006 sp<EffectChain> ec = chains[i]; 7007 int sessionid = ec->sessionId(); 7008 sp<ThreadBase> t = ec->mThread.promote(); 7009 if (t == 0) { 7010 continue; 7011 } 7012 size_t numsessionrefs = mAudioSessionRefs.size(); 7013 bool found = false; 7014 for (size_t k = 0; k < numsessionrefs; k++) { 7015 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7016 if (ref->mSessionid == sessionid) { 7017 ALOGV(" session %d still exists for %d with %d refs", 7018 sessionid, ref->mPid, ref->mCnt); 7019 found = true; 7020 break; 7021 } 7022 } 7023 if (!found) { 7024 // remove all effects from the chain 7025 while (ec->mEffects.size()) { 7026 sp<EffectModule> effect = ec->mEffects[0]; 7027 effect->unPin(); 7028 Mutex::Autolock _l (t->mLock); 7029 t->removeEffect_l(effect); 7030 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7031 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7032 if (handle != 0) { 7033 handle->mEffect.clear(); 7034 if (handle->mHasControl && handle->mEnabled) { 7035 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7036 } 7037 } 7038 } 7039 AudioSystem::unregisterEffect(effect->id()); 7040 } 7041 } 7042 } 7043 return; 7044} 7045 7046// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7047AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7048{ 7049 return mPlaybackThreads.valueFor(output).get(); 7050} 7051 7052// checkMixerThread_l() must be called with AudioFlinger::mLock held 7053AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7054{ 7055 PlaybackThread *thread = checkPlaybackThread_l(output); 7056 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7057} 7058 7059// checkRecordThread_l() must be called with AudioFlinger::mLock held 7060AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7061{ 7062 return mRecordThreads.valueFor(input).get(); 7063} 7064 7065uint32_t AudioFlinger::nextUniqueId() 7066{ 7067 return android_atomic_inc(&mNextUniqueId); 7068} 7069 7070AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7071{ 7072 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7073 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7074 AudioStreamOut *output = thread->getOutput(); 7075 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7076 return thread; 7077 } 7078 } 7079 return NULL; 7080} 7081 7082uint32_t AudioFlinger::primaryOutputDevice_l() const 7083{ 7084 PlaybackThread *thread = primaryPlaybackThread_l(); 7085 7086 if (thread == NULL) { 7087 return 0; 7088 } 7089 7090 return thread->device(); 7091} 7092 7093sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7094 int triggerSession, 7095 int listenerSession, 7096 sync_event_callback_t callBack, 7097 void *cookie) 7098{ 7099 Mutex::Autolock _l(mLock); 7100 7101 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7102 status_t playStatus = NAME_NOT_FOUND; 7103 status_t recStatus = NAME_NOT_FOUND; 7104 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7105 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7106 if (playStatus == NO_ERROR) { 7107 return event; 7108 } 7109 } 7110 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7111 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7112 if (recStatus == NO_ERROR) { 7113 return event; 7114 } 7115 } 7116 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7117 mPendingSyncEvents.add(event); 7118 } else { 7119 ALOGV("createSyncEvent() invalid event %d", event->type()); 7120 event.clear(); 7121 } 7122 return event; 7123} 7124 7125// ---------------------------------------------------------------------------- 7126// Effect management 7127// ---------------------------------------------------------------------------- 7128 7129 7130status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7131{ 7132 Mutex::Autolock _l(mLock); 7133 return EffectQueryNumberEffects(numEffects); 7134} 7135 7136status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7137{ 7138 Mutex::Autolock _l(mLock); 7139 return EffectQueryEffect(index, descriptor); 7140} 7141 7142status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7143 effect_descriptor_t *descriptor) const 7144{ 7145 Mutex::Autolock _l(mLock); 7146 return EffectGetDescriptor(pUuid, descriptor); 7147} 7148 7149 7150sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7151 effect_descriptor_t *pDesc, 7152 const sp<IEffectClient>& effectClient, 7153 int32_t priority, 7154 audio_io_handle_t io, 7155 int sessionId, 7156 status_t *status, 7157 int *id, 7158 int *enabled) 7159{ 7160 status_t lStatus = NO_ERROR; 7161 sp<EffectHandle> handle; 7162 effect_descriptor_t desc; 7163 7164 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7165 pid, effectClient.get(), priority, sessionId, io); 7166 7167 if (pDesc == NULL) { 7168 lStatus = BAD_VALUE; 7169 goto Exit; 7170 } 7171 7172 // check audio settings permission for global effects 7173 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7174 lStatus = PERMISSION_DENIED; 7175 goto Exit; 7176 } 7177 7178 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7179 // that can only be created by audio policy manager (running in same process) 7180 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7181 lStatus = PERMISSION_DENIED; 7182 goto Exit; 7183 } 7184 7185 if (io == 0) { 7186 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7187 // output must be specified by AudioPolicyManager when using session 7188 // AUDIO_SESSION_OUTPUT_STAGE 7189 lStatus = BAD_VALUE; 7190 goto Exit; 7191 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7192 // if the output returned by getOutputForEffect() is removed before we lock the 7193 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7194 // and we will exit safely 7195 io = AudioSystem::getOutputForEffect(&desc); 7196 } 7197 } 7198 7199 { 7200 Mutex::Autolock _l(mLock); 7201 7202 7203 if (!EffectIsNullUuid(&pDesc->uuid)) { 7204 // if uuid is specified, request effect descriptor 7205 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7206 if (lStatus < 0) { 7207 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7208 goto Exit; 7209 } 7210 } else { 7211 // if uuid is not specified, look for an available implementation 7212 // of the required type in effect factory 7213 if (EffectIsNullUuid(&pDesc->type)) { 7214 ALOGW("createEffect() no effect type"); 7215 lStatus = BAD_VALUE; 7216 goto Exit; 7217 } 7218 uint32_t numEffects = 0; 7219 effect_descriptor_t d; 7220 d.flags = 0; // prevent compiler warning 7221 bool found = false; 7222 7223 lStatus = EffectQueryNumberEffects(&numEffects); 7224 if (lStatus < 0) { 7225 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7226 goto Exit; 7227 } 7228 for (uint32_t i = 0; i < numEffects; i++) { 7229 lStatus = EffectQueryEffect(i, &desc); 7230 if (lStatus < 0) { 7231 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7232 continue; 7233 } 7234 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7235 // If matching type found save effect descriptor. If the session is 7236 // 0 and the effect is not auxiliary, continue enumeration in case 7237 // an auxiliary version of this effect type is available 7238 found = true; 7239 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7240 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7241 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7242 break; 7243 } 7244 } 7245 } 7246 if (!found) { 7247 lStatus = BAD_VALUE; 7248 ALOGW("createEffect() effect not found"); 7249 goto Exit; 7250 } 7251 // For same effect type, chose auxiliary version over insert version if 7252 // connect to output mix (Compliance to OpenSL ES) 7253 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7254 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7255 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7256 } 7257 } 7258 7259 // Do not allow auxiliary effects on a session different from 0 (output mix) 7260 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7261 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7262 lStatus = INVALID_OPERATION; 7263 goto Exit; 7264 } 7265 7266 // check recording permission for visualizer 7267 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7268 !recordingAllowed()) { 7269 lStatus = PERMISSION_DENIED; 7270 goto Exit; 7271 } 7272 7273 // return effect descriptor 7274 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7275 7276 // If output is not specified try to find a matching audio session ID in one of the 7277 // output threads. 7278 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7279 // because of code checking output when entering the function. 