AudioFlinger.cpp revision 109347d421413303eb1678dd9e2aa9d40acf89d2
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 168 // AudioFlinger::setParameters() updates, other threads read w/o lock 169 170// ---------------------------------------------------------------------------- 171 172#ifdef ADD_BATTERY_DATA 173// To collect the amplifier usage 174static void addBatteryData(uint32_t params) { 175 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 176 if (service == NULL) { 177 // it already logged 178 return; 179 } 180 181 service->addBatteryData(params); 182} 183#endif 184 185static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 186{ 187 const hw_module_t *mod; 188 int rc; 189 190 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 191 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 rc = audio_hw_device_open(mod, dev); 197 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 198 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 199 if (rc) { 200 goto out; 201 } 202 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 203 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 204 rc = BAD_VALUE; 205 goto out; 206 } 207 return 0; 208 209out: 210 *dev = NULL; 211 return rc; 212} 213 214// ---------------------------------------------------------------------------- 215 216AudioFlinger::AudioFlinger() 217 : BnAudioFlinger(), 218 mPrimaryHardwareDev(NULL), 219 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 220 mMasterVolume(1.0f), 221 mMasterVolumeSupportLvl(MVS_NONE), 222 mMasterMute(false), 223 mNextUniqueId(1), 224 mMode(AUDIO_MODE_INVALID), 225 mBtNrecIsOff(false) 226{ 227} 228 229void AudioFlinger::onFirstRef() 230{ 231 int rc = 0; 232 233 Mutex::Autolock _l(mLock); 234 235 /* TODO: move all this work into an Init() function */ 236 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 237 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 238 uint32_t int_val; 239 if (1 == sscanf(val_str, "%u", &int_val)) { 240 mStandbyTimeInNsecs = milliseconds(int_val); 241 ALOGI("Using %u mSec as standby time.", int_val); 242 } else { 243 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 244 ALOGI("Using default %u mSec as standby time.", 245 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 246 } 247 } 248 249 mMode = AUDIO_MODE_NORMAL; 250 mMasterVolumeSW = 1.0; 251 mMasterVolume = 1.0; 252 mHardwareStatus = AUDIO_HW_IDLE; 253} 254 255AudioFlinger::~AudioFlinger() 256{ 257 258 while (!mRecordThreads.isEmpty()) { 259 // closeInput() will remove first entry from mRecordThreads 260 closeInput(mRecordThreads.keyAt(0)); 261 } 262 while (!mPlaybackThreads.isEmpty()) { 263 // closeOutput() will remove first entry from mPlaybackThreads 264 closeOutput(mPlaybackThreads.keyAt(0)); 265 } 266 267 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 268 // no mHardwareLock needed, as there are no other references to this 269 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 270 delete mAudioHwDevs.valueAt(i); 271 } 272} 273 274static const char * const audio_interfaces[] = { 275 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 276 AUDIO_HARDWARE_MODULE_ID_A2DP, 277 AUDIO_HARDWARE_MODULE_ID_USB, 278}; 279#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 280 281audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 282{ 283 // if module is 0, the request comes from an old policy manager and we should load 284 // well known modules 285 if (module == 0) { 286 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 287 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 288 loadHwModule_l(audio_interfaces[i]); 289 } 290 } else { 291 // check a match for the requested module handle 292 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 293 if (audioHwdevice != NULL) { 294 return audioHwdevice->hwDevice(); 295 } 296 } 297 // then try to find a module supporting the requested device. 298 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 299 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 300 if ((dev->get_supported_devices(dev) & devices) == devices) 301 return dev; 302 } 303 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 IAudioFlinger::track_flags_t flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 pid_t tid, 449 int *sessionId, 450 status_t *status) 451{ 452 sp<PlaybackThread::Track> track; 453 sp<TrackHandle> trackHandle; 454 sp<Client> client; 455 status_t lStatus; 456 int lSessionId; 457 458 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 459 // but if someone uses binder directly they could bypass that and cause us to crash 460 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 461 ALOGE("createTrack() invalid stream type %d", streamType); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 466 { 467 Mutex::Autolock _l(mLock); 468 PlaybackThread *thread = checkPlaybackThread_l(output); 469 PlaybackThread *effectThread = NULL; 470 if (thread == NULL) { 471 ALOGE("unknown output thread"); 472 lStatus = BAD_VALUE; 473 goto Exit; 474 } 475 476 client = registerPid_l(pid); 477 478 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 479 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 480 // check if an effect chain with the same session ID is present on another 481 // output thread and move it here. 482 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 483 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 484 if (mPlaybackThreads.keyAt(i) != output) { 485 uint32_t sessions = t->hasAudioSession(*sessionId); 486 if (sessions & PlaybackThread::EFFECT_SESSION) { 487 effectThread = t.get(); 488 break; 489 } 490 } 491 } 492 lSessionId = *sessionId; 493 } else { 494 // if no audio session id is provided, create one here 495 lSessionId = nextUniqueId(); 496 if (sessionId != NULL) { 497 *sessionId = lSessionId; 498 } 499 } 500 ALOGV("createTrack() lSessionId: %d", lSessionId); 501 502 track = thread->createTrack_l(client, streamType, sampleRate, format, 503 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 504 505 // move effect chain to this output thread if an effect on same session was waiting 506 // for a track to be created 507 if (lStatus == NO_ERROR && effectThread != NULL) { 508 Mutex::Autolock _dl(thread->mLock); 509 Mutex::Autolock _sl(effectThread->mLock); 510 moveEffectChain_l(lSessionId, effectThread, thread, true); 511 } 512 513 // Look for sync events awaiting for a session to be used. 514 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 515 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 516 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 517 if (lStatus == NO_ERROR) { 518 track->setSyncEvent(mPendingSyncEvents[i]); 519 } else { 520 mPendingSyncEvents[i]->cancel(); 521 } 522 mPendingSyncEvents.removeAt(i); 523 i--; 524 } 525 } 526 } 527 } 528 if (lStatus == NO_ERROR) { 529 trackHandle = new TrackHandle(track); 530 } else { 531 // remove local strong reference to Client before deleting the Track so that the Client 532 // destructor is called by the TrackBase destructor with mLock held 533 client.clear(); 534 track.clear(); 535 } 536 537Exit: 538 if (status != NULL) { 539 *status = lStatus; 540 } 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 586 // should examine all callers and fix them to handle smaller counts 587 return thread->frameCount(); 588} 589 590uint32_t AudioFlinger::latency(audio_io_handle_t output) const 591{ 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGW("latency() unknown thread %d", output); 596 return 0; 597 } 598 return thread->latency(); 599} 600 601status_t AudioFlinger::setMasterVolume(float value) 602{ 603 status_t ret = initCheck(); 604 if (ret != NO_ERROR) { 605 return ret; 606 } 607 608 // check calling permissions 609 if (!settingsAllowed()) { 610 return PERMISSION_DENIED; 611 } 612 613 float swmv = value; 614 615 Mutex::Autolock _l(mLock); 616 617 // when hw supports master volume, don't scale in sw mixer 618 if (MVS_NONE != mMasterVolumeSupportLvl) { 619 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 620 AutoMutex lock(mHardwareLock); 621 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 622 623 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 624 if (NULL != dev->set_master_volume) { 625 dev->set_master_volume(dev, value); 626 } 627 mHardwareStatus = AUDIO_HW_IDLE; 628 } 629 630 swmv = 1.0; 631 } 632 633 mMasterVolume = value; 634 mMasterVolumeSW = swmv; 635 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 636 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 637 638 return NO_ERROR; 639} 640 641status_t AudioFlinger::setMode(audio_mode_t mode) 642{ 643 status_t ret = initCheck(); 644 if (ret != NO_ERROR) { 645 return ret; 646 } 647 648 // check calling permissions 649 if (!settingsAllowed()) { 650 return PERMISSION_DENIED; 651 } 652 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 653 ALOGW("Illegal value: setMode(%d)", mode); 654 return BAD_VALUE; 655 } 656 657 { // scope for the lock 658 AutoMutex lock(mHardwareLock); 659 mHardwareStatus = AUDIO_HW_SET_MODE; 660 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 661 mHardwareStatus = AUDIO_HW_IDLE; 662 } 663 664 if (NO_ERROR == ret) { 665 Mutex::Autolock _l(mLock); 666 mMode = mode; 667 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 668 mPlaybackThreads.valueAt(i)->setMode(mode); 669 } 670 671 return ret; 672} 673 674status_t AudioFlinger::setMicMute(bool state) 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return ret; 679 } 680 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 AutoMutex lock(mHardwareLock); 687 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 688 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 689 mHardwareStatus = AUDIO_HW_IDLE; 690 return ret; 691} 692 693bool AudioFlinger::getMicMute() const 694{ 695 status_t ret = initCheck(); 696 if (ret != NO_ERROR) { 697 return false; 698 } 699 700 bool state = AUDIO_MODE_INVALID; 701 AutoMutex lock(mHardwareLock); 702 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 703 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 704 mHardwareStatus = AUDIO_HW_IDLE; 705 return state; 706} 707 708status_t AudioFlinger::setMasterMute(bool muted) 709{ 710 // check calling permissions 711 if (!settingsAllowed()) { 712 return PERMISSION_DENIED; 713 } 714 715 Mutex::Autolock _l(mLock); 716 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 717 mMasterMute = muted; 718 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 719 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 720 721 return NO_ERROR; 722} 723 724float AudioFlinger::masterVolume() const 725{ 726 Mutex::Autolock _l(mLock); 727 return masterVolume_l(); 728} 729 730float AudioFlinger::masterVolumeSW() const 731{ 732 Mutex::Autolock _l(mLock); 733 return masterVolumeSW_l(); 734} 735 736bool AudioFlinger::masterMute() const 737{ 738 Mutex::Autolock _l(mLock); 739 return masterMute_l(); 740} 741 742float AudioFlinger::masterVolume_l() const 743{ 744 if (MVS_FULL == mMasterVolumeSupportLvl) { 745 float ret_val; 746 AutoMutex lock(mHardwareLock); 747 748 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 749 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 750 (NULL != mPrimaryHardwareDev->get_master_volume), 751 "can't get master volume"); 752 753 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 754 mHardwareStatus = AUDIO_HW_IDLE; 755 return ret_val; 756 } 757 758 return mMasterVolume; 759} 760 761status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 762 audio_io_handle_t output) 763{ 764 // check calling permissions 765 if (!settingsAllowed()) { 766 return PERMISSION_DENIED; 767 } 768 769 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 770 ALOGE("setStreamVolume() invalid stream %d", stream); 771 return BAD_VALUE; 772 } 773 774 AutoMutex lock(mLock); 775 PlaybackThread *thread = NULL; 776 if (output) { 777 thread = checkPlaybackThread_l(output); 778 if (thread == NULL) { 779 return BAD_VALUE; 780 } 781 } 782 783 mStreamTypes[stream].volume = value; 784 785 if (thread == NULL) { 786 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 787 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 788 } 789 } else { 790 thread->setStreamVolume(stream, value); 791 } 792 793 return NO_ERROR; 794} 795 796status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 797{ 798 // check calling permissions 799 if (!settingsAllowed()) { 800 return PERMISSION_DENIED; 801 } 802 803 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 804 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 805 ALOGE("setStreamMute() invalid stream %d", stream); 806 return BAD_VALUE; 807 } 808 809 AutoMutex lock(mLock); 810 mStreamTypes[stream].mute = muted; 811 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 812 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 813 814 return NO_ERROR; 815} 816 817float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 818{ 819 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 820 return 0.0f; 821 } 822 823 AutoMutex lock(mLock); 824 float volume; 825 if (output) { 826 PlaybackThread *thread = checkPlaybackThread_l(output); 827 if (thread == NULL) { 828 return 0.0f; 829 } 830 volume = thread->streamVolume(stream); 831 } else { 832 volume = streamVolume_l(stream); 833 } 834 835 return volume; 836} 837 838bool AudioFlinger::streamMute(audio_stream_type_t stream) const 839{ 840 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 841 return true; 842 } 843 844 AutoMutex lock(mLock); 845 return streamMute_l(stream); 846} 847 848status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 849{ 850 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 851 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 852 // check calling permissions 853 if (!settingsAllowed()) { 854 return PERMISSION_DENIED; 855 } 856 857 // ioHandle == 0 means the parameters are global to the audio hardware interface 858 if (ioHandle == 0) { 859 Mutex::Autolock _l(mLock); 860 status_t final_result = NO_ERROR; 861 { 862 AutoMutex lock(mHardwareLock); 863 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 864 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 865 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 866 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 867 final_result = result ?: final_result; 868 } 869 mHardwareStatus = AUDIO_HW_IDLE; 870 } 871 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 872 AudioParameter param = AudioParameter(keyValuePairs); 873 String8 value; 874 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 875 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 876 if (mBtNrecIsOff != btNrecIsOff) { 877 for (size_t i = 0; i < mRecordThreads.size(); i++) { 878 sp<RecordThread> thread = mRecordThreads.valueAt(i); 879 RecordThread::RecordTrack *track = thread->track(); 880 if (track != NULL) { 881 audio_devices_t device = (audio_devices_t)( 882 thread->device() & AUDIO_DEVICE_IN_ALL); 883 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 884 thread->setEffectSuspended(FX_IID_AEC, 885 suspend, 886 track->sessionId()); 887 thread->setEffectSuspended(FX_IID_NS, 888 suspend, 889 track->sessionId()); 890 } 891 } 892 mBtNrecIsOff = btNrecIsOff; 893 } 894 } 895 String8 screenState; 896 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 897 bool isOff = screenState == "off"; 898 if (isOff != (gScreenState & 1)) { 899 gScreenState = ((gScreenState & ~1) + 2) | isOff; 900 } 901 } 902 return final_result; 903 } 904 905 // hold a strong ref on thread in case closeOutput() or closeInput() is called 906 // and the thread is exited once the lock is released 907 sp<ThreadBase> thread; 908 { 909 Mutex::Autolock _l(mLock); 910 thread = checkPlaybackThread_l(ioHandle); 911 if (thread == NULL) { 912 thread = checkRecordThread_l(ioHandle); 913 } else if (thread == primaryPlaybackThread_l()) { 914 // indicate output device change to all input threads for pre processing 915 AudioParameter param = AudioParameter(keyValuePairs); 916 int value; 917 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 918 (value != 0)) { 919 for (size_t i = 0; i < mRecordThreads.size(); i++) { 920 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 921 } 922 } 923 } 924 } 925 if (thread != 0) { 926 return thread->setParameters(keyValuePairs); 927 } 928 return BAD_VALUE; 929} 930 931String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 932{ 933// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 934// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 935 936 Mutex::Autolock _l(mLock); 937 938 if (ioHandle == 0) { 939 String8 out_s8; 940 941 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 942 char *s; 943 { 944 AutoMutex lock(mHardwareLock); 945 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 946 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 947 s = dev->get_parameters(dev, keys.string()); 948 mHardwareStatus = AUDIO_HW_IDLE; 949 } 950 out_s8 += String8(s ? s : ""); 951 free(s); 952 } 953 return out_s8; 954 } 955 956 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 957 if (playbackThread != NULL) { 958 return playbackThread->getParameters(keys); 959 } 960 RecordThread *recordThread = checkRecordThread_l(ioHandle); 961 if (recordThread != NULL) { 962 return recordThread->getParameters(keys); 963 } 964 return String8(""); 965} 966 967size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 968{ 969 status_t ret = initCheck(); 970 if (ret != NO_ERROR) { 971 return 0; 972 } 973 974 AutoMutex lock(mHardwareLock); 975 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 976 struct audio_config config = { 977 sample_rate: sampleRate, 978 channel_mask: audio_channel_in_mask_from_count(channelCount), 979 format: format, 980 }; 981 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 982 mHardwareStatus = AUDIO_HW_IDLE; 983 return size; 984} 985 986unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 987{ 988 if (ioHandle == 0) { 989 return 0; 990 } 991 992 Mutex::Autolock _l(mLock); 993 994 RecordThread *recordThread = checkRecordThread_l(ioHandle); 995 if (recordThread != NULL) { 996 return recordThread->getInputFramesLost(); 997 } 998 return 0; 999} 1000 1001status_t AudioFlinger::setVoiceVolume(float value) 1002{ 1003 status_t ret = initCheck(); 1004 if (ret != NO_ERROR) { 1005 return ret; 1006 } 1007 1008 // check calling permissions 1009 if (!settingsAllowed()) { 1010 return PERMISSION_DENIED; 1011 } 1012 1013 AutoMutex lock(mHardwareLock); 1014 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1015 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1016 mHardwareStatus = AUDIO_HW_IDLE; 1017 1018 return ret; 1019} 1020 1021status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1022 audio_io_handle_t output) const 1023{ 1024 status_t status; 1025 1026 Mutex::Autolock _l(mLock); 1027 1028 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1029 if (playbackThread != NULL) { 1030 return playbackThread->getRenderPosition(halFrames, dspFrames); 1031 } 1032 1033 return BAD_VALUE; 1034} 1035 1036void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1037{ 1038 1039 Mutex::Autolock _l(mLock); 1040 1041 pid_t pid = IPCThreadState::self()->getCallingPid(); 1042 if (mNotificationClients.indexOfKey(pid) < 0) { 1043 sp<NotificationClient> notificationClient = new NotificationClient(this, 1044 client, 1045 pid); 1046 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1047 1048 mNotificationClients.add(pid, notificationClient); 1049 1050 sp<IBinder> binder = client->asBinder(); 1051 binder->linkToDeath(notificationClient); 1052 1053 // the config change is always sent from playback or record threads to avoid deadlock 1054 // with AudioSystem::gLock 1055 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1056 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1057 } 1058 1059 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1060 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1061 } 1062 } 1063} 1064 1065void AudioFlinger::removeNotificationClient(pid_t pid) 1066{ 1067 Mutex::Autolock _l(mLock); 1068 1069 mNotificationClients.removeItem(pid); 1070 1071 ALOGV("%d died, releasing its sessions", pid); 1072 size_t num = mAudioSessionRefs.size(); 1073 bool removed = false; 1074 for (size_t i = 0; i< num; ) { 1075 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1076 ALOGV(" pid %d @ %d", ref->mPid, i); 1077 if (ref->mPid == pid) { 1078 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1079 mAudioSessionRefs.removeAt(i); 1080 delete ref; 1081 removed = true; 1082 num--; 1083 } else { 1084 i++; 1085 } 1086 } 1087 if (removed) { 1088 purgeStaleEffects_l(); 1089 } 1090} 1091 1092// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1094{ 1095 size_t size = mNotificationClients.size(); 1096 for (size_t i = 0; i < size; i++) { 1097 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1098 param2); 1099 } 1100} 1101 1102// removeClient_l() must be called with AudioFlinger::mLock held 1103void AudioFlinger::removeClient_l(pid_t pid) 1104{ 1105 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1106 mClients.removeItem(pid); 1107} 1108 1109// getEffectThread_l() must be called with AudioFlinger::mLock held 1110sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1111{ 1112 sp<PlaybackThread> thread; 1113 1114 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1115 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1116 ALOG_ASSERT(thread == 0); 1117 thread = mPlaybackThreads.valueAt(i); 1118 } 1119 } 1120 1121 return thread; 1122} 1123 1124// ---------------------------------------------------------------------------- 1125 1126AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1127 uint32_t device, type_t type) 1128 : Thread(false), 1129 mType(type), 1130 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1131 // mChannelMask 1132 mChannelCount(0), 1133 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1134 mParamStatus(NO_ERROR), 1135 mStandby(false), mId(id), 1136 mDevice(device), 1137 mDeathRecipient(new PMDeathRecipient(this)) 1138{ 1139} 1140 1141AudioFlinger::ThreadBase::~ThreadBase() 1142{ 1143 mParamCond.broadcast(); 1144 // do not lock the mutex in destructor 1145 releaseWakeLock_l(); 1146 if (mPowerManager != 0) { 1147 sp<IBinder> binder = mPowerManager->asBinder(); 1148 binder->unlinkToDeath(mDeathRecipient); 1149 } 1150} 1151 1152void AudioFlinger::ThreadBase::exit() 1153{ 1154 ALOGV("ThreadBase::exit"); 1155 { 1156 // This lock prevents the following race in thread (uniprocessor for illustration): 1157 // if (!exitPending()) { 1158 // // context switch from here to exit() 1159 // // exit() calls requestExit(), what exitPending() observes 1160 // // exit() calls signal(), which is dropped since no waiters 1161 // // context switch back from exit() to here 1162 // mWaitWorkCV.wait(...); 1163 // // now thread is hung 1164 // } 1165 AutoMutex lock(mLock); 1166 requestExit(); 1167 mWaitWorkCV.signal(); 1168 } 1169 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1170 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1171 requestExitAndWait(); 1172} 1173 1174status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1175{ 1176 status_t status; 1177 1178 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1179 Mutex::Autolock _l(mLock); 1180 1181 mNewParameters.add(keyValuePairs); 1182 mWaitWorkCV.signal(); 1183 // wait condition with timeout in case the thread loop has exited 1184 // before the request could be processed 1185 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1186 status = mParamStatus; 1187 mWaitWorkCV.signal(); 1188 } else { 1189 status = TIMED_OUT; 1190 } 1191 return status; 1192} 1193 1194void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1195{ 1196 Mutex::Autolock _l(mLock); 1197 sendConfigEvent_l(event, param); 1198} 1199 1200// sendConfigEvent_l() must be called with ThreadBase::mLock held 1201void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1202{ 1203 ConfigEvent configEvent; 1204 configEvent.mEvent = event; 1205 configEvent.mParam = param; 1206 mConfigEvents.add(configEvent); 1207 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1208 mWaitWorkCV.signal(); 1209} 1210 1211void AudioFlinger::ThreadBase::processConfigEvents() 1212{ 1213 mLock.lock(); 1214 while (!mConfigEvents.isEmpty()) { 1215 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1216 ConfigEvent configEvent = mConfigEvents[0]; 1217 mConfigEvents.removeAt(0); 1218 // release mLock before locking AudioFlinger mLock: lock order is always 1219 // AudioFlinger then ThreadBase to avoid cross deadlock 1220 mLock.unlock(); 1221 mAudioFlinger->mLock.lock(); 1222 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1223 mAudioFlinger->mLock.unlock(); 1224 mLock.lock(); 1225 } 1226 mLock.unlock(); 1227} 1228 1229status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1230{ 1231 const size_t SIZE = 256; 1232 char buffer[SIZE]; 1233 String8 result; 1234 1235 bool locked = tryLock(mLock); 1236 if (!locked) { 1237 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1238 write(fd, buffer, strlen(buffer)); 1239 } 1240 1241 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1246 result.append(buffer); 1247 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1250 result.append(buffer); 1251 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1252 result.append(buffer); 1253 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1254 result.append(buffer); 1255 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1256 result.append(buffer); 1257 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1258 result.append(buffer); 1259 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1260 result.append(buffer); 1261 1262 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1263 result.append(buffer); 1264 result.append(" Index Command"); 1265 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1266 snprintf(buffer, SIZE, "\n %02d ", i); 1267 result.append(buffer); 1268 result.append(mNewParameters[i]); 1269 } 1270 1271 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1272 result.append(buffer); 1273 snprintf(buffer, SIZE, " Index event param\n"); 1274 result.append(buffer); 1275 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1276 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1277 result.append(buffer); 1278 } 1279 result.append("\n"); 1280 1281 write(fd, result.string(), result.size()); 1282 1283 if (locked) { 1284 mLock.unlock(); 1285 } 1286 return NO_ERROR; 1287} 1288 1289status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1290{ 1291 const size_t SIZE = 256; 1292 char buffer[SIZE]; 1293 String8 result; 1294 1295 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1296 write(fd, buffer, strlen(buffer)); 1297 1298 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1299 sp<EffectChain> chain = mEffectChains[i]; 1300 if (chain != 0) { 1301 chain->dump(fd, args); 1302 } 1303 } 1304 return NO_ERROR; 1305} 1306 1307void AudioFlinger::ThreadBase::acquireWakeLock() 1308{ 1309 Mutex::Autolock _l(mLock); 1310 acquireWakeLock_l(); 1311} 1312 1313void AudioFlinger::ThreadBase::acquireWakeLock_l() 1314{ 1315 if (mPowerManager == 0) { 1316 // use checkService() to avoid blocking if power service is not up yet 1317 sp<IBinder> binder = 1318 defaultServiceManager()->checkService(String16("power")); 1319 if (binder == 0) { 1320 ALOGW("Thread %s cannot connect to the power manager service", mName); 1321 } else { 1322 mPowerManager = interface_cast<IPowerManager>(binder); 1323 binder->linkToDeath(mDeathRecipient); 1324 } 1325 } 1326 if (mPowerManager != 0) { 1327 sp<IBinder> binder = new BBinder(); 1328 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1329 binder, 1330 String16(mName)); 1331 if (status == NO_ERROR) { 1332 mWakeLockToken = binder; 1333 } 1334 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1335 } 1336} 1337 1338void AudioFlinger::ThreadBase::releaseWakeLock() 1339{ 1340 Mutex::Autolock _l(mLock); 1341 releaseWakeLock_l(); 1342} 1343 1344void AudioFlinger::ThreadBase::releaseWakeLock_l() 1345{ 1346 if (mWakeLockToken != 0) { 1347 ALOGV("releaseWakeLock_l() %s", mName); 1348 if (mPowerManager != 0) { 1349 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1350 } 1351 mWakeLockToken.clear(); 1352 } 1353} 1354 1355void AudioFlinger::ThreadBase::clearPowerManager() 1356{ 1357 Mutex::Autolock _l(mLock); 1358 releaseWakeLock_l(); 1359 mPowerManager.clear(); 1360} 1361 1362void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1363{ 1364 sp<ThreadBase> thread = mThread.promote(); 1365 if (thread != 0) { 1366 thread->clearPowerManager(); 1367 } 1368 ALOGW("power manager service died !!!"); 1369} 1370 1371void AudioFlinger::ThreadBase::setEffectSuspended( 1372 const effect_uuid_t *type, bool suspend, int sessionId) 1373{ 1374 Mutex::Autolock _l(mLock); 1375 setEffectSuspended_l(type, suspend, sessionId); 1376} 1377 1378void AudioFlinger::ThreadBase::setEffectSuspended_l( 1379 const effect_uuid_t *type, bool suspend, int sessionId) 1380{ 1381 sp<EffectChain> chain = getEffectChain_l(sessionId); 1382 if (chain != 0) { 1383 if (type != NULL) { 1384 chain->setEffectSuspended_l(type, suspend); 1385 } else { 1386 chain->setEffectSuspendedAll_l(suspend); 1387 } 1388 } 1389 1390 updateSuspendedSessions_l(type, suspend, sessionId); 1391} 1392 1393void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1394{ 1395 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1396 if (index < 0) { 1397 return; 1398 } 1399 1400 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1401 mSuspendedSessions.editValueAt(index); 1402 1403 for (size_t i = 0; i < sessionEffects.size(); i++) { 1404 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1405 for (int j = 0; j < desc->mRefCount; j++) { 1406 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1407 chain->setEffectSuspendedAll_l(true); 1408 } else { 1409 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1410 desc->mType.timeLow); 1411 chain->setEffectSuspended_l(&desc->mType, true); 1412 } 1413 } 1414 } 1415} 1416 1417void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1418 bool suspend, 1419 int sessionId) 1420{ 1421 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1422 1423 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1424 1425 if (suspend) { 1426 if (index >= 0) { 1427 sessionEffects = mSuspendedSessions.editValueAt(index); 1428 } else { 1429 mSuspendedSessions.add(sessionId, sessionEffects); 1430 } 1431 } else { 1432 if (index < 0) { 1433 return; 1434 } 1435 sessionEffects = mSuspendedSessions.editValueAt(index); 1436 } 1437 1438 1439 int key = EffectChain::kKeyForSuspendAll; 1440 if (type != NULL) { 1441 key = type->timeLow; 1442 } 1443 index = sessionEffects.indexOfKey(key); 1444 1445 sp<SuspendedSessionDesc> desc; 1446 if (suspend) { 1447 if (index >= 0) { 1448 desc = sessionEffects.valueAt(index); 1449 } else { 1450 desc = new SuspendedSessionDesc(); 1451 if (type != NULL) { 1452 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1453 } 1454 sessionEffects.add(key, desc); 1455 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1456 } 1457 desc->mRefCount++; 1458 } else { 1459 if (index < 0) { 1460 return; 1461 } 1462 desc = sessionEffects.valueAt(index); 1463 if (--desc->mRefCount == 0) { 1464 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1465 sessionEffects.removeItemsAt(index); 1466 if (sessionEffects.isEmpty()) { 1467 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1468 sessionId); 1469 mSuspendedSessions.removeItem(sessionId); 1470 } 1471 } 1472 } 1473 if (!sessionEffects.isEmpty()) { 1474 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1475 } 1476} 1477 1478void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1479 bool enabled, 1480 int sessionId) 1481{ 1482 Mutex::Autolock _l(mLock); 1483 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1484} 1485 1486void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1487 bool enabled, 1488 int sessionId) 1489{ 1490 if (mType != RECORD) { 1491 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1492 // another session. This gives the priority to well behaved effect control panels 1493 // and applications not using global effects. 