AudioFlinger.cpp revision a4f7e0e9a0e92a063f1b3a08988cf46e2cf1fa94
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167// ---------------------------------------------------------------------------- 168 169#ifdef ADD_BATTERY_DATA 170// To collect the amplifier usage 171static void addBatteryData(uint32_t params) { 172 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 173 if (service == NULL) { 174 // it already logged 175 return; 176 } 177 178 service->addBatteryData(params); 179} 180#endif 181 182static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183{ 184 const hw_module_t *mod; 185 int rc; 186 187 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190 if (rc) { 191 goto out; 192 } 193 rc = audio_hw_device_open(mod, dev); 194 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 195 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196 if (rc) { 197 goto out; 198 } 199 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201 rc = BAD_VALUE; 202 goto out; 203 } 204 return 0; 205 206out: 207 *dev = NULL; 208 return rc; 209} 210 211// ---------------------------------------------------------------------------- 212 213AudioFlinger::AudioFlinger() 214 : BnAudioFlinger(), 215 mPrimaryHardwareDev(NULL), 216 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217 mMasterVolume(1.0f), 218 mMasterVolumeSupportLvl(MVS_NONE), 219 mMasterMute(false), 220 mNextUniqueId(1), 221 mMode(AUDIO_MODE_INVALID), 222 mBtNrecIsOff(false) 223{ 224} 225 226void AudioFlinger::onFirstRef() 227{ 228 int rc = 0; 229 230 Mutex::Autolock _l(mLock); 231 232 /* TODO: move all this work into an Init() function */ 233 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 235 uint32_t int_val; 236 if (1 == sscanf(val_str, "%u", &int_val)) { 237 mStandbyTimeInNsecs = milliseconds(int_val); 238 ALOGI("Using %u mSec as standby time.", int_val); 239 } else { 240 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241 ALOGI("Using default %u mSec as standby time.", 242 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243 } 244 } 245 246 mMode = AUDIO_MODE_NORMAL; 247 mMasterVolumeSW = 1.0; 248 mMasterVolume = 1.0; 249 mHardwareStatus = AUDIO_HW_IDLE; 250} 251 252AudioFlinger::~AudioFlinger() 253{ 254 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 // check if an effect chain with the same session ID is present on another 478 // output thread and move it here. 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 uint32_t sessions = t->hasAudioSession(*sessionId); 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 break; 486 } 487 } 488 } 489 lSessionId = *sessionId; 490 } else { 491 // if no audio session id is provided, create one here 492 lSessionId = nextUniqueId(); 493 if (sessionId != NULL) { 494 *sessionId = lSessionId; 495 } 496 } 497 ALOGV("createTrack() lSessionId: %d", lSessionId); 498 499 track = thread->createTrack_l(client, streamType, sampleRate, format, 500 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 501 502 // move effect chain to this output thread if an effect on same session was waiting 503 // for a track to be created 504 if (lStatus == NO_ERROR && effectThread != NULL) { 505 Mutex::Autolock _dl(thread->mLock); 506 Mutex::Autolock _sl(effectThread->mLock); 507 moveEffectChain_l(lSessionId, effectThread, thread, true); 508 } 509 510 // Look for sync events awaiting for a session to be used. 511 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 512 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 513 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 514 if (lStatus == NO_ERROR) { 515 track->setSyncEvent(mPendingSyncEvents[i]); 516 } else { 517 mPendingSyncEvents[i]->cancel(); 518 } 519 mPendingSyncEvents.removeAt(i); 520 i--; 521 } 522 } 523 } 524 } 525 if (lStatus == NO_ERROR) { 526 trackHandle = new TrackHandle(track); 527 } else { 528 // remove local strong reference to Client before deleting the Track so that the Client 529 // destructor is called by the TrackBase destructor with mLock held 530 client.clear(); 531 track.clear(); 532 } 533 534Exit: 535 if (status != NULL) { 536 *status = lStatus; 537 } 538 return trackHandle; 539} 540 541uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("sampleRate() unknown thread %d", output); 547 return 0; 548 } 549 return thread->sampleRate(); 550} 551 552int AudioFlinger::channelCount(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("channelCount() unknown thread %d", output); 558 return 0; 559 } 560 return thread->channelCount(); 561} 562 563audio_format_t AudioFlinger::format(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("format() unknown thread %d", output); 569 return AUDIO_FORMAT_INVALID; 570 } 571 return thread->format(); 572} 573 574size_t AudioFlinger::frameCount(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("frameCount() unknown thread %d", output); 580 return 0; 581 } 582 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 583 // should examine all callers and fix them to handle smaller counts 584 return thread->frameCount(); 585} 586 587uint32_t AudioFlinger::latency(audio_io_handle_t output) const 588{ 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGW("latency() unknown thread %d", output); 593 return 0; 594 } 595 return thread->latency(); 596} 597 598status_t AudioFlinger::setMasterVolume(float value) 599{ 600 status_t ret = initCheck(); 601 if (ret != NO_ERROR) { 602 return ret; 603 } 604 605 // check calling permissions 606 if (!settingsAllowed()) { 607 return PERMISSION_DENIED; 608 } 609 610 float swmv = value; 611 612 Mutex::Autolock _l(mLock); 613 614 // when hw supports master volume, don't scale in sw mixer 615 if (MVS_NONE != mMasterVolumeSupportLvl) { 616 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 617 AutoMutex lock(mHardwareLock); 618 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 619 620 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 621 if (NULL != dev->set_master_volume) { 622 dev->set_master_volume(dev, value); 623 } 624 mHardwareStatus = AUDIO_HW_IDLE; 625 } 626 627 swmv = 1.0; 628 } 629 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 Mutex::Autolock _l(mLock); 857 status_t final_result = NO_ERROR; 858 { 859 AutoMutex lock(mHardwareLock); 860 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 861 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 862 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 863 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 864 final_result = result ?: final_result; 865 } 866 mHardwareStatus = AUDIO_HW_IDLE; 867 } 868 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 869 AudioParameter param = AudioParameter(keyValuePairs); 870 String8 value; 871 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 Mutex::Autolock _l(mLock); 927 928 if (ioHandle == 0) { 929 String8 out_s8; 930 931 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 932 char *s; 933 { 934 AutoMutex lock(mHardwareLock); 935 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 936 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 937 s = dev->get_parameters(dev, keys.string()); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 } 940 out_s8 += String8(s ? s : ""); 941 free(s); 942 } 943 return out_s8; 944 } 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 struct audio_config config = { 967 sample_rate: sampleRate, 968 channel_mask: audio_channel_in_mask_from_count(channelCount), 969 format: format, 970 }; 971 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 return size; 974} 975 976unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 977{ 978 if (ioHandle == 0) { 979 return 0; 980 } 981 982 Mutex::Autolock _l(mLock); 983 984 RecordThread *recordThread = checkRecordThread_l(ioHandle); 985 if (recordThread != NULL) { 986 return recordThread->getInputFramesLost(); 987 } 988 return 0; 989} 990 991status_t AudioFlinger::setVoiceVolume(float value) 992{ 993 status_t ret = initCheck(); 994 if (ret != NO_ERROR) { 995 return ret; 996 } 997 998 // check calling permissions 999 if (!settingsAllowed()) { 1000 return PERMISSION_DENIED; 1001 } 1002 1003 AutoMutex lock(mHardwareLock); 1004 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1005 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1006 mHardwareStatus = AUDIO_HW_IDLE; 1007 1008 return ret; 1009} 1010 1011status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1012 audio_io_handle_t output) const 1013{ 1014 status_t status; 1015 1016 Mutex::Autolock _l(mLock); 1017 1018 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1019 if (playbackThread != NULL) { 1020 return playbackThread->getRenderPosition(halFrames, dspFrames); 1021 } 1022 1023 return BAD_VALUE; 1024} 1025 1026void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1027{ 1028 1029 Mutex::Autolock _l(mLock); 1030 1031 pid_t pid = IPCThreadState::self()->getCallingPid(); 1032 if (mNotificationClients.indexOfKey(pid) < 0) { 1033 sp<NotificationClient> notificationClient = new NotificationClient(this, 1034 client, 1035 pid); 1036 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1037 1038 mNotificationClients.add(pid, notificationClient); 1039 1040 sp<IBinder> binder = client->asBinder(); 1041 binder->linkToDeath(notificationClient); 1042 1043 // the config change is always sent from playback or record threads to avoid deadlock 1044 // with AudioSystem::gLock 1045 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1046 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1047 } 1048 1049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1050 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1051 } 1052 } 1053} 1054 1055void AudioFlinger::removeNotificationClient(pid_t pid) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 1059 mNotificationClients.removeItem(pid); 1060 1061 ALOGV("%d died, releasing its sessions", pid); 1062 size_t num = mAudioSessionRefs.size(); 1063 bool removed = false; 1064 for (size_t i = 0; i< num; ) { 1065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1066 ALOGV(" pid %d @ %d", ref->mPid, i); 1067 if (ref->mPid == pid) { 1068 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1069 mAudioSessionRefs.removeAt(i); 1070 delete ref; 1071 removed = true; 1072 num--; 1073 } else { 1074 i++; 1075 } 1076 } 1077 if (removed) { 1078 purgeStaleEffects_l(); 1079 } 1080} 1081 1082// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1083void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1084{ 1085 size_t size = mNotificationClients.size(); 1086 for (size_t i = 0; i < size; i++) { 1087 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1088 param2); 1089 } 1090} 1091 1092// removeClient_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::removeClient_l(pid_t pid) 1094{ 1095 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1096 mClients.removeItem(pid); 1097} 1098 1099 1100// ---------------------------------------------------------------------------- 1101 1102AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1103 uint32_t device, type_t type) 1104 : Thread(false), 1105 mType(type), 1106 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1107 // mChannelMask 1108 mChannelCount(0), 1109 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1110 mParamStatus(NO_ERROR), 1111 mStandby(false), mId(id), 1112 mDevice(device), 1113 mDeathRecipient(new PMDeathRecipient(this)) 1114{ 1115} 1116 1117AudioFlinger::ThreadBase::~ThreadBase() 1118{ 1119 mParamCond.broadcast(); 1120 // do not lock the mutex in destructor 1121 releaseWakeLock_l(); 1122 if (mPowerManager != 0) { 1123 sp<IBinder> binder = mPowerManager->asBinder(); 1124 binder->unlinkToDeath(mDeathRecipient); 1125 } 1126} 1127 1128void AudioFlinger::ThreadBase::exit() 1129{ 1130 ALOGV("ThreadBase::exit"); 1131 { 1132 // This lock prevents the following race in thread (uniprocessor for illustration): 1133 // if (!exitPending()) { 1134 // // context switch from here to exit() 1135 // // exit() calls requestExit(), what exitPending() observes 1136 // // exit() calls signal(), which is dropped since no waiters 1137 // // context switch back from exit() to here 1138 // mWaitWorkCV.wait(...); 1139 // // now thread is hung 1140 // } 1141 AutoMutex lock(mLock); 1142 requestExit(); 1143 mWaitWorkCV.signal(); 1144 } 1145 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1146 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1147 requestExitAndWait(); 1148} 1149 1150status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1151{ 1152 status_t status; 1153 1154 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1155 Mutex::Autolock _l(mLock); 1156 1157 mNewParameters.add(keyValuePairs); 1158 mWaitWorkCV.signal(); 1159 // wait condition with timeout in case the thread loop has exited 1160 // before the request could be processed 1161 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1162 status = mParamStatus; 1163 mWaitWorkCV.signal(); 1164 } else { 1165 status = TIMED_OUT; 1166 } 1167 return status; 1168} 1169 1170void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1171{ 1172 Mutex::Autolock _l(mLock); 1173 sendConfigEvent_l(event, param); 1174} 1175 1176// sendConfigEvent_l() must be called with ThreadBase::mLock held 1177void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1178{ 1179 ConfigEvent configEvent; 1180 configEvent.mEvent = event; 1181 configEvent.mParam = param; 1182 mConfigEvents.add(configEvent); 1183 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1184 mWaitWorkCV.signal(); 1185} 1186 1187void AudioFlinger::ThreadBase::processConfigEvents() 1188{ 1189 mLock.lock(); 1190 while (!mConfigEvents.isEmpty()) { 1191 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1192 ConfigEvent configEvent = mConfigEvents[0]; 1193 mConfigEvents.removeAt(0); 1194 // release mLock before locking AudioFlinger mLock: lock order is always 1195 // AudioFlinger then ThreadBase to avoid cross deadlock 1196 mLock.unlock(); 1197 mAudioFlinger->mLock.lock(); 1198 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1199 mAudioFlinger->mLock.unlock(); 1200 mLock.lock(); 1201 } 1202 mLock.unlock(); 1203} 1204 1205status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1206{ 1207 const size_t SIZE = 256; 1208 char buffer[SIZE]; 1209 String8 result; 1210 1211 bool locked = tryLock(mLock); 1212 if (!locked) { 1213 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1214 write(fd, buffer, strlen(buffer)); 1215 } 1216 1217 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1218 result.append(buffer); 1219 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1236 result.append(buffer); 1237 1238 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1239 result.append(buffer); 1240 result.append(" Index Command"); 1241 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1242 snprintf(buffer, SIZE, "\n %02d ", i); 1243 result.append(buffer); 1244 result.append(mNewParameters[i]); 1245 } 1246 1247 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, " Index event param\n"); 1250 result.append(buffer); 1251 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1252 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1253 result.append(buffer); 1254 } 1255 result.append("\n"); 1256 1257 write(fd, result.string(), result.size()); 1258 1259 if (locked) { 1260 mLock.unlock(); 1261 } 1262 return NO_ERROR; 1263} 1264 1265status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1266{ 1267 const size_t SIZE = 256; 1268 char buffer[SIZE]; 1269 String8 result; 1270 1271 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1272 write(fd, buffer, strlen(buffer)); 1273 1274 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1275 sp<EffectChain> chain = mEffectChains[i]; 1276 if (chain != 0) { 1277 chain->dump(fd, args); 1278 } 1279 } 1280 return NO_ERROR; 1281} 1282 1283void AudioFlinger::ThreadBase::acquireWakeLock() 1284{ 1285 Mutex::Autolock _l(mLock); 1286 acquireWakeLock_l(); 1287} 1288 1289void AudioFlinger::ThreadBase::acquireWakeLock_l() 1290{ 1291 if (mPowerManager == 0) { 1292 // use checkService() to avoid blocking if power service is not up yet 1293 sp<IBinder> binder = 1294 defaultServiceManager()->checkService(String16("power")); 1295 if (binder == 0) { 1296 ALOGW("Thread %s cannot connect to the power manager service", mName); 1297 } else { 1298 mPowerManager = interface_cast<IPowerManager>(binder); 1299 binder->linkToDeath(mDeathRecipient); 1300 } 1301 } 1302 if (mPowerManager != 0) { 1303 sp<IBinder> binder = new BBinder(); 1304 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1305 binder, 1306 String16(mName)); 1307 if (status == NO_ERROR) { 1308 mWakeLockToken = binder; 1309 } 1310 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::releaseWakeLock() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 releaseWakeLock_l(); 1318} 1319 1320void AudioFlinger::ThreadBase::releaseWakeLock_l() 1321{ 1322 if (mWakeLockToken != 0) { 1323 ALOGV("releaseWakeLock_l() %s", mName); 1324 if (mPowerManager != 0) { 1325 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1326 } 1327 mWakeLockToken.clear(); 1328 } 1329} 1330 1331void AudioFlinger::ThreadBase::clearPowerManager() 1332{ 1333 Mutex::Autolock _l(mLock); 1334 releaseWakeLock_l(); 1335 mPowerManager.clear(); 1336} 1337 1338void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1339{ 1340 sp<ThreadBase> thread = mThread.promote(); 1341 if (thread != 0) { 1342 thread->clearPowerManager(); 1343 } 1344 ALOGW("power manager service died !!!"); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 setEffectSuspended_l(type, suspend, sessionId); 1352} 1353 1354void AudioFlinger::ThreadBase::setEffectSuspended_l( 1355 const effect_uuid_t *type, bool suspend, int sessionId) 1356{ 1357 sp<EffectChain> chain = getEffectChain_l(sessionId); 1358 if (chain != 0) { 1359 if (type != NULL) { 1360 chain->setEffectSuspended_l(type, suspend); 1361 } else { 1362 chain->setEffectSuspendedAll_l(suspend); 1363 } 1364 } 1365 1366 updateSuspendedSessions_l(type, suspend, sessionId); 1367} 1368 1369void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1370{ 1371 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1372 if (index < 0) { 1373 return; 1374 } 1375 1376 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1377 mSuspendedSessions.editValueAt(index); 1378 1379 for (size_t i = 0; i < sessionEffects.size(); i++) { 1380 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1381 for (int j = 0; j < desc->mRefCount; j++) { 1382 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1383 chain->setEffectSuspendedAll_l(true); 1384 } else { 1385 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1386 desc->mType.timeLow); 1387 chain->setEffectSuspended_l(&desc->mType, true); 1388 } 1389 } 1390 } 1391} 1392 1393void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1394 bool suspend, 1395 int sessionId) 1396{ 1397 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1398 1399 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1400 1401 if (suspend) { 1402 if (index >= 0) { 1403 sessionEffects = mSuspendedSessions.editValueAt(index); 1404 } else { 1405 mSuspendedSessions.add(sessionId, sessionEffects); 1406 } 1407 } else { 1408 if (index < 0) { 1409 return; 1410 } 1411 sessionEffects = mSuspendedSessions.editValueAt(index); 1412 } 1413 1414 1415 int key = EffectChain::kKeyForSuspendAll; 1416 if (type != NULL) { 1417 key = type->timeLow; 1418 } 1419 index = sessionEffects.indexOfKey(key); 1420 1421 sp<SuspendedSessionDesc> desc; 1422 if (suspend) { 1423 if (index >= 0) { 1424 desc = sessionEffects.valueAt(index); 1425 } else { 1426 desc = new SuspendedSessionDesc(); 1427 if (type != NULL) { 1428 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1429 } 1430 sessionEffects.add(key, desc); 1431 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1432 } 1433 desc->mRefCount++; 1434 } else { 1435 if (index < 0) { 1436 return; 1437 } 1438 desc = sessionEffects.valueAt(index); 1439 if (--desc->mRefCount == 0) { 1440 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1441 sessionEffects.removeItemsAt(index); 1442 if (sessionEffects.isEmpty()) { 1443 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1444 sessionId); 1445 mSuspendedSessions.removeItem(sessionId); 1446 } 1447 } 1448 } 1449 if (!sessionEffects.isEmpty()) { 1450 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1451 } 1452} 1453 1454void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1455 bool enabled, 1456 int sessionId) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1460} 1461 1462void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1463 bool enabled, 1464 int sessionId) 1465{ 1466 if (mType != RECORD) { 1467 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1468 // another session. This gives the priority to well behaved effect control panels 1469 // and applications not using global effects. 