AudioFlinger.cpp revision 717e128691f083a9469a1d0e363ac6ecd5c65d58
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 if (srcThread == 0) { 4700 return INVALID_OPERATION; 4701 } 4702 4703 if (EffectId != 0 && playbackThread != srcThread.get()) { 4704 Mutex::Autolock _dl(playbackThread->mLock); 4705 Mutex::Autolock _sl(srcThread->mLock); 4706 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4707 if (chain == 0) { 4708 return INVALID_OPERATION; 4709 } 4710 4711 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4712 if (effect == 0) { 4713 return INVALID_OPERATION; 4714 } 4715 srcThread->removeEffect_l(effect); 4716 playbackThread->addEffect_l(effect); 4717 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4718 if (effect->state() == EffectModule::ACTIVE || 4719 effect->state() == EffectModule::STOPPING) { 4720 effect->start(); 4721 } 4722 4723 sp<EffectChain> dstChain = effect->chain().promote(); 4724 if (dstChain == 0) { 4725 srcThread->addEffect_l(effect); 4726 return INVALID_OPERATION; 4727 } 4728 AudioSystem::unregisterEffect(effect->id()); 4729 AudioSystem::registerEffect(&effect->desc(), 4730 srcThread->id(), 4731 dstChain->strategy(), 4732 AUDIO_SESSION_OUTPUT_MIX, 4733 effect->id()); 4734 } 4735 status = playbackThread->attachAuxEffect(this, EffectId); 4736 } 4737 return status; 4738} 4739 4740void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4741{ 4742 mAuxEffectId = EffectId; 4743 mAuxBuffer = buffer; 4744} 4745 4746bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4747 size_t audioHalFrames) 4748{ 4749 // a track is considered presented when the total number of frames written to audio HAL 4750 // corresponds to the number of frames written when presentationComplete() is called for the 4751 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4752 if (mPresentationCompleteFrames == 0) { 4753 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4754 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4755 mPresentationCompleteFrames, audioHalFrames); 4756 } 4757 if (framesWritten >= mPresentationCompleteFrames) { 4758 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4759 mSessionId, framesWritten); 4760 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4761 return true; 4762 } 4763 return false; 4764} 4765 4766void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4767{ 4768 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4769 if (mSyncEvents[i]->type() == type) { 4770 mSyncEvents[i]->trigger(); 4771 mSyncEvents.removeAt(i); 4772 i--; 4773 } 4774 } 4775} 4776 4777// implement VolumeBufferProvider interface 4778 4779uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4780{ 4781 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4782 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4783 uint32_t vlr = mCblk->getVolumeLR(); 4784 uint32_t vl = vlr & 0xFFFF; 4785 uint32_t vr = vlr >> 16; 4786 // track volumes come from shared memory, so can't be trusted and must be clamped 4787 if (vl > MAX_GAIN_INT) { 4788 vl = MAX_GAIN_INT; 4789 } 4790 if (vr > MAX_GAIN_INT) { 4791 vr = MAX_GAIN_INT; 4792 } 4793 // now apply the cached master volume and stream type volume; 4794 // this is trusted but lacks any synchronization or barrier so may be stale 4795 float v = mCachedVolume; 4796 vl *= v; 4797 vr *= v; 4798 // re-combine into U4.16 4799 vlr = (vr << 16) | (vl & 0xFFFF); 4800 // FIXME look at mute, pause, and stop flags 4801 return vlr; 4802} 4803 4804status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4805{ 4806 if (mState == TERMINATED || mState == PAUSED || 4807 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4808 (mState == STOPPED)))) { 4809 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4810 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4811 event->cancel(); 4812 return INVALID_OPERATION; 4813 } 4814 TrackBase::setSyncEvent(event); 4815 return NO_ERROR; 4816} 4817 4818// timed audio tracks 4819 4820sp<AudioFlinger::PlaybackThread::TimedTrack> 4821AudioFlinger::PlaybackThread::TimedTrack::create( 4822 PlaybackThread *thread, 4823 const sp<Client>& client, 4824 audio_stream_type_t streamType, 4825 uint32_t sampleRate, 4826 audio_format_t format, 4827 uint32_t channelMask, 4828 int frameCount, 4829 const sp<IMemory>& sharedBuffer, 4830 int sessionId) { 4831 if (!client->reserveTimedTrack()) 4832 return NULL; 4833 4834 return new TimedTrack( 4835 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4836 sharedBuffer, sessionId); 4837} 4838 4839AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4840 PlaybackThread *thread, 4841 const sp<Client>& client, 4842 audio_stream_type_t streamType, 4843 uint32_t sampleRate, 4844 audio_format_t format, 4845 uint32_t channelMask, 4846 int frameCount, 4847 const sp<IMemory>& sharedBuffer, 4848 int sessionId) 4849 : Track(thread, client, streamType, sampleRate, format, channelMask, 4850 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4851 mQueueHeadInFlight(false), 4852 mTrimQueueHeadOnRelease(false), 4853 mFramesPendingInQueue(0), 4854 mTimedSilenceBuffer(NULL), 4855 mTimedSilenceBufferSize(0), 4856 mTimedAudioOutputOnTime(false), 4857 mMediaTimeTransformValid(false) 4858{ 4859 LocalClock lc; 4860 mLocalTimeFreq = lc.getLocalFreq(); 4861 4862 mLocalTimeToSampleTransform.a_zero = 0; 4863 mLocalTimeToSampleTransform.b_zero = 0; 4864 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4865 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4866 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4867 &mLocalTimeToSampleTransform.a_to_b_denom); 4868 4869 mMediaTimeToSampleTransform.a_zero = 0; 4870 mMediaTimeToSampleTransform.b_zero = 0; 4871 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4872 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4873 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4874 &mMediaTimeToSampleTransform.a_to_b_denom); 4875} 4876 4877AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4878 mClient->releaseTimedTrack(); 4879 delete [] mTimedSilenceBuffer; 4880} 4881 4882status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4883 size_t size, sp<IMemory>* buffer) { 4884 4885 Mutex::Autolock _l(mTimedBufferQueueLock); 4886 4887 trimTimedBufferQueue_l(); 4888 4889 // lazily initialize the shared memory heap for timed buffers 4890 if (mTimedMemoryDealer == NULL) { 4891 const int kTimedBufferHeapSize = 512 << 10; 4892 4893 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4894 "AudioFlingerTimed"); 4895 if (mTimedMemoryDealer == NULL) 4896 return NO_MEMORY; 4897 } 4898 4899 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4900 if (newBuffer == NULL) { 4901 newBuffer = mTimedMemoryDealer->allocate(size); 4902 if (newBuffer == NULL) 4903 return NO_MEMORY; 4904 } 4905 4906 *buffer = newBuffer; 4907 return NO_ERROR; 4908} 4909 4910// caller must hold mTimedBufferQueueLock 4911void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4912 int64_t mediaTimeNow; 4913 { 4914 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4915 if (!mMediaTimeTransformValid) 4916 return; 4917 4918 int64_t targetTimeNow; 4919 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4920 ? mCCHelper.getCommonTime(&targetTimeNow) 4921 : mCCHelper.getLocalTime(&targetTimeNow); 4922 4923 if (OK != res) 4924 return; 4925 4926 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4927 &mediaTimeNow)) { 4928 return; 4929 } 4930 } 4931 4932 size_t trimEnd; 4933 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4934 int64_t bufEnd; 4935 4936 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4937 // We have a next buffer. Just use its PTS as the PTS of the frame 4938 // following the last frame in this buffer. If the stream is sparse 4939 // (ie, there are deliberate gaps left in the stream which should be 4940 // filled with silence by the TimedAudioTrack), then this can result 4941 // in one extra buffer being left un-trimmed when it could have 4942 // been. In general, this is not typical, and we would rather 4943 // optimized away the TS calculation below for the more common case 4944 // where PTSes are contiguous. 4945 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4946 } else { 4947 // We have no next buffer. Compute the PTS of the frame following 4948 // the last frame in this buffer by computing the duration of of 4949 // this frame in media time units and adding it to the PTS of the 4950 // buffer. 4951 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4952 / mCblk->frameSize; 4953 4954 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4955 &bufEnd)) { 4956 ALOGE("Failed to convert frame count of %lld to media time" 4957 " duration" " (scale factor %d/%u) in %s", 4958 frameCount, 4959 mMediaTimeToSampleTransform.a_to_b_numer, 4960 mMediaTimeToSampleTransform.a_to_b_denom, 4961 __PRETTY_FUNCTION__); 4962 break; 4963 } 4964 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4965 } 4966 4967 if (bufEnd > mediaTimeNow) 4968 break; 4969 4970 // Is the buffer we want to use in the middle of a mix operation right 4971 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4972 // from the mixer which should be coming back shortly. 4973 if (!trimEnd && mQueueHeadInFlight) { 4974 mTrimQueueHeadOnRelease = true; 4975 } 4976 } 4977 4978 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4979 if (trimStart < trimEnd) { 4980 // Update the bookkeeping for framesReady() 4981 for (size_t i = trimStart; i < trimEnd; ++i) { 4982 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4983 } 4984 4985 // Now actually remove the buffers from the queue. 4986 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4987 } 4988} 4989 4990void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4991 const char* logTag) { 4992 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4993 "%s called (reason \"%s\"), but timed buffer queue has no" 4994 " elements to trim.", __FUNCTION__, logTag); 4995 4996 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4997 mTimedBufferQueue.removeAt(0); 4998} 4999 5000void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5001 const TimedBuffer& buf, 5002 const char* logTag) { 5003 uint32_t bufBytes = buf.buffer()->size(); 5004 uint32_t consumedAlready = buf.position(); 5005 5006 ALOG_ASSERT(consumedAlready <= bufBytes, 5007 "Bad bookkeeping while updating frames pending. Timed buffer is" 5008 " only %u bytes long, but claims to have consumed %u" 5009 " bytes. (update reason: \"%s\")", 5010 bufBytes, consumedAlready, logTag); 5011 5012 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 5013 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5014 "Bad bookkeeping while updating frames pending. Should have at" 5015 " least %u queued frames, but we think we have only %u. (update" 5016 " reason: \"%s\")", 5017 bufFrames, mFramesPendingInQueue, logTag); 5018 5019 mFramesPendingInQueue -= bufFrames; 5020} 5021 5022status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5023 const sp<IMemory>& buffer, int64_t pts) { 5024 5025 { 5026 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5027 if (!mMediaTimeTransformValid) 5028 return INVALID_OPERATION; 5029 } 5030 5031 Mutex::Autolock _l(mTimedBufferQueueLock); 5032 5033 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5034 mFramesPendingInQueue += bufFrames; 5035 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5036 5037 return NO_ERROR; 5038} 5039 5040status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5041 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5042 5043 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5044 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5045 target); 5046 5047 if (!(target == TimedAudioTrack::LOCAL_TIME || 5048 target == TimedAudioTrack::COMMON_TIME)) { 5049 return BAD_VALUE; 5050 } 5051 5052 Mutex::Autolock lock(mMediaTimeTransformLock); 5053 mMediaTimeTransform = xform; 5054 mMediaTimeTransformTarget = target; 5055 mMediaTimeTransformValid = true; 5056 5057 return NO_ERROR; 5058} 5059 5060#define min(a, b) ((a) < (b) ? (a) : (b)) 5061 5062// implementation of getNextBuffer for tracks whose buffers have timestamps 5063status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5064 AudioBufferProvider::Buffer* buffer, int64_t pts) 5065{ 5066 if (pts == AudioBufferProvider::kInvalidPTS) { 5067 buffer->raw = 0; 5068 buffer->frameCount = 0; 5069 mTimedAudioOutputOnTime = false; 5070 return INVALID_OPERATION; 5071 } 5072 5073 Mutex::Autolock _l(mTimedBufferQueueLock); 5074 5075 ALOG_ASSERT(!mQueueHeadInFlight, 5076 "getNextBuffer called without releaseBuffer!"); 5077 5078 while (true) { 5079 5080 // if we have no timed buffers, then fail 5081 if (mTimedBufferQueue.isEmpty()) { 5082 buffer->raw = 0; 5083 buffer->frameCount = 0; 5084 return NOT_ENOUGH_DATA; 5085 } 5086 5087 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5088 5089 // calculate the PTS of the head of the timed buffer queue expressed in 5090 // local time 5091 int64_t headLocalPTS; 5092 { 5093 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5094 5095 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5096 5097 if (mMediaTimeTransform.a_to_b_denom == 0) { 5098 // the transform represents a pause, so yield silence 5099 timedYieldSilence_l(buffer->frameCount, buffer); 5100 return NO_ERROR; 5101 } 5102 5103 int64_t transformedPTS; 5104 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5105 &transformedPTS)) { 5106 // the transform failed. this shouldn't happen, but if it does 5107 // then just drop this buffer 5108 ALOGW("timedGetNextBuffer transform failed"); 5109 buffer->raw = 0; 5110 buffer->frameCount = 0; 5111 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5112 return NO_ERROR; 5113 } 5114 5115 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5116 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5117 &headLocalPTS)) { 5118 buffer->raw = 0; 5119 buffer->frameCount = 0; 5120 return INVALID_OPERATION; 5121 } 5122 } else { 5123 headLocalPTS = transformedPTS; 5124 } 5125 } 5126 5127 // adjust the head buffer's PTS to reflect the portion of the head buffer 5128 // that has already been consumed 5129 int64_t effectivePTS = headLocalPTS + 5130 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5131 5132 // Calculate the delta in samples between the head of the input buffer 5133 // queue and the start of the next output buffer that will be written. 5134 // If the transformation fails because of over or underflow, it means 5135 // that the sample's position in the output stream is so far out of 5136 // whack that it should just be dropped. 5137 int64_t sampleDelta; 5138 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5139 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5140 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5141 " mix"); 5142 continue; 5143 } 5144 if (!mLocalTimeToSampleTransform.doForwardTransform( 5145 (effectivePTS - pts) << 32, &sampleDelta)) { 5146 ALOGV("*** too late during sample rate transform: dropped buffer"); 5147 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5148 continue; 5149 } 5150 5151 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5152 " sampleDelta=[%d.%08x]", 5153 head.pts(), head.position(), pts, 5154 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5155 + (sampleDelta >> 32)), 5156 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5157 5158 // if the delta between the ideal placement for the next input sample and 5159 // the current output position is within this threshold, then we will 5160 // concatenate the next input samples to the previous output 5161 const int64_t kSampleContinuityThreshold = 5162 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5163 5164 // if this is the first buffer of audio that we're emitting from this track 5165 // then it should be almost exactly on time. 5166 const int64_t kSampleStartupThreshold = 1LL << 32; 5167 5168 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5169 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5170 // the next input is close enough to being on time, so concatenate it 5171 // with the last output 5172 timedYieldSamples_l(buffer); 5173 5174 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5175 head.position(), buffer->frameCount); 5176 return NO_ERROR; 5177 } 5178 5179 // Looks like our output is not on time. Reset our on timed status. 5180 // Next time we mix samples from our input queue, then should be within 5181 // the StartupThreshold. 5182 mTimedAudioOutputOnTime = false; 5183 if (sampleDelta > 0) { 5184 // the gap between the current output position and the proper start of 5185 // the next input sample is too big, so fill it with silence 5186 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5187 5188 timedYieldSilence_l(framesUntilNextInput, buffer); 5189 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5190 return NO_ERROR; 5191 } else { 5192 // the next input sample is late 5193 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5194 size_t onTimeSamplePosition = 5195 head.position() + lateFrames * mCblk->frameSize; 5196 5197 if (onTimeSamplePosition > head.buffer()->size()) { 5198 // all the remaining samples in the head are too late, so 5199 // drop it and move on 5200 ALOGV("*** too late: dropped buffer"); 5201 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5202 continue; 5203 } else { 5204 // skip over the late samples 5205 head.setPosition(onTimeSamplePosition); 5206 5207 // yield the available samples 5208 timedYieldSamples_l(buffer); 5209 5210 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5211 return NO_ERROR; 5212 } 5213 } 5214 } 5215} 5216 5217// Yield samples from the timed buffer queue head up to the given output 5218// buffer's capacity. 