1494 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1495 // global effects 1496 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1497 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1498 } 1499 } 1500 1501 sp<EffectChain> chain = getEffectChain_l(sessionId); 1502 if (chain != 0) { 1503 chain->checkSuspendOnEffectEnabled(effect, enabled); 1504 } 1505} 1506 1507// ---------------------------------------------------------------------------- 1508 1509AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1510 AudioStreamOut* output, 1511 audio_io_handle_t id, 1512 uint32_t device, 1513 type_t type) 1514 : ThreadBase(audioFlinger, id, device, type), 1515 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1516 // Assumes constructor is called by AudioFlinger with it's mLock held, 1517 // but it would be safer to explicitly pass initial masterMute as parameter 1518 mMasterMute(audioFlinger->masterMute_l()), 1519 // mStreamTypes[] initialized in constructor body 1520 mOutput(output), 1521 // Assumes constructor is called by AudioFlinger with it's mLock held, 1522 // but it would be safer to explicitly pass initial masterVolume as parameter 1523 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1524 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1525 mMixerStatus(MIXER_IDLE), 1526 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1527 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1528 mScreenState(gScreenState), 1529 // index 0 is reserved for normal mixer's submix 1530 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1531{ 1532 snprintf(mName, kNameLength, "AudioOut_%X", id); 1533 1534 readOutputParameters(); 1535 1536 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1537 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1538 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1539 stream = (audio_stream_type_t) (stream + 1)) { 1540 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1541 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1542 } 1543 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1544 // because mAudioFlinger doesn't have one to copy from 1545} 1546 1547AudioFlinger::PlaybackThread::~PlaybackThread() 1548{ 1549 delete [] mMixBuffer; 1550} 1551 1552status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1553{ 1554 dumpInternals(fd, args); 1555 dumpTracks(fd, args); 1556 dumpEffectChains(fd, args); 1557 return NO_ERROR; 1558} 1559 1560status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1561{ 1562 const size_t SIZE = 256; 1563 char buffer[SIZE]; 1564 String8 result; 1565 1566 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1567 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1568 const stream_type_t *st = &mStreamTypes[i]; 1569 if (i > 0) { 1570 result.appendFormat(", "); 1571 } 1572 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1573 if (st->mute) { 1574 result.append("M"); 1575 } 1576 } 1577 result.append("\n"); 1578 write(fd, result.string(), result.length()); 1579 result.clear(); 1580 1581 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1582 result.append(buffer); 1583 Track::appendDumpHeader(result); 1584 for (size_t i = 0; i < mTracks.size(); ++i) { 1585 sp<Track> track = mTracks[i]; 1586 if (track != 0) { 1587 track->dump(buffer, SIZE); 1588 result.append(buffer); 1589 } 1590 } 1591 1592 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1593 result.append(buffer); 1594 Track::appendDumpHeader(result); 1595 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1596 sp<Track> track = mActiveTracks[i].promote(); 1597 if (track != 0) { 1598 track->dump(buffer, SIZE); 1599 result.append(buffer); 1600 } 1601 } 1602 write(fd, result.string(), result.size()); 1603 1604 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1605 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1606 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1607 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1608 1609 return NO_ERROR; 1610} 1611 1612status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1613{ 1614 const size_t SIZE = 256; 1615 char buffer[SIZE]; 1616 String8 result; 1617 1618 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1619 result.append(buffer); 1620 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1621 result.append(buffer); 1622 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1623 result.append(buffer); 1624 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1625 result.append(buffer); 1626 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1627 result.append(buffer); 1628 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1629 result.append(buffer); 1630 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1631 result.append(buffer); 1632 write(fd, result.string(), result.size()); 1633 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1634 1635 dumpBase(fd, args); 1636 1637 return NO_ERROR; 1638} 1639 1640// Thread virtuals 1641status_t AudioFlinger::PlaybackThread::readyToRun() 1642{ 1643 status_t status = initCheck(); 1644 if (status == NO_ERROR) { 1645 ALOGI("AudioFlinger's thread %p ready to run", this); 1646 } else { 1647 ALOGE("No working audio driver found."); 1648 } 1649 return status; 1650} 1651 1652void AudioFlinger::PlaybackThread::onFirstRef() 1653{ 1654 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1655} 1656 1657// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1658sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1659 const sp<AudioFlinger::Client>& client, 1660 audio_stream_type_t streamType, 1661 uint32_t sampleRate, 1662 audio_format_t format, 1663 uint32_t channelMask, 1664 int frameCount, 1665 const sp<IMemory>& sharedBuffer, 1666 int sessionId, 1667 IAudioFlinger::track_flags_t flags, 1668 pid_t tid, 1669 status_t *status) 1670{ 1671 sp<Track> track; 1672 status_t lStatus; 1673 1674 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1675 1676 // client expresses a preference for FAST, but we get the final say 1677 if (flags & IAudioFlinger::TRACK_FAST) { 1678 if ( 1679 // not timed 1680 (!isTimed) && 1681 // either of these use cases: 1682 ( 1683 // use case 1: shared buffer with any frame count 1684 ( 1685 (sharedBuffer != 0) 1686 ) || 1687 // use case 2: callback handler and frame count is default or at least as large as HAL 1688 ( 1689 (tid != -1) && 1690 ((frameCount == 0) || 1691 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1692 ) 1693 ) && 1694 // PCM data 1695 audio_is_linear_pcm(format) && 1696 // mono or stereo 1697 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1698 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1699#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1700 // hardware sample rate 1701 (sampleRate == mSampleRate) && 1702#endif 1703 // normal mixer has an associated fast mixer 1704 hasFastMixer() && 1705 // there are sufficient fast track slots available 1706 (mFastTrackAvailMask != 0) 1707 // FIXME test that MixerThread for this fast track has a capable output HAL 1708 // FIXME add a permission test also? 1709 ) { 1710 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1711 if (frameCount == 0) { 1712 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1713 } 1714 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1715 frameCount, mFrameCount); 1716 } else { 1717 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1718 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1719 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1720 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1721 audio_is_linear_pcm(format), 1722 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1723 flags &= ~IAudioFlinger::TRACK_FAST; 1724 // For compatibility with AudioTrack calculation, buffer depth is forced 1725 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1726 // This is probably too conservative, but legacy application code may depend on it. 1727 // If you change this calculation, also review the start threshold which is related. 1728 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1729 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1730 if (minBufCount < 2) { 1731 minBufCount = 2; 1732 } 1733 int minFrameCount = mNormalFrameCount * minBufCount; 1734 if (frameCount < minFrameCount) { 1735 frameCount = minFrameCount; 1736 } 1737 } 1738 } 1739 1740 if (mType == DIRECT) { 1741 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1742 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1743 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1744 "for output %p with format %d", 1745 sampleRate, format, channelMask, mOutput, mFormat); 1746 lStatus = BAD_VALUE; 1747 goto Exit; 1748 } 1749 } 1750 } else { 1751 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1752 if (sampleRate > mSampleRate*2) { 1753 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1754 lStatus = BAD_VALUE; 1755 goto Exit; 1756 } 1757 } 1758 1759 lStatus = initCheck(); 1760 if (lStatus != NO_ERROR) { 1761 ALOGE("Audio driver not initialized."); 1762 goto Exit; 1763 } 1764 1765 { // scope for mLock 1766 Mutex::Autolock _l(mLock); 1767 1768 // all tracks in same audio session must share the same routing strategy otherwise 1769 // conflicts will happen when tracks are moved from one output to another by audio policy 1770 // manager 1771 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1772 for (size_t i = 0; i < mTracks.size(); ++i) { 1773 sp<Track> t = mTracks[i]; 1774 if (t != 0 && !t->isOutputTrack()) { 1775 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1776 if (sessionId == t->sessionId() && strategy != actual) { 1777 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1778 strategy, actual); 1779 lStatus = BAD_VALUE; 1780 goto Exit; 1781 } 1782 } 1783 } 1784 1785 if (!isTimed) { 1786 track = new Track(this, client, streamType, sampleRate, format, 1787 channelMask, frameCount, sharedBuffer, sessionId, flags); 1788 } else { 1789 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1790 channelMask, frameCount, sharedBuffer, sessionId); 1791 } 1792 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1793 lStatus = NO_MEMORY; 1794 goto Exit; 1795 } 1796 mTracks.add(track); 1797 1798 sp<EffectChain> chain = getEffectChain_l(sessionId); 1799 if (chain != 0) { 1800 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1801 track->setMainBuffer(chain->inBuffer()); 1802 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1803 chain->incTrackCnt(); 1804 } 1805 } 1806 1807#ifdef HAVE_REQUEST_PRIORITY 1808 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1809 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1810 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1811 // so ask activity manager to do this on our behalf 1812 int err = requestPriority(callingPid, tid, 1); 1813 if (err != 0) { 1814 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1815 1, callingPid, tid, err); 1816 } 1817 } 1818#endif 1819 1820 lStatus = NO_ERROR; 1821 1822Exit: 1823 if (status) { 1824 *status = lStatus; 1825 } 1826 return track; 1827} 1828 1829uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1830{ 1831 if (mFastMixer != NULL) { 1832 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1833 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1834 } 1835 return latency; 1836} 1837 1838uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1839{ 1840 return latency; 1841} 1842 1843uint32_t AudioFlinger::PlaybackThread::latency() const 1844{ 1845 Mutex::Autolock _l(mLock); 1846 return latency_l(); 1847} 1848uint32_t AudioFlinger::PlaybackThread::latency_l() const 1849{ 1850 if (initCheck() == NO_ERROR) { 1851 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1852 } else { 1853 return 0; 1854 } 1855} 1856 1857void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1858{ 1859 Mutex::Autolock _l(mLock); 1860 mMasterVolume = value; 1861} 1862 1863void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1864{ 1865 Mutex::Autolock _l(mLock); 1866 setMasterMute_l(muted); 1867} 1868 1869void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1870{ 1871 Mutex::Autolock _l(mLock); 1872 mStreamTypes[stream].volume = value; 1873} 1874 1875void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1876{ 1877 Mutex::Autolock _l(mLock); 1878 mStreamTypes[stream].mute = muted; 1879} 1880 1881float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1882{ 1883 Mutex::Autolock _l(mLock); 1884 return mStreamTypes[stream].volume; 1885} 1886 1887// addTrack_l() must be called with ThreadBase::mLock held 1888status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1889{ 1890 status_t status = ALREADY_EXISTS; 1891 1892 // set retry count for buffer fill 1893 track->mRetryCount = kMaxTrackStartupRetries; 1894 if (mActiveTracks.indexOf(track) < 0) { 1895 // the track is newly added, make sure it fills up all its 1896 // buffers before playing. This is to ensure the client will 1897 // effectively get the latency it requested. 1898 track->mFillingUpStatus = Track::FS_FILLING; 1899 track->mResetDone = false; 1900 track->mPresentationCompleteFrames = 0; 1901 mActiveTracks.add(track); 1902 if (track->mainBuffer() != mMixBuffer) { 1903 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1904 if (chain != 0) { 1905 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1906 chain->incActiveTrackCnt(); 1907 } 1908 } 1909 1910 status = NO_ERROR; 1911 } 1912 1913 ALOGV("mWaitWorkCV.broadcast"); 1914 mWaitWorkCV.broadcast(); 1915 1916 return status; 1917} 1918 1919// destroyTrack_l() must be called with ThreadBase::mLock held 1920void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1921{ 1922 track->mState = TrackBase::TERMINATED; 1923 // active tracks are removed by threadLoop() 1924 if (mActiveTracks.indexOf(track) < 0) { 1925 removeTrack_l(track); 1926 } 1927} 1928 1929void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1930{ 1931 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1932 mTracks.remove(track); 1933 deleteTrackName_l(track->name()); 1934 // redundant as track is about to be destroyed, for dumpsys only 1935 track->mName = -1; 1936 if (track->isFastTrack()) { 1937 int index = track->mFastIndex; 1938 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1939 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1940 mFastTrackAvailMask |= 1 << index; 1941 // redundant as track is about to be destroyed, for dumpsys only 1942 track->mFastIndex = -1; 1943 } 1944 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1945 if (chain != 0) { 1946 chain->decTrackCnt(); 1947 } 1948} 1949 1950String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1951{ 1952 String8 out_s8 = String8(""); 1953 char *s; 1954 1955 Mutex::Autolock _l(mLock); 1956 if (initCheck() != NO_ERROR) { 1957 return out_s8; 1958 } 1959 1960 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1961 out_s8 = String8(s); 1962 free(s); 1963 return out_s8; 1964} 1965 1966// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1967void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1968 AudioSystem::OutputDescriptor desc; 1969 void *param2 = NULL; 1970 1971 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1972 1973 switch (event) { 1974 case AudioSystem::OUTPUT_OPENED: 1975 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1976 desc.channels = mChannelMask; 1977 desc.samplingRate = mSampleRate; 1978 desc.format = mFormat; 1979 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1980 desc.latency = latency(); 1981 param2 = &desc; 1982 break; 1983 1984 case AudioSystem::STREAM_CONFIG_CHANGED: 1985 param2 = ¶m; 1986 case AudioSystem::OUTPUT_CLOSED: 1987 default: 1988 break; 1989 } 1990 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1991} 1992 1993void AudioFlinger::PlaybackThread::readOutputParameters() 1994{ 1995 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1996 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1997 mChannelCount = (uint16_t)popcount(mChannelMask); 1998 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1999 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2000 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2001 if (mFrameCount & 15) { 2002 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2003 mFrameCount); 2004 } 2005 2006 // Calculate size of normal mix buffer relative to the HAL output buffer size 2007 double multiplier = 1.0; 2008 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2009 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2010 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2011 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2012 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2013 maxNormalFrameCount = maxNormalFrameCount & ~15; 2014 if (maxNormalFrameCount < minNormalFrameCount) { 2015 maxNormalFrameCount = minNormalFrameCount; 2016 } 2017 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2018 if (multiplier <= 1.0) { 2019 multiplier = 1.0; 2020 } else if (multiplier <= 2.0) { 2021 if (2 * mFrameCount <= maxNormalFrameCount) { 2022 multiplier = 2.0; 2023 } else { 2024 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2025 } 2026 } else { 2027 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2028 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2029 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2030 // FIXME this rounding up should not be done if no HAL SRC 2031 uint32_t truncMult = (uint32_t) multiplier; 2032 if ((truncMult & 1)) { 2033 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2034 ++truncMult; 2035 } 2036 } 2037 multiplier = (double) truncMult; 2038 } 2039 } 2040 mNormalFrameCount = multiplier * mFrameCount; 2041 // round up to nearest 16 frames to satisfy AudioMixer 2042 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2043 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2044 2045 delete[] mMixBuffer; 2046 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2047 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2048 2049 // force reconfiguration of effect chains and engines to take new buffer size and audio 2050 // parameters into account 2051 // Note that mLock is not held when readOutputParameters() is called from the constructor 2052 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2053 // matter. 2054 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2055 Vector< sp<EffectChain> > effectChains = mEffectChains; 2056 for (size_t i = 0; i < effectChains.size(); i ++) { 2057 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2058 } 2059} 2060 2061 2062status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2063{ 2064 if (halFrames == NULL || dspFrames == NULL) { 2065 return BAD_VALUE; 2066 } 2067 Mutex::Autolock _l(mLock); 2068 if (initCheck() != NO_ERROR) { 2069 return INVALID_OPERATION; 2070 } 2071 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2072 2073 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2074} 2075 2076uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2077{ 2078 Mutex::Autolock _l(mLock); 2079 uint32_t result = 0; 2080 if (getEffectChain_l(sessionId) != 0) { 2081 result = EFFECT_SESSION; 2082 } 2083 2084 for (size_t i = 0; i < mTracks.size(); ++i) { 2085 sp<Track> track = mTracks[i]; 2086 if (sessionId == track->sessionId() && 2087 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2088 result |= TRACK_SESSION; 2089 break; 2090 } 2091 } 2092 2093 return result; 2094} 2095 2096uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2097{ 2098 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2099 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2100 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2101 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2102 } 2103 for (size_t i = 0; i < mTracks.size(); i++) { 2104 sp<Track> track = mTracks[i]; 2105 if (sessionId == track->sessionId() && 2106 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2107 return AudioSystem::getStrategyForStream(track->streamType()); 2108 } 2109 } 2110 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2111} 2112 2113 2114AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2115{ 2116 Mutex::Autolock _l(mLock); 2117 return mOutput; 2118} 2119 2120AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2121{ 2122 Mutex::Autolock _l(mLock); 2123 AudioStreamOut *output = mOutput; 2124 mOutput = NULL; 2125 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2126 // must push a NULL and wait for ack 2127 mOutputSink.clear(); 2128 mPipeSink.clear(); 2129 mNormalSink.clear(); 2130 return output; 2131} 2132 2133// this method must always be called either with ThreadBase mLock held or inside the thread loop 2134audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2135{ 2136 if (mOutput == NULL) { 2137 return NULL; 2138 } 2139 return &mOutput->stream->common; 2140} 2141 2142uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2143{ 2144 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2145} 2146 2147status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2148{ 2149 if (!isValidSyncEvent(event)) { 2150 return BAD_VALUE; 2151 } 2152 2153 Mutex::Autolock _l(mLock); 2154 2155 for (size_t i = 0; i < mTracks.size(); ++i) { 2156 sp<Track> track = mTracks[i]; 2157 if (event->triggerSession() == track->sessionId()) { 2158 track->setSyncEvent(event); 2159 return NO_ERROR; 2160 } 2161 } 2162 2163 return NAME_NOT_FOUND; 2164} 2165 2166bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2167{ 2168 switch (event->type()) { 2169 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2170 return true; 2171 default: 2172 break; 2173 } 2174 return false; 2175} 2176 2177void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2178{ 2179 size_t count = tracksToRemove.size(); 2180 if (CC_UNLIKELY(count)) { 2181 for (size_t i = 0 ; i < count ; i++) { 2182 const sp<Track>& track = tracksToRemove.itemAt(i); 2183 if ((track->sharedBuffer() != 0) && 2184 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2185 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2186 } 2187 } 2188 } 2189 2190} 2191 2192// ---------------------------------------------------------------------------- 2193 2194AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2195 audio_io_handle_t id, uint32_t device, type_t type) 2196 : PlaybackThread(audioFlinger, output, id, device, type), 2197 // mAudioMixer below 2198#ifdef SOAKER 2199 mSoaker(NULL), 2200#endif 2201 // mFastMixer below 2202 mFastMixerFutex(0) 2203 // mOutputSink below 2204 // mPipeSink below 2205 // mNormalSink below 2206{ 2207 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2208 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2209 "mFrameCount=%d, mNormalFrameCount=%d", 2210 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2211 mNormalFrameCount); 2212 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2213 2214 // FIXME - Current mixer implementation only supports stereo output 2215 if (mChannelCount == 1) { 2216 ALOGE("Invalid audio hardware channel count"); 2217 } 2218 2219 // create an NBAIO sink for the HAL output stream, and negotiate 2220 mOutputSink = new AudioStreamOutSink(output->stream); 2221 size_t numCounterOffers = 0; 2222 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2223 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2224 ALOG_ASSERT(index == 0); 2225 2226 // initialize fast mixer depending on configuration 2227 bool initFastMixer; 2228 switch (kUseFastMixer) { 2229 case FastMixer_Never: 2230 initFastMixer = false; 2231 break; 2232 case FastMixer_Always: 2233 initFastMixer = true; 2234 break; 2235 case FastMixer_Static: 2236 case FastMixer_Dynamic: 2237 initFastMixer = mFrameCount < mNormalFrameCount; 2238 break; 2239 } 2240 if (initFastMixer) { 2241 2242 // create a MonoPipe to connect our submix to FastMixer 2243 NBAIO_Format format = mOutputSink->format(); 2244 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2245 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2246 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2247 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2248 const NBAIO_Format offers[1] = {format}; 2249 size_t numCounterOffers = 0; 2250 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2251 ALOG_ASSERT(index == 0); 2252 monoPipe->setAvgFrames((mScreenState & 1) ? 2253 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2254 mPipeSink = monoPipe; 2255 2256#ifdef TEE_SINK_FRAMES 2257 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2258 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2259 numCounterOffers = 0; 2260 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2261 ALOG_ASSERT(index == 0); 2262 mTeeSink = teeSink; 2263 PipeReader *teeSource = new PipeReader(*teeSink); 2264 numCounterOffers = 0; 2265 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2266 ALOG_ASSERT(index == 0); 2267 mTeeSource = teeSource; 2268#endif 2269 2270#ifdef SOAKER 2271 // create a soaker as workaround for governor issues 2272 mSoaker = new Soaker(); 2273 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2274 mSoaker->run("Soaker", PRIORITY_LOWEST); 2275#endif 2276 2277 // create fast mixer and configure it initially with just one fast track for our submix 2278 mFastMixer = new FastMixer(); 2279 FastMixerStateQueue *sq = mFastMixer->sq(); 2280#ifdef STATE_QUEUE_DUMP 2281 sq->setObserverDump(&mStateQueueObserverDump); 2282 sq->setMutatorDump(&mStateQueueMutatorDump); 2283#endif 2284 FastMixerState *state = sq->begin(); 2285 FastTrack *fastTrack = &state->mFastTracks[0]; 2286 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2287 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2288 fastTrack->mVolumeProvider = NULL; 2289 fastTrack->mGeneration++; 2290 state->mFastTracksGen++; 2291 state->mTrackMask = 1; 2292 // fast mixer will use the HAL output sink 2293 state->mOutputSink = mOutputSink.get(); 2294 state->mOutputSinkGen++; 2295 state->mFrameCount = mFrameCount; 2296 state->mCommand = FastMixerState::COLD_IDLE; 2297 // already done in constructor initialization list 2298 //mFastMixerFutex = 0; 2299 state->mColdFutexAddr = &mFastMixerFutex; 2300 state->mColdGen++; 2301 state->mDumpState = &mFastMixerDumpState; 2302 state->mTeeSink = mTeeSink.get(); 2303 sq->end(); 2304 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2305 2306 // start the fast mixer 2307 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2308#ifdef HAVE_REQUEST_PRIORITY 2309 pid_t tid = mFastMixer->getTid(); 2310 int err = requestPriority(getpid_cached, tid, 2); 2311 if (err != 0) { 2312 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2313 2, getpid_cached, tid, err); 2314 } 2315#endif 2316 2317#ifdef AUDIO_WATCHDOG 2318 // create and start the watchdog 2319 mAudioWatchdog = new AudioWatchdog(); 2320 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2321 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2322 tid = mAudioWatchdog->getTid(); 2323 err = requestPriority(getpid_cached, tid, 1); 2324 if (err != 0) { 2325 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2326 1, getpid_cached, tid, err); 2327 } 2328#endif 2329 2330 } else { 2331 mFastMixer = NULL; 2332 } 2333 2334 switch (kUseFastMixer) { 2335 case FastMixer_Never: 2336 case FastMixer_Dynamic: 2337 mNormalSink = mOutputSink; 2338 break; 2339 case FastMixer_Always: 2340 mNormalSink = mPipeSink; 2341 break; 2342 case FastMixer_Static: 2343 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2344 break; 2345 } 2346} 2347 2348AudioFlinger::MixerThread::~MixerThread() 2349{ 2350 if (mFastMixer != NULL) { 2351 FastMixerStateQueue *sq = mFastMixer->sq(); 2352 FastMixerState *state = sq->begin(); 2353 if (state->mCommand == FastMixerState::COLD_IDLE) { 2354 int32_t old = android_atomic_inc(&mFastMixerFutex); 2355 if (old == -1) { 2356 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2357 } 2358 } 2359 state->mCommand = FastMixerState::EXIT; 2360 sq->end(); 2361 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2362 mFastMixer->join(); 2363 // Though the fast mixer thread has exited, it's state queue is still valid. 2364 // We'll use that extract the final state which contains one remaining fast track 2365 // corresponding to our sub-mix. 2366 state = sq->begin(); 2367 ALOG_ASSERT(state->mTrackMask == 1); 2368 FastTrack *fastTrack = &state->mFastTracks[0]; 2369 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2370 delete fastTrack->mBufferProvider; 2371 sq->end(false /*didModify*/); 2372 delete mFastMixer; 2373#ifdef SOAKER 2374 if (mSoaker != NULL) { 2375 mSoaker->requestExitAndWait(); 2376 } 2377 delete mSoaker; 2378#endif 2379 if (mAudioWatchdog != 0) { 2380 mAudioWatchdog->requestExit(); 2381 mAudioWatchdog->requestExitAndWait(); 2382 mAudioWatchdog.clear(); 2383 } 2384 } 2385 delete mAudioMixer; 2386} 2387 2388class CpuStats { 2389public: 2390 CpuStats(); 2391 void sample(const String8 &title); 2392#ifdef DEBUG_CPU_USAGE 2393private: 2394 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2395 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2396 2397 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2398 2399 int mCpuNum; // thread's current CPU number 2400 int mCpukHz; // frequency of thread's current CPU in kHz 2401#endif 2402}; 2403 2404CpuStats::CpuStats() 2405#ifdef DEBUG_CPU_USAGE 2406 : mCpuNum(-1), mCpukHz(-1) 2407#endif 2408{ 2409} 2410 2411void CpuStats::sample(const String8 &title) { 2412#ifdef DEBUG_CPU_USAGE 2413 // get current thread's delta CPU time in wall clock ns 2414 double wcNs; 2415 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2416 2417 // record sample for wall clock statistics 2418 if (valid) { 2419 mWcStats.sample(wcNs); 2420 } 2421 2422 // get the current CPU number 2423 int cpuNum = sched_getcpu(); 2424 2425 // get the current CPU frequency in kHz 2426 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2427 2428 // check if either CPU number or frequency changed 2429 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2430 mCpuNum = cpuNum; 2431 mCpukHz = cpukHz; 2432 // ignore sample for purposes of cycles 2433 valid = false; 2434 } 2435 2436 // if no change in CPU number or frequency, then record sample for cycle statistics 2437 if (valid && mCpukHz > 0) { 2438 double cycles = wcNs * cpukHz * 0.000001; 2439 mHzStats.sample(cycles); 2440 } 2441 2442 unsigned n = mWcStats.n(); 2443 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2444 if ((n & 127) == 1) { 2445 long long elapsed = mCpuUsage.elapsed(); 2446 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2447 double perLoop = elapsed / (double) n; 2448 double perLoop100 = perLoop * 0.01; 2449 double perLoop1k = perLoop * 0.001; 2450 double mean = mWcStats.mean(); 2451 double stddev = mWcStats.stddev(); 2452 double minimum = mWcStats.minimum(); 2453 double maximum = mWcStats.maximum(); 2454 double meanCycles = mHzStats.mean(); 2455 double stddevCycles = mHzStats.stddev(); 2456 double minCycles = mHzStats.minimum(); 2457 double maxCycles = mHzStats.maximum(); 2458 mCpuUsage.resetElapsed(); 2459 mWcStats.reset(); 2460 mHzStats.reset(); 2461 ALOGD("CPU usage for %s over past %.1f secs\n" 2462 " (%u mixer loops at %.1f mean ms per loop):\n" 2463 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2464 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2465 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2466 title.string(), 2467 elapsed * .000000001, n, perLoop * .000001, 2468 mean * .001, 2469 stddev * .001, 2470 minimum * .001, 2471 maximum * .001, 2472 mean / perLoop100, 2473 stddev / perLoop100, 2474 minimum / perLoop100, 2475 maximum / perLoop100, 2476 meanCycles / perLoop1k, 2477 stddevCycles / perLoop1k, 2478 minCycles / perLoop1k, 2479 maxCycles / perLoop1k); 2480 2481 } 2482 } 2483#endif 2484}; 2485 2486void AudioFlinger::PlaybackThread::checkSilentMode_l() 2487{ 2488 if (!mMasterMute) { 2489 char value[PROPERTY_VALUE_MAX]; 2490 if (property_get("ro.audio.silent", value, "0") > 0) { 2491 char *endptr; 2492 unsigned long ul = strtoul(value, &endptr, 0); 2493 if (*endptr == '\0' && ul != 0) { 2494 ALOGD("Silence is golden"); 2495 // The setprop command will not allow a property to be changed after 2496 // the first time it is set, so we don't have to worry about un-muting. 2497 setMasterMute_l(true); 2498 } 2499 } 2500 } 2501} 2502 2503bool AudioFlinger::PlaybackThread::threadLoop() 2504{ 2505 Vector< sp<Track> > tracksToRemove; 2506 2507 standbyTime = systemTime(); 2508 2509 // MIXER 2510 nsecs_t lastWarning = 0; 2511if (mType == MIXER) { 2512 longStandbyExit = false; 2513} 2514 2515 // DUPLICATING 2516 // FIXME could this be made local to while loop? 2517 writeFrames = 0; 2518 2519 cacheParameters_l(); 2520 sleepTime = idleSleepTime; 2521 2522if (mType == MIXER) { 2523 sleepTimeShift = 0; 2524} 2525 2526 CpuStats cpuStats; 2527 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2528 2529 acquireWakeLock(); 2530 2531 while (!exitPending()) 2532 { 2533 cpuStats.sample(myName); 2534 2535 Vector< sp<EffectChain> > effectChains; 2536 2537 processConfigEvents(); 2538 2539 { // scope for mLock 2540 2541 Mutex::Autolock _l(mLock); 2542 2543 if (checkForNewParameters_l()) { 2544 cacheParameters_l(); 2545 } 2546 2547 saveOutputTracks(); 2548 2549 // put audio hardware into standby after short delay 2550 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2551 mSuspended > 0)) { 2552 if (!mStandby) { 2553 2554 threadLoop_standby(); 2555 2556 mStandby = true; 2557 mBytesWritten = 0; 2558 } 2559 2560 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2561 // we're about to wait, flush the binder command buffer 2562 IPCThreadState::self()->flushCommands(); 2563 2564 clearOutputTracks(); 2565 2566 if (exitPending()) break; 2567 2568 releaseWakeLock_l(); 2569 // wait until we have something to do... 2570 ALOGV("%s going to sleep", myName.string()); 2571 mWaitWorkCV.wait(mLock); 2572 ALOGV("%s waking up", myName.string()); 2573 acquireWakeLock_l(); 2574 2575 mMixerStatus = MIXER_IDLE; 2576 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2577 2578 checkSilentMode_l(); 2579 2580 standbyTime = systemTime() + standbyDelay; 2581 sleepTime = idleSleepTime; 2582 if (mType == MIXER) { 2583 sleepTimeShift = 0; 2584 } 2585 2586 continue; 2587 } 2588 } 2589 2590 // mMixerStatusIgnoringFastTracks is also updated internally 2591 mMixerStatus = prepareTracks_l(&tracksToRemove); 2592 2593 // prevent any changes in effect chain list and in each effect chain 2594 // during mixing and effect process as the audio buffers could be deleted 2595 // or modified if an effect is created or deleted 2596 lockEffectChains_l(effectChains); 2597 } 2598 2599 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2600 threadLoop_mix(); 2601 } else { 2602 threadLoop_sleepTime(); 2603 } 2604 2605 if (mSuspended > 0) { 2606 sleepTime = suspendSleepTimeUs(); 2607 } 2608 2609 // only process effects if we're going to write 2610 if (sleepTime == 0) { 2611 for (size_t i = 0; i < effectChains.size(); i ++) { 2612 effectChains[i]->process_l(); 2613 } 2614 } 2615 2616 // enable changes in effect chain 2617 unlockEffectChains(effectChains); 2618 2619 // sleepTime == 0 means we must write to audio hardware 2620 if (sleepTime == 0) { 2621 2622 threadLoop_write(); 2623 2624if (mType == MIXER) { 2625 // write blocked detection 2626 nsecs_t now = systemTime(); 2627 nsecs_t delta = now - mLastWriteTime; 2628 if (!mStandby && delta > maxPeriod) { 2629 mNumDelayedWrites++; 2630 if ((now - lastWarning) > kWarningThrottleNs) { 2631#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2632 ScopedTrace st(ATRACE_TAG, "underrun"); 2633#endif 2634 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2635 ns2ms(delta), mNumDelayedWrites, this); 2636 lastWarning = now; 2637 } 2638 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2639 // a different threshold. Or completely removed for what it is worth anyway... 2640 if (mStandby) { 2641 longStandbyExit = true; 2642 } 2643 } 2644} 2645 2646 mStandby = false; 2647 } else { 2648 usleep(sleepTime); 2649 } 2650 2651 // Finally let go of removed track(s), without the lock held 2652 // since we can't guarantee the destructors won't acquire that 2653 // same lock. This will also mutate and push a new fast mixer state. 2654 threadLoop_removeTracks(tracksToRemove); 2655 tracksToRemove.clear(); 2656 2657 // FIXME I don't understand the need for this here; 2658 // it was in the original code but maybe the 2659 // assignment in saveOutputTracks() makes this unnecessary? 2660 clearOutputTracks(); 2661 2662 // Effect chains will be actually deleted here if they were removed from 2663 // mEffectChains list during mixing or effects processing 2664 effectChains.clear(); 2665 2666 // FIXME Note that the above .clear() is no longer necessary since effectChains 2667 // is now local to this block, but will keep it for now (at least until merge done). 2668 } 2669 2670if (mType == MIXER || mType == DIRECT) { 2671 // put output stream into standby mode 2672 if (!mStandby) { 2673 mOutput->stream->common.standby(&mOutput->stream->common); 2674 } 2675} 2676if (mType == DUPLICATING) { 2677 // for DuplicatingThread, standby mode is handled by the outputTracks 2678} 2679 2680 releaseWakeLock(); 2681 2682 ALOGV("Thread %p type %d exiting", this, mType); 2683 return false; 2684} 2685 2686void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2687{ 2688 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2689} 2690 2691void AudioFlinger::MixerThread::threadLoop_write() 2692{ 2693 // FIXME we should only do one push per cycle; confirm this is true 2694 // Start the fast mixer if it's not already running 2695 if (mFastMixer != NULL) { 2696 FastMixerStateQueue *sq = mFastMixer->sq(); 2697 FastMixerState *state = sq->begin(); 2698 if (state->mCommand != FastMixerState::MIX_WRITE && 2699 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2700 if (state->mCommand == FastMixerState::COLD_IDLE) { 2701 int32_t old = android_atomic_inc(&mFastMixerFutex); 2702 if (old == -1) { 2703 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2704 } 2705 if (mAudioWatchdog != 0) { 2706 mAudioWatchdog->resume(); 2707 } 2708 } 2709 state->mCommand = FastMixerState::MIX_WRITE; 2710 sq->end(); 2711 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2712 if (kUseFastMixer == FastMixer_Dynamic) { 2713 mNormalSink = mPipeSink; 2714 } 2715 } else { 2716 sq->end(false /*didModify*/); 2717 } 2718 } 2719 PlaybackThread::threadLoop_write(); 2720} 2721 2722// shared by MIXER and DIRECT, overridden by DUPLICATING 2723void AudioFlinger::PlaybackThread::threadLoop_write() 2724{ 2725 // FIXME rewrite to reduce number of system calls 2726 mLastWriteTime = systemTime(); 2727 mInWrite = true; 2728 int bytesWritten; 2729 2730 // If an NBAIO sink is present, use it to write the normal mixer's submix 2731 if (mNormalSink != 0) { 2732#define mBitShift 2 // FIXME 2733 size_t count = mixBufferSize >> mBitShift; 2734#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2735 Tracer::traceBegin(ATRACE_TAG, "write"); 2736#endif 2737 // update the setpoint when gScreenState changes 2738 uint32_t screenState = gScreenState; 2739 if (screenState != mScreenState) { 2740 mScreenState = screenState; 2741 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2742 if (pipe != NULL) { 2743 pipe->setAvgFrames((mScreenState & 1) ? 