1470 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1471 // global effects 1472 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1473 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1474 } 1475 } 1476 1477 sp<EffectChain> chain = getEffectChain_l(sessionId); 1478 if (chain != 0) { 1479 chain->checkSuspendOnEffectEnabled(effect, enabled); 1480 } 1481} 1482 1483// ---------------------------------------------------------------------------- 1484 1485AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1486 AudioStreamOut* output, 1487 audio_io_handle_t id, 1488 uint32_t device, 1489 type_t type) 1490 : ThreadBase(audioFlinger, id, device, type), 1491 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1492 // Assumes constructor is called by AudioFlinger with it's mLock held, 1493 // but it would be safer to explicitly pass initial masterMute as parameter 1494 mMasterMute(audioFlinger->masterMute_l()), 1495 // mStreamTypes[] initialized in constructor body 1496 mOutput(output), 1497 // Assumes constructor is called by AudioFlinger with it's mLock held, 1498 // but it would be safer to explicitly pass initial masterVolume as parameter 1499 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1500 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1501 mMixerStatus(MIXER_IDLE), 1502 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1503 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1504 // index 0 is reserved for normal mixer's submix 1505 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1506{ 1507 snprintf(mName, kNameLength, "AudioOut_%X", id); 1508 1509 readOutputParameters(); 1510 1511 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1512 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1513 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1514 stream = (audio_stream_type_t) (stream + 1)) { 1515 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1516 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1517 } 1518 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1519 // because mAudioFlinger doesn't have one to copy from 1520} 1521 1522AudioFlinger::PlaybackThread::~PlaybackThread() 1523{ 1524 delete [] mMixBuffer; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1528{ 1529 dumpInternals(fd, args); 1530 dumpTracks(fd, args); 1531 dumpEffectChains(fd, args); 1532 return NO_ERROR; 1533} 1534 1535status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1536{ 1537 const size_t SIZE = 256; 1538 char buffer[SIZE]; 1539 String8 result; 1540 1541 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1542 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1543 const stream_type_t *st = &mStreamTypes[i]; 1544 if (i > 0) { 1545 result.appendFormat(", "); 1546 } 1547 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1548 if (st->mute) { 1549 result.append("M"); 1550 } 1551 } 1552 result.append("\n"); 1553 write(fd, result.string(), result.length()); 1554 result.clear(); 1555 1556 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1557 result.append(buffer); 1558 Track::appendDumpHeader(result); 1559 for (size_t i = 0; i < mTracks.size(); ++i) { 1560 sp<Track> track = mTracks[i]; 1561 if (track != 0) { 1562 track->dump(buffer, SIZE); 1563 result.append(buffer); 1564 } 1565 } 1566 1567 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1571 sp<Track> track = mActiveTracks[i].promote(); 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 write(fd, result.string(), result.size()); 1578 1579 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1580 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1581 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1582 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1583 1584 return NO_ERROR; 1585} 1586 1587status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1588{ 1589 const size_t SIZE = 256; 1590 char buffer[SIZE]; 1591 String8 result; 1592 1593 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1606 result.append(buffer); 1607 write(fd, result.string(), result.size()); 1608 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1609 1610 dumpBase(fd, args); 1611 1612 return NO_ERROR; 1613} 1614 1615// Thread virtuals 1616status_t AudioFlinger::PlaybackThread::readyToRun() 1617{ 1618 status_t status = initCheck(); 1619 if (status == NO_ERROR) { 1620 ALOGI("AudioFlinger's thread %p ready to run", this); 1621 } else { 1622 ALOGE("No working audio driver found."); 1623 } 1624 return status; 1625} 1626 1627void AudioFlinger::PlaybackThread::onFirstRef() 1628{ 1629 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1630} 1631 1632// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1633sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1634 const sp<AudioFlinger::Client>& client, 1635 audio_stream_type_t streamType, 1636 uint32_t sampleRate, 1637 audio_format_t format, 1638 uint32_t channelMask, 1639 int frameCount, 1640 const sp<IMemory>& sharedBuffer, 1641 int sessionId, 1642 IAudioFlinger::track_flags_t flags, 1643 pid_t tid, 1644 status_t *status) 1645{ 1646 sp<Track> track; 1647 status_t lStatus; 1648 1649 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1650 1651 // client expresses a preference for FAST, but we get the final say 1652 if (flags & IAudioFlinger::TRACK_FAST) { 1653 if ( 1654 // not timed 1655 (!isTimed) && 1656 // either of these use cases: 1657 ( 1658 // use case 1: shared buffer with any frame count 1659 ( 1660 (sharedBuffer != 0) 1661 ) || 1662 // use case 2: callback handler and frame count is default or at least as large as HAL 1663 ( 1664 (tid != -1) && 1665 ((frameCount == 0) || 1666 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1667 ) 1668 ) && 1669 // PCM data 1670 audio_is_linear_pcm(format) && 1671 // mono or stereo 1672 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1673 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1674#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1675 // hardware sample rate 1676 (sampleRate == mSampleRate) && 1677#endif 1678 // normal mixer has an associated fast mixer 1679 hasFastMixer() && 1680 // there are sufficient fast track slots available 1681 (mFastTrackAvailMask != 0) 1682 // FIXME test that MixerThread for this fast track has a capable output HAL 1683 // FIXME add a permission test also? 1684 ) { 1685 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1686 if (frameCount == 0) { 1687 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1688 } 1689 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1690 frameCount, mFrameCount); 1691 } else { 1692 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1693 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1694 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1695 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1696 audio_is_linear_pcm(format), 1697 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1698 flags &= ~IAudioFlinger::TRACK_FAST; 1699 // For compatibility with AudioTrack calculation, buffer depth is forced 1700 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1701 // This is probably too conservative, but legacy application code may depend on it. 1702 // If you change this calculation, also review the start threshold which is related. 1703 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1704 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1705 if (minBufCount < 2) { 1706 minBufCount = 2; 1707 } 1708 int minFrameCount = mNormalFrameCount * minBufCount; 1709 if (frameCount < minFrameCount) { 1710 frameCount = minFrameCount; 1711 } 1712 } 1713 } 1714 1715 if (mType == DIRECT) { 1716 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1717 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1718 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1719 "for output %p with format %d", 1720 sampleRate, format, channelMask, mOutput, mFormat); 1721 lStatus = BAD_VALUE; 1722 goto Exit; 1723 } 1724 } 1725 } else { 1726 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1727 if (sampleRate > mSampleRate*2) { 1728 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1729 lStatus = BAD_VALUE; 1730 goto Exit; 1731 } 1732 } 1733 1734 lStatus = initCheck(); 1735 if (lStatus != NO_ERROR) { 1736 ALOGE("Audio driver not initialized."); 1737 goto Exit; 1738 } 1739 1740 { // scope for mLock 1741 Mutex::Autolock _l(mLock); 1742 1743 // all tracks in same audio session must share the same routing strategy otherwise 1744 // conflicts will happen when tracks are moved from one output to another by audio policy 1745 // manager 1746 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1747 for (size_t i = 0; i < mTracks.size(); ++i) { 1748 sp<Track> t = mTracks[i]; 1749 if (t != 0 && !t->isOutputTrack()) { 1750 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1751 if (sessionId == t->sessionId() && strategy != actual) { 1752 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1753 strategy, actual); 1754 lStatus = BAD_VALUE; 1755 goto Exit; 1756 } 1757 } 1758 } 1759 1760 if (!isTimed) { 1761 track = new Track(this, client, streamType, sampleRate, format, 1762 channelMask, frameCount, sharedBuffer, sessionId, flags); 1763 } else { 1764 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1765 channelMask, frameCount, sharedBuffer, sessionId); 1766 } 1767 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1768 lStatus = NO_MEMORY; 1769 goto Exit; 1770 } 1771 mTracks.add(track); 1772 1773 sp<EffectChain> chain = getEffectChain_l(sessionId); 1774 if (chain != 0) { 1775 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1776 track->setMainBuffer(chain->inBuffer()); 1777 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1778 chain->incTrackCnt(); 1779 } 1780 } 1781 1782#ifdef HAVE_REQUEST_PRIORITY 1783 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1784 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1785 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1786 // so ask activity manager to do this on our behalf 1787 int err = requestPriority(callingPid, tid, 1); 1788 if (err != 0) { 1789 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1790 1, callingPid, tid, err); 1791 } 1792 } 1793#endif 1794 1795 lStatus = NO_ERROR; 1796 1797Exit: 1798 if (status) { 1799 *status = lStatus; 1800 } 1801 return track; 1802} 1803 1804uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1805{ 1806 if (mFastMixer != NULL) { 1807 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1808 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1809 } 1810 return latency; 1811} 1812 1813uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1814{ 1815 return latency; 1816} 1817 1818uint32_t AudioFlinger::PlaybackThread::latency() const 1819{ 1820 Mutex::Autolock _l(mLock); 1821 return latency_l(); 1822} 1823uint32_t AudioFlinger::PlaybackThread::latency_l() const 1824{ 1825 if (initCheck() == NO_ERROR) { 1826 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1827 } else { 1828 return 0; 1829 } 1830} 1831 1832void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1833{ 1834 Mutex::Autolock _l(mLock); 1835 mMasterVolume = value; 1836} 1837 1838void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1839{ 1840 Mutex::Autolock _l(mLock); 1841 setMasterMute_l(muted); 1842} 1843 1844void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 mStreamTypes[stream].volume = value; 1848} 1849 1850void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1851{ 1852 Mutex::Autolock _l(mLock); 1853 mStreamTypes[stream].mute = muted; 1854} 1855 1856float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1857{ 1858 Mutex::Autolock _l(mLock); 1859 return mStreamTypes[stream].volume; 1860} 1861 1862// addTrack_l() must be called with ThreadBase::mLock held 1863status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1864{ 1865 status_t status = ALREADY_EXISTS; 1866 1867 // set retry count for buffer fill 1868 track->mRetryCount = kMaxTrackStartupRetries; 1869 if (mActiveTracks.indexOf(track) < 0) { 1870 // the track is newly added, make sure it fills up all its 1871 // buffers before playing. This is to ensure the client will 1872 // effectively get the latency it requested. 1873 track->mFillingUpStatus = Track::FS_FILLING; 1874 track->mResetDone = false; 1875 track->mPresentationCompleteFrames = 0; 1876 mActiveTracks.add(track); 1877 if (track->mainBuffer() != mMixBuffer) { 1878 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1879 if (chain != 0) { 1880 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1881 chain->incActiveTrackCnt(); 1882 } 1883 } 1884 1885 status = NO_ERROR; 1886 } 1887 1888 ALOGV("mWaitWorkCV.broadcast"); 1889 mWaitWorkCV.broadcast(); 1890 1891 return status; 1892} 1893 1894// destroyTrack_l() must be called with ThreadBase::mLock held 1895void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1896{ 1897 track->mState = TrackBase::TERMINATED; 1898 // active tracks are removed by threadLoop() 1899 if (mActiveTracks.indexOf(track) < 0) { 1900 removeTrack_l(track); 1901 } 1902} 1903 1904void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1905{ 1906 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1907 mTracks.remove(track); 1908 deleteTrackName_l(track->name()); 1909 // redundant as track is about to be destroyed, for dumpsys only 1910 track->mName = -1; 1911 if (track->isFastTrack()) { 1912 int index = track->mFastIndex; 1913 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1914 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1915 mFastTrackAvailMask |= 1 << index; 1916 // redundant as track is about to be destroyed, for dumpsys only 1917 track->mFastIndex = -1; 1918 } 1919 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1920 if (chain != 0) { 1921 chain->decTrackCnt(); 1922 } 1923} 1924 1925String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1926{ 1927 String8 out_s8 = String8(""); 1928 char *s; 1929 1930 Mutex::Autolock _l(mLock); 1931 if (initCheck() != NO_ERROR) { 1932 return out_s8; 1933 } 1934 1935 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1936 out_s8 = String8(s); 1937 free(s); 1938 return out_s8; 1939} 1940 1941// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1942void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1943 AudioSystem::OutputDescriptor desc; 1944 void *param2 = NULL; 1945 1946 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1947 1948 switch (event) { 1949 case AudioSystem::OUTPUT_OPENED: 1950 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1951 desc.channels = mChannelMask; 1952 desc.samplingRate = mSampleRate; 1953 desc.format = mFormat; 1954 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1955 desc.latency = latency(); 1956 param2 = &desc; 1957 break; 1958 1959 case AudioSystem::STREAM_CONFIG_CHANGED: 1960 param2 = ¶m; 1961 case AudioSystem::OUTPUT_CLOSED: 1962 default: 1963 break; 1964 } 1965 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1966} 1967 1968void AudioFlinger::PlaybackThread::readOutputParameters() 1969{ 1970 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1971 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1972 mChannelCount = (uint16_t)popcount(mChannelMask); 1973 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1974 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1975 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1976 if (mFrameCount & 15) { 1977 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1978 mFrameCount); 1979 } 1980 1981 // Calculate size of normal mix buffer relative to the HAL output buffer size 1982 double multiplier = 1.0; 1983 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1984 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1985 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1986 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1987 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1988 maxNormalFrameCount = maxNormalFrameCount & ~15; 1989 if (maxNormalFrameCount < minNormalFrameCount) { 1990 maxNormalFrameCount = minNormalFrameCount; 1991 } 1992 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1993 if (multiplier <= 1.0) { 1994 multiplier = 1.0; 1995 } else if (multiplier <= 2.0) { 1996 if (2 * mFrameCount <= maxNormalFrameCount) { 1997 multiplier = 2.0; 1998 } else { 1999 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2000 } 2001 } else { 2002 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2003 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2004 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2005 // FIXME this rounding up should not be done if no HAL SRC 2006 uint32_t truncMult = (uint32_t) multiplier; 2007 if ((truncMult & 1)) { 2008 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2009 ++truncMult; 2010 } 2011 } 2012 multiplier = (double) truncMult; 2013 } 2014 } 2015 mNormalFrameCount = multiplier * mFrameCount; 2016 // round up to nearest 16 frames to satisfy AudioMixer 2017 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2018 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2019 2020 delete[] mMixBuffer; 2021 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2022 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2023 2024 // force reconfiguration of effect chains and engines to take new buffer size and audio 2025 // parameters into account 2026 // Note that mLock is not held when readOutputParameters() is called from the constructor 2027 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2028 // matter. 2029 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2030 Vector< sp<EffectChain> > effectChains = mEffectChains; 2031 for (size_t i = 0; i < effectChains.size(); i ++) { 2032 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2033 } 2034} 2035 2036 2037status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2038{ 2039 if (halFrames == NULL || dspFrames == NULL) { 2040 return BAD_VALUE; 2041 } 2042 Mutex::Autolock _l(mLock); 2043 if (initCheck() != NO_ERROR) { 2044 return INVALID_OPERATION; 2045 } 2046 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2047 2048 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2049} 2050 2051uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2052{ 2053 Mutex::Autolock _l(mLock); 2054 uint32_t result = 0; 2055 if (getEffectChain_l(sessionId) != 0) { 2056 result = EFFECT_SESSION; 2057 } 2058 2059 for (size_t i = 0; i < mTracks.size(); ++i) { 2060 sp<Track> track = mTracks[i]; 2061 if (sessionId == track->sessionId() && 2062 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2063 result |= TRACK_SESSION; 2064 break; 2065 } 2066 } 2067 2068 return result; 2069} 2070 2071uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2072{ 2073 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2074 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2075 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2076 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2077 } 2078 for (size_t i = 0; i < mTracks.size(); i++) { 2079 sp<Track> track = mTracks[i]; 2080 if (sessionId == track->sessionId() && 2081 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2082 return AudioSystem::getStrategyForStream(track->streamType()); 2083 } 2084 } 2085 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2086} 2087 2088 2089AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2090{ 2091 Mutex::Autolock _l(mLock); 2092 return mOutput; 2093} 2094 2095AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2096{ 2097 Mutex::Autolock _l(mLock); 2098 AudioStreamOut *output = mOutput; 2099 mOutput = NULL; 2100 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2101 // must push a NULL and wait for ack 2102 mOutputSink.clear(); 2103 mPipeSink.clear(); 2104 mNormalSink.clear(); 2105 return output; 2106} 2107 2108// this method must always be called either with ThreadBase mLock held or inside the thread loop 2109audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2110{ 2111 if (mOutput == NULL) { 2112 return NULL; 2113 } 2114 return &mOutput->stream->common; 2115} 2116 2117uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2118{ 2119 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2120} 2121 2122status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2123{ 2124 if (!isValidSyncEvent(event)) { 2125 return BAD_VALUE; 2126 } 2127 2128 Mutex::Autolock _l(mLock); 2129 2130 for (size_t i = 0; i < mTracks.size(); ++i) { 2131 sp<Track> track = mTracks[i]; 2132 if (event->triggerSession() == track->sessionId()) { 2133 track->setSyncEvent(event); 2134 return NO_ERROR; 2135 } 2136 } 2137 2138 return NAME_NOT_FOUND; 2139} 2140 2141bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2142{ 2143 switch (event->type()) { 2144 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2145 return true; 2146 default: 2147 break; 2148 } 2149 return false; 2150} 2151 2152void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2153{ 2154 size_t count = tracksToRemove.size(); 2155 if (CC_UNLIKELY(count)) { 2156 for (size_t i = 0 ; i < count ; i++) { 2157 const sp<Track>& track = tracksToRemove.itemAt(i); 2158 if ((track->sharedBuffer() != 0) && 2159 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2160 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2161 } 2162 } 2163 } 2164 2165} 2166 2167// ---------------------------------------------------------------------------- 2168 2169AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2170 audio_io_handle_t id, uint32_t device, type_t type) 2171 : PlaybackThread(audioFlinger, output, id, device, type), 2172 // mAudioMixer below 2173#ifdef SOAKER 2174 mSoaker(NULL), 2175#endif 2176 // mFastMixer below 2177 mFastMixerFutex(0) 2178 // mOutputSink below 2179 // mPipeSink below 2180 // mNormalSink below 2181{ 2182 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2183 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2184 "mFrameCount=%d, mNormalFrameCount=%d", 2185 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2186 mNormalFrameCount); 2187 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2188 2189 // FIXME - Current mixer implementation only supports stereo output 2190 if (mChannelCount == 1) { 2191 ALOGE("Invalid audio hardware channel count"); 2192 } 2193 2194 // create an NBAIO sink for the HAL output stream, and negotiate 2195 mOutputSink = new AudioStreamOutSink(output->stream); 2196 size_t numCounterOffers = 0; 2197 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2198 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2199 ALOG_ASSERT(index == 0); 2200 2201 // initialize fast mixer depending on configuration 2202 bool initFastMixer; 2203 switch (kUseFastMixer) { 2204 case FastMixer_Never: 2205 initFastMixer = false; 2206 break; 2207 case FastMixer_Always: 2208 initFastMixer = true; 2209 break; 2210 case FastMixer_Static: 2211 case FastMixer_Dynamic: 2212 initFastMixer = mFrameCount < mNormalFrameCount; 2213 break; 2214 } 2215 if (initFastMixer) { 2216 2217 // create a MonoPipe to connect our submix to FastMixer 2218 NBAIO_Format format = mOutputSink->format(); 2219 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2220 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2221 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2222 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2223 const NBAIO_Format offers[1] = {format}; 2224 size_t numCounterOffers = 0; 2225 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2226 ALOG_ASSERT(index == 0); 2227 mPipeSink = monoPipe; 2228 2229#ifdef TEE_SINK_FRAMES 2230 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2231 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2232 numCounterOffers = 0; 2233 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2234 ALOG_ASSERT(index == 0); 2235 mTeeSink = teeSink; 2236 PipeReader *teeSource = new PipeReader(*teeSink); 2237 numCounterOffers = 0; 2238 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2239 ALOG_ASSERT(index == 0); 2240 mTeeSource = teeSource; 2241#endif 2242 2243#ifdef SOAKER 2244 // create a soaker as workaround for governor issues 2245 mSoaker = new Soaker(); 2246 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2247 mSoaker->run("Soaker", PRIORITY_LOWEST); 2248#endif 2249 2250 // create fast mixer and configure it initially with just one fast track for our submix 2251 mFastMixer = new FastMixer(); 2252 FastMixerStateQueue *sq = mFastMixer->sq(); 2253#ifdef STATE_QUEUE_DUMP 2254 sq->setObserverDump(&mStateQueueObserverDump); 2255 sq->setMutatorDump(&mStateQueueMutatorDump); 2256#endif 2257 FastMixerState *state = sq->begin(); 2258 FastTrack *fastTrack = &state->mFastTracks[0]; 2259 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2260 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2261 fastTrack->mVolumeProvider = NULL; 2262 fastTrack->mGeneration++; 2263 state->mFastTracksGen++; 2264 state->mTrackMask = 1; 2265 // fast mixer will use the HAL output sink 2266 state->mOutputSink = mOutputSink.get(); 2267 state->mOutputSinkGen++; 2268 state->mFrameCount = mFrameCount; 2269 state->mCommand = FastMixerState::COLD_IDLE; 2270 // already done in constructor initialization list 2271 //mFastMixerFutex = 0; 2272 state->mColdFutexAddr = &mFastMixerFutex; 2273 state->mColdGen++; 2274 state->mDumpState = &mFastMixerDumpState; 2275 state->mTeeSink = mTeeSink.get(); 2276 sq->end(); 2277 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2278 2279 // start the fast mixer 2280 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2281#ifdef HAVE_REQUEST_PRIORITY 2282 pid_t tid = mFastMixer->getTid(); 2283 int err = requestPriority(getpid_cached, tid, 2); 2284 if (err != 0) { 2285 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2286 2, getpid_cached, tid, err); 2287 } 2288#endif 2289 2290 } else { 2291 mFastMixer = NULL; 2292 } 2293 2294 switch (kUseFastMixer) { 2295 case FastMixer_Never: 2296 case FastMixer_Dynamic: 2297 mNormalSink = mOutputSink; 2298 break; 2299 case FastMixer_Always: 2300 mNormalSink = mPipeSink; 2301 break; 2302 case FastMixer_Static: 2303 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2304 break; 2305 } 2306} 2307 2308AudioFlinger::MixerThread::~MixerThread() 2309{ 2310 if (mFastMixer != NULL) { 2311 FastMixerStateQueue *sq = mFastMixer->sq(); 2312 FastMixerState *state = sq->begin(); 2313 if (state->mCommand == FastMixerState::COLD_IDLE) { 2314 int32_t old = android_atomic_inc(&mFastMixerFutex); 2315 if (old == -1) { 2316 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2317 } 2318 } 2319 state->mCommand = FastMixerState::EXIT; 2320 sq->end(); 2321 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2322 mFastMixer->join(); 2323 // Though the fast mixer thread has exited, it's state queue is still valid. 