5219// 5220// Caller must hold mTimedBufferQueueLock 5221void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5222 AudioBufferProvider::Buffer* buffer) { 5223 5224 const TimedBuffer& head = mTimedBufferQueue[0]; 5225 5226 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5227 head.position()); 5228 5229 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5230 mCblk->frameSize); 5231 size_t framesRequested = buffer->frameCount; 5232 buffer->frameCount = min(framesLeftInHead, framesRequested); 5233 5234 mQueueHeadInFlight = true; 5235 mTimedAudioOutputOnTime = true; 5236} 5237 5238// Yield samples of silence up to the given output buffer's capacity 5239// 5240// Caller must hold mTimedBufferQueueLock 5241void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5242 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5243 5244 // lazily allocate a buffer filled with silence 5245 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5246 delete [] mTimedSilenceBuffer; 5247 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5248 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5249 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5250 } 5251 5252 buffer->raw = mTimedSilenceBuffer; 5253 size_t framesRequested = buffer->frameCount; 5254 buffer->frameCount = min(numFrames, framesRequested); 5255 5256 mTimedAudioOutputOnTime = false; 5257} 5258 5259// AudioBufferProvider interface 5260void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5261 AudioBufferProvider::Buffer* buffer) { 5262 5263 Mutex::Autolock _l(mTimedBufferQueueLock); 5264 5265 // If the buffer which was just released is part of the buffer at the head 5266 // of the queue, be sure to update the amt of the buffer which has been 5267 // consumed. If the buffer being returned is not part of the head of the 5268 // queue, its either because the buffer is part of the silence buffer, or 5269 // because the head of the timed queue was trimmed after the mixer called 5270 // getNextBuffer but before the mixer called releaseBuffer. 5271 if (buffer->raw == mTimedSilenceBuffer) { 5272 ALOG_ASSERT(!mQueueHeadInFlight, 5273 "Queue head in flight during release of silence buffer!"); 5274 goto done; 5275 } 5276 5277 ALOG_ASSERT(mQueueHeadInFlight, 5278 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5279 " head in flight."); 5280 5281 if (mTimedBufferQueue.size()) { 5282 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5283 5284 void* start = head.buffer()->pointer(); 5285 void* end = reinterpret_cast<void*>( 5286 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5287 + head.buffer()->size()); 5288 5289 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5290 "released buffer not within the head of the timed buffer" 5291 " queue; qHead = [%p, %p], released buffer = %p", 5292 start, end, buffer->raw); 5293 5294 head.setPosition(head.position() + 5295 (buffer->frameCount * mCblk->frameSize)); 5296 mQueueHeadInFlight = false; 5297 5298 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5299 "Bad bookkeeping during releaseBuffer! Should have at" 5300 " least %u queued frames, but we think we have only %u", 5301 buffer->frameCount, mFramesPendingInQueue); 5302 5303 mFramesPendingInQueue -= buffer->frameCount; 5304 5305 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5306 || mTrimQueueHeadOnRelease) { 5307 trimTimedBufferQueueHead_l("releaseBuffer"); 5308 mTrimQueueHeadOnRelease = false; 5309 } 5310 } else { 5311 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5312 " buffers in the timed buffer queue"); 5313 } 5314 5315done: 5316 buffer->raw = 0; 5317 buffer->frameCount = 0; 5318} 5319 5320size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5321 Mutex::Autolock _l(mTimedBufferQueueLock); 5322 return mFramesPendingInQueue; 5323} 5324 5325AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5326 : mPTS(0), mPosition(0) {} 5327 5328AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5329 const sp<IMemory>& buffer, int64_t pts) 5330 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5331 5332// ---------------------------------------------------------------------------- 5333 5334// RecordTrack constructor must be called with AudioFlinger::mLock held 5335AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5336 RecordThread *thread, 5337 const sp<Client>& client, 5338 uint32_t sampleRate, 5339 audio_format_t format, 5340 uint32_t channelMask, 5341 int frameCount, 5342 int sessionId) 5343 : TrackBase(thread, client, sampleRate, format, 5344 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5345 mOverflow(false) 5346{ 5347 if (mCblk != NULL) { 5348 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5349 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5350 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5351 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5352 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5353 } else { 5354 mCblk->frameSize = sizeof(int8_t); 5355 } 5356 } 5357} 5358 5359AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5360{ 5361 sp<ThreadBase> thread = mThread.promote(); 5362 if (thread != 0) { 5363 AudioSystem::releaseInput(thread->id()); 5364 } 5365} 5366 5367// AudioBufferProvider interface 5368status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5369{ 5370 audio_track_cblk_t* cblk = this->cblk(); 5371 uint32_t framesAvail; 5372 uint32_t framesReq = buffer->frameCount; 5373 5374 // Check if last stepServer failed, try to step now 5375 if (mStepServerFailed) { 5376 if (!step()) goto getNextBuffer_exit; 5377 ALOGV("stepServer recovered"); 5378 mStepServerFailed = false; 5379 } 5380 5381 framesAvail = cblk->framesAvailable_l(); 5382 5383 if (CC_LIKELY(framesAvail)) { 5384 uint32_t s = cblk->server; 5385 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5386 5387 if (framesReq > framesAvail) { 5388 framesReq = framesAvail; 5389 } 5390 if (framesReq > bufferEnd - s) { 5391 framesReq = bufferEnd - s; 5392 } 5393 5394 buffer->raw = getBuffer(s, framesReq); 5395 if (buffer->raw == NULL) goto getNextBuffer_exit; 5396 5397 buffer->frameCount = framesReq; 5398 return NO_ERROR; 5399 } 5400 5401getNextBuffer_exit: 5402 buffer->raw = NULL; 5403 buffer->frameCount = 0; 5404 return NOT_ENOUGH_DATA; 5405} 5406 5407status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5408 int triggerSession) 5409{ 5410 sp<ThreadBase> thread = mThread.promote(); 5411 if (thread != 0) { 5412 RecordThread *recordThread = (RecordThread *)thread.get(); 5413 return recordThread->start(this, event, triggerSession); 5414 } else { 5415 return BAD_VALUE; 5416 } 5417} 5418 5419void AudioFlinger::RecordThread::RecordTrack::stop() 5420{ 5421 sp<ThreadBase> thread = mThread.promote(); 5422 if (thread != 0) { 5423 RecordThread *recordThread = (RecordThread *)thread.get(); 5424 recordThread->stop(this); 5425 TrackBase::reset(); 5426 // Force overrun condition to avoid false overrun callback until first data is 5427 // read from buffer 5428 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5429 } 5430} 5431 5432void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5433{ 5434 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5435 (mClient == 0) ? getpid_cached : mClient->pid(), 5436 mFormat, 5437 mChannelMask, 5438 mSessionId, 5439 mFrameCount, 5440 mState, 5441 mCblk->sampleRate, 5442 mCblk->server, 5443 mCblk->user); 5444} 5445 5446 5447// ---------------------------------------------------------------------------- 5448 5449AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5450 PlaybackThread *playbackThread, 5451 DuplicatingThread *sourceThread, 5452 uint32_t sampleRate, 5453 audio_format_t format, 5454 uint32_t channelMask, 5455 int frameCount) 5456 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5457 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5458 mActive(false), mSourceThread(sourceThread) 5459{ 5460 5461 if (mCblk != NULL) { 5462 mCblk->flags |= CBLK_DIRECTION_OUT; 5463 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5464 mOutBuffer.frameCount = 0; 5465 playbackThread->mTracks.add(this); 5466 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5467 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5468 mCblk, mBuffer, mCblk->buffers, 5469 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5470 } else { 5471 ALOGW("Error creating output track on thread %p", playbackThread); 5472 } 5473} 5474 5475AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5476{ 5477 clearBufferQueue(); 5478} 5479 5480status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5481 int triggerSession) 5482{ 5483 status_t status = Track::start(event, triggerSession); 5484 if (status != NO_ERROR) { 5485 return status; 5486 } 5487 5488 mActive = true; 5489 mRetryCount = 127; 5490 return status; 5491} 5492 5493void AudioFlinger::PlaybackThread::OutputTrack::stop() 5494{ 5495 Track::stop(); 5496 clearBufferQueue(); 5497 mOutBuffer.frameCount = 0; 5498 mActive = false; 5499} 5500 5501bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5502{ 5503 Buffer *pInBuffer; 5504 Buffer inBuffer; 5505 uint32_t channelCount = mChannelCount; 5506 bool outputBufferFull = false; 5507 inBuffer.frameCount = frames; 5508 inBuffer.i16 = data; 5509 5510 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5511 5512 if (!mActive && frames != 0) { 5513 start(); 5514 sp<ThreadBase> thread = mThread.promote(); 5515 if (thread != 0) { 5516 MixerThread *mixerThread = (MixerThread *)thread.get(); 5517 if (mCblk->frameCount > frames){ 5518 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5519 uint32_t startFrames = (mCblk->frameCount - frames); 5520 pInBuffer = new Buffer; 5521 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5522 pInBuffer->frameCount = startFrames; 5523 pInBuffer->i16 = pInBuffer->mBuffer; 5524 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5525 mBufferQueue.add(pInBuffer); 5526 } else { 5527 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5528 } 5529 } 5530 } 5531 } 5532 5533 while (waitTimeLeftMs) { 5534 // First write pending buffers, then new data 5535 if (mBufferQueue.size()) { 5536 pInBuffer = mBufferQueue.itemAt(0); 5537 } else { 5538 pInBuffer = &inBuffer; 5539 } 5540 5541 if (pInBuffer->frameCount == 0) { 5542 break; 5543 } 5544 5545 if (mOutBuffer.frameCount == 0) { 5546 mOutBuffer.frameCount = pInBuffer->frameCount; 5547 nsecs_t startTime = systemTime(); 5548 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5549 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5550 outputBufferFull = true; 5551 break; 5552 } 5553 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5554 if (waitTimeLeftMs >= waitTimeMs) { 5555 waitTimeLeftMs -= waitTimeMs; 5556 } else { 5557 waitTimeLeftMs = 0; 5558 } 5559 } 5560 5561 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5562 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5563 mCblk->stepUser(outFrames); 5564 pInBuffer->frameCount -= outFrames; 5565 pInBuffer->i16 += outFrames * channelCount; 5566 mOutBuffer.frameCount -= outFrames; 5567 mOutBuffer.i16 += outFrames * channelCount; 5568 5569 if (pInBuffer->frameCount == 0) { 5570 if (mBufferQueue.size()) { 5571 mBufferQueue.removeAt(0); 5572 delete [] pInBuffer->mBuffer; 5573 delete pInBuffer; 5574 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5575 } else { 5576 break; 5577 } 5578 } 5579 } 5580 5581 // If we could not write all frames, allocate a buffer and queue it for next time. 5582 if (inBuffer.frameCount) { 5583 sp<ThreadBase> thread = mThread.promote(); 5584 if (thread != 0 && !thread->standby()) { 5585 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5586 pInBuffer = new Buffer; 5587 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5588 pInBuffer->frameCount = inBuffer.frameCount; 5589 pInBuffer->i16 = pInBuffer->mBuffer; 5590 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5591 mBufferQueue.add(pInBuffer); 5592 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5593 } else { 5594 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5595 } 5596 } 5597 } 5598 5599 // Calling write() with a 0 length buffer, means that no more data will be written: 5600 // If no more buffers are pending, fill output track buffer to make sure it is started 5601 // by output mixer. 5602 if (frames == 0 && mBufferQueue.size() == 0) { 5603 if (mCblk->user < mCblk->frameCount) { 5604 frames = mCblk->frameCount - mCblk->user; 5605 pInBuffer = new Buffer; 5606 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5607 pInBuffer->frameCount = frames; 5608 pInBuffer->i16 = pInBuffer->mBuffer; 5609 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5610 mBufferQueue.add(pInBuffer); 5611 } else if (mActive) { 5612 stop(); 5613 } 5614 } 5615 5616 return outputBufferFull; 5617} 5618 5619status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5620{ 5621 int active; 5622 status_t result; 5623 audio_track_cblk_t* cblk = mCblk; 5624 uint32_t framesReq = buffer->frameCount; 5625 5626// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5627 buffer->frameCount = 0; 5628 5629 uint32_t framesAvail = cblk->framesAvailable(); 5630 5631 5632 if (framesAvail == 0) { 5633 Mutex::Autolock _l(cblk->lock); 5634 goto start_loop_here; 5635 while (framesAvail == 0) { 5636 active = mActive; 5637 if (CC_UNLIKELY(!active)) { 5638 ALOGV("Not active and NO_MORE_BUFFERS"); 5639 return NO_MORE_BUFFERS; 5640 } 5641 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5642 if (result != NO_ERROR) { 5643 return NO_MORE_BUFFERS; 5644 } 5645 // read the server count again 5646 start_loop_here: 5647 framesAvail = cblk->framesAvailable_l(); 5648 } 5649 } 5650 5651// if (framesAvail < framesReq) { 5652// return NO_MORE_BUFFERS; 5653// } 5654 5655 if (framesReq > framesAvail) { 5656 framesReq = framesAvail; 5657 } 5658 5659 uint32_t u = cblk->user; 5660 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5661 5662 if (framesReq > bufferEnd - u) { 5663 framesReq = bufferEnd - u; 5664 } 5665 5666 buffer->frameCount = framesReq; 5667 buffer->raw = (void *)cblk->buffer(u); 5668 return NO_ERROR; 5669} 5670 5671 5672void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5673{ 5674 size_t size = mBufferQueue.size(); 5675 5676 for (size_t i = 0; i < size; i++) { 5677 Buffer *pBuffer = mBufferQueue.itemAt(i); 5678 delete [] pBuffer->mBuffer; 5679 delete pBuffer; 5680 } 5681 mBufferQueue.clear(); 5682} 5683 5684// ---------------------------------------------------------------------------- 5685 5686AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5687 : RefBase(), 5688 mAudioFlinger(audioFlinger), 5689 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5690 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5691 mPid(pid), 5692 mTimedTrackCount(0) 5693{ 5694 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5695} 5696 5697// Client destructor must be called with AudioFlinger::mLock held 5698AudioFlinger::Client::~Client() 5699{ 5700 mAudioFlinger->removeClient_l(mPid); 5701} 5702 5703sp<MemoryDealer> AudioFlinger::Client::heap() const 5704{ 5705 return mMemoryDealer; 5706} 5707 5708// Reserve one of the limited slots for a timed audio track associated 5709// with this client 5710bool AudioFlinger::Client::reserveTimedTrack() 5711{ 5712 const int kMaxTimedTracksPerClient = 4; 5713 5714 Mutex::Autolock _l(mTimedTrackLock); 5715 5716 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5717 ALOGW("can not create timed track - pid %d has exceeded the limit", 5718 mPid); 5719 return false; 5720 } 5721 5722 mTimedTrackCount++; 5723 return true; 5724} 5725 5726// Release a slot for a timed audio track 5727void AudioFlinger::Client::releaseTimedTrack() 5728{ 5729 Mutex::Autolock _l(mTimedTrackLock); 5730 mTimedTrackCount--; 5731} 5732 5733// ---------------------------------------------------------------------------- 5734 5735AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5736 const sp<IAudioFlingerClient>& client, 5737 pid_t pid) 5738 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5739{ 5740} 5741 5742AudioFlinger::NotificationClient::~NotificationClient() 5743{ 5744} 5745 5746void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5747{ 5748 sp<NotificationClient> keep(this); 5749 mAudioFlinger->removeNotificationClient(mPid); 5750} 5751 5752// ---------------------------------------------------------------------------- 5753 5754AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5755 : BnAudioTrack(), 5756 mTrack(track) 5757{ 5758} 5759 5760AudioFlinger::TrackHandle::~TrackHandle() { 5761 // just stop the track on deletion, associated resources 5762 // will be freed from the main thread once all pending buffers have 5763 // been played. Unless it's not in the active track list, in which 5764 // case we free everything now... 5765 mTrack->destroy(); 5766} 5767 5768sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5769 return mTrack->getCblk(); 5770} 5771 5772status_t AudioFlinger::TrackHandle::start() { 5773 return mTrack->start(); 5774} 5775 5776void AudioFlinger::TrackHandle::stop() { 5777 mTrack->stop(); 5778} 5779 5780void AudioFlinger::TrackHandle::flush() { 5781 mTrack->flush(); 5782} 5783 5784void AudioFlinger::TrackHandle::mute(bool e) { 5785 mTrack->mute(e); 5786} 5787 5788void AudioFlinger::TrackHandle::pause() { 5789 mTrack->pause(); 5790} 5791 5792status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5793{ 5794 return mTrack->attachAuxEffect(EffectId); 5795} 5796 5797status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5798 sp<IMemory>* buffer) { 5799 if (!mTrack->isTimedTrack()) 5800 return INVALID_OPERATION; 5801 5802 PlaybackThread::TimedTrack* tt = 5803 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5804 return tt->allocateTimedBuffer(size, buffer); 5805} 5806 5807status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5808 int64_t pts) { 5809 if (!mTrack->isTimedTrack()) 5810 return INVALID_OPERATION; 5811 5812 PlaybackThread::TimedTrack* tt = 5813 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5814 return tt->queueTimedBuffer(buffer, pts); 5815} 5816 5817status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5818 const LinearTransform& xform, int target) { 5819 5820 if (!mTrack->isTimedTrack()) 5821 return INVALID_OPERATION; 5822 5823 PlaybackThread::TimedTrack* tt = 5824 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5825 return tt->setMediaTimeTransform( 5826 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5827} 5828 5829status_t AudioFlinger::TrackHandle::onTransact( 5830 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5831{ 5832 return BnAudioTrack::onTransact(code, data, reply, flags); 5833} 5834 5835// ---------------------------------------------------------------------------- 5836 5837sp<IAudioRecord> AudioFlinger::openRecord( 5838 pid_t pid, 5839 audio_io_handle_t input, 5840 uint32_t sampleRate, 5841 audio_format_t format, 5842 uint32_t channelMask, 5843 int frameCount, 5844 IAudioFlinger::track_flags_t flags, 5845 int *sessionId, 5846 status_t *status) 5847{ 5848 sp<RecordThread::RecordTrack> recordTrack; 5849 sp<RecordHandle> recordHandle; 5850 sp<Client> client; 5851 status_t lStatus; 5852 RecordThread *thread; 5853 size_t inFrameCount; 5854 int lSessionId; 5855 5856 // check calling permissions 5857 if (!recordingAllowed()) { 5858 lStatus = PERMISSION_DENIED; 5859 goto Exit; 5860 } 5861 5862 // add client to list 5863 { // scope for mLock 5864 Mutex::Autolock _l(mLock); 5865 thread = checkRecordThread_l(input); 5866 if (thread == NULL) { 5867 lStatus = BAD_VALUE; 5868 goto Exit; 5869 } 5870 5871 client = registerPid_l(pid); 5872 5873 // If no audio session id is provided, create one here 5874 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5875 lSessionId = *sessionId; 5876 } else { 5877 lSessionId = nextUniqueId(); 5878 if (sessionId != NULL) { 5879 *sessionId = lSessionId; 5880 } 5881 } 5882 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5883 recordTrack = thread->createRecordTrack_l(client, 5884 sampleRate, 5885 format, 5886 channelMask, 5887 frameCount, 5888 lSessionId, 5889 &lStatus); 5890 } 5891 if (lStatus != NO_ERROR) { 5892 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5893 // destructor is called by the TrackBase destructor with mLock held 5894 client.clear(); 5895 recordTrack.clear(); 5896 goto Exit; 5897 } 5898 5899 // return to handle to client 5900 recordHandle = new RecordHandle(recordTrack); 5901 lStatus = NO_ERROR; 5902 5903Exit: 5904 if (status) { 5905 *status = lStatus; 5906 } 5907 return recordHandle; 5908} 5909 5910// ---------------------------------------------------------------------------- 5911 5912AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5913 : BnAudioRecord(), 5914 mRecordTrack(recordTrack) 5915{ 5916} 5917 5918AudioFlinger::RecordHandle::~RecordHandle() { 5919 stop(); 5920} 5921 5922sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5923 return mRecordTrack->getCblk(); 5924} 5925 5926status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5927 ALOGV("RecordHandle::start()"); 5928 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5929} 5930 5931void AudioFlinger::RecordHandle::stop() { 5932 ALOGV("RecordHandle::stop()"); 5933 mRecordTrack->stop(); 5934} 5935 5936status_t AudioFlinger::RecordHandle::onTransact( 5937 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5938{ 5939 return BnAudioRecord::onTransact(code, data, reply, flags); 5940} 5941 5942// ---------------------------------------------------------------------------- 5943 5944AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5945 AudioStreamIn *input, 5946 uint32_t sampleRate, 5947 uint32_t channels, 5948 audio_io_handle_t id, 5949 uint32_t device) : 5950 ThreadBase(audioFlinger, id, device, RECORD), 5951 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5952 // mRsmpInIndex and mInputBytes set by readInputParameters() 5953 mReqChannelCount(popcount(channels)), 5954 mReqSampleRate(sampleRate) 5955 // mBytesRead is only meaningful while active, and so is cleared in start() 5956 // (but might be better to also clear here for dump?) 5957{ 5958 snprintf(mName, kNameLength, "AudioIn_%X", id); 5959 5960 readInputParameters(); 5961} 5962 5963 5964AudioFlinger::RecordThread::~RecordThread() 5965{ 5966 delete[] mRsmpInBuffer; 5967 delete mResampler; 5968 delete[] mRsmpOutBuffer; 5969} 5970 5971void AudioFlinger::RecordThread::onFirstRef() 5972{ 5973 run(mName, PRIORITY_URGENT_AUDIO); 5974} 5975 5976status_t AudioFlinger::RecordThread::readyToRun() 5977{ 5978 status_t status = initCheck(); 5979 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5980 return status; 5981} 5982 5983bool AudioFlinger::RecordThread::threadLoop() 5984{ 5985 AudioBufferProvider::Buffer buffer; 5986 sp<RecordTrack> activeTrack; 5987 Vector< sp<EffectChain> > effectChains; 5988 5989 nsecs_t lastWarning = 0; 5990 5991 acquireWakeLock(); 5992 5993 // start recording 5994 while (!exitPending()) { 5995 5996 processConfigEvents(); 5997 5998 { // scope for mLock 5999 Mutex::Autolock _l(mLock); 6000 checkForNewParameters_l(); 6001 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6002 if (!mStandby) { 6003 mInput->stream->common.standby(&mInput->stream->common); 6004 mStandby = true; 6005 } 6006 6007 if (exitPending()) break; 6008 6009 releaseWakeLock_l(); 6010 ALOGV("RecordThread: loop stopping"); 6011 // go to sleep 6012 mWaitWorkCV.wait(mLock); 6013 ALOGV("RecordThread: loop starting"); 6014 acquireWakeLock_l(); 6015 continue; 6016 } 6017 if (mActiveTrack != 0) { 6018 if (mActiveTrack->mState == TrackBase::PAUSING) { 6019 if (!mStandby) { 6020 mInput->stream->common.standby(&mInput->stream->common); 6021 mStandby = true; 6022 } 6023 mActiveTrack.clear(); 6024 mStartStopCond.broadcast(); 6025 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6026 if (mReqChannelCount != mActiveTrack->channelCount()) { 6027 mActiveTrack.clear(); 6028 mStartStopCond.broadcast(); 6029 } else if (mBytesRead != 0) { 6030 // record start succeeds only if first read from audio input 6031 // succeeds 6032 if (mBytesRead > 0) { 6033 mActiveTrack->mState = TrackBase::ACTIVE; 6034 } else { 6035 mActiveTrack.clear(); 6036 } 6037 mStartStopCond.broadcast(); 6038 } 6039 mStandby = false; 6040 } 6041 } 6042 lockEffectChains_l(effectChains); 6043 } 6044 6045 if (mActiveTrack != 0) { 6046 if (mActiveTrack->mState != TrackBase::ACTIVE && 6047 mActiveTrack->mState != TrackBase::RESUMING) { 6048 unlockEffectChains(effectChains); 6049 usleep(kRecordThreadSleepUs); 6050 continue; 6051 } 6052 for (size_t i = 0; i < effectChains.size(); i ++) { 6053 effectChains[i]->process_l(); 6054 } 6055 6056 buffer.frameCount = mFrameCount; 6057 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6058 size_t framesOut = buffer.frameCount; 6059 if (mResampler == NULL) { 6060 // no resampling 6061 while (framesOut) { 6062 size_t framesIn = mFrameCount - mRsmpInIndex; 6063 if (framesIn) { 6064 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6065 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6066 if (framesIn > framesOut) 6067 framesIn = framesOut; 6068 mRsmpInIndex += framesIn; 6069 framesOut -= framesIn; 6070 if ((int)mChannelCount == mReqChannelCount || 6071 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6072 memcpy(dst, src, framesIn * mFrameSize); 6073 } else { 6074 int16_t *src16 = (int16_t *)src; 6075 int16_t *dst16 = (int16_t *)dst; 6076 if (mChannelCount == 1) { 6077 while (framesIn--) { 6078 *dst16++ = *src16; 6079 *dst16++ = *src16++; 6080 } 6081 } else { 6082 while (framesIn--) { 6083 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6084 src16 += 2; 6085 } 6086 } 6087 } 6088 } 6089 if (framesOut && mFrameCount == mRsmpInIndex) { 6090 if (framesOut == mFrameCount && 6091 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6092 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6093 framesOut = 0; 6094 } else { 6095 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6096 mRsmpInIndex = 0; 6097 } 6098 if (mBytesRead < 0) { 6099 ALOGE("Error reading audio input"); 6100 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6101 // Force input into standby so that it tries to 6102 // recover at next read attempt 6103 mInput->stream->common.standby(&mInput->stream->common); 6104 usleep(kRecordThreadSleepUs); 6105 } 6106 mRsmpInIndex = mFrameCount; 6107 framesOut = 0; 6108 buffer.frameCount = 0; 6109 } 6110 } 6111 } 6112 } else { 6113 // resampling 6114 6115 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6116 // alter output frame count as if we were expecting stereo samples 6117 if (mChannelCount == 1 && mReqChannelCount == 1) { 6118 framesOut >>= 1; 6119 } 6120 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6121 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6122 // are 32 bit aligned which should be always true. 6123 if (mChannelCount == 2 && mReqChannelCount == 1) { 6124 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6125 // the resampler always outputs stereo samples: do post stereo to mono conversion 6126 int16_t *src = (int16_t *)mRsmpOutBuffer; 6127 int16_t *dst = buffer.i16; 6128 while (framesOut--) { 6129 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6130 src += 2; 6131 } 6132 } else { 6133 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6134 } 6135 6136 } 6137 if (mFramestoDrop == 0) { 6138 mActiveTrack->releaseBuffer(&buffer); 6139 } else { 6140 if (mFramestoDrop > 0) { 6141 mFramestoDrop -= buffer.frameCount; 6142 if (mFramestoDrop <= 0) { 6143 clearSyncStartEvent(); 6144 } 6145 } else { 6146 mFramestoDrop += buffer.frameCount; 6147 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6148 mSyncStartEvent->isCancelled()) { 6149 ALOGW("Synced record %s, session %d, trigger session %d", 6150 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6151 mActiveTrack->sessionId(), 6152 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6153 clearSyncStartEvent(); 6154 } 6155 } 6156 } 6157 mActiveTrack->overflow(); 6158 } 6159 // client isn't retrieving buffers fast enough 6160 else { 6161 if (!mActiveTrack->setOverflow()) { 6162 nsecs_t now = systemTime(); 6163 if ((now - lastWarning) > kWarningThrottleNs) { 6164 ALOGW("RecordThread: buffer overflow"); 6165 lastWarning = now; 6166 } 6167 } 6168 // Release the processor for a while before asking for a new buffer. 6169 // This will give the application more chance to read from the buffer and 6170 // clear the overflow. 6171 usleep(kRecordThreadSleepUs); 6172 } 6173 } 6174 // enable changes in effect chain 6175 unlockEffectChains(effectChains); 6176 effectChains.clear(); 6177 } 6178 6179 if (!mStandby) { 6180 mInput->stream->common.standby(&mInput->stream->common); 6181 } 6182 mActiveTrack.clear(); 6183 6184 mStartStopCond.broadcast(); 6185 6186 releaseWakeLock(); 6187 6188 ALOGV("RecordThread %p exiting", this); 6189 return false; 6190} 6191 6192 6193sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6194 const sp<AudioFlinger::Client>& client, 6195 uint32_t sampleRate, 6196 audio_format_t format, 6197 int channelMask, 6198 int frameCount, 6199 int sessionId, 6200 status_t *status) 6201{ 6202 sp<RecordTrack> track; 6203 status_t lStatus; 6204 6205 lStatus = initCheck(); 6206 if (lStatus != NO_ERROR) { 6207 ALOGE("Audio driver not initialized."); 6208 goto Exit; 6209 } 6210 6211 { // scope for mLock 6212 Mutex::Autolock _l(mLock); 6213 6214 track = new RecordTrack(this, client, sampleRate, 6215 format, channelMask, frameCount, sessionId); 6216 6217 if (track->getCblk() == 0) { 6218 lStatus = NO_MEMORY; 6219 goto Exit; 6220 } 6221 6222 mTrack = track.get(); 6223 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6224 bool suspend = audio_is_bluetooth_sco_device( 6225 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6226 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6227 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6228 } 6229 lStatus = NO_ERROR; 6230 6231Exit: 6232 if (status) { 6233 *status = lStatus; 6234 } 6235 return track; 6236} 6237 6238status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6239 AudioSystem::sync_event_t event, 6240 int triggerSession) 6241{ 6242 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6243 sp<ThreadBase> strongMe = this; 6244 status_t status = NO_ERROR; 6245 6246 if (event == AudioSystem::SYNC_EVENT_NONE) { 6247 clearSyncStartEvent(); 6248 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6249 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6250 triggerSession, 6251 recordTrack->sessionId(), 6252 syncStartEventCallback, 6253 this); 6254 // Sync event can be cancelled by the trigger session if the track is not in a 6255 // compatible state in which case we start record immediately 6256 if (mSyncStartEvent->isCancelled()) { 6257 clearSyncStartEvent(); 6258 } else { 6259 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6260 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6261 } 6262 } 6263 6264 { 6265 AutoMutex lock(mLock); 6266 if (mActiveTrack != 0) { 6267 if (recordTrack != mActiveTrack.get()) { 6268 status = -EBUSY; 6269 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6270 mActiveTrack->mState = TrackBase::ACTIVE; 6271 } 6272 return status; 6273 } 6274 6275 recordTrack->mState = TrackBase::IDLE; 6276 mActiveTrack = recordTrack; 6277 mLock.unlock(); 6278 status_t status = AudioSystem::startInput(mId); 6279 mLock.lock(); 6280 if (status != NO_ERROR) { 6281 mActiveTrack.clear(); 6282 clearSyncStartEvent(); 6283 return status; 6284 } 6285 mRsmpInIndex = mFrameCount; 6286 mBytesRead = 0; 6287 if (mResampler != NULL) { 6288 mResampler->reset(); 6289 } 6290 mActiveTrack->mState = TrackBase::RESUMING; 6291 // signal thread to start 6292 ALOGV("Signal record thread"); 6293 mWaitWorkCV.signal(); 6294 // do not wait for mStartStopCond if exiting 6295 if (exitPending()) { 6296 mActiveTrack.clear(); 6297 status = INVALID_OPERATION; 6298 goto startError; 6299 } 6300 mStartStopCond.wait(mLock); 6301 if (mActiveTrack == 0) { 6302 ALOGV("Record failed to start"); 6303 status = BAD_VALUE; 6304 goto startError; 6305 } 6306 ALOGV("Record started OK"); 6307 return status; 6308 } 6309startError: 6310 AudioSystem::stopInput(mId); 6311 clearSyncStartEvent(); 6312 return status; 6313} 6314 6315void AudioFlinger::RecordThread::clearSyncStartEvent() 6316{ 6317 if (mSyncStartEvent != 0) { 6318 mSyncStartEvent->cancel(); 6319 } 6320 mSyncStartEvent.clear(); 6321 mFramestoDrop = 0; 6322} 6323 6324void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6325{ 6326 sp<SyncEvent> strongEvent = event.promote(); 6327 6328 if (strongEvent != 0) { 6329 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6330 me->handleSyncStartEvent(strongEvent); 6331 } 6332} 6333 6334void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6335{ 6336 if (event == mSyncStartEvent) { 6337 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6338 // from audio HAL 6339 mFramestoDrop = mFrameCount * 2; 6340 } 6341} 6342 6343void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6344 ALOGV("RecordThread::stop"); 6345 sp<ThreadBase> strongMe = this; 6346 { 6347 AutoMutex lock(mLock); 6348 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6349 mActiveTrack->mState = TrackBase::PAUSING; 6350 // do not wait for mStartStopCond if exiting 6351 if (exitPending()) { 6352 return; 6353 } 6354 mStartStopCond.wait(mLock); 6355 // if we have been restarted, recordTrack == mActiveTrack.get() here 6356 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6357 mLock.unlock(); 6358 AudioSystem::stopInput(mId); 6359 mLock.lock(); 6360 ALOGV("Record stopped OK"); 6361 } 6362 } 6363 } 6364} 6365 6366bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6367{ 6368 return false; 6369} 6370 6371status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6372{ 6373 if (!isValidSyncEvent(event)) { 6374 return BAD_VALUE; 6375 } 6376 6377 Mutex::Autolock _l(mLock); 6378 6379 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6380 mTrack->setSyncEvent(event); 6381 return NO_ERROR; 6382 } 6383 return NAME_NOT_FOUND; 6384} 6385 6386status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6387{ 6388 const size_t SIZE = 256; 6389 char buffer[SIZE]; 6390 String8 result; 6391 6392 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6393 result.append(buffer); 6394 6395 if (mActiveTrack != 0) { 6396 result.append("Active Track:\n"); 6397 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6398 mActiveTrack->dump(buffer, SIZE); 6399 result.append(buffer); 6400 6401 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6402 result.append(buffer); 6403 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6404 result.append(buffer); 6405 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6406 result.append(buffer); 6407 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6408 result.append(buffer); 6409 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6410 result.append(buffer); 6411 6412 6413 } else { 6414 result.append("No record client\n"); 6415 } 6416 write(fd, result.string(), result.size()); 6417 6418 dumpBase(fd, args); 6419 dumpEffectChains(fd, args); 6420 6421 return NO_ERROR; 6422} 6423 6424// AudioBufferProvider interface 6425status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6426{ 6427 size_t framesReq = buffer->frameCount; 6428 size_t framesReady = mFrameCount - mRsmpInIndex; 6429 int channelCount; 6430 6431 if (framesReady == 0) { 6432 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6433 if (mBytesRead < 0) { 6434 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6435 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6436 // Force input into standby so that it tries to 6437 // recover at next read attempt 6438 mInput->stream->common.standby(&mInput->stream->common); 6439 usleep(kRecordThreadSleepUs); 6440 } 6441 buffer->raw = NULL; 6442 buffer->frameCount = 0; 6443 return NOT_ENOUGH_DATA; 6444 } 6445 mRsmpInIndex = 0; 6446 framesReady = mFrameCount; 6447 } 6448 6449 if (framesReq > framesReady) { 6450 framesReq = framesReady; 6451 } 6452 6453 if (mChannelCount == 1 && mReqChannelCount == 2) { 6454 channelCount = 1; 6455 } else { 6456 channelCount = 2; 6457 } 6458 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6459 buffer->frameCount = framesReq; 6460 return NO_ERROR; 6461} 6462 6463// AudioBufferProvider interface 6464void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6465{ 6466 mRsmpInIndex += buffer->frameCount; 6467 buffer->frameCount = 0; 6468} 6469 6470bool AudioFlinger::RecordThread::checkForNewParameters_l() 6471{ 6472 bool reconfig = false; 6473 6474 while (!mNewParameters.isEmpty()) { 6475 status_t status = NO_ERROR; 6476 String8 keyValuePair = mNewParameters[0]; 6477 AudioParameter param = AudioParameter(keyValuePair); 6478 int value; 6479 audio_format_t reqFormat = mFormat; 6480 int reqSamplingRate = mReqSampleRate; 6481 int reqChannelCount = mReqChannelCount; 6482 6483 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6484 reqSamplingRate = value; 6485 reconfig = true; 6486 } 6487 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6488 reqFormat = (audio_format_t) value; 6489 reconfig = true; 6490 } 6491 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6492 reqChannelCount = popcount(value); 6493 reconfig = true; 6494 } 6495 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6496 // do not accept frame count changes if tracks are open as the track buffer 6497 // size depends on frame count and correct behavior would not be guaranteed 6498 // if frame count is changed after track creation 6499 if (mActiveTrack != 0) { 6500 status = INVALID_OPERATION; 6501 } else { 6502 reconfig = true; 6503 } 6504 } 6505 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6506 // forward device change to effects that have requested to be 6507 // aware of attached audio device. 