2744 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2745 } 2746 } 2747 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2748#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2749 Tracer::traceEnd(ATRACE_TAG); 2750#endif 2751 if (framesWritten > 0) { 2752 bytesWritten = framesWritten << mBitShift; 2753 } else { 2754 bytesWritten = framesWritten; 2755 } 2756 // otherwise use the HAL / AudioStreamOut directly 2757 } else { 2758 // Direct output thread. 2759 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2760 } 2761 2762 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2763 mNumWrites++; 2764 mInWrite = false; 2765} 2766 2767void AudioFlinger::MixerThread::threadLoop_standby() 2768{ 2769 // Idle the fast mixer if it's currently running 2770 if (mFastMixer != NULL) { 2771 FastMixerStateQueue *sq = mFastMixer->sq(); 2772 FastMixerState *state = sq->begin(); 2773 if (!(state->mCommand & FastMixerState::IDLE)) { 2774 state->mCommand = FastMixerState::COLD_IDLE; 2775 state->mColdFutexAddr = &mFastMixerFutex; 2776 state->mColdGen++; 2777 mFastMixerFutex = 0; 2778 sq->end(); 2779 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2780 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2781 if (kUseFastMixer == FastMixer_Dynamic) { 2782 mNormalSink = mOutputSink; 2783 } 2784 if (mAudioWatchdog != 0) { 2785 mAudioWatchdog->pause(); 2786 } 2787 } else { 2788 sq->end(false /*didModify*/); 2789 } 2790 } 2791 PlaybackThread::threadLoop_standby(); 2792} 2793 2794// shared by MIXER and DIRECT, overridden by DUPLICATING 2795void AudioFlinger::PlaybackThread::threadLoop_standby() 2796{ 2797 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2798 mOutput->stream->common.standby(&mOutput->stream->common); 2799} 2800 2801void AudioFlinger::MixerThread::threadLoop_mix() 2802{ 2803 // obtain the presentation timestamp of the next output buffer 2804 int64_t pts; 2805 status_t status = INVALID_OPERATION; 2806 2807 if (NULL != mOutput->stream->get_next_write_timestamp) { 2808 status = mOutput->stream->get_next_write_timestamp( 2809 mOutput->stream, &pts); 2810 } 2811 2812 if (status != NO_ERROR) { 2813 pts = AudioBufferProvider::kInvalidPTS; 2814 } 2815 2816 // mix buffers... 2817 mAudioMixer->process(pts); 2818 // increase sleep time progressively when application underrun condition clears. 2819 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2820 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2821 // such that we would underrun the audio HAL. 2822 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2823 sleepTimeShift--; 2824 } 2825 sleepTime = 0; 2826 standbyTime = systemTime() + standbyDelay; 2827 //TODO: delay standby when effects have a tail 2828} 2829 2830void AudioFlinger::MixerThread::threadLoop_sleepTime() 2831{ 2832 // If no tracks are ready, sleep once for the duration of an output 2833 // buffer size, then write 0s to the output 2834 if (sleepTime == 0) { 2835 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2836 sleepTime = activeSleepTime >> sleepTimeShift; 2837 if (sleepTime < kMinThreadSleepTimeUs) { 2838 sleepTime = kMinThreadSleepTimeUs; 2839 } 2840 // reduce sleep time in case of consecutive application underruns to avoid 2841 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2842 // duration we would end up writing less data than needed by the audio HAL if 2843 // the condition persists. 2844 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2845 sleepTimeShift++; 2846 } 2847 } else { 2848 sleepTime = idleSleepTime; 2849 } 2850 } else if (mBytesWritten != 0 || 2851 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2852 memset (mMixBuffer, 0, mixBufferSize); 2853 sleepTime = 0; 2854 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2855 } 2856 // TODO add standby time extension fct of effect tail 2857} 2858 2859// prepareTracks_l() must be called with ThreadBase::mLock held 2860AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2861 Vector< sp<Track> > *tracksToRemove) 2862{ 2863 2864 mixer_state mixerStatus = MIXER_IDLE; 2865 // find out which tracks need to be processed 2866 size_t count = mActiveTracks.size(); 2867 size_t mixedTracks = 0; 2868 size_t tracksWithEffect = 0; 2869 // counts only _active_ fast tracks 2870 size_t fastTracks = 0; 2871 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2872 2873 float masterVolume = mMasterVolume; 2874 bool masterMute = mMasterMute; 2875 2876 if (masterMute) { 2877 masterVolume = 0; 2878 } 2879 // Delegate master volume control to effect in output mix effect chain if needed 2880 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2881 if (chain != 0) { 2882 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2883 chain->setVolume_l(&v, &v); 2884 masterVolume = (float)((v + (1 << 23)) >> 24); 2885 chain.clear(); 2886 } 2887 2888 // prepare a new state to push 2889 FastMixerStateQueue *sq = NULL; 2890 FastMixerState *state = NULL; 2891 bool didModify = false; 2892 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2893 if (mFastMixer != NULL) { 2894 sq = mFastMixer->sq(); 2895 state = sq->begin(); 2896 } 2897 2898 for (size_t i=0 ; i<count ; i++) { 2899 sp<Track> t = mActiveTracks[i].promote(); 2900 if (t == 0) continue; 2901 2902 // this const just means the local variable doesn't change 2903 Track* const track = t.get(); 2904 2905 // process fast tracks 2906 if (track->isFastTrack()) { 2907 2908 // It's theoretically possible (though unlikely) for a fast track to be created 2909 // and then removed within the same normal mix cycle. This is not a problem, as 2910 // the track never becomes active so it's fast mixer slot is never touched. 2911 // The converse, of removing an (active) track and then creating a new track 2912 // at the identical fast mixer slot within the same normal mix cycle, 2913 // is impossible because the slot isn't marked available until the end of each cycle. 2914 int j = track->mFastIndex; 2915 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2916 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2917 FastTrack *fastTrack = &state->mFastTracks[j]; 2918 2919 // Determine whether the track is currently in underrun condition, 2920 // and whether it had a recent underrun. 2921 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2922 FastTrackUnderruns underruns = ftDump->mUnderruns; 2923 uint32_t recentFull = (underruns.mBitFields.mFull - 2924 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2925 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2926 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2927 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2928 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2929 uint32_t recentUnderruns = recentPartial + recentEmpty; 2930 track->mObservedUnderruns = underruns; 2931 // don't count underruns that occur while stopping or pausing 2932 // or stopped which can occur when flush() is called while active 2933 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2934 track->mUnderrunCount += recentUnderruns; 2935 } 2936 2937 // This is similar to the state machine for normal tracks, 2938 // with a few modifications for fast tracks. 2939 bool isActive = true; 2940 switch (track->mState) { 2941 case TrackBase::STOPPING_1: 2942 // track stays active in STOPPING_1 state until first underrun 2943 if (recentUnderruns > 0) { 2944 track->mState = TrackBase::STOPPING_2; 2945 } 2946 break; 2947 case TrackBase::PAUSING: 2948 // ramp down is not yet implemented 2949 track->setPaused(); 2950 break; 2951 case TrackBase::RESUMING: 2952 // ramp up is not yet implemented 2953 track->mState = TrackBase::ACTIVE; 2954 break; 2955 case TrackBase::ACTIVE: 2956 if (recentFull > 0 || recentPartial > 0) { 2957 // track has provided at least some frames recently: reset retry count 2958 track->mRetryCount = kMaxTrackRetries; 2959 } 2960 if (recentUnderruns == 0) { 2961 // no recent underruns: stay active 2962 break; 2963 } 2964 // there has recently been an underrun of some kind 2965 if (track->sharedBuffer() == 0) { 2966 // were any of the recent underruns "empty" (no frames available)? 2967 if (recentEmpty == 0) { 2968 // no, then ignore the partial underruns as they are allowed indefinitely 2969 break; 2970 } 2971 // there has recently been an "empty" underrun: decrement the retry counter 2972 if (--(track->mRetryCount) > 0) { 2973 break; 2974 } 2975 // indicate to client process that the track was disabled because of underrun; 2976 // it will then automatically call start() when data is available 2977 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2978 // remove from active list, but state remains ACTIVE [confusing but true] 2979 isActive = false; 2980 break; 2981 } 2982 // fall through 2983 case TrackBase::STOPPING_2: 2984 case TrackBase::PAUSED: 2985 case TrackBase::TERMINATED: 2986 case TrackBase::STOPPED: 2987 case TrackBase::FLUSHED: // flush() while active 2988 // Check for presentation complete if track is inactive 2989 // We have consumed all the buffers of this track. 2990 // This would be incomplete if we auto-paused on underrun 2991 { 2992 size_t audioHALFrames = 2993 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2994 size_t framesWritten = 2995 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2996 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2997 // track stays in active list until presentation is complete 2998 break; 2999 } 3000 } 3001 if (track->isStopping_2()) { 3002 track->mState = TrackBase::STOPPED; 3003 } 3004 if (track->isStopped()) { 3005 // Can't reset directly, as fast mixer is still polling this track 3006 // track->reset(); 3007 // So instead mark this track as needing to be reset after push with ack 3008 resetMask |= 1 << i; 3009 } 3010 isActive = false; 3011 break; 3012 case TrackBase::IDLE: 3013 default: 3014 LOG_FATAL("unexpected track state %d", track->mState); 3015 } 3016 3017 if (isActive) { 3018 // was it previously inactive? 3019 if (!(state->mTrackMask & (1 << j))) { 3020 ExtendedAudioBufferProvider *eabp = track; 3021 VolumeProvider *vp = track; 3022 fastTrack->mBufferProvider = eabp; 3023 fastTrack->mVolumeProvider = vp; 3024 fastTrack->mSampleRate = track->mSampleRate; 3025 fastTrack->mChannelMask = track->mChannelMask; 3026 fastTrack->mGeneration++; 3027 state->mTrackMask |= 1 << j; 3028 didModify = true; 3029 // no acknowledgement required for newly active tracks 3030 } 3031 // cache the combined master volume and stream type volume for fast mixer; this 3032 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3033 track->mCachedVolume = track->isMuted() ? 3034 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3035 ++fastTracks; 3036 } else { 3037 // was it previously active? 3038 if (state->mTrackMask & (1 << j)) { 3039 fastTrack->mBufferProvider = NULL; 3040 fastTrack->mGeneration++; 3041 state->mTrackMask &= ~(1 << j); 3042 didModify = true; 3043 // If any fast tracks were removed, we must wait for acknowledgement 3044 // because we're about to decrement the last sp<> on those tracks. 3045 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3046 } else { 3047 LOG_FATAL("fast track %d should have been active", j); 3048 } 3049 tracksToRemove->add(track); 3050 // Avoids a misleading display in dumpsys 3051 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3052 } 3053 continue; 3054 } 3055 3056 { // local variable scope to avoid goto warning 3057 3058 audio_track_cblk_t* cblk = track->cblk(); 3059 3060 // The first time a track is added we wait 3061 // for all its buffers to be filled before processing it 3062 int name = track->name(); 3063 // make sure that we have enough frames to mix one full buffer. 3064 // enforce this condition only once to enable draining the buffer in case the client 3065 // app does not call stop() and relies on underrun to stop: 3066 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3067 // during last round 3068 uint32_t minFrames = 1; 3069 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3070 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3071 if (t->sampleRate() == (int)mSampleRate) { 3072 minFrames = mNormalFrameCount; 3073 } else { 3074 // +1 for rounding and +1 for additional sample needed for interpolation 3075 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3076 // add frames already consumed but not yet released by the resampler 3077 // because cblk->framesReady() will include these frames 3078 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3079 // the minimum track buffer size is normally twice the number of frames necessary 3080 // to fill one buffer and the resampler should not leave more than one buffer worth 3081 // of unreleased frames after each pass, but just in case... 3082 ALOG_ASSERT(minFrames <= cblk->frameCount); 3083 } 3084 } 3085 if ((track->framesReady() >= minFrames) && track->isReady() && 3086 !track->isPaused() && !track->isTerminated()) 3087 { 3088 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3089 3090 mixedTracks++; 3091 3092 // track->mainBuffer() != mMixBuffer means there is an effect chain 3093 // connected to the track 3094 chain.clear(); 3095 if (track->mainBuffer() != mMixBuffer) { 3096 chain = getEffectChain_l(track->sessionId()); 3097 // Delegate volume control to effect in track effect chain if needed 3098 if (chain != 0) { 3099 tracksWithEffect++; 3100 } else { 3101 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3102 name, track->sessionId()); 3103 } 3104 } 3105 3106 3107 int param = AudioMixer::VOLUME; 3108 if (track->mFillingUpStatus == Track::FS_FILLED) { 3109 // no ramp for the first volume setting 3110 track->mFillingUpStatus = Track::FS_ACTIVE; 3111 if (track->mState == TrackBase::RESUMING) { 3112 track->mState = TrackBase::ACTIVE; 3113 param = AudioMixer::RAMP_VOLUME; 3114 } 3115 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3116 } else if (cblk->server != 0) { 3117 // If the track is stopped before the first frame was mixed, 3118 // do not apply ramp 3119 param = AudioMixer::RAMP_VOLUME; 3120 } 3121 3122 // compute volume for this track 3123 uint32_t vl, vr, va; 3124 if (track->isMuted() || track->isPausing() || 3125 mStreamTypes[track->streamType()].mute) { 3126 vl = vr = va = 0; 3127 if (track->isPausing()) { 3128 track->setPaused(); 3129 } 3130 } else { 3131 3132 // read original volumes with volume control 3133 float typeVolume = mStreamTypes[track->streamType()].volume; 3134 float v = masterVolume * typeVolume; 3135 uint32_t vlr = cblk->getVolumeLR(); 3136 vl = vlr & 0xFFFF; 3137 vr = vlr >> 16; 3138 // track volumes come from shared memory, so can't be trusted and must be clamped 3139 if (vl > MAX_GAIN_INT) { 3140 ALOGV("Track left volume out of range: %04X", vl); 3141 vl = MAX_GAIN_INT; 3142 } 3143 if (vr > MAX_GAIN_INT) { 3144 ALOGV("Track right volume out of range: %04X", vr); 3145 vr = MAX_GAIN_INT; 3146 } 3147 // now apply the master volume and stream type volume 3148 vl = (uint32_t)(v * vl) << 12; 3149 vr = (uint32_t)(v * vr) << 12; 3150 // assuming master volume and stream type volume each go up to 1.0, 3151 // vl and vr are now in 8.24 format 3152 3153 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3154 // send level comes from shared memory and so may be corrupt 3155 if (sendLevel > MAX_GAIN_INT) { 3156 ALOGV("Track send level out of range: %04X", sendLevel); 3157 sendLevel = MAX_GAIN_INT; 3158 } 3159 va = (uint32_t)(v * sendLevel); 3160 } 3161 // Delegate volume control to effect in track effect chain if needed 3162 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3163 // Do not ramp volume if volume is controlled by effect 3164 param = AudioMixer::VOLUME; 3165 track->mHasVolumeController = true; 3166 } else { 3167 // force no volume ramp when volume controller was just disabled or removed 3168 // from effect chain to avoid volume spike 3169 if (track->mHasVolumeController) { 3170 param = AudioMixer::VOLUME; 3171 } 3172 track->mHasVolumeController = false; 3173 } 3174 3175 // Convert volumes from 8.24 to 4.12 format 3176 // This additional clamping is needed in case chain->setVolume_l() overshot 3177 vl = (vl + (1 << 11)) >> 12; 3178 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3179 vr = (vr + (1 << 11)) >> 12; 3180 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3181 3182 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3183 3184 // XXX: these things DON'T need to be done each time 3185 mAudioMixer->setBufferProvider(name, track); 3186 mAudioMixer->enable(name); 3187 3188 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3189 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3190 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3191 mAudioMixer->setParameter( 3192 name, 3193 AudioMixer::TRACK, 3194 AudioMixer::FORMAT, (void *)track->format()); 3195 mAudioMixer->setParameter( 3196 name, 3197 AudioMixer::TRACK, 3198 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3199 mAudioMixer->setParameter( 3200 name, 3201 AudioMixer::RESAMPLE, 3202 AudioMixer::SAMPLE_RATE, 3203 (void *)(cblk->sampleRate)); 3204 mAudioMixer->setParameter( 3205 name, 3206 AudioMixer::TRACK, 3207 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3208 mAudioMixer->setParameter( 3209 name, 3210 AudioMixer::TRACK, 3211 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3212 3213 // reset retry count 3214 track->mRetryCount = kMaxTrackRetries; 3215 3216 // If one track is ready, set the mixer ready if: 3217 // - the mixer was not ready during previous round OR 3218 // - no other track is not ready 3219 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3220 mixerStatus != MIXER_TRACKS_ENABLED) { 3221 mixerStatus = MIXER_TRACKS_READY; 3222 } 3223 } else { 3224 // clear effect chain input buffer if an active track underruns to avoid sending 3225 // previous audio buffer again to effects 3226 chain = getEffectChain_l(track->sessionId()); 3227 if (chain != 0) { 3228 chain->clearInputBuffer(); 3229 } 3230 3231 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3232 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3233 track->isStopped() || track->isPaused()) { 3234 // We have consumed all the buffers of this track. 3235 // Remove it from the list of active tracks. 3236 // TODO: use actual buffer filling status instead of latency when available from 3237 // audio HAL 3238 size_t audioHALFrames = 3239 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3240 size_t framesWritten = 3241 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3242 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3243 if (track->isStopped()) { 3244 track->reset(); 3245 } 3246 tracksToRemove->add(track); 3247 } 3248 } else { 3249 track->mUnderrunCount++; 3250 // No buffers for this track. Give it a few chances to 3251 // fill a buffer, then remove it from active list. 3252 if (--(track->mRetryCount) <= 0) { 3253 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3254 tracksToRemove->add(track); 3255 // indicate to client process that the track was disabled because of underrun; 3256 // it will then automatically call start() when data is available 3257 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3258 // If one track is not ready, mark the mixer also not ready if: 3259 // - the mixer was ready during previous round OR 3260 // - no other track is ready 3261 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3262 mixerStatus != MIXER_TRACKS_READY) { 3263 mixerStatus = MIXER_TRACKS_ENABLED; 3264 } 3265 } 3266 mAudioMixer->disable(name); 3267 } 3268 3269 } // local variable scope to avoid goto warning 3270track_is_ready: ; 3271 3272 } 3273 3274 // Push the new FastMixer state if necessary 3275 bool pauseAudioWatchdog = false; 3276 if (didModify) { 3277 state->mFastTracksGen++; 3278 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3279 if (kUseFastMixer == FastMixer_Dynamic && 3280 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3281 state->mCommand = FastMixerState::COLD_IDLE; 3282 state->mColdFutexAddr = &mFastMixerFutex; 3283 state->mColdGen++; 3284 mFastMixerFutex = 0; 3285 if (kUseFastMixer == FastMixer_Dynamic) { 3286 mNormalSink = mOutputSink; 3287 } 3288 // If we go into cold idle, need to wait for acknowledgement 3289 // so that fast mixer stops doing I/O. 3290 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3291 pauseAudioWatchdog = true; 3292 } 3293 sq->end(); 3294 } 3295 if (sq != NULL) { 3296 sq->end(didModify); 3297 sq->push(block); 3298 } 3299 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3300 mAudioWatchdog->pause(); 3301 } 3302 3303 // Now perform the deferred reset on fast tracks that have stopped 3304 while (resetMask != 0) { 3305 size_t i = __builtin_ctz(resetMask); 3306 ALOG_ASSERT(i < count); 3307 resetMask &= ~(1 << i); 3308 sp<Track> t = mActiveTracks[i].promote(); 3309 if (t == 0) continue; 3310 Track* track = t.get(); 3311 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3312 track->reset(); 3313 } 3314 3315 // remove all the tracks that need to be... 3316 count = tracksToRemove->size(); 3317 if (CC_UNLIKELY(count)) { 3318 for (size_t i=0 ; i<count ; i++) { 3319 const sp<Track>& track = tracksToRemove->itemAt(i); 3320 mActiveTracks.remove(track); 3321 if (track->mainBuffer() != mMixBuffer) { 3322 chain = getEffectChain_l(track->sessionId()); 3323 if (chain != 0) { 3324 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3325 chain->decActiveTrackCnt(); 3326 } 3327 } 3328 if (track->isTerminated()) { 3329 removeTrack_l(track); 3330 } 3331 } 3332 } 3333 3334 // mix buffer must be cleared if all tracks are connected to an 3335 // effect chain as in this case the mixer will not write to 3336 // mix buffer and track effects will accumulate into it 3337 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3338 // FIXME as a performance optimization, should remember previous zero status 3339 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3340 } 3341 3342 // if any fast tracks, then status is ready 3343 mMixerStatusIgnoringFastTracks = mixerStatus; 3344 if (fastTracks > 0) { 3345 mixerStatus = MIXER_TRACKS_READY; 3346 } 3347 return mixerStatus; 3348} 3349 3350/* 3351The derived values that are cached: 3352 - mixBufferSize from frame count * frame size 3353 - activeSleepTime from activeSleepTimeUs() 3354 - idleSleepTime from idleSleepTimeUs() 3355 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3356 - maxPeriod from frame count and sample rate (MIXER only) 3357 3358The parameters that affect these derived values are: 3359 - frame count 3360 - frame size 3361 - sample rate 3362 - device type: A2DP or not 3363 - device latency 3364 - format: PCM or not 3365 - active sleep time 3366 - idle sleep time 3367*/ 3368 3369void AudioFlinger::PlaybackThread::cacheParameters_l() 3370{ 3371 mixBufferSize = mNormalFrameCount * mFrameSize; 3372 activeSleepTime = activeSleepTimeUs(); 3373 idleSleepTime = idleSleepTimeUs(); 3374} 3375 3376void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3377{ 3378 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3379 this, streamType, mTracks.size()); 3380 Mutex::Autolock _l(mLock); 3381 3382 size_t size = mTracks.size(); 3383 for (size_t i = 0; i < size; i++) { 3384 sp<Track> t = mTracks[i]; 3385 if (t->streamType() == streamType) { 3386 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3387 t->mCblk->cv.signal(); 3388 } 3389 } 3390} 3391 3392// getTrackName_l() must be called with ThreadBase::mLock held 3393int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3394{ 3395 return mAudioMixer->getTrackName(channelMask); 3396} 3397 3398// deleteTrackName_l() must be called with ThreadBase::mLock held 3399void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3400{ 3401 ALOGV("remove track (%d) and delete from mixer", name); 3402 mAudioMixer->deleteTrackName(name); 3403} 3404 3405// checkForNewParameters_l() must be called with ThreadBase::mLock held 3406bool AudioFlinger::MixerThread::checkForNewParameters_l() 3407{ 3408 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3409 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3410 bool reconfig = false; 3411 3412 while (!mNewParameters.isEmpty()) { 3413 3414 if (mFastMixer != NULL) { 3415 FastMixerStateQueue *sq = mFastMixer->sq(); 3416 FastMixerState *state = sq->begin(); 3417 if (!(state->mCommand & FastMixerState::IDLE)) { 3418 previousCommand = state->mCommand; 3419 state->mCommand = FastMixerState::HOT_IDLE; 3420 sq->end(); 3421 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3422 } else { 3423 sq->end(false /*didModify*/); 3424 } 3425 } 3426 3427 status_t status = NO_ERROR; 3428 String8 keyValuePair = mNewParameters[0]; 3429 AudioParameter param = AudioParameter(keyValuePair); 3430 int value; 3431 3432 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3433 reconfig = true; 3434 } 3435 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3436 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3437 status = BAD_VALUE; 3438 } else { 3439 reconfig = true; 3440 } 3441 } 3442 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3443 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3444 status = BAD_VALUE; 3445 } else { 3446 reconfig = true; 3447 } 3448 } 3449 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3450 // do not accept frame count changes if tracks are open as the track buffer 3451 // size depends on frame count and correct behavior would not be guaranteed 3452 // if frame count is changed after track creation 3453 if (!mTracks.isEmpty()) { 3454 status = INVALID_OPERATION; 3455 } else { 3456 reconfig = true; 3457 } 3458 } 3459 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3460#ifdef ADD_BATTERY_DATA 3461 // when changing the audio output device, call addBatteryData to notify 3462 // the change 3463 if ((int)mDevice != value) { 3464 uint32_t params = 0; 3465 // check whether speaker is on 3466 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3467 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3468 } 3469 3470 int deviceWithoutSpeaker 3471 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3472 // check if any other device (except speaker) is on 3473 if (value & deviceWithoutSpeaker ) { 3474 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3475 } 3476 3477 if (params != 0) { 3478 addBatteryData(params); 3479 } 3480 } 3481#endif 3482 3483 // forward device change to effects that have requested to be 3484 // aware of attached audio device. 3485 mDevice = (uint32_t)value; 3486 for (size_t i = 0; i < mEffectChains.size(); i++) { 3487 mEffectChains[i]->setDevice_l(mDevice); 3488 } 3489 } 3490 3491 if (status == NO_ERROR) { 3492 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3493 keyValuePair.string()); 3494 if (!mStandby && status == INVALID_OPERATION) { 3495 mOutput->stream->common.standby(&mOutput->stream->common); 3496 mStandby = true; 3497 mBytesWritten = 0; 3498 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3499 keyValuePair.string()); 3500 } 3501 if (status == NO_ERROR && reconfig) { 3502 delete mAudioMixer; 3503 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3504 mAudioMixer = NULL; 3505 readOutputParameters(); 3506 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3507 for (size_t i = 0; i < mTracks.size() ; i++) { 3508 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3509 if (name < 0) break; 3510 mTracks[i]->mName = name; 3511 // limit track sample rate to 2 x new output sample rate 3512 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3513 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3514 } 3515 } 3516 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3517 } 3518 } 3519 3520 mNewParameters.removeAt(0); 3521 3522 mParamStatus = status; 3523 mParamCond.signal(); 3524 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3525 // already timed out waiting for the status and will never signal the condition. 3526 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3527 } 3528 3529 if (!(previousCommand & FastMixerState::IDLE)) { 3530 ALOG_ASSERT(mFastMixer != NULL); 3531 FastMixerStateQueue *sq = mFastMixer->sq(); 3532 FastMixerState *state = sq->begin(); 3533 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3534 state->mCommand = previousCommand; 3535 sq->end(); 3536 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3537 } 3538 3539 return reconfig; 3540} 3541 3542status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3543{ 3544 const size_t SIZE = 256; 3545 char buffer[SIZE]; 3546 String8 result; 3547 3548 PlaybackThread::dumpInternals(fd, args); 3549 3550 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3551 result.append(buffer); 3552 write(fd, result.string(), result.size()); 3553 3554 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3555 FastMixerDumpState copy = mFastMixerDumpState; 3556 copy.dump(fd); 3557 3558#ifdef STATE_QUEUE_DUMP 3559 // Similar for state queue 3560 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3561 observerCopy.dump(fd); 3562 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3563 mutatorCopy.dump(fd); 3564#endif 3565 3566 // Write the tee output to a .wav file 3567 NBAIO_Source *teeSource = mTeeSource.get(); 3568 if (teeSource != NULL) { 3569 char teePath[64]; 3570 struct timeval tv; 3571 gettimeofday(&tv, NULL); 3572 struct tm tm; 3573 localtime_r(&tv.tv_sec, &tm); 3574 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3575 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3576 if (teeFd >= 0) { 3577 char wavHeader[44]; 3578 memcpy(wavHeader, 3579 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3580 sizeof(wavHeader)); 3581 NBAIO_Format format = teeSource->format(); 3582 unsigned channelCount = Format_channelCount(format); 3583 ALOG_ASSERT(channelCount <= FCC_2); 3584 unsigned sampleRate = Format_sampleRate(format); 3585 wavHeader[22] = channelCount; // number of channels 3586 wavHeader[24] = sampleRate; // sample rate 3587 wavHeader[25] = sampleRate >> 8; 3588 wavHeader[32] = channelCount * 2; // block alignment 3589 write(teeFd, wavHeader, sizeof(wavHeader)); 3590 size_t total = 0; 3591 bool firstRead = true; 3592 for (;;) { 3593#define TEE_SINK_READ 1024 3594 short buffer[TEE_SINK_READ * FCC_2]; 3595 size_t count = TEE_SINK_READ; 3596 ssize_t actual = teeSource->read(buffer, count); 3597 bool wasFirstRead = firstRead; 3598 firstRead = false; 3599 if (actual <= 0) { 3600 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3601 continue; 3602 } 3603 break; 3604 } 3605 ALOG_ASSERT(actual <= count); 3606 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3607 total += actual; 3608 } 3609 lseek(teeFd, (off_t) 4, SEEK_SET); 3610 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3611 write(teeFd, &temp, sizeof(temp)); 3612 lseek(teeFd, (off_t) 40, SEEK_SET); 3613 temp = total * channelCount * sizeof(short); 3614 write(teeFd, &temp, sizeof(temp)); 3615 close(teeFd); 3616 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3617 } else { 3618 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3619 } 3620 } 3621 3622 if (mAudioWatchdog != 0) { 3623 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3624 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3625 wdCopy.dump(fd); 3626 } 3627 3628 return NO_ERROR; 3629} 3630 3631uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3632{ 3633 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3634} 3635 3636uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3637{ 3638 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3639} 3640 3641void AudioFlinger::MixerThread::cacheParameters_l() 3642{ 3643 PlaybackThread::cacheParameters_l(); 3644 3645 // FIXME: Relaxed timing because of a certain device that can't meet latency 3646 // Should be reduced to 2x after the vendor fixes the driver issue 3647 // increase threshold again due to low power audio mode. The way this warning 3648 // threshold is calculated and its usefulness should be reconsidered anyway. 3649 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3650} 3651 3652// ---------------------------------------------------------------------------- 3653AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3654 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3655 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3656 // mLeftVolFloat, mRightVolFloat 3657{ 3658} 3659 3660AudioFlinger::DirectOutputThread::~DirectOutputThread() 3661{ 3662} 3663 3664AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3665 Vector< sp<Track> > *tracksToRemove 3666) 3667{ 3668 sp<Track> trackToRemove; 3669 3670 mixer_state mixerStatus = MIXER_IDLE; 3671 3672 // find out which tracks need to be processed 3673 if (mActiveTracks.size() != 0) { 3674 sp<Track> t = mActiveTracks[0].promote(); 3675 // The track died recently 3676 if (t == 0) return MIXER_IDLE; 3677 3678 Track* const track = t.get(); 3679 audio_track_cblk_t* cblk = track->cblk(); 3680 3681 // The first time a track is added we wait 3682 // for all its buffers to be filled before processing it 3683 uint32_t minFrames; 3684 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3685 minFrames = mNormalFrameCount; 3686 } else { 3687 minFrames = 1; 3688 } 3689 if ((track->framesReady() >= minFrames) && track->isReady() && 3690 !track->isPaused() && !track->isTerminated()) 3691 { 3692 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3693 3694 if (track->mFillingUpStatus == Track::FS_FILLED) { 3695 track->mFillingUpStatus = Track::FS_ACTIVE; 3696 mLeftVolFloat = mRightVolFloat = 0; 3697 if (track->mState == TrackBase::RESUMING) { 3698 track->mState = TrackBase::ACTIVE; 3699 } 3700 } 3701 3702 // compute volume for this track 3703 float left, right; 3704 if (track->isMuted() || mMasterMute || track->isPausing() || 3705 mStreamTypes[track->streamType()].mute) { 3706 left = right = 0; 3707 if (track->isPausing()) { 3708 track->setPaused(); 3709 } 3710 } else { 3711 float typeVolume = mStreamTypes[track->streamType()].volume; 3712 float v = mMasterVolume * typeVolume; 3713 uint32_t vlr = cblk->getVolumeLR(); 3714 float v_clamped = v * (vlr & 0xFFFF); 3715 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3716 left = v_clamped/MAX_GAIN; 3717 v_clamped = v * (vlr >> 16); 3718 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3719 right = v_clamped/MAX_GAIN; 3720 } 3721 3722 if (left != mLeftVolFloat || right != mRightVolFloat) { 3723 mLeftVolFloat = left; 3724 mRightVolFloat = right; 3725 3726 // Convert volumes from float to 8.24 3727 uint32_t vl = (uint32_t)(left * (1 << 24)); 3728 uint32_t vr = (uint32_t)(right * (1 << 24)); 3729 3730 // Delegate volume control to effect in track effect chain if needed 3731 // only one effect chain can be present on DirectOutputThread, so if 3732 // there is one, the track is connected to it 3733 if (!mEffectChains.isEmpty()) { 3734 // Do not ramp volume if volume is controlled by effect 3735 mEffectChains[0]->setVolume_l(&vl, &vr); 3736 left = (float)vl / (1 << 24); 3737 right = (float)vr / (1 << 24); 3738 } 3739 mOutput->stream->set_volume(mOutput->stream, left, right); 3740 } 3741 3742 // reset retry count 3743 track->mRetryCount = kMaxTrackRetriesDirect; 3744 mActiveTrack = t; 3745 mixerStatus = MIXER_TRACKS_READY; 3746 } else { 3747 // clear effect chain input buffer if an active track underruns to avoid sending 3748 // previous audio buffer again to effects 3749 if (!mEffectChains.isEmpty()) { 3750 mEffectChains[0]->clearInputBuffer(); 3751 } 3752 3753 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3754 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3755 track->isStopped() || track->isPaused()) { 3756 // We have consumed all the buffers of this track. 