2324 // We'll use that extract the final state which contains one remaining fast track 2325 // corresponding to our sub-mix. 2326 state = sq->begin(); 2327 ALOG_ASSERT(state->mTrackMask == 1); 2328 FastTrack *fastTrack = &state->mFastTracks[0]; 2329 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2330 delete fastTrack->mBufferProvider; 2331 sq->end(false /*didModify*/); 2332 delete mFastMixer; 2333#ifdef SOAKER 2334 if (mSoaker != NULL) { 2335 mSoaker->requestExitAndWait(); 2336 } 2337 delete mSoaker; 2338#endif 2339 } 2340 delete mAudioMixer; 2341} 2342 2343class CpuStats { 2344public: 2345 CpuStats(); 2346 void sample(const String8 &title); 2347#ifdef DEBUG_CPU_USAGE 2348private: 2349 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2350 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2351 2352 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2353 2354 int mCpuNum; // thread's current CPU number 2355 int mCpukHz; // frequency of thread's current CPU in kHz 2356#endif 2357}; 2358 2359CpuStats::CpuStats() 2360#ifdef DEBUG_CPU_USAGE 2361 : mCpuNum(-1), mCpukHz(-1) 2362#endif 2363{ 2364} 2365 2366void CpuStats::sample(const String8 &title) { 2367#ifdef DEBUG_CPU_USAGE 2368 // get current thread's delta CPU time in wall clock ns 2369 double wcNs; 2370 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2371 2372 // record sample for wall clock statistics 2373 if (valid) { 2374 mWcStats.sample(wcNs); 2375 } 2376 2377 // get the current CPU number 2378 int cpuNum = sched_getcpu(); 2379 2380 // get the current CPU frequency in kHz 2381 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2382 2383 // check if either CPU number or frequency changed 2384 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2385 mCpuNum = cpuNum; 2386 mCpukHz = cpukHz; 2387 // ignore sample for purposes of cycles 2388 valid = false; 2389 } 2390 2391 // if no change in CPU number or frequency, then record sample for cycle statistics 2392 if (valid && mCpukHz > 0) { 2393 double cycles = wcNs * cpukHz * 0.000001; 2394 mHzStats.sample(cycles); 2395 } 2396 2397 unsigned n = mWcStats.n(); 2398 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2399 if ((n & 127) == 1) { 2400 long long elapsed = mCpuUsage.elapsed(); 2401 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2402 double perLoop = elapsed / (double) n; 2403 double perLoop100 = perLoop * 0.01; 2404 double perLoop1k = perLoop * 0.001; 2405 double mean = mWcStats.mean(); 2406 double stddev = mWcStats.stddev(); 2407 double minimum = mWcStats.minimum(); 2408 double maximum = mWcStats.maximum(); 2409 double meanCycles = mHzStats.mean(); 2410 double stddevCycles = mHzStats.stddev(); 2411 double minCycles = mHzStats.minimum(); 2412 double maxCycles = mHzStats.maximum(); 2413 mCpuUsage.resetElapsed(); 2414 mWcStats.reset(); 2415 mHzStats.reset(); 2416 ALOGD("CPU usage for %s over past %.1f secs\n" 2417 " (%u mixer loops at %.1f mean ms per loop):\n" 2418 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2419 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2420 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2421 title.string(), 2422 elapsed * .000000001, n, perLoop * .000001, 2423 mean * .001, 2424 stddev * .001, 2425 minimum * .001, 2426 maximum * .001, 2427 mean / perLoop100, 2428 stddev / perLoop100, 2429 minimum / perLoop100, 2430 maximum / perLoop100, 2431 meanCycles / perLoop1k, 2432 stddevCycles / perLoop1k, 2433 minCycles / perLoop1k, 2434 maxCycles / perLoop1k); 2435 2436 } 2437 } 2438#endif 2439}; 2440 2441void AudioFlinger::PlaybackThread::checkSilentMode_l() 2442{ 2443 if (!mMasterMute) { 2444 char value[PROPERTY_VALUE_MAX]; 2445 if (property_get("ro.audio.silent", value, "0") > 0) { 2446 char *endptr; 2447 unsigned long ul = strtoul(value, &endptr, 0); 2448 if (*endptr == '\0' && ul != 0) { 2449 ALOGD("Silence is golden"); 2450 // The setprop command will not allow a property to be changed after 2451 // the first time it is set, so we don't have to worry about un-muting. 2452 setMasterMute_l(true); 2453 } 2454 } 2455 } 2456} 2457 2458bool AudioFlinger::PlaybackThread::threadLoop() 2459{ 2460 Vector< sp<Track> > tracksToRemove; 2461 2462 standbyTime = systemTime(); 2463 2464 // MIXER 2465 nsecs_t lastWarning = 0; 2466if (mType == MIXER) { 2467 longStandbyExit = false; 2468} 2469 2470 // DUPLICATING 2471 // FIXME could this be made local to while loop? 2472 writeFrames = 0; 2473 2474 cacheParameters_l(); 2475 sleepTime = idleSleepTime; 2476 2477if (mType == MIXER) { 2478 sleepTimeShift = 0; 2479} 2480 2481 CpuStats cpuStats; 2482 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2483 2484 acquireWakeLock(); 2485 2486 while (!exitPending()) 2487 { 2488 cpuStats.sample(myName); 2489 2490 Vector< sp<EffectChain> > effectChains; 2491 2492 processConfigEvents(); 2493 2494 { // scope for mLock 2495 2496 Mutex::Autolock _l(mLock); 2497 2498 if (checkForNewParameters_l()) { 2499 cacheParameters_l(); 2500 } 2501 2502 saveOutputTracks(); 2503 2504 // put audio hardware into standby after short delay 2505 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2506 mSuspended > 0)) { 2507 if (!mStandby) { 2508 2509 threadLoop_standby(); 2510 2511 mStandby = true; 2512 mBytesWritten = 0; 2513 } 2514 2515 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2516 // we're about to wait, flush the binder command buffer 2517 IPCThreadState::self()->flushCommands(); 2518 2519 clearOutputTracks(); 2520 2521 if (exitPending()) break; 2522 2523 releaseWakeLock_l(); 2524 // wait until we have something to do... 2525 ALOGV("%s going to sleep", myName.string()); 2526 mWaitWorkCV.wait(mLock); 2527 ALOGV("%s waking up", myName.string()); 2528 acquireWakeLock_l(); 2529 2530 mMixerStatus = MIXER_IDLE; 2531 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2532 2533 checkSilentMode_l(); 2534 2535 standbyTime = systemTime() + standbyDelay; 2536 sleepTime = idleSleepTime; 2537 if (mType == MIXER) { 2538 sleepTimeShift = 0; 2539 } 2540 2541 continue; 2542 } 2543 } 2544 2545 // mMixerStatusIgnoringFastTracks is also updated internally 2546 mMixerStatus = prepareTracks_l(&tracksToRemove); 2547 2548 // prevent any changes in effect chain list and in each effect chain 2549 // during mixing and effect process as the audio buffers could be deleted 2550 // or modified if an effect is created or deleted 2551 lockEffectChains_l(effectChains); 2552 } 2553 2554 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2555 threadLoop_mix(); 2556 } else { 2557 threadLoop_sleepTime(); 2558 } 2559 2560 if (mSuspended > 0) { 2561 sleepTime = suspendSleepTimeUs(); 2562 } 2563 2564 // only process effects if we're going to write 2565 if (sleepTime == 0) { 2566 for (size_t i = 0; i < effectChains.size(); i ++) { 2567 effectChains[i]->process_l(); 2568 } 2569 } 2570 2571 // enable changes in effect chain 2572 unlockEffectChains(effectChains); 2573 2574 // sleepTime == 0 means we must write to audio hardware 2575 if (sleepTime == 0) { 2576 2577 threadLoop_write(); 2578 2579if (mType == MIXER) { 2580 // write blocked detection 2581 nsecs_t now = systemTime(); 2582 nsecs_t delta = now - mLastWriteTime; 2583 if (!mStandby && delta > maxPeriod) { 2584 mNumDelayedWrites++; 2585 if ((now - lastWarning) > kWarningThrottleNs) { 2586#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2587 ScopedTrace st(ATRACE_TAG, "underrun"); 2588#endif 2589 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2590 ns2ms(delta), mNumDelayedWrites, this); 2591 lastWarning = now; 2592 } 2593 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2594 // a different threshold. Or completely removed for what it is worth anyway... 2595 if (mStandby) { 2596 longStandbyExit = true; 2597 } 2598 } 2599} 2600 2601 mStandby = false; 2602 } else { 2603 usleep(sleepTime); 2604 } 2605 2606 // Finally let go of removed track(s), without the lock held 2607 // since we can't guarantee the destructors won't acquire that 2608 // same lock. This will also mutate and push a new fast mixer state. 2609 threadLoop_removeTracks(tracksToRemove); 2610 tracksToRemove.clear(); 2611 2612 // FIXME I don't understand the need for this here; 2613 // it was in the original code but maybe the 2614 // assignment in saveOutputTracks() makes this unnecessary? 2615 clearOutputTracks(); 2616 2617 // Effect chains will be actually deleted here if they were removed from 2618 // mEffectChains list during mixing or effects processing 2619 effectChains.clear(); 2620 2621 // FIXME Note that the above .clear() is no longer necessary since effectChains 2622 // is now local to this block, but will keep it for now (at least until merge done). 2623 } 2624 2625if (mType == MIXER || mType == DIRECT) { 2626 // put output stream into standby mode 2627 if (!mStandby) { 2628 mOutput->stream->common.standby(&mOutput->stream->common); 2629 } 2630} 2631if (mType == DUPLICATING) { 2632 // for DuplicatingThread, standby mode is handled by the outputTracks 2633} 2634 2635 releaseWakeLock(); 2636 2637 ALOGV("Thread %p type %d exiting", this, mType); 2638 return false; 2639} 2640 2641void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2642{ 2643 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_write() 2647{ 2648 // FIXME we should only do one push per cycle; confirm this is true 2649 // Start the fast mixer if it's not already running 2650 if (mFastMixer != NULL) { 2651 FastMixerStateQueue *sq = mFastMixer->sq(); 2652 FastMixerState *state = sq->begin(); 2653 if (state->mCommand != FastMixerState::MIX_WRITE && 2654 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2655 if (state->mCommand == FastMixerState::COLD_IDLE) { 2656 int32_t old = android_atomic_inc(&mFastMixerFutex); 2657 if (old == -1) { 2658 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2659 } 2660 } 2661 state->mCommand = FastMixerState::MIX_WRITE; 2662 sq->end(); 2663 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2664 if (kUseFastMixer == FastMixer_Dynamic) { 2665 mNormalSink = mPipeSink; 2666 } 2667 } else { 2668 sq->end(false /*didModify*/); 2669 } 2670 } 2671 PlaybackThread::threadLoop_write(); 2672} 2673 2674// shared by MIXER and DIRECT, overridden by DUPLICATING 2675void AudioFlinger::PlaybackThread::threadLoop_write() 2676{ 2677 // FIXME rewrite to reduce number of system calls 2678 mLastWriteTime = systemTime(); 2679 mInWrite = true; 2680 int bytesWritten; 2681 2682 // If an NBAIO sink is present, use it to write the normal mixer's submix 2683 if (mNormalSink != 0) { 2684#define mBitShift 2 // FIXME 2685 size_t count = mixBufferSize >> mBitShift; 2686#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2687 Tracer::traceBegin(ATRACE_TAG, "write"); 2688#endif 2689 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2690#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2691 Tracer::traceEnd(ATRACE_TAG); 2692#endif 2693 if (framesWritten > 0) { 2694 bytesWritten = framesWritten << mBitShift; 2695 } else { 2696 bytesWritten = framesWritten; 2697 } 2698 // otherwise use the HAL / AudioStreamOut directly 2699 } else { 2700 // Direct output thread. 2701 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2702 } 2703 2704 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2705 mNumWrites++; 2706 mInWrite = false; 2707} 2708 2709void AudioFlinger::MixerThread::threadLoop_standby() 2710{ 2711 // Idle the fast mixer if it's currently running 2712 if (mFastMixer != NULL) { 2713 FastMixerStateQueue *sq = mFastMixer->sq(); 2714 FastMixerState *state = sq->begin(); 2715 if (!(state->mCommand & FastMixerState::IDLE)) { 2716 state->mCommand = FastMixerState::COLD_IDLE; 2717 state->mColdFutexAddr = &mFastMixerFutex; 2718 state->mColdGen++; 2719 mFastMixerFutex = 0; 2720 sq->end(); 2721 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2722 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2723 if (kUseFastMixer == FastMixer_Dynamic) { 2724 mNormalSink = mOutputSink; 2725 } 2726 } else { 2727 sq->end(false /*didModify*/); 2728 } 2729 } 2730 PlaybackThread::threadLoop_standby(); 2731} 2732 2733// shared by MIXER and DIRECT, overridden by DUPLICATING 2734void AudioFlinger::PlaybackThread::threadLoop_standby() 2735{ 2736 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2737 mOutput->stream->common.standby(&mOutput->stream->common); 2738} 2739 2740void AudioFlinger::MixerThread::threadLoop_mix() 2741{ 2742 // obtain the presentation timestamp of the next output buffer 2743 int64_t pts; 2744 status_t status = INVALID_OPERATION; 2745 2746 if (NULL != mOutput->stream->get_next_write_timestamp) { 2747 status = mOutput->stream->get_next_write_timestamp( 2748 mOutput->stream, &pts); 2749 } 2750 2751 if (status != NO_ERROR) { 2752 pts = AudioBufferProvider::kInvalidPTS; 2753 } 2754 2755 // mix buffers... 2756 mAudioMixer->process(pts); 2757 // increase sleep time progressively when application underrun condition clears. 2758 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2759 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2760 // such that we would underrun the audio HAL. 2761 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2762 sleepTimeShift--; 2763 } 2764 sleepTime = 0; 2765 standbyTime = systemTime() + standbyDelay; 2766 //TODO: delay standby when effects have a tail 2767} 2768 2769void AudioFlinger::MixerThread::threadLoop_sleepTime() 2770{ 2771 // If no tracks are ready, sleep once for the duration of an output 2772 // buffer size, then write 0s to the output 2773 if (sleepTime == 0) { 2774 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2775 sleepTime = activeSleepTime >> sleepTimeShift; 2776 if (sleepTime < kMinThreadSleepTimeUs) { 2777 sleepTime = kMinThreadSleepTimeUs; 2778 } 2779 // reduce sleep time in case of consecutive application underruns to avoid 2780 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2781 // duration we would end up writing less data than needed by the audio HAL if 2782 // the condition persists. 2783 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2784 sleepTimeShift++; 2785 } 2786 } else { 2787 sleepTime = idleSleepTime; 2788 } 2789 } else if (mBytesWritten != 0 || 2790 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2791 memset (mMixBuffer, 0, mixBufferSize); 2792 sleepTime = 0; 2793 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2794 } 2795 // TODO add standby time extension fct of effect tail 2796} 2797 2798// prepareTracks_l() must be called with ThreadBase::mLock held 2799AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2800 Vector< sp<Track> > *tracksToRemove) 2801{ 2802 2803 mixer_state mixerStatus = MIXER_IDLE; 2804 // find out which tracks need to be processed 2805 size_t count = mActiveTracks.size(); 2806 size_t mixedTracks = 0; 2807 size_t tracksWithEffect = 0; 2808 // counts only _active_ fast tracks 2809 size_t fastTracks = 0; 2810 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2811 2812 float masterVolume = mMasterVolume; 2813 bool masterMute = mMasterMute; 2814 2815 if (masterMute) { 2816 masterVolume = 0; 2817 } 2818 // Delegate master volume control to effect in output mix effect chain if needed 2819 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2820 if (chain != 0) { 2821 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2822 chain->setVolume_l(&v, &v); 2823 masterVolume = (float)((v + (1 << 23)) >> 24); 2824 chain.clear(); 2825 } 2826 2827 // prepare a new state to push 2828 FastMixerStateQueue *sq = NULL; 2829 FastMixerState *state = NULL; 2830 bool didModify = false; 2831 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2832 if (mFastMixer != NULL) { 2833 sq = mFastMixer->sq(); 2834 state = sq->begin(); 2835 } 2836 2837 for (size_t i=0 ; i<count ; i++) { 2838 sp<Track> t = mActiveTracks[i].promote(); 2839 if (t == 0) continue; 2840 2841 // this const just means the local variable doesn't change 2842 Track* const track = t.get(); 2843 2844 // process fast tracks 2845 if (track->isFastTrack()) { 2846 2847 // It's theoretically possible (though unlikely) for a fast track to be created 2848 // and then removed within the same normal mix cycle. This is not a problem, as 2849 // the track never becomes active so it's fast mixer slot is never touched. 2850 // The converse, of removing an (active) track and then creating a new track 2851 // at the identical fast mixer slot within the same normal mix cycle, 2852 // is impossible because the slot isn't marked available until the end of each cycle. 2853 int j = track->mFastIndex; 2854 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2855 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2856 FastTrack *fastTrack = &state->mFastTracks[j]; 2857 2858 // Determine whether the track is currently in underrun condition, 2859 // and whether it had a recent underrun. 2860 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2861 FastTrackUnderruns underruns = ftDump->mUnderruns; 2862 uint32_t recentFull = (underruns.mBitFields.mFull - 2863 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2864 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2865 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2866 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2867 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2868 uint32_t recentUnderruns = recentPartial + recentEmpty; 2869 track->mObservedUnderruns = underruns; 2870 // don't count underruns that occur while stopping or pausing 2871 // or stopped which can occur when flush() is called while active 2872 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2873 track->mUnderrunCount += recentUnderruns; 2874 } 2875 2876 // This is similar to the state machine for normal tracks, 2877 // with a few modifications for fast tracks. 2878 bool isActive = true; 2879 switch (track->mState) { 2880 case TrackBase::STOPPING_1: 2881 // track stays active in STOPPING_1 state until first underrun 2882 if (recentUnderruns > 0) { 2883 track->mState = TrackBase::STOPPING_2; 2884 } 2885 break; 2886 case TrackBase::PAUSING: 2887 // ramp down is not yet implemented 2888 track->setPaused(); 2889 break; 2890 case TrackBase::RESUMING: 2891 // ramp up is not yet implemented 2892 track->mState = TrackBase::ACTIVE; 2893 break; 2894 case TrackBase::ACTIVE: 2895 if (recentFull > 0 || recentPartial > 0) { 2896 // track has provided at least some frames recently: reset retry count 2897 track->mRetryCount = kMaxTrackRetries; 2898 } 2899 if (recentUnderruns == 0) { 2900 // no recent underruns: stay active 2901 break; 2902 } 2903 // there has recently been an underrun of some kind 2904 if (track->sharedBuffer() == 0) { 2905 // were any of the recent underruns "empty" (no frames available)? 2906 if (recentEmpty == 0) { 2907 // no, then ignore the partial underruns as they are allowed indefinitely 2908 break; 2909 } 2910 // there has recently been an "empty" underrun: decrement the retry counter 2911 if (--(track->mRetryCount) > 0) { 2912 break; 2913 } 2914 // indicate to client process that the track was disabled because of underrun; 2915 // it will then automatically call start() when data is available 2916 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2917 // remove from active list, but state remains ACTIVE [confusing but true] 2918 isActive = false; 2919 break; 2920 } 2921 // fall through 2922 case TrackBase::STOPPING_2: 2923 case TrackBase::PAUSED: 2924 case TrackBase::TERMINATED: 2925 case TrackBase::STOPPED: 2926 case TrackBase::FLUSHED: // flush() while active 2927 // Check for presentation complete if track is inactive 2928 // We have consumed all the buffers of this track. 2929 // This would be incomplete if we auto-paused on underrun 2930 { 2931 size_t audioHALFrames = 2932 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2933 size_t framesWritten = 2934 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2935 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2936 // track stays in active list until presentation is complete 2937 break; 2938 } 2939 } 2940 if (track->isStopping_2()) { 2941 track->mState = TrackBase::STOPPED; 2942 } 2943 if (track->isStopped()) { 2944 // Can't reset directly, as fast mixer is still polling this track 2945 // track->reset(); 2946 // So instead mark this track as needing to be reset after push with ack 2947 resetMask |= 1 << i; 2948 } 2949 isActive = false; 2950 break; 2951 case TrackBase::IDLE: 2952 default: 2953 LOG_FATAL("unexpected track state %d", track->mState); 2954 } 2955 2956 if (isActive) { 2957 // was it previously inactive? 2958 if (!(state->mTrackMask & (1 << j))) { 2959 ExtendedAudioBufferProvider *eabp = track; 2960 VolumeProvider *vp = track; 2961 fastTrack->mBufferProvider = eabp; 2962 fastTrack->mVolumeProvider = vp; 2963 fastTrack->mSampleRate = track->mSampleRate; 2964 fastTrack->mChannelMask = track->mChannelMask; 2965 fastTrack->mGeneration++; 2966 state->mTrackMask |= 1 << j; 2967 didModify = true; 2968 // no acknowledgement required for newly active tracks 2969 } 2970 // cache the combined master volume and stream type volume for fast mixer; this 2971 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2972 track->mCachedVolume = track->isMuted() ? 2973 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2974 ++fastTracks; 2975 } else { 2976 // was it previously active? 2977 if (state->mTrackMask & (1 << j)) { 2978 fastTrack->mBufferProvider = NULL; 2979 fastTrack->mGeneration++; 2980 state->mTrackMask &= ~(1 << j); 2981 didModify = true; 2982 // If any fast tracks were removed, we must wait for acknowledgement 2983 // because we're about to decrement the last sp<> on those tracks. 2984 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2985 } else { 2986 LOG_FATAL("fast track %d should have been active", j); 2987 } 2988 tracksToRemove->add(track); 2989 // Avoids a misleading display in dumpsys 2990 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2991 } 2992 continue; 2993 } 2994 2995 { // local variable scope to avoid goto warning 2996 2997 audio_track_cblk_t* cblk = track->cblk(); 2998 2999 // The first time a track is added we wait 3000 // for all its buffers to be filled before processing it 3001 int name = track->name(); 3002 // make sure that we have enough frames to mix one full buffer. 