6508 for (size_t i = 0; i < mEffectChains.size(); i++) { 6509 mEffectChains[i]->setDevice_l(value); 6510 } 6511 // store input device and output device but do not forward output device to audio HAL. 6512 // Note that status is ignored by the caller for output device 6513 // (see AudioFlinger::setParameters() 6514 if (value & AUDIO_DEVICE_OUT_ALL) { 6515 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6516 status = BAD_VALUE; 6517 } else { 6518 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6519 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6520 if (mTrack != NULL) { 6521 bool suspend = audio_is_bluetooth_sco_device( 6522 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6523 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6524 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6525 } 6526 } 6527 mDevice |= (uint32_t)value; 6528 } 6529 if (status == NO_ERROR) { 6530 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6531 if (status == INVALID_OPERATION) { 6532 mInput->stream->common.standby(&mInput->stream->common); 6533 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6534 keyValuePair.string()); 6535 } 6536 if (reconfig) { 6537 if (status == BAD_VALUE && 6538 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6539 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6540 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6541 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6542 (reqChannelCount <= FCC_2)) { 6543 status = NO_ERROR; 6544 } 6545 if (status == NO_ERROR) { 6546 readInputParameters(); 6547 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6548 } 6549 } 6550 } 6551 6552 mNewParameters.removeAt(0); 6553 6554 mParamStatus = status; 6555 mParamCond.signal(); 6556 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6557 // already timed out waiting for the status and will never signal the condition. 6558 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6559 } 6560 return reconfig; 6561} 6562 6563String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6564{ 6565 char *s; 6566 String8 out_s8 = String8(); 6567 6568 Mutex::Autolock _l(mLock); 6569 if (initCheck() != NO_ERROR) { 6570 return out_s8; 6571 } 6572 6573 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6574 out_s8 = String8(s); 6575 free(s); 6576 return out_s8; 6577} 6578 6579void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6580 AudioSystem::OutputDescriptor desc; 6581 void *param2 = NULL; 6582 6583 switch (event) { 6584 case AudioSystem::INPUT_OPENED: 6585 case AudioSystem::INPUT_CONFIG_CHANGED: 6586 desc.channels = mChannelMask; 6587 desc.samplingRate = mSampleRate; 6588 desc.format = mFormat; 6589 desc.frameCount = mFrameCount; 6590 desc.latency = 0; 6591 param2 = &desc; 6592 break; 6593 6594 case AudioSystem::INPUT_CLOSED: 6595 default: 6596 break; 6597 } 6598 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6599} 6600 6601void AudioFlinger::RecordThread::readInputParameters() 6602{ 6603 delete mRsmpInBuffer; 6604 // mRsmpInBuffer is always assigned a new[] below 6605 delete mRsmpOutBuffer; 6606 mRsmpOutBuffer = NULL; 6607 delete mResampler; 6608 mResampler = NULL; 6609 6610 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6611 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6612 mChannelCount = (uint16_t)popcount(mChannelMask); 6613 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6614 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6615 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6616 mFrameCount = mInputBytes / mFrameSize; 6617 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6618 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6619 6620 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6621 { 6622 int channelCount; 6623 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6624 // stereo to mono post process as the resampler always outputs stereo. 6625 if (mChannelCount == 1 && mReqChannelCount == 2) { 6626 channelCount = 1; 6627 } else { 6628 channelCount = 2; 6629 } 6630 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6631 mResampler->setSampleRate(mSampleRate); 6632 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6633 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6634 6635 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6636 if (mChannelCount == 1 && mReqChannelCount == 1) { 6637 mFrameCount >>= 1; 6638 } 6639 6640 } 6641 mRsmpInIndex = mFrameCount; 6642} 6643 6644unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6645{ 6646 Mutex::Autolock _l(mLock); 6647 if (initCheck() != NO_ERROR) { 6648 return 0; 6649 } 6650 6651 return mInput->stream->get_input_frames_lost(mInput->stream); 6652} 6653 6654uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6655{ 6656 Mutex::Autolock _l(mLock); 6657 uint32_t result = 0; 6658 if (getEffectChain_l(sessionId) != 0) { 6659 result = EFFECT_SESSION; 6660 } 6661 6662 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6663 result |= TRACK_SESSION; 6664 } 6665 6666 return result; 6667} 6668 6669AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6670{ 6671 Mutex::Autolock _l(mLock); 6672 return mTrack; 6673} 6674 6675AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6676{ 6677 Mutex::Autolock _l(mLock); 6678 return mInput; 6679} 6680 6681AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6682{ 6683 Mutex::Autolock _l(mLock); 6684 AudioStreamIn *input = mInput; 6685 mInput = NULL; 6686 return input; 6687} 6688 6689// this method must always be called either with ThreadBase mLock held or inside the thread loop 6690audio_stream_t* AudioFlinger::RecordThread::stream() const 6691{ 6692 if (mInput == NULL) { 6693 return NULL; 6694 } 6695 return &mInput->stream->common; 6696} 6697 6698 6699// ---------------------------------------------------------------------------- 6700 6701audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6702{ 6703 if (!settingsAllowed()) { 6704 return 0; 6705 } 6706 Mutex::Autolock _l(mLock); 6707 return loadHwModule_l(name); 6708} 6709 6710// loadHwModule_l() must be called with AudioFlinger::mLock held 6711audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6712{ 6713 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6714 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6715 ALOGW("loadHwModule() module %s already loaded", name); 6716 return mAudioHwDevs.keyAt(i); 6717 } 6718 } 6719 6720 audio_hw_device_t *dev; 6721 6722 int rc = load_audio_interface(name, &dev); 6723 if (rc) { 6724 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6725 return 0; 6726 } 6727 6728 mHardwareStatus = AUDIO_HW_INIT; 6729 rc = dev->init_check(dev); 6730 mHardwareStatus = AUDIO_HW_IDLE; 6731 if (rc) { 6732 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6733 return 0; 6734 } 6735 6736 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6737 (NULL != dev->set_master_volume)) { 6738 AutoMutex lock(mHardwareLock); 6739 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6740 dev->set_master_volume(dev, mMasterVolume); 6741 mHardwareStatus = AUDIO_HW_IDLE; 6742 } 6743 6744 audio_module_handle_t handle = nextUniqueId(); 6745 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6746 6747 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6748 name, dev->common.module->name, dev->common.module->id, handle); 6749 6750 return handle; 6751 6752} 6753 6754audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6755 audio_devices_t *pDevices, 6756 uint32_t *pSamplingRate, 6757 audio_format_t *pFormat, 6758 audio_channel_mask_t *pChannelMask, 6759 uint32_t *pLatencyMs, 6760 audio_output_flags_t flags) 6761{ 6762 status_t status; 6763 PlaybackThread *thread = NULL; 6764 struct audio_config config = { 6765 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6766 channel_mask: pChannelMask ? *pChannelMask : 0, 6767 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6768 }; 6769 audio_stream_out_t *outStream = NULL; 6770 audio_hw_device_t *outHwDev; 6771 6772 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6773 module, 6774 (pDevices != NULL) ? (int)*pDevices : 0, 6775 config.sample_rate, 6776 config.format, 6777 config.channel_mask, 6778 flags); 6779 6780 if (pDevices == NULL || *pDevices == 0) { 6781 return 0; 6782 } 6783 6784 Mutex::Autolock _l(mLock); 6785 6786 outHwDev = findSuitableHwDev_l(module, *pDevices); 6787 if (outHwDev == NULL) 6788 return 0; 6789 6790 audio_io_handle_t id = nextUniqueId(); 6791 6792 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6793 6794 status = outHwDev->open_output_stream(outHwDev, 6795 id, 6796 *pDevices, 6797 (audio_output_flags_t)flags, 6798 &config, 6799 &outStream); 6800 6801 mHardwareStatus = AUDIO_HW_IDLE; 6802 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6803 outStream, 6804 config.sample_rate, 6805 config.format, 6806 config.channel_mask, 6807 status); 6808 6809 if (status == NO_ERROR && outStream != NULL) { 6810 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6811 6812 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6813 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6814 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6815 thread = new DirectOutputThread(this, output, id, *pDevices); 6816 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6817 } else { 6818 thread = new MixerThread(this, output, id, *pDevices); 6819 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6820 } 6821 mPlaybackThreads.add(id, thread); 6822 6823 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6824 if (pFormat != NULL) *pFormat = config.format; 6825 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6826 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6827 6828 // notify client processes of the new output creation 6829 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6830 6831 // the first primary output opened designates the primary hw device 6832 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6833 ALOGI("Using module %d has the primary audio interface", module); 6834 mPrimaryHardwareDev = outHwDev; 6835 6836 AutoMutex lock(mHardwareLock); 6837 mHardwareStatus = AUDIO_HW_SET_MODE; 6838 outHwDev->set_mode(outHwDev, mMode); 6839 6840 // Determine the level of master volume support the primary audio HAL has, 6841 // and set the initial master volume at the same time. 6842 float initialVolume = 1.0; 6843 mMasterVolumeSupportLvl = MVS_NONE; 6844 6845 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6846 if ((NULL != outHwDev->get_master_volume) && 6847 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6848 mMasterVolumeSupportLvl = MVS_FULL; 6849 } else { 6850 mMasterVolumeSupportLvl = MVS_SETONLY; 6851 initialVolume = 1.0; 6852 } 6853 6854 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6855 if ((NULL == outHwDev->set_master_volume) || 6856 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6857 mMasterVolumeSupportLvl = MVS_NONE; 6858 } 6859 // now that we have a primary device, initialize master volume on other devices 6860 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6861 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6862 6863 if ((dev != mPrimaryHardwareDev) && 6864 (NULL != dev->set_master_volume)) { 6865 dev->set_master_volume(dev, initialVolume); 6866 } 6867 } 6868 mHardwareStatus = AUDIO_HW_IDLE; 6869 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6870 ? initialVolume 6871 : 1.0; 6872 mMasterVolume = initialVolume; 6873 } 6874 return id; 6875 } 6876 6877 return 0; 6878} 6879 6880audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6881 audio_io_handle_t output2) 6882{ 6883 Mutex::Autolock _l(mLock); 6884 MixerThread *thread1 = checkMixerThread_l(output1); 6885 MixerThread *thread2 = checkMixerThread_l(output2); 6886 6887 if (thread1 == NULL || thread2 == NULL) { 6888 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6889 return 0; 6890 } 6891 6892 audio_io_handle_t id = nextUniqueId(); 6893 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6894 thread->addOutputTrack(thread2); 6895 mPlaybackThreads.add(id, thread); 6896 // notify client processes of the new output creation 6897 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6898 return id; 6899} 6900 6901status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6902{ 6903 // keep strong reference on the playback thread so that 6904 // it is not destroyed while exit() is executed 6905 sp<PlaybackThread> thread; 6906 { 6907 Mutex::Autolock _l(mLock); 6908 thread = checkPlaybackThread_l(output); 6909 if (thread == NULL) { 6910 return BAD_VALUE; 6911 } 6912 6913 ALOGV("closeOutput() %d", output); 6914 6915 if (thread->type() == ThreadBase::MIXER) { 6916 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6917 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6918 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6919 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6920 } 6921 } 6922 } 6923 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6924 mPlaybackThreads.removeItem(output); 6925 } 6926 thread->exit(); 6927 // The thread entity (active unit of execution) is no longer running here, 6928 // but the ThreadBase container still exists. 6929 6930 if (thread->type() != ThreadBase::DUPLICATING) { 6931 AudioStreamOut *out = thread->clearOutput(); 6932 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6933 // from now on thread->mOutput is NULL 6934 out->hwDev->close_output_stream(out->hwDev, out->stream); 6935 delete out; 6936 } 6937 return NO_ERROR; 6938} 6939 6940status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6941{ 6942 Mutex::Autolock _l(mLock); 6943 PlaybackThread *thread = checkPlaybackThread_l(output); 6944 6945 if (thread == NULL) { 6946 return BAD_VALUE; 6947 } 6948 6949 ALOGV("suspendOutput() %d", output); 6950 thread->suspend(); 6951 6952 return NO_ERROR; 6953} 6954 6955status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6956{ 6957 Mutex::Autolock _l(mLock); 6958 PlaybackThread *thread = checkPlaybackThread_l(output); 6959 6960 if (thread == NULL) { 6961 return BAD_VALUE; 6962 } 6963 6964 ALOGV("restoreOutput() %d", output); 6965 6966 thread->restore(); 6967 6968 return NO_ERROR; 6969} 6970 6971audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6972 audio_devices_t *pDevices, 6973 uint32_t *pSamplingRate, 6974 audio_format_t *pFormat, 6975 uint32_t *pChannelMask) 6976{ 6977 status_t status; 6978 RecordThread *thread = NULL; 6979 struct audio_config config = { 6980 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6981 channel_mask: pChannelMask ? *pChannelMask : 0, 6982 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6983 }; 6984 uint32_t reqSamplingRate = config.sample_rate; 6985 audio_format_t reqFormat = config.format; 6986 audio_channel_mask_t reqChannels = config.channel_mask; 6987 audio_stream_in_t *inStream = NULL; 6988 audio_hw_device_t *inHwDev; 6989 6990 if (pDevices == NULL || *pDevices == 0) { 6991 return 0; 6992 } 6993 6994 Mutex::Autolock _l(mLock); 6995 6996 inHwDev = findSuitableHwDev_l(module, *pDevices); 6997 if (inHwDev == NULL) 6998 return 0; 6999 7000 audio_io_handle_t id = nextUniqueId(); 7001 7002 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 7003 &inStream); 7004 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7005 inStream, 7006 config.sample_rate, 7007 config.format, 7008 config.channel_mask, 7009 status); 7010 7011 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7012 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7013 // or stereo to mono conversions on 16 bit PCM inputs. 7014 if (status == BAD_VALUE && 7015 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7016 (config.sample_rate <= 2 * reqSamplingRate) && 7017 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7018 ALOGV("openInput() reopening with proposed sampling rate and channels"); 7019 inStream = NULL; 7020 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 7021 } 7022 7023 if (status == NO_ERROR && inStream != NULL) { 7024 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7025 7026 // Start record thread 7027 // RecorThread require both input and output device indication to forward to audio 7028 // pre processing modules 7029 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 7030 thread = new RecordThread(this, 7031 input, 7032 reqSamplingRate, 7033 reqChannels, 7034 id, 7035 device); 7036 mRecordThreads.add(id, thread); 7037 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7038 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7039 if (pFormat != NULL) *pFormat = config.format; 7040 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7041 7042 input->stream->common.standby(&input->stream->common); 7043 7044 // notify client processes of the new input creation 7045 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7046 return id; 7047 } 7048 7049 return 0; 7050} 7051 7052status_t AudioFlinger::closeInput(audio_io_handle_t input) 7053{ 7054 // keep strong reference on the record thread so that 7055 // it is not destroyed while exit() is executed 7056 sp<RecordThread> thread; 7057 { 7058 Mutex::Autolock _l(mLock); 7059 thread = checkRecordThread_l(input); 7060 if (thread == NULL) { 7061 return BAD_VALUE; 7062 } 7063 7064 ALOGV("closeInput() %d", input); 7065 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7066 mRecordThreads.removeItem(input); 7067 } 7068 thread->exit(); 7069 // The thread entity (active unit of execution) is no longer running here, 7070 // but the ThreadBase container still exists. 