3757 // Remove it from the list of active tracks. 3758 // TODO: implement behavior for compressed audio 3759 size_t audioHALFrames = 3760 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3761 size_t framesWritten = 3762 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3763 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3764 if (track->isStopped()) { 3765 track->reset(); 3766 } 3767 trackToRemove = track; 3768 } 3769 } else { 3770 // No buffers for this track. Give it a few chances to 3771 // fill a buffer, then remove it from active list. 3772 if (--(track->mRetryCount) <= 0) { 3773 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3774 trackToRemove = track; 3775 } else { 3776 mixerStatus = MIXER_TRACKS_ENABLED; 3777 } 3778 } 3779 } 3780 } 3781 3782 // FIXME merge this with similar code for removing multiple tracks 3783 // remove all the tracks that need to be... 3784 if (CC_UNLIKELY(trackToRemove != 0)) { 3785 tracksToRemove->add(trackToRemove); 3786 mActiveTracks.remove(trackToRemove); 3787 if (!mEffectChains.isEmpty()) { 3788 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3789 trackToRemove->sessionId()); 3790 mEffectChains[0]->decActiveTrackCnt(); 3791 } 3792 if (trackToRemove->isTerminated()) { 3793 removeTrack_l(trackToRemove); 3794 } 3795 } 3796 3797 return mixerStatus; 3798} 3799 3800void AudioFlinger::DirectOutputThread::threadLoop_mix() 3801{ 3802 AudioBufferProvider::Buffer buffer; 3803 size_t frameCount = mFrameCount; 3804 int8_t *curBuf = (int8_t *)mMixBuffer; 3805 // output audio to hardware 3806 while (frameCount) { 3807 buffer.frameCount = frameCount; 3808 mActiveTrack->getNextBuffer(&buffer); 3809 if (CC_UNLIKELY(buffer.raw == NULL)) { 3810 memset(curBuf, 0, frameCount * mFrameSize); 3811 break; 3812 } 3813 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3814 frameCount -= buffer.frameCount; 3815 curBuf += buffer.frameCount * mFrameSize; 3816 mActiveTrack->releaseBuffer(&buffer); 3817 } 3818 sleepTime = 0; 3819 standbyTime = systemTime() + standbyDelay; 3820 mActiveTrack.clear(); 3821 3822} 3823 3824void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3825{ 3826 if (sleepTime == 0) { 3827 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3828 sleepTime = activeSleepTime; 3829 } else { 3830 sleepTime = idleSleepTime; 3831 } 3832 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3833 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3834 sleepTime = 0; 3835 } 3836} 3837 3838// getTrackName_l() must be called with ThreadBase::mLock held 3839int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3840{ 3841 return 0; 3842} 3843 3844// deleteTrackName_l() must be called with ThreadBase::mLock held 3845void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3846{ 3847} 3848 3849// checkForNewParameters_l() must be called with ThreadBase::mLock held 3850bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3851{ 3852 bool reconfig = false; 3853 3854 while (!mNewParameters.isEmpty()) { 3855 status_t status = NO_ERROR; 3856 String8 keyValuePair = mNewParameters[0]; 3857 AudioParameter param = AudioParameter(keyValuePair); 3858 int value; 3859 3860 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3861 // do not accept frame count changes if tracks are open as the track buffer 3862 // size depends on frame count and correct behavior would not be garantied 3863 // if frame count is changed after track creation 3864 if (!mTracks.isEmpty()) { 3865 status = INVALID_OPERATION; 3866 } else { 3867 reconfig = true; 3868 } 3869 } 3870 if (status == NO_ERROR) { 3871 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3872 keyValuePair.string()); 3873 if (!mStandby && status == INVALID_OPERATION) { 3874 mOutput->stream->common.standby(&mOutput->stream->common); 3875 mStandby = true; 3876 mBytesWritten = 0; 3877 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3878 keyValuePair.string()); 3879 } 3880 if (status == NO_ERROR && reconfig) { 3881 readOutputParameters(); 3882 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3883 } 3884 } 3885 3886 mNewParameters.removeAt(0); 3887 3888 mParamStatus = status; 3889 mParamCond.signal(); 3890 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3891 // already timed out waiting for the status and will never signal the condition. 3892 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3893 } 3894 return reconfig; 3895} 3896 3897uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3898{ 3899 uint32_t time; 3900 if (audio_is_linear_pcm(mFormat)) { 3901 time = PlaybackThread::activeSleepTimeUs(); 3902 } else { 3903 time = 10000; 3904 } 3905 return time; 3906} 3907 3908uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3909{ 3910 uint32_t time; 3911 if (audio_is_linear_pcm(mFormat)) { 3912 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3913 } else { 3914 time = 10000; 3915 } 3916 return time; 3917} 3918 3919uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3920{ 3921 uint32_t time; 3922 if (audio_is_linear_pcm(mFormat)) { 3923 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3924 } else { 3925 time = 10000; 3926 } 3927 return time; 3928} 3929 3930void AudioFlinger::DirectOutputThread::cacheParameters_l() 3931{ 3932 PlaybackThread::cacheParameters_l(); 3933 3934 // use shorter standby delay as on normal output to release 3935 // hardware resources as soon as possible 3936 standbyDelay = microseconds(activeSleepTime*2); 3937} 3938 3939// ---------------------------------------------------------------------------- 3940 3941AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3942 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3943 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3944 mWaitTimeMs(UINT_MAX) 3945{ 3946 addOutputTrack(mainThread); 3947} 3948 3949AudioFlinger::DuplicatingThread::~DuplicatingThread() 3950{ 3951 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3952 mOutputTracks[i]->destroy(); 3953 } 3954} 3955 3956void AudioFlinger::DuplicatingThread::threadLoop_mix() 3957{ 3958 // mix buffers... 3959 if (outputsReady(outputTracks)) { 3960 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3961 } else { 3962 memset(mMixBuffer, 0, mixBufferSize); 3963 } 3964 sleepTime = 0; 3965 writeFrames = mNormalFrameCount; 3966 standbyTime = systemTime() + standbyDelay; 3967} 3968 3969void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3970{ 3971 if (sleepTime == 0) { 3972 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3973 sleepTime = activeSleepTime; 3974 } else { 3975 sleepTime = idleSleepTime; 3976 } 3977 } else if (mBytesWritten != 0) { 3978 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3979 writeFrames = mNormalFrameCount; 3980 memset(mMixBuffer, 0, mixBufferSize); 3981 } else { 3982 // flush remaining overflow buffers in output tracks 3983 writeFrames = 0; 3984 } 3985 sleepTime = 0; 3986 } 3987} 3988 3989void AudioFlinger::DuplicatingThread::threadLoop_write() 3990{ 3991 for (size_t i = 0; i < outputTracks.size(); i++) { 3992 outputTracks[i]->write(mMixBuffer, writeFrames); 3993 } 3994 mBytesWritten += mixBufferSize; 3995} 3996 3997void AudioFlinger::DuplicatingThread::threadLoop_standby() 3998{ 3999 // DuplicatingThread implements standby by stopping all tracks 4000 for (size_t i = 0; i < outputTracks.size(); i++) { 4001 outputTracks[i]->stop(); 4002 } 4003} 4004 4005void AudioFlinger::DuplicatingThread::saveOutputTracks() 4006{ 4007 outputTracks = mOutputTracks; 4008} 4009 4010void AudioFlinger::DuplicatingThread::clearOutputTracks() 4011{ 4012 outputTracks.clear(); 4013} 4014 4015void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4016{ 4017 Mutex::Autolock _l(mLock); 4018 // FIXME explain this formula 4019 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4020 OutputTrack *outputTrack = new OutputTrack(thread, 4021 this, 4022 mSampleRate, 4023 mFormat, 4024 mChannelMask, 4025 frameCount); 4026 if (outputTrack->cblk() != NULL) { 4027 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4028 mOutputTracks.add(outputTrack); 4029 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4030 updateWaitTime_l(); 4031 } 4032} 4033 4034void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4035{ 4036 Mutex::Autolock _l(mLock); 4037 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4038 if (mOutputTracks[i]->thread() == thread) { 4039 mOutputTracks[i]->destroy(); 4040 mOutputTracks.removeAt(i); 4041 updateWaitTime_l(); 4042 return; 4043 } 4044 } 4045 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4046} 4047 4048// caller must hold mLock 4049void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4050{ 4051 mWaitTimeMs = UINT_MAX; 4052 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4053 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4054 if (strong != 0) { 4055 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4056 if (waitTimeMs < mWaitTimeMs) { 4057 mWaitTimeMs = waitTimeMs; 4058 } 4059 } 4060 } 4061} 4062 4063 4064bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4065{ 4066 for (size_t i = 0; i < outputTracks.size(); i++) { 4067 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4068 if (thread == 0) { 4069 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4070 return false; 4071 } 4072 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4073 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4074 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4075 return false; 4076 } 4077 } 4078 return true; 4079} 4080 4081uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4082{ 4083 return (mWaitTimeMs * 1000) / 2; 4084} 4085 4086void AudioFlinger::DuplicatingThread::cacheParameters_l() 4087{ 4088 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4089 updateWaitTime_l(); 4090 4091 MixerThread::cacheParameters_l(); 4092} 4093 4094// ---------------------------------------------------------------------------- 4095 4096// TrackBase constructor must be called with AudioFlinger::mLock held 4097AudioFlinger::ThreadBase::TrackBase::TrackBase( 4098 ThreadBase *thread, 4099 const sp<Client>& client, 4100 uint32_t sampleRate, 4101 audio_format_t format, 4102 uint32_t channelMask, 4103 int frameCount, 4104 const sp<IMemory>& sharedBuffer, 4105 int sessionId) 4106 : RefBase(), 4107 mThread(thread), 4108 mClient(client), 4109 mCblk(NULL), 4110 // mBuffer 4111 // mBufferEnd 4112 mFrameCount(0), 4113 mState(IDLE), 4114 mSampleRate(sampleRate), 4115 mFormat(format), 4116 mStepServerFailed(false), 4117 mSessionId(sessionId) 4118 // mChannelCount 4119 // mChannelMask 4120{ 4121 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4122 4123 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4124 size_t size = sizeof(audio_track_cblk_t); 4125 uint8_t channelCount = popcount(channelMask); 4126 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4127 if (sharedBuffer == 0) { 4128 size += bufferSize; 4129 } 4130 4131 if (client != NULL) { 4132 mCblkMemory = client->heap()->allocate(size); 4133 if (mCblkMemory != 0) { 4134 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4135 if (mCblk != NULL) { // construct the shared structure in-place. 4136 new(mCblk) audio_track_cblk_t(); 4137 // clear all buffers 4138 mCblk->frameCount = frameCount; 4139 mCblk->sampleRate = sampleRate; 4140// uncomment the following lines to quickly test 32-bit wraparound 4141// mCblk->user = 0xffff0000; 4142// mCblk->server = 0xffff0000; 4143// mCblk->userBase = 0xffff0000; 4144// mCblk->serverBase = 0xffff0000; 4145 mChannelCount = channelCount; 4146 mChannelMask = channelMask; 4147 if (sharedBuffer == 0) { 4148 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4149 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4150 // Force underrun condition to avoid false underrun callback until first data is 4151 // written to buffer (other flags are cleared) 4152 mCblk->flags = CBLK_UNDERRUN_ON; 4153 } else { 4154 mBuffer = sharedBuffer->pointer(); 4155 } 4156 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4157 } 4158 } else { 4159 ALOGE("not enough memory for AudioTrack size=%u", size); 4160 client->heap()->dump("AudioTrack"); 4161 return; 4162 } 4163 } else { 4164 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4165 // construct the shared structure in-place. 4166 new(mCblk) audio_track_cblk_t(); 4167 // clear all buffers 4168 mCblk->frameCount = frameCount; 4169 mCblk->sampleRate = sampleRate; 4170// uncomment the following lines to quickly test 32-bit wraparound 4171// mCblk->user = 0xffff0000; 4172// mCblk->server = 0xffff0000; 4173// mCblk->userBase = 0xffff0000; 4174// mCblk->serverBase = 0xffff0000; 4175 mChannelCount = channelCount; 4176 mChannelMask = channelMask; 4177 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4178 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4179 // Force underrun condition to avoid false underrun callback until first data is 4180 // written to buffer (other flags are cleared) 4181 mCblk->flags = CBLK_UNDERRUN_ON; 4182 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4183 } 4184} 4185 4186AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4187{ 4188 if (mCblk != NULL) { 4189 if (mClient == 0) { 4190 delete mCblk; 4191 } else { 4192 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4193 } 4194 } 4195 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4196 if (mClient != 0) { 4197 // Client destructor must run with AudioFlinger mutex locked 4198 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4199 // If the client's reference count drops to zero, the associated destructor 4200 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4201 // relying on the automatic clear() at end of scope. 4202 mClient.clear(); 4203 } 4204} 4205 4206// AudioBufferProvider interface 4207// getNextBuffer() = 0; 4208// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4209void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4210{ 4211 buffer->raw = NULL; 4212 mFrameCount = buffer->frameCount; 4213 // FIXME See note at getNextBuffer() 4214 (void) step(); // ignore return value of step() 4215 buffer->frameCount = 0; 4216} 4217 4218bool AudioFlinger::ThreadBase::TrackBase::step() { 4219 bool result; 4220 audio_track_cblk_t* cblk = this->cblk(); 4221 4222 result = cblk->stepServer(mFrameCount); 4223 if (!result) { 4224 ALOGV("stepServer failed acquiring cblk mutex"); 4225 mStepServerFailed = true; 4226 } 4227 return result; 4228} 4229 4230void AudioFlinger::ThreadBase::TrackBase::reset() { 4231 audio_track_cblk_t* cblk = this->cblk(); 4232 4233 cblk->user = 0; 4234 cblk->server = 0; 4235 cblk->userBase = 0; 4236 cblk->serverBase = 0; 4237 mStepServerFailed = false; 4238 ALOGV("TrackBase::reset"); 4239} 4240 4241int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4242 return (int)mCblk->sampleRate; 4243} 4244 4245void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4246 audio_track_cblk_t* cblk = this->cblk(); 4247 size_t frameSize = cblk->frameSize; 4248 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4249 int8_t *bufferEnd = bufferStart + frames * frameSize; 4250 4251 // Check validity of returned pointer in case the track control block would have been corrupted. 4252 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4253 "TrackBase::getBuffer buffer out of range:\n" 4254 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4255 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4256 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4257 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4258 4259 return bufferStart; 4260} 4261 4262status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4263{ 4264 mSyncEvents.add(event); 4265 return NO_ERROR; 4266} 4267 4268// ---------------------------------------------------------------------------- 4269 4270// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4271AudioFlinger::PlaybackThread::Track::Track( 4272 PlaybackThread *thread, 4273 const sp<Client>& client, 4274 audio_stream_type_t streamType, 4275 uint32_t sampleRate, 4276 audio_format_t format, 4277 uint32_t channelMask, 4278 int frameCount, 4279 const sp<IMemory>& sharedBuffer, 4280 int sessionId, 4281 IAudioFlinger::track_flags_t flags) 4282 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4283 mMute(false), 4284 mFillingUpStatus(FS_INVALID), 4285 // mRetryCount initialized later when needed 4286 mSharedBuffer(sharedBuffer), 4287 mStreamType(streamType), 4288 mName(-1), // see note below 4289 mMainBuffer(thread->mixBuffer()), 4290 mAuxBuffer(NULL), 4291 mAuxEffectId(0), mHasVolumeController(false), 4292 mPresentationCompleteFrames(0), 4293 mFlags(flags), 4294 mFastIndex(-1), 4295 mUnderrunCount(0), 4296 mCachedVolume(1.0) 4297{ 4298 if (mCblk != NULL) { 4299 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4300 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4301 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4302 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4303 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4304 mCblk->mName = mName; 4305 if (mName < 0) { 4306 ALOGE("no more track names available"); 4307 return; 4308 } 4309 // only allocate a fast track index if we were able to allocate a normal track name 4310 if (flags & IAudioFlinger::TRACK_FAST) { 4311 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4312 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4313 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4314 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4315 // FIXME This is too eager. We allocate a fast track index before the 4316 // fast track becomes active. Since fast tracks are a scarce resource, 4317 // this means we are potentially denying other more important fast tracks from 4318 // being created. It would be better to allocate the index dynamically. 4319 mFastIndex = i; 4320 mCblk->mName = i; 4321 // Read the initial underruns because this field is never cleared by the fast mixer 4322 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4323 thread->mFastTrackAvailMask &= ~(1 << i); 4324 } 4325 } 4326 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4327} 4328 4329AudioFlinger::PlaybackThread::Track::~Track() 4330{ 4331 ALOGV("PlaybackThread::Track destructor"); 4332 sp<ThreadBase> thread = mThread.promote(); 4333 if (thread != 0) { 4334 Mutex::Autolock _l(thread->mLock); 4335 mState = TERMINATED; 4336 } 4337} 4338 4339void AudioFlinger::PlaybackThread::Track::destroy() 4340{ 4341 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4342 // by removing it from mTracks vector, so there is a risk that this Tracks's 4343 // destructor is called. As the destructor needs to lock mLock, 4344 // we must acquire a strong reference on this Track before locking mLock 4345 // here so that the destructor is called only when exiting this function. 4346 // On the other hand, as long as Track::destroy() is only called by 4347 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4348 // this Track with its member mTrack. 4349 sp<Track> keep(this); 4350 { // scope for mLock 4351 sp<ThreadBase> thread = mThread.promote(); 4352 if (thread != 0) { 4353 if (!isOutputTrack()) { 4354 if (mState == ACTIVE || mState == RESUMING) { 4355 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4356 4357#ifdef ADD_BATTERY_DATA 4358 // to track the speaker usage 4359 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4360#endif 4361 } 4362 AudioSystem::releaseOutput(thread->id()); 4363 } 4364 Mutex::Autolock _l(thread->mLock); 4365 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4366 playbackThread->destroyTrack_l(this); 4367 } 4368 } 4369} 4370 4371/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4372{ 4373 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4374 " Server User Main buf Aux Buf Flags Underruns\n"); 4375} 4376 4377void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4378{ 4379 uint32_t vlr = mCblk->getVolumeLR(); 4380 if (isFastTrack()) { 4381 sprintf(buffer, " F %2d", mFastIndex); 4382 } else { 4383 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4384 } 4385 track_state state = mState; 4386 char stateChar; 4387 switch (state) { 4388 case IDLE: 4389 stateChar = 'I'; 4390 break; 4391 case TERMINATED: 4392 stateChar = 'T'; 4393 break; 4394 case STOPPING_1: 4395 stateChar = 's'; 4396 break; 4397 case STOPPING_2: 4398 stateChar = '5'; 4399 break; 4400 case STOPPED: 4401 stateChar = 'S'; 4402 break; 4403 case RESUMING: 4404 stateChar = 'R'; 4405 break; 4406 case ACTIVE: 4407 stateChar = 'A'; 4408 break; 4409 case PAUSING: 4410 stateChar = 'p'; 4411 break; 4412 case PAUSED: 4413 stateChar = 'P'; 4414 break; 4415 case FLUSHED: 4416 stateChar = 'F'; 4417 break; 4418 default: 4419 stateChar = '?'; 4420 break; 4421 } 4422 char nowInUnderrun; 4423 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4424 case UNDERRUN_FULL: 4425 nowInUnderrun = ' '; 4426 break; 4427 case UNDERRUN_PARTIAL: 4428 nowInUnderrun = '<'; 4429 break; 4430 case UNDERRUN_EMPTY: 4431 nowInUnderrun = '*'; 4432 break; 4433 default: 4434 nowInUnderrun = '?'; 4435 break; 4436 } 4437 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4438 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4439 (mClient == 0) ? getpid_cached : mClient->pid(), 4440 mStreamType, 4441 mFormat, 4442 mChannelMask, 4443 mSessionId, 4444 mFrameCount, 4445 mCblk->frameCount, 4446 stateChar, 4447 mMute, 4448 mFillingUpStatus, 4449 mCblk->sampleRate, 4450 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4451 20.0 * log10((vlr >> 16) / 4096.0), 4452 mCblk->server, 4453 mCblk->user, 4454 (int)mMainBuffer, 4455 (int)mAuxBuffer, 4456 mCblk->flags, 4457 mUnderrunCount, 4458 nowInUnderrun); 4459} 4460 4461// AudioBufferProvider interface 4462status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4463 AudioBufferProvider::Buffer* buffer, int64_t pts) 4464{ 4465 audio_track_cblk_t* cblk = this->cblk(); 4466 uint32_t framesReady; 4467 uint32_t framesReq = buffer->frameCount; 4468 4469 // Check if last stepServer failed, try to step now 4470 if (mStepServerFailed) { 4471 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4472 // Since the fast mixer is higher priority than client callback thread, 4473 // it does not result in priority inversion for client. 4474 // But a non-blocking solution would be preferable to avoid 4475 // fast mixer being unable to tryLock(), and 4476 // to avoid the extra context switches if the client wakes up, 4477 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4478 if (!step()) goto getNextBuffer_exit; 4479 ALOGV("stepServer recovered"); 4480 mStepServerFailed = false; 4481 } 4482 4483 // FIXME Same as above 4484 framesReady = cblk->framesReady(); 4485 4486 if (CC_LIKELY(framesReady)) { 4487 uint32_t s = cblk->server; 4488 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4489 4490 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4491 if (framesReq > framesReady) { 4492 framesReq = framesReady; 4493 } 4494 if (framesReq > bufferEnd - s) { 4495 framesReq = bufferEnd - s; 4496 } 4497 4498 buffer->raw = getBuffer(s, framesReq); 4499 if (buffer->raw == NULL) goto getNextBuffer_exit; 4500 4501 buffer->frameCount = framesReq; 4502 return NO_ERROR; 4503 } 4504 4505getNextBuffer_exit: 4506 buffer->raw = NULL; 4507 buffer->frameCount = 0; 4508 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4509 return NOT_ENOUGH_DATA; 4510} 4511 4512// Note that framesReady() takes a mutex on the control block using tryLock(). 4513// This could result in priority inversion if framesReady() is called by the normal mixer, 4514// as the normal mixer thread runs at lower 4515// priority than the client's callback thread: there is a short window within framesReady() 4516// during which the normal mixer could be preempted, and the client callback would block. 4517// Another problem can occur if framesReady() is called by the fast mixer: 4518// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4519// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4520size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4521 return mCblk->framesReady(); 4522} 4523 4524// Don't call for fast tracks; the framesReady() could result in priority inversion 4525bool AudioFlinger::PlaybackThread::Track::isReady() const { 4526 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4527 4528 if (framesReady() >= mCblk->frameCount || 4529 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4530 mFillingUpStatus = FS_FILLED; 4531 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4532 return true; 4533 } 4534 return false; 4535} 4536 4537status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4538 int triggerSession) 4539{ 4540 status_t status = NO_ERROR; 4541 ALOGV("start(%d), calling pid %d session %d", 4542 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4543 4544 sp<ThreadBase> thread = mThread.promote(); 4545 if (thread != 0) { 4546 Mutex::Autolock _l(thread->mLock); 4547 track_state state = mState; 4548 // here the track could be either new, or restarted 4549 // in both cases "unstop" the track 4550 if (mState == PAUSED) { 4551 mState = TrackBase::RESUMING; 4552 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4553 } else { 4554 mState = TrackBase::ACTIVE; 4555 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4556 } 4557 4558 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4559 thread->mLock.unlock(); 4560 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4561 thread->mLock.lock(); 4562 4563#ifdef ADD_BATTERY_DATA 4564 // to track the speaker usage 4565 if (status == NO_ERROR) { 4566 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4567 } 4568#endif 4569 } 4570 if (status == NO_ERROR) { 4571 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4572 playbackThread->addTrack_l(this); 4573 } else { 4574 mState = state; 4575 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4576 } 4577 } else { 4578 status = BAD_VALUE; 4579 } 4580 return status; 4581} 4582 4583void AudioFlinger::PlaybackThread::Track::stop() 4584{ 4585 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4586 sp<ThreadBase> thread = mThread.promote(); 4587 if (thread != 0) { 4588 Mutex::Autolock _l(thread->mLock); 4589 track_state state = mState; 4590 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4591 // If the track is not active (PAUSED and buffers full), flush buffers 4592 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4593 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4594 reset(); 4595 mState = STOPPED; 4596 } else if (!isFastTrack()) { 4597 mState = STOPPED; 4598 } else { 4599 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4600 // and then to STOPPED and reset() when presentation is complete 4601 mState = STOPPING_1; 4602 } 4603 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4604 } 4605 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4606 thread->mLock.unlock(); 4607 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4608 thread->mLock.lock(); 4609 4610#ifdef ADD_BATTERY_DATA 4611 // to track the speaker usage 4612 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4613#endif 4614 } 4615 } 4616} 4617 4618void AudioFlinger::PlaybackThread::Track::pause() 4619{ 4620 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4621 sp<ThreadBase> thread = mThread.promote(); 4622 if (thread != 0) { 4623 Mutex::Autolock _l(thread->mLock); 4624 if (mState == ACTIVE || mState == RESUMING) { 4625 mState = PAUSING; 4626 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4627 if (!isOutputTrack()) { 4628 thread->mLock.unlock(); 4629 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4630 thread->mLock.lock(); 4631 4632#ifdef ADD_BATTERY_DATA 4633 // to track the speaker usage 4634 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4635#endif 4636 } 4637 } 4638 } 4639} 4640 4641void AudioFlinger::PlaybackThread::Track::flush() 4642{ 4643 ALOGV("flush(%d)", mName); 4644 sp<ThreadBase> thread = mThread.promote(); 4645 if (thread != 0) { 4646 Mutex::Autolock _l(thread->mLock); 4647 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4648 mState != PAUSING) { 4649 return; 4650 } 4651 // No point remaining in PAUSED state after a flush => go to 4652 // FLUSHED state 4653 mState = FLUSHED; 4654 // do not reset the track if it is still in the process of being stopped or paused. 4655 // this will be done by prepareTracks_l() when the track is stopped. 4656 // prepareTracks_l() will see mState == FLUSHED, then 4657 // remove from active track list, reset(), and trigger presentation complete 4658 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4659 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4660 reset(); 4661 } 4662 } 4663} 4664 4665void AudioFlinger::PlaybackThread::Track::reset() 4666{ 4667 // Do not reset twice to avoid discarding data written just after a flush and before 4668 // the audioflinger thread detects the track is stopped. 4669 if (!mResetDone) { 4670 TrackBase::reset(); 4671 // Force underrun condition to avoid false underrun callback until first data is 4672 // written to buffer 4673 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4674 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4675 mFillingUpStatus = FS_FILLING; 4676 mResetDone = true; 4677 if (mState == FLUSHED) { 4678 mState = IDLE; 4679 } 4680 } 4681} 4682 4683void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4684{ 4685 mMute = muted; 4686} 4687 4688status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4689{ 4690 status_t status = DEAD_OBJECT; 4691 sp<ThreadBase> thread = mThread.promote(); 4692 if (thread != 0) { 4693 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4694 sp<AudioFlinger> af = mClient->audioFlinger(); 4695 4696 Mutex::Autolock _l(af->mLock); 4697 4698 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4699 4700 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4701 Mutex::Autolock _dl(playbackThread->mLock); 4702 Mutex::Autolock _sl(srcThread->mLock); 4703 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4704 if (chain == 0) { 4705 return INVALID_OPERATION; 4706 } 4707 4708 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4709 if (effect == 0) { 4710 return INVALID_OPERATION; 4711 } 4712 srcThread->removeEffect_l(effect); 4713 playbackThread->addEffect_l(effect); 4714 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4715 if (effect->state() == EffectModule::ACTIVE || 4716 effect->state() == EffectModule::STOPPING) { 4717 effect->start(); 4718 } 4719 4720 sp<EffectChain> dstChain = effect->chain().promote(); 4721 if (dstChain == 0) { 4722 srcThread->addEffect_l(effect); 4723 return INVALID_OPERATION; 4724 } 4725 AudioSystem::unregisterEffect(effect->id()); 4726 AudioSystem::registerEffect(&effect->desc(), 4727 srcThread->id(), 4728 dstChain->strategy(), 4729 AUDIO_SESSION_OUTPUT_MIX, 4730 effect->id()); 4731 } 4732 status = playbackThread->attachAuxEffect(this, EffectId); 4733 } 4734 return status; 4735} 4736 4737void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4738{ 4739 mAuxEffectId = EffectId; 4740 mAuxBuffer = buffer; 4741} 4742 4743bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4744 size_t audioHalFrames) 4745{ 4746 // a track is considered presented when the total number of frames written to audio HAL 4747 // corresponds to the number of frames written when presentationComplete() is called for the 4748 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4749 if (mPresentationCompleteFrames == 0) { 4750 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4751 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4752 mPresentationCompleteFrames, audioHalFrames); 4753 } 4754 if (framesWritten >= mPresentationCompleteFrames) { 4755 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4756 mSessionId, framesWritten); 4757 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4758 return true; 4759 } 4760 return false; 4761} 4762 4763void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4764{ 4765 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4766 if (mSyncEvents[i]->type() == type) { 4767 mSyncEvents[i]->trigger(); 4768 mSyncEvents.removeAt(i); 4769 i--; 4770 } 4771 } 4772} 4773 4774// implement VolumeBufferProvider interface 4775 4776uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4777{ 4778 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4779 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4780 uint32_t vlr = mCblk->getVolumeLR(); 4781 uint32_t vl = vlr & 0xFFFF; 4782 uint32_t vr = vlr >> 16; 4783 // track volumes come from shared memory, so can't be trusted and must be clamped 4784 if (vl > MAX_GAIN_INT) { 4785 vl = MAX_GAIN_INT; 4786 } 4787 if (vr > MAX_GAIN_INT) { 4788 vr = MAX_GAIN_INT; 4789 } 4790 // now apply the cached master volume and stream type volume; 4791 // this is trusted but lacks any synchronization or barrier so may be stale 4792 float v = mCachedVolume; 4793 vl *= v; 4794 vr *= v; 4795 // re-combine into U4.16 4796 vlr = (vr << 16) | (vl & 0xFFFF); 4797 // FIXME look at mute, pause, and stop flags 4798 return vlr; 4799} 4800 4801status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4802{ 4803 if (mState == TERMINATED || mState == PAUSED || 4804 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4805 (mState == STOPPED)))) { 4806 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4807 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4808 event->cancel(); 4809 return INVALID_OPERATION; 4810 } 4811 TrackBase::setSyncEvent(event); 4812 return NO_ERROR; 4813} 4814 4815// timed audio tracks 4816 4817sp<AudioFlinger::PlaybackThread::TimedTrack> 4818AudioFlinger::PlaybackThread::TimedTrack::create( 4819 PlaybackThread *thread, 4820 const sp<Client>& client, 4821 audio_stream_type_t streamType, 4822 uint32_t sampleRate, 4823 audio_format_t format, 4824 uint32_t channelMask, 4825 int frameCount, 4826 const sp<IMemory>& sharedBuffer, 4827 int sessionId) { 4828 if (!client->reserveTimedTrack()) 4829 return NULL; 4830 4831 return new TimedTrack( 4832 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4833 sharedBuffer, sessionId); 4834} 4835 4836AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4837 PlaybackThread *thread, 4838 const sp<Client>& client, 4839 audio_stream_type_t streamType, 4840 uint32_t sampleRate, 4841 audio_format_t format, 4842 uint32_t channelMask, 4843 int frameCount, 4844 const sp<IMemory>& sharedBuffer, 4845 int sessionId) 4846 : Track(thread, client, streamType, sampleRate, format, channelMask, 4847 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4848 mQueueHeadInFlight(false), 4849 mTrimQueueHeadOnRelease(false), 4850 mFramesPendingInQueue(0), 4851 mTimedSilenceBuffer(NULL), 4852 mTimedSilenceBufferSize(0), 4853 mTimedAudioOutputOnTime(false), 4854 mMediaTimeTransformValid(false) 4855{ 4856 LocalClock lc; 4857 mLocalTimeFreq = lc.getLocalFreq(); 4858 4859 mLocalTimeToSampleTransform.a_zero = 0; 4860 mLocalTimeToSampleTransform.b_zero = 0; 4861 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4862 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4863 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4864 &mLocalTimeToSampleTransform.a_to_b_denom); 4865 4866 mMediaTimeToSampleTransform.a_zero = 0; 4867 mMediaTimeToSampleTransform.b_zero = 0; 4868 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4869 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4870 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4871 &mMediaTimeToSampleTransform.a_to_b_denom); 4872} 4873 4874AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4875 mClient->releaseTimedTrack(); 4876 delete [] mTimedSilenceBuffer; 4877} 4878 4879status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4880 size_t size, sp<IMemory>* buffer) { 4881 4882 Mutex::Autolock _l(mTimedBufferQueueLock); 4883 4884 trimTimedBufferQueue_l(); 4885 4886 // lazily initialize the shared memory heap for timed buffers 4887 if (mTimedMemoryDealer == NULL) { 4888 const int kTimedBufferHeapSize = 512 << 10; 4889 4890 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4891 "AudioFlingerTimed"); 4892 if (mTimedMemoryDealer == NULL) 4893 return NO_MEMORY; 4894 } 4895 4896 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4897 if (newBuffer == NULL) { 4898 newBuffer = mTimedMemoryDealer->allocate(size); 4899 if (newBuffer == NULL) 4900 return NO_MEMORY; 4901 } 4902 4903 *buffer = newBuffer; 4904 return NO_ERROR; 4905} 4906 4907// caller must hold mTimedBufferQueueLock 4908void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4909 int64_t mediaTimeNow; 4910 { 4911 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4912 if (!mMediaTimeTransformValid) 4913 return; 4914 4915 int64_t targetTimeNow; 4916 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4917 ? mCCHelper.getCommonTime(&targetTimeNow) 4918 : mCCHelper.getLocalTime(&targetTimeNow); 4919 4920 if (OK != res) 4921 return; 4922 4923 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4924 &mediaTimeNow)) { 4925 return; 4926 } 4927 } 4928 4929 size_t trimEnd; 4930 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4931 int64_t bufEnd; 4932 4933 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4934 // We have a next buffer. Just use its PTS as the PTS of the frame 4935 // following the last frame in this buffer. If the stream is sparse 4936 // (ie, there are deliberate gaps left in the stream which should be 4937 // filled with silence by the TimedAudioTrack), then this can result 4938 // in one extra buffer being left un-trimmed when it could have 4939 // been. In general, this is not typical, and we would rather 4940 // optimized away the TS calculation below for the more common case 4941 // where PTSes are contiguous. 