7280 // Note: io is never 0 when creating an effect on an input 7281 if (io == 0) { 7282 // look for the thread where the specified audio session is present 7283 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7284 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7285 io = mPlaybackThreads.keyAt(i); 7286 break; 7287 } 7288 } 7289 if (io == 0) { 7290 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7291 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7292 io = mRecordThreads.keyAt(i); 7293 break; 7294 } 7295 } 7296 } 7297 // If no output thread contains the requested session ID, default to 7298 // first output. The effect chain will be moved to the correct output 7299 // thread when a track with the same session ID is created 7300 if (io == 0 && mPlaybackThreads.size()) { 7301 io = mPlaybackThreads.keyAt(0); 7302 } 7303 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7304 } 7305 ThreadBase *thread = checkRecordThread_l(io); 7306 if (thread == NULL) { 7307 thread = checkPlaybackThread_l(io); 7308 if (thread == NULL) { 7309 ALOGE("createEffect() unknown output thread"); 7310 lStatus = BAD_VALUE; 7311 goto Exit; 7312 } 7313 } 7314 7315 sp<Client> client = registerPid_l(pid); 7316 7317 // create effect on selected output thread 7318 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7319 &desc, enabled, &lStatus); 7320 if (handle != 0 && id != NULL) { 7321 *id = handle->id(); 7322 } 7323 } 7324 7325Exit: 7326 if (status != NULL) { 7327 *status = lStatus; 7328 } 7329 return handle; 7330} 7331 7332status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7333 audio_io_handle_t dstOutput) 7334{ 7335 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7336 sessionId, srcOutput, dstOutput); 7337 Mutex::Autolock _l(mLock); 7338 if (srcOutput == dstOutput) { 7339 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7340 return NO_ERROR; 7341 } 7342 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7343 if (srcThread == NULL) { 7344 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7345 return BAD_VALUE; 7346 } 7347 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7348 if (dstThread == NULL) { 7349 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7350 return BAD_VALUE; 7351 } 7352 7353 Mutex::Autolock _dl(dstThread->mLock); 7354 Mutex::Autolock _sl(srcThread->mLock); 7355 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7356 7357 return NO_ERROR; 7358} 7359 7360// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7361status_t AudioFlinger::moveEffectChain_l(int sessionId, 7362 AudioFlinger::PlaybackThread *srcThread, 7363 AudioFlinger::PlaybackThread *dstThread, 7364 bool reRegister) 7365{ 7366 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7367 sessionId, srcThread, dstThread); 7368 7369 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7370 if (chain == 0) { 7371 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7372 sessionId, srcThread); 7373 return INVALID_OPERATION; 7374 } 7375 7376 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7377 // so that a new chain is created with correct parameters when first effect is added. This is 7378 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7379 // removed. 7380 srcThread->removeEffectChain_l(chain); 7381 7382 // transfer all effects one by one so that new effect chain is created on new thread with 7383 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7384 audio_io_handle_t dstOutput = dstThread->id(); 7385 sp<EffectChain> dstChain; 7386 uint32_t strategy = 0; // prevent compiler warning 7387 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7388 while (effect != 0) { 7389 srcThread->removeEffect_l(effect); 7390 dstThread->addEffect_l(effect); 7391 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7392 if (effect->state() == EffectModule::ACTIVE || 7393 effect->state() == EffectModule::STOPPING) { 7394 effect->start(); 7395 } 7396 // if the move request is not received from audio policy manager, the effect must be 7397 // re-registered with the new strategy and output 7398 if (dstChain == 0) { 7399 dstChain = effect->chain().promote(); 7400 if (dstChain == 0) { 7401 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7402 srcThread->addEffect_l(effect); 7403 return NO_INIT; 7404 } 7405 strategy = dstChain->strategy(); 7406 } 7407 if (reRegister) { 7408 AudioSystem::unregisterEffect(effect->id()); 7409 AudioSystem::registerEffect(&effect->desc(), 7410 dstOutput, 7411 strategy, 7412 sessionId, 7413 effect->id()); 7414 } 7415 effect = chain->getEffectFromId_l(0); 7416 } 7417 7418 return NO_ERROR; 7419} 7420 7421 7422// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7423sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7424 const sp<AudioFlinger::Client>& client, 7425 const sp<IEffectClient>& effectClient, 7426 int32_t priority, 7427 int sessionId, 7428 effect_descriptor_t *desc, 7429 int *enabled, 7430 status_t *status 7431 ) 7432{ 7433 sp<EffectModule> effect; 7434 sp<EffectHandle> handle; 7435 status_t lStatus; 7436 sp<EffectChain> chain; 7437 bool chainCreated = false; 7438 bool effectCreated = false; 7439 bool effectRegistered = false; 7440 7441 lStatus = initCheck(); 7442 if (lStatus != NO_ERROR) { 7443 ALOGW("createEffect_l() Audio driver not initialized."); 7444 goto Exit; 7445 } 7446 7447 // Do not allow effects with session ID 0 on direct output or duplicating threads 7448 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7449 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7450 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7451 desc->name, sessionId); 7452 lStatus = BAD_VALUE; 7453 goto Exit; 7454 } 7455 // Only Pre processor effects are allowed on input threads and only on input threads 7456 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7457 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7458 desc->name, desc->flags, mType); 7459 lStatus = BAD_VALUE; 7460 goto Exit; 7461 } 7462 7463 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7464 7465 { // scope for mLock 7466 Mutex::Autolock _l(mLock); 7467 7468 // check for existing effect chain with the requested audio session 7469 chain = getEffectChain_l(sessionId); 7470 if (chain == 0) { 7471 // create a new chain for this session 7472 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7473 chain = new EffectChain(this, sessionId); 7474 addEffectChain_l(chain); 7475 chain->setStrategy(getStrategyForSession_l(sessionId)); 7476 chainCreated = true; 7477 } else { 7478 effect = chain->getEffectFromDesc_l(desc); 7479 } 7480 7481 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7482 7483 if (effect == 0) { 7484 int id = mAudioFlinger->nextUniqueId(); 7485 // Check CPU and memory usage 7486 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7487 if (lStatus != NO_ERROR) { 7488 goto Exit; 7489 } 7490 effectRegistered = true; 7491 // create a new effect module if none present in the chain 7492 effect = new EffectModule(this, chain, desc, id, sessionId); 7493 lStatus = effect->status(); 7494 if (lStatus != NO_ERROR) { 7495 goto Exit; 7496 } 7497 lStatus = chain->addEffect_l(effect); 7498 if (lStatus != NO_ERROR) { 7499 goto Exit; 7500 } 7501 effectCreated = true; 7502 7503 effect->setDevice(mDevice); 7504 effect->setMode(mAudioFlinger->getMode()); 7505 } 7506 // create effect handle and connect it to effect module 7507 handle = new EffectHandle(effect, client, effectClient, priority); 7508 lStatus = effect->addHandle(handle); 7509 if (enabled != NULL) { 7510 *enabled = (int)effect->isEnabled(); 7511 } 7512 } 7513 7514Exit: 7515 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7516 Mutex::Autolock _l(mLock); 7517 if (effectCreated) { 7518 chain->removeEffect_l(effect); 7519 } 7520 if (effectRegistered) { 7521 AudioSystem::unregisterEffect(effect->id()); 7522 } 7523 if (chainCreated) { 7524 removeEffectChain_l(chain); 7525 } 7526 handle.clear(); 7527 } 7528 7529 if (status != NULL) { 7530 *status = lStatus; 7531 } 7532 return handle; 7533} 7534 7535sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7536{ 7537 sp<EffectChain> chain = getEffectChain_l(sessionId); 7538 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7539} 7540 7541// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7542// PlaybackThread::mLock held 7543status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7544{ 7545 // check for existing effect chain with the requested audio session 7546 int sessionId = effect->sessionId(); 7547 sp<EffectChain> chain = getEffectChain_l(sessionId); 7548 bool chainCreated = false; 7549 7550 if (chain == 0) { 7551 // create a new chain for this session 7552 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7553 chain = new EffectChain(this, sessionId); 7554 addEffectChain_l(chain); 7555 chain->setStrategy(getStrategyForSession_l(sessionId)); 7556 chainCreated = true; 7557 } 7558 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7559 7560 if (chain->getEffectFromId_l(effect->id()) != 0) { 7561 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7562 this, effect->desc().name, chain.get()); 7563 return BAD_VALUE; 7564 } 7565 7566 status_t status = chain->addEffect_l(effect); 7567 if (status != NO_ERROR) { 7568 if (chainCreated) { 7569 