7071 7072 AudioStreamIn *in = thread->clearInput(); 7073 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7074 // from now on thread->mInput is NULL 7075 in->hwDev->close_input_stream(in->hwDev, in->stream); 7076 delete in; 7077 7078 return NO_ERROR; 7079} 7080 7081status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7082{ 7083 Mutex::Autolock _l(mLock); 7084 MixerThread *dstThread = checkMixerThread_l(output); 7085 if (dstThread == NULL) { 7086 ALOGW("setStreamOutput() bad output id %d", output); 7087 return BAD_VALUE; 7088 } 7089 7090 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7091 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7092 7093 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7094 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7095 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7096 MixerThread *srcThread = (MixerThread *)thread; 7097 srcThread->invalidateTracks(stream); 7098 } 7099 } 7100 7101 return NO_ERROR; 7102} 7103 7104 7105int AudioFlinger::newAudioSessionId() 7106{ 7107 return nextUniqueId(); 7108} 7109 7110void AudioFlinger::acquireAudioSessionId(int audioSession) 7111{ 7112 Mutex::Autolock _l(mLock); 7113 pid_t caller = IPCThreadState::self()->getCallingPid(); 7114 ALOGV("acquiring %d from %d", audioSession, caller); 7115 size_t num = mAudioSessionRefs.size(); 7116 for (size_t i = 0; i< num; i++) { 7117 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7118 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7119 ref->mCnt++; 7120 ALOGV(" incremented refcount to %d", ref->mCnt); 7121 return; 7122 } 7123 } 7124 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7125 ALOGV(" added new entry for %d", audioSession); 7126} 7127 7128void AudioFlinger::releaseAudioSessionId(int audioSession) 7129{ 7130 Mutex::Autolock _l(mLock); 7131 pid_t caller = IPCThreadState::self()->getCallingPid(); 7132 ALOGV("releasing %d from %d", audioSession, caller); 7133 size_t num = mAudioSessionRefs.size(); 7134 for (size_t i = 0; i< num; i++) { 7135 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7136 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7137 ref->mCnt--; 7138 ALOGV(" decremented refcount to %d", ref->mCnt); 7139 if (ref->mCnt == 0) { 7140 mAudioSessionRefs.removeAt(i); 7141 delete ref; 7142 purgeStaleEffects_l(); 7143 } 7144 return; 7145 } 7146 } 7147 ALOGW("session id %d not found for pid %d", audioSession, caller); 7148} 7149 7150void AudioFlinger::purgeStaleEffects_l() { 7151 7152 ALOGV("purging stale effects"); 7153 7154 Vector< sp<EffectChain> > chains; 7155 7156 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7157 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7158 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7159 sp<EffectChain> ec = t->mEffectChains[j]; 7160 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7161 chains.push(ec); 7162 } 7163 } 7164 } 7165 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7166 sp<RecordThread> t = mRecordThreads.valueAt(i); 7167 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7168 sp<EffectChain> ec = t->mEffectChains[j]; 7169 chains.push(ec); 7170 } 7171 } 7172 7173 for (size_t i = 0; i < chains.size(); i++) { 7174 sp<EffectChain> ec = chains[i]; 7175 int sessionid = ec->sessionId(); 7176 sp<ThreadBase> t = ec->mThread.promote(); 7177 if (t == 0) { 7178 continue; 7179 } 7180 size_t numsessionrefs = mAudioSessionRefs.size(); 7181 bool found = false; 7182 for (size_t k = 0; k < numsessionrefs; k++) { 7183 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7184 if (ref->mSessionid == sessionid) { 7185 ALOGV(" session %d still exists for %d with %d refs", 7186 sessionid, ref->mPid, ref->mCnt); 7187 found = true; 7188 break; 7189 } 7190 } 7191 if (!found) { 7192 // remove all effects from the chain 7193 while (ec->mEffects.size()) { 7194 sp<EffectModule> effect = ec->mEffects[0]; 7195 effect->unPin(); 7196 Mutex::Autolock _l (t->mLock); 7197 t->removeEffect_l(effect); 7198 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7199 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7200 if (handle != 0) { 7201 handle->mEffect.clear(); 7202 if (handle->mHasControl && handle->mEnabled) { 7203 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7204 } 7205 } 7206 } 7207 AudioSystem::unregisterEffect(effect->id()); 7208 } 7209 } 7210 } 7211 return; 7212} 7213 7214// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7215AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7216{ 7217 return mPlaybackThreads.valueFor(output).get(); 7218} 7219 7220// checkMixerThread_l() must be called with AudioFlinger::mLock held 7221AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7222{ 7223 PlaybackThread *thread = checkPlaybackThread_l(output); 7224 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7225} 7226 7227// checkRecordThread_l() must be called with AudioFlinger::mLock held 7228AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7229{ 7230 return mRecordThreads.valueFor(input).get(); 7231} 7232 7233uint32_t AudioFlinger::nextUniqueId() 7234{ 7235 return android_atomic_inc(&mNextUniqueId); 7236} 7237 7238AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7239{ 7240 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7241 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7242 AudioStreamOut *output = thread->getOutput(); 7243 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7244 return thread; 7245 } 7246 } 7247 return NULL; 7248} 7249 7250uint32_t AudioFlinger::primaryOutputDevice_l() const 7251{ 7252 PlaybackThread *thread = primaryPlaybackThread_l(); 7253 7254 if (thread == NULL) { 7255 return 0; 7256 } 7257 7258 return thread->device(); 7259} 7260 7261sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7262 int triggerSession, 7263 int listenerSession, 7264 sync_event_callback_t callBack, 7265 void *cookie) 7266{ 7267 Mutex::Autolock _l(mLock); 7268 7269 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7270 status_t playStatus = NAME_NOT_FOUND; 7271 status_t recStatus = NAME_NOT_FOUND; 7272 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7273 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7274 if (playStatus == NO_ERROR) { 7275 return event; 7276 } 7277 } 7278 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7279 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7280 if (recStatus == NO_ERROR) { 7281 return event; 7282 } 7283 } 7284 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7285 mPendingSyncEvents.add(event); 7286 } else { 7287 ALOGV("createSyncEvent() invalid event %d", event->type()); 7288 event.clear(); 7289 } 7290 return event; 7291} 7292 7293// ---------------------------------------------------------------------------- 7294// Effect management 7295// ---------------------------------------------------------------------------- 7296 7297 7298status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7299{ 7300 Mutex::Autolock _l(mLock); 7301 return EffectQueryNumberEffects(numEffects); 7302} 7303 7304status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7305{ 7306 Mutex::Autolock _l(mLock); 7307 return EffectQueryEffect(index, descriptor); 7308} 7309 7310status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7311 effect_descriptor_t *descriptor) const 7312{ 7313 Mutex::Autolock _l(mLock); 7314 return EffectGetDescriptor(pUuid, descriptor); 7315} 7316 7317 7318sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7319 effect_descriptor_t *pDesc, 7320 const sp<IEffectClient>& effectClient, 7321 int32_t priority, 7322 audio_io_handle_t io, 7323 int sessionId, 7324 status_t *status, 7325 int *id, 7326 int *enabled) 7327{ 7328 status_t lStatus = NO_ERROR; 7329 sp<EffectHandle> handle; 7330 effect_descriptor_t desc; 7331 7332 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7333 pid, effectClient.get(), priority, sessionId, io); 7334 7335 if (pDesc == NULL) { 7336 lStatus = BAD_VALUE; 7337 goto Exit; 7338 } 7339 7340 // check audio settings permission for global effects 7341 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7342 lStatus = PERMISSION_DENIED; 7343 goto Exit; 7344 } 7345 7346 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7347 // that can only be created by audio policy manager (running in same process) 7348 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7349 lStatus = PERMISSION_DENIED; 7350 goto Exit; 7351 } 7352 7353 if (io == 0) { 7354 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7355 // output must be specified by AudioPolicyManager when using session 7356 // AUDIO_SESSION_OUTPUT_STAGE 7357 lStatus = BAD_VALUE; 7358 goto Exit; 7359 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7360 // if the output returned by getOutputForEffect() is removed before we lock the 7361 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7362 // and we will exit safely 7363 io = AudioSystem::getOutputForEffect(&desc); 7364 } 7365 } 7366 7367 { 7368 Mutex::Autolock _l(mLock); 7369 7370 7371 if (!EffectIsNullUuid(&pDesc->uuid)) { 7372 // if uuid is specified, request effect descriptor 7373 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7374 if (lStatus < 0) { 7375 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7376 goto Exit; 7377 } 7378 } else { 7379 // if uuid is not specified, look for an available implementation 7380 // of the required type in effect factory 7381 if (EffectIsNullUuid(&pDesc->type)) { 7382 ALOGW("createEffect() no effect type"); 7383 lStatus = BAD_VALUE; 7384 goto Exit; 7385 } 7386 uint32_t numEffects = 0; 7387 effect_descriptor_t d; 7388 d.flags = 0; // prevent compiler warning 7389 bool found = false; 7390 7391 lStatus = EffectQueryNumberEffects(&numEffects); 7392 if (lStatus < 0) { 7393 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7394 goto Exit; 7395 } 7396 for (uint32_t i = 0; i < numEffects; i++) { 7397 lStatus = EffectQueryEffect(i, &desc); 7398 if (lStatus < 0) { 7399 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7400 continue; 7401 } 7402 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7403 // If matching type found save effect descriptor. If the session is 7404 // 0 and the effect is not auxiliary, continue enumeration in case 7405 // an auxiliary version of this effect type is available 7406 found = true; 7407 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7408 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7409 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7410 break; 7411 } 7412 } 7413 } 7414 if (!found) { 7415 lStatus = BAD_VALUE; 7416 ALOGW("createEffect() effect not found"); 7417 goto Exit; 7418 } 7419 // For same effect type, chose auxiliary version over insert version if 7420 // connect to output mix (Compliance to OpenSL ES) 7421 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7422 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7423 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7424 } 7425 } 7426 7427 // Do not allow auxiliary effects on a session different from 0 (output mix) 7428 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7429 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7430 lStatus = INVALID_OPERATION; 7431 goto Exit; 7432 } 7433 7434 // check recording permission for visualizer 7435 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7436 !recordingAllowed()) { 7437 lStatus = PERMISSION_DENIED; 7438 goto Exit; 7439 } 7440 7441 // return effect descriptor 7442 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7443 7444 // If output is not specified try to find a matching audio session ID in one of the 7445 // output threads. 7446 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7447 // because of code checking output when entering the function. 7448 // Note: io is never 0 when creating an effect on an input 7449 if (io == 0) { 7450 // look for the thread where the specified audio session is present 7451 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7452 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7453 io = mPlaybackThreads.keyAt(i); 7454 break; 7455 } 7456 } 7457 if (io == 0) { 7458 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7459 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7460 io = mRecordThreads.keyAt(i); 7461 break; 7462 } 7463 } 7464 } 7465 // If no output thread contains the requested session ID, default to 7466 // first output. The effect chain will be moved to the correct output 7467 // thread when a track with the same session ID is created 7468 if (io == 0 && mPlaybackThreads.size()) { 7469 io = mPlaybackThreads.keyAt(0); 7470 } 7471 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7472 } 7473 ThreadBase *thread = checkRecordThread_l(io); 7474 if (thread == NULL) { 7475 thread = checkPlaybackThread_l(io); 7476 if (thread == NULL) { 7477 ALOGE("createEffect() unknown output thread"); 7478 lStatus = BAD_VALUE; 7479 goto Exit; 7480 } 7481 } 7482 7483 sp<Client> client = registerPid_l(pid); 7484 7485 // create effect on selected output thread 7486 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7487 &desc, enabled, &lStatus); 7488 if (handle != 0 && id != NULL) { 7489 *id = handle->id(); 7490 } 7491 } 7492 7493Exit: 7494 if (status != NULL) { 7495 *status = lStatus; 7496 } 7497 return handle; 7498} 7499 7500status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7501 audio_io_handle_t dstOutput) 7502{ 7503 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7504 sessionId, srcOutput, dstOutput); 7505 Mutex::Autolock _l(mLock); 7506 if (srcOutput == dstOutput) { 7507 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7508 return NO_ERROR; 7509 } 7510 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7511 if (srcThread == NULL) { 7512 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7513 return BAD_VALUE; 7514 } 7515 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7516 if (dstThread == NULL) { 7517 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7518 return BAD_VALUE; 7519 } 7520 7521 Mutex::Autolock _dl(dstThread->mLock); 7522 Mutex::Autolock _sl(srcThread->mLock); 7523 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7524 7525 return NO_ERROR; 7526} 7527 7528// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7529status_t AudioFlinger::moveEffectChain_l(int sessionId, 7530 AudioFlinger::PlaybackThread *srcThread, 7531 AudioFlinger::PlaybackThread *dstThread, 7532 bool reRegister) 7533{ 7534 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7535 sessionId, srcThread, dstThread); 7536 7537 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7538 if (chain == 0) { 7539 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7540 sessionId, srcThread); 7541 return INVALID_OPERATION; 7542 } 7543 7544 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7545 // so that a new chain is created with correct parameters when first effect is added. This is 7546 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7547 // removed. 7548 srcThread->removeEffectChain_l(chain); 7549 7550 // transfer all effects one by one so that new effect chain is created on new thread with 7551 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7552 audio_io_handle_t dstOutput = dstThread->id(); 7553 sp<EffectChain> dstChain; 7554 uint32_t strategy = 0; // prevent compiler warning 7555 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7556 while (effect != 0) { 7557 srcThread->removeEffect_l(effect); 7558 dstThread->addEffect_l(effect); 7559 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7560 if (effect->state() == EffectModule::ACTIVE || 7561 effect->state() == EffectModule::STOPPING) { 7562 effect->start(); 7563 } 7564 // if the move request is not received from audio policy manager, the effect must be 7565 // re-registered with the new strategy and output 7566 if (dstChain == 0) { 7567 dstChain = effect->chain().promote(); 7568 if (dstChain == 0) { 7569 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7570 srcThread->addEffect_l(effect); 7571 return NO_INIT; 7572 } 7573 strategy = dstChain->strategy(); 7574 } 7575 if (reRegister) { 7576 AudioSystem::unregisterEffect(effect->id()); 7577 AudioSystem::registerEffect(&effect->desc(), 7578 dstOutput, 7579 strategy, 7580 sessionId, 7581 effect->id()); 7582 } 7583 effect = chain->getEffectFromId_l(0); 7584 } 7585 7586 return NO_ERROR; 7587} 7588 7589 7590// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7591sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7592 const sp<AudioFlinger::Client>& client, 7593 const sp<IEffectClient>& effectClient, 7594 int32_t priority, 7595 int sessionId, 7596 effect_descriptor_t *desc, 7597 int *enabled, 7598 status_t *status 7599 ) 7600{ 7601 sp<EffectModule> effect; 7602 sp<EffectHandle> handle; 7603 status_t lStatus; 7604 sp<EffectChain> chain; 7605 bool chainCreated = false; 7606 bool effectCreated = false; 7607 bool effectRegistered = false; 7608 7609 lStatus = initCheck(); 7610 if (lStatus != NO_ERROR) { 7611 ALOGW("createEffect_l() Audio driver not initialized."); 7612 goto Exit; 7613 } 7614 7615 // Do not allow effects with session ID 0 on direct output or duplicating threads 7616 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7617 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7618 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7619 desc->name, sessionId); 7620 lStatus = BAD_VALUE; 7621 goto Exit; 7622 } 7623 // Only Pre processor effects are allowed on input threads and only on input threads 7624 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7625 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7626 desc->name, desc->flags, mType); 7627 lStatus = BAD_VALUE; 7628 goto Exit; 7629 } 7630 7631 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7632 7633 { // scope for mLock 7634 Mutex::Autolock _l(mLock); 7635 7636 // check for existing effect chain with the requested audio session 7637 chain = getEffectChain_l(sessionId); 7638 if (chain == 0) { 7639 // create a new chain for this session 7640 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7641 chain = new EffectChain(this, sessionId); 7642 addEffectChain_l(chain); 7643 chain->setStrategy(getStrategyForSession_l(sessionId)); 7644 chainCreated = true; 7645 } else { 7646 effect = chain->getEffectFromDesc_l(desc); 7647 } 7648 7649 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7650 7651 if (effect == 0) { 7652 int id = mAudioFlinger->nextUniqueId(); 7653 // Check CPU and memory usage 7654 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7655 if (lStatus != NO_ERROR) { 7656 goto Exit; 7657 } 7658 effectRegistered = true; 7659 // create a new effect module if none present in the chain 7660 effect = new EffectModule(this, chain, desc, id, sessionId); 7661 lStatus = effect->status(); 7662 if (lStatus != NO_ERROR) { 7663 goto Exit; 7664 } 7665 lStatus = chain->addEffect_l(effect); 7666 if (lStatus != NO_ERROR) { 7667 goto Exit; 7668 } 7669 effectCreated = true; 7670 7671 effect->setDevice(mDevice); 7672 effect->setMode(mAudioFlinger->getMode()); 7673 } 7674 // create effect handle and connect it to effect module 7675 handle = new EffectHandle(effect, client, effectClient, priority); 7676 lStatus = effect->addHandle(handle); 7677 if (enabled != NULL) { 7678 *enabled = (int)effect->isEnabled(); 7679 } 7680 } 7681 7682Exit: 7683 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7684 Mutex::Autolock _l(mLock); 7685 if (effectCreated) { 7686 chain->removeEffect_l(effect); 7687 } 7688 if (effectRegistered) { 7689 AudioSystem::unregisterEffect(effect->id()); 7690 } 7691 if (chainCreated) { 7692 removeEffectChain_l(chain); 7693 } 7694 handle.clear(); 7695 } 7696 7697 if (status != NULL) { 7698 *status = lStatus; 7699 } 7700 return handle; 7701} 7702 7703sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7704{ 7705 Mutex::Autolock _l(mLock); 