4942 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4943 } else { 4944 // We have no next buffer. Compute the PTS of the frame following 4945 // the last frame in this buffer by computing the duration of of 4946 // this frame in media time units and adding it to the PTS of the 4947 // buffer. 4948 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4949 / mCblk->frameSize; 4950 4951 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4952 &bufEnd)) { 4953 ALOGE("Failed to convert frame count of %lld to media time" 4954 " duration" " (scale factor %d/%u) in %s", 4955 frameCount, 4956 mMediaTimeToSampleTransform.a_to_b_numer, 4957 mMediaTimeToSampleTransform.a_to_b_denom, 4958 __PRETTY_FUNCTION__); 4959 break; 4960 } 4961 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4962 } 4963 4964 if (bufEnd > mediaTimeNow) 4965 break; 4966 4967 // Is the buffer we want to use in the middle of a mix operation right 4968 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4969 // from the mixer which should be coming back shortly. 4970 if (!trimEnd && mQueueHeadInFlight) { 4971 mTrimQueueHeadOnRelease = true; 4972 } 4973 } 4974 4975 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4976 if (trimStart < trimEnd) { 4977 // Update the bookkeeping for framesReady() 4978 for (size_t i = trimStart; i < trimEnd; ++i) { 4979 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4980 } 4981 4982 // Now actually remove the buffers from the queue. 4983 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4984 } 4985} 4986 4987void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4988 const char* logTag) { 4989 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4990 "%s called (reason \"%s\"), but timed buffer queue has no" 4991 " elements to trim.", __FUNCTION__, logTag); 4992 4993 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4994 mTimedBufferQueue.removeAt(0); 4995} 4996 4997void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4998 const TimedBuffer& buf, 4999 const char* logTag) { 5000 uint32_t bufBytes = buf.buffer()->size(); 5001 uint32_t consumedAlready = buf.position(); 5002 5003 ALOG_ASSERT(consumedAlready <= bufBytes, 5004 "Bad bookkeeping while updating frames pending. Timed buffer is" 5005 " only %u bytes long, but claims to have consumed %u" 5006 " bytes. (update reason: \"%s\")", 5007 bufBytes, consumedAlready, logTag); 5008 5009 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 5010 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5011 "Bad bookkeeping while updating frames pending. Should have at" 5012 " least %u queued frames, but we think we have only %u. (update" 5013 " reason: \"%s\")", 5014 bufFrames, mFramesPendingInQueue, logTag); 5015 5016 mFramesPendingInQueue -= bufFrames; 5017} 5018 5019status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5020 const sp<IMemory>& buffer, int64_t pts) { 5021 5022 { 5023 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5024 if (!mMediaTimeTransformValid) 5025 return INVALID_OPERATION; 5026 } 5027 5028 Mutex::Autolock _l(mTimedBufferQueueLock); 5029 5030 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5031 mFramesPendingInQueue += bufFrames; 5032 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5033 5034 return NO_ERROR; 5035} 5036 5037status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5038 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5039 5040 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5041 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5042 target); 5043 5044 if (!(target == TimedAudioTrack::LOCAL_TIME || 5045 target == TimedAudioTrack::COMMON_TIME)) { 5046 return BAD_VALUE; 5047 } 5048 5049 Mutex::Autolock lock(mMediaTimeTransformLock); 5050 mMediaTimeTransform = xform; 5051 mMediaTimeTransformTarget = target; 5052 mMediaTimeTransformValid = true; 5053 5054 return NO_ERROR; 5055} 5056 5057#define min(a, b) ((a) < (b) ? (a) : (b)) 5058 5059// implementation of getNextBuffer for tracks whose buffers have timestamps 5060status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5061 AudioBufferProvider::Buffer* buffer, int64_t pts) 5062{ 5063 if (pts == AudioBufferProvider::kInvalidPTS) { 5064 buffer->raw = 0; 5065 buffer->frameCount = 0; 5066 mTimedAudioOutputOnTime = false; 5067 return INVALID_OPERATION; 5068 } 5069 5070 Mutex::Autolock _l(mTimedBufferQueueLock); 5071 5072 ALOG_ASSERT(!mQueueHeadInFlight, 5073 "getNextBuffer called without releaseBuffer!"); 5074 5075 while (true) { 5076 5077 // if we have no timed buffers, then fail 5078 if (mTimedBufferQueue.isEmpty()) { 5079 buffer->raw = 0; 5080 buffer->frameCount = 0; 5081 return NOT_ENOUGH_DATA; 5082 } 5083 5084 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5085 5086 // calculate the PTS of the head of the timed buffer queue expressed in 5087 // local time 5088 int64_t headLocalPTS; 5089 { 5090 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5091 5092 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5093 5094 if (mMediaTimeTransform.a_to_b_denom == 0) { 5095 // the transform represents a pause, so yield silence 5096 timedYieldSilence_l(buffer->frameCount, buffer); 5097 return NO_ERROR; 5098 } 5099 5100 int64_t transformedPTS; 5101 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5102 &transformedPTS)) { 5103 // the transform failed. this shouldn't happen, but if it does 5104 // then just drop this buffer 5105 ALOGW("timedGetNextBuffer transform failed"); 5106 buffer->raw = 0; 5107 buffer->frameCount = 0; 5108 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5109 return NO_ERROR; 5110 } 5111 5112 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5113 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5114 &headLocalPTS)) { 5115 buffer->raw = 0; 5116 buffer->frameCount = 0; 5117 return INVALID_OPERATION; 5118 } 5119 } else { 5120 headLocalPTS = transformedPTS; 5121 } 5122 } 5123 5124 // adjust the head buffer's PTS to reflect the portion of the head buffer 5125 // that has already been consumed 5126 int64_t effectivePTS = headLocalPTS + 5127 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5128 5129 // Calculate the delta in samples between the head of the input buffer 5130 // queue and the start of the next output buffer that will be written. 5131 // If the transformation fails because of over or underflow, it means 5132 // that the sample's position in the output stream is so far out of 5133 // whack that it should just be dropped. 5134 int64_t sampleDelta; 5135 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5136 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5137 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5138 " mix"); 5139 continue; 5140 } 5141 if (!mLocalTimeToSampleTransform.doForwardTransform( 5142 (effectivePTS - pts) << 32, &sampleDelta)) { 5143 ALOGV("*** too late during sample rate transform: dropped buffer"); 5144 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5145 continue; 5146 } 5147 5148 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5149 " sampleDelta=[%d.%08x]", 5150 head.pts(), head.position(), pts, 5151 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5152 + (sampleDelta >> 32)), 5153 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5154 5155 // if the delta between the ideal placement for the next input sample and 5156 // the current output position is within this threshold, then we will 5157 // concatenate the next input samples to the previous output 5158 const int64_t kSampleContinuityThreshold = 5159 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5160 5161 // if this is the first buffer of audio that we're emitting from this track 5162 // then it should be almost exactly on time. 5163 const int64_t kSampleStartupThreshold = 1LL << 32; 5164 5165 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5166 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5167 // the next input is close enough to being on time, so concatenate it 5168 // with the last output 5169 timedYieldSamples_l(buffer); 5170 5171 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5172 head.position(), buffer->frameCount); 5173 return NO_ERROR; 5174 } 5175 5176 // Looks like our output is not on time. Reset our on timed status. 5177 // Next time we mix samples from our input queue, then should be within 5178 // the StartupThreshold. 5179 mTimedAudioOutputOnTime = false; 5180 if (sampleDelta > 0) { 5181 // the gap between the current output position and the proper start of 5182 // the next input sample is too big, so fill it with silence 5183 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5184 5185 timedYieldSilence_l(framesUntilNextInput, buffer); 5186 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5187 return NO_ERROR; 5188 } else { 5189 // the next input sample is late 5190 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5191 size_t onTimeSamplePosition = 5192 head.position() + lateFrames * mCblk->frameSize; 5193 5194 if (onTimeSamplePosition > head.buffer()->size()) { 5195 // all the remaining samples in the head are too late, so 5196 // drop it and move on 5197 ALOGV("*** too late: dropped buffer"); 5198 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5199 continue; 5200 } else { 5201 // skip over the late samples 5202 head.setPosition(onTimeSamplePosition); 5203 5204 // yield the available samples 5205 timedYieldSamples_l(buffer); 5206 5207 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5208 return NO_ERROR; 5209 } 5210 } 5211 } 5212} 5213 5214// Yield samples from the timed buffer queue head up to the given output 5215// buffer's capacity. 5216// 5217// Caller must hold mTimedBufferQueueLock 5218void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5219 AudioBufferProvider::Buffer* buffer) { 5220 5221 const TimedBuffer& head = mTimedBufferQueue[0]; 5222 5223 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5224 head.position()); 5225 5226 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5227 mCblk->frameSize); 5228 size_t framesRequested = buffer->frameCount; 5229 buffer->frameCount = min(framesLeftInHead, framesRequested); 5230 5231 mQueueHeadInFlight = true; 5232 mTimedAudioOutputOnTime = true; 5233} 5234 5235// Yield samples of silence up to the given output buffer's capacity 5236// 5237// Caller must hold mTimedBufferQueueLock 5238void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5239 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5240 5241 // lazily allocate a buffer filled with silence 5242 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5243 delete [] mTimedSilenceBuffer; 5244 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5245 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5246 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5247 } 5248 5249 buffer->raw = mTimedSilenceBuffer; 5250 size_t framesRequested = buffer->frameCount; 5251 buffer->frameCount = min(numFrames, framesRequested); 5252 5253 mTimedAudioOutputOnTime = false; 5254} 5255 5256// AudioBufferProvider interface 5257void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5258 AudioBufferProvider::Buffer* buffer) { 5259 5260 Mutex::Autolock _l(mTimedBufferQueueLock); 5261 5262 // If the buffer which was just released is part of the buffer at the head 5263 // of the queue, be sure to update the amt of the buffer which has been 5264 // consumed. If the buffer being returned is not part of the head of the 5265 // queue, its either because the buffer is part of the silence buffer, or 5266 // because the head of the timed queue was trimmed after the mixer called 5267 // getNextBuffer but before the mixer called releaseBuffer. 5268 if (buffer->raw == mTimedSilenceBuffer) { 5269 ALOG_ASSERT(!mQueueHeadInFlight, 5270 "Queue head in flight during release of silence buffer!"); 5271 goto done; 5272 } 5273 5274 ALOG_ASSERT(mQueueHeadInFlight, 5275 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5276 " head in flight."); 5277 5278 if (mTimedBufferQueue.size()) { 5279 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5280 5281 void* start = head.buffer()->pointer(); 5282 void* end = reinterpret_cast<void*>( 5283 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5284 + head.buffer()->size()); 5285 5286 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5287 "released buffer not within the head of the timed buffer" 5288 " queue; qHead = [%p, %p], released buffer = %p", 5289 start, end, buffer->raw); 5290 5291 head.setPosition(head.position() + 5292 (buffer->frameCount * mCblk->frameSize)); 5293 mQueueHeadInFlight = false; 5294 5295 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5296 "Bad bookkeeping during releaseBuffer! Should have at" 5297 " least %u queued frames, but we think we have only %u", 5298 buffer->frameCount, mFramesPendingInQueue); 5299 5300 mFramesPendingInQueue -= buffer->frameCount; 5301 5302 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5303 || mTrimQueueHeadOnRelease) { 5304 trimTimedBufferQueueHead_l("releaseBuffer"); 5305 mTrimQueueHeadOnRelease = false; 5306 } 5307 } else { 5308 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5309 " buffers in the timed buffer queue"); 5310 } 5311 5312done: 5313 buffer->raw = 0; 5314 buffer->frameCount = 0; 5315} 5316 5317size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5318 Mutex::Autolock _l(mTimedBufferQueueLock); 5319 return mFramesPendingInQueue; 5320} 5321 5322AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5323 : mPTS(0), mPosition(0) {} 5324 5325AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5326 const sp<IMemory>& buffer, int64_t pts) 5327 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5328 5329// ---------------------------------------------------------------------------- 5330 5331// RecordTrack constructor must be called with AudioFlinger::mLock held 5332AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5333 RecordThread *thread, 5334 const sp<Client>& client, 5335 uint32_t sampleRate, 5336 audio_format_t format, 5337 uint32_t channelMask, 5338 int frameCount, 5339 int sessionId) 5340 : TrackBase(thread, client, sampleRate, format, 5341 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5342 mOverflow(false) 5343{ 5344 if (mCblk != NULL) { 5345 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5346 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5347 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5348 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5349 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5350 } else { 5351 mCblk->frameSize = sizeof(int8_t); 5352 } 5353 } 5354} 5355 5356AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5357{ 5358 sp<ThreadBase> thread = mThread.promote(); 5359 if (thread != 0) { 5360 AudioSystem::releaseInput(thread->id()); 5361 } 5362} 5363 5364// AudioBufferProvider interface 5365status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5366{ 5367 audio_track_cblk_t* cblk = this->cblk(); 5368 uint32_t framesAvail; 5369 uint32_t framesReq = buffer->frameCount; 5370 5371 // Check if last stepServer failed, try to step now 5372 if (mStepServerFailed) { 5373 if (!step()) goto getNextBuffer_exit; 5374 ALOGV("stepServer recovered"); 5375 mStepServerFailed = false; 5376 } 5377 5378 framesAvail = cblk->framesAvailable_l(); 5379 5380 if (CC_LIKELY(framesAvail)) { 5381 uint32_t s = cblk->server; 5382 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5383 5384 if (framesReq > framesAvail) { 5385 framesReq = framesAvail; 5386 } 5387 if (framesReq > bufferEnd - s) { 5388 framesReq = bufferEnd - s; 5389 } 5390 5391 buffer->raw = getBuffer(s, framesReq); 5392 if (buffer->raw == NULL) goto getNextBuffer_exit; 5393 5394 buffer->frameCount = framesReq; 5395 return NO_ERROR; 5396 } 5397 5398getNextBuffer_exit: 5399 buffer->raw = NULL; 5400 buffer->frameCount = 0; 5401 return NOT_ENOUGH_DATA; 5402} 5403 5404status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5405 int triggerSession) 5406{ 5407 sp<ThreadBase> thread = mThread.promote(); 5408 if (thread != 0) { 5409 RecordThread *recordThread = (RecordThread *)thread.get(); 5410 return recordThread->start(this, event, triggerSession); 5411 } else { 5412 return BAD_VALUE; 5413 } 5414} 5415 5416void AudioFlinger::RecordThread::RecordTrack::stop() 5417{ 5418 sp<ThreadBase> thread = mThread.promote(); 5419 if (thread != 0) { 5420 RecordThread *recordThread = (RecordThread *)thread.get(); 5421 recordThread->stop(this); 5422 TrackBase::reset(); 5423 // Force overrun condition to avoid false overrun callback until first data is 5424 // read from buffer 5425 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5426 } 5427} 5428 5429void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5430{ 5431 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5432 (mClient == 0) ? getpid_cached : mClient->pid(), 5433 mFormat, 5434 mChannelMask, 5435 mSessionId, 5436 mFrameCount, 5437 mState, 5438 mCblk->sampleRate, 5439 mCblk->server, 5440 mCblk->user); 5441} 5442 5443 5444// ---------------------------------------------------------------------------- 5445 5446AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5447 PlaybackThread *playbackThread, 5448 DuplicatingThread *sourceThread, 5449 uint32_t sampleRate, 5450 audio_format_t format, 5451 uint32_t channelMask, 5452 int frameCount) 5453 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5454 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5455 mActive(false), mSourceThread(sourceThread) 5456{ 5457 5458 if (mCblk != NULL) { 5459 mCblk->flags |= CBLK_DIRECTION_OUT; 5460 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5461 mOutBuffer.frameCount = 0; 5462 playbackThread->mTracks.add(this); 5463 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5464 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5465 mCblk, mBuffer, mCblk->buffers, 5466 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5467 } else { 5468 ALOGW("Error creating output track on thread %p", playbackThread); 5469 } 5470} 5471 5472AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5473{ 5474 clearBufferQueue(); 5475} 5476 5477status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5478 int triggerSession) 5479{ 5480 status_t status = Track::start(event, triggerSession); 5481 if (status != NO_ERROR) { 5482 return status; 5483 } 5484 5485 mActive = true; 5486 mRetryCount = 127; 5487 return status; 5488} 5489 5490void AudioFlinger::PlaybackThread::OutputTrack::stop() 5491{ 5492 Track::stop(); 5493 clearBufferQueue(); 5494 mOutBuffer.frameCount = 0; 5495 mActive = false; 5496} 5497 5498bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5499{ 5500 Buffer *pInBuffer; 5501 Buffer inBuffer; 5502 uint32_t channelCount = mChannelCount; 5503 bool outputBufferFull = false; 5504 inBuffer.frameCount = frames; 5505 inBuffer.i16 = data; 5506 5507 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5508 5509 if (!mActive && frames != 0) { 5510 start(); 5511 sp<ThreadBase> thread = mThread.promote(); 5512 if (thread != 0) { 5513 MixerThread *mixerThread = (MixerThread *)thread.get(); 5514 if (mCblk->frameCount > frames){ 5515 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5516 uint32_t startFrames = (mCblk->frameCount - frames); 5517 pInBuffer = new Buffer; 5518 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5519 pInBuffer->frameCount = startFrames; 5520 pInBuffer->i16 = pInBuffer->mBuffer; 5521 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5522 mBufferQueue.add(pInBuffer); 5523 } else { 5524 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5525 } 5526 } 5527 } 5528 } 5529 5530 while (waitTimeLeftMs) { 5531 // First write pending buffers, then new data 5532 if (mBufferQueue.size()) { 5533 pInBuffer = mBufferQueue.itemAt(0); 5534 } else { 5535 pInBuffer = &inBuffer; 5536 } 5537 5538 if (pInBuffer->frameCount == 0) { 5539 break; 5540 } 5541 5542 if (mOutBuffer.frameCount == 0) { 5543 mOutBuffer.frameCount = pInBuffer->frameCount; 5544 nsecs_t startTime = systemTime(); 5545 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5546 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5547 outputBufferFull = true; 5548 break; 5549 } 5550 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5551 if (waitTimeLeftMs >= waitTimeMs) { 5552 waitTimeLeftMs -= waitTimeMs; 5553 } else { 5554 waitTimeLeftMs = 0; 5555 } 5556 } 5557 5558 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5559 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5560 mCblk->stepUser(outFrames); 5561 pInBuffer->frameCount -= outFrames; 5562 pInBuffer->i16 += outFrames * channelCount; 5563 mOutBuffer.frameCount -= outFrames; 5564 mOutBuffer.i16 += outFrames * channelCount; 5565 5566 if (pInBuffer->frameCount == 0) { 5567 if (mBufferQueue.size()) { 5568 mBufferQueue.removeAt(0); 5569 delete [] pInBuffer->mBuffer; 5570 delete pInBuffer; 5571 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5572 } else { 5573 break; 5574 } 5575 } 5576 } 5577 5578 // If we could not write all frames, allocate a buffer and queue it for next time. 5579 if (inBuffer.frameCount) { 5580 sp<ThreadBase> thread = mThread.promote(); 5581 if (thread != 0 && !thread->standby()) { 5582 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5583 pInBuffer = new Buffer; 5584 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5585 pInBuffer->frameCount = inBuffer.frameCount; 5586 pInBuffer->i16 = pInBuffer->mBuffer; 5587 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5588 mBufferQueue.add(pInBuffer); 5589 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5590 } else { 5591 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5592 } 5593 } 5594 } 5595 5596 // Calling write() with a 0 length buffer, means that no more data will be written: 5597 // If no more buffers are pending, fill output track buffer to make sure it is started 5598 // by output mixer. 5599 if (frames == 0 && mBufferQueue.size() == 0) { 5600 if (mCblk->user < mCblk->frameCount) { 5601 frames = mCblk->frameCount - mCblk->user; 5602 pInBuffer = new Buffer; 5603 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5604 pInBuffer->frameCount = frames; 5605 pInBuffer->i16 = pInBuffer->mBuffer; 5606 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5607 mBufferQueue.add(pInBuffer); 5608 } else if (mActive) { 5609 stop(); 5610 } 5611 } 5612 5613 return outputBufferFull; 5614} 5615 5616status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5617{ 5618 int active; 5619 status_t result; 5620 audio_track_cblk_t* cblk = mCblk; 5621 uint32_t framesReq = buffer->frameCount; 5622 5623// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5624 buffer->frameCount = 0; 5625 5626 uint32_t framesAvail = cblk->framesAvailable(); 5627 5628 5629 if (framesAvail == 0) { 5630 Mutex::Autolock _l(cblk->lock); 5631 goto start_loop_here; 5632 while (framesAvail == 0) { 5633 active = mActive; 5634 if (CC_UNLIKELY(!active)) { 5635 ALOGV("Not active and NO_MORE_BUFFERS"); 5636 return NO_MORE_BUFFERS; 5637 } 5638 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5639 if (result != NO_ERROR) { 5640 return NO_MORE_BUFFERS; 5641 } 5642 // read the server count again 5643 start_loop_here: 5644 framesAvail = cblk->framesAvailable_l(); 5645 } 5646 } 5647 5648// if (framesAvail < framesReq) { 5649// return NO_MORE_BUFFERS; 5650// } 5651 5652 if (framesReq > framesAvail) { 5653 framesReq = framesAvail; 5654 } 5655 5656 uint32_t u = cblk->user; 5657 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5658 5659 if (framesReq > bufferEnd - u) { 5660 framesReq = bufferEnd - u; 5661 } 5662 5663 buffer->frameCount = framesReq; 5664 buffer->raw = (void *)cblk->buffer(u); 5665 return NO_ERROR; 5666} 5667 5668 5669void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5670{ 5671 size_t size = mBufferQueue.size(); 5672 5673 for (size_t i = 0; i < size; i++) { 5674 Buffer *pBuffer = mBufferQueue.itemAt(i); 5675 delete [] pBuffer->mBuffer; 5676 delete pBuffer; 5677 } 5678 mBufferQueue.clear(); 5679} 5680 5681// ---------------------------------------------------------------------------- 5682 5683AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5684 : RefBase(), 5685 mAudioFlinger(audioFlinger), 5686 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5687 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5688 mPid(pid), 5689 mTimedTrackCount(0) 5690{ 5691 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5692} 5693 5694// Client destructor must be called with AudioFlinger::mLock held 5695AudioFlinger::Client::~Client() 5696{ 5697 mAudioFlinger->removeClient_l(mPid); 5698} 5699 5700sp<MemoryDealer> AudioFlinger::Client::heap() const 5701{ 5702 return mMemoryDealer; 5703} 5704 5705// Reserve one of the limited slots for a timed audio track associated 5706// with this client 5707bool AudioFlinger::Client::reserveTimedTrack() 5708{ 5709 const int kMaxTimedTracksPerClient = 4; 5710 5711 Mutex::Autolock _l(mTimedTrackLock); 5712 5713 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5714 ALOGW("can not create timed track - pid %d has exceeded the limit", 5715 mPid); 5716 return false; 5717 } 5718 5719 mTimedTrackCount++; 5720 return true; 5721} 5722 5723// Release a slot for a timed audio track 5724void AudioFlinger::Client::releaseTimedTrack() 5725{ 5726 Mutex::Autolock _l(mTimedTrackLock); 5727 mTimedTrackCount--; 5728} 5729 5730// ---------------------------------------------------------------------------- 5731 5732AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5733 const sp<IAudioFlingerClient>& client, 5734 pid_t pid) 5735 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5736{ 5737} 5738 5739AudioFlinger::NotificationClient::~NotificationClient() 5740{ 5741} 5742 5743void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5744{ 5745 sp<NotificationClient> keep(this); 5746 mAudioFlinger->removeNotificationClient(mPid); 5747} 5748 5749// ---------------------------------------------------------------------------- 5750 5751AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5752 : BnAudioTrack(), 5753 mTrack(track) 5754{ 5755} 5756 5757AudioFlinger::TrackHandle::~TrackHandle() { 5758 // just stop the track on deletion, associated resources 5759 // will be freed from the main thread once all pending buffers have 5760 // been played. Unless it's not in the active track list, in which 5761 // case we free everything now... 5762 mTrack->destroy(); 5763} 5764 5765sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5766 return mTrack->getCblk(); 5767} 5768 5769status_t AudioFlinger::TrackHandle::start() { 5770 return mTrack->start(); 5771} 5772 5773void AudioFlinger::TrackHandle::stop() { 5774 mTrack->stop(); 5775} 5776 5777void AudioFlinger::TrackHandle::flush() { 5778 mTrack->flush(); 5779} 5780 5781void AudioFlinger::TrackHandle::mute(bool e) { 5782 mTrack->mute(e); 5783} 5784 5785void AudioFlinger::TrackHandle::pause() { 5786 mTrack->pause(); 5787} 5788 5789status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5790{ 5791 return mTrack->attachAuxEffect(EffectId); 5792} 5793 5794status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5795 sp<IMemory>* buffer) { 5796 if (!mTrack->isTimedTrack()) 5797 return INVALID_OPERATION; 5798 5799 PlaybackThread::TimedTrack* tt = 5800 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5801 return tt->allocateTimedBuffer(size, buffer); 5802} 5803 5804status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5805 int64_t pts) { 5806 if (!mTrack->isTimedTrack()) 5807 return INVALID_OPERATION; 5808 5809 PlaybackThread::TimedTrack* tt = 5810 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5811 return tt->queueTimedBuffer(buffer, pts); 5812} 5813 5814status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5815 const LinearTransform& xform, int target) { 5816 5817 if (!mTrack->isTimedTrack()) 5818 return INVALID_OPERATION; 5819 5820 PlaybackThread::TimedTrack* tt = 5821 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5822 return tt->setMediaTimeTransform( 5823 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5824} 5825 5826status_t AudioFlinger::TrackHandle::onTransact( 5827 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5828{ 5829 return BnAudioTrack::onTransact(code, data, reply, flags); 5830} 5831 5832// ---------------------------------------------------------------------------- 5833 5834sp<IAudioRecord> AudioFlinger::openRecord( 5835 pid_t pid, 5836 audio_io_handle_t input, 5837 uint32_t sampleRate, 5838 audio_format_t format, 5839 uint32_t channelMask, 5840 int frameCount, 5841 IAudioFlinger::track_flags_t flags, 5842 int *sessionId, 5843 status_t *status) 5844{ 5845 sp<RecordThread::RecordTrack> recordTrack; 5846 sp<RecordHandle> recordHandle; 5847 sp<Client> client; 5848 status_t lStatus; 5849 RecordThread *thread; 5850 size_t inFrameCount; 5851 int lSessionId; 5852 5853 // check calling permissions 5854 if (!recordingAllowed()) { 5855 lStatus = PERMISSION_DENIED; 5856 goto Exit; 5857 } 5858 5859 // add client to list 5860 { // scope for mLock 5861 Mutex::Autolock _l(mLock); 5862 thread = checkRecordThread_l(input); 5863 if (thread == NULL) { 5864 lStatus = BAD_VALUE; 5865 goto Exit; 5866 } 5867 5868 client = registerPid_l(pid); 5869 5870 // If no audio session id is provided, create one here 5871 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5872 lSessionId = *sessionId; 5873 } else { 5874 lSessionId = nextUniqueId(); 5875 if (sessionId != NULL) { 5876 *sessionId = lSessionId; 5877 } 5878 } 5879 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5880 recordTrack = thread->createRecordTrack_l(client, 5881 sampleRate, 5882 format, 5883 channelMask, 5884 frameCount, 5885 lSessionId, 5886 &lStatus); 5887 } 5888 if (lStatus != NO_ERROR) { 5889 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5890 // destructor is called by the TrackBase destructor with mLock held 5891 client.clear(); 5892 recordTrack.clear(); 5893 goto Exit; 5894 } 5895 5896 // return to handle to client 5897 recordHandle = new RecordHandle(recordTrack); 5898 lStatus = NO_ERROR; 5899 5900Exit: 5901 if (status) { 5902 *status = lStatus; 5903 } 5904 return recordHandle; 5905} 5906 5907// ---------------------------------------------------------------------------- 5908 5909AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5910 : BnAudioRecord(), 5911 mRecordTrack(recordTrack) 5912{ 5913} 5914 5915AudioFlinger::RecordHandle::~RecordHandle() { 5916 stop(); 5917} 5918 5919sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5920 return mRecordTrack->getCblk(); 5921} 5922 5923status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5924 ALOGV("RecordHandle::start()"); 5925 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5926} 5927 5928void AudioFlinger::RecordHandle::stop() { 5929 ALOGV("RecordHandle::stop()"); 5930 mRecordTrack->stop(); 5931} 5932 5933status_t AudioFlinger::RecordHandle::onTransact( 5934 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5935{ 5936 return BnAudioRecord::onTransact(code, data, reply, flags); 5937} 5938 5939// ---------------------------------------------------------------------------- 5940 5941AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5942 AudioStreamIn *input, 5943 uint32_t sampleRate, 5944 uint32_t channels, 5945 audio_io_handle_t id, 5946 uint32_t device) : 5947 ThreadBase(audioFlinger, id, device, RECORD), 5948 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5949 // mRsmpInIndex and mInputBytes set by readInputParameters() 5950 mReqChannelCount(popcount(channels)), 5951 mReqSampleRate(sampleRate) 5952 // mBytesRead is only meaningful while active, and so is cleared in start() 5953 // (but might be better to also clear here for dump?) 5954{ 5955 snprintf(mName, kNameLength, "AudioIn_%X", id); 5956 5957 readInputParameters(); 5958} 5959 5960 5961AudioFlinger::RecordThread::~RecordThread() 5962{ 5963 delete[] mRsmpInBuffer; 5964 delete mResampler; 5965 delete[] mRsmpOutBuffer; 5966} 5967 5968void AudioFlinger::RecordThread::onFirstRef() 5969{ 5970 run(mName, PRIORITY_URGENT_AUDIO); 5971} 5972 5973status_t AudioFlinger::RecordThread::readyToRun() 5974{ 5975 status_t status = initCheck(); 5976 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5977 return status; 5978} 5979 5980bool AudioFlinger::RecordThread::threadLoop() 5981{ 5982 AudioBufferProvider::Buffer buffer; 5983 sp<RecordTrack> activeTrack; 5984 Vector< sp<EffectChain> > effectChains; 5985 5986 nsecs_t lastWarning = 0; 5987 5988 acquireWakeLock(); 5989 5990 // start recording 5991 while (!exitPending()) { 5992 5993 processConfigEvents(); 5994 5995 { // scope for mLock 5996 Mutex::Autolock _l(mLock); 5997 checkForNewParameters_l(); 5998 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5999 if (!mStandby) { 6000 mInput->stream->common.standby(&mInput->stream->common); 6001 mStandby = true; 6002 } 6003 6004 if (exitPending()) break; 6005 6006 releaseWakeLock_l(); 6007 ALOGV("RecordThread: loop stopping"); 6008 // go to sleep 6009 mWaitWorkCV.wait(mLock); 6010 ALOGV("RecordThread: loop starting"); 6011 acquireWakeLock_l(); 6012 continue; 6013 } 6014 if (mActiveTrack != 0) { 6015 if (mActiveTrack->mState == TrackBase::PAUSING) { 6016 if (!mStandby) { 6017 mInput->stream->common.standby(&mInput->stream->common); 6018 mStandby = true; 6019 } 6020 mActiveTrack.clear(); 6021 mStartStopCond.broadcast(); 6022 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6023 if (mReqChannelCount != mActiveTrack->channelCount()) { 6024 mActiveTrack.clear(); 6025 mStartStopCond.broadcast(); 6026 } else if (mBytesRead != 0) { 6027 // record start succeeds only if first read from audio input 6028 // succeeds 6029 if (mBytesRead > 0) { 6030 mActiveTrack->mState = TrackBase::ACTIVE; 6031 } else { 6032 mActiveTrack.clear(); 6033 } 6034 mStartStopCond.broadcast(); 6035 } 6036 mStandby = false; 6037 } 6038 } 6039 lockEffectChains_l(effectChains); 6040 } 6041 6042 if (mActiveTrack != 0) { 6043 if (mActiveTrack->mState != TrackBase::ACTIVE && 6044 mActiveTrack->mState != TrackBase::RESUMING) { 6045 unlockEffectChains(effectChains); 6046 usleep(kRecordThreadSleepUs); 6047 continue; 6048 } 6049 for (size_t i = 0; i < effectChains.size(); i ++) { 6050 effectChains[i]->process_l(); 6051 } 6052 6053 buffer.frameCount = mFrameCount; 6054 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6055 size_t framesOut = buffer.frameCount; 6056 if (mResampler == NULL) { 6057 // no resampling 6058 while (framesOut) { 6059 size_t framesIn = mFrameCount - mRsmpInIndex; 6060 if (framesIn) { 6061 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6062 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6063 if (framesIn > framesOut) 6064 framesIn = framesOut; 6065 mRsmpInIndex += framesIn; 6066 framesOut -= framesIn; 6067 if ((int)mChannelCount == mReqChannelCount || 6068 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6069 memcpy(dst, src, framesIn * mFrameSize); 6070 } else { 6071 int16_t *src16 = (int16_t *)src; 6072 int16_t *dst16 = (int16_t *)dst; 6073 if (mChannelCount == 1) { 6074 while (framesIn--) { 6075 *dst16++ = *src16; 6076 *dst16++ = *src16++; 6077 } 6078 } else { 6079 while (framesIn--) { 6080 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6081 src16 += 2; 6082 } 6083 } 6084 } 6085 } 6086 if (framesOut && mFrameCount == mRsmpInIndex) { 6087 if (framesOut == mFrameCount && 6088 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6089 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6090 framesOut = 0; 6091 } else { 6092 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6093 mRsmpInIndex = 0; 6094 } 6095 if (mBytesRead < 0) { 6096 ALOGE("Error reading audio input"); 6097 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6098 // Force input into standby so that it tries to 6099 // recover at next read attempt 6100 mInput->stream->common.standby(&mInput->stream->common); 6101 usleep(kRecordThreadSleepUs); 6102 } 6103 mRsmpInIndex = mFrameCount; 6104 framesOut = 0; 6105 buffer.frameCount = 0; 6106 } 6107 } 6108 } 6109 } else { 6110 // resampling 6111 6112 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6113 // alter output frame count as if we were expecting stereo samples 6114 if (mChannelCount == 1 && mReqChannelCount == 1) { 6115 framesOut >>= 1; 6116 } 6117 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6118 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6119 // are 32 bit aligned which should be always true. 6120 if (mChannelCount == 2 && mReqChannelCount == 1) { 6121 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6122 // the resampler always outputs stereo samples: do post stereo to mono conversion 6123 int16_t *src = (int16_t *)mRsmpOutBuffer; 6124 int16_t *dst = buffer.i16; 6125 while (framesOut--) { 6126 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6127 src += 2; 6128 } 6129 } else { 6130 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6131 } 6132 6133 } 6134 if (mFramestoDrop == 0) { 6135 mActiveTrack->releaseBuffer(&buffer); 6136 } else { 6137 if (mFramestoDrop > 0) { 6138 mFramestoDrop -= buffer.frameCount; 6139 if (mFramestoDrop <= 0) { 6140 clearSyncStartEvent(); 6141 } 6142 } else { 6143 mFramestoDrop += buffer.frameCount; 6144 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6145 mSyncStartEvent->isCancelled()) { 6146 ALOGW("Synced record %s, session %d, trigger session %d", 6147 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6148 mActiveTrack->sessionId(), 6149 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6150 clearSyncStartEvent(); 6151 } 6152 } 6153 } 6154 mActiveTrack->overflow(); 6155 } 6156 // client isn't retrieving buffers fast enough 6157 else { 6158 if (!mActiveTrack->setOverflow()) { 6159 nsecs_t now = systemTime(); 6160 if ((now - lastWarning) > kWarningThrottleNs) { 6161 ALOGW("RecordThread: buffer overflow"); 6162 lastWarning = now; 6163 } 6164 } 6165 // Release the processor for a while before asking for a new buffer. 