3003 // enforce this condition only once to enable draining the buffer in case the client 3004 // app does not call stop() and relies on underrun to stop: 3005 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3006 // during last round 3007 uint32_t minFrames = 1; 3008 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3009 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3010 if (t->sampleRate() == (int)mSampleRate) { 3011 minFrames = mNormalFrameCount; 3012 } else { 3013 // +1 for rounding and +1 for additional sample needed for interpolation 3014 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3015 // add frames already consumed but not yet released by the resampler 3016 // because cblk->framesReady() will include these frames 3017 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3018 // the minimum track buffer size is normally twice the number of frames necessary 3019 // to fill one buffer and the resampler should not leave more than one buffer worth 3020 // of unreleased frames after each pass, but just in case... 3021 ALOG_ASSERT(minFrames <= cblk->frameCount); 3022 } 3023 } 3024 if ((track->framesReady() >= minFrames) && track->isReady() && 3025 !track->isPaused() && !track->isTerminated()) 3026 { 3027 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3028 3029 mixedTracks++; 3030 3031 // track->mainBuffer() != mMixBuffer means there is an effect chain 3032 // connected to the track 3033 chain.clear(); 3034 if (track->mainBuffer() != mMixBuffer) { 3035 chain = getEffectChain_l(track->sessionId()); 3036 // Delegate volume control to effect in track effect chain if needed 3037 if (chain != 0) { 3038 tracksWithEffect++; 3039 } else { 3040 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3041 name, track->sessionId()); 3042 } 3043 } 3044 3045 3046 int param = AudioMixer::VOLUME; 3047 if (track->mFillingUpStatus == Track::FS_FILLED) { 3048 // no ramp for the first volume setting 3049 track->mFillingUpStatus = Track::FS_ACTIVE; 3050 if (track->mState == TrackBase::RESUMING) { 3051 track->mState = TrackBase::ACTIVE; 3052 param = AudioMixer::RAMP_VOLUME; 3053 } 3054 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3055 } else if (cblk->server != 0) { 3056 // If the track is stopped before the first frame was mixed, 3057 // do not apply ramp 3058 param = AudioMixer::RAMP_VOLUME; 3059 } 3060 3061 // compute volume for this track 3062 uint32_t vl, vr, va; 3063 if (track->isMuted() || track->isPausing() || 3064 mStreamTypes[track->streamType()].mute) { 3065 vl = vr = va = 0; 3066 if (track->isPausing()) { 3067 track->setPaused(); 3068 } 3069 } else { 3070 3071 // read original volumes with volume control 3072 float typeVolume = mStreamTypes[track->streamType()].volume; 3073 float v = masterVolume * typeVolume; 3074 uint32_t vlr = cblk->getVolumeLR(); 3075 vl = vlr & 0xFFFF; 3076 vr = vlr >> 16; 3077 // track volumes come from shared memory, so can't be trusted and must be clamped 3078 if (vl > MAX_GAIN_INT) { 3079 ALOGV("Track left volume out of range: %04X", vl); 3080 vl = MAX_GAIN_INT; 3081 } 3082 if (vr > MAX_GAIN_INT) { 3083 ALOGV("Track right volume out of range: %04X", vr); 3084 vr = MAX_GAIN_INT; 3085 } 3086 // now apply the master volume and stream type volume 3087 vl = (uint32_t)(v * vl) << 12; 3088 vr = (uint32_t)(v * vr) << 12; 3089 // assuming master volume and stream type volume each go up to 1.0, 3090 // vl and vr are now in 8.24 format 3091 3092 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3093 // send level comes from shared memory and so may be corrupt 3094 if (sendLevel > MAX_GAIN_INT) { 3095 ALOGV("Track send level out of range: %04X", sendLevel); 3096 sendLevel = MAX_GAIN_INT; 3097 } 3098 va = (uint32_t)(v * sendLevel); 3099 } 3100 // Delegate volume control to effect in track effect chain if needed 3101 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3102 // Do not ramp volume if volume is controlled by effect 3103 param = AudioMixer::VOLUME; 3104 track->mHasVolumeController = true; 3105 } else { 3106 // force no volume ramp when volume controller was just disabled or removed 3107 // from effect chain to avoid volume spike 3108 if (track->mHasVolumeController) { 3109 param = AudioMixer::VOLUME; 3110 } 3111 track->mHasVolumeController = false; 3112 } 3113 3114 // Convert volumes from 8.24 to 4.12 format 3115 // This additional clamping is needed in case chain->setVolume_l() overshot 3116 vl = (vl + (1 << 11)) >> 12; 3117 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3118 vr = (vr + (1 << 11)) >> 12; 3119 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3120 3121 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3122 3123 // XXX: these things DON'T need to be done each time 3124 mAudioMixer->setBufferProvider(name, track); 3125 mAudioMixer->enable(name); 3126 3127 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3128 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3129 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3130 mAudioMixer->setParameter( 3131 name, 3132 AudioMixer::TRACK, 3133 AudioMixer::FORMAT, (void *)track->format()); 3134 mAudioMixer->setParameter( 3135 name, 3136 AudioMixer::TRACK, 3137 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3138 mAudioMixer->setParameter( 3139 name, 3140 AudioMixer::RESAMPLE, 3141 AudioMixer::SAMPLE_RATE, 3142 (void *)(cblk->sampleRate)); 3143 mAudioMixer->setParameter( 3144 name, 3145 AudioMixer::TRACK, 3146 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3147 mAudioMixer->setParameter( 3148 name, 3149 AudioMixer::TRACK, 3150 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3151 3152 // reset retry count 3153 track->mRetryCount = kMaxTrackRetries; 3154 3155 // If one track is ready, set the mixer ready if: 3156 // - the mixer was not ready during previous round OR 3157 // - no other track is not ready 3158 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3159 mixerStatus != MIXER_TRACKS_ENABLED) { 3160 mixerStatus = MIXER_TRACKS_READY; 3161 } 3162 } else { 3163 // clear effect chain input buffer if an active track underruns to avoid sending 3164 // previous audio buffer again to effects 3165 chain = getEffectChain_l(track->sessionId()); 3166 if (chain != 0) { 3167 chain->clearInputBuffer(); 3168 } 3169 3170 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3171 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3172 track->isStopped() || track->isPaused()) { 3173 // We have consumed all the buffers of this track. 3174 // Remove it from the list of active tracks. 3175 // TODO: use actual buffer filling status instead of latency when available from 3176 // audio HAL 3177 size_t audioHALFrames = 3178 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3179 size_t framesWritten = 3180 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3181 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3182 if (track->isStopped()) { 3183 track->reset(); 3184 } 3185 tracksToRemove->add(track); 3186 } 3187 } else { 3188 track->mUnderrunCount++; 3189 // No buffers for this track. Give it a few chances to 3190 // fill a buffer, then remove it from active list. 3191 if (--(track->mRetryCount) <= 0) { 3192 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3193 tracksToRemove->add(track); 3194 // indicate to client process that the track was disabled because of underrun; 3195 // it will then automatically call start() when data is available 3196 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3197 // If one track is not ready, mark the mixer also not ready if: 3198 // - the mixer was ready during previous round OR 3199 // - no other track is ready 3200 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3201 mixerStatus != MIXER_TRACKS_READY) { 3202 mixerStatus = MIXER_TRACKS_ENABLED; 3203 } 3204 } 3205 mAudioMixer->disable(name); 3206 } 3207 3208 } // local variable scope to avoid goto warning 3209track_is_ready: ; 3210 3211 } 3212 3213 // Push the new FastMixer state if necessary 3214 if (didModify) { 3215 state->mFastTracksGen++; 3216 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3217 if (kUseFastMixer == FastMixer_Dynamic && 3218 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3219 state->mCommand = FastMixerState::COLD_IDLE; 3220 state->mColdFutexAddr = &mFastMixerFutex; 3221 state->mColdGen++; 3222 mFastMixerFutex = 0; 3223 if (kUseFastMixer == FastMixer_Dynamic) { 3224 mNormalSink = mOutputSink; 3225 } 3226 // If we go into cold idle, need to wait for acknowledgement 3227 // so that fast mixer stops doing I/O. 3228 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3229 } 3230 sq->end(); 3231 } 3232 if (sq != NULL) { 3233 sq->end(didModify); 3234 sq->push(block); 3235 } 3236 3237 // Now perform the deferred reset on fast tracks that have stopped 3238 while (resetMask != 0) { 3239 size_t i = __builtin_ctz(resetMask); 3240 ALOG_ASSERT(i < count); 3241 resetMask &= ~(1 << i); 3242 sp<Track> t = mActiveTracks[i].promote(); 3243 if (t == 0) continue; 3244 Track* track = t.get(); 3245 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3246 track->reset(); 3247 } 3248 3249 // remove all the tracks that need to be... 3250 count = tracksToRemove->size(); 3251 if (CC_UNLIKELY(count)) { 3252 for (size_t i=0 ; i<count ; i++) { 3253 const sp<Track>& track = tracksToRemove->itemAt(i); 3254 mActiveTracks.remove(track); 3255 if (track->mainBuffer() != mMixBuffer) { 3256 chain = getEffectChain_l(track->sessionId()); 3257 if (chain != 0) { 3258 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3259 chain->decActiveTrackCnt(); 3260 } 3261 } 3262 if (track->isTerminated()) { 3263 removeTrack_l(track); 3264 } 3265 } 3266 } 3267 3268 // mix buffer must be cleared if all tracks are connected to an 3269 // effect chain as in this case the mixer will not write to 3270 // mix buffer and track effects will accumulate into it 3271 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3272 // FIXME as a performance optimization, should remember previous zero status 3273 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3274 } 3275 3276 // if any fast tracks, then status is ready 3277 mMixerStatusIgnoringFastTracks = mixerStatus; 3278 if (fastTracks > 0) { 3279 mixerStatus = MIXER_TRACKS_READY; 3280 } 3281 return mixerStatus; 3282} 3283 3284/* 3285The derived values that are cached: 3286 - mixBufferSize from frame count * frame size 3287 - activeSleepTime from activeSleepTimeUs() 3288 - idleSleepTime from idleSleepTimeUs() 3289 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3290 - maxPeriod from frame count and sample rate (MIXER only) 3291 3292The parameters that affect these derived values are: 3293 - frame count 3294 - frame size 3295 - sample rate 3296 - device type: A2DP or not 3297 - device latency 3298 - format: PCM or not 3299 - active sleep time 3300 - idle sleep time 3301*/ 3302 3303void AudioFlinger::PlaybackThread::cacheParameters_l() 3304{ 3305 mixBufferSize = mNormalFrameCount * mFrameSize; 3306 activeSleepTime = activeSleepTimeUs(); 3307 idleSleepTime = idleSleepTimeUs(); 3308} 3309 3310void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3311{ 3312 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3313 this, streamType, mTracks.size()); 3314 Mutex::Autolock _l(mLock); 3315 3316 size_t size = mTracks.size(); 3317 for (size_t i = 0; i < size; i++) { 3318 sp<Track> t = mTracks[i]; 3319 if (t->streamType() == streamType) { 3320 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3321 t->mCblk->cv.signal(); 3322 } 3323 } 3324} 3325 3326// getTrackName_l() must be called with ThreadBase::mLock held 3327int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3328{ 3329 return mAudioMixer->getTrackName(channelMask); 3330} 3331 3332// deleteTrackName_l() must be called with ThreadBase::mLock held 3333void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3334{ 3335 ALOGV("remove track (%d) and delete from mixer", name); 3336 mAudioMixer->deleteTrackName(name); 3337} 3338 3339// checkForNewParameters_l() must be called with ThreadBase::mLock held 3340bool AudioFlinger::MixerThread::checkForNewParameters_l() 3341{ 3342 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3343 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3344 bool reconfig = false; 3345 3346 while (!mNewParameters.isEmpty()) { 3347 3348 if (mFastMixer != NULL) { 3349 FastMixerStateQueue *sq = mFastMixer->sq(); 3350 FastMixerState *state = sq->begin(); 3351 if (!(state->mCommand & FastMixerState::IDLE)) { 3352 previousCommand = state->mCommand; 3353 state->mCommand = FastMixerState::HOT_IDLE; 3354 sq->end(); 3355 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3356 } else { 3357 sq->end(false /*didModify*/); 3358 } 3359 } 3360 3361 status_t status = NO_ERROR; 3362 String8 keyValuePair = mNewParameters[0]; 3363 AudioParameter param = AudioParameter(keyValuePair); 3364 int value; 3365 3366 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3367 reconfig = true; 3368 } 3369 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3370 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3371 status = BAD_VALUE; 3372 } else { 3373 reconfig = true; 3374 } 3375 } 3376 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3377 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3378 status = BAD_VALUE; 3379 } else { 3380 reconfig = true; 3381 } 3382 } 3383 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3384 // do not accept frame count changes if tracks are open as the track buffer 3385 // size depends on frame count and correct behavior would not be guaranteed 3386 // if frame count is changed after track creation 3387 if (!mTracks.isEmpty()) { 3388 status = INVALID_OPERATION; 3389 } else { 3390 reconfig = true; 3391 } 3392 } 3393 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3394#ifdef ADD_BATTERY_DATA 3395 // when changing the audio output device, call addBatteryData to notify 3396 // the change 3397 if ((int)mDevice != value) { 3398 uint32_t params = 0; 3399 // check whether speaker is on 3400 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3401 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3402 } 3403 3404 int deviceWithoutSpeaker 3405 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3406 // check if any other device (except speaker) is on 3407 if (value & deviceWithoutSpeaker ) { 3408 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3409 } 3410 3411 if (params != 0) { 3412 addBatteryData(params); 3413 } 3414 } 3415#endif 3416 3417 // forward device change to effects that have requested to be 3418 // aware of attached audio device. 3419 mDevice = (uint32_t)value; 3420 for (size_t i = 0; i < mEffectChains.size(); i++) { 3421 mEffectChains[i]->setDevice_l(mDevice); 3422 } 3423 } 3424 3425 if (status == NO_ERROR) { 3426 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3427 keyValuePair.string()); 3428 if (!mStandby && status == INVALID_OPERATION) { 3429 mOutput->stream->common.standby(&mOutput->stream->common); 3430 mStandby = true; 3431 mBytesWritten = 0; 3432 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3433 keyValuePair.string()); 3434 } 3435 if (status == NO_ERROR && reconfig) { 3436 delete mAudioMixer; 3437 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3438 mAudioMixer = NULL; 3439 readOutputParameters(); 3440 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3441 for (size_t i = 0; i < mTracks.size() ; i++) { 3442 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3443 if (name < 0) break; 3444 mTracks[i]->mName = name; 3445 // limit track sample rate to 2 x new output sample rate 3446 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3447 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3448 } 3449 } 3450 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3451 } 3452 } 3453 3454 mNewParameters.removeAt(0); 3455 3456 mParamStatus = status; 3457 mParamCond.signal(); 3458 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3459 // already timed out waiting for the status and will never signal the condition. 3460 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3461 } 3462 3463 if (!(previousCommand & FastMixerState::IDLE)) { 3464 ALOG_ASSERT(mFastMixer != NULL); 3465 FastMixerStateQueue *sq = mFastMixer->sq(); 3466 FastMixerState *state = sq->begin(); 3467 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3468 state->mCommand = previousCommand; 3469 sq->end(); 3470 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3471 } 3472 3473 return reconfig; 3474} 3475 3476status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3477{ 3478 const size_t SIZE = 256; 3479 char buffer[SIZE]; 3480 String8 result; 3481 3482 PlaybackThread::dumpInternals(fd, args); 3483 3484 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3485 result.append(buffer); 3486 write(fd, result.string(), result.size()); 3487 3488 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3489 FastMixerDumpState copy = mFastMixerDumpState; 3490 copy.dump(fd); 3491 3492#ifdef STATE_QUEUE_DUMP 3493 // Similar for state queue 3494 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3495 observerCopy.dump(fd); 3496 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3497 mutatorCopy.dump(fd); 3498#endif 3499 3500 // Write the tee output to a .wav file 3501 NBAIO_Source *teeSource = mTeeSource.get(); 3502 if (teeSource != NULL) { 3503 char teePath[64]; 3504 struct timeval tv; 3505 gettimeofday(&tv, NULL); 3506 struct tm tm; 3507 localtime_r(&tv.tv_sec, &tm); 3508 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3509 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3510 if (teeFd >= 0) { 3511 char wavHeader[44]; 3512 memcpy(wavHeader, 3513 "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", 3514 sizeof(wavHeader)); 3515 NBAIO_Format format = teeSource->format(); 3516 unsigned channelCount = Format_channelCount(format); 3517 ALOG_ASSERT(channelCount <= FCC_2); 3518 unsigned sampleRate = Format_sampleRate(format); 3519 wavHeader[22] = channelCount; // number of channels 3520 wavHeader[24] = sampleRate; // sample rate 3521 wavHeader[25] = sampleRate >> 8; 3522 wavHeader[32] = channelCount * 2; // block alignment 3523 write(teeFd, wavHeader, sizeof(wavHeader)); 3524 size_t total = 0; 3525 bool firstRead = true; 3526 for (;;) { 3527#define TEE_SINK_READ 1024 3528 short buffer[TEE_SINK_READ * FCC_2]; 3529 size_t count = TEE_SINK_READ; 3530 ssize_t actual = teeSource->read(buffer, count); 3531 bool wasFirstRead = firstRead; 3532 firstRead = false; 3533 if (actual <= 0) { 3534 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3535 continue; 3536 } 3537 break; 3538 } 3539 ALOG_ASSERT(actual <= count); 3540 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3541 total += actual; 3542 } 3543 lseek(teeFd, (off_t) 4, SEEK_SET); 3544 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3545 write(teeFd, &temp, sizeof(temp)); 3546 lseek(teeFd, (off_t) 40, SEEK_SET); 3547 temp = total * channelCount * sizeof(short); 3548 write(teeFd, &temp, sizeof(temp)); 3549 close(teeFd); 3550 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3551 } else { 3552 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3553 } 3554 } 3555 3556 return NO_ERROR; 3557} 3558 3559uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3560{ 3561 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3562} 3563 3564uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3565{ 3566 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3567} 3568 3569void AudioFlinger::MixerThread::cacheParameters_l() 3570{ 3571 PlaybackThread::cacheParameters_l(); 3572 3573 // FIXME: Relaxed timing because of a certain device that can't meet latency 3574 // Should be reduced to 2x after the vendor fixes the driver issue 3575 // increase threshold again due to low power audio mode. The way this warning 3576 // threshold is calculated and its usefulness should be reconsidered anyway. 3577 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3578} 3579 3580// ---------------------------------------------------------------------------- 3581AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3582 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3583 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3584 // mLeftVolFloat, mRightVolFloat 3585{ 3586} 3587 3588AudioFlinger::DirectOutputThread::~DirectOutputThread() 3589{ 3590} 3591 3592AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3593 Vector< sp<Track> > *tracksToRemove 3594) 3595{ 3596 sp<Track> trackToRemove; 3597 3598 mixer_state mixerStatus = MIXER_IDLE; 3599 3600 // find out which tracks need to be processed 3601 if (mActiveTracks.size() != 0) { 3602 sp<Track> t = mActiveTracks[0].promote(); 3603 // The track died recently 3604 if (t == 0) return MIXER_IDLE; 3605 3606 Track* const track = t.get(); 3607 audio_track_cblk_t* cblk = track->cblk(); 3608 3609 // The first time a track is added we wait 3610 // for all its buffers to be filled before processing it 3611 uint32_t minFrames; 3612 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3613 minFrames = mNormalFrameCount; 3614 } else { 3615 minFrames = 1; 3616 } 3617 if ((track->framesReady() >= minFrames) && track->isReady() && 3618 !track->isPaused() && !track->isTerminated()) 3619 { 3620 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3621 3622 if (track->mFillingUpStatus == Track::FS_FILLED) { 3623 track->mFillingUpStatus = Track::FS_ACTIVE; 3624 mLeftVolFloat = mRightVolFloat = 0; 3625 if (track->mState == TrackBase::RESUMING) { 3626 track->mState = TrackBase::ACTIVE; 3627 } 3628 } 3629 3630 // compute volume for this track 3631 float left, right; 3632 if (track->isMuted() || mMasterMute || track->isPausing() || 3633 mStreamTypes[track->streamType()].mute) { 3634 left = right = 0; 3635 if (track->isPausing()) { 3636 track->setPaused(); 3637 } 3638 } else { 3639 float typeVolume = mStreamTypes[track->streamType()].volume; 3640 float v = mMasterVolume * typeVolume; 3641 uint32_t vlr = cblk->getVolumeLR(); 3642 float v_clamped = v * (vlr & 0xFFFF); 3643 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3644 left = v_clamped/MAX_GAIN; 3645 v_clamped = v * (vlr >> 16); 3646 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3647 right = v_clamped/MAX_GAIN; 3648 } 3649 3650 if (left != mLeftVolFloat || right != mRightVolFloat) { 3651 mLeftVolFloat = left; 3652 mRightVolFloat = right; 3653 3654 // Convert volumes from float to 8.24 3655 uint32_t vl = (uint32_t)(left * (1 << 24)); 3656 uint32_t vr = (uint32_t)(right * (1 << 24)); 3657 3658 // Delegate volume control to effect in track effect chain if needed 3659 // only one effect chain can be present on DirectOutputThread, so if 3660 // there is one, the track is connected to it 3661 if (!mEffectChains.isEmpty()) { 3662 // Do not ramp volume if volume is controlled by effect 3663 mEffectChains[0]->setVolume_l(&vl, &vr); 3664 left = (float)vl / (1 << 24); 3665 right = (float)vr / (1 << 24); 3666 } 3667 mOutput->stream->set_volume(mOutput->stream, left, right); 3668 } 3669 3670 // reset retry count 3671 track->mRetryCount = kMaxTrackRetriesDirect; 3672 mActiveTrack = t; 3673 mixerStatus = MIXER_TRACKS_READY; 3674 } else { 3675 // clear effect chain input buffer if an active track underruns to avoid sending 3676 // previous audio buffer again to effects 3677 if (!mEffectChains.isEmpty()) { 3678 mEffectChains[0]->clearInputBuffer(); 3679 } 3680 3681 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3682 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3683 track->isStopped() || track->isPaused()) { 3684 // We have consumed all the buffers of this track. 