7706 return getEffect_l(sessionId, effectId); 7707} 7708 7709sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7710{ 7711 sp<EffectChain> chain = getEffectChain_l(sessionId); 7712 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7713} 7714 7715// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7716// PlaybackThread::mLock held 7717status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7718{ 7719 // check for existing effect chain with the requested audio session 7720 int sessionId = effect->sessionId(); 7721 sp<EffectChain> chain = getEffectChain_l(sessionId); 7722 bool chainCreated = false; 7723 7724 if (chain == 0) { 7725 // create a new chain for this session 7726 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7727 chain = new EffectChain(this, sessionId); 7728 addEffectChain_l(chain); 7729 chain->setStrategy(getStrategyForSession_l(sessionId)); 7730 chainCreated = true; 7731 } 7732 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7733 7734 if (chain->getEffectFromId_l(effect->id()) != 0) { 7735 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7736 this, effect->desc().name, chain.get()); 7737 return BAD_VALUE; 7738 } 7739 7740 status_t status = chain->addEffect_l(effect); 7741 if (status != NO_ERROR) { 7742 if (chainCreated) { 7743 removeEffectChain_l(chain); 7744 } 7745 return status; 7746 } 7747 7748 effect->setDevice(mDevice); 7749 effect->setMode(mAudioFlinger->getMode()); 7750 return NO_ERROR; 7751} 7752 7753void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7754 7755 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7756 effect_descriptor_t desc = effect->desc(); 7757 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7758 detachAuxEffect_l(effect->id()); 7759 } 7760 7761 sp<EffectChain> chain = effect->chain().promote(); 7762 if (chain != 0) { 7763 // remove effect chain if removing last effect 7764 if (chain->removeEffect_l(effect) == 0) { 7765 removeEffectChain_l(chain); 7766 } 7767 } else { 7768 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7769 } 7770} 7771 7772void AudioFlinger::ThreadBase::lockEffectChains_l( 7773 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7774{ 7775 effectChains = mEffectChains; 7776 for (size_t i = 0; i < mEffectChains.size(); i++) { 7777 mEffectChains[i]->lock(); 7778 } 7779} 7780 7781void AudioFlinger::ThreadBase::unlockEffectChains( 7782 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7783{ 7784 for (size_t i = 0; i < effectChains.size(); i++) { 7785 effectChains[i]->unlock(); 7786 } 7787} 7788 7789sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7790{ 7791 Mutex::Autolock _l(mLock); 7792 return getEffectChain_l(sessionId); 7793} 7794 7795sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7796{ 7797 size_t size = mEffectChains.size(); 7798 for (size_t i = 0; i < size; i++) { 7799 if (mEffectChains[i]->sessionId() == sessionId) { 7800 return mEffectChains[i]; 7801 } 7802 } 7803 return 0; 7804} 7805 7806void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7807{ 7808 Mutex::Autolock _l(mLock); 7809 size_t size = mEffectChains.size(); 7810 for (size_t i = 0; i < size; i++) { 7811 mEffectChains[i]->setMode_l(mode); 7812 } 7813} 7814 7815void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7816 const wp<EffectHandle>& handle, 7817 bool unpinIfLast) { 7818 7819 Mutex::Autolock _l(mLock); 7820 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7821 // delete the effect module if removing last handle on it 7822 if (effect->removeHandle(handle) == 0) { 7823 if (!effect->isPinned() || unpinIfLast) { 7824 removeEffect_l(effect); 7825 AudioSystem::unregisterEffect(effect->id()); 7826 } 7827 } 7828} 7829 7830status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7831{ 7832 int session = chain->sessionId(); 7833 int16_t *buffer = mMixBuffer; 7834 bool ownsBuffer = false; 7835 7836 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7837 if (session > 0) { 7838 // Only one effect chain can be present in direct output thread and it uses 7839 // the mix buffer as input 7840 if (mType != DIRECT) { 7841 size_t numSamples = mNormalFrameCount * mChannelCount; 7842 buffer = new int16_t[numSamples]; 7843 memset(buffer, 0, numSamples * sizeof(int16_t)); 7844 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7845 ownsBuffer = true; 7846 } 7847 7848 // Attach all tracks with same session ID to this chain. 7849 for (size_t i = 0; i < mTracks.size(); ++i) { 7850 sp<Track> track = mTracks[i]; 7851 if (session == track->sessionId()) { 7852 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7853 track->setMainBuffer(buffer); 7854 chain->incTrackCnt(); 7855 } 7856 } 7857 7858 // indicate all active tracks in the chain 7859 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7860 sp<Track> track = mActiveTracks[i].promote(); 7861 if (track == 0) continue; 7862 if (session == track->sessionId()) { 7863 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7864 chain->incActiveTrackCnt(); 7865 } 7866 } 7867 } 7868 7869 chain->setInBuffer(buffer, ownsBuffer); 7870 chain->setOutBuffer(mMixBuffer); 7871 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7872 // chains list in order to be processed last as it contains output stage effects 7873 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7874 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7875 // after track specific effects and before output stage 7876 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7877 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7878 // Effect chain for other sessions are inserted at beginning of effect 7879 // chains list to be processed before output mix effects. Relative order between other 7880 // sessions is not important 7881 size_t size = mEffectChains.size(); 7882 size_t i = 0; 7883 for (i = 0; i < size; i++) { 7884 if (mEffectChains[i]->sessionId() < session) break; 7885 } 7886 mEffectChains.insertAt(chain, i); 7887 checkSuspendOnAddEffectChain_l(chain); 7888 7889 return NO_ERROR; 7890} 7891 7892size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7893{ 7894 int session = chain->sessionId(); 7895 7896 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7897 7898 for (size_t i = 0; i < mEffectChains.size(); i++) { 7899 if (chain == mEffectChains[i]) { 7900 mEffectChains.removeAt(i); 7901 // detach all active tracks from the chain 7902 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7903 sp<Track> track = mActiveTracks[i].promote(); 7904 if (track == 0) continue; 7905 if (session == track->sessionId()) { 7906 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7907 chain.get(), session); 7908 chain->decActiveTrackCnt(); 7909 } 7910 } 7911 7912 // detach all tracks with same session ID from this chain 7913 for (size_t i = 0; i < mTracks.size(); ++i) { 7914 sp<Track> track = mTracks[i]; 7915 if (session == track->sessionId()) { 7916 track->setMainBuffer(mMixBuffer); 7917 chain->decTrackCnt(); 7918 } 7919 } 7920 break; 7921 } 7922 } 7923 return mEffectChains.size(); 7924} 7925 7926status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7927 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7928{ 7929 Mutex::Autolock _l(mLock); 7930 return attachAuxEffect_l(track, EffectId); 7931} 7932 7933status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7934 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7935{ 7936 status_t status = NO_ERROR; 7937 7938 if (EffectId == 0) { 7939 track->setAuxBuffer(0, NULL); 7940 } else { 7941 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7942 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7943 if (effect != 0) { 7944 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7945 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7946 } else { 7947 status = INVALID_OPERATION; 7948 } 7949 } else { 7950 status = BAD_VALUE; 7951 } 7952 } 7953 return status; 7954} 7955 7956void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7957{ 7958 for (size_t i = 0; i < mTracks.size(); ++i) { 7959 sp<Track> track = mTracks[i]; 7960 if (track->auxEffectId() == effectId) { 7961 attachAuxEffect_l(track, 0); 7962 } 7963 } 7964} 7965 7966status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7967{ 7968 // only one chain per input thread 7969 if (mEffectChains.size() != 0) { 7970 return INVALID_OPERATION; 7971 } 7972 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7973 7974 chain->setInBuffer(NULL); 7975 chain->setOutBuffer(NULL); 7976 7977 checkSuspendOnAddEffectChain_l(chain); 7978 7979 mEffectChains.add(chain); 7980 7981 return NO_ERROR; 7982} 7983 7984size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7985{ 7986 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7987 ALOGW_IF(mEffectChains.size() != 1, 7988 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7989 chain.get(), mEffectChains.size(), this); 7990 if (mEffectChains.size() == 1) { 7991 mEffectChains.removeAt(0); 7992 } 7993 return 0; 7994} 7995 7996// ---------------------------------------------------------------------------- 7997// EffectModule implementation 7998// ---------------------------------------------------------------------------- 7999 8000#undef LOG_TAG 8001#define LOG_TAG "AudioFlinger::EffectModule" 8002 8003AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8004 const wp<AudioFlinger::EffectChain>& chain, 8005 effect_descriptor_t *desc, 8006 int id, 8007 int sessionId) 8008 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 8009 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 8010{ 8011 ALOGV("Constructor %p", this); 8012 int lStatus; 8013 if (thread == NULL) { 8014 return; 8015 } 8016 8017 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 8018 8019 // create effect engine from effect factory 8020 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8021 8022 if (mStatus != NO_ERROR) { 8023 return; 8024 } 8025 lStatus = init(); 8026 if (lStatus < 0) { 8027 mStatus = lStatus; 8028 goto Error; 8029 } 8030 8031 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 8032 mPinned = true; 8033 } 8034 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8035 return; 8036Error: 8037 EffectRelease(mEffectInterface); 8038 mEffectInterface = NULL; 8039 ALOGV("Constructor Error %d", mStatus); 8040} 8041 8042AudioFlinger::EffectModule::~EffectModule() 8043{ 8044 ALOGV("Destructor %p", this); 8045 if (mEffectInterface != NULL) { 8046 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8047 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8048 sp<ThreadBase> thread = mThread.promote(); 8049 if (thread != 0) { 8050 audio_stream_t *stream = thread->stream(); 8051 if (stream != NULL) { 8052 stream->remove_audio_effect(stream, mEffectInterface); 8053 } 8054 } 8055 } 8056 // release effect engine 8057 EffectRelease(mEffectInterface); 8058 } 8059} 8060 8061status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8062{ 8063 status_t status; 8064 8065 Mutex::Autolock _l(mLock); 8066 int priority = handle->priority(); 8067 size_t size = mHandles.size(); 8068 sp<EffectHandle> h; 8069 size_t i; 8070 for (i = 0; i < size; i++) { 8071 h = mHandles[i].promote(); 8072 if (h == 0) continue; 8073 if (h->priority() <= priority) break; 8074 } 8075 // if inserted in first place, move effect control from previous owner to this handle 8076 if (i == 0) { 8077 bool enabled = false; 8078 if (h != 0) { 8079 enabled = h->enabled(); 8080 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8081 } 8082 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8083 status = NO_ERROR; 8084 } else { 8085 status = ALREADY_EXISTS; 8086 } 8087 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8088 mHandles.insertAt(handle, i); 8089 return status; 8090} 8091 8092size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8093{ 8094 Mutex::Autolock _l(mLock); 8095 size_t size = mHandles.size(); 8096 size_t i; 8097 for (i = 0; i < size; i++) { 8098 if (mHandles[i] == handle) break; 8099 } 8100 if (i == size) { 8101 return size; 8102 } 8103 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8104 8105 bool enabled = false; 8106 EffectHandle *hdl = handle.unsafe_get(); 8107 if (hdl != NULL) { 8108 ALOGV("removeHandle() unsafe_get OK"); 8109 enabled = hdl->enabled(); 8110 } 8111 mHandles.removeAt(i); 8112 size = mHandles.size(); 8113 // if removed from first place, move effect control from this handle to next in line 8114 if (i == 0 && size != 0) { 8115 sp<EffectHandle> h = mHandles[0].promote(); 8116 if (h != 0) { 8117 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8118 } 8119 } 8120 8121 // Prevent calls to process() and other functions on effect interface from now on. 8122 // The effect engine will be released by the destructor when the last strong reference on 8123 // this object is released which can happen after next process is called. 8124 if (size == 0 && !mPinned) { 8125 mState = DESTROYED; 8126 } 8127 8128 return size; 8129} 8130 8131sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8132{ 8133 Mutex::Autolock _l(mLock); 8134 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8135} 8136 8137void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8138{ 8139 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8140 // keep a strong reference on this EffectModule to avoid calling the 8141 // destructor before we exit 8142 sp<EffectModule> keep(this); 8143 { 8144 sp<ThreadBase> thread = mThread.promote(); 8145 if (thread != 0) { 8146 thread->disconnectEffect(keep, handle, unpinIfLast); 8147 } 8148 } 8149} 8150 8151void AudioFlinger::EffectModule::updateState() { 8152 Mutex::Autolock _l(mLock); 8153 8154 switch (mState) { 8155 case RESTART: 8156 reset_l(); 8157 // FALL THROUGH 8158 8159 case STARTING: 8160 // clear auxiliary effect input buffer for next accumulation 8161 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8162 memset(mConfig.inputCfg.buffer.raw, 8163 0, 8164 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8165 } 8166 start_l(); 8167 mState = ACTIVE; 8168 break; 8169 case STOPPING: 8170 stop_l(); 8171 mDisableWaitCnt = mMaxDisableWaitCnt; 8172 mState = STOPPED; 8173 break; 8174 case STOPPED: 8175 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8176 // turn off sequence. 8177 if (--mDisableWaitCnt == 0) { 8178 reset_l(); 8179 mState = IDLE; 8180 } 8181 break; 8182 default: //IDLE , ACTIVE, DESTROYED 8183 break; 8184 } 8185} 8186 8187void AudioFlinger::EffectModule::process() 8188{ 8189 Mutex::Autolock _l(mLock); 8190 8191 if (mState == DESTROYED || mEffectInterface == NULL || 8192 mConfig.inputCfg.buffer.raw == NULL || 8193 mConfig.outputCfg.buffer.raw == NULL) { 8194 return; 8195 } 8196 8197 if (isProcessEnabled()) { 8198 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8199 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8200 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8201 mConfig.inputCfg.buffer.s32, 8202 mConfig.inputCfg.buffer.frameCount/2); 8203 } 8204 8205 // do the actual processing in the effect engine 8206 int ret = (*mEffectInterface)->process(mEffectInterface, 8207 &mConfig.inputCfg.buffer, 8208 &mConfig.outputCfg.buffer); 8209 8210 // force transition to IDLE state when engine is ready 8211 if (mState == STOPPED && ret == -ENODATA) { 8212 mDisableWaitCnt = 1; 8213 } 8214 8215 // clear auxiliary effect input buffer for next accumulation 8216 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8217 memset(mConfig.inputCfg.buffer.raw, 0, 8218 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8219 } 8220 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8221 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8222 // If an insert effect is idle and input buffer is different from output buffer, 8223 // accumulate input onto output 8224 sp<EffectChain> chain = mChain.promote(); 8225 if (chain != 0 && chain->activeTrackCnt() != 0) { 8226 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8227 int16_t *in = mConfig.inputCfg.buffer.s16; 8228 int16_t *out = mConfig.outputCfg.buffer.s16; 8229 for (size_t i = 0; i < frameCnt; i++) { 8230 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8231 } 8232 } 8233 } 8234} 8235 8236void AudioFlinger::EffectModule::reset_l() 8237{ 8238 if (mEffectInterface == NULL) { 8239 return; 8240 } 8241 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8242} 8243 8244status_t AudioFlinger::EffectModule::configure() 8245{ 8246 uint32_t channels; 8247 if (mEffectInterface == NULL) { 8248 return NO_INIT; 8249 } 8250 8251 sp<ThreadBase> thread = mThread.promote(); 8252 if (thread == 0) { 8253 return DEAD_OBJECT; 8254 } 8255 8256 // TODO: handle configuration of effects replacing track process 8257 if (thread->channelCount() == 1) { 8258 channels = AUDIO_CHANNEL_OUT_MONO; 8259 } else { 8260 channels = AUDIO_CHANNEL_OUT_STEREO; 8261 } 8262 8263 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8264 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8265 } else { 8266 mConfig.inputCfg.channels = channels; 8267 } 8268 mConfig.outputCfg.channels = channels; 8269 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8270 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8271 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8272 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8273 mConfig.inputCfg.bufferProvider.cookie = NULL; 8274 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8275 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8276 mConfig.outputCfg.bufferProvider.cookie = NULL; 8277 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8278 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8279 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8280 // Insert effect: 8281 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8282 // always overwrites output buffer: input buffer == output buffer 8283 // - in other sessions: 8284 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8285 // other effect: overwrites output buffer: input buffer == output buffer 8286 // Auxiliary effect: 8287 // accumulates in output buffer: input buffer != output buffer 8288 // Therefore: accumulate <=> input buffer != output buffer 8289 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8290 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8291 } else { 8292 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8293 } 8294 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8295 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8296 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8297 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8298 8299 ALOGV("configure() %p thread %p buffer %p