6166 // This will give the application more chance to read from the buffer and 6167 // clear the overflow. 6168 usleep(kRecordThreadSleepUs); 6169 } 6170 } 6171 // enable changes in effect chain 6172 unlockEffectChains(effectChains); 6173 effectChains.clear(); 6174 } 6175 6176 if (!mStandby) { 6177 mInput->stream->common.standby(&mInput->stream->common); 6178 } 6179 mActiveTrack.clear(); 6180 6181 mStartStopCond.broadcast(); 6182 6183 releaseWakeLock(); 6184 6185 ALOGV("RecordThread %p exiting", this); 6186 return false; 6187} 6188 6189 6190sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6191 const sp<AudioFlinger::Client>& client, 6192 uint32_t sampleRate, 6193 audio_format_t format, 6194 int channelMask, 6195 int frameCount, 6196 int sessionId, 6197 status_t *status) 6198{ 6199 sp<RecordTrack> track; 6200 status_t lStatus; 6201 6202 lStatus = initCheck(); 6203 if (lStatus != NO_ERROR) { 6204 ALOGE("Audio driver not initialized."); 6205 goto Exit; 6206 } 6207 6208 { // scope for mLock 6209 Mutex::Autolock _l(mLock); 6210 6211 track = new RecordTrack(this, client, sampleRate, 6212 format, channelMask, frameCount, sessionId); 6213 6214 if (track->getCblk() == 0) { 6215 lStatus = NO_MEMORY; 6216 goto Exit; 6217 } 6218 6219 mTrack = track.get(); 6220 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6221 bool suspend = audio_is_bluetooth_sco_device( 6222 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6223 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6224 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6225 } 6226 lStatus = NO_ERROR; 6227 6228Exit: 6229 if (status) { 6230 *status = lStatus; 6231 } 6232 return track; 6233} 6234 6235status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6236 AudioSystem::sync_event_t event, 6237 int triggerSession) 6238{ 6239 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6240 sp<ThreadBase> strongMe = this; 6241 status_t status = NO_ERROR; 6242 6243 if (event == AudioSystem::SYNC_EVENT_NONE) { 6244 clearSyncStartEvent(); 6245 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6246 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6247 triggerSession, 6248 recordTrack->sessionId(), 6249 syncStartEventCallback, 6250 this); 6251 // Sync event can be cancelled by the trigger session if the track is not in a 6252 // compatible state in which case we start record immediately 6253 if (mSyncStartEvent->isCancelled()) { 6254 clearSyncStartEvent(); 6255 } else { 6256 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6257 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6258 } 6259 } 6260 6261 { 6262 AutoMutex lock(mLock); 6263 if (mActiveTrack != 0) { 6264 if (recordTrack != mActiveTrack.get()) { 6265 status = -EBUSY; 6266 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6267 mActiveTrack->mState = TrackBase::ACTIVE; 6268 } 6269 return status; 6270 } 6271 6272 recordTrack->mState = TrackBase::IDLE; 6273 mActiveTrack = recordTrack; 6274 mLock.unlock(); 6275 status_t status = AudioSystem::startInput(mId); 6276 mLock.lock(); 6277 if (status != NO_ERROR) { 6278 mActiveTrack.clear(); 6279 clearSyncStartEvent(); 6280 return status; 6281 } 6282 mRsmpInIndex = mFrameCount; 6283 mBytesRead = 0; 6284 if (mResampler != NULL) { 6285 mResampler->reset(); 6286 } 6287 mActiveTrack->mState = TrackBase::RESUMING; 6288 // signal thread to start 6289 ALOGV("Signal record thread"); 6290 mWaitWorkCV.signal(); 6291 // do not wait for mStartStopCond if exiting 6292 if (exitPending()) { 6293 mActiveTrack.clear(); 6294 status = INVALID_OPERATION; 6295 goto startError; 6296 } 6297 mStartStopCond.wait(mLock); 6298 if (mActiveTrack == 0) { 6299 ALOGV("Record failed to start"); 6300 status = BAD_VALUE; 6301 goto startError; 6302 } 6303 ALOGV("Record started OK"); 6304 return status; 6305 } 6306startError: 6307 AudioSystem::stopInput(mId); 6308 clearSyncStartEvent(); 6309 return status; 6310} 6311 6312void AudioFlinger::RecordThread::clearSyncStartEvent() 6313{ 6314 if (mSyncStartEvent != 0) { 6315 mSyncStartEvent->cancel(); 6316 } 6317 mSyncStartEvent.clear(); 6318 mFramestoDrop = 0; 6319} 6320 6321void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6322{ 6323 sp<SyncEvent> strongEvent = event.promote(); 6324 6325 if (strongEvent != 0) { 6326 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6327 me->handleSyncStartEvent(strongEvent); 6328 } 6329} 6330 6331void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6332{ 6333 if (event == mSyncStartEvent) { 6334 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6335 // from audio HAL 6336 mFramestoDrop = mFrameCount * 2; 6337 } 6338} 6339 6340void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6341 ALOGV("RecordThread::stop"); 6342 sp<ThreadBase> strongMe = this; 6343 { 6344 AutoMutex lock(mLock); 6345 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6346 mActiveTrack->mState = TrackBase::PAUSING; 6347 // do not wait for mStartStopCond if exiting 6348 if (exitPending()) { 6349 return; 6350 } 6351 mStartStopCond.wait(mLock); 6352 // if we have been restarted, recordTrack == mActiveTrack.get() here 6353 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6354 mLock.unlock(); 6355 AudioSystem::stopInput(mId); 6356 mLock.lock(); 6357 ALOGV("Record stopped OK"); 6358 } 6359 } 6360 } 6361} 6362 6363bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6364{ 6365 return false; 6366} 6367 6368status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6369{ 6370 if (!isValidSyncEvent(event)) { 6371 return BAD_VALUE; 6372 } 6373 6374 Mutex::Autolock _l(mLock); 6375 6376 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6377 mTrack->setSyncEvent(event); 6378 return NO_ERROR; 6379 } 6380 return NAME_NOT_FOUND; 6381} 6382 6383status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6384{ 6385 const size_t SIZE = 256; 6386 char buffer[SIZE]; 6387 String8 result; 6388 6389 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6390 result.append(buffer); 6391 6392 if (mActiveTrack != 0) { 6393 result.append("Active Track:\n"); 6394 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6395 mActiveTrack->dump(buffer, SIZE); 6396 result.append(buffer); 6397 6398 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6399 result.append(buffer); 6400 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6401 result.append(buffer); 6402 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6403 result.append(buffer); 6404 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6405 result.append(buffer); 6406 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6407 result.append(buffer); 6408 6409 6410 } else { 6411 result.append("No record client\n"); 6412 } 6413 write(fd, result.string(), result.size()); 6414 6415 dumpBase(fd, args); 6416 dumpEffectChains(fd, args); 6417 6418 return NO_ERROR; 6419} 6420 6421// AudioBufferProvider interface 6422status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6423{ 6424 size_t framesReq = buffer->frameCount; 6425 size_t framesReady = mFrameCount - mRsmpInIndex; 6426 int channelCount; 6427 6428 if (framesReady == 0) { 6429 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6430 if (mBytesRead < 0) { 6431 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6432 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6433 // Force input into standby so that it tries to 6434 // recover at next read attempt 6435 mInput->stream->common.standby(&mInput->stream->common); 6436 usleep(kRecordThreadSleepUs); 6437 } 6438 buffer->raw = NULL; 6439 buffer->frameCount = 0; 6440 return NOT_ENOUGH_DATA; 6441 } 6442 mRsmpInIndex = 0; 6443 framesReady = mFrameCount; 6444 } 6445 6446 if (framesReq > framesReady) { 6447 framesReq = framesReady; 6448 } 6449 6450 if (mChannelCount == 1 && mReqChannelCount == 2) { 6451 channelCount = 1; 6452 } else { 6453 channelCount = 2; 6454 } 6455 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6456 buffer->frameCount = framesReq; 6457 return NO_ERROR; 6458} 6459 6460// AudioBufferProvider interface 6461void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6462{ 6463 mRsmpInIndex += buffer->frameCount; 6464 buffer->frameCount = 0; 6465} 6466 6467bool AudioFlinger::RecordThread::checkForNewParameters_l() 6468{ 6469 bool reconfig = false; 6470 6471 while (!mNewParameters.isEmpty()) { 6472 status_t status = NO_ERROR; 6473 String8 keyValuePair = mNewParameters[0]; 6474 AudioParameter param = AudioParameter(keyValuePair); 6475 int value; 6476 audio_format_t reqFormat = mFormat; 6477 int reqSamplingRate = mReqSampleRate; 6478 int reqChannelCount = mReqChannelCount; 6479 6480 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6481 reqSamplingRate = value; 6482 reconfig = true; 6483 } 6484 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6485 reqFormat = (audio_format_t) value; 6486 reconfig = true; 6487 } 6488 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6489 reqChannelCount = popcount(value); 6490 reconfig = true; 6491 } 6492 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6493 // do not accept frame count changes if tracks are open as the track buffer 6494 // size depends on frame count and correct behavior would not be guaranteed 6495 // if frame count is changed after track creation 6496 if (mActiveTrack != 0) { 6497 status = INVALID_OPERATION; 6498 } else { 6499 reconfig = true; 6500 } 6501 } 6502 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6503 // forward device change to effects that have requested to be 6504 // aware of attached audio device. 6505 for (size_t i = 0; i < mEffectChains.size(); i++) { 6506 mEffectChains[i]->setDevice_l(value); 6507 } 6508 // store input device and output device but do not forward output device to audio HAL. 6509 // Note that status is ignored by the caller for output device 6510 // (see AudioFlinger::setParameters() 6511 if (value & AUDIO_DEVICE_OUT_ALL) { 6512 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6513 status = BAD_VALUE; 6514 } else { 6515 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6516 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6517 if (mTrack != NULL) { 6518 bool suspend = audio_is_bluetooth_sco_device( 6519 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6520 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6521 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6522 } 6523 } 6524 mDevice |= (uint32_t)value; 6525 } 6526 if (status == NO_ERROR) { 6527 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6528 if (status == INVALID_OPERATION) { 6529 mInput->stream->common.standby(&mInput->stream->common); 6530 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6531 keyValuePair.string()); 6532 } 6533 if (reconfig) { 6534 if (status == BAD_VALUE && 6535 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6536 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6537 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6538 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6539 (reqChannelCount <= FCC_2)) { 6540 status = NO_ERROR; 6541 } 6542 if (status == NO_ERROR) { 6543 readInputParameters(); 6544 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6545 } 6546 } 6547 } 6548 6549 mNewParameters.removeAt(0); 6550 6551 mParamStatus = status; 6552 mParamCond.signal(); 6553 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6554 // already timed out waiting for the status and will never signal the condition. 6555 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6556 } 6557 return reconfig; 6558} 6559 6560String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6561{ 6562 char *s; 6563 String8 out_s8 = String8(); 6564 6565 Mutex::Autolock _l(mLock); 6566 if (initCheck() != NO_ERROR) { 6567 return out_s8; 6568 } 6569 6570 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6571 out_s8 = String8(s); 6572 free(s); 6573 return out_s8; 6574} 6575 6576void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6577 AudioSystem::OutputDescriptor desc; 6578 void *param2 = NULL; 6579 6580 switch (event) { 6581 case AudioSystem::INPUT_OPENED: 6582 case AudioSystem::INPUT_CONFIG_CHANGED: 6583 desc.channels = mChannelMask; 6584 desc.samplingRate = mSampleRate; 6585 desc.format = mFormat; 6586 desc.frameCount = mFrameCount; 6587 desc.latency = 0; 6588 param2 = &desc; 6589 break; 6590 6591 case AudioSystem::INPUT_CLOSED: 6592 default: 6593 break; 6594 } 6595 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6596} 6597 6598void AudioFlinger::RecordThread::readInputParameters() 6599{ 6600 delete mRsmpInBuffer; 6601 // mRsmpInBuffer is always assigned a new[] below 6602 delete mRsmpOutBuffer; 6603 mRsmpOutBuffer = NULL; 6604 delete mResampler; 6605 mResampler = NULL; 6606 6607 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6608 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6609 mChannelCount = (uint16_t)popcount(mChannelMask); 6610 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6611 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6612 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6613 mFrameCount = mInputBytes / mFrameSize; 6614 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6615 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6616 6617 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6618 { 6619 int channelCount; 6620 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6621 // stereo to mono post process as the resampler always outputs stereo. 6622 if (mChannelCount == 1 && mReqChannelCount == 2) { 6623 channelCount = 1; 6624 } else { 6625 channelCount = 2; 6626 } 6627 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6628 mResampler->setSampleRate(mSampleRate); 6629 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6630 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6631 6632 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6633 if (mChannelCount == 1 && mReqChannelCount == 1) { 6634 mFrameCount >>= 1; 6635 } 6636 6637 } 6638 mRsmpInIndex = mFrameCount; 6639} 6640 6641unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6642{ 6643 Mutex::Autolock _l(mLock); 6644 if (initCheck() != NO_ERROR) { 6645 return 0; 6646 } 6647 6648 return mInput->stream->get_input_frames_lost(mInput->stream); 6649} 6650 6651uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6652{ 6653 Mutex::Autolock _l(mLock); 6654 uint32_t result = 0; 6655 if (getEffectChain_l(sessionId) != 0) { 6656 result = EFFECT_SESSION; 6657 } 6658 6659 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6660 result |= TRACK_SESSION; 6661 } 6662 6663 return result; 6664} 6665 6666AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6667{ 6668 Mutex::Autolock _l(mLock); 6669 return mTrack; 6670} 6671 6672AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6673{ 6674 Mutex::Autolock _l(mLock); 6675 return mInput; 6676} 6677 6678AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6679{ 6680 Mutex::Autolock _l(mLock); 6681 AudioStreamIn *input = mInput; 6682 mInput = NULL; 6683 return input; 6684} 6685 6686// this method must always be called either with ThreadBase mLock held or inside the thread loop 6687audio_stream_t* AudioFlinger::RecordThread::stream() const 6688{ 6689 if (mInput == NULL) { 6690 return NULL; 6691 } 6692 return &mInput->stream->common; 6693} 6694 6695 6696// ---------------------------------------------------------------------------- 6697 6698audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6699{ 6700 if (!settingsAllowed()) { 6701 return 0; 6702 } 6703 Mutex::Autolock _l(mLock); 6704 return loadHwModule_l(name); 6705} 6706 6707// loadHwModule_l() must be called with AudioFlinger::mLock held 6708audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6709{ 6710 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6711 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6712 ALOGW("loadHwModule() module %s already loaded", name); 6713 return mAudioHwDevs.keyAt(i); 6714 } 6715 } 6716 6717 audio_hw_device_t *dev; 6718 6719 int rc = load_audio_interface(name, &dev); 6720 if (rc) { 6721 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6722 return 0; 6723 } 6724 6725 mHardwareStatus = AUDIO_HW_INIT; 6726 rc = dev->init_check(dev); 6727 mHardwareStatus = AUDIO_HW_IDLE; 6728 if (rc) { 6729 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6730 return 0; 6731 } 6732 6733 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6734 (NULL != dev->set_master_volume)) { 6735 AutoMutex lock(mHardwareLock); 6736 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6737 dev->set_master_volume(dev, mMasterVolume); 6738 mHardwareStatus = AUDIO_HW_IDLE; 6739 } 6740 6741 audio_module_handle_t handle = nextUniqueId(); 6742 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6743 6744 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6745 name, dev->common.module->name, dev->common.module->id, handle); 6746 6747 return handle; 6748 6749} 6750 6751audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6752 audio_devices_t *pDevices, 6753 uint32_t *pSamplingRate, 6754 audio_format_t *pFormat, 6755 audio_channel_mask_t *pChannelMask, 6756 uint32_t *pLatencyMs, 6757 audio_output_flags_t flags) 6758{ 6759 status_t status; 6760 PlaybackThread *thread = NULL; 6761 struct audio_config config = { 6762 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6763 channel_mask: pChannelMask ? *pChannelMask : 0, 6764 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6765 }; 6766 audio_stream_out_t *outStream = NULL; 6767 audio_hw_device_t *outHwDev; 6768 6769 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6770 module, 6771 (pDevices != NULL) ? (int)*pDevices : 0, 6772 config.sample_rate, 6773 config.format, 6774 config.channel_mask, 6775 flags); 6776 6777 if (pDevices == NULL || *pDevices == 0) { 6778 return 0; 6779 } 6780 6781 Mutex::Autolock _l(mLock); 6782 6783 outHwDev = findSuitableHwDev_l(module, *pDevices); 6784 if (outHwDev == NULL) 6785 return 0; 6786 6787 audio_io_handle_t id = nextUniqueId(); 6788 6789 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6790 6791 status = outHwDev->open_output_stream(outHwDev, 6792 id, 6793 *pDevices, 6794 (audio_output_flags_t)flags, 6795 &config, 6796 &outStream); 6797 6798 mHardwareStatus = AUDIO_HW_IDLE; 6799 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6800 outStream, 6801 config.sample_rate, 6802 config.format, 6803 config.channel_mask, 6804 status); 6805 6806 if (status == NO_ERROR && outStream != NULL) { 6807 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6808 6809 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6810 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6811 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6812 thread = new DirectOutputThread(this, output, id, *pDevices); 6813 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6814 } else { 6815 thread = new MixerThread(this, output, id, *pDevices); 6816 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6817 } 6818 mPlaybackThreads.add(id, thread); 6819 6820 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6821 if (pFormat != NULL) *pFormat = config.format; 6822 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6823 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6824 6825 // notify client processes of the new output creation 6826 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6827 6828 // the first primary output opened designates the primary hw device 6829 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6830 ALOGI("Using module %d has the primary audio interface", module); 6831 mPrimaryHardwareDev = outHwDev; 6832 6833 AutoMutex lock(mHardwareLock); 6834 mHardwareStatus = AUDIO_HW_SET_MODE; 6835 outHwDev->set_mode(outHwDev, mMode); 6836 6837 // Determine the level of master volume support the primary audio HAL has, 6838 // and set the initial master volume at the same time. 6839 float initialVolume = 1.0; 6840 mMasterVolumeSupportLvl = MVS_NONE; 6841 6842 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6843 if ((NULL != outHwDev->get_master_volume) && 6844 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6845 mMasterVolumeSupportLvl = MVS_FULL; 6846 } else { 6847 mMasterVolumeSupportLvl = MVS_SETONLY; 6848 initialVolume = 1.0; 6849 } 6850 6851 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6852 if ((NULL == outHwDev->set_master_volume) || 6853 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6854 mMasterVolumeSupportLvl = MVS_NONE; 6855 } 6856 // now that we have a primary device, initialize master volume on other devices 6857 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6858 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6859 6860 if ((dev != mPrimaryHardwareDev) && 6861 (NULL != dev->set_master_volume)) { 6862 dev->set_master_volume(dev, initialVolume); 6863 } 6864 } 6865 mHardwareStatus = AUDIO_HW_IDLE; 6866 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6867 ? initialVolume 6868 : 1.0; 6869 mMasterVolume = initialVolume; 6870 } 6871 return id; 6872 } 6873 6874 return 0; 6875} 6876 6877audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6878 audio_io_handle_t output2) 6879{ 6880 Mutex::Autolock _l(mLock); 6881 MixerThread *thread1 = checkMixerThread_l(output1); 6882 MixerThread *thread2 = checkMixerThread_l(output2); 6883 6884 if (thread1 == NULL || thread2 == NULL) { 6885 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6886 return 0; 6887 } 6888 6889 audio_io_handle_t id = nextUniqueId(); 6890 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6891 thread->addOutputTrack(thread2); 6892 mPlaybackThreads.add(id, thread); 6893 // notify client processes of the new output creation 6894 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6895 return id; 6896} 6897 6898status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6899{ 6900 // keep strong reference on the playback thread so that 6901 // it is not destroyed while exit() is executed 6902 sp<PlaybackThread> thread; 6903 { 6904 Mutex::Autolock _l(mLock); 6905 thread = checkPlaybackThread_l(output); 6906 if (thread == NULL) { 6907 return BAD_VALUE; 6908 } 6909 6910 ALOGV("closeOutput() %d", output); 6911 6912 if (thread->type() == ThreadBase::MIXER) { 6913 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6914 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6915 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6916 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6917 } 6918 } 6919 } 6920 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6921 mPlaybackThreads.removeItem(output); 6922 } 6923 thread->exit(); 6924 // The thread entity (active unit of execution) is no longer running here, 6925 // but the ThreadBase container still exists. 6926 6927 if (thread->type() != ThreadBase::DUPLICATING) { 6928 AudioStreamOut *out = thread->clearOutput(); 6929 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6930 // from now on thread->mOutput is NULL 6931 out->hwDev->close_output_stream(out->hwDev, out->stream); 6932 delete out; 6933 } 6934 return NO_ERROR; 6935} 6936 6937status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6938{ 6939 Mutex::Autolock _l(mLock); 6940 PlaybackThread *thread = checkPlaybackThread_l(output); 6941 6942 if (thread == NULL) { 6943 return BAD_VALUE; 6944 } 6945 6946 ALOGV("suspendOutput() %d", output); 6947 thread->suspend(); 6948 6949 return NO_ERROR; 6950} 6951 6952status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6953{ 6954 Mutex::Autolock _l(mLock); 6955 PlaybackThread *thread = checkPlaybackThread_l(output); 6956 6957 if (thread == NULL) { 6958 return BAD_VALUE; 6959 } 6960 6961 ALOGV("restoreOutput() %d", output); 6962 6963 thread->restore(); 6964 6965 return NO_ERROR; 6966} 6967 6968audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6969 audio_devices_t *pDevices, 6970 uint32_t *pSamplingRate, 6971 audio_format_t *pFormat, 6972 uint32_t *pChannelMask) 6973{ 6974 status_t status; 6975 RecordThread *thread = NULL; 6976 struct audio_config config = { 6977 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6978 channel_mask: pChannelMask ? *pChannelMask : 0, 6979 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6980 }; 6981 uint32_t reqSamplingRate = config.sample_rate; 6982 audio_format_t reqFormat = config.format; 6983 audio_channel_mask_t reqChannels = config.channel_mask; 6984 audio_stream_in_t *inStream = NULL; 6985 audio_hw_device_t *inHwDev; 6986 6987 if (pDevices == NULL || *pDevices == 0) { 6988 return 0; 6989 } 6990 6991 Mutex::Autolock _l(mLock); 6992 6993 inHwDev = findSuitableHwDev_l(module, *pDevices); 6994 if (inHwDev == NULL) 6995 return 0; 6996 6997 audio_io_handle_t id = nextUniqueId(); 6998 6999 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 7000 &inStream); 7001 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7002 inStream, 7003 config.sample_rate, 7004 config.format, 7005 config.channel_mask, 7006 status); 7007 7008 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7009 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7010 // or stereo to mono conversions on 16 bit PCM inputs. 7011 if (status == BAD_VALUE && 7012 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7013 (config.sample_rate <= 2 * reqSamplingRate) && 7014 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7015 ALOGV("openInput() reopening with proposed sampling rate and channels"); 7016 inStream = NULL; 7017 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 7018 } 7019 7020 if (status == NO_ERROR && inStream != NULL) { 7021 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7022 7023 // Start record thread 7024 // RecorThread require both input and output device indication to forward to audio 7025 // pre processing modules 7026 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 7027 thread = new RecordThread(this, 7028 input, 7029 reqSamplingRate, 7030 reqChannels, 7031 id, 7032 device); 7033 mRecordThreads.add(id, thread); 7034 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7035 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7036 if (pFormat != NULL) *pFormat = config.format; 7037 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7038 7039 input->stream->common.standby(&input->stream->common); 7040 7041 // notify client processes of the new input creation 7042 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7043 return id; 7044 } 7045 7046 return 0; 7047} 7048 7049status_t AudioFlinger::closeInput(audio_io_handle_t input) 7050{ 7051 // keep strong reference on the record thread so that 7052 // it is not destroyed while exit() is executed 7053 sp<RecordThread> thread; 7054 { 7055 Mutex::Autolock _l(mLock); 7056 thread = checkRecordThread_l(input); 7057 if (thread == NULL) { 7058 return BAD_VALUE; 7059 } 7060 7061 ALOGV("closeInput() %d", input); 7062 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7063 mRecordThreads.removeItem(input); 7064 } 7065 thread->exit(); 7066 // The thread entity (active unit of execution) is no longer running here, 7067 // but the ThreadBase container still exists. 