removeEffectChain_l(chain); 7570 } 7571 return status; 7572 } 7573 7574 effect->setDevice(mDevice); 7575 effect->setMode(mAudioFlinger->getMode()); 7576 return NO_ERROR; 7577} 7578 7579void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7580 7581 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7582 effect_descriptor_t desc = effect->desc(); 7583 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7584 detachAuxEffect_l(effect->id()); 7585 } 7586 7587 sp<EffectChain> chain = effect->chain().promote(); 7588 if (chain != 0) { 7589 // remove effect chain if removing last effect 7590 if (chain->removeEffect_l(effect) == 0) { 7591 removeEffectChain_l(chain); 7592 } 7593 } else { 7594 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7595 } 7596} 7597 7598void AudioFlinger::ThreadBase::lockEffectChains_l( 7599 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7600{ 7601 effectChains = mEffectChains; 7602 for (size_t i = 0; i < mEffectChains.size(); i++) { 7603 mEffectChains[i]->lock(); 7604 } 7605} 7606 7607void AudioFlinger::ThreadBase::unlockEffectChains( 7608 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7609{ 7610 for (size_t i = 0; i < effectChains.size(); i++) { 7611 effectChains[i]->unlock(); 7612 } 7613} 7614 7615sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7616{ 7617 Mutex::Autolock _l(mLock); 7618 return getEffectChain_l(sessionId); 7619} 7620 7621sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7622{ 7623 size_t size = mEffectChains.size(); 7624 for (size_t i = 0; i < size; i++) { 7625 if (mEffectChains[i]->sessionId() == sessionId) { 7626 return mEffectChains[i]; 7627 } 7628 } 7629 return 0; 7630} 7631 7632void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7633{ 7634 Mutex::Autolock _l(mLock); 7635 size_t size = mEffectChains.size(); 7636 for (size_t i = 0; i < size; i++) { 7637 mEffectChains[i]->setMode_l(mode); 7638 } 7639} 7640 7641void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7642 const wp<EffectHandle>& handle, 7643 bool unpinIfLast) { 7644 7645 Mutex::Autolock _l(mLock); 7646 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7647 // delete the effect module if removing last handle on it 7648 if (effect->removeHandle(handle) == 0) { 7649 if (!effect->isPinned() || unpinIfLast) { 7650 removeEffect_l(effect); 7651 AudioSystem::unregisterEffect(effect->id()); 7652 } 7653 } 7654} 7655 7656status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7657{ 7658 int session = chain->sessionId(); 7659 int16_t *buffer = mMixBuffer; 7660 bool ownsBuffer = false; 7661 7662 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7663 if (session > 0) { 7664 // Only one effect chain can be present in direct output thread and it uses 7665 // the mix buffer as input 7666 if (mType != DIRECT) { 7667 size_t numSamples = mNormalFrameCount * mChannelCount; 7668 buffer = new int16_t[numSamples]; 7669 memset(buffer, 0, numSamples * sizeof(int16_t)); 7670 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7671 ownsBuffer = true; 7672 } 7673 7674 // Attach all tracks with same session ID to this chain. 7675 for (size_t i = 0; i < mTracks.size(); ++i) { 7676 sp<Track> track = mTracks[i]; 7677 if (session == track->sessionId()) { 7678 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7679 track->setMainBuffer(buffer); 7680 chain->incTrackCnt(); 7681 } 7682 } 7683 7684 // indicate all active tracks in the chain 7685 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7686 sp<Track> track = mActiveTracks[i].promote(); 7687 if (track == 0) continue; 7688 if (session == track->sessionId()) { 7689 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7690 chain->incActiveTrackCnt(); 7691 } 7692 } 7693 } 7694 7695 chain->setInBuffer(buffer, ownsBuffer); 7696 chain->setOutBuffer(mMixBuffer); 7697 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7698 // chains list in order to be processed last as it contains output stage effects 7699 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7700 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7701 // after track specific effects and before output stage 7702 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7703 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7704 // Effect chain for other sessions are inserted at beginning of effect 7705 // chains list to be processed before output mix effects. Relative order between other 7706 // sessions is not important 7707 size_t size = mEffectChains.size(); 7708 size_t i = 0; 7709 for (i = 0; i < size; i++) { 7710 if (mEffectChains[i]->sessionId() < session) break; 7711 } 7712 mEffectChains.insertAt(chain, i); 7713 checkSuspendOnAddEffectChain_l(chain); 7714 7715 return NO_ERROR; 7716} 7717 7718size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7719{ 7720 int session = chain->sessionId(); 7721 7722 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7723 7724 for (size_t i = 0; i < mEffectChains.size(); i++) { 7725 if (chain == mEffectChains[i]) { 7726 mEffectChains.removeAt(i); 7727 // detach all active tracks from the chain 7728 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7729 sp<Track> track = mActiveTracks[i].promote(); 7730 if (track == 0) continue; 7731 if (session == track->sessionId()) { 7732 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7733 chain.get(), session); 7734 chain->decActiveTrackCnt(); 7735 } 7736 } 7737 7738 // detach all tracks with same session ID from this chain 7739 for (size_t i = 0; i < mTracks.size(); ++i) { 7740 sp<Track> track = mTracks[i]; 7741 if (session == track->sessionId()) { 7742 track->setMainBuffer(mMixBuffer); 7743 chain->decTrackCnt(); 7744 } 7745 } 7746 break; 7747 } 7748 } 7749 return mEffectChains.size(); 7750} 7751 7752status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7753 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7754{ 7755 Mutex::Autolock _l(mLock); 7756 return attachAuxEffect_l(track, EffectId); 7757} 7758 7759status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7760 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7761{ 7762 status_t status = NO_ERROR; 7763 7764 if (EffectId == 0) { 7765 track->setAuxBuffer(0, NULL); 7766 } else { 7767 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7768 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7769 if (effect != 0) { 7770 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7771 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7772 } else { 7773 status = INVALID_OPERATION; 7774 } 7775 } else { 7776 status = BAD_VALUE; 7777 } 7778 } 7779 return status; 7780} 7781 7782void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7783{ 7784 for (size_t i = 0; i < mTracks.size(); ++i) { 7785 sp<Track> track = mTracks[i]; 7786 if (track->auxEffectId() == effectId) { 7787 attachAuxEffect_l(track, 0); 7788 } 7789 } 7790} 7791 7792status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7793{ 7794 // only one chain per input thread 7795 if (mEffectChains.size() != 0) { 7796 return INVALID_OPERATION; 7797 } 7798 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7799 7800 chain->setInBuffer(NULL); 7801 chain->setOutBuffer(NULL); 7802 7803 checkSuspendOnAddEffectChain_l(chain); 7804 7805 mEffectChains.add(chain); 7806 7807 return NO_ERROR; 7808} 7809 7810size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7811{ 7812 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7813 ALOGW_IF(mEffectChains.size() != 1, 7814 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7815 chain.get(), mEffectChains.size(), this); 7816 if (mEffectChains.size() == 1) { 7817 mEffectChains.removeAt(0); 7818 } 7819 return 0; 7820} 7821 7822// ---------------------------------------------------------------------------- 7823// EffectModule implementation 7824// ---------------------------------------------------------------------------- 7825 7826#undef LOG_TAG 7827#define LOG_TAG "AudioFlinger::EffectModule" 7828 7829AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7830 const wp<AudioFlinger::EffectChain>& chain, 7831 effect_descriptor_t *desc, 7832 int id, 7833 int sessionId) 7834 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7835 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7836{ 7837 ALOGV("Constructor %p", this); 7838 int lStatus; 7839 if (thread == NULL) { 7840 return; 7841 } 7842 7843 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7844 7845 // create effect engine from effect factory 7846 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7847 7848 if (mStatus != NO_ERROR) { 7849 return; 7850 } 7851 lStatus = init(); 7852 if (lStatus < 0) { 7853 mStatus = lStatus; 7854 goto Error; 7855 } 7856 7857 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7858 mPinned = true; 7859 } 7860 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7861 return; 7862Error: 7863 EffectRelease(mEffectInterface); 7864 mEffectInterface = NULL; 7865 ALOGV("Constructor Error %d", mStatus); 7866} 7867 7868AudioFlinger::EffectModule::~EffectModule() 7869{ 7870 ALOGV("Destructor %p", this); 7871 if (mEffectInterface != NULL) { 7872 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7873 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7874 sp<ThreadBase> thread = mThread.promote(); 7875 if (thread != 0) { 7876 audio_stream_t *stream = thread->stream(); 7877 if (stream != NULL) { 7878 stream->remove_audio_effect(stream, mEffectInterface); 7879 } 7880 } 7881 } 7882 // release effect engine 7883 EffectRelease(mEffectInterface); 7884 } 7885} 7886 7887status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7888{ 7889 status_t status; 7890 7891 Mutex::Autolock _l(mLock); 7892 int priority = handle->priority(); 7893 size_t size = mHandles.size(); 7894 sp<EffectHandle> h; 7895 size_t i; 7896 for (i = 0; i < size; i++) { 7897 h = mHandles[i].promote(); 7898 if (h == 0) continue; 7899 if (h->priority() <= priority) break; 7900 } 7901 // if inserted in first place, move effect control from previous owner to this handle 7902 if (i == 0) { 7903 bool enabled = false; 7904 if (h != 0) { 7905 enabled = h->enabled(); 7906 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7907 } 7908 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7909 status = NO_ERROR; 7910 } else { 7911 status = ALREADY_EXISTS; 7912 } 7913 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7914 mHandles.insertAt(handle, i); 7915 return status; 7916} 7917 7918size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7919{ 7920 Mutex::Autolock _l(mLock); 7921 size_t size = mHandles.size(); 7922 size_t i; 7923 for (i = 0; i < size; i++) { 7924 if (mHandles[i] == handle) break; 7925 } 7926 if (i == size) { 7927 return size; 7928 } 7929 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7930 7931 bool enabled = false; 7932 EffectHandle *hdl = handle.unsafe_get(); 7933 if (hdl != NULL) { 7934 ALOGV("removeHandle() unsafe_get OK"); 7935 enabled = hdl->enabled(); 7936 } 7937 mHandles.removeAt(i); 7938 size = mHandles.size(); 7939 // if removed from first place, move effect control from this handle to next in line 7940 if (i == 0 && size != 0) { 7941 sp<EffectHandle> h = mHandles[0].promote(); 7942 if (h != 0) { 7943 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7944 } 7945 } 7946 7947 // Prevent calls to process() and other functions on effect interface from now on. 7948 // The effect engine will be released by the destructor when the last strong reference on 7949 // this object is released which can happen after next process is called. 7950 if (size == 0 && !mPinned) { 7951 mState = DESTROYED; 7952 } 7953 7954 return size; 7955} 7956 7957sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7958{ 7959 Mutex::Autolock _l(mLock); 7960 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7961} 7962 7963void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7964{ 7965 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7966 // keep a strong reference on this EffectModule to avoid calling the 7967 // destructor before we exit 7968 sp<EffectModule> keep(this); 7969 { 7970 sp<ThreadBase> thread = mThread.promote(); 7971 if (thread != 0) { 7972 thread->disconnectEffect(keep, handle, unpinIfLast); 7973 } 7974 } 7975} 7976 7977void AudioFlinger::EffectModule::updateState() { 7978 Mutex::Autolock _l(mLock); 7979 7980 switch (mState) { 7981 case RESTART: 7982 reset_l(); 7983 // FALL THROUGH 7984 7985 case STARTING: 7986 // clear auxiliary effect input buffer for next accumulation 7987 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7988 memset(mConfig.inputCfg.buffer.raw, 7989 0, 7990 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7991 } 7992 start_l(); 7993 mState = ACTIVE; 7994 break; 7995 case STOPPING: 7996 stop_l(); 7997 mDisableWaitCnt = mMaxDisableWaitCnt; 7998 mState = STOPPED; 7999 break; 8000 case STOPPED: 8001 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8002 // turn off sequence. 8003 if (--mDisableWaitCnt == 0) { 8004 reset_l(); 8005 mState = IDLE; 8006 } 8007 break; 8008 default: //IDLE , ACTIVE, DESTROYED 8009 break; 8010 } 8011} 8012 8013void AudioFlinger::EffectModule::process() 8014{ 8015 Mutex::Autolock _l(mLock); 8016 8017 if (mState == DESTROYED || mEffectInterface == NULL || 8018 mConfig.inputCfg.buffer.raw == NULL || 8019 mConfig.outputCfg.buffer.raw == NULL) { 8020 return; 8021 } 8022 8023 if (isProcessEnabled()) { 8024 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8025 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8026 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8027 mConfig.inputCfg.buffer.s32, 8028 mConfig.inputCfg.buffer.frameCount/2); 8029 } 8030 8031 // do the actual processing in the effect engine 8032 int ret = (*mEffectInterface)->process(mEffectInterface, 8033 &mConfig.inputCfg.buffer, 8034 &mConfig.outputCfg.buffer); 8035 8036 // force transition to IDLE state when engine is ready 8037 if (mState == STOPPED && ret == -ENODATA) { 8038 mDisableWaitCnt = 1; 8039 } 8040 8041 // clear auxiliary effect input buffer for next accumulation 8042 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8043 memset(mConfig.inputCfg.buffer.raw, 0, 8044 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8045 } 8046 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8047 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8048 // If an insert effect is idle and input buffer is different from output buffer, 8049 // accumulate input onto output 8050 sp<EffectChain> chain = mChain.promote(); 8051 if (chain != 0 && chain->activeTrackCnt() != 0) { 8052 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8053 int16_t *in = mConfig.inputCfg.buffer.s16; 8054 int16_t *out = mConfig.outputCfg.buffer.s16; 8055 for (size_t i = 0; i < frameCnt; i++) { 8056 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8057 } 8058 } 8059 } 8060} 8061 8062void AudioFlinger::EffectModule::reset_l() 8063{ 8064 if (mEffectInterface == NULL) { 8065 return; 8066 } 8067 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8068} 8069 8070status_t AudioFlinger::EffectModule::configure() 8071{ 8072 uint32_t channels; 8073 if (mEffectInterface == NULL) { 8074 return NO_INIT; 8075 } 8076 8077 sp<ThreadBase> thread = mThread.promote(); 8078 if (thread == 0) { 8079 return DEAD_OBJECT; 8080 } 8081 8082 // TODO: handle configuration of effects replacing track process 8083 if (thread->channelCount() == 1) { 8084 channels = AUDIO_CHANNEL_OUT_MONO; 8085 } else { 8086 channels = AUDIO_CHANNEL_OUT_STEREO; 8087 } 8088 8089 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8090 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8091 } else { 8092 mConfig.inputCfg.channels = channels; 8093 } 8094 mConfig.outputCfg.channels = channels; 8095 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8096 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8097 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8098 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8099 mConfig.inputCfg.bufferProvider.cookie = NULL; 8100 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8101 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8102 mConfig.outputCfg.bufferProvider.cookie = NULL; 8103 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8104 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8105 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8106 // Insert effect: 8107 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8108 // always overwrites output buffer: input buffer == output buffer 8109 // - in other sessions: 8110 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8111 // other effect: overwrites output buffer: input buffer == output buffer 8112 // Auxiliary effect: 8113 // accumulates in output buffer: input buffer != output buffer 8114 // Therefore: accumulate <=> input buffer != output buffer 8115 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8116 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8117 } else { 8118 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8119 } 8120 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8121 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8122 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8123 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8124 8125 ALOGV("configure() %p thread %p buffer %p framecount %d", 8126 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8127 8128 status_t cmdStatus; 8129 uint32_t size = sizeof(int); 8130 status_t status = (*mEffectInterface)->command(mEffectInterface, 8131 EFFECT_CMD_SET_CONFIG, 8132 sizeof(effect_config_t), 8133 &mConfig, 8134 &size, 8135 &cmdStatus); 8136 if (status == 0) { 8137 status = cmdStatus; 8138 } 8139 8140 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8141 (1000 * mConfig.outputCfg.buffer.frameCount); 