framecount %d", 8300 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8301 8302 status_t cmdStatus; 8303 uint32_t size = sizeof(int); 8304 status_t status = (*mEffectInterface)->command(mEffectInterface, 8305 EFFECT_CMD_SET_CONFIG, 8306 sizeof(effect_config_t), 8307 &mConfig, 8308 &size, 8309 &cmdStatus); 8310 if (status == 0) { 8311 status = cmdStatus; 8312 } 8313 8314 if (status == 0 && 8315 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8316 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8317 effect_param_t *p = (effect_param_t *)buf32; 8318 8319 p->psize = sizeof(uint32_t); 8320 p->vsize = sizeof(uint32_t); 8321 size = sizeof(int); 8322 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8323 8324 uint32_t latency = 0; 8325 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8326 if (pbt != NULL) { 8327 latency = pbt->latency_l(); 8328 } 8329 8330 *((int32_t *)p->data + 1)= latency; 8331 (*mEffectInterface)->command(mEffectInterface, 8332 EFFECT_CMD_SET_PARAM, 8333 sizeof(effect_param_t) + 8, 8334 &buf32, 8335 &size, 8336 &cmdStatus); 8337 } 8338 8339 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8340 (1000 * mConfig.outputCfg.buffer.frameCount); 8341 8342 return status; 8343} 8344 8345status_t AudioFlinger::EffectModule::init() 8346{ 8347 Mutex::Autolock _l(mLock); 8348 if (mEffectInterface == NULL) { 8349 return NO_INIT; 8350 } 8351 status_t cmdStatus; 8352 uint32_t size = sizeof(status_t); 8353 status_t status = (*mEffectInterface)->command(mEffectInterface, 8354 EFFECT_CMD_INIT, 8355 0, 8356 NULL, 8357 &size, 8358 &cmdStatus); 8359 if (status == 0) { 8360 status = cmdStatus; 8361 } 8362 return status; 8363} 8364 8365status_t AudioFlinger::EffectModule::start() 8366{ 8367 Mutex::Autolock _l(mLock); 8368 return start_l(); 8369} 8370 8371status_t AudioFlinger::EffectModule::start_l() 8372{ 8373 if (mEffectInterface == NULL) { 8374 return NO_INIT; 8375 } 8376 status_t cmdStatus; 8377 uint32_t size = sizeof(status_t); 8378 status_t status = (*mEffectInterface)->command(mEffectInterface, 8379 EFFECT_CMD_ENABLE, 8380 0, 8381 NULL, 8382 &size, 8383 &cmdStatus); 8384 if (status == 0) { 8385 status = cmdStatus; 8386 } 8387 if (status == 0 && 8388 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8389 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8390 sp<ThreadBase> thread = mThread.promote(); 8391 if (thread != 0) { 8392 audio_stream_t *stream = thread->stream(); 8393 if (stream != NULL) { 8394 stream->add_audio_effect(stream, mEffectInterface); 8395 } 8396 } 8397 } 8398 return status; 8399} 8400 8401status_t AudioFlinger::EffectModule::stop() 8402{ 8403 Mutex::Autolock _l(mLock); 8404 return stop_l(); 8405} 8406 8407status_t AudioFlinger::EffectModule::stop_l() 8408{ 8409 if (mEffectInterface == NULL) { 8410 return NO_INIT; 8411 } 8412 status_t cmdStatus; 8413 uint32_t size = sizeof(status_t); 8414 status_t status = (*mEffectInterface)->command(mEffectInterface, 8415 EFFECT_CMD_DISABLE, 8416 0, 8417 NULL, 8418 &size, 8419 &cmdStatus); 8420 if (status == 0) { 8421 status = cmdStatus; 8422 } 8423 if (status == 0 && 8424 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8425 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8426 sp<ThreadBase> thread = mThread.promote(); 8427 if (thread != 0) { 8428 audio_stream_t *stream = thread->stream(); 8429 if (stream != NULL) { 8430 stream->remove_audio_effect(stream, mEffectInterface); 8431 } 8432 } 8433 } 8434 return status; 8435} 8436 8437status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8438 uint32_t cmdSize, 8439 void *pCmdData, 8440 uint32_t *replySize, 8441 void *pReplyData) 8442{ 8443 Mutex::Autolock _l(mLock); 8444// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8445 8446 if (mState == DESTROYED || mEffectInterface == NULL) { 8447 return NO_INIT; 8448 } 8449 status_t status = (*mEffectInterface)->command(mEffectInterface, 8450 cmdCode, 8451 cmdSize, 8452 pCmdData, 8453 replySize, 8454 pReplyData); 8455 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8456 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8457 for (size_t i = 1; i < mHandles.size(); i++) { 8458 sp<EffectHandle> h = mHandles[i].promote(); 8459 if (h != 0) { 8460 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8461 } 8462 } 8463 } 8464 return status; 8465} 8466 8467status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8468{ 8469 8470 Mutex::Autolock _l(mLock); 8471 ALOGV("setEnabled %p enabled %d", this, enabled); 8472 8473 if (enabled != isEnabled()) { 8474 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8475 if (enabled && status != NO_ERROR) { 8476 return status; 8477 } 8478 8479 switch (mState) { 8480 // going from disabled to enabled 8481 case IDLE: 8482 mState = STARTING; 8483 break; 8484 case STOPPED: 8485 mState = RESTART; 8486 break; 8487 case STOPPING: 8488 mState = ACTIVE; 8489 break; 8490 8491 // going from enabled to disabled 8492 case RESTART: 8493 mState = STOPPED; 8494 break; 8495 case STARTING: 8496 mState = IDLE; 8497 break; 8498 case ACTIVE: 8499 mState = STOPPING; 8500 break; 8501 case DESTROYED: 8502 return NO_ERROR; // simply ignore as we are being destroyed 8503 } 8504 for (size_t i = 1; i < mHandles.size(); i++) { 8505 sp<EffectHandle> h = mHandles[i].promote(); 8506 if (h != 0) { 8507 h->setEnabled(enabled); 8508 } 8509 } 8510 } 8511 return NO_ERROR; 8512} 8513 8514bool AudioFlinger::EffectModule::isEnabled() const 8515{ 8516 switch (mState) { 8517 case RESTART: 8518 case STARTING: 8519 case ACTIVE: 8520 return true; 8521 case IDLE: 8522 case STOPPING: 8523 case STOPPED: 8524 case DESTROYED: 8525 default: 8526 return false; 8527 } 8528} 8529 8530bool AudioFlinger::EffectModule::isProcessEnabled() const 8531{ 8532 switch (mState) { 8533 case RESTART: 8534 case ACTIVE: 8535 case STOPPING: 8536 case STOPPED: 8537 return true; 8538 case IDLE: 8539 case STARTING: 8540 case DESTROYED: 8541 default: 8542 return false; 8543 } 8544} 8545 8546status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8547{ 8548 Mutex::Autolock _l(mLock); 8549 status_t status = NO_ERROR; 8550 8551 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8552 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8553 if (isProcessEnabled() && 8554 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8555 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8556 status_t cmdStatus; 8557 uint32_t volume[2]; 8558 uint32_t *pVolume = NULL; 8559 uint32_t size = sizeof(volume); 8560 volume[0] = *left; 8561 volume[1] = *right; 8562 if (controller) { 8563 pVolume = volume; 8564 } 8565 status = (*mEffectInterface)->command(mEffectInterface, 8566 EFFECT_CMD_SET_VOLUME, 8567 size, 8568 volume, 8569 &size, 8570 pVolume); 8571 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8572 *left = volume[0]; 8573 *right = volume[1]; 8574 } 8575 } 8576 return status; 8577} 8578 8579status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8580{ 8581 Mutex::Autolock _l(mLock); 8582 status_t status = NO_ERROR; 8583 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8584 // audio pre processing modules on RecordThread can receive both output and 8585 // input device indication in the same call 8586 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8587 if (dev) { 8588 status_t cmdStatus; 8589 uint32_t size = sizeof(status_t); 8590 8591 status = (*mEffectInterface)->command(mEffectInterface, 8592 EFFECT_CMD_SET_DEVICE, 8593 sizeof(uint32_t), 8594 &dev, 8595 &size, 8596 &cmdStatus); 8597 if (status == NO_ERROR) { 8598 status = cmdStatus; 8599 } 8600 } 8601 dev = device & AUDIO_DEVICE_IN_ALL; 8602 if (dev) { 8603 status_t cmdStatus; 8604 uint32_t size = sizeof(status_t); 8605 8606 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8607 EFFECT_CMD_SET_INPUT_DEVICE, 8608 sizeof(uint32_t), 8609 &dev, 8610 &size, 8611 &cmdStatus); 8612 if (status2 == NO_ERROR) { 8613 status2 = cmdStatus; 8614 } 8615 if (status == NO_ERROR) { 8616 status = status2; 8617 } 8618 } 8619 } 8620 return status; 8621} 8622 8623status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8624{ 8625 Mutex::Autolock _l(mLock); 8626 status_t status = NO_ERROR; 8627 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8628 status_t cmdStatus; 8629 uint32_t size = sizeof(status_t); 8630 status = (*mEffectInterface)->command(mEffectInterface, 8631 EFFECT_CMD_SET_AUDIO_MODE, 8632 sizeof(audio_mode_t), 8633 &mode, 8634 &size, 8635 &cmdStatus); 8636 if (status == NO_ERROR) { 8637 status = cmdStatus; 8638 } 8639 } 8640 return status; 8641} 8642 8643void AudioFlinger::EffectModule::setSuspended(bool suspended) 8644{ 8645 Mutex::Autolock _l(mLock); 8646 mSuspended = suspended; 8647} 8648 8649bool AudioFlinger::EffectModule::suspended() const 8650{ 8651 Mutex::Autolock _l(mLock); 8652 return mSuspended; 8653} 8654 8655status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8656{ 8657 const size_t SIZE = 256; 8658 char buffer[SIZE]; 8659 String8 result; 8660 8661 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8662 result.append(buffer); 8663 8664 bool locked = tryLock(mLock); 8665 // failed to lock - AudioFlinger is probably deadlocked 8666 if (!locked) { 8667 result.append("\t\tCould not lock Fx mutex:\n"); 8668 } 8669 8670 result.append("\t\tSession Status State Engine:\n"); 8671 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8672 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8673 result.append(buffer); 8674 8675 result.append("\t\tDescriptor:\n"); 8676 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8677 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8678 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8679 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8680 result.append(buffer); 8681 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8682 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8683 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8684 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8685 result.append(buffer); 8686 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8687 mDescriptor.apiVersion, 8688 mDescriptor.flags); 8689 result.append(buffer); 8690 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8691 mDescriptor.name); 8692 result.append(buffer); 8693 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8694 mDescriptor.implementor); 8695 result.append(buffer); 8696 8697 result.append("\t\t- Input configuration:\n"); 8698 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8699 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8700 (uint32_t)mConfig.inputCfg.buffer.raw, 8701 mConfig.inputCfg.buffer.frameCount, 8702 mConfig.inputCfg.samplingRate, 8703 mConfig.inputCfg.channels, 8704 mConfig.inputCfg.format); 8705 result.append(buffer); 8706 8707 result.append("\t\t- Output configuration:\n"); 8708 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8709 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8710 (uint32_t)mConfig.outputCfg.buffer.raw, 8711 mConfig.outputCfg.buffer.frameCount, 8712 mConfig.outputCfg.samplingRate, 8713 mConfig.outputCfg.channels, 8714 mConfig.outputCfg.format); 8715 result.append(buffer); 8716 8717 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8718 result.append(buffer); 8719 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8720 for (size_t i = 0; i < mHandles.size(); ++i) { 8721 sp<EffectHandle> handle = mHandles[i].promote(); 8722 if (handle != 0) { 8723 handle->dump(buffer, SIZE); 8724 result.append(buffer); 8725 } 8726 } 8727 8728 result.append("\n"); 8729 8730 write(fd, result.string(), result.length()); 8731 8732 if (locked) { 8733 mLock.unlock(); 8734 } 8735 8736 return NO_ERROR; 8737} 8738 8739// ---------------------------------------------------------------------------- 8740// EffectHandle implementation 8741// ---------------------------------------------------------------------------- 8742 8743#undef LOG_TAG 8744#define LOG_TAG "AudioFlinger::EffectHandle" 8745 8746AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8747 const sp<AudioFlinger::Client>& client, 8748 const sp<IEffectClient>& effectClient, 8749 int32_t priority) 8750 : BnEffect(), 8751 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8752 mPriority(priority), mHasControl(false), mEnabled(false) 8753{ 8754 ALOGV("constructor %p", this); 8755 8756 if (client == 0) { 8757 return; 8758 } 8759 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8760 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8761 if (mCblkMemory != 0) { 8762 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8763 8764 if (mCblk != NULL) { 8765 new(mCblk) effect_param_cblk_t(); 8766 mBuffer = (uint8_t *)mCblk + bufOffset; 8767 } 8768 } else { 8769 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8770 return; 8771 } 8772} 8773 8774AudioFlinger::EffectHandle::~EffectHandle() 8775{ 8776 ALOGV("Destructor %p", this); 8777 disconnect(false); 8778 ALOGV("Destructor DONE %p", this); 8779} 8780 8781status_t AudioFlinger::EffectHandle::enable() 8782{ 8783 ALOGV("enable %p", this); 8784 if (!mHasControl) return INVALID_OPERATION; 8785 if (mEffect == 0) return DEAD_OBJECT; 8786 8787 if (mEnabled) { 8788 return NO_ERROR; 8789 } 8790 8791 mEnabled = true; 8792 8793 sp<ThreadBase> thread = mEffect->thread().promote(); 8794 if (thread != 0) { 8795 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8796 } 8797 8798 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8799 if (mEffect->suspended()) { 8800 return NO_ERROR; 8801 } 8802 8803 status_t status = mEffect->setEnabled(true); 8804 if (status != NO_ERROR) { 8805 if (thread != 0) { 8806 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8807 } 8808 mEnabled = false; 8809 } 8810 return status; 8811} 8812 8813status_t AudioFlinger::EffectHandle::disable() 8814{ 8815 ALOGV("disable %p", this); 8816 if (!mHasControl) return INVALID_OPERATION; 8817 if (mEffect == 0) return DEAD_OBJECT; 8818 8819 if (!mEnabled) { 8820 return NO_ERROR; 8821 } 8822 mEnabled = false; 8823 8824 if (mEffect->suspended()) { 8825 return NO_ERROR; 8826 } 8827 8828 status_t status = mEffect->setEnabled(false); 8829 8830 sp<ThreadBase> thread = mEffect->thread().promote(); 8831 if (thread != 0) { 8832 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8833 } 8834 8835 return status; 8836} 8837 8838void AudioFlinger::EffectHandle::disconnect() 8839{ 8840 disconnect(true); 8841} 8842 8843void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8844{ 8845 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8846 if (mEffect == 0) { 8847 return; 8848 } 8849 mEffect->disconnect(this, unpinIfLast); 8850 8851 if (mHasControl && mEnabled) { 8852 sp<ThreadBase> thread = mEffect->thread().promote(); 8853 if (thread != 0) { 8854 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8855 } 8856 } 8857 8858 // release sp on module => module destructor can be called now 8859 mEffect.clear(); 8860 if (mClient != 0) { 8861 if (mCblk != NULL) { 8862 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8863 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8864 } 8865 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8866 // Client destructor must run with AudioFlinger mutex locked 8867 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8868 mClient.clear(); 8869 } 8870} 8871 8872status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8873 uint32_t cmdSize, 8874 void *pCmdData, 8875 uint32_t *replySize, 8876 void *pReplyData) 8877{ 8878// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8879// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8880 8881 // only get parameter command is permitted for applications not controlling the effect 8882 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8883 return INVALID_OPERATION; 8884 } 8885 if (mEffect == 0) return DEAD_OBJECT; 8886 if (mClient == 0) return INVALID_OPERATION; 8887 8888 // handle commands that are not forwarded transparently to effect engine 8889 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8890 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8891 // no risk to block the whole media server process or mixer threads is we are stuck here 8892 Mutex::Autolock _l(mCblk->lock); 8893 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8894 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8895 mCblk->serverIndex = 0; 8896 mCblk->clientIndex = 0; 8897 return BAD_VALUE; 8898 } 8899 status_t status = NO_ERROR; 8900 while (mCblk->serverIndex < mCblk->clientIndex) { 8901 int reply; 8902 uint32_t rsize = sizeof(int); 8903 int *p = (int *)(mBuffer + mCblk->serverIndex); 8904 int size = *p++; 8905 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8906 ALOGW("command(): invalid parameter block size"); 8907 break; 8908 } 8909 effect_param_t *param = (effect_param_t *)p; 8910 if (param->psize == 0 || param->vsize == 0) { 8911 ALOGW("command(): null parameter or value size"); 8912 mCblk->serverIndex += size; 8913 continue; 8914 } 8915 uint32_t psize = sizeof(effect_param_t) + 8916 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8917 param->vsize; 8918 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8919 psize, 8920 p, 8921 &rsize, 8922 &reply); 8923 // stop at first error encountered 8924 if (ret != NO_ERROR) { 8925 status = ret; 8926 *(int *)pReplyData = reply; 8927 break; 8928 } else if (reply != NO_ERROR) { 8929 *(int *)pReplyData = reply; 8930 break; 8931 } 8932 mCblk->serverIndex += size; 8933 } 8934 mCblk->serverIndex = 0; 8935 mCblk->clientIndex = 0; 8936 return status; 8937 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8938 *(int *)pReplyData = NO_ERROR; 8939 return enable(); 8940 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8941 *(int *)pReplyData = NO_ERROR; 8942 return disable(); 8943 } 8944 8945 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8946} 8947 8948void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8949{ 8950 ALOGV("setControl %p control %d", this, hasControl); 8951 8952 mHasControl = hasControl; 8953 mEnabled = enabled; 8954 8955 if (signal && mEffectClient != 0) { 8956 mEffectClient->controlStatusChanged(hasControl); 8957 } 8958} 8959 8960void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8961 uint32_t cmdSize, 8962 void *pCmdData, 8963 uint32_t replySize, 8964 void *pReplyData) 8965{ 8966 if (mEffectClient != 0) { 8967 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8968 } 8969} 8970 8971 8972 8973void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8974{ 8975 if (mEffectClient != 0) { 8976 mEffectClient->enableStatusChanged(enabled); 8977 } 8978} 8979 8980status_t