3685 // Remove it from the list of active tracks. 3686 // TODO: implement behavior for compressed audio 3687 size_t audioHALFrames = 3688 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3689 size_t framesWritten = 3690 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3691 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3692 if (track->isStopped()) { 3693 track->reset(); 3694 } 3695 trackToRemove = track; 3696 } 3697 } else { 3698 // No buffers for this track. Give it a few chances to 3699 // fill a buffer, then remove it from active list. 3700 if (--(track->mRetryCount) <= 0) { 3701 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3702 trackToRemove = track; 3703 } else { 3704 mixerStatus = MIXER_TRACKS_ENABLED; 3705 } 3706 } 3707 } 3708 } 3709 3710 // FIXME merge this with similar code for removing multiple tracks 3711 // remove all the tracks that need to be... 3712 if (CC_UNLIKELY(trackToRemove != 0)) { 3713 tracksToRemove->add(trackToRemove); 3714 mActiveTracks.remove(trackToRemove); 3715 if (!mEffectChains.isEmpty()) { 3716 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3717 trackToRemove->sessionId()); 3718 mEffectChains[0]->decActiveTrackCnt(); 3719 } 3720 if (trackToRemove->isTerminated()) { 3721 removeTrack_l(trackToRemove); 3722 } 3723 } 3724 3725 return mixerStatus; 3726} 3727 3728void AudioFlinger::DirectOutputThread::threadLoop_mix() 3729{ 3730 AudioBufferProvider::Buffer buffer; 3731 size_t frameCount = mFrameCount; 3732 int8_t *curBuf = (int8_t *)mMixBuffer; 3733 // output audio to hardware 3734 while (frameCount) { 3735 buffer.frameCount = frameCount; 3736 mActiveTrack->getNextBuffer(&buffer); 3737 if (CC_UNLIKELY(buffer.raw == NULL)) { 3738 memset(curBuf, 0, frameCount * mFrameSize); 3739 break; 3740 } 3741 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3742 frameCount -= buffer.frameCount; 3743 curBuf += buffer.frameCount * mFrameSize; 3744 mActiveTrack->releaseBuffer(&buffer); 3745 } 3746 sleepTime = 0; 3747 standbyTime = systemTime() + standbyDelay; 3748 mActiveTrack.clear(); 3749 3750} 3751 3752void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3753{ 3754 if (sleepTime == 0) { 3755 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3756 sleepTime = activeSleepTime; 3757 } else { 3758 sleepTime = idleSleepTime; 3759 } 3760 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3761 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3762 sleepTime = 0; 3763 } 3764} 3765 3766// getTrackName_l() must be called with ThreadBase::mLock held 3767int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3768{ 3769 return 0; 3770} 3771 3772// deleteTrackName_l() must be called with ThreadBase::mLock held 3773void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3774{ 3775} 3776 3777// checkForNewParameters_l() must be called with ThreadBase::mLock held 3778bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3779{ 3780 bool reconfig = false; 3781 3782 while (!mNewParameters.isEmpty()) { 3783 status_t status = NO_ERROR; 3784 String8 keyValuePair = mNewParameters[0]; 3785 AudioParameter param = AudioParameter(keyValuePair); 3786 int value; 3787 3788 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3789 // do not accept frame count changes if tracks are open as the track buffer 3790 // size depends on frame count and correct behavior would not be garantied 3791 // if frame count is changed after track creation 3792 if (!mTracks.isEmpty()) { 3793 status = INVALID_OPERATION; 3794 } else { 3795 reconfig = true; 3796 } 3797 } 3798 if (status == NO_ERROR) { 3799 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3800 keyValuePair.string()); 3801 if (!mStandby && status == INVALID_OPERATION) { 3802 mOutput->stream->common.standby(&mOutput->stream->common); 3803 mStandby = true; 3804 mBytesWritten = 0; 3805 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3806 keyValuePair.string()); 3807 } 3808 if (status == NO_ERROR && reconfig) { 3809 readOutputParameters(); 3810 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3811 } 3812 } 3813 3814 mNewParameters.removeAt(0); 3815 3816 mParamStatus = status; 3817 mParamCond.signal(); 3818 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3819 // already timed out waiting for the status and will never signal the condition. 3820 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3821 } 3822 return reconfig; 3823} 3824 3825uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3826{ 3827 uint32_t time; 3828 if (audio_is_linear_pcm(mFormat)) { 3829 time = PlaybackThread::activeSleepTimeUs(); 3830 } else { 3831 time = 10000; 3832 } 3833 return time; 3834} 3835 3836uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3837{ 3838 uint32_t time; 3839 if (audio_is_linear_pcm(mFormat)) { 3840 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3841 } else { 3842 time = 10000; 3843 } 3844 return time; 3845} 3846 3847uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3848{ 3849 uint32_t time; 3850 if (audio_is_linear_pcm(mFormat)) { 3851 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3852 } else { 3853 time = 10000; 3854 } 3855 return time; 3856} 3857 3858void AudioFlinger::DirectOutputThread::cacheParameters_l() 3859{ 3860 PlaybackThread::cacheParameters_l(); 3861 3862 // use shorter standby delay as on normal output to release 3863 // hardware resources as soon as possible 3864 standbyDelay = microseconds(activeSleepTime*2); 3865} 3866 3867// ---------------------------------------------------------------------------- 3868 3869AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3870 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3871 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3872 mWaitTimeMs(UINT_MAX) 3873{ 3874 addOutputTrack(mainThread); 3875} 3876 3877AudioFlinger::DuplicatingThread::~DuplicatingThread() 3878{ 3879 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3880 mOutputTracks[i]->destroy(); 3881 } 3882} 3883 3884void AudioFlinger::DuplicatingThread::threadLoop_mix() 3885{ 3886 // mix buffers... 3887 if (outputsReady(outputTracks)) { 3888 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3889 } else { 3890 memset(mMixBuffer, 0, mixBufferSize); 3891 } 3892 sleepTime = 0; 3893 writeFrames = mNormalFrameCount; 3894 standbyTime = systemTime() + standbyDelay; 3895} 3896 3897void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3898{ 3899 if (sleepTime == 0) { 3900 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3901 sleepTime = activeSleepTime; 3902 } else { 3903 sleepTime = idleSleepTime; 3904 } 3905 } else if (mBytesWritten != 0) { 3906 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3907 writeFrames = mNormalFrameCount; 3908 memset(mMixBuffer, 0, mixBufferSize); 3909 } else { 3910 // flush remaining overflow buffers in output tracks 3911 writeFrames = 0; 3912 } 3913 sleepTime = 0; 3914 } 3915} 3916 3917void AudioFlinger::DuplicatingThread::threadLoop_write() 3918{ 3919 for (size_t i = 0; i < outputTracks.size(); i++) { 3920 outputTracks[i]->write(mMixBuffer, writeFrames); 3921 } 3922 mBytesWritten += mixBufferSize; 3923} 3924 3925void AudioFlinger::DuplicatingThread::threadLoop_standby() 3926{ 3927 // DuplicatingThread implements standby by stopping all tracks 3928 for (size_t i = 0; i < outputTracks.size(); i++) { 3929 outputTracks[i]->stop(); 3930 } 3931} 3932 3933void AudioFlinger::DuplicatingThread::saveOutputTracks() 3934{ 3935 outputTracks = mOutputTracks; 3936} 3937 3938void AudioFlinger::DuplicatingThread::clearOutputTracks() 3939{ 3940 outputTracks.clear(); 3941} 3942 3943void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3944{ 3945 Mutex::Autolock _l(mLock); 3946 // FIXME explain this formula 3947 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3948 OutputTrack *outputTrack = new OutputTrack(thread, 3949 this, 3950 mSampleRate, 3951 mFormat, 3952 mChannelMask, 3953 frameCount); 3954 if (outputTrack->cblk() != NULL) { 3955 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3956 mOutputTracks.add(outputTrack); 3957 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3958 updateWaitTime_l(); 3959 } 3960} 3961 3962void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3963{ 3964 Mutex::Autolock _l(mLock); 3965 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3966 if (mOutputTracks[i]->thread() == thread) { 3967 mOutputTracks[i]->destroy(); 3968 mOutputTracks.removeAt(i); 3969 updateWaitTime_l(); 3970 return; 3971 } 3972 } 3973 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3974} 3975 3976// caller must hold mLock 3977void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3978{ 3979 mWaitTimeMs = UINT_MAX; 3980 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3981 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3982 if (strong != 0) { 3983 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3984 if (waitTimeMs < mWaitTimeMs) { 3985 mWaitTimeMs = waitTimeMs; 3986 } 3987 } 3988 } 3989} 3990 3991 3992bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3993{ 3994 for (size_t i = 0; i < outputTracks.size(); i++) { 3995 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3996 if (thread == 0) { 3997 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3998 return false; 3999 } 4000 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4001 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4002 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4003 return false; 4004 } 4005 } 4006 return true; 4007} 4008 4009uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4010{ 4011 return (mWaitTimeMs * 1000) / 2; 4012} 4013 4014void AudioFlinger::DuplicatingThread::cacheParameters_l() 4015{ 4016 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4017 updateWaitTime_l(); 4018 4019 MixerThread::cacheParameters_l(); 4020} 4021 4022// ---------------------------------------------------------------------------- 4023 4024// TrackBase constructor must be called with AudioFlinger::mLock held 4025AudioFlinger::ThreadBase::TrackBase::TrackBase( 4026 ThreadBase *thread, 4027 const sp<Client>& client, 4028 uint32_t sampleRate, 4029 audio_format_t format, 4030 uint32_t channelMask, 4031 int frameCount, 4032 const sp<IMemory>& sharedBuffer, 4033 int sessionId) 4034 : RefBase(), 4035 mThread(thread), 4036 mClient(client), 4037 mCblk(NULL), 4038 // mBuffer 4039 // mBufferEnd 4040 mFrameCount(0), 4041 mState(IDLE), 4042 mSampleRate(sampleRate), 4043 mFormat(format), 4044 mStepServerFailed(false), 4045 mSessionId(sessionId) 4046 // mChannelCount 4047 // mChannelMask 4048{ 4049 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4050 4051 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4052 size_t size = sizeof(audio_track_cblk_t); 4053 uint8_t channelCount = popcount(channelMask); 4054 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4055 if (sharedBuffer == 0) { 4056 size += bufferSize; 4057 } 4058 4059 if (client != NULL) { 4060 mCblkMemory = client->heap()->allocate(size); 4061 if (mCblkMemory != 0) { 4062 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4063 if (mCblk != NULL) { // construct the shared structure in-place. 4064 new(mCblk) audio_track_cblk_t(); 4065 // clear all buffers 4066 mCblk->frameCount = frameCount; 4067 mCblk->sampleRate = sampleRate; 4068// uncomment the following lines to quickly test 32-bit wraparound 4069// mCblk->user = 0xffff0000; 4070// mCblk->server = 0xffff0000; 4071// mCblk->userBase = 0xffff0000; 4072// mCblk->serverBase = 0xffff0000; 4073 mChannelCount = channelCount; 4074 mChannelMask = channelMask; 4075 if (sharedBuffer == 0) { 4076 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4077 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4078 // Force underrun condition to avoid false underrun callback until first data is 4079 // written to buffer (other flags are cleared) 4080 mCblk->flags = CBLK_UNDERRUN_ON; 4081 } else { 4082 mBuffer = sharedBuffer->pointer(); 4083 } 4084 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4085 } 4086 } else { 4087 ALOGE("not enough memory for AudioTrack size=%u", size); 4088 client->heap()->dump("AudioTrack"); 4089 return; 4090 } 4091 } else { 4092 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4093 // construct the shared structure in-place. 4094 new(mCblk) audio_track_cblk_t(); 4095 // clear all buffers 4096 mCblk->frameCount = frameCount; 4097 mCblk->sampleRate = sampleRate; 4098// uncomment the following lines to quickly test 32-bit wraparound 4099// mCblk->user = 0xffff0000; 4100// mCblk->server = 0xffff0000; 4101// mCblk->userBase = 0xffff0000; 4102// mCblk->serverBase = 0xffff0000; 4103 mChannelCount = channelCount; 4104 mChannelMask = channelMask; 4105 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4106 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4107 // Force underrun condition to avoid false underrun callback until first data is 4108 // written to buffer (other flags are cleared) 4109 mCblk->flags = CBLK_UNDERRUN_ON; 4110 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4111 } 4112} 4113 4114AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4115{ 4116 if (mCblk != NULL) { 4117 if (mClient == 0) { 4118 delete mCblk; 4119 } else { 4120 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4121 } 4122 } 4123 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4124 if (mClient != 0) { 4125 // Client destructor must run with AudioFlinger mutex locked 4126 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4127 // If the client's reference count drops to zero, the associated destructor 4128 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4129 // relying on the automatic clear() at end of scope. 4130 mClient.clear(); 4131 } 4132} 4133 4134// AudioBufferProvider interface 4135// getNextBuffer() = 0; 4136// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4137void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4138{ 4139 buffer->raw = NULL; 4140 mFrameCount = buffer->frameCount; 4141 // FIXME See note at getNextBuffer() 4142 (void) step(); // ignore return value of step() 4143 buffer->frameCount = 0; 4144} 4145 4146bool AudioFlinger::ThreadBase::TrackBase::step() { 4147 bool result; 4148 audio_track_cblk_t* cblk = this->cblk(); 4149 4150 result = cblk->stepServer(mFrameCount); 4151 if (!result) { 4152 ALOGV("stepServer failed acquiring cblk mutex"); 4153 mStepServerFailed = true; 4154 } 4155 return result; 4156} 4157 4158void AudioFlinger::ThreadBase::TrackBase::reset() { 4159 audio_track_cblk_t* cblk = this->cblk(); 4160 4161 cblk->user = 0; 4162 cblk->server = 0; 4163 cblk->userBase = 0; 4164 cblk->serverBase = 0; 4165 mStepServerFailed = false; 4166 ALOGV("TrackBase::reset"); 4167} 4168 4169int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4170 return (int)mCblk->sampleRate; 4171} 4172 4173void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4174 audio_track_cblk_t* cblk = this->cblk(); 4175 size_t frameSize = cblk->frameSize; 4176 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4177 int8_t *bufferEnd = bufferStart + frames * frameSize; 4178 4179 // Check validity of returned pointer in case the track control block would have been corrupted. 4180 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4181 "TrackBase::getBuffer buffer out of range:\n" 4182 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4183 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4184 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4185 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4186 4187 return bufferStart; 4188} 4189 4190status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4191{ 4192 mSyncEvents.add(event); 4193 return NO_ERROR; 4194} 4195 4196// ---------------------------------------------------------------------------- 4197 4198// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4199AudioFlinger::PlaybackThread::Track::Track( 4200 PlaybackThread *thread, 4201 const sp<Client>& client, 4202 audio_stream_type_t streamType, 4203 uint32_t sampleRate, 4204 audio_format_t format, 4205 uint32_t channelMask, 4206 int frameCount, 4207 const sp<IMemory>& sharedBuffer, 4208 int sessionId, 4209 IAudioFlinger::track_flags_t flags) 4210 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4211 mMute(false), 4212 mFillingUpStatus(FS_INVALID), 4213 // mRetryCount initialized later when needed 4214 mSharedBuffer(sharedBuffer), 4215 mStreamType(streamType), 4216 mName(-1), // see note below 4217 mMainBuffer(thread->mixBuffer()), 4218 mAuxBuffer(NULL), 4219 mAuxEffectId(0), mHasVolumeController(false), 4220 mPresentationCompleteFrames(0), 4221 mFlags(flags), 4222 mFastIndex(-1), 4223 mUnderrunCount(0), 4224 mCachedVolume(1.0) 4225{ 4226 if (mCblk != NULL) { 4227 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4228 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4229 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4230 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4231 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4232 if (mName < 0) { 4233 ALOGE("no more track names available"); 4234 return; 4235 } 4236 // only allocate a fast track index if we were able to allocate a normal track name 4237 if (flags & IAudioFlinger::TRACK_FAST) { 4238 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4239 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4240 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4241 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4242 // FIXME This is too eager. We allocate a fast track index before the 4243 // fast track becomes active. Since fast tracks are a scarce resource, 4244 // this means we are potentially denying other more important fast tracks from 4245 // being created. It would be better to allocate the index dynamically. 4246 mFastIndex = i; 4247 // Read the initial underruns because this field is never cleared by the fast mixer 4248 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4249 thread->mFastTrackAvailMask &= ~(1 << i); 4250 } 4251 } 4252 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4253} 4254 4255AudioFlinger::PlaybackThread::Track::~Track() 4256{ 4257 ALOGV("PlaybackThread::Track destructor"); 4258 sp<ThreadBase> thread = mThread.promote(); 4259 if (thread != 0) { 4260 Mutex::Autolock _l(thread->mLock); 4261 mState = TERMINATED; 4262 } 4263} 4264 4265void AudioFlinger::PlaybackThread::Track::destroy() 4266{ 4267 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4268 // by removing it from mTracks vector, so there is a risk that this Tracks's 4269 // destructor is called. As the destructor needs to lock mLock, 4270 // we must acquire a strong reference on this Track before locking mLock 4271 // here so that the destructor is called only when exiting this function. 4272 // On the other hand, as long as Track::destroy() is only called by 4273 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4274 // this Track with its member mTrack. 4275 sp<Track> keep(this); 4276 { // scope for mLock 4277 sp<ThreadBase> thread = mThread.promote(); 4278 if (thread != 0) { 4279 if (!isOutputTrack()) { 4280 if (mState == ACTIVE || mState == RESUMING) { 4281 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4282 4283#ifdef ADD_BATTERY_DATA 4284 // to track the speaker usage 4285 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4286#endif 4287 } 4288 AudioSystem::releaseOutput(thread->id()); 4289 } 4290 Mutex::Autolock _l(thread->mLock); 4291 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4292 playbackThread->destroyTrack_l(this); 4293 } 4294 } 4295} 4296 4297/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4298{ 4299 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4300 " Server User Main buf Aux Buf Flags Underruns\n"); 4301} 4302 4303void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4304{ 4305 uint32_t vlr = mCblk->getVolumeLR(); 4306 if (isFastTrack()) { 4307 sprintf(buffer, " F %2d", mFastIndex); 4308 } else { 4309 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4310 } 4311 track_state state = mState; 4312 char stateChar; 4313 switch (state) { 4314 case IDLE: 4315 stateChar = 'I'; 4316 break; 4317 case TERMINATED: 4318 stateChar = 'T'; 4319 break; 4320 case STOPPING_1: 4321 stateChar = 's'; 4322 break; 4323 case STOPPING_2: 4324 stateChar = '5'; 4325 break; 4326 case STOPPED: 4327 stateChar = 'S'; 4328 break; 4329 case RESUMING: 4330 stateChar = 'R'; 4331 break; 4332 case ACTIVE: 4333 stateChar = 'A'; 4334 break; 4335 case PAUSING: 4336 stateChar = 'p'; 4337 break; 4338 case PAUSED: 4339 stateChar = 'P'; 4340 break; 4341 case FLUSHED: 4342 stateChar = 'F'; 4343 break; 4344 default: 4345 stateChar = '?'; 4346 break; 4347 } 4348 char nowInUnderrun; 4349 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4350 case UNDERRUN_FULL: 4351 nowInUnderrun = ' '; 4352 break; 4353 case UNDERRUN_PARTIAL: 4354 nowInUnderrun = '<'; 4355 break; 4356 case UNDERRUN_EMPTY: 4357 nowInUnderrun = '*'; 4358 break; 4359 default: 4360 nowInUnderrun = '?'; 4361 break; 4362 } 4363 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4364 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4365 (mClient == 0) ? getpid_cached : mClient->pid(), 4366 mStreamType, 4367 mFormat, 4368 mChannelMask, 4369 mSessionId, 4370 mFrameCount, 4371 mCblk->frameCount, 4372 stateChar, 4373 mMute, 4374 mFillingUpStatus, 4375 mCblk->sampleRate, 4376 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4377 20.0 * log10((vlr >> 16) / 4096.0), 4378 mCblk->server, 4379 mCblk->user, 4380 (int)mMainBuffer, 4381 (int)mAuxBuffer, 4382 mCblk->flags, 4383 mUnderrunCount, 4384 nowInUnderrun); 4385} 4386 4387// AudioBufferProvider interface 4388status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4389 AudioBufferProvider::Buffer* buffer, int64_t pts) 4390{ 4391 audio_track_cblk_t* cblk = this->cblk(); 4392 uint32_t framesReady; 4393 uint32_t framesReq = buffer->frameCount; 4394 4395 // Check if last stepServer failed, try to step now 4396 if (mStepServerFailed) { 4397 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4398 // Since the fast mixer is higher priority than client callback thread, 4399 // it does not result in priority inversion for client. 4400 // But a non-blocking solution would be preferable to avoid 4401 // fast mixer being unable to tryLock(), and 4402 // to avoid the extra context switches if the client wakes up, 4403 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4404 if (!step()) goto getNextBuffer_exit; 4405 ALOGV("stepServer recovered"); 4406 mStepServerFailed = false; 4407 } 4408 4409 // FIXME Same as above 4410 framesReady = cblk->framesReady(); 4411 4412 if (CC_LIKELY(framesReady)) { 4413 uint32_t s = cblk->server; 4414 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4415 4416 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4417 if (framesReq > framesReady) { 4418 framesReq = framesReady; 4419 } 4420 if (framesReq > bufferEnd - s) { 4421 framesReq = bufferEnd - s; 4422 } 4423 4424 buffer->raw = getBuffer(s, framesReq); 4425 if (buffer->raw == NULL) goto getNextBuffer_exit; 4426 4427 buffer->frameCount = framesReq; 4428 return NO_ERROR; 4429 } 4430 4431getNextBuffer_exit: 4432 buffer->raw = NULL; 4433 buffer->frameCount = 0; 4434 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4435 return NOT_ENOUGH_DATA; 4436} 4437 4438// Note that framesReady() takes a mutex on the control block using tryLock(). 