8142 8143 return status; 8144} 8145 8146status_t AudioFlinger::EffectModule::init() 8147{ 8148 Mutex::Autolock _l(mLock); 8149 if (mEffectInterface == NULL) { 8150 return NO_INIT; 8151 } 8152 status_t cmdStatus; 8153 uint32_t size = sizeof(status_t); 8154 status_t status = (*mEffectInterface)->command(mEffectInterface, 8155 EFFECT_CMD_INIT, 8156 0, 8157 NULL, 8158 &size, 8159 &cmdStatus); 8160 if (status == 0) { 8161 status = cmdStatus; 8162 } 8163 return status; 8164} 8165 8166status_t AudioFlinger::EffectModule::start() 8167{ 8168 Mutex::Autolock _l(mLock); 8169 return start_l(); 8170} 8171 8172status_t AudioFlinger::EffectModule::start_l() 8173{ 8174 if (mEffectInterface == NULL) { 8175 return NO_INIT; 8176 } 8177 status_t cmdStatus; 8178 uint32_t size = sizeof(status_t); 8179 status_t status = (*mEffectInterface)->command(mEffectInterface, 8180 EFFECT_CMD_ENABLE, 8181 0, 8182 NULL, 8183 &size, 8184 &cmdStatus); 8185 if (status == 0) { 8186 status = cmdStatus; 8187 } 8188 if (status == 0 && 8189 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8190 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8191 sp<ThreadBase> thread = mThread.promote(); 8192 if (thread != 0) { 8193 audio_stream_t *stream = thread->stream(); 8194 if (stream != NULL) { 8195 stream->add_audio_effect(stream, mEffectInterface); 8196 } 8197 } 8198 } 8199 return status; 8200} 8201 8202status_t AudioFlinger::EffectModule::stop() 8203{ 8204 Mutex::Autolock _l(mLock); 8205 return stop_l(); 8206} 8207 8208status_t AudioFlinger::EffectModule::stop_l() 8209{ 8210 if (mEffectInterface == NULL) { 8211 return NO_INIT; 8212 } 8213 status_t cmdStatus; 8214 uint32_t size = sizeof(status_t); 8215 status_t status = (*mEffectInterface)->command(mEffectInterface, 8216 EFFECT_CMD_DISABLE, 8217 0, 8218 NULL, 8219 &size, 8220 &cmdStatus); 8221 if (status == 0) { 8222 status = cmdStatus; 8223 } 8224 if (status == 0 && 8225 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8226 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8227 sp<ThreadBase> thread = mThread.promote(); 8228 if (thread != 0) { 8229 audio_stream_t *stream = thread->stream(); 8230 if (stream != NULL) { 8231 stream->remove_audio_effect(stream, mEffectInterface); 8232 } 8233 } 8234 } 8235 return status; 8236} 8237 8238status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8239 uint32_t cmdSize, 8240 void *pCmdData, 8241 uint32_t *replySize, 8242 void *pReplyData) 8243{ 8244 Mutex::Autolock _l(mLock); 8245// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8246 8247 if (mState == DESTROYED || mEffectInterface == NULL) { 8248 return NO_INIT; 8249 } 8250 status_t status = (*mEffectInterface)->command(mEffectInterface, 8251 cmdCode, 8252 cmdSize, 8253 pCmdData, 8254 replySize, 8255 pReplyData); 8256 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8257 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8258 for (size_t i = 1; i < mHandles.size(); i++) { 8259 sp<EffectHandle> h = mHandles[i].promote(); 8260 if (h != 0) { 8261 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8262 } 8263 } 8264 } 8265 return status; 8266} 8267 8268status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8269{ 8270 8271 Mutex::Autolock _l(mLock); 8272 ALOGV("setEnabled %p enabled %d", this, enabled); 8273 8274 if (enabled != isEnabled()) { 8275 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8276 if (enabled && status != NO_ERROR) { 8277 return status; 8278 } 8279 8280 switch (mState) { 8281 // going from disabled to enabled 8282 case IDLE: 8283 mState = STARTING; 8284 break; 8285 case STOPPED: 8286 mState = RESTART; 8287 break; 8288 case STOPPING: 8289 mState = ACTIVE; 8290 break; 8291 8292 // going from enabled to disabled 8293 case RESTART: 8294 mState = STOPPED; 8295 break; 8296 case STARTING: 8297 mState = IDLE; 8298 break; 8299 case ACTIVE: 8300 mState = STOPPING; 8301 break; 8302 case DESTROYED: 8303 return NO_ERROR; // simply ignore as we are being destroyed 8304 } 8305 for (size_t i = 1; i < mHandles.size(); i++) { 8306 sp<EffectHandle> h = mHandles[i].promote(); 8307 if (h != 0) { 8308 h->setEnabled(enabled); 8309 } 8310 } 8311 } 8312 return NO_ERROR; 8313} 8314 8315bool AudioFlinger::EffectModule::isEnabled() const 8316{ 8317 switch (mState) { 8318 case RESTART: 8319 case STARTING: 8320 case ACTIVE: 8321 return true; 8322 case IDLE: 8323 case STOPPING: 8324 case STOPPED: 8325 case DESTROYED: 8326 default: 8327 return false; 8328 } 8329} 8330 8331bool AudioFlinger::EffectModule::isProcessEnabled() const 8332{ 8333 switch (mState) { 8334 case RESTART: 8335 case ACTIVE: 8336 case STOPPING: 8337 case STOPPED: 8338 return true; 8339 case IDLE: 8340 case STARTING: 8341 case DESTROYED: 8342 default: 8343 return false; 8344 } 8345} 8346 8347status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8348{ 8349 Mutex::Autolock _l(mLock); 8350 status_t status = NO_ERROR; 8351 8352 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8353 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8354 if (isProcessEnabled() && 8355 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8356 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8357 status_t cmdStatus; 8358 uint32_t volume[2]; 8359 uint32_t *pVolume = NULL; 8360 uint32_t size = sizeof(volume); 8361 volume[0] = *left; 8362 volume[1] = *right; 8363 if (controller) { 8364 pVolume = volume; 8365 } 8366 status = (*mEffectInterface)->command(mEffectInterface, 8367 EFFECT_CMD_SET_VOLUME, 8368 size, 8369 volume, 8370 &size, 8371 pVolume); 8372 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8373 *left = volume[0]; 8374 *right = volume[1]; 8375 } 8376 } 8377 return status; 8378} 8379 8380status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8381{ 8382 Mutex::Autolock _l(mLock); 8383 status_t status = NO_ERROR; 8384 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8385 // audio pre processing modules on RecordThread can receive both output and 8386 // input device indication in the same call 8387 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8388 if (dev) { 8389 status_t cmdStatus; 8390 uint32_t size = sizeof(status_t); 8391 8392 status = (*mEffectInterface)->command(mEffectInterface, 8393 EFFECT_CMD_SET_DEVICE, 8394 sizeof(uint32_t), 8395 &dev, 8396 &size, 8397 &cmdStatus); 8398 if (status == NO_ERROR) { 8399 status = cmdStatus; 8400 } 8401 } 8402 dev = device & AUDIO_DEVICE_IN_ALL; 8403 if (dev) { 8404 status_t cmdStatus; 8405 uint32_t size = sizeof(status_t); 8406 8407 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8408 EFFECT_CMD_SET_INPUT_DEVICE, 8409 sizeof(uint32_t), 8410 &dev, 8411 &size, 8412 &cmdStatus); 8413 if (status2 == NO_ERROR) { 8414 status2 = cmdStatus; 8415 } 8416 if (status == NO_ERROR) { 8417 status = status2; 8418 } 8419 } 8420 } 8421 return status; 8422} 8423 8424status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8425{ 8426 Mutex::Autolock _l(mLock); 8427 status_t status = NO_ERROR; 8428 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8429 status_t cmdStatus; 8430 uint32_t size = sizeof(status_t); 8431 status = (*mEffectInterface)->command(mEffectInterface, 8432 EFFECT_CMD_SET_AUDIO_MODE, 8433 sizeof(audio_mode_t), 8434 &mode, 8435 &size, 8436 &cmdStatus); 8437 if (status == NO_ERROR) { 8438 status = cmdStatus; 8439 } 8440 } 8441 return status; 8442} 8443 8444void AudioFlinger::EffectModule::setSuspended(bool suspended) 8445{ 8446 Mutex::Autolock _l(mLock); 8447 mSuspended = suspended; 8448} 8449 8450bool AudioFlinger::EffectModule::suspended() const 8451{ 8452 Mutex::Autolock _l(mLock); 8453 return mSuspended; 8454} 8455 8456status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8457{ 8458 const size_t SIZE = 256; 8459 char buffer[SIZE]; 8460 String8 result; 8461 8462 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8463 result.append(buffer); 8464 8465 bool locked = tryLock(mLock); 8466 // failed to lock - AudioFlinger is probably deadlocked 8467 if (!locked) { 8468 result.append("\t\tCould not lock Fx mutex:\n"); 8469 } 8470 8471 result.append("\t\tSession Status State Engine:\n"); 8472 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8473 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8474 result.append(buffer); 8475 8476 result.append("\t\tDescriptor:\n"); 8477 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8478 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8479 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8480 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8481 result.append(buffer); 8482 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8483 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8484 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8485 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8486 result.append(buffer); 8487 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8488 