AudioFlinger::EffectHandle::onTransact( 8981 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8982{ 8983 return BnEffect::onTransact(code, data, reply, flags); 8984} 8985 8986 8987void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8988{ 8989 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8990 8991 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8992 (mClient == 0) ? getpid_cached : mClient->pid(), 8993 mPriority, 8994 mHasControl, 8995 !locked, 8996 mCblk ? mCblk->clientIndex : 0, 8997 mCblk ? mCblk->serverIndex : 0 8998 ); 8999 9000 if (locked) { 9001 mCblk->lock.unlock(); 9002 } 9003} 9004 9005#undef LOG_TAG 9006#define LOG_TAG "AudioFlinger::EffectChain" 9007 9008AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9009 int sessionId) 9010 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9011 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9012 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9013{ 9014 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9015 if (thread == NULL) { 9016 return; 9017 } 9018 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9019 thread->frameCount(); 9020} 9021 9022AudioFlinger::EffectChain::~EffectChain() 9023{ 9024 if (mOwnInBuffer) { 9025 delete mInBuffer; 9026 } 9027 9028} 9029 9030// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9031sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9032{ 9033 size_t size = mEffects.size(); 9034 9035 for (size_t i = 0; i < size; i++) { 9036 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9037 return mEffects[i]; 9038 } 9039 } 9040 return 0; 9041} 9042 9043// getEffectFromId_l() must be called with ThreadBase::mLock held 9044sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9045{ 9046 size_t size = mEffects.size(); 9047 9048 for (size_t i = 0; i < size; i++) { 9049 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9050 if (id == 0 || mEffects[i]->id() == id) { 9051 return mEffects[i]; 9052 } 9053 } 9054 return 0; 9055} 9056 9057// getEffectFromType_l() must be called with ThreadBase::mLock held 9058sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9059 const effect_uuid_t *type) 9060{ 9061 size_t size = mEffects.size(); 9062 9063 for (size_t i = 0; i < size; i++) { 9064 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9065 return mEffects[i]; 9066 } 9067 } 9068 return 0; 9069} 9070 9071void AudioFlinger::EffectChain::clearInputBuffer() 9072{ 9073 Mutex::Autolock _l(mLock); 9074 sp<ThreadBase> thread = mThread.promote(); 9075 if (thread == 0) { 9076 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9077 return; 9078 } 9079 clearInputBuffer_l(thread); 9080} 9081 9082// Must be called with EffectChain::mLock locked 9083void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9084{ 9085 size_t numSamples = thread->frameCount() * thread->channelCount(); 9086 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9087 9088} 9089 9090// Must be called with EffectChain::mLock locked 9091void AudioFlinger::EffectChain::process_l() 9092{ 9093 sp<ThreadBase> thread = mThread.promote(); 9094 if (thread == 0) { 9095 ALOGW("process_l(): cannot promote mixer thread"); 9096 return; 9097 } 9098 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9099 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9100 // always process effects unless no more tracks are on the session and the effect tail 9101 // has been rendered 9102 bool doProcess = true; 9103 if (!isGlobalSession) { 9104 bool tracksOnSession = (trackCnt() != 0); 9105 9106 if (!tracksOnSession && mTailBufferCount == 0) { 9107 doProcess = false; 9108 } 9109 9110 if (activeTrackCnt() == 0) { 9111 // if no track is active and the effect tail has not been rendered, 9112 // the input buffer must be cleared here as the mixer process will not do it 9113 if (tracksOnSession || mTailBufferCount > 0) { 9114 clearInputBuffer_l(thread); 9115 if (mTailBufferCount > 0) { 9116 mTailBufferCount--; 9117 } 9118 } 9119 } 9120 } 9121 9122 size_t size = mEffects.size(); 9123 if (doProcess) { 9124 for (size_t i = 0; i < size; i++) { 9125 mEffects[i]->process(); 9126 } 9127 } 9128 for (size_t i = 0; i < size; i++) { 9129 mEffects[i]->updateState(); 9130 } 9131} 9132 9133// addEffect_l() must be called with PlaybackThread::mLock held 9134status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9135{ 9136 effect_descriptor_t desc = effect->desc(); 9137 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9138 9139 Mutex::Autolock _l(mLock); 9140 effect->setChain(this); 9141 sp<ThreadBase> thread = mThread.promote(); 9142 if (thread == 0) { 9143 return NO_INIT; 9144 } 9145 effect->setThread(thread); 9146 9147 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9148 // Auxiliary effects are inserted at the beginning of mEffects vector as 9149 // they are processed first and accumulated in chain input buffer 9150 mEffects.insertAt(effect, 0); 9151 9152 // the input buffer for auxiliary effect contains mono samples in 9153 // 32 bit format. This is to avoid saturation in AudoMixer 9154 // accumulation stage. Saturation is done in EffectModule::process() before 9155 // calling the process in effect engine 9156 size_t numSamples = thread->frameCount(); 9157 int32_t *buffer = new int32_t[numSamples]; 9158 memset(buffer, 0, numSamples * sizeof(int32_t)); 9159 effect->setInBuffer((int16_t *)buffer); 9160 // auxiliary effects output samples to chain input buffer for further processing 9161 // by insert effects 9162 effect->setOutBuffer(mInBuffer); 9163 } else { 9164 // Insert effects are inserted at the end of mEffects vector as they are processed 9165 // after track and auxiliary effects. 9166 // Insert effect order as a function of indicated preference: 9167 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9168 // another effect is present 9169 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9170 // last effect claiming first position 9171 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9172 // first effect claiming last position 9173 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9174 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9175 // already present 9176 9177 size_t size = mEffects.size(); 9178 size_t idx_insert = size; 9179 ssize_t idx_insert_first = -1; 9180 ssize_t idx_insert_last = -1; 9181 9182 for (size_t i = 0; i < size; i++) { 9183 effect_descriptor_t d = mEffects[i]->desc(); 9184 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9185 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9186 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9187 // check invalid effect chaining combinations 9188 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9189 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9190 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9191 return INVALID_OPERATION; 9192 } 9193 // remember position of first insert effect and by default 9194 // select this as insert position for new effect 9195 if (idx_insert == size) { 9196 idx_insert = i; 9197 } 9198 // remember position of last insert effect claiming 9199 // first position 9200 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9201 idx_insert_first = i; 9202 } 9203 // remember position of first insert effect claiming 9204 // last position 9205 if (iPref == EFFECT_FLAG_INSERT_LAST && 9206 idx_insert_last == -1) { 9207 idx_insert_last = i; 9208 } 9209 } 9210 } 9211 9212 // modify idx_insert from first position if needed 9213 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9214 if (idx_insert_last != -1) { 9215 idx_insert = idx_insert_last; 9216 } else { 9217 idx_insert = size; 9218 } 9219 } else { 9220 if (idx_insert_first != -1) { 9221 idx_insert = idx_insert_first + 1; 9222 } 9223 } 9224 9225 // always read samples from chain input buffer 9226 effect->setInBuffer(mInBuffer); 9227 9228 // if last effect in the chain, output samples to chain 9229 // output buffer, otherwise to chain input buffer 9230 if (idx_insert == size) { 9231 if (idx_insert != 0) { 9232 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9233 mEffects[idx_insert-1]->configure(); 9234 } 9235 effect->setOutBuffer(mOutBuffer); 9236 } else { 9237 effect->setOutBuffer(mInBuffer); 9238 } 9239 mEffects.insertAt(effect, idx_insert); 9240 9241 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9242 } 9243 effect->configure(); 9244 return NO_ERROR; 9245} 9246 9247// removeEffect_l() must be called with PlaybackThread::mLock held 9248size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9249{ 9250 Mutex::Autolock _l(mLock); 9251 size_t size = mEffects.size(); 9252 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9253 9254 for (size_t i = 0; i < size; i++) { 9255 if (effect == mEffects[i]) { 9256 // calling stop here will remove pre-processing effect from the audio HAL. 9257 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9258 // the middle of a read from audio HAL 9259 if (mEffects[i]->state() == EffectModule::ACTIVE || 9260 mEffects[i]->state() == EffectModule::STOPPING) { 9261 mEffects[i]->stop(); 9262 } 9263 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9264 delete[] effect->inBuffer(); 9265 } else { 9266 if (i == size - 1 && i != 0) { 9267 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9268 mEffects[i - 1]->configure(); 9269 } 9270 } 9271 mEffects.removeAt(i); 9272 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9273 break; 9274 } 9275 } 9276 9277 return mEffects.size(); 9278} 9279 9280// setDevice_l() must be called with PlaybackThread::mLock held 9281void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9282{ 9283 size_t size = mEffects.size(); 9284 for (size_t i = 0; i < size; i++) { 9285 mEffects[i]->setDevice(device); 9286 } 9287} 9288 9289// setMode_l() must be called with PlaybackThread::mLock held 9290void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9291{ 9292 size_t size = mEffects.size(); 9293 for (size_t i = 0; i < size; i++) { 9294 mEffects[i]->setMode(mode); 9295 } 9296} 9297 9298// setVolume_l() must be called with PlaybackThread::mLock held 9299bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9300{ 9301 uint32_t newLeft = *left; 9302 uint32_t newRight = *right; 9303 bool hasControl = false; 9304 int ctrlIdx = -1; 9305 size_t size = mEffects.size(); 9306 9307 // first update volume controller 9308 for (size_t i = size; i > 0; i--) { 9309 if (mEffects[i - 1]->isProcessEnabled() && 9310 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9311 ctrlIdx = i - 1; 9312 hasControl = true; 9313 break; 9314 } 9315 } 9316 9317 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9318 if (hasControl) { 9319 *left = mNewLeftVolume; 9320 *right = mNewRightVolume; 9321 } 9322 return hasControl; 9323 } 9324 9325 mVolumeCtrlIdx = ctrlIdx; 9326 mLeftVolume = newLeft; 9327 mRightVolume = newRight; 9328 9329 // second get volume update from volume controller 9330 if (ctrlIdx >= 0) { 9331 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9332 mNewLeftVolume = newLeft; 9333 mNewRightVolume = newRight; 9334 } 9335 // then indicate volume to all other effects in chain. 9336 // Pass altered volume to effects before volume controller 9337 // and requested volume to effects after controller 9338 uint32_t lVol = newLeft; 9339 uint32_t rVol = newRight; 9340 9341 for (size_t i = 0; i < size; i++) { 9342 if ((int)i == ctrlIdx) continue; 9343 // this also works for ctrlIdx == -1 when there is no volume controller 9344 if ((int)i > ctrlIdx) { 9345 lVol = *left; 9346 rVol = *right; 9347 } 9348 mEffects[i]->setVolume(&lVol, &rVol, false); 9349 } 9350 *left = newLeft; 9351 *right = newRight; 9352 9353 return hasControl; 9354} 9355 9356status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9357{ 9358 const size_t SIZE = 256; 9359 char buffer[SIZE]; 9360 String8 result; 9361 9362 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9363 result.append(buffer); 9364 9365 bool locked = tryLock(mLock); 9366 // failed to lock - AudioFlinger is probably deadlocked 9367 if (!locked) { 9368 result.append("\tCould not lock mutex:\n"); 9369 } 9370 9371 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9372 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9373 mEffects.size(), 9374 (uint32_t)mInBuffer, 9375 (uint32_t)mOutBuffer, 9376 mActiveTrackCnt); 9377 result.append(buffer); 9378 write(fd, result.string(), result.size()); 9379 9380 for (size_t i = 0; i < mEffects.size(); ++i) { 9381 sp<EffectModule> effect = mEffects[i]; 9382 if (effect != 0) { 9383 effect->dump(fd, args); 9384 } 9385 } 9386 9387 if (locked) { 9388 mLock.unlock(); 9389 } 9390 9391 return NO_ERROR; 9392} 9393 9394// must be called with ThreadBase::mLock held 9395void AudioFlinger::EffectChain::setEffectSuspended_l( 9396 const effect_uuid_t *type, bool suspend) 9397{ 9398 sp<SuspendedEffectDesc> desc; 9399 // use effect type UUID timelow as key as there is no real risk of identical 9400 // timeLow fields among effect type UUIDs. 9401 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9402 if (suspend) { 9403 if (index >= 0) { 9404 desc = mSuspendedEffects.valueAt(index); 9405 } else { 9406 desc = new SuspendedEffectDesc(); 9407 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9408 mSuspendedEffects.add(type->timeLow, desc); 9409 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9410 } 9411 if (desc->mRefCount++ == 0) { 9412 sp<EffectModule> effect = getEffectIfEnabled(type); 9413 if (effect != 0) { 9414 desc->mEffect = effect; 9415 effect->setSuspended(true); 9416 effect->setEnabled(false); 9417 } 9418 } 9419 } else { 9420 if (index < 0) { 9421 return; 9422 } 9423 desc = mSuspendedEffects.valueAt(index); 9424 if (desc->mRefCount <= 0) { 9425 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9426 desc->mRefCount = 1; 9427 } 9428 if (--desc->mRefCount == 0) { 9429 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9430 if (desc->mEffect != 0) { 9431 sp<EffectModule> effect = desc->mEffect.promote(); 9432 if (effect != 0) { 9433 effect->setSuspended(false); 9434 sp<EffectHandle> handle = effect->controlHandle(); 9435 if (handle != 0) { 9436 effect->setEnabled(handle->enabled()); 9437 } 9438 } 9439 desc->mEffect.clear(); 9440 } 9441 mSuspendedEffects.removeItemsAt(index); 9442 } 9443 } 9444} 9445 9446// must be called with ThreadBase::mLock held 9447void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9448{ 9449 sp<SuspendedEffectDesc> desc; 9450 9451 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9452 if (suspend) { 9453 if (index >= 0) { 9454 desc = mSuspendedEffects.valueAt(index); 9455 } else { 9456 desc = new SuspendedEffectDesc(); 9457 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9458 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9459 } 9460 if (desc->mRefCount++ == 0) { 9461 Vector< sp<EffectModule> > effects; 9462 getSuspendEligibleEffects(effects); 9463 for (size_t i = 0; i < effects.size(); i++) { 9464 setEffectSuspended_l(&effects[i]->desc().type, true); 9465 } 9466 } 9467 } else { 9468 if (index < 0) { 9469 return; 9470 } 9471 desc = mSuspendedEffects.valueAt(index); 9472 if (desc->mRefCount <= 0) { 9473 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9474 desc->mRefCount = 1; 9475 } 9476 if (--desc->mRefCount == 0) { 9477 Vector<const effect_uuid_t *> types; 9478 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9479 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9480 continue; 9481 } 9482 types.add(&mSuspendedEffects.valueAt(i)->mType); 9483 } 9484 for (size_t i = 0; i < types.size(); i++) { 9485 setEffectSuspended_l(types[i], false); 9486 } 9487 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9488 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9489 } 9490 } 9491} 9492 9493 9494// The volume effect is used for automated tests only 9495#ifndef OPENSL_ES_H_ 9496static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9497 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9498const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9499#endif //OPENSL_ES_H_ 9500 9501bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9502{ 9503 // auxiliary effects and visualizer are never suspended on output mix 9504 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9505 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9506 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9507 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9508 return false; 9509 } 9510 return true; 9511} 9512 9513void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9514{ 9515 effects.clear(); 9516 for (size_t i = 0; i < mEffects.size(); i++) { 9517 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9518 effects.add(mEffects[i]); 9519 } 9520 } 9521} 9522 9523sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9524 const effect_uuid_t *type) 9525{ 9526 sp<EffectModule> effect = getEffectFromType_l(type); 9527 return effect != 0 && effect->isEnabled() ? effect : 0; 9528} 9529 9530void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9531 bool enabled) 9532{ 9533 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9534 if (enabled) { 9535 if (index < 0) { 9536 // if the effect is not suspend check if all effects are suspended 9537 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9538 if (index < 0) { 9539 return; 9540 } 9541 if (!isEffectEligibleForSuspend(effect->desc())) { 9542 return; 9543 } 9544 setEffectSuspended_l(&effect->desc().type, enabled); 9545 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9546 if (index < 0) { 9547 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9548 return; 9549 } 9550 } 9551 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9552 effect->desc().type.timeLow); 9553 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9554 // if effect is requested to suspended but was not yet enabled, supend it now. 9555 if (desc->mEffect == 0) { 9556 desc->mEffect = effect; 9557 effect->setEnabled(false); 9558 effect->setSuspended(true); 9559 } 9560 } else { 9561 if (index < 0) { 9562 return; 9563 } 9564 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9565 effect->desc().type.timeLow); 9566 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9567 desc->mEffect.clear(); 9568 effect->setSuspended(false); 9569 } 9570} 9571 9572#undef LOG_TAG 9573#define LOG_TAG "AudioFlinger" 9574 9575// ---------------------------------------------------------------------------- 9576 9577status_t AudioFlinger::onTransact( 9578 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9579{ 9580 return BnAudioFlinger::onTransact(code, data, reply, flags); 9581} 9582 9583}; // namespace android 9584