4439// This could result in priority inversion if framesReady() is called by the normal mixer, 4440// as the normal mixer thread runs at lower 4441// priority than the client's callback thread: there is a short window within framesReady() 4442// during which the normal mixer could be preempted, and the client callback would block. 4443// Another problem can occur if framesReady() is called by the fast mixer: 4444// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4445// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4446size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4447 return mCblk->framesReady(); 4448} 4449 4450// Don't call for fast tracks; the framesReady() could result in priority inversion 4451bool AudioFlinger::PlaybackThread::Track::isReady() const { 4452 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4453 4454 if (framesReady() >= mCblk->frameCount || 4455 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4456 mFillingUpStatus = FS_FILLED; 4457 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4458 return true; 4459 } 4460 return false; 4461} 4462 4463status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4464 int triggerSession) 4465{ 4466 status_t status = NO_ERROR; 4467 ALOGV("start(%d), calling pid %d session %d", 4468 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4469 4470 sp<ThreadBase> thread = mThread.promote(); 4471 if (thread != 0) { 4472 Mutex::Autolock _l(thread->mLock); 4473 track_state state = mState; 4474 // here the track could be either new, or restarted 4475 // in both cases "unstop" the track 4476 if (mState == PAUSED) { 4477 mState = TrackBase::RESUMING; 4478 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4479 } else { 4480 mState = TrackBase::ACTIVE; 4481 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4482 } 4483 4484 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4485 thread->mLock.unlock(); 4486 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4487 thread->mLock.lock(); 4488 4489#ifdef ADD_BATTERY_DATA 4490 // to track the speaker usage 4491 if (status == NO_ERROR) { 4492 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4493 } 4494#endif 4495 } 4496 if (status == NO_ERROR) { 4497 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4498 playbackThread->addTrack_l(this); 4499 } else { 4500 mState = state; 4501 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4502 } 4503 } else { 4504 status = BAD_VALUE; 4505 } 4506 return status; 4507} 4508 4509void AudioFlinger::PlaybackThread::Track::stop() 4510{ 4511 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4512 sp<ThreadBase> thread = mThread.promote(); 4513 if (thread != 0) { 4514 Mutex::Autolock _l(thread->mLock); 4515 track_state state = mState; 4516 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4517 // If the track is not active (PAUSED and buffers full), flush buffers 4518 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4519 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4520 reset(); 4521 mState = STOPPED; 4522 } else if (!isFastTrack()) { 4523 mState = STOPPED; 4524 } else { 4525 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4526 // and then to STOPPED and reset() when presentation is complete 4527 mState = STOPPING_1; 4528 } 4529 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4530 } 4531 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4532 thread->mLock.unlock(); 4533 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4534 thread->mLock.lock(); 4535 4536#ifdef ADD_BATTERY_DATA 4537 // to track the speaker usage 4538 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4539#endif 4540 } 4541 } 4542} 4543 4544void AudioFlinger::PlaybackThread::Track::pause() 4545{ 4546 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4547 sp<ThreadBase> thread = mThread.promote(); 4548 if (thread != 0) { 4549 Mutex::Autolock _l(thread->mLock); 4550 if (mState == ACTIVE || mState == RESUMING) { 4551 mState = PAUSING; 4552 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4553 if (!isOutputTrack()) { 4554 thread->mLock.unlock(); 4555 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4556 thread->mLock.lock(); 4557 4558#ifdef ADD_BATTERY_DATA 4559 // to track the speaker usage 4560 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4561#endif 4562 } 4563 } 4564 } 4565} 4566 4567void AudioFlinger::PlaybackThread::Track::flush() 4568{ 4569 ALOGV("flush(%d)", mName); 4570 sp<ThreadBase> thread = mThread.promote(); 4571 if (thread != 0) { 4572 Mutex::Autolock _l(thread->mLock); 4573 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4574 mState != PAUSING) { 4575 return; 4576 } 4577 // No point remaining in PAUSED state after a flush => go to 4578 // FLUSHED state 4579 mState = FLUSHED; 4580 // do not reset the track if it is still in the process of being stopped or paused. 4581 // this will be done by prepareTracks_l() when the track is stopped. 4582 // prepareTracks_l() will see mState == FLUSHED, then 4583 // remove from active track list, reset(), and trigger presentation complete 4584 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4585 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4586 reset(); 4587 } 4588 } 4589} 4590 4591void AudioFlinger::PlaybackThread::Track::reset() 4592{ 4593 // Do not reset twice to avoid discarding data written just after a flush and before 4594 // the audioflinger thread detects the track is stopped. 4595 if (!mResetDone) { 4596 TrackBase::reset(); 4597 // Force underrun condition to avoid false underrun callback until first data is 4598 // written to buffer 4599 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4600 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4601 mFillingUpStatus = FS_FILLING; 4602 mResetDone = true; 4603 if (mState == FLUSHED) { 4604 mState = IDLE; 4605 } 4606 } 4607} 4608 4609void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4610{ 4611 mMute = muted; 4612} 4613 4614status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4615{ 4616 status_t status = DEAD_OBJECT; 4617 sp<ThreadBase> thread = mThread.promote(); 4618 if (thread != 0) { 4619 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4620 status = playbackThread->attachAuxEffect(this, EffectId); 4621 } 4622 return status; 4623} 4624 4625void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4626{ 4627 mAuxEffectId = EffectId; 4628 mAuxBuffer = buffer; 4629} 4630 4631bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4632 size_t audioHalFrames) 4633{ 4634 // a track is considered presented when the total number of frames written to audio HAL 4635 // corresponds to the number of frames written when presentationComplete() is called for the 4636 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4637 if (mPresentationCompleteFrames == 0) { 4638 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4639 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4640 mPresentationCompleteFrames, audioHalFrames); 4641 } 4642 if (framesWritten >= mPresentationCompleteFrames) { 4643 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4644 mSessionId, framesWritten); 4645 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4646 return true; 4647 } 4648 return false; 4649} 4650 4651void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4652{ 4653 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4654 if (mSyncEvents[i]->type() == type) { 4655 mSyncEvents[i]->trigger(); 4656 mSyncEvents.removeAt(i); 4657 i--; 4658 } 4659 } 4660} 4661 4662// implement VolumeBufferProvider interface 4663 4664uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4665{ 4666 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4667 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4668 uint32_t vlr = mCblk->getVolumeLR(); 4669 uint32_t vl = vlr & 0xFFFF; 4670 uint32_t vr = vlr >> 16; 4671 // track volumes come from shared memory, so can't be trusted and must be clamped 4672 if (vl > MAX_GAIN_INT) { 4673 vl = MAX_GAIN_INT; 4674 } 4675 if (vr > MAX_GAIN_INT) { 4676 vr = MAX_GAIN_INT; 4677 } 4678 // now apply the cached master volume and stream type volume; 4679 // this is trusted but lacks any synchronization or barrier so may be stale 4680 float v = mCachedVolume; 4681 vl *= v; 4682 vr *= v; 4683 // re-combine into U4.16 4684 vlr = (vr << 16) | (vl & 0xFFFF); 4685 // FIXME look at mute, pause, and stop flags 4686 return vlr; 4687} 4688 4689status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4690{ 4691 if (mState == TERMINATED || mState == PAUSED || 4692 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4693 (mState == STOPPED)))) { 4694 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4695 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4696 event->cancel(); 4697 return INVALID_OPERATION; 4698 } 4699 TrackBase::setSyncEvent(event); 4700 return NO_ERROR; 4701} 4702 4703// timed audio tracks 4704 4705sp<AudioFlinger::PlaybackThread::TimedTrack> 4706AudioFlinger::PlaybackThread::TimedTrack::create( 4707 PlaybackThread *thread, 4708 const sp<Client>& client, 4709 audio_stream_type_t streamType, 4710 uint32_t sampleRate, 4711 audio_format_t format, 4712 uint32_t channelMask, 4713 int frameCount, 4714 const sp<IMemory>& sharedBuffer, 4715 int sessionId) { 4716 if (!client->reserveTimedTrack()) 4717 return NULL; 4718 4719 return new TimedTrack( 4720 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4721 sharedBuffer, sessionId); 4722} 4723 4724AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4725 PlaybackThread *thread, 4726 const sp<Client>& client, 4727 audio_stream_type_t streamType, 4728 uint32_t sampleRate, 4729 audio_format_t format, 4730 uint32_t channelMask, 4731 int frameCount, 4732 const sp<IMemory>& sharedBuffer, 4733 int sessionId) 4734 : Track(thread, client, streamType, sampleRate, format, channelMask, 4735 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4736 mQueueHeadInFlight(false), 4737 mTrimQueueHeadOnRelease(false), 4738 mFramesPendingInQueue(0), 4739 mTimedSilenceBuffer(NULL), 4740 mTimedSilenceBufferSize(0), 4741 mTimedAudioOutputOnTime(false), 4742 mMediaTimeTransformValid(false) 4743{ 4744 LocalClock lc; 4745 mLocalTimeFreq = lc.getLocalFreq(); 4746 4747 mLocalTimeToSampleTransform.a_zero = 0; 4748 mLocalTimeToSampleTransform.b_zero = 0; 4749 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4750 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4751 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4752 &mLocalTimeToSampleTransform.a_to_b_denom); 4753 4754 mMediaTimeToSampleTransform.a_zero = 0; 4755 mMediaTimeToSampleTransform.b_zero = 0; 4756 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4757 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4758 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4759 &mMediaTimeToSampleTransform.a_to_b_denom); 4760} 4761 4762AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4763 mClient->releaseTimedTrack(); 4764 delete [] mTimedSilenceBuffer; 4765} 4766 4767status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4768 size_t size, sp<IMemory>* buffer) { 4769 4770 Mutex::Autolock _l(mTimedBufferQueueLock); 4771 4772 trimTimedBufferQueue_l(); 4773 4774 // lazily initialize the shared memory heap for timed buffers 4775 if (mTimedMemoryDealer == NULL) { 4776 const int kTimedBufferHeapSize = 512 << 10; 4777 4778 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4779 "AudioFlingerTimed"); 4780 if (mTimedMemoryDealer == NULL) 4781 return NO_MEMORY; 4782 } 4783 4784 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4785 if (newBuffer == NULL) { 4786 newBuffer = mTimedMemoryDealer->allocate(size); 4787 if (newBuffer == NULL) 4788 return NO_MEMORY; 4789 } 4790 4791 *buffer = newBuffer; 4792 return NO_ERROR; 4793} 4794 4795// caller must hold mTimedBufferQueueLock 4796void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4797 int64_t mediaTimeNow; 4798 { 4799 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4800 if (!mMediaTimeTransformValid) 4801 return; 4802 4803 int64_t targetTimeNow; 4804 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4805 ? mCCHelper.getCommonTime(&targetTimeNow) 4806 : mCCHelper.getLocalTime(&targetTimeNow); 4807 4808 if (OK != res) 4809 return; 4810 4811 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4812 &mediaTimeNow)) { 4813 return; 4814 } 4815 } 4816 4817 size_t trimEnd; 4818 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4819 int64_t bufEnd; 4820 4821 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4822 // We have a next buffer. Just use its PTS as the PTS of the frame 4823 // following the last frame in this buffer. If the stream is sparse 4824 // (ie, there are deliberate gaps left in the stream which should be 4825 // filled with silence by the TimedAudioTrack), then this can result 4826 // in one extra buffer being left un-trimmed when it could have 4827 // been. In general, this is not typical, and we would rather 4828 // optimized away the TS calculation below for the more common case 4829 // where PTSes are contiguous. 4830 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4831 } else { 4832 // We have no next buffer. Compute the PTS of the frame following 4833 // the last frame in this buffer by computing the duration of of 4834 // this frame in media time units and adding it to the PTS of the 4835 // buffer. 4836 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4837 / mCblk->frameSize; 4838 4839 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4840 &bufEnd)) { 4841 ALOGE("Failed to convert frame count of %lld to media time" 4842 " duration" " (scale factor %d/%u) in %s", 4843 frameCount, 4844 mMediaTimeToSampleTransform.a_to_b_numer, 4845 mMediaTimeToSampleTransform.a_to_b_denom, 4846 __PRETTY_FUNCTION__); 4847 break; 4848 } 4849 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4850 } 4851 4852 if (bufEnd > mediaTimeNow) 4853 break; 4854 4855 // Is the buffer we want to use in the middle of a mix operation right 4856 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4857 // from the mixer which should be coming back shortly. 4858 if (!trimEnd && mQueueHeadInFlight) { 4859 mTrimQueueHeadOnRelease = true; 4860 } 4861 } 4862 4863 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4864 if (trimStart < trimEnd) { 4865 // Update the bookkeeping for framesReady() 4866 for (size_t i = trimStart; i < trimEnd; ++i) { 4867 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4868 } 4869 4870 // Now actually remove the buffers from the queue. 4871 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4872 } 4873} 4874 4875void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4876 const char* logTag) { 4877 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4878 "%s called (reason \"%s\"), but timed buffer queue has no" 4879 " elements to trim.", __FUNCTION__, logTag); 4880 4881 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4882 mTimedBufferQueue.removeAt(0); 4883} 4884 4885void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4886 const TimedBuffer& buf, 4887 const char* logTag) { 4888 uint32_t bufBytes = buf.buffer()->size(); 4889 uint32_t consumedAlready = buf.position(); 4890 4891 ALOG_ASSERT(consumedAlready <= bufBytes, 4892 "Bad bookkeeping while updating frames pending. Timed buffer is" 4893 " only %u bytes long, but claims to have consumed %u" 4894 " bytes. (update reason: \"%s\")", 4895 bufBytes, consumedAlready, logTag); 4896 4897 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4898 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4899 "Bad bookkeeping while updating frames pending. Should have at" 4900 " least %u queued frames, but we think we have only %u. (update" 4901 " reason: \"%s\")", 4902 bufFrames, mFramesPendingInQueue, logTag); 4903 4904 mFramesPendingInQueue -= bufFrames; 4905} 4906 4907status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4908 const sp<IMemory>& buffer, int64_t pts) { 4909 4910 { 4911 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4912 if (!mMediaTimeTransformValid) 4913 return INVALID_OPERATION; 4914 } 4915 4916 Mutex::Autolock _l(mTimedBufferQueueLock); 4917 4918 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4919 mFramesPendingInQueue += bufFrames; 4920 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4921 4922 return NO_ERROR; 4923} 4924 4925status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4926 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4927 4928 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4929 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4930 target); 4931 4932 if (!(target == TimedAudioTrack::LOCAL_TIME || 4933 target == TimedAudioTrack::COMMON_TIME)) { 4934 return BAD_VALUE; 4935 } 4936 4937 Mutex::Autolock lock(mMediaTimeTransformLock); 4938 mMediaTimeTransform = xform; 4939 mMediaTimeTransformTarget = target; 4940 mMediaTimeTransformValid = true; 4941 4942 return NO_ERROR; 4943} 4944 4945#define min(a, b) ((a) < (b) ? (a) : (b)) 4946 4947// implementation of getNextBuffer for tracks whose buffers have timestamps 4948status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4949 AudioBufferProvider::Buffer* buffer, int64_t pts) 4950{ 4951 if (pts == AudioBufferProvider::kInvalidPTS) { 4952 buffer->raw = 0; 4953 buffer->frameCount = 0; 4954 mTimedAudioOutputOnTime = false; 4955 return INVALID_OPERATION; 4956 } 4957 4958 Mutex::Autolock _l(mTimedBufferQueueLock); 4959 4960 ALOG_ASSERT(!mQueueHeadInFlight, 4961 "getNextBuffer called without releaseBuffer!"); 4962 4963 while (true) { 4964 4965 // if we have no timed buffers, then fail 4966 if (mTimedBufferQueue.isEmpty()) { 4967 buffer->raw = 0; 4968 buffer->frameCount = 0; 4969 return NOT_ENOUGH_DATA; 4970 } 4971 4972 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4973 4974 // calculate the PTS of the head of the timed buffer queue expressed in 4975 // local time 4976 int64_t headLocalPTS; 4977 { 4978 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4979 4980 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4981 4982 if (mMediaTimeTransform.a_to_b_denom == 0) { 4983 // the transform represents a pause, so yield silence 4984 timedYieldSilence_l(buffer->frameCount, buffer); 4985 return NO_ERROR; 4986 } 4987 4988 int64_t transformedPTS; 4989 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4990 &transformedPTS)) { 4991 // the transform failed. this shouldn't happen, but if it does 4992 // then just drop this buffer 4993 ALOGW("timedGetNextBuffer transform failed"); 4994 buffer->raw = 0; 4995 buffer->frameCount = 0; 4996 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4997 return NO_ERROR; 4998 } 4999 5000 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5001 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5002 &headLocalPTS)) { 5003 buffer->raw = 0; 5004 buffer->frameCount = 0; 5005 return INVALID_OPERATION; 5006 } 5007 } else { 5008 headLocalPTS = transformedPTS; 5009 } 5010 } 5011 5012 // adjust the head buffer's PTS to reflect the portion of the head buffer 5013 // that has already been consumed 5014 int64_t effectivePTS = headLocalPTS + 5015 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5016 5017 // Calculate the delta in samples between the head of the input buffer 5018 // queue and the start of the next output buffer that will be written. 5019 // If the transformation fails because of over or underflow, it means 5020 // that the sample's position in the output stream is so far out of 5021 // whack that it should just be dropped. 5022 int64_t sampleDelta; 5023 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5024 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5025 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5026 " mix"); 5027 continue; 5028 } 5029 if (!mLocalTimeToSampleTransform.doForwardTransform( 5030 (effectivePTS - pts) << 32, &sampleDelta)) { 5031 ALOGV("*** too late during sample rate transform: dropped buffer"); 5032 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5033 continue; 5034 } 5035 5036 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5037 " sampleDelta=[%d.%08x]", 5038 head.pts(), head.position(), pts, 5039 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5040 + (sampleDelta >> 32)), 5041 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5042 5043 // if the delta between the ideal placement for the next input sample and 5044 // the current output position is within this threshold, then we will 5045 // concatenate the next input samples to the previous output 5046 const int64_t kSampleContinuityThreshold = 5047 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5048 5049 // if this is the first buffer of audio that we're emitting from this track 5050 // then it should be almost exactly on time. 5051 const int64_t kSampleStartupThreshold = 1LL << 32; 5052 5053 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5054 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5055 // the next input is close enough to being on time, so concatenate it 5056 // with the last output 5057 timedYieldSamples_l(buffer); 5058 5059 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5060 head.position(), buffer->frameCount); 5061 return NO_ERROR; 5062 } 5063 5064 // Looks like our output is not on time. Reset our on timed status. 5065 // Next time we mix samples from our input queue, then should be within 5066 // the StartupThreshold. 5067 mTimedAudioOutputOnTime = false; 5068 if (sampleDelta > 0) { 5069 // the gap between the current output position and the proper start of 5070 // the next input sample is too big, so fill it with silence 5071 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5072 5073 timedYieldSilence_l(framesUntilNextInput, buffer); 5074 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5075 return NO_ERROR; 5076 } else { 5077 // the next input sample is late 5078 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5079 size_t onTimeSamplePosition = 5080 head.position() + lateFrames * mCblk->frameSize; 5081 5082 if (onTimeSamplePosition > head.buffer()->size()) { 5083 // all the remaining samples in the head are too late, so 5084 // drop it and move on 5085 ALOGV("*** too late: dropped buffer"); 5086 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5087 continue; 5088 } else { 5089 // skip over the late samples 5090 head.setPosition(onTimeSamplePosition); 5091 5092 // yield the available samples 5093 timedYieldSamples_l(buffer); 5094 5095 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5096 return NO_ERROR; 5097 } 5098 } 5099 } 5100} 5101 5102// Yield samples from the timed buffer queue head up to the given output 5103// buffer's capacity. 5104// 5105// Caller must hold mTimedBufferQueueLock 5106void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5107 AudioBufferProvider::Buffer* buffer) { 5108 5109 const TimedBuffer& head = mTimedBufferQueue[0]; 5110 5111 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5112 head.position()); 5113 5114 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5115 mCblk->frameSize); 5116 size_t framesRequested = buffer->frameCount; 5117 buffer->frameCount = min(framesLeftInHead, framesRequested); 5118 5119 mQueueHeadInFlight = true; 5120 mTimedAudioOutputOnTime = true; 5121} 5122 5123// Yield samples of silence up to the given output buffer's capacity 5124// 5125// Caller must hold mTimedBufferQueueLock 5126void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5127 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5128 5129 // lazily allocate a buffer filled with silence 5130 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5131 delete [] mTimedSilenceBuffer; 5132 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5133 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5134 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5135 } 5136 5137 buffer->raw = mTimedSilenceBuffer; 5138 size_t framesRequested = buffer->frameCount; 5139 buffer->frameCount = min(numFrames, framesRequested); 5140 5141 mTimedAudioOutputOnTime = false; 5142} 5143 5144// AudioBufferProvider interface 5145void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5146 AudioBufferProvider::Buffer* buffer) { 5147 5148 Mutex::Autolock _l(mTimedBufferQueueLock); 5149 5150 // If the buffer which was just released is part of the buffer at the head 5151 // of the queue, be sure to update the amt of the buffer which has been 5152 // consumed. If the buffer being returned is not part of the head of the 5153 // queue, its either because the buffer is part of the silence buffer, or 5154 // because the head of the timed queue was trimmed after the mixer called 5155 // getNextBuffer but before the mixer called releaseBuffer. 5156 if (buffer->raw == mTimedSilenceBuffer) { 5157 ALOG_ASSERT(!mQueueHeadInFlight, 5158 "Queue head in flight during release of silence buffer!"); 5159 goto done; 5160 } 5161 5162 ALOG_ASSERT(mQueueHeadInFlight, 5163 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5164 " head in flight."); 5165 5166 if (mTimedBufferQueue.size()) { 5167 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5168 5169 void* start = head.buffer()->pointer(); 5170 void* end = reinterpret_cast<void*>( 5171 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5172 + head.buffer()->size()); 5173 5174 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5175 "released buffer not within the head of the timed buffer" 5176 " queue; qHead = [%p, %p], released buffer = %p", 5177 start, end, buffer->raw); 5178 5179 head.setPosition(head.position() + 5180 (buffer->frameCount * mCblk->frameSize)); 5181 mQueueHeadInFlight = false; 5182 5183 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5184 "Bad bookkeeping during releaseBuffer! Should have at" 5185 " least %u queued frames, but we think we have only %u", 5186 buffer->frameCount, mFramesPendingInQueue); 5187 5188 mFramesPendingInQueue -= buffer->frameCount; 5189 5190 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5191 || mTrimQueueHeadOnRelease) { 5192 trimTimedBufferQueueHead_l("releaseBuffer"); 5193 mTrimQueueHeadOnRelease = false; 5194 } 5195 } else { 5196 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5197 " buffers in the timed buffer queue"); 5198 } 5199 5200done: 5201 buffer->raw = 0; 5202 buffer->frameCount = 0; 5203} 5204 5205size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5206 Mutex::Autolock _l(mTimedBufferQueueLock); 5207 return mFramesPendingInQueue; 5208} 5209 5210AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5211 : mPTS(0), mPosition(0) {} 5212 5213AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5214 const sp<IMemory>& buffer, int64_t pts) 5215 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5216 5217// ---------------------------------------------------------------------------- 5218 5219// RecordTrack constructor must be called with AudioFlinger::mLock held 5220AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5221 RecordThread *thread, 5222 const sp<Client>& client, 5223 uint32_t sampleRate, 5224 audio_format_t format, 5225 uint32_t channelMask, 5226 int frameCount, 5227 int sessionId) 5228 : TrackBase(thread, client, sampleRate, format, 5229 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5230 mOverflow(false) 5231{ 5232 if (mCblk != NULL) { 5233 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5234 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5235 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5236 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5237 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5238 } else { 5239 mCblk->frameSize = sizeof(int8_t); 5240 } 5241 } 5242} 5243 5244AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5245{ 5246 sp<ThreadBase> thread = mThread.promote(); 5247 if (thread != 0) { 5248 AudioSystem::releaseInput(thread->id()); 5249 } 5250} 5251 5252// AudioBufferProvider interface 5253status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5254{ 5255 audio_track_cblk_t* cblk = this->cblk(); 5256 uint32_t framesAvail; 5257 uint32_t framesReq = buffer->frameCount; 5258 5259 // Check if last stepServer failed, try to step now 5260 if (mStepServerFailed) { 5261 if (!step()) goto getNextBuffer_exit; 5262 ALOGV("stepServer recovered"); 5263 mStepServerFailed = false; 5264 } 5265 5266 framesAvail = cblk->framesAvailable_l(); 5267 5268 if (CC_LIKELY(framesAvail)) { 5269 uint32_t s = cblk->server; 5270 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5271 5272 if (framesReq > framesAvail) { 5273 framesReq = framesAvail; 5274 } 5275 if (framesReq > bufferEnd - s) { 5276 framesReq = bufferEnd - s; 5277 } 5278 5279 buffer->raw = getBuffer(s, framesReq); 5280 if (buffer->raw == NULL) goto getNextBuffer_exit; 5281 5282 buffer->frameCount = framesReq; 5283 return NO_ERROR; 5284 } 5285 5286getNextBuffer_exit: 5287 buffer->raw = NULL; 5288 buffer->frameCount = 0; 5289 return NOT_ENOUGH_DATA; 5290} 5291 5292status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5293 int triggerSession) 5294{ 5295 sp<ThreadBase> thread = mThread.promote(); 5296 if (thread != 0) { 5297 RecordThread *recordThread = (RecordThread *)thread.get(); 5298 return recordThread->start(this, event, triggerSession); 5299 } else { 5300 return BAD_VALUE; 5301 } 5302} 5303 5304void AudioFlinger::RecordThread::RecordTrack::stop() 5305{ 5306 sp<ThreadBase> thread = mThread.promote(); 5307 if (thread != 0) { 5308 RecordThread *recordThread = (RecordThread *)thread.get(); 5309 recordThread->stop(this); 5310 TrackBase::reset(); 5311 // Force overrun condition to avoid false overrun callback until first data is 5312 // read from buffer 5313 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5314 } 5315} 5316 5317void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5318{ 5319 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5320 (mClient == 0) ? getpid_cached : mClient->pid(), 5321 mFormat, 5322 mChannelMask, 5323 mSessionId, 5324 mFrameCount, 5325 mState, 5326 mCblk->sampleRate, 5327 mCblk->server, 5328 mCblk->user); 5329} 5330 5331 5332// ---------------------------------------------------------------------------- 5333 5334AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5335 PlaybackThread *playbackThread, 5336 DuplicatingThread *sourceThread, 5337 uint32_t sampleRate, 5338 audio_format_t format, 5339 uint32_t channelMask, 5340 int frameCount) 5341 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5342 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5343 mActive(false), mSourceThread(sourceThread) 5344{ 5345 5346 if (mCblk != NULL) { 5347 mCblk->flags |= CBLK_DIRECTION_OUT; 5348 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5349 mOutBuffer.frameCount = 0; 5350 playbackThread->mTracks.add(this); 5351 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5352 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5353 mCblk, mBuffer, mCblk->buffers, 5354 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5355 } else { 5356 ALOGW("Error creating output track on thread %p", playbackThread); 5357 } 5358} 5359 5360AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5361{ 5362 clearBufferQueue(); 5363} 5364 5365status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5366 int triggerSession) 5367{ 5368 status_t status = Track::start(event, triggerSession); 5369 if (status != NO_ERROR) { 5370 return status; 5371 } 5372 5373 mActive = true; 5374 mRetryCount = 127; 5375 return status; 5376} 5377 5378void AudioFlinger::PlaybackThread::OutputTrack::stop() 5379{ 5380 Track::stop(); 5381 clearBufferQueue(); 5382 mOutBuffer.frameCount = 0; 5383 mActive = false; 5384} 5385 5386bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5387{ 5388 Buffer *pInBuffer; 5389 Buffer inBuffer; 5390 uint32_t channelCount = mChannelCount; 5391 bool outputBufferFull = false; 5392 inBuffer.frameCount = frames; 5393 inBuffer.i16 = data; 5394 5395 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5396 5397 if (!mActive && frames != 0) { 5398 start(); 5399 sp<ThreadBase> thread = mThread.promote(); 5400 if (thread != 0) { 5401 MixerThread *mixerThread = (MixerThread *)thread.get(); 5402 if (mCblk->frameCount > frames){ 5403 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5404 uint32_t startFrames = (mCblk->frameCount - frames); 5405 pInBuffer = new Buffer; 5406 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5407 pInBuffer->frameCount = startFrames; 5408 pInBuffer->i16 = pInBuffer->mBuffer; 5409 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5410 mBufferQueue.add(pInBuffer); 5411 } else { 5412 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5413 } 5414 } 5415 } 5416 } 5417 5418 while (waitTimeLeftMs) { 5419 // First write pending buffers, then new data 5420 if (mBufferQueue.size()) { 5421 pInBuffer = mBufferQueue.itemAt(0); 5422 } else { 5423 pInBuffer = &inBuffer; 5424 } 5425 5426 if (pInBuffer->frameCount == 0) { 5427 break; 5428 } 5429 5430 if (mOutBuffer.frameCount == 0) { 5431 mOutBuffer.frameCount = pInBuffer->frameCount; 5432 nsecs_t startTime = systemTime(); 5433 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5434 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5435 outputBufferFull = true; 5436 break; 5437 } 5438 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5439 if (waitTimeLeftMs >= waitTimeMs) { 5440 waitTimeLeftMs -= waitTimeMs; 5441 } else { 5442 waitTimeLeftMs = 0; 5443 } 5444 } 5445 5446 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5447 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5448 mCblk->stepUser(outFrames); 5449 pInBuffer->frameCount -= outFrames; 5450 pInBuffer->i16 += outFrames * channelCount; 5451 mOutBuffer.frameCount -= outFrames; 5452 mOutBuffer.i16 += outFrames * channelCount; 5453 5454 if (pInBuffer->frameCount == 0) { 5455 if (mBufferQueue.size()) { 5456 mBufferQueue.removeAt(0); 5457 delete [] pInBuffer->mBuffer; 5458 delete pInBuffer; 5459 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5460 } else { 5461 break; 5462 } 5463 } 5464 } 5465 5466 // If we could not write all frames, allocate a buffer and queue it for next time. 5467 if (inBuffer.frameCount) { 5468 sp<ThreadBase> thread = mThread.promote(); 5469 if (thread != 0 && !thread->standby()) { 5470 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5471 pInBuffer = new Buffer; 5472 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5473 pInBuffer->frameCount = inBuffer.frameCount; 5474 pInBuffer->i16 = pInBuffer->mBuffer; 5475 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5476 mBufferQueue.add(pInBuffer); 5477 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5478 } else { 5479 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5480 } 5481 } 5482 } 5483 5484 // Calling write() with a 0 length buffer, means that no more data will be written: 5485 // If no more buffers are pending, fill output track buffer to make sure it is started 5486 // by output mixer. 5487 if (frames == 0 && mBufferQueue.size() == 0) { 5488 if (mCblk->user < mCblk->frameCount) { 5489 frames = mCblk->frameCount - mCblk->user; 5490 pInBuffer = new Buffer; 5491 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5492 pInBuffer->frameCount = frames; 5493 pInBuffer->i16 = pInBuffer->mBuffer; 5494 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5495 mBufferQueue.add(pInBuffer); 5496 } else if (mActive) { 5497 stop(); 5498 } 5499 } 5500 5501 return outputBufferFull; 5502} 5503 5504status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5505{ 5506 int active; 5507 status_t result; 5508 audio_track_cblk_t* cblk = mCblk; 5509 uint32_t framesReq = buffer->frameCount; 5510 5511// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5512 buffer->frameCount = 0; 5513 5514 uint32_t framesAvail = cblk->framesAvailable(); 5515 5516 5517 if (framesAvail == 0) { 5518 Mutex::Autolock _l(cblk->lock); 5519 goto start_loop_here; 5520 while (framesAvail == 0) { 5521 active = mActive; 5522 if (CC_UNLIKELY(!active)) { 5523 ALOGV("Not active and NO_MORE_BUFFERS"); 5524 return NO_MORE_BUFFERS; 5525 } 5526 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5527 if (result != NO_ERROR) { 5528 return NO_MORE_BUFFERS; 5529 } 5530 // read the server count again 5531 start_loop_here: 5532 framesAvail = cblk->framesAvailable_l(); 5533 } 5534 } 5535 5536// if (framesAvail < framesReq) { 5537// return NO_MORE_BUFFERS; 5538// } 5539 5540 if (framesReq > framesAvail) { 5541 framesReq = framesAvail; 5542 } 5543 5544 uint32_t u = cblk->user; 5545 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5546 5547 if (framesReq > bufferEnd - u) { 5548 framesReq = bufferEnd - u; 5549 } 5550 5551 buffer->frameCount = framesReq; 5552 buffer->raw = (void *)cblk->buffer(u); 5553 return NO_ERROR; 5554} 5555 5556 5557void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5558{ 5559 size_t size = mBufferQueue.size(); 5560 5561 for (size_t i = 0; i < size; i++) { 5562 Buffer *pBuffer = mBufferQueue.itemAt(i); 5563 delete [] pBuffer->mBuffer; 5564 delete pBuffer; 5565 } 5566 mBufferQueue.clear(); 5567} 5568 5569// ---------------------------------------------------------------------------- 5570 5571AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5572 : RefBase(), 5573 mAudioFlinger(audioFlinger), 5574 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5575 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5576 mPid(pid), 5577 mTimedTrackCount(0) 5578{ 5579 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5580} 5581 5582// Client destructor must be called with AudioFlinger::mLock held 5583AudioFlinger::Client::~Client() 5584{ 5585 mAudioFlinger->removeClient_l(mPid); 5586} 5587 5588sp<MemoryDealer> AudioFlinger::Client::heap() const 5589{ 5590 return mMemoryDealer; 5591} 5592 5593// Reserve one of the limited slots for a timed audio track associated 5594// with this client 5595bool AudioFlinger::Client::reserveTimedTrack() 5596{ 5597 const int kMaxTimedTracksPerClient = 4; 5598 5599 Mutex::Autolock _l(mTimedTrackLock); 5600 5601 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5602 ALOGW("can not create timed track - pid %d has exceeded the limit", 5603 mPid); 5604 return false; 5605 } 5606 5607 mTimedTrackCount++; 5608 return true; 5609} 5610 5611// Release a slot for a timed audio track 5612void AudioFlinger::Client::releaseTimedTrack() 5613{ 5614 Mutex::Autolock _l(mTimedTrackLock); 5615 mTimedTrackCount--; 5616} 5617 5618// ---------------------------------------------------------------------------- 5619 5620AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5621 const sp<IAudioFlingerClient>& client, 5622 pid_t pid) 5623 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5624{ 5625} 5626 5627AudioFlinger::NotificationClient::~NotificationClient() 5628{ 5629} 5630 5631void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5632{ 5633 sp<NotificationClient> keep(this); 5634 mAudioFlinger->removeNotificationClient(mPid); 5635} 5636 5637// ---------------------------------------------------------------------------- 5638 5639AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5640 : BnAudioTrack(), 5641 mTrack(track) 5642{ 5643} 5644 5645AudioFlinger::TrackHandle::~TrackHandle() { 5646 // just stop the track on deletion, associated resources 5647 // will be freed from the main thread once all pending buffers have 5648 // been played. Unless it's not in the active track list, in which 5649 // case we free everything now... 5650 mTrack->destroy(); 5651} 5652 5653sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5654 return mTrack->getCblk(); 5655} 5656 5657status_t AudioFlinger::TrackHandle::start() { 5658 return mTrack->start(); 5659} 5660 5661void AudioFlinger::TrackHandle::stop() { 5662 mTrack->stop(); 5663} 5664 5665void AudioFlinger::TrackHandle::flush() { 5666 mTrack->flush(); 5667} 5668 5669void AudioFlinger::TrackHandle::mute(bool e) { 5670 mTrack->mute(e); 5671} 5672 5673void AudioFlinger::TrackHandle::pause() { 5674 mTrack->pause(); 5675} 5676 5677status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5678{ 5679 return mTrack->attachAuxEffect(EffectId); 5680} 5681 5682status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5683 sp<IMemory>* buffer) { 5684 if (!mTrack->isTimedTrack()) 5685 return INVALID_OPERATION; 5686 5687 PlaybackThread::TimedTrack* tt = 5688 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5689 return tt->allocateTimedBuffer(size, buffer); 5690} 5691 5692status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5693 int64_t pts) { 5694 if (!mTrack->isTimedTrack()) 5695 return INVALID_OPERATION; 5696 5697 PlaybackThread::TimedTrack* tt = 5698 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5699 return tt->queueTimedBuffer(buffer, pts); 5700} 5701 5702status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5703 const LinearTransform& xform, int target) { 5704 5705 if (!mTrack->isTimedTrack()) 5706 return INVALID_OPERATION; 5707 5708 PlaybackThread::TimedTrack* tt = 5709 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5710 return tt->setMediaTimeTransform( 5711 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5712} 5713 5714status_t AudioFlinger::TrackHandle::onTransact( 5715 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5716{ 5717 return BnAudioTrack::onTransact(code, data, reply, flags); 5718} 5719 5720// ---------------------------------------------------------------------------- 5721 5722sp<IAudioRecord> AudioFlinger::openRecord( 5723 pid_t pid, 5724 audio_io_handle_t input, 5725 uint32_t sampleRate, 5726 audio_format_t format, 5727 uint32_t channelMask, 5728 int frameCount, 5729 IAudioFlinger::track_flags_t flags, 5730 int *sessionId, 5731 status_t *status) 5732{ 5733 sp<RecordThread::RecordTrack> recordTrack; 5734 sp<RecordHandle> recordHandle; 5735 sp<Client> client; 5736 status_t lStatus; 5737 RecordThread *thread; 5738 size_t inFrameCount; 5739 int lSessionId; 5740 5741 // check calling permissions 5742 if (!recordingAllowed()) { 5743 lStatus = PERMISSION_DENIED; 5744 goto Exit; 5745 } 5746 5747 // add client to list 5748 { // scope for mLock 5749 Mutex::Autolock _l(mLock); 5750 thread = checkRecordThread_l(input); 5751 if (thread == NULL) { 5752 lStatus = BAD_VALUE; 5753 goto Exit; 5754 } 5755 5756 client = registerPid_l(pid); 5757 5758 // If no audio session id is provided, create one here 5759 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5760 lSessionId = *sessionId; 5761 } else { 5762 lSessionId = nextUniqueId(); 5763 if (sessionId != NULL) { 5764 *sessionId = lSessionId; 5765 } 5766 } 5767 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5768 recordTrack = thread->createRecordTrack_l(client, 5769 sampleRate, 5770 format, 5771 channelMask, 5772 frameCount, 5773 lSessionId, 5774 &lStatus); 5775 } 5776 if (lStatus != NO_ERROR) { 5777 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5778 // destructor is called by the TrackBase destructor with mLock held 5779 client.clear(); 5780 recordTrack.clear(); 5781 goto Exit; 5782 } 5783 5784 // return to handle to client 5785 recordHandle = new RecordHandle(recordTrack); 5786 lStatus = NO_ERROR; 5787 5788Exit: 5789 if (status) { 5790 *status = lStatus; 5791 } 5792 return recordHandle; 5793} 5794 5795// ---------------------------------------------------------------------------- 5796 5797AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5798 : BnAudioRecord(), 5799 mRecordTrack(recordTrack) 5800{ 5801} 5802 5803AudioFlinger::RecordHandle::~RecordHandle() { 5804 stop(); 5805} 5806 5807sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5808 return mRecordTrack->getCblk(); 5809} 5810 5811status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5812 ALOGV("RecordHandle::start()"); 5813 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5814} 5815 5816void AudioFlinger::RecordHandle::stop() { 5817 ALOGV("RecordHandle::stop()"); 5818 mRecordTrack->stop(); 5819} 5820 5821status_t AudioFlinger::RecordHandle::onTransact( 5822 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5823{ 5824 return BnAudioRecord::onTransact(code, data, reply, flags); 5825} 5826 5827// ---------------------------------------------------------------------------- 5828 5829AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5830 AudioStreamIn *input, 5831 uint32_t sampleRate, 5832 uint32_t channels, 5833 audio_io_handle_t id, 5834 uint32_t device) : 5835 ThreadBase(audioFlinger, id, device, RECORD), 5836 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5837 // mRsmpInIndex and mInputBytes set by readInputParameters() 5838 mReqChannelCount(popcount(channels)), 5839 mReqSampleRate(sampleRate) 5840 // mBytesRead is only meaningful while active, and so is cleared in start() 5841 // (but might be better to also clear here for dump?) 5842{ 5843 snprintf(mName, kNameLength, "AudioIn_%X", id); 5844 5845 readInputParameters(); 5846} 5847 5848 5849AudioFlinger::RecordThread::~RecordThread() 5850{ 5851 delete[] mRsmpInBuffer; 5852 delete mResampler; 5853 delete[] mRsmpOutBuffer; 5854} 5855 5856void AudioFlinger::RecordThread::onFirstRef() 5857{ 5858 run(mName, PRIORITY_URGENT_AUDIO); 5859} 5860 5861status_t AudioFlinger::RecordThread::readyToRun() 5862{ 5863 status_t status = initCheck(); 5864 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5865 return status; 5866} 5867 5868bool AudioFlinger::RecordThread::threadLoop() 5869{ 5870 AudioBufferProvider::Buffer buffer; 5871 sp<RecordTrack> activeTrack; 5872 Vector< sp<EffectChain> > effectChains; 5873 5874 nsecs_t lastWarning = 0; 5875 5876 acquireWakeLock(); 5877 5878 // start recording 5879 while (!exitPending()) { 5880 5881 processConfigEvents(); 5882 5883 { // scope for mLock 5884 Mutex::Autolock _l(mLock); 5885 checkForNewParameters_l(); 5886 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5887 if (!mStandby) { 5888 mInput->stream->common.standby(&mInput->stream->common); 5889 mStandby = true; 5890 } 5891 5892 if (exitPending()) break; 5893 5894 releaseWakeLock_l(); 5895 ALOGV("RecordThread: loop stopping"); 5896 // go to sleep 5897 mWaitWorkCV.wait(mLock); 5898 ALOGV("RecordThread: loop starting"); 5899 acquireWakeLock_l(); 5900 continue; 5901 } 5902 if (mActiveTrack != 0) { 5903 if (mActiveTrack->mState == TrackBase::PAUSING) { 5904 if (!mStandby) { 5905 mInput->stream->common.standby(&mInput->stream->common); 5906 mStandby = true; 5907 } 5908 mActiveTrack.clear(); 5909 mStartStopCond.broadcast(); 5910 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5911 if (mReqChannelCount != mActiveTrack->channelCount()) { 5912 mActiveTrack.clear(); 5913 mStartStopCond.broadcast(); 5914 } else if (mBytesRead != 0) { 5915 // record start succeeds only if first read from audio input 5916 // succeeds 5917 if (mBytesRead > 0) { 5918 mActiveTrack->mState = TrackBase::ACTIVE; 5919 } else { 5920 mActiveTrack.clear(); 5921 } 5922 mStartStopCond.broadcast(); 5923 } 5924 mStandby = false; 5925 } 5926 } 5927 lockEffectChains_l(effectChains); 5928 } 5929 5930 if (mActiveTrack != 0) { 5931 if (mActiveTrack->mState != TrackBase::ACTIVE && 5932 mActiveTrack->mState != TrackBase::RESUMING) { 5933 unlockEffectChains(effectChains); 5934 usleep(kRecordThreadSleepUs); 5935 continue; 5936 } 5937 for (size_t i = 0; i < effectChains.size(); i ++) { 5938 effectChains[i]->process_l(); 5939 } 5940 5941 buffer.frameCount = mFrameCount; 5942 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5943 size_t framesOut = buffer.frameCount; 5944 if (mResampler == NULL) { 5945 // no resampling 5946 while (framesOut) { 5947 size_t framesIn = mFrameCount - mRsmpInIndex; 5948 if (framesIn) { 5949 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5950 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5951 if (framesIn > framesOut) 5952 framesIn = framesOut; 5953 mRsmpInIndex += framesIn; 5954 framesOut -= framesIn; 5955 if ((int)mChannelCount == mReqChannelCount || 5956 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5957 memcpy(dst, src, framesIn * mFrameSize); 5958 } else { 5959 int16_t *src16 = (int16_t *)src; 5960 int16_t *dst16 = (int16_t *)dst; 5961 if (mChannelCount == 1) { 5962 while (framesIn--) { 5963 *dst16++ = *src16; 5964 *dst16++ = *src16++; 5965 } 5966 } else { 5967 while (framesIn--) { 5968 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5969 src16 += 2; 5970 } 5971 } 5972 } 5973 } 5974 if (framesOut && mFrameCount == mRsmpInIndex) { 5975 if (framesOut == mFrameCount && 5976 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5977 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5978 framesOut = 0; 5979 } else { 5980 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5981 mRsmpInIndex = 0; 5982 } 5983 if (mBytesRead < 0) { 5984 ALOGE("Error reading audio input"); 5985 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5986 // Force input into standby so that it tries to 5987 // recover at next read attempt 5988 mInput->stream->common.standby(&mInput->stream->common); 5989 usleep(kRecordThreadSleepUs); 5990 } 5991 mRsmpInIndex = mFrameCount; 5992 framesOut = 0; 5993 buffer.frameCount = 0; 5994 } 5995 } 5996 } 5997 } else { 5998 // resampling 5999 6000 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6001 // alter output frame count as if we were expecting stereo samples 6002 if (mChannelCount == 1 && mReqChannelCount == 1) { 6003 framesOut >>= 1; 6004 } 6005 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6006 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6007 // are 32 bit aligned which should be always true. 6008 if (mChannelCount == 2 && mReqChannelCount == 1) { 6009 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6010 // the resampler always outputs stereo samples: do post stereo to mono conversion 6011 int16_t *src = (int16_t *)mRsmpOutBuffer; 6012 int16_t *dst = buffer.i16; 6013 while (framesOut--) { 6014 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6015 src += 2; 6016 } 6017 } else { 6018 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6019 } 6020 6021 } 6022 if (mFramestoDrop == 0) { 6023 mActiveTrack->releaseBuffer(&buffer); 6024 } else { 6025 if (mFramestoDrop > 0) { 6026 mFramestoDrop -= buffer.frameCount; 6027 if (mFramestoDrop <= 0) { 6028 clearSyncStartEvent(); 6029 } 6030 } else { 6031 mFramestoDrop += buffer.frameCount; 6032 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6033 mSyncStartEvent->isCancelled()) { 6034 ALOGW("Synced record %s, session %d, trigger session %d", 6035 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6036 mActiveTrack->sessionId(), 6037 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6038 clearSyncStartEvent(); 6039 } 6040 } 6041 } 6042 mActiveTrack->overflow(); 6043 } 6044 // client isn't retrieving buffers fast enough 6045 else { 6046 if (!mActiveTrack->setOverflow()) { 6047 nsecs_t now = systemTime(); 6048 if ((now - lastWarning) > kWarningThrottleNs) { 6049 ALOGW("RecordThread: buffer overflow"); 6050 lastWarning = now; 6051 } 6052 } 6053 // Release the processor for a while before asking for a new buffer. 6054 // This will give the application more chance to read from the buffer and 6055 // clear the overflow. 6056 usleep(kRecordThreadSleepUs); 6057 } 6058 } 6059 // enable changes in effect chain 6060 unlockEffectChains(effectChains); 6061 effectChains.clear(); 6062 } 6063 6064 if (!mStandby) { 6065 mInput->stream->common.standby(&mInput->stream->common); 6066 } 6067 mActiveTrack.clear(); 6068 6069 mStartStopCond.broadcast(); 6070 6071 releaseWakeLock(); 6072 6073 ALOGV("RecordThread %p exiting", this); 6074 return false; 6075} 6076 6077 6078sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6079 const sp<AudioFlinger::Client>& client, 6080 uint32_t sampleRate, 6081 audio_format_t format, 6082 int channelMask, 6083 int frameCount, 6084 int sessionId, 6085 status_t *status) 6086{ 6087 sp<RecordTrack> track; 6088 status_t lStatus; 6089 6090 lStatus = initCheck(); 6091 if (lStatus != NO_ERROR) { 6092 ALOGE("Audio driver not initialized."); 6093 goto Exit; 6094 } 6095 6096 { // scope for mLock 6097 Mutex::Autolock _l(mLock); 6098 6099 track = new RecordTrack(this, client, sampleRate, 6100 format, channelMask, frameCount, sessionId); 6101 6102 if (track->getCblk() == 0) { 6103 lStatus = NO_MEMORY; 6104 goto Exit; 6105 } 6106 6107 mTrack = track.get(); 6108 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6109 bool suspend = audio_is_bluetooth_sco_device( 6110 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6111 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6112 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6113 } 6114 lStatus = NO_ERROR; 6115 6116Exit: 6117 if (status) { 6118 *status = lStatus; 6119 } 6120 return track; 6121} 6122 6123status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6124 AudioSystem::sync_event_t event, 6125 int triggerSession) 6126{ 6127 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6128 sp<ThreadBase> strongMe = this; 6129 status_t status = NO_ERROR; 6130 6131 if (event == AudioSystem::SYNC_EVENT_NONE) { 6132 clearSyncStartEvent(); 6133 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6134 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6135 triggerSession, 6136 recordTrack->sessionId(), 6137 syncStartEventCallback, 6138 this); 6139 // Sync event can be cancelled by the trigger session if the track is not in a 6140 // compatible state in which case we start record immediately 6141 if (mSyncStartEvent->isCancelled()) { 6142 clearSyncStartEvent(); 6143 } else { 6144 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6145 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6146 } 6147 } 6148 6149 { 6150 AutoMutex lock(mLock); 6151 if (mActiveTrack != 0) { 6152 if (recordTrack != mActiveTrack.get()) { 6153 status = -EBUSY; 6154 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6155 mActiveTrack->mState = TrackBase::ACTIVE; 6156 } 6157 return status; 6158 } 6159 6160 recordTrack->mState = TrackBase::IDLE; 6161 mActiveTrack = recordTrack; 6162 mLock.unlock(); 6163 status_t status = AudioSystem::startInput(mId); 6164 mLock.lock(); 6165 if (status != NO_ERROR) { 6166 mActiveTrack.clear(); 6167 clearSyncStartEvent(); 6168 return status; 6169 } 6170 mRsmpInIndex = mFrameCount; 6171 mBytesRead = 0; 6172 if (mResampler != NULL) { 6173 mResampler->reset(); 6174 } 6175 mActiveTrack->mState = TrackBase::RESUMING; 6176 // signal thread to start 6177 ALOGV("Signal record thread"); 6178 mWaitWorkCV.signal(); 6179 // do not wait for mStartStopCond if exiting 6180 if (exitPending()) { 6181 mActiveTrack.clear(); 6182 status = INVALID_OPERATION; 6183 goto startError; 6184 } 6185 mStartStopCond.wait(mLock); 6186 if (mActiveTrack == 0) { 6187 ALOGV("Record failed to start"); 6188 status = BAD_VALUE; 6189 goto startError; 6190 } 6191 ALOGV("Record started OK"); 6192 return status; 6193 } 6194startError: 6195 AudioSystem::stopInput(mId); 6196 clearSyncStartEvent(); 6197 return status; 6198} 6199 6200void AudioFlinger::RecordThread::clearSyncStartEvent() 6201{ 6202 if (mSyncStartEvent != 0) { 6203 mSyncStartEvent->cancel(); 6204 } 6205 mSyncStartEvent.clear(); 6206 mFramestoDrop = 0; 6207} 6208 6209void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6210{ 6211 sp<SyncEvent> strongEvent = event.promote(); 6212 6213 if (strongEvent != 0) { 6214 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6215 me->handleSyncStartEvent(strongEvent); 6216 } 6217} 6218 6219void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6220{ 6221 if (event == mSyncStartEvent) { 6222 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6223 // from audio HAL 6224 mFramestoDrop = mFrameCount * 2; 6225 } 6226} 6227 6228void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6229 ALOGV("RecordThread::stop"); 6230 sp<ThreadBase> strongMe = this; 6231 { 6232 AutoMutex lock(mLock); 6233 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6234 mActiveTrack->mState = TrackBase::PAUSING; 6235 // do not wait for mStartStopCond if exiting 6236 if (exitPending()) { 6237 return; 6238 } 6239 mStartStopCond.wait(mLock); 6240 // if we have been restarted, recordTrack == mActiveTrack.get() here 6241 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6242 mLock.unlock(); 6243 AudioSystem::stopInput(mId); 6244 mLock.lock(); 6245 ALOGV("Record stopped OK"); 6246 } 6247 } 6248 } 6249} 6250 6251bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6252{ 6253 return false; 6254} 6255 6256status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6257{ 6258 if (!isValidSyncEvent(event)) { 6259 return BAD_VALUE; 6260 } 6261 6262 Mutex::Autolock _l(mLock); 6263 6264 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6265 mTrack->setSyncEvent(event); 6266 return NO_ERROR; 6267 } 6268 return NAME_NOT_FOUND; 6269} 6270 6271status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6272{ 6273 const size_t SIZE = 256; 6274 char buffer[SIZE]; 6275 String8 result; 6276 6277 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6278 result.append(buffer); 6279 6280 if (mActiveTrack != 0) { 6281 result.append("Active Track:\n"); 6282 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6283 mActiveTrack->dump(buffer, SIZE); 6284 result.append(buffer); 6285 6286 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6287 result.append(buffer); 6288 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6289 result.append(buffer); 6290 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6291 result.append(buffer); 6292 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6293 result.append(buffer); 6294 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6295 result.append(buffer); 6296 6297 6298 } else { 6299 result.append("No record client\n"); 6300 } 6301 write(fd, result.string(), result.size()); 6302 6303 dumpBase(fd, args); 6304 dumpEffectChains(fd, args); 6305 6306 return NO_ERROR; 6307} 6308 6309// AudioBufferProvider interface 6310status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6311{ 6312 size_t framesReq = buffer->frameCount; 6313 size_t framesReady = mFrameCount - mRsmpInIndex; 6314 int channelCount; 6315 6316 if (framesReady == 0) { 6317 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6318 if (mBytesRead < 0) { 6319 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6320 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6321 // Force input into standby so that it tries to 6322 // recover at next read attempt 6323 mInput->stream->common.standby(&mInput->stream->common); 6324 usleep(kRecordThreadSleepUs); 6325 } 6326 buffer->raw = NULL; 6327 buffer->frameCount = 0; 6328 return NOT_ENOUGH_DATA; 6329 } 6330 mRsmpInIndex = 0; 6331 framesReady = mFrameCount; 6332 } 6333 6334 if (framesReq > framesReady) { 6335 framesReq = framesReady; 6336 } 6337 6338 if (mChannelCount == 1 && mReqChannelCount == 2) { 6339 channelCount = 1; 6340 } else { 6341 channelCount = 2; 6342 } 6343 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6344 buffer->frameCount = framesReq; 6345 return NO_ERROR; 6346} 6347 6348// AudioBufferProvider interface 6349void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6350{ 6351 mRsmpInIndex += buffer->frameCount; 6352 buffer->frameCount = 0; 6353} 6354 6355bool AudioFlinger::RecordThread::checkForNewParameters_l() 6356{ 6357 bool reconfig = false; 6358 6359 while (!mNewParameters.isEmpty()) { 6360 status_t status = NO_ERROR; 6361 String8 keyValuePair = mNewParameters[0]; 6362 AudioParameter param = AudioParameter(keyValuePair); 6363 int value; 6364 audio_format_t reqFormat = mFormat; 6365 int reqSamplingRate = mReqSampleRate; 6366 int reqChannelCount = mReqChannelCount; 6367 6368 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6369 reqSamplingRate = value; 6370 reconfig = true; 6371 } 6372 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6373 reqFormat = (audio_format_t) value; 6374 reconfig = true; 6375 } 6376 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6377 reqChannelCount = popcount(value); 6378 reconfig = true; 6379 } 6380 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6381 // do not accept frame count changes if tracks are open as the track buffer 6382 // size depends on frame count and correct behavior would not be guaranteed 6383 // if frame count is changed after track creation 6384 if (mActiveTrack != 0) { 6385 status = INVALID_OPERATION; 6386 } else { 6387 reconfig = true; 6388 } 6389 } 6390 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6391 // forward device change to effects that have requested to be 6392 // aware of attached audio device. 6393 for (size_t i = 0; i < mEffectChains.size(); i++) { 6394 mEffectChains[i]->setDevice_l(value); 6395 } 6396 // store input device and output device but do not forward output device to audio HAL. 6397 // Note that status is ignored by the caller for output device 6398 // (see AudioFlinger::setParameters() 6399 if (value & AUDIO_DEVICE_OUT_ALL) { 6400 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6401 status = BAD_VALUE; 6402 } else { 6403 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6404 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6405 if (mTrack != NULL) { 6406 bool suspend = audio_is_bluetooth_sco_device( 6407 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6408 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6409 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6410 } 6411 } 6412 mDevice |= (uint32_t)value; 6413 } 6414 if (status == NO_ERROR) { 6415 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6416 if (status == INVALID_OPERATION) { 6417 mInput->stream->common.standby(&mInput->stream->common); 6418 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6419 keyValuePair.string()); 6420 } 6421 if (reconfig) { 6422 if (status == BAD_VALUE && 6423 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6424 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6425 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6426 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6427 (reqChannelCount <= FCC_2)) { 6428 status = NO_ERROR; 6429 } 6430 if (status == NO_ERROR) { 6431 readInputParameters(); 6432 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6433 } 6434 } 6435 } 6436 6437 mNewParameters.removeAt(0); 6438 6439 mParamStatus = status; 6440 mParamCond.signal(); 6441 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6442 // already timed out waiting for the status and will never signal the condition. 6443 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6444 } 6445 return reconfig; 6446} 6447 6448String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6449{ 6450 char *s; 6451 String8 out_s8 = String8(); 6452 6453 Mutex::Autolock _l(mLock); 6454 if (initCheck() != NO_ERROR) { 6455 return out_s8; 6456 } 6457 6458 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6459 out_s8 = String8(s); 6460 free(s); 6461 return out_s8; 6462} 6463 6464void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6465 AudioSystem::OutputDescriptor desc; 6466 void *param2 = NULL; 6467 6468 switch (event) { 6469 case AudioSystem::INPUT_OPENED: 6470 case AudioSystem::INPUT_CONFIG_CHANGED: 6471 desc.channels = mChannelMask; 6472 desc.samplingRate = mSampleRate; 6473 desc.format = mFormat; 6474 desc.frameCount = mFrameCount; 6475 desc.latency = 0; 6476 param2 = &desc; 6477 break; 6478 6479 case AudioSystem::INPUT_CLOSED: 6480 default: 6481 break; 6482 } 6483 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6484} 6485 6486void AudioFlinger::RecordThread::readInputParameters() 6487{ 6488 delete mRsmpInBuffer; 6489 // mRsmpInBuffer is always assigned a new[] below 6490 delete mRsmpOutBuffer; 6491 mRsmpOutBuffer = NULL; 6492 delete mResampler; 6493 mResampler = NULL; 6494 6495 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6496 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6497 mChannelCount = (uint16_t)popcount(mChannelMask); 6498 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6499 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6500 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6501 mFrameCount = mInputBytes / mFrameSize; 6502 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6503 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6504 6505 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6506 { 6507 int channelCount; 6508 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6509 // stereo to mono post process as the resampler always outputs stereo. 6510 if (mChannelCount == 1 && mReqChannelCount == 2) { 6511 channelCount = 1; 6512 } else { 6513 channelCount = 2; 6514 } 6515 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6516 mResampler->setSampleRate(mSampleRate); 6517 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6518 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6519 6520 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6521 if (mChannelCount == 1 && mReqChannelCount == 1) { 6522 mFrameCount >>= 1; 6523 } 6524 6525 } 6526 mRsmpInIndex = mFrameCount; 6527} 6528 6529unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6530{ 6531 Mutex::Autolock _l(mLock); 6532 if (initCheck() != NO_ERROR) { 6533 return 0; 6534 } 6535 6536 return mInput->stream->get_input_frames_lost(mInput->stream); 6537} 6538 6539uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6540{ 6541 Mutex::Autolock _l(mLock); 6542 uint32_t result = 0; 6543 if (getEffectChain_l(sessionId) != 0) { 6544 result = EFFECT_SESSION; 6545 } 6546 6547 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6548 result |= TRACK_SESSION; 6549 } 6550 6551 return result; 6552} 6553 6554AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6555{ 6556 Mutex::Autolock _l(mLock); 6557 return mTrack; 6558} 6559 6560AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6561{ 6562 Mutex::Autolock _l(mLock); 6563 return mInput; 6564} 6565 6566AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6567{ 6568 Mutex::Autolock _l(mLock); 6569 AudioStreamIn *input = mInput; 6570 mInput = NULL; 6571 return input; 6572} 6573 6574// this method must always be called either with ThreadBase mLock held or inside the thread loop 6575audio_stream_t* AudioFlinger::RecordThread::stream() const 6576{ 6577 if (mInput == NULL) { 6578 return NULL; 6579 } 6580 return &mInput->stream->common; 6581} 6582 6583 6584// ---------------------------------------------------------------------------- 6585 6586audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6587{ 6588 if (!settingsAllowed()) { 6589 return 0; 6590 } 6591 Mutex::Autolock _l(mLock); 6592 return loadHwModule_l(name); 6593} 6594 6595// loadHwModule_l() must be called with AudioFlinger::mLock held 6596audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6597{ 6598 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6599 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6600 ALOGW("loadHwModule() module %s already loaded", name); 6601 return mAudioHwDevs.keyAt(i); 6602 } 6603 } 6604 6605 audio_hw_device_t *dev; 6606 6607 int rc = load_audio_interface(name, &dev); 6608 if (rc) { 6609 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6610 return 0; 6611 } 6612 6613 mHardwareStatus = AUDIO_HW_INIT; 6614 rc = dev->init_check(dev); 6615 mHardwareStatus = AUDIO_HW_IDLE; 6616 if (rc) { 6617 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6618 return 0; 6619 } 6620 6621 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6622 (NULL != dev->set_master_volume)) { 6623 AutoMutex lock(mHardwareLock); 6624 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6625 dev->set_master_volume(dev, mMasterVolume); 6626 mHardwareStatus = AUDIO_HW_IDLE; 6627 } 6628 6629 audio_module_handle_t handle = nextUniqueId(); 6630 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6631 6632 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6633 name, dev->common.module->name, dev->common.module->id, handle); 6634 6635 return handle; 6636 6637} 6638 6639audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6640 audio_devices_t *pDevices, 6641 uint32_t *pSamplingRate, 6642 audio_format_t *pFormat, 6643 audio_channel_mask_t *pChannelMask, 6644 uint32_t *pLatencyMs, 6645 audio_output_flags_t flags) 6646{ 6647 status_t status; 6648 PlaybackThread *thread = NULL; 6649 struct audio_config config = { 6650 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6651 channel_mask: pChannelMask ? *pChannelMask : 0, 6652 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6653 }; 6654 audio_stream_out_t *outStream = NULL; 6655 audio_hw_device_t *outHwDev; 6656 6657 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6658 module, 6659 (pDevices != NULL) ? (int)*pDevices : 0, 6660 config.sample_rate, 6661 config.format, 6662 config.channel_mask, 6663 flags); 6664 6665 if (pDevices == NULL || *pDevices == 0) { 6666 return 0; 6667 } 6668 6669 Mutex::Autolock _l(mLock); 6670 6671 outHwDev = findSuitableHwDev_l(module, *pDevices); 6672 if (outHwDev == NULL) 6673 return 0; 6674 6675 audio_io_handle_t id = nextUniqueId(); 6676 6677 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6678 6679 status = outHwDev->open_output_stream(outHwDev, 6680 id, 6681 *pDevices, 6682 (audio_output_flags_t)flags, 6683 &config, 6684 &outStream); 6685 6686 mHardwareStatus = AUDIO_HW_IDLE; 6687 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6688 outStream, 6689 config.sample_rate, 6690 config.format, 6691 config.channel_mask, 6692 status); 6693 6694 if (status == NO_ERROR && outStream != NULL) { 6695 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6696 6697 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6698 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6699 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6700 thread = new DirectOutputThread(this, output, id, *pDevices); 6701 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