7068 7069 AudioStreamIn *in = thread->clearInput(); 7070 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7071 // from now on thread->mInput is NULL 7072 in->hwDev->close_input_stream(in->hwDev, in->stream); 7073 delete in; 7074 7075 return NO_ERROR; 7076} 7077 7078status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7079{ 7080 Mutex::Autolock _l(mLock); 7081 MixerThread *dstThread = checkMixerThread_l(output); 7082 if (dstThread == NULL) { 7083 ALOGW("setStreamOutput() bad output id %d", output); 7084 return BAD_VALUE; 7085 } 7086 7087 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7088 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7089 7090 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7091 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7092 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7093 MixerThread *srcThread = (MixerThread *)thread; 7094 srcThread->invalidateTracks(stream); 7095 } 7096 } 7097 7098 return NO_ERROR; 7099} 7100 7101 7102int AudioFlinger::newAudioSessionId() 7103{ 7104 return nextUniqueId(); 7105} 7106 7107void AudioFlinger::acquireAudioSessionId(int audioSession) 7108{ 7109 Mutex::Autolock _l(mLock); 7110 pid_t caller = IPCThreadState::self()->getCallingPid(); 7111 ALOGV("acquiring %d from %d", audioSession, caller); 7112 size_t num = mAudioSessionRefs.size(); 7113 for (size_t i = 0; i< num; i++) { 7114 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7115 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7116 ref->mCnt++; 7117 ALOGV(" incremented refcount to %d", ref->mCnt); 7118 return; 7119 } 7120 } 7121 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7122 ALOGV(" added new entry for %d", audioSession); 7123} 7124 7125void AudioFlinger::releaseAudioSessionId(int audioSession) 7126{ 7127 Mutex::Autolock _l(mLock); 7128 pid_t caller = IPCThreadState::self()->getCallingPid(); 7129 ALOGV("releasing %d from %d", audioSession, caller); 7130 size_t num = mAudioSessionRefs.size(); 7131 for (size_t i = 0; i< num; i++) { 7132 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7133 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7134 ref->mCnt--; 7135 ALOGV(" decremented refcount to %d", ref->mCnt); 7136 if (ref->mCnt == 0) { 7137 mAudioSessionRefs.removeAt(i); 7138 delete ref; 7139 purgeStaleEffects_l(); 7140 } 7141 return; 7142 } 7143 } 7144 ALOGW("session id %d not found for pid %d", audioSession, caller); 7145} 7146 7147void AudioFlinger::purgeStaleEffects_l() { 7148 7149 ALOGV("purging stale effects"); 7150 7151 Vector< sp<EffectChain> > chains; 7152 7153 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7154 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7155 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7156 sp<EffectChain> ec = t->mEffectChains[j]; 7157 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7158 chains.push(ec); 7159 } 7160 } 7161 } 7162 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7163 sp<RecordThread> t = mRecordThreads.valueAt(i); 7164 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7165 sp<EffectChain> ec = t->mEffectChains[j]; 7166 chains.push(ec); 7167 } 7168 } 7169 7170 for (size_t i = 0; i < chains.size(); i++) { 7171 sp<EffectChain> ec = chains[i]; 7172 int sessionid = ec->sessionId(); 7173 sp<ThreadBase> t = ec->mThread.promote(); 7174 if (t == 0) { 7175 continue; 7176 } 7177 size_t numsessionrefs = mAudioSessionRefs.size(); 7178 bool found = false; 7179 for (size_t k = 0; k < numsessionrefs; k++) { 7180 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7181 if (ref->mSessionid == sessionid) { 7182 ALOGV(" session %d still exists for %d with %d refs", 7183 sessionid, ref->mPid, ref->mCnt); 7184 found = true; 7185 break; 7186 } 7187 } 7188 if (!found) { 7189 // remove all effects from the chain 7190 while (ec->mEffects.size()) { 7191 sp<EffectModule> effect = ec->mEffects[0]; 7192 effect->unPin(); 7193 Mutex::Autolock _l (t->mLock); 7194 t->removeEffect_l(effect); 7195 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7196 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7197 if (handle != 0) { 7198 handle->mEffect.clear(); 7199 if (handle->mHasControl && handle->mEnabled) { 7200 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7201 } 7202 } 7203 } 7204 AudioSystem::unregisterEffect(effect->id()); 7205 } 7206 } 7207 } 7208 return; 7209} 7210 7211// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7212AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7213{ 7214 return mPlaybackThreads.valueFor(output).get(); 7215} 7216 7217// checkMixerThread_l() must be called with AudioFlinger::mLock held 7218AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7219{ 7220 PlaybackThread *thread = checkPlaybackThread_l(output); 7221 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7222} 7223 7224// checkRecordThread_l() must be called with AudioFlinger::mLock held 7225AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7226{ 7227 return mRecordThreads.valueFor(input).get(); 7228} 7229 7230uint32_t AudioFlinger::nextUniqueId() 7231{ 7232 return android_atomic_inc(&mNextUniqueId); 7233} 7234 7235AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7236{ 7237 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7238 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7239 AudioStreamOut *output = thread->getOutput(); 7240 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7241 return thread; 7242 } 7243 } 7244 return NULL; 7245} 7246 7247uint32_t AudioFlinger::primaryOutputDevice_l() const 7248{ 7249 PlaybackThread *thread = primaryPlaybackThread_l(); 7250 7251 if (thread == NULL) { 7252 return 0; 7253 } 7254 7255 return thread->device(); 7256} 7257 7258sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7259 int triggerSession, 7260 int listenerSession, 7261 sync_event_callback_t callBack, 7262 void *cookie) 7263{ 7264 Mutex::Autolock _l(mLock); 7265 7266 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7267 status_t playStatus = NAME_NOT_FOUND; 7268 status_t recStatus = NAME_NOT_FOUND; 7269 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7270 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7271 if (playStatus == NO_ERROR) { 7272 return event; 7273 } 7274 } 7275 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7276 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7277 if (recStatus == NO_ERROR) { 7278 return event; 7279 } 7280 } 7281 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7282 mPendingSyncEvents.add(event); 7283 } else { 7284 ALOGV("createSyncEvent() invalid event %d", event->type()); 7285 event.clear(); 7286 } 7287 return event; 7288} 7289 7290// ---------------------------------------------------------------------------- 7291// Effect management 7292// ---------------------------------------------------------------------------- 7293 7294 7295status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7296{ 7297 Mutex::Autolock _l(mLock); 7298 return EffectQueryNumberEffects(numEffects); 7299} 7300 7301status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7302{ 7303 Mutex::Autolock _l(mLock); 7304 return EffectQueryEffect(index, descriptor); 7305} 7306 7307status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7308 effect_descriptor_t *descriptor) const 7309{ 7310 Mutex::Autolock _l(mLock); 7311 return EffectGetDescriptor(pUuid, descriptor); 7312} 7313 7314 7315sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7316 effect_descriptor_t *pDesc, 7317 const sp<IEffectClient>& effectClient, 7318 int32_t priority, 7319 audio_io_handle_t io, 7320 int sessionId, 7321 status_t *status, 7322 int *id, 7323 int *enabled) 7324{ 7325 status_t lStatus = NO_ERROR; 7326 sp<EffectHandle> handle; 7327 effect_descriptor_t desc; 7328 7329 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7330 pid, effectClient.get(), priority, sessionId, io); 7331 7332 if (pDesc == NULL) { 7333 lStatus = BAD_VALUE; 7334 goto Exit; 7335 } 7336 7337 // check audio settings permission for global effects 7338 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7339 lStatus = PERMISSION_DENIED; 7340 goto Exit; 7341 } 7342 7343 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7344 // that can only be created by audio policy manager (running in same process) 7345 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7346 lStatus = PERMISSION_DENIED; 7347 goto Exit; 7348 } 7349 7350 if (io == 0) { 7351 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7352 // output must be specified by AudioPolicyManager when using session 7353 // AUDIO_SESSION_OUTPUT_STAGE 7354 lStatus = BAD_VALUE; 7355 goto Exit; 7356 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7357 // if the output returned by getOutputForEffect() is removed before we lock the 7358 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7359 // and we will exit safely 7360 io = AudioSystem::getOutputForEffect(&desc); 7361 } 7362 } 7363 7364 { 7365 Mutex::Autolock _l(mLock); 7366 7367 7368 if (!EffectIsNullUuid(&pDesc->uuid)) { 7369 // if uuid is specified, request effect descriptor 7370 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7371 if (lStatus < 0) { 7372 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7373 goto Exit; 7374 } 7375 } else { 7376 // if uuid is not specified, look for an available implementation 7377 // of the required type in effect factory 7378 if (EffectIsNullUuid(&pDesc->type)) { 7379 ALOGW("createEffect() no effect type"); 7380 lStatus = BAD_VALUE; 7381 goto Exit; 7382 } 7383 uint32_t numEffects = 0; 7384 effect_descriptor_t d; 7385 d.flags = 0; // prevent compiler warning 7386 bool found = false; 7387 7388 lStatus = EffectQueryNumberEffects(&numEffects); 7389 if (lStatus < 0) { 7390 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7391 goto Exit; 7392 } 7393 for (uint32_t i = 0; i < numEffects; i++) { 7394 lStatus = EffectQueryEffect(i, &desc); 7395 if (lStatus < 0) { 7396 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7397 continue; 7398 } 7399 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7400 // If matching type found save effect descriptor. If the session is 7401 // 0 and the effect is not auxiliary, continue enumeration in case 7402 // an auxiliary version of this effect type is available 7403 found = true; 7404 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7405 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7406 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7407 break; 7408 } 7409 } 7410 } 7411 if (!found) { 7412 lStatus = BAD_VALUE; 7413 ALOGW("createEffect() effect not found"); 7414 goto Exit; 7415 } 7416 // For same effect type, chose auxiliary version over insert version if 7417 // connect to output mix (Compliance to OpenSL ES) 7418 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7419 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7420 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7421 } 7422 } 7423 7424 // Do not allow auxiliary effects on a session different from 0 (output mix) 7425 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7426 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7427 lStatus = INVALID_OPERATION; 7428 goto Exit; 7429 } 7430 7431 // check recording permission for visualizer 7432 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7433 !recordingAllowed()) { 7434 lStatus = PERMISSION_DENIED; 7435 goto Exit; 7436 } 7437 7438 // return effect descriptor 7439 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7440 7441 // If output is not specified try to find a matching audio session ID in one of the 7442 // output threads. 7443 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7444 // because of code checking output when entering the function. 7445 // Note: io is never 0 when creating an effect on an input 7446 if (io == 0) { 7447 // look for the thread where the specified audio session is present 7448 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7449 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7450 io = mPlaybackThreads.keyAt(i); 7451 break; 7452 } 7453 } 7454 if (io == 0) { 7455 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7456 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7457 io = mRecordThreads.keyAt(i); 7458 break; 7459 } 7460 } 7461 } 7462 // If no output thread contains the requested session ID, default to 7463 // first output. The effect chain will be moved to the correct output 7464 // thread when a track with the same session ID is created 7465 if (io == 0 && mPlaybackThreads.size()) { 7466 io = mPlaybackThreads.keyAt(0); 7467 } 7468 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7469 } 7470 ThreadBase *thread = checkRecordThread_l(io); 7471 if (thread == NULL) { 7472 thread = checkPlaybackThread_l(io); 7473 if (thread == NULL) { 7474 ALOGE("createEffect() unknown output thread"); 7475 lStatus = BAD_VALUE; 7476 goto Exit; 7477 } 7478 } 7479 7480 sp<Client> client = registerPid_l(pid); 7481 7482 // create effect on selected output thread 7483 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7484 &desc, enabled, &lStatus); 7485 if (handle != 0 && id != NULL) { 7486 *id = handle->id(); 7487 } 7488 } 7489 7490Exit: 7491 if (status != NULL) { 7492 *status = lStatus; 7493 } 7494 return handle; 7495} 7496 7497status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7498 audio_io_handle_t dstOutput) 7499{ 7500 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7501 sessionId, srcOutput, dstOutput); 7502 Mutex::Autolock _l(mLock); 7503 if (srcOutput == dstOutput) { 7504 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7505 return NO_ERROR; 7506 } 7507 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7508 if (srcThread == NULL) { 7509 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7510 return BAD_VALUE; 7511 } 7512 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7513 if (dstThread == NULL) { 7514 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7515 return BAD_VALUE; 7516 } 7517 7518 Mutex::Autolock _dl(dstThread->mLock); 7519 Mutex::Autolock _sl(srcThread->mLock); 7520 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7521 7522 return NO_ERROR; 7523} 7524 7525// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7526status_t AudioFlinger::moveEffectChain_l(int sessionId, 7527 AudioFlinger::PlaybackThread *srcThread, 7528 AudioFlinger::PlaybackThread *dstThread, 7529 bool reRegister) 7530{ 7531 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7532 sessionId, srcThread, dstThread); 7533 7534 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7535 if (chain == 0) { 7536 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7537 sessionId, srcThread); 7538 return INVALID_OPERATION; 7539 } 7540 7541 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7542 // so that a new chain is created with correct parameters when first effect is added. This is 7543 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7544 // removed. 7545 srcThread->removeEffectChain_l(chain); 7546 7547 // transfer all effects one by one so that new effect chain is created on new thread with 7548 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7549 audio_io_handle_t dstOutput = dstThread->id(); 7550 sp<EffectChain> dstChain; 7551 uint32_t strategy = 0; // prevent compiler warning 7552 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7553 while (effect != 0) { 7554 srcThread->removeEffect_l(effect); 7555 dstThread->addEffect_l(effect); 7556 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7557 if (effect->state() == EffectModule::ACTIVE || 7558 effect->state() == EffectModule::STOPPING) { 7559 effect->start(); 7560 } 7561 // if the move request is not received from audio policy manager, the effect must be 7562 // re-registered with the new strategy and output 7563 if (dstChain == 0) { 7564 dstChain = effect->chain().promote(); 7565 if (dstChain == 0) { 7566 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7567 srcThread->addEffect_l(effect); 7568 return NO_INIT; 7569 } 7570 strategy = dstChain->strategy(); 7571 } 7572 if (reRegister) { 7573 AudioSystem::unregisterEffect(effect->id()); 7574 AudioSystem::registerEffect(&effect->desc(), 7575 dstOutput, 7576 strategy, 7577 sessionId, 7578 effect->id()); 7579 } 7580 effect = chain->getEffectFromId_l(0); 7581 } 7582 7583 return NO_ERROR; 7584} 7585 7586 7587// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7588sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7589 const sp<AudioFlinger::Client>& client, 7590 const sp<IEffectClient>& effectClient, 7591 int32_t priority, 7592 int sessionId, 7593 effect_descriptor_t *desc, 7594 int *enabled, 7595 status_t *status 7596 ) 7597{ 7598 sp<EffectModule> effect; 7599 sp<EffectHandle> handle; 7600 status_t lStatus; 7601 sp<EffectChain> chain; 7602 bool chainCreated = false; 7603 bool effectCreated = false; 7604 bool effectRegistered = false; 7605 7606 lStatus = initCheck(); 7607 if (lStatus != NO_ERROR) { 7608 ALOGW("createEffect_l() Audio driver not initialized."); 7609 goto Exit; 7610 } 7611 7612 // Do not allow effects with session ID 0 on direct output or duplicating threads 7613 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7614 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7615 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7616 desc->name, sessionId); 7617 lStatus = BAD_VALUE; 7618 goto Exit; 7619 } 7620 // Only Pre processor effects are allowed on input threads and only on input threads 7621 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7622 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7623 desc->name, desc->flags, mType); 7624 lStatus = BAD_VALUE; 7625 goto Exit; 7626 } 7627 7628 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7629 7630 { // scope for mLock 7631 Mutex::Autolock _l(mLock); 7632 7633 // check for existing effect chain with the requested audio session 7634 chain = getEffectChain_l(sessionId); 7635 if (chain == 0) { 7636 // create a new chain for this session 7637 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7638 chain = new EffectChain(this, sessionId); 7639 addEffectChain_l(chain); 7640 chain->setStrategy(getStrategyForSession_l(sessionId)); 7641 chainCreated = true; 7642 } else { 7643 effect = chain->getEffectFromDesc_l(desc); 7644 } 7645 7646 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7647 7648 if (effect == 0) { 7649 int id = mAudioFlinger->nextUniqueId(); 7650 // Check CPU and memory usage 7651 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7652 if (lStatus != NO_ERROR) { 7653 goto Exit; 7654 } 7655 effectRegistered = true; 7656 // create a new effect module if none present in the chain 7657 effect = new EffectModule(this, chain, desc, id, sessionId); 7658 lStatus = effect->status(); 7659 if (lStatus != NO_ERROR) { 7660 goto Exit; 7661 } 7662 lStatus = chain->addEffect_l(effect); 7663 if (lStatus != NO_ERROR) { 7664 goto Exit; 7665 } 7666 effectCreated = true; 7667 7668 effect->setDevice(mDevice); 7669 effect->setMode(mAudioFlinger->getMode()); 7670 } 7671 // create effect handle and connect it to effect module 7672 handle = new EffectHandle(effect, client, effectClient, priority); 7673 lStatus = effect->addHandle(handle); 7674 if (enabled != NULL) { 7675 *enabled = (int)effect->isEnabled(); 7676 } 7677 } 7678 7679Exit: 7680 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7681 Mutex::Autolock _l(mLock); 7682 if (effectCreated) { 7683 chain->removeEffect_l(effect); 7684 } 7685 if (effectRegistered) { 7686 AudioSystem::unregisterEffect(effect->id()); 7687 } 7688 if (chainCreated) { 7689 removeEffectChain_l(chain); 7690 } 7691 handle.clear(); 7692 } 7693 7694 if (status != NULL) { 7695 *status = lStatus; 7696 } 7697 return handle; 7698} 7699 7700sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7701{ 7702 Mutex::Autolock _l(mLock); 7703 return getEffect_l(sessionId, effectId); 7704} 7705 7706sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7707{ 7708 sp<EffectChain> chain = getEffectChain_l(sessionId); 7709 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7710} 7711 7712// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7713// PlaybackThread::mLock held 7714status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7715{ 7716 // check for existing effect chain with the requested audio session 7717 int sessionId = effect->sessionId(); 7718 sp<EffectChain> chain = getEffectChain_l(sessionId); 7719 bool chainCreated = false; 7720 7721 if (chain == 0) { 7722 // create a new chain for this session 7723 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7724 chain = new EffectChain(this, sessionId); 7725 addEffectChain_l(chain); 7726 chain->setStrategy(getStrategyForSession_l(sessionId)); 7727 chainCreated = true; 7728 } 7729 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7730 7731 if (chain->getEffectFromId_l(effect->id()) != 0) { 7732 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7733 this, effect->desc().name, chain.get()); 7734 return BAD_VALUE; 7735 } 7736 7737 status_t status = chain->addEffect_l(effect); 7738 if (status != NO_ERROR) { 7739 if (chainCreated) { 7740 removeEffectChain_l(chain); 7741 } 7742 return status; 7743 } 7744 7745 effect->setDevice(mDevice); 7746 effect->setMode(mAudioFlinger->getMode()); 7747 return NO_ERROR; 7748} 7749 7750void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7751 7752 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7753 effect_descriptor_t desc = effect->desc(); 7754 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7755 detachAuxEffect_l(effect->id()); 7756 } 7757 7758 sp<EffectChain> chain = effect->chain().promote(); 7759 if (chain != 0) { 7760 // remove effect chain if removing last effect 7761 if (chain->removeEffect_l(effect) == 0) { 7762 removeEffectChain_l(chain); 7763 } 7764 } else { 7765 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7766 } 7767} 7768 7769void AudioFlinger::ThreadBase::lockEffectChains_l( 7770 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7771{ 7772 effectChains = mEffectChains; 7773 for (size_t i = 0; i < mEffectChains.size(); i++) { 7774 mEffectChains[i]->lock(); 7775 } 7776} 7777 7778void AudioFlinger::ThreadBase::unlockEffectChains( 7779 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7780{ 7781 for (size_t i = 0; i < effectChains.size(); i++) { 7782 effectChains[i]->unlock(); 7783 } 7784} 7785 7786sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7787{ 7788 Mutex::Autolock _l(mLock); 7789 return getEffectChain_l(sessionId); 7790} 7791 7792sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7793{ 7794 size_t size = mEffectChains.size(); 7795 for (size_t i = 0; i < size; i++) { 7796 if (mEffectChains[i]->sessionId() == sessionId) { 7797 return mEffectChains[i]; 7798 } 7799 } 7800 return 0; 7801} 7802 7803void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7804{ 7805 Mutex::Autolock _l(mLock); 7806 size_t size = mEffectChains.size(); 7807 for (size_t i = 0; i < size; i++) { 7808 mEffectChains[i]->setMode_l(mode); 7809 } 7810} 7811 7812void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7813 const wp<EffectHandle>& handle, 7814 bool unpinIfLast) { 7815 7816 Mutex::Autolock _l(mLock); 7817 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7818 // delete the effect module if removing last handle on it 7819 if (effect->removeHandle(handle) == 0) { 7820 if (!effect->isPinned() || unpinIfLast) { 7821 removeEffect_l(effect); 7822 AudioSystem::unregisterEffect(effect->id()); 7823 } 7824 } 7825} 7826 7827status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7828{ 7829 int session = chain->sessionId(); 7830 int16_t *buffer = mMixBuffer; 7831 bool ownsBuffer = false; 7832 7833 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7834 if (session > 0) { 7835 // Only one effect chain can be present in direct output thread and it uses 7836 // the mix buffer as input 7837 if (mType != DIRECT) { 7838 size_t numSamples = mNormalFrameCount * mChannelCount; 7839 buffer = new int16_t[numSamples]; 7840 memset(buffer, 0, numSamples * sizeof(int16_t)); 7841 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7842 ownsBuffer = true; 7843 } 7844 7845 // Attach all tracks with same session ID to this chain. 7846 for (size_t i = 0; i < mTracks.size(); ++i) { 7847 sp<Track> track = mTracks[i]; 7848 if (session == track->sessionId()) { 7849 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7850 track->setMainBuffer(buffer); 7851 chain->incTrackCnt(); 7852 } 7853 } 7854 7855 // indicate all active tracks in the chain 7856 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7857 sp<Track> track = mActiveTracks[i].promote(); 7858 if (track == 0) continue; 7859 if (session == track->sessionId()) { 7860 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7861 chain->incActiveTrackCnt(); 7862 } 7863 } 7864 } 7865 7866 chain->setInBuffer(buffer, ownsBuffer); 7867 chain->setOutBuffer(mMixBuffer); 7868 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7869 // chains list in order to be processed last as it contains output stage effects 7870 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7871 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7872 // after track specific effects and before output stage 7873 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7874 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7875 // Effect chain for other sessions are inserted at beginning of effect 7876 // chains list to be processed before output mix effects. Relative order between other 7877 // sessions is not important 7878 size_t size = mEffectChains.size(); 7879 size_t i = 0; 7880 for (i = 0; i < size; i++) { 7881 if (mEffectChains[i]->sessionId() < session) break; 7882 } 7883 mEffectChains.insertAt(chain, i); 7884 checkSuspendOnAddEffectChain_l(chain); 7885 7886 return NO_ERROR; 7887} 7888 7889size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7890{ 7891 int session = chain->sessionId(); 7892 7893 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7894 7895 for (size_t i = 0; i < mEffectChains.size(); i++) { 7896 if (chain == mEffectChains[i]) { 7897 mEffectChains.removeAt(i); 7898 // detach all active tracks from the chain 7899 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7900 sp<Track> track = mActiveTracks[i].promote(); 7901 if (track == 0) continue; 7902 if (session == track->sessionId()) { 7903 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7904 chain.get(), session); 7905 chain->decActiveTrackCnt(); 7906 } 7907 } 7908 7909 // detach all tracks with same session ID from this chain 7910 for (size_t i = 0; i < mTracks.size(); ++i) { 7911 sp<Track> track = mTracks[i]; 7912 if (session == track->sessionId()) { 7913 track->setMainBuffer(mMixBuffer); 7914 chain->decTrackCnt(); 7915 } 7916 } 7917 break; 7918 } 7919 } 7920 return mEffectChains.size(); 7921} 7922 7923status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7924 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7925{ 7926 Mutex::Autolock _l(mLock); 7927 return attachAuxEffect_l(track, EffectId); 7928} 7929 7930status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7931 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7932{ 7933 status_t status = NO_ERROR; 7934 7935 if (EffectId == 0) { 7936 track->setAuxBuffer(0, NULL); 7937 } else { 7938 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7939 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7940 if (effect != 0) { 7941 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7942 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7943 } else { 7944 status = INVALID_OPERATION; 7945 } 7946 } else { 7947 status = BAD_VALUE; 7948 } 7949 } 7950 return status; 7951} 7952 7953void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7954{ 7955 for (size_t i = 0; i < mTracks.size(); ++i) { 7956 sp<Track> track = mTracks[i]; 7957 if (track->auxEffectId() == effectId) { 7958 attachAuxEffect_l(track, 0); 7959 } 7960 } 7961} 7962 7963status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7964{ 7965 // only one chain per input thread 7966 if (mEffectChains.size() != 0) { 7967 return INVALID_OPERATION; 7968 } 7969 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7970 7971 chain->setInBuffer(NULL); 7972 chain->setOutBuffer(NULL); 7973 7974 checkSuspendOnAddEffectChain_l(chain); 7975 7976 mEffectChains.add(chain); 7977 7978 return NO_ERROR; 7979} 7980 7981size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7982{ 7983 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7984 ALOGW_IF(mEffectChains.size() != 1, 7985 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7986 chain.get(), mEffectChains.size(), this); 7987 if (mEffectChains.size() == 1) { 7988 mEffectChains.removeAt(0); 7989 } 7990 return 0; 7991} 7992 7993// ---------------------------------------------------------------------------- 7994// EffectModule implementation 7995// ---------------------------------------------------------------------------- 7996 7997#undef LOG_TAG 7998#define LOG_TAG "AudioFlinger::EffectModule" 7999 8000AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8001 const wp<AudioFlinger::EffectChain>& chain, 8002 effect_descriptor_t *desc, 8003 int id, 8004 int sessionId) 8005 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 8006 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 8007{ 8008 ALOGV("Constructor %p", this); 8009 int lStatus; 8010 if (thread == NULL) { 8011 return; 8012 } 8013 8014 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 8015 8016 // create effect engine from effect factory 8017 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8018 8019 if (mStatus != NO_ERROR) { 8020 return; 8021 } 8022 lStatus = init(); 8023 if (lStatus < 0) { 8024 mStatus = lStatus; 8025 goto Error; 8026 } 8027 8028 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 8029 mPinned = true; 8030 } 8031 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8032 return; 8033Error: 8034 EffectRelease(mEffectInterface); 8035 mEffectInterface = NULL; 8036 ALOGV("Constructor Error %d", mStatus); 8037} 8038 8039AudioFlinger::EffectModule::~EffectModule() 8040{ 8041 ALOGV("Destructor %p", this); 8042 if (mEffectInterface != NULL) { 8043 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8044 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8045 sp<ThreadBase> thread = mThread.promote(); 8046 if (thread != 0) { 8047 audio_stream_t *stream = thread->stream(); 8048 if (stream != NULL) { 8049 stream->remove_audio_effect(stream, mEffectInterface); 8050 } 8051 } 8052 } 8053 // release effect engine 8054 EffectRelease(mEffectInterface); 8055 } 8056} 8057 8058status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8059{ 8060 status_t status; 8061 8062 Mutex::Autolock _l(mLock); 8063 int priority = handle->priority(); 8064 size_t size = mHandles.size(); 8065 sp<EffectHandle> h; 8066 size_t i; 8067 for (i = 0; i < size; i++) { 8068 h = mHandles[i].promote(); 8069 if (h == 0) continue; 8070 if (h->priority() <= priority) break; 8071 } 8072 // if inserted in first place, move effect control from previous owner to this handle 8073 if (i == 0) { 8074 bool enabled = false; 8075 if (h != 0) { 8076 enabled = h->enabled(); 8077 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8078 } 8079 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8080 status = NO_ERROR; 8081 } else { 8082 status = ALREADY_EXISTS; 8083 } 8084 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8085 mHandles.insertAt(handle, i); 8086 return status; 8087} 8088 8089size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8090{ 8091 Mutex::Autolock _l(mLock); 8092 size_t size = mHandles.size(); 8093 size_t i; 8094 for (i = 0; i < size; i++) { 8095 if (mHandles[i] == handle) break; 8096 } 8097 if (i == size) { 8098 return size; 8099 } 8100 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8101 8102 bool enabled = false; 8103 EffectHandle *hdl = handle.unsafe_get(); 8104 if (hdl != NULL) { 8105 ALOGV("removeHandle() unsafe_get OK"); 8106 enabled = hdl->enabled(); 8107 } 8108 mHandles.removeAt(i); 8109 size = mHandles.size(); 8110 // if removed from first place, move effect control from this handle to next in line 8111 if (i == 0 && size != 0) { 8112 sp<EffectHandle> h = mHandles[0].promote(); 8113 if (h != 0) { 8114 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8115 } 8116 } 8117 8118 // Prevent calls to process() and other functions on effect interface from now on. 8119 // The effect engine will be released by the destructor when the last strong reference on 8120 // this object is released which can happen after next process is called. 8121 if (size == 0 && !mPinned) { 8122 mState = DESTROYED; 8123 } 8124 8125 return size; 8126} 8127 8128sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8129{ 8130 Mutex::Autolock _l(mLock); 8131 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8132} 8133 8134void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8135{ 8136 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8137 // keep a strong reference on this EffectModule to avoid calling the 8138 // destructor before we exit 8139 sp<EffectModule> keep(this); 8140 { 8141 sp<ThreadBase> thread = mThread.promote(); 8142 if (thread != 0) { 8143 thread->disconnectEffect(keep, handle, unpinIfLast); 8144 } 8145 } 8146} 8147 8148void AudioFlinger::EffectModule::updateState() { 8149 Mutex::Autolock _l(mLock); 8150 8151 switch (mState) { 8152 case RESTART: 8153 reset_l(); 8154 // FALL THROUGH 8155 8156 case STARTING: 8157 // clear auxiliary effect input buffer for next accumulation 8158 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8159 memset(mConfig.inputCfg.buffer.raw, 8160 0, 8161 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8162 } 8163 start_l(); 8164 mState = ACTIVE; 8165 break; 8166 case STOPPING: 8167 stop_l(); 8168 mDisableWaitCnt = mMaxDisableWaitCnt; 8169 mState = STOPPED; 8170 break; 8171 case STOPPED: 8172 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8173 // turn off sequence. 8174 if (--mDisableWaitCnt == 0) { 8175 reset_l(); 8176 mState = IDLE; 8177 } 8178 break; 8179 default: //IDLE , ACTIVE, DESTROYED 8180 break; 8181 } 8182} 8183 8184void AudioFlinger::EffectModule::process() 8185{ 8186 Mutex::Autolock _l(mLock); 8187 8188 if (mState == DESTROYED || mEffectInterface == NULL || 8189 mConfig.inputCfg.buffer.raw == NULL || 8190 mConfig.outputCfg.buffer.raw == NULL) { 8191 return; 8192 } 8193 8194 if (isProcessEnabled()) { 8195 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8196 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8197 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8198 mConfig.inputCfg.buffer.s32, 8199 mConfig.inputCfg.buffer.frameCount/2); 8200 } 8201 8202 // do the actual processing in the effect engine 8203 int ret = (*mEffectInterface)->process(mEffectInterface, 8204 &mConfig.inputCfg.buffer, 8205 &mConfig.outputCfg.buffer); 8206 8207 // force transition to IDLE state when engine is ready 8208 if (mState == STOPPED && ret == -ENODATA) { 8209 mDisableWaitCnt = 1; 8210 } 8211 8212 // clear auxiliary effect input buffer for next accumulation 8213 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8214 memset(mConfig.inputCfg.buffer.raw, 0, 8215 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8216 } 8217 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8218 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8219 // If an insert effect is idle and input buffer is different from output buffer, 8220 // accumulate input onto output 8221 sp<EffectChain> chain = mChain.promote(); 8222 if (chain != 0 && chain->activeTrackCnt() != 0) { 8223 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8224 int16_t *in = mConfig.inputCfg.buffer.s16; 8225 int16_t *out = mConfig.outputCfg.buffer.s16; 8226 for (size_t i = 0; i < frameCnt; i++) { 8227 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8228 } 8229 } 8230 } 8231} 8232 8233void AudioFlinger::EffectModule::reset_l() 8234{ 8235 if (mEffectInterface == NULL) { 8236 return; 8237 } 8238 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8239} 8240 8241status_t AudioFlinger::EffectModule::configure() 8242{ 8243 uint32_t channels; 8244 if (mEffectInterface == NULL) { 8245 return NO_INIT; 8246 } 8247 8248 sp<ThreadBase> thread = mThread.promote(); 8249 if (thread == 0) { 8250 return DEAD_OBJECT; 8251 } 8252 8253 // TODO: handle configuration of effects replacing track process 8254 if (thread->channelCount() == 1) { 8255 channels = AUDIO_CHANNEL_OUT_MONO; 8256 } else { 8257 channels = AUDIO_CHANNEL_OUT_STEREO; 8258 } 8259 8260 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8261 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8262 } else { 8263 mConfig.inputCfg.channels = channels; 8264 } 8265 mConfig.outputCfg.channels = channels; 8266 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8267 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8268 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8269 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8270 mConfig.inputCfg.bufferProvider.cookie = NULL; 8271 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8272 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8273 mConfig.outputCfg.bufferProvider.cookie = NULL; 8274 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8275 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8276 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8277 // Insert effect: 8278 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8279 // always overwrites output buffer: input buffer == output buffer 8280 // - in other sessions: 8281 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8282 // other effect: overwrites output buffer: input buffer == output buffer 8283 // Auxiliary effect: 8284 // accumulates in output buffer: input buffer != output buffer 8285 // Therefore: accumulate <=> input buffer != output buffer 8286 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8287 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8288 } else { 8289 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8290 } 8291 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8292 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8293 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8294 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8295 8296 ALOGV("configure() %p thread %p buffer %p framecount %d", 8297 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8298 8299 status_t cmdStatus; 8300 uint32_t size = sizeof(int); 8301 status_t status = (*mEffectInterface)->command(mEffectInterface, 8302 EFFECT_CMD_SET_CONFIG, 8303 sizeof(effect_config_t), 8304 &mConfig, 8305 &size, 8306 &cmdStatus); 8307 if (status == 0) { 8308 status = cmdStatus; 8309 } 8310 8311 if (status == 0 && 8312 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8313 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8314 effect_param_t *p = (effect_param_t *)buf32; 8315 8316 p->psize = sizeof(uint32_t); 8317 p->vsize = sizeof(uint32_t); 8318 size = sizeof(int); 8319 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8320 8321 uint32_t latency = 0; 8322 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8323 if (pbt != NULL) { 8324 latency = pbt->latency_l(); 8325 } 8326 8327 *((int32_t *)p->data + 1)= latency; 8328 (*mEffectInterface)->command(mEffectInterface, 8329 EFFECT_CMD_SET_PARAM, 8330 sizeof(effect_param_t) + 8, 8331 &buf32, 8332 &size, 8333 &cmdStatus); 8334 } 8335 8336 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8337 (1000 * mConfig.outputCfg.buffer.frameCount); 8338 8339 return status; 8340} 8341 8342status_t AudioFlinger::EffectModule::init() 8343{ 8344 Mutex::Autolock _l(mLock); 8345 if (mEffectInterface == NULL) { 8346 return NO_INIT; 8347 } 8348 status_t cmdStatus; 8349 uint32_t size = sizeof(status_t); 8350 status_t status = (*mEffectInterface)->command(mEffectInterface, 8351 EFFECT_CMD_INIT, 8352 0, 8353 NULL, 8354 &size, 8355 &cmdStatus); 8356 if (status == 0) { 8357 status = cmdStatus; 8358 } 8359 return status; 8360} 8361 8362status_t AudioFlinger::EffectModule::start() 8363{ 8364 Mutex::Autolock _l(mLock); 8365 return start_l(); 8366} 8367 8368status_t AudioFlinger::EffectModule::start_l() 8369{ 8370 if (mEffectInterface == NULL) { 8371 return NO_INIT; 8372 } 8373 status_t cmdStatus; 8374 uint32_t size = sizeof(status_t); 8375 status_t status = (*mEffectInterface)->command(mEffectInterface, 8376 EFFECT_CMD_ENABLE, 8377 0, 8378 NULL, 8379 &size, 8380 &cmdStatus); 8381 if (status == 0) { 8382 status = cmdStatus; 8383 } 8384 if (status == 0 && 8385 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8386 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8387 sp<ThreadBase> thread = mThread.promote(); 8388 if (thread != 0) { 8389 audio_stream_t *stream = thread->stream(); 8390 if (stream != NULL) { 8391 stream->add_audio_effect(stream, mEffectInterface); 8392 } 8393 } 8394 } 8395 return status; 8396} 8397 8398status_t AudioFlinger::EffectModule::stop() 8399{ 8400 Mutex::Autolock _l(mLock); 8401 return stop_l(); 8402} 8403 8404status_t AudioFlinger::EffectModule::stop_l() 8405{ 8406 if (mEffectInterface == NULL) { 8407 return NO_INIT; 8408 } 8409 status_t cmdStatus; 8410 uint32_t size = sizeof(status_t); 8411 status_t status = (*mEffectInterface)->command(mEffectInterface, 8412 EFFECT_CMD_DISABLE, 8413 0, 8414 NULL, 8415 &size, 8416 &cmdStatus); 8417 if (status == 0) { 8418 status = cmdStatus; 8419 } 8420 if (status == 0 && 8421 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8422 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8423 sp<ThreadBase> thread = mThread.promote(); 8424 if (thread != 0) { 8425 audio_stream_t *stream = thread->stream(); 8426 if (stream != NULL) { 8427 stream->remove_audio_effect(stream, mEffectInterface); 8428 } 8429 } 8430 } 8431 return status; 8432} 8433 8434status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8435 uint32_t cmdSize, 8436 void *pCmdData, 8437 uint32_t *replySize, 8438 void *pReplyData) 8439{ 8440 Mutex::Autolock _l(mLock); 8441// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8442 8443 if (mState == DESTROYED || mEffectInterface == NULL) { 8444 return NO_INIT; 8445 } 8446 status_t status = (*mEffectInterface)->command(mEffectInterface, 8447 cmdCode, 8448 cmdSize, 8449 pCmdData, 8450 replySize, 8451 pReplyData); 8452 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8453 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8454 for (size_t i = 1; i < mHandles.size(); i++) { 8455 sp<EffectHandle> h = mHandles[i].promote(); 8456 if (h != 0) { 8457 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8458 } 8459 } 8460 } 8461 return status; 8462} 8463 8464status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8465{ 8466 8467 Mutex::Autolock _l(mLock); 8468 ALOGV("setEnabled %p enabled %d", this, enabled); 8469 8470 if (enabled != isEnabled()) { 8471 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8472 if (enabled && status != NO_ERROR) { 8473 return status; 8474 } 8475 8476 switch (mState) { 8477 // going from disabled to enabled 8478 case IDLE: 8479 mState = STARTING; 8480 break; 8481 case STOPPED: 8482 mState = RESTART; 8483 break; 8484 case STOPPING: 8485 mState = ACTIVE; 8486 break; 8487 8488 // going from enabled to disabled 8489 case RESTART: 8490 mState = STOPPED; 8491 break; 8492 case STARTING: 8493 mState = IDLE; 8494 break; 8495 case ACTIVE: 8496 mState = STOPPING; 8497 break; 8498 case DESTROYED: 8499 return NO_ERROR; // simply ignore as we are being destroyed 8500 } 8501 for (size_t i = 1; i < mHandles.size(); i++) { 8502 sp<EffectHandle> h = mHandles[i].promote(); 8503 if (h != 0) { 8504 h->setEnabled(enabled); 8505 } 8506 } 8507 } 8508 return NO_ERROR; 8509} 8510 8511bool AudioFlinger::EffectModule::isEnabled() const 8512{ 8513 switch (mState) { 8514 case RESTART: 8515 case STARTING: 8516 case ACTIVE: 8517 return true; 8518 case IDLE: 8519 case STOPPING: 8520 case STOPPED: 8521 case DESTROYED: 8522 default: 8523 return false; 8524 } 8525} 8526 8527bool AudioFlinger::EffectModule::isProcessEnabled() const 8528{ 8529 switch (mState) { 8530 case RESTART: 8531 case ACTIVE: 8532 case STOPPING: 8533 case STOPPED: 8534 return true; 8535 case IDLE: 8536 case STARTING: 8537 case DESTROYED: 8538 default: 8539 return false; 8540 } 8541} 8542 8543status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8544{ 8545 Mutex::Autolock _l(mLock); 8546 status_t status = NO_ERROR; 8547 8548 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8549 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8550 if (isProcessEnabled() && 8551 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8552 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8553 status_t cmdStatus; 8554 uint32_t volume[2]; 8555 uint32_t *pVolume = NULL; 8556 uint32_t size = sizeof(volume); 8557 volume[0] = *left; 8558 volume[1] = *right; 8559 if (controller) { 8560 pVolume = volume; 8561 } 8562 status = (*mEffectInterface)->command(mEffectInterface, 8563 EFFECT_CMD_SET_VOLUME, 8564 size, 8565 volume, 8566 &size, 8567 pVolume); 8568 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8569 *left = volume[0]; 8570 *right = volume[1]; 8571 } 8572 } 8573 return status; 8574} 8575 8576status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8577{ 8578 Mutex::Autolock _l(mLock); 8579 status_t status = NO_ERROR; 8580 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8581 // audio pre processing modules on RecordThread can receive both output and 8582 // input device indication in the same call 8583 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8584 if (dev) { 8585 status_t cmdStatus; 8586 uint32_t size = sizeof(status_t); 8587 8588 status = (*mEffectInterface)->command(mEffectInterface, 8589 EFFECT_CMD_SET_DEVICE, 8590 sizeof(uint32_t), 8591 &dev, 8592 &size, 8593 &cmdStatus); 8594 if (status == NO_ERROR) { 8595 status = cmdStatus; 8596 } 8597 } 8598 dev = device & AUDIO_DEVICE_IN_ALL; 8599 if (dev) { 8600 status_t cmdStatus; 8601 uint32_t size = sizeof(status_t); 8602 8603 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8604 EFFECT_CMD_SET_INPUT_DEVICE, 8605 sizeof(uint32_t), 8606 &dev, 8607 &size, 8608 &cmdStatus); 8609 if (status2 == NO_ERROR) { 8610 status2 = cmdStatus; 8611 } 8612 if (status == NO_ERROR) { 8613 status = status2; 8614 } 8615 } 8616 } 8617 return status; 8618} 8619 8620status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8621{ 8622 Mutex::Autolock _l(mLock); 8623 status_t status = NO_ERROR; 8624 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8625 status_t cmdStatus; 8626 uint32_t size = sizeof(status_t); 8627 status = (*mEffectInterface)->command(mEffectInterface, 8628 EFFECT_CMD_SET_AUDIO_MODE, 8629 sizeof(audio_mode_t), 8630 &mode, 8631 &size, 8632 &cmdStatus); 8633 if (status == NO_ERROR) { 8634 status = cmdStatus; 8635 } 8636 } 8637 return status; 8638} 8639 8640void AudioFlinger::EffectModule::setSuspended(bool suspended) 8641{ 8642 Mutex::Autolock _l(mLock); 8643 mSuspended = suspended; 8644} 8645 8646bool AudioFlinger::EffectModule::suspended() const 8647{ 8648 Mutex::Autolock _l(mLock); 8649 return mSuspended; 8650} 8651 8652status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8653{ 8654 const size_t SIZE = 256; 8655 char buffer[SIZE]; 8656 String8 result; 8657 8658 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8659 result.append(buffer); 8660 8661 bool locked = tryLock(mLock); 8662 // failed to lock - AudioFlinger is probably deadlocked 8663 if (!locked) { 8664 result.append("\t\tCould not lock Fx mutex:\n"); 8665 } 8666 8667 result.append("\t\tSession Status State Engine:\n"); 8668 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8669 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8670 result.append(buffer); 8671 8672 result.append("\t\tDescriptor:\n"); 8673 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8674 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8675 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8676 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8677 result.append(buffer); 8678 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8679 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8680 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8681 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8682 result.append(buffer); 8683 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8684 mDescriptor.apiVersion, 8685 mDescriptor.flags); 8686 result.append(buffer); 8687 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8688 mDescriptor.name); 8689 result.append(buffer); 8690 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8691 mDescriptor.implementor); 8692 result.append(buffer); 8693 8694 result.append("\t\t- Input configuration:\n"); 8695 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8696 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8697 (uint32_t)mConfig.inputCfg.buffer.raw, 8698 mConfig.inputCfg.buffer.frameCount, 8699 mConfig.inputCfg.samplingRate, 8700 mConfig.inputCfg.channels, 8701 mConfig.inputCfg.format); 8702 result.append(buffer); 8703 8704 result.append("\t\t- Output configuration:\n"); 8705 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8706 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8707 (uint32_t)mConfig.outputCfg.buffer.raw, 8708 mConfig.outputCfg.buffer.frameCount, 8709 mConfig.outputCfg.samplingRate, 8710 mConfig.outputCfg.channels, 8711 mConfig.outputCfg.format); 8712 result.append(buffer); 8713 8714 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8715 result.append(buffer); 8716 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8717 for (size_t i = 0; i < mHandles.size(); ++i) { 8718 sp<EffectHandle> handle = mHandles[i].promote(); 8719 if (handle != 0) { 8720 handle->dump(buffer, SIZE); 8721 result.append(buffer); 8722 } 8723 } 8724 8725 result.append("\n"); 8726 8727 write(fd, result.string(), result.length()); 8728 8729 if (locked) { 8730 mLock.unlock(); 8731 } 8732 8733 return NO_ERROR; 8734} 8735 8736// ---------------------------------------------------------------------------- 8737// EffectHandle implementation 8738// ---------------------------------------------------------------------------- 8739 8740#undef LOG_TAG 8741#define LOG_TAG "AudioFlinger::EffectHandle" 8742 8743AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8744 const sp<AudioFlinger::Client>& client, 8745 const sp<IEffectClient>& effectClient, 8746 int32_t priority) 8747 : BnEffect(), 8748 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8749 mPriority(priority), mHasControl(false), mEnabled(false) 8750{ 8751 ALOGV("constructor %p", this); 8752 8753 if (client == 0) { 8754 return; 8755 } 8756 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8757 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8758 if (mCblkMemory != 0) { 8759 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8760 8761 if (mCblk != NULL) { 8762 new(mCblk) effect_param_cblk_t(); 8763 mBuffer = (uint8_t *)mCblk + bufOffset; 8764 } 8765 } else { 8766 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8767 return; 8768 } 8769} 8770 8771AudioFlinger::EffectHandle::~EffectHandle() 8772{ 8773 ALOGV("Destructor %p", this); 8774 disconnect(false); 8775 ALOGV("Destructor DONE %p", this); 8776} 8777 8778status_t AudioFlinger::EffectHandle::enable() 8779{ 8780 ALOGV("enable %p", this); 8781 if (!mHasControl) return INVALID_OPERATION; 8782 if (mEffect == 0) return DEAD_OBJECT; 8783 8784 if (mEnabled) { 8785 return NO_ERROR; 8786 } 8787 8788 mEnabled = true; 8789 8790 sp<ThreadBase> thread = mEffect->thread().promote(); 8791 if (thread != 0) { 8792 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8793 } 8794 8795 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8796 if (mEffect->suspended()) { 8797 return NO_ERROR; 8798 } 8799 8800 status_t status = mEffect->setEnabled(true); 8801 if (status != NO_ERROR) { 8802 if (thread != 0) { 8803 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8804 } 8805 mEnabled = false; 8806 } 8807 return status; 8808} 8809 8810status_t AudioFlinger::EffectHandle::disable() 8811{ 8812 ALOGV("disable %p", this); 8813 if (!mHasControl) return INVALID_OPERATION; 8814 if (mEffect == 0) return DEAD_OBJECT; 8815 8816 if (!mEnabled) { 8817 return NO_ERROR; 8818 } 8819 mEnabled = false; 8820 8821 if (mEffect->suspended()) { 8822 return NO_ERROR; 8823 } 8824 8825 status_t status = mEffect->setEnabled(false); 8826 8827 sp<ThreadBase> thread = mEffect->thread().promote(); 8828 if (thread != 0) { 8829 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8830 } 8831 8832 return status; 8833} 8834 8835void AudioFlinger::EffectHandle::disconnect() 8836{ 8837 disconnect(true); 8838} 8839 8840void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8841{ 8842 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8843 if (mEffect == 0) { 8844 return; 8845 } 8846 mEffect->disconnect(this, unpinIfLast); 8847 8848 if (mHasControl && mEnabled) { 8849 sp<ThreadBase> thread = mEffect->thread().promote(); 8850 if (thread != 0) { 8851 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8852 } 8853 } 8854 8855 // release sp on module => module destructor can be called now 8856 mEffect.clear(); 8857 if (mClient != 0) { 8858 if (mCblk != NULL) { 8859 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8860 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8861 } 8862 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8863 // Client destructor must run with AudioFlinger mutex locked 8864 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8865 mClient.clear(); 8866 } 8867} 8868 8869status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8870 uint32_t cmdSize, 8871 void *pCmdData, 8872 uint32_t *replySize, 8873 void *pReplyData) 8874{ 8875// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8876// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8877 8878 // only get parameter command is permitted for applications not controlling the effect 8879 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8880 return INVALID_OPERATION; 8881 } 8882 if (mEffect == 0) return DEAD_OBJECT; 8883 if (mClient == 0) return INVALID_OPERATION; 8884 8885 // handle commands that are not forwarded transparently to effect engine 8886 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8887 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8888 // no risk to block the whole media server process or mixer threads is we are stuck here 8889 Mutex::Autolock _l(mCblk->lock); 8890 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8891 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8892 mCblk->serverIndex = 0; 8893 mCblk->clientIndex = 0; 8894 return BAD_VALUE; 8895 } 8896 status_t status = NO_ERROR; 8897 while (mCblk->serverIndex < mCblk->clientIndex) { 8898 int reply; 8899 uint32_t rsize = sizeof(int); 8900 int *p = (int *)(mBuffer + mCblk->serverIndex); 8901 int size = *p++; 8902 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8903 ALOGW("command(): invalid parameter block size"); 8904 break; 8905 } 8906 effect_param_t *param = (effect_param_t *)p; 8907 if (param->psize == 0 || param->vsize == 0) { 8908 ALOGW("command(): null parameter or value size"); 8909 mCblk->serverIndex += size; 8910 continue; 8911 } 8912 uint32_t psize = sizeof(effect_param_t) + 8913 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8914 param->vsize; 8915 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8916 psize, 8917 p, 8918 &rsize, 8919 &reply); 8920 // stop at first error encountered 8921 if (ret != NO_ERROR) { 8922 status = ret; 8923 *(int *)pReplyData = reply; 8924 break; 8925 } else if (reply != NO_ERROR) { 8926 *(int *)pReplyData = reply; 8927 break; 8928 } 8929 mCblk->serverIndex += size; 8930 } 8931 mCblk->serverIndex = 0; 8932 mCblk->clientIndex = 0; 8933 return status; 8934 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8935 *(int *)pReplyData = NO_ERROR; 8936 return enable(); 8937 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8938 *(int *)pReplyData = NO_ERROR; 8939 return disable(); 8940 } 8941 8942 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8943} 8944 8945void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8946{ 8947 ALOGV("setControl %p control %d", this, hasControl); 8948 8949 mHasControl = hasControl; 8950 mEnabled = enabled; 8951 8952 if (signal && mEffectClient != 0) { 8953 mEffectClient->controlStatusChanged(hasControl); 8954 } 8955} 8956 8957void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8958 uint32_t cmdSize, 8959 void *pCmdData, 8960 uint32_t replySize, 8961 void *pReplyData) 8962{ 8963 if (mEffectClient != 0) { 8964 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8965 } 8966} 8967 8968 8969 8970void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8971{ 8972 if (mEffectClient != 0) { 8973 mEffectClient->enableStatusChanged(enabled); 8974 } 8975} 8976 8977status_t AudioFlinger::EffectHandle::onTransact( 8978 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8979{ 8980 return BnEffect::onTransact(code, data, reply, flags); 8981} 8982 8983 8984void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8985{ 8986 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8987 8988 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8989 (mClient == 0) ? getpid_cached : mClient->pid(), 8990 mPriority, 8991 mHasControl, 8992 !locked, 8993 mCblk ? mCblk->clientIndex : 0, 8994 mCblk ? mCblk->serverIndex : 0 8995 ); 8996 8997 if (locked) { 8998 mCblk->lock.unlock(); 8999 } 9000} 9001 9002#undef LOG_TAG 9003#define LOG_TAG "AudioFlinger::EffectChain" 9004 9005AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9006 int sessionId) 9007 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9008 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9009 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9010{ 9011 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9012 if (thread == NULL) { 9013 return; 9014 } 9015 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9016 thread->frameCount(); 9017} 9018 9019AudioFlinger::EffectChain::~EffectChain() 9020{ 9021 if (mOwnInBuffer) { 9022 delete mInBuffer; 9023 } 9024 9025} 9026 9027// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9028sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9029{ 9030 size_t size = mEffects.size(); 9031 9032 for (size_t i = 0; i < size; i++) { 9033 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9034 return mEffects[i]; 9035 } 9036 } 9037 return 0; 9038} 9039 9040// getEffectFromId_l() must be called with ThreadBase::mLock held 9041sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9042{ 9043 size_t size = mEffects.size(); 9044 9045 for (size_t i = 0; i < size; i++) { 9046 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9047 if (id == 0 || mEffects[i]->id() == id) { 9048 return mEffects[i]; 9049 } 9050 } 9051 return 0; 9052} 9053 9054// getEffectFromType_l() must be called with ThreadBase::mLock held 9055sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9056 const effect_uuid_t *type) 9057{ 9058 size_t size = mEffects.size(); 9059 9060 for (size_t i = 0; i < size; i++) { 9061 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9062 return mEffects[i]; 9063 } 9064 } 9065 return 0; 9066} 9067 9068void AudioFlinger::EffectChain::clearInputBuffer() 9069{ 9070 Mutex::Autolock _l(mLock); 9071 sp<ThreadBase> thread = mThread.promote(); 9072 if (thread == 0) { 9073 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9074 return; 9075 } 9076 clearInputBuffer_l(thread); 9077} 9078 9079// Must be called with EffectChain::mLock locked 9080void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9081{ 9082 size_t numSamples = thread->frameCount() * thread->channelCount(); 9083 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9084 9085} 9086 9087// Must be called with EffectChain::mLock locked 9088void AudioFlinger::EffectChain::process_l() 9089{ 9090 sp<ThreadBase> thread = mThread.promote(); 9091 if (thread == 0) { 9092 ALOGW("process_l(): cannot promote mixer thread"); 9093 return; 9094 } 9095 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9096 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9097 // always process effects unless no more tracks are on the session and the effect tail 9098 // has been rendered 9099 bool doProcess = true; 9100 if (!isGlobalSession) { 9101 bool tracksOnSession = (trackCnt() != 0); 9102 9103 if (!tracksOnSession && mTailBufferCount == 0) { 9104 doProcess = false; 9105 } 9106 9107 if (activeTrackCnt() == 0) { 9108 // if no track is active and the effect tail has not been rendered, 9109 // the input buffer must be cleared here as the mixer process will not do it 9110 if (tracksOnSession || mTailBufferCount > 0) { 9111 clearInputBuffer_l(thread); 9112 if (mTailBufferCount > 0) { 9113 mTailBufferCount--; 9114 } 9115 } 9116 } 9117 } 9118 9119 size_t size = mEffects.size(); 9120 if (doProcess) { 9121 for (size_t i = 0; i < size; i++) { 9122 mEffects[i]->process(); 9123 } 9124 } 9125 for (size_t i = 0; i < size; i++) { 9126 mEffects[i]->updateState(); 9127 } 9128} 9129 9130// addEffect_l() must be called with PlaybackThread::mLock held 9131status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9132{ 9133 effect_descriptor_t desc = effect->desc(); 9134 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9135 9136 Mutex::Autolock _l(mLock); 9137 effect->setChain(this); 9138 sp<ThreadBase> thread = mThread.promote(); 9139 if (thread == 0) { 9140 return NO_INIT; 9141 } 9142 effect->setThread(thread); 9143 9144 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9145 // Auxiliary effects are inserted at the beginning of mEffects vector as 9146 // they are processed first and accumulated in chain input buffer 9147 mEffects.insertAt(effect, 0); 9148 9149 // the input buffer for auxiliary effect contains mono samples in 9150 // 32 bit format. This is to avoid saturation in AudoMixer 9151 // accumulation stage. Saturation is done in EffectModule::process() before 9152 // calling the process in effect engine 9153 size_t numSamples = thread->frameCount(); 9154 int32_t *buffer = new int32_t[numSamples]; 9155 memset(buffer, 0, numSamples * sizeof(int32_t)); 9156 effect->setInBuffer((int16_t *)buffer); 9157 // auxiliary effects output samples to chain input buffer for further processing 9158 // by insert effects 9159 effect->setOutBuffer(mInBuffer); 9160 } else { 9161 // Insert effects are inserted at the end of mEffects vector as they are processed 9162 // after track and auxiliary effects. 9163 // Insert effect order as a function of indicated preference: 9164 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9165 // another effect is present 9166 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9167 // last effect claiming first position 9168 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9169 // first effect claiming last position 9170 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9171 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9172 // already present 9173 9174 size_t size = mEffects.size(); 9175 size_t idx_insert = size; 9176 ssize_t idx_insert_first = -1; 9177 ssize_t idx_insert_last = -1; 9178 9179 for (size_t i = 0; i < size; i++) { 9180 effect_descriptor_t d = mEffects[i]->desc(); 9181 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9182 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9183 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9184 // check invalid effect chaining combinations 9185 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9186 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9187 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9188 return INVALID_OPERATION; 9189 } 9190 // remember position of first insert effect and by default 9191 // select this as insert position for new effect 9192 if (idx_insert == size) { 9193 idx_insert = i; 9194 } 9195 // remember position of last insert effect claiming 9196 // first position 9197 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9198 idx_insert_first = i; 9199 } 9200 // remember position of first insert effect claiming 9201 // last position 9202 if (iPref == EFFECT_FLAG_INSERT_LAST && 9203 idx_insert_last == -1) { 9204 idx_insert_last = i; 9205 } 9206 } 9207 } 9208 9209 // modify idx_insert from first position if needed 9210 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9211 if (idx_insert_last != -1) { 9212 idx_insert = idx_insert_last; 9213 } else { 9214 idx_insert = size; 9215 } 9216 } else { 9217 if (idx_insert_first != -1) { 9218 idx_insert = idx_insert_first + 1; 9219 } 9220 } 9221 9222 // always read samples from chain input buffer 9223 effect->setInBuffer(mInBuffer); 9224 9225 // if last effect in the chain, output samples to chain 9226 // output buffer, otherwise to chain input buffer 9227 if (idx_insert == size) { 9228 if (idx_insert != 0) { 9229 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9230 mEffects[idx_insert-1]->configure(); 9231 } 9232 effect->setOutBuffer(mOutBuffer); 9233 } else { 9234 effect->setOutBuffer(mInBuffer); 9235 } 9236 mEffects.insertAt(effect, idx_insert); 9237 9238 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9239 } 9240 effect->configure(); 9241 return NO_ERROR; 9242} 9243 9244// removeEffect_l() must be called with PlaybackThread::mLock held 9245size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9246{ 9247 Mutex::Autolock _l(mLock); 9248 size_t size = mEffects.size(); 9249 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9250 9251 for (size_t i = 0; i < size; i++) { 9252 if (effect == mEffects[i]) { 9253 // calling stop here will remove pre-processing effect from the audio HAL. 9254 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9255 // the middle of a read from audio HAL 9256 if (mEffects[i]->state() == EffectModule::ACTIVE || 9257 mEffects[i]->state() == EffectModule::STOPPING) { 9258 mEffects[i]->stop(); 9259 } 9260 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9261 delete[] effect->inBuffer(); 9262 } else { 9263 if (i == size - 1 && i != 0) { 9264 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9265 mEffects[i - 1]->configure(); 9266 } 9267 } 9268 mEffects.removeAt(i); 9269 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9270 break; 9271 } 9272 } 9273 9274 return mEffects.size(); 9275} 9276 9277// setDevice_l() must be called with PlaybackThread::mLock held 9278void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9279{ 9280 size_t size = mEffects.size(); 9281 for (size_t i = 0; i < size; i++) { 9282 mEffects[i]->setDevice(device); 9283 } 9284} 9285 9286// setMode_l() must be called with PlaybackThread::mLock held 9287void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9288{ 9289 size_t size = mEffects.size(); 9290 for (size_t i = 0; i < size; i++) { 9291 mEffects[i]->setMode(mode); 9292 } 9293} 9294 9295// setVolume_l() must be called with PlaybackThread::mLock held 9296bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9297{ 9298 uint32_t newLeft = *left; 9299 uint32_t newRight = *right; 9300 bool hasControl = false; 9301 int ctrlIdx = -1; 9302 size_t size = mEffects.size(); 9303 9304 // first update volume controller 9305 for (size_t i = size; i > 0; i--) { 9306 if (mEffects[i - 1]->isProcessEnabled() && 9307 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9308 ctrlIdx = i - 1; 9309 hasControl = true; 9310 break; 9311 } 9312 } 9313 9314 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9315 if (hasControl) { 9316 *left = mNewLeftVolume; 9317 *right = mNewRightVolume; 9318 } 9319 return hasControl; 9320 } 9321 9322 mVolumeCtrlIdx = ctrlIdx; 9323 mLeftVolume = newLeft; 9324 mRightVolume = newRight; 9325 9326 // second get volume update from volume controller 9327 if (ctrlIdx >= 0) { 9328 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9329 mNewLeftVolume = newLeft; 9330 mNewRightVolume = newRight; 9331 } 9332 // then indicate volume to all other effects in chain. 9333 // Pass altered volume to effects before volume controller 9334 // and requested volume to effects after controller 9335 uint32_t lVol = newLeft; 9336 uint32_t rVol = newRight; 9337 9338 for (size_t i = 0; i < size; i++) { 9339 if ((int)i == ctrlIdx) continue; 9340 // this also works for ctrlIdx == -1 when there is no volume controller 9341 if ((int)i > ctrlIdx) { 9342 lVol = *left; 9343 rVol = *right; 9344 } 9345 mEffects[i]->setVolume(&lVol, &rVol, false); 9346 } 9347 *left = newLeft; 9348 *right = newRight; 9349 9350 return hasControl; 9351} 9352 9353status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9354{ 9355 const size_t SIZE = 256; 9356 char buffer[SIZE]; 9357 String8 result; 9358 9359 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9360 result.append(buffer); 9361 9362 bool locked = tryLock(mLock); 9363 // failed to lock - AudioFlinger is probably deadlocked 9364 if (!locked) { 9365 result.append("\tCould not lock mutex:\n"); 9366 } 9367 9368 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9369 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9370 mEffects.size(), 9371 (uint32_t)mInBuffer, 9372 (uint32_t)mOutBuffer, 9373 mActiveTrackCnt); 9374 result.append(buffer); 9375 write(fd, result.string(), result.size()); 9376 9377 for (size_t i = 0; i < mEffects.size(); ++i) { 9378 sp<EffectModule> effect = mEffects[i]; 9379 if (effect != 0) { 9380 effect->dump(fd, args); 9381 } 9382 } 9383 9384 if (locked) { 9385 mLock.unlock(); 9386 } 9387 9388 return NO_ERROR; 9389} 9390 9391// must be called with ThreadBase::mLock held 9392void AudioFlinger::EffectChain::setEffectSuspended_l( 9393 const effect_uuid_t *type, bool suspend) 9394{ 9395 sp<SuspendedEffectDesc> desc; 9396 // use effect type UUID timelow as key as there is no real risk of identical 9397 // timeLow fields among effect type UUIDs. 9398 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9399 if (suspend) { 9400 if (index >= 0) { 9401 desc = mSuspendedEffects.valueAt(index); 9402 } else { 9403 desc = new SuspendedEffectDesc(); 9404 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9405 mSuspendedEffects.add(type->timeLow, desc); 9406 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9407 } 9408 if (desc->mRefCount++ == 0) { 9409 sp<EffectModule> effect = getEffectIfEnabled(type); 9410 if (effect != 0) { 9411 desc->mEffect = effect; 9412 effect->setSuspended(true); 9413 effect->setEnabled(false); 9414 } 9415 } 9416 } else { 9417 if (index < 0) { 9418 return; 9419 } 9420 desc = mSuspendedEffects.valueAt(index); 9421 if (desc->mRefCount <= 0) { 9422 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9423 desc->mRefCount = 1; 9424 } 9425 if (--desc->mRefCount == 0) { 9426 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9427 if (desc->mEffect != 0) { 9428 sp<EffectModule> effect = desc->mEffect.promote(); 9429 if (effect != 0) { 9430 effect->setSuspended(false); 9431 sp<EffectHandle> handle = effect->controlHandle(); 9432 if (handle != 0) { 9433 effect->setEnabled(handle->enabled()); 9434 } 9435 } 9436 desc->mEffect.clear(); 9437 } 9438 mSuspendedEffects.removeItemsAt(index); 9439 } 9440 } 9441} 9442 9443// must be called with ThreadBase::mLock held 9444void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9445{ 9446 sp<SuspendedEffectDesc> desc; 9447 9448 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9449 if (suspend) { 9450 if (index >= 0) { 9451 desc = mSuspendedEffects.valueAt(index); 9452 } else { 9453 desc = new SuspendedEffectDesc(); 9454 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9455 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9456 } 9457 if (desc->mRefCount++ == 0) { 9458 Vector< sp<EffectModule> > effects; 9459 getSuspendEligibleEffects(effects); 9460 for (size_t i = 0; i < effects.size(); i++) { 9461 setEffectSuspended_l(&effects[i]->desc().type, true); 9462 } 9463 } 9464 } else { 9465 if (index < 0) { 9466 return; 9467 } 9468 desc = mSuspendedEffects.valueAt(index); 9469 if (desc->mRefCount <= 0) { 9470 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9471 desc->mRefCount = 1; 9472 } 9473 if (--desc->mRefCount == 0) { 9474 Vector<const effect_uuid_t *> types; 9475 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9476 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9477 continue; 9478 } 9479 types.add(&mSuspendedEffects.valueAt(i)->mType); 9480 } 9481 for (size_t i = 0; i < types.size(); i++) { 9482 setEffectSuspended_l(types[i], false); 9483 } 9484 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9485 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9486 } 9487 } 9488} 9489 9490 9491// The volume effect is used for automated tests only 9492#ifndef OPENSL_ES_H_ 9493static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9494 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9495const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9496#endif //OPENSL_ES_H_ 9497 9498bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9499{ 9500 // auxiliary effects and visualizer are never suspended on output mix 9501 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9502 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9503 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9504 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9505 return false; 9506 } 9507 return true; 9508} 9509 9510void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9511{ 9512 effects.clear(); 9513 for (size_t i = 0; i < mEffects.size(); i++) { 9514 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9515 effects.add(mEffects[i]); 9516 } 9517 } 9518} 9519 9520sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9521 const effect_uuid_t *type) 9522{ 9523 sp<EffectModule> effect = getEffectFromType_l(type); 9524 return effect != 0 && effect->isEnabled() ? effect : 0; 9525} 9526 9527void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9528 bool enabled) 9529{ 9530 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9531 if (enabled) { 9532 if (index < 0) { 9533 // if the effect is not suspend check if all effects are suspended 9534 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9535 if (index < 0) { 9536 return; 9537 } 9538 if (!isEffectEligibleForSuspend(effect->desc())) { 9539 return; 9540 } 9541 setEffectSuspended_l(&effect->desc().type, enabled); 9542 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9543 if (index < 0) { 9544 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9545 return; 9546 } 9547 } 9548 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9549 effect->desc().type.timeLow); 9550 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9551 // if effect is requested to suspended but was not yet enabled, supend it now. 9552 if (desc->mEffect == 0) { 9553 desc->mEffect = effect; 9554 effect->setEnabled(false); 9555 effect->setSuspended(true); 9556 } 9557 } else { 9558 if (index < 0) { 9559 return; 9560 } 9561 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9562 effect->desc().type.timeLow); 9563 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9564 desc->mEffect.clear(); 9565 effect->setSuspended(false); 9566 } 9567} 9568 9569#undef LOG_TAG 9570#define LOG_TAG "AudioFlinger" 9571 9572// ---------------------------------------------------------------------------- 9573 9574status_t AudioFlinger::onTransact( 9575 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9576{ 9577 return BnAudioFlinger::onTransact(code, data, reply, flags); 9578} 9579 9580}; // namespace android 9581