mDescriptor.apiVersion, 8489 mDescriptor.flags); 8490 result.append(buffer); 8491 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8492 mDescriptor.name); 8493 result.append(buffer); 8494 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8495 mDescriptor.implementor); 8496 result.append(buffer); 8497 8498 result.append("\t\t- Input configuration:\n"); 8499 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8500 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8501 (uint32_t)mConfig.inputCfg.buffer.raw, 8502 mConfig.inputCfg.buffer.frameCount, 8503 mConfig.inputCfg.samplingRate, 8504 mConfig.inputCfg.channels, 8505 mConfig.inputCfg.format); 8506 result.append(buffer); 8507 8508 result.append("\t\t- Output configuration:\n"); 8509 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8510 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8511 (uint32_t)mConfig.outputCfg.buffer.raw, 8512 mConfig.outputCfg.buffer.frameCount, 8513 mConfig.outputCfg.samplingRate, 8514 mConfig.outputCfg.channels, 8515 mConfig.outputCfg.format); 8516 result.append(buffer); 8517 8518 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8519 result.append(buffer); 8520 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8521 for (size_t i = 0; i < mHandles.size(); ++i) { 8522 sp<EffectHandle> handle = mHandles[i].promote(); 8523 if (handle != 0) { 8524 handle->dump(buffer, SIZE); 8525 result.append(buffer); 8526 } 8527 } 8528 8529 result.append("\n"); 8530 8531 write(fd, result.string(), result.length()); 8532 8533 if (locked) { 8534 mLock.unlock(); 8535 } 8536 8537 return NO_ERROR; 8538} 8539 8540// ---------------------------------------------------------------------------- 8541// EffectHandle implementation 8542// ---------------------------------------------------------------------------- 8543 8544#undef LOG_TAG 8545#define LOG_TAG "AudioFlinger::EffectHandle" 8546 8547AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8548 const sp<AudioFlinger::Client>& client, 8549 const sp<IEffectClient>& effectClient, 8550 int32_t priority) 8551 : BnEffect(), 8552 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8553 mPriority(priority), mHasControl(false), mEnabled(false) 8554{ 8555 ALOGV("constructor %p", this); 8556 8557 if (client == 0) { 8558 return; 8559 } 8560 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8561 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8562 if (mCblkMemory != 0) { 8563 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8564 8565 if (mCblk != NULL) { 8566 new(mCblk) effect_param_cblk_t(); 8567 mBuffer = (uint8_t *)mCblk + bufOffset; 8568 } 8569 } else { 8570 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8571 return; 8572 } 8573} 8574 8575AudioFlinger::EffectHandle::~EffectHandle() 8576{ 8577 ALOGV("Destructor %p", this); 8578 disconnect(false); 8579 ALOGV("Destructor DONE %p", this); 8580} 8581 8582status_t AudioFlinger::EffectHandle::enable() 8583{ 8584 ALOGV("enable %p", this); 8585 if (!mHasControl) return INVALID_OPERATION; 8586 if (mEffect == 0) return DEAD_OBJECT; 8587 8588 if (mEnabled) { 8589 return NO_ERROR; 8590 } 8591 8592 mEnabled = true; 8593 8594 sp<ThreadBase> thread = mEffect->thread().promote(); 8595 if (thread != 0) { 8596 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8597 } 8598 8599 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8600 if (mEffect->suspended()) { 8601 return NO_ERROR; 8602 } 8603 8604 status_t status = mEffect->setEnabled(true); 8605 if (status != NO_ERROR) { 8606 if (thread != 0) { 8607 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8608 } 8609 mEnabled = false; 8610 } 8611 return status; 8612} 8613 8614status_t AudioFlinger::EffectHandle::disable() 8615{ 8616 ALOGV("disable %p", this); 8617 if (!mHasControl) return INVALID_OPERATION; 8618 if (mEffect == 0) return DEAD_OBJECT; 8619 8620 if (!mEnabled) { 8621 return NO_ERROR; 8622 } 8623 mEnabled = false; 8624 8625 if (mEffect->suspended()) { 8626 return NO_ERROR; 8627 } 8628 8629 status_t status = mEffect->setEnabled(false); 8630 8631 sp<ThreadBase> thread = mEffect->thread().promote(); 8632 if (thread != 0) { 8633 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8634 } 8635 8636 return status; 8637} 8638 8639void AudioFlinger::EffectHandle::disconnect() 8640{ 8641 disconnect(true); 8642} 8643 8644void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8645{ 8646 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8647 if (mEffect == 0) { 8648 return; 8649 } 8650 mEffect->disconnect(this, unpinIfLast); 8651 8652 if (mHasControl && mEnabled) { 8653 sp<ThreadBase> thread = mEffect->thread().promote(); 8654 if (thread != 0) { 8655 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8656 } 8657 } 8658 8659 // release sp on module => module destructor can be called now 8660 mEffect.clear(); 8661 if (mClient != 0) { 8662 if (mCblk != NULL) { 8663 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8664 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8665 } 8666 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8667 // Client destructor must run with AudioFlinger mutex locked 8668 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8669 mClient.clear(); 8670 } 8671} 8672 8673status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8674 uint32_t cmdSize, 8675 void *pCmdData, 8676 uint32_t *replySize, 8677 void *pReplyData) 8678{ 8679// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8680// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8681 8682 // only get parameter command is permitted for applications not controlling the effect 8683 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8684 return INVALID_OPERATION; 8685 } 8686 if (mEffect == 0) return DEAD_OBJECT; 8687 if (mClient == 0) return INVALID_OPERATION; 8688 8689 // handle commands that are not forwarded transparently to effect engine 8690 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8691 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8692 // no risk to block the whole media server process or mixer threads is we are stuck here 8693 Mutex::Autolock _l(mCblk->lock); 8694 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8695 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8696 mCblk->serverIndex = 0; 8697 mCblk->clientIndex = 0; 8698 return BAD_VALUE; 8699 } 8700 status_t status = NO_ERROR; 8701 while (mCblk->serverIndex < mCblk->clientIndex) { 8702 int reply; 8703 uint32_t rsize = sizeof(int); 8704 int *p = (int *)(mBuffer + mCblk->serverIndex); 8705 int size = *p++; 8706 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8707 ALOGW("command(): invalid parameter block size"); 8708 break; 8709 } 8710 effect_param_t *param = (effect_param_t *)p; 8711 if (param->psize == 0 || param->vsize == 0) { 8712 ALOGW("command(): null parameter or value size"); 8713 mCblk->serverIndex += size; 8714 continue; 8715 } 8716 uint32_t psize = sizeof(effect_param_t) + 8717 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8718 param->vsize; 8719 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8720 psize, 8721 p, 8722 &rsize, 8723 &reply); 8724 // stop at first error encountered 8725 if (ret != NO_ERROR) { 8726 status = ret; 8727 *(int *)pReplyData = reply; 8728 break; 8729 } else if (reply != NO_ERROR) { 8730 *(int *)pReplyData = reply; 8731 break; 8732 } 8733 mCblk->serverIndex += size; 8734 } 8735 mCblk->serverIndex = 0; 8736 mCblk->clientIndex = 0; 8737 return status; 8738 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8739 *(int *)pReplyData = NO_ERROR; 8740 return enable(); 8741 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8742 *(int *)pReplyData = NO_ERROR; 8743 return disable(); 8744 } 8745 8746 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8747} 8748 8749void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8750{ 8751 ALOGV("setControl %p control %d", this, hasControl); 8752 8753 mHasControl = hasControl; 8754 mEnabled = enabled; 8755 8756 if (signal && mEffectClient != 0) { 8757 mEffectClient->controlStatusChanged(hasControl); 8758 } 8759} 8760 8761void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8762 uint32_t cmdSize, 8763 void *pCmdData, 8764 uint32_t replySize, 8765 void *pReplyData) 8766{ 8767 if (mEffectClient != 0) { 8768 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8769 } 8770} 8771 8772 8773 8774void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8775{ 8776 if (mEffectClient != 0) { 8777 mEffectClient->enableStatusChanged(enabled); 8778 } 8779} 8780 8781status_t AudioFlinger::EffectHandle::onTransact( 8782 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8783{ 8784 return BnEffect::onTransact(code, data, reply, flags); 8785} 8786 8787 8788void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8789{ 8790 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8791 8792 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8793 (mClient == 0) ? getpid_cached : mClient->pid(), 8794 mPriority, 8795 mHasControl, 8796 !locked, 8797 mCblk ? mCblk->clientIndex : 0, 8798 mCblk ? mCblk->serverIndex : 0 8799 ); 8800 8801 if (locked) { 8802 mCblk->lock.unlock(); 8803 } 8804} 8805 8806#undef LOG_TAG 8807#define LOG_TAG "AudioFlinger::EffectChain" 8808 8809AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8810 int sessionId) 8811 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8812 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8813 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8814{ 8815 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8816 if (thread == NULL) { 8817 return; 8818 } 8819 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8820 thread->frameCount(); 8821} 8822 8823AudioFlinger::EffectChain::~EffectChain() 8824{ 8825 if (mOwnInBuffer) { 8826 delete mInBuffer; 8827 } 8828 8829} 8830 8831// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8832sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8833{ 8834 size_t size = mEffects.size(); 8835 8836 for (size_t i = 0; i < size; i++) { 8837 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8838 return mEffects[i]; 8839 } 8840 } 8841 return 0; 8842} 8843 8844// getEffectFromId_l() must be called with ThreadBase::mLock held 8845sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8846{ 8847 size_t size = mEffects.size(); 8848 8849 for (size_t i = 0; i < size; i++) { 8850 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8851 if (id == 0 || mEffects[i]->id() == id) { 8852 return mEffects[i]; 8853 } 8854 } 8855 return 0; 8856} 8857 8858// getEffectFromType_l() must be called with ThreadBase::mLock held 8859sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8860 const effect_uuid_t *type) 8861{ 8862 size_t size = mEffects.size(); 8863 8864 for (size_t i = 0; i < size; i++) { 8865 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8866 return mEffects[i]; 8867 } 8868 } 8869 return 0; 8870} 8871 8872// Must be called with EffectChain::mLock locked 8873void AudioFlinger::EffectChain::process_l() 8874{ 8875 sp<ThreadBase> thread = mThread.promote(); 8876 if (thread == 0) { 8877 ALOGW("process_l(): cannot promote mixer thread"); 8878 return; 8879 } 8880 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8881 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8882 // always process effects unless no more tracks are on the session and the effect tail 8883 // has been rendered 8884 bool doProcess = true; 8885 if (!isGlobalSession) { 8886 bool tracksOnSession = (trackCnt() != 0); 8887 8888 if (!tracksOnSession && mTailBufferCount == 0) { 8889 doProcess = false; 8890 } 8891 8892 if (activeTrackCnt() == 0) { 8893 // if no track is active and the effect tail has not been rendered, 8894 // the input buffer must be cleared here as the mixer process will not do it 8895 if (tracksOnSession || mTailBufferCount > 0) { 8896 size_t numSamples = thread->frameCount() * thread->channelCount(); 8897 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8898 if (mTailBufferCount > 0) { 8899 mTailBufferCount--; 8900 } 8901 } 8902 } 8903 } 8904 8905 size_t size = mEffects.size(); 8906 if (doProcess) { 8907 for (size_t i = 0; i < size; i++) { 8908 mEffects[i]->process(); 8909 } 8910 } 8911 for (size_t i = 0; i < size; i++) { 8912 mEffects[i]->updateState(); 8913 } 8914} 8915 8916// addEffect_l() must be called with PlaybackThread::mLock held 8917status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8918{ 8919 effect_descriptor_t desc = effect->desc(); 8920 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8921 8922 Mutex::Autolock _l(mLock); 8923 effect->setChain(this); 8924 sp<ThreadBase> thread = mThread.promote(); 8925 if (thread == 0) { 8926 return NO_INIT; 8927 } 8928 effect->setThread(thread); 8929 8930 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8931 // Auxiliary effects are inserted at the beginning of mEffects vector as 8932 // they are processed first and accumulated in chain input buffer 8933 mEffects.insertAt(effect, 0); 8934 8935 // the input buffer for auxiliary effect contains mono samples in 8936 // 32 bit format. This is to avoid saturation in AudoMixer 8937 // accumulation stage. Saturation is done in EffectModule::process() before 8938 // calling the process in effect engine 8939 size_t numSamples = thread->frameCount(); 8940 int32_t *buffer = new int32_t[numSamples]; 8941 memset(buffer, 0, numSamples * sizeof(int32_t)); 8942 effect->setInBuffer((int16_t *)buffer); 8943 // auxiliary effects output samples to chain input buffer for further processing 8944 // by insert effects 8945 effect->setOutBuffer(mInBuffer); 8946 } else { 8947 // Insert effects are inserted at the end of mEffects vector as they are processed 8948 // after track and auxiliary effects. 8949 // Insert effect order as a function of indicated preference: 8950 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8951 // another effect is present 8952 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8953 // last effect claiming first position 8954 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8955 // first effect claiming last position 8956 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8957 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8958 // already present 8959 8960 size_t size = mEffects.size(); 8961 size_t idx_insert = size; 8962 ssize_t idx_insert_first = -1; 8963 ssize_t idx_insert_last = -1; 8964 8965 for (size_t i = 0; i < size; i++) { 8966 effect_descriptor_t d = mEffects[i]->desc(); 8967 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8968 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8969 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8970 // check invalid effect chaining combinations 8971 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8972 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8973 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8974 return INVALID_OPERATION; 8975 } 8976 // remember position of first insert effect and by default 8977 // select this as insert position for new effect 8978 if (idx_insert == size) { 8979 idx_insert = i; 8980 } 8981 // remember position of last insert effect claiming 8982 // first position 8983 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8984 idx_insert_first = i; 8985 } 8986 // remember position of first insert effect claiming 8987 // last position 8988 if (iPref == EFFECT_FLAG_INSERT_LAST && 8989 idx_insert_last == -1) { 8990 idx_insert_last = i; 8991 } 8992 } 8993 } 8994 8995 // modify idx_insert from first position if needed 8996 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8997 if (idx_insert_last != -1) { 8998 idx_insert = idx_insert_last; 8999 } else { 9000 idx_insert = size; 9001 } 9002 } else { 9003 if (idx_insert_first != -1) { 9004 idx_insert = idx_insert_first + 1; 9005 } 9006 } 9007 9008 // always read samples from chain input buffer 9009 effect->setInBuffer(mInBuffer); 9010 9011 // if last effect in the chain, output samples to chain 9012 // output buffer, otherwise to chain input buffer 9013 if (idx_insert == size) { 9014 if (idx_insert != 0) { 9015 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9016 mEffects[idx_insert-1]->configure(); 9017 } 9018 effect->setOutBuffer(mOutBuffer); 9019 } else { 9020 effect->setOutBuffer(mInBuffer); 9021 } 9022 mEffects.insertAt(effect, idx_insert); 9023 9024 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9025 } 9026 effect->configure(); 9027 return NO_ERROR; 9028} 9029 9030// removeEffect_l() must be called with PlaybackThread::mLock held 9031size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9032{ 9033 Mutex::Autolock _l(mLock); 9034 size_t size = mEffects.size(); 9035 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9036 9037 for (size_t i = 0; i < size; i++) { 9038 if (effect == mEffects[i]) { 9039 // calling stop here will remove pre-processing effect from the audio HAL. 9040 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9041 // the middle of a read from audio HAL 9042 if (mEffects[i]->state() == EffectModule::ACTIVE || 9043 mEffects[i]->state() == EffectModule::STOPPING) { 9044 mEffects[i]->stop(); 9045 } 9046 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9047 delete[] effect->inBuffer(); 9048 } else { 9049 if (i == size - 1 && i != 0) { 9050 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9051 mEffects[i - 1]->configure(); 9052 } 9053 } 9054 mEffects.removeAt(i); 9055 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9056 break; 9057 } 9058 } 9059 9060 return mEffects.size(); 9061} 9062 9063// setDevice_l() must be called with PlaybackThread::mLock held 9064void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9065{ 9066 size_t size = mEffects.size(); 9067 for (size_t i = 0; i < size; i++) { 9068 mEffects[i]->setDevice(device); 9069 } 9070} 9071 9072// setMode_l() must be called with PlaybackThread::mLock held 9073void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9074{ 9075 size_t size = mEffects.size(); 9076 for (size_t i = 0; i < size; i++) { 9077 mEffects[i]->setMode(mode); 9078 } 9079} 9080 9081// setVolume_l() must be called with PlaybackThread::mLock held 9082bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9083{ 9084 uint32_t newLeft = *left; 9085 uint32_t newRight = *right; 9086 bool hasControl = false; 9087 int ctrlIdx = -1; 9088 size_t size = mEffects.size(); 9089 9090 // first update volume controller 9091 for (size_t i = size; i > 0; i--) { 9092 if (mEffects[i - 1]->isProcessEnabled() && 9093 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9094 ctrlIdx = i - 1; 9095 hasControl = true; 9096 break; 9097 } 9098 } 9099 9100 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9101 if (hasControl) { 9102 *left = mNewLeftVolume; 9103 *right = mNewRightVolume; 9104 } 9105 return hasControl; 9106 } 9107 9108 mVolumeCtrlIdx = ctrlIdx; 9109 mLeftVolume = newLeft; 9110 mRightVolume = newRight; 9111 9112 // second get volume update from volume controller 9113 if (ctrlIdx >= 0) { 9114 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9115 mNewLeftVolume = newLeft; 9116 mNewRightVolume = newRight; 9117 } 9118 // then indicate volume to all other effects in chain. 9119 // Pass altered volume to effects before volume controller 9120 // and requested volume to effects after controller 9121 uint32_t lVol = newLeft; 9122 uint32_t rVol = newRight; 9123 9124 for (size_t i = 0; i < size; i++) { 9125 if ((int)i == ctrlIdx) continue; 9126 // this also works for ctrlIdx == -1 when there is no volume controller 9127 if ((int)i > ctrlIdx) { 9128 lVol = *left; 9129 rVol = *right; 9130 } 9131 mEffects[i]->setVolume(&lVol, &rVol, false); 9132 } 9133 *left = newLeft; 9134 *right = newRight; 9135 9136 return hasControl; 9137} 9138 9139status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9140{ 9141 const size_t SIZE = 256; 9142 char buffer[SIZE]; 9143 String8 result; 9144 9145 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9146 result.append(buffer); 9147 9148 bool locked = tryLock(mLock); 9149 // failed to lock - AudioFlinger is probably deadlocked 9150 if (!locked) { 9151 result.append("\tCould not lock mutex:\n"); 9152 } 9153 9154 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9155 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9156 mEffects.size(), 9157 (uint32_t)mInBuffer, 9158 (uint32_t)mOutBuffer, 9159 mActiveTrackCnt); 9160 result.append(buffer); 9161 write(fd, result.string(), result.size()); 9162 9163 for (size_t i = 0; i < mEffects.size(); ++i) { 9164 sp<EffectModule> effect = mEffects[i]; 9165 if (effect != 0) { 9166 effect->dump(fd, args); 9167 } 9168 } 9169 9170 if (locked) { 9171 mLock.unlock(); 9172 } 9173 9174 return NO_ERROR; 9175} 9176 9177// must be called with ThreadBase::mLock held 9178void AudioFlinger::EffectChain::setEffectSuspended_l( 9179 const effect_uuid_t *type, bool suspend) 9180{ 9181 sp<SuspendedEffectDesc> desc; 9182 // use effect type UUID timelow as key as there is no real risk of identical 9183 // timeLow fields among effect type UUIDs. 9184 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9185 if (suspend) { 9186 if (index >= 0) { 9187 desc = mSuspendedEffects.valueAt(index); 9188 } else { 9189 desc = new SuspendedEffectDesc(); 9190 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9191 mSuspendedEffects.add(type->timeLow, desc); 9192 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9193 } 9194 if (desc->mRefCount++ == 0) { 9195 sp<EffectModule> effect = getEffectIfEnabled(type); 9196 if (effect != 0) { 9197 desc->mEffect = effect; 9198 effect->setSuspended(true); 9199 effect->setEnabled(false); 9200 } 9201 } 9202 } else { 9203 if (index < 0) { 9204 return; 9205 } 9206 desc = mSuspendedEffects.valueAt(index); 9207 if (desc->mRefCount <= 0) { 9208 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9209 desc->mRefCount = 1; 9210 } 9211 if (--desc->mRefCount == 0) { 9212 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9213 if (desc->mEffect != 0) { 9214 sp<EffectModule> effect = desc->mEffect.promote(); 9215 if (effect != 0) { 9216 effect->setSuspended(false); 9217 sp<EffectHandle> handle = effect->controlHandle(); 9218 if (handle != 0) { 9219 effect->setEnabled(handle->enabled()); 9220 } 9221 } 9222 desc->mEffect.clear(); 9223 } 9224 mSuspendedEffects.removeItemsAt(index); 9225 } 9226 } 9227} 9228 9229// must be called with ThreadBase::mLock held 9230void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9231{ 9232 sp<SuspendedEffectDesc> desc; 9233 9234 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9235 if (suspend) { 9236 if (index >= 0) { 9237 desc = mSuspendedEffects.valueAt(index); 9238 } else { 9239 desc = new SuspendedEffectDesc(); 9240 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9241 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9242 } 9243 if (desc->mRefCount++ == 0) { 9244 Vector< sp<EffectModule> > effects; 9245 getSuspendEligibleEffects(effects); 9246 for (size_t i = 0; i < effects.size(); i++) { 9247 setEffectSuspended_l(&effects[i]->desc().type, true); 9248 } 9249 } 9250 } else { 9251 if (index < 0) { 9252 return; 9253 } 9254 desc = mSuspendedEffects.valueAt(index); 9255 if (desc->mRefCount <= 0) { 9256 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9257 desc->mRefCount = 1; 9258 } 9259 if (--desc->mRefCount == 0) { 9260 Vector<const effect_uuid_t *> types; 9261 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9262 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9263 continue; 9264 } 9265 types.add(&mSuspendedEffects.valueAt(i)->mType); 9266 } 9267 for (size_t i = 0; i < types.size(); i++) { 9268 setEffectSuspended_l(types[i], false); 9269 } 9270 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9271 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9272 } 9273 } 9274} 9275 9276 9277// The volume effect is used for automated tests only 9278#ifndef OPENSL_ES_H_ 9279static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9280 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9281const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9282#endif //OPENSL_ES_H_ 9283 9284bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9285{ 9286 // auxiliary effects and visualizer are never suspended on output mix 9287 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9288 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9289 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9290 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9291 return false; 9292 } 9293 return true; 9294} 9295 9296void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9297{ 9298 effects.clear(); 9299 for (size_t i = 0; i < mEffects.size(); i++) { 9300 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9301 effects.add(mEffects[i]); 9302 } 9303 } 9304} 9305 9306sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9307 const effect_uuid_t *type) 9308{ 9309 sp<EffectModule> effect = getEffectFromType_l(type); 9310 return effect != 0 && effect->isEnabled() ? effect : 0; 9311} 9312 9313void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9314 bool enabled) 9315{ 9316 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9317 if (enabled) { 9318 if (index < 0) { 9319 // if the effect is not suspend check if all effects are suspended 9320 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9321 if (index < 0) { 9322 return; 9323 } 9324 if (!isEffectEligibleForSuspend(effect->desc())) { 9325 return; 9326 } 9327 setEffectSuspended_l(&effect->desc().type, enabled); 9328 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9329 if (index < 0) { 9330 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9331 return; 9332 } 9333 } 9334 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9335 effect->desc().type.timeLow); 9336 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9337 // if effect is requested to suspended but was not yet enabled, supend it now. 9338 if (desc->mEffect == 0) { 9339 desc->mEffect = effect; 9340 effect->setEnabled(false); 9341 effect->setSuspended(true); 9342 } 9343 } else { 9344 if (index < 0) { 9345 return; 9346 } 9347 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9348 effect->desc().type.timeLow); 9349 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9350 desc->mEffect.clear(); 9351 effect->setSuspended(false); 9352 } 9353} 9354 9355#undef LOG_TAG 9356#define LOG_TAG "AudioFlinger" 9357 9358// ---------------------------------------------------------------------------- 9359 9360status_t AudioFlinger::onTransact( 9361 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9362{ 9363 return BnAudioFlinger::onTransact(code, data, reply, flags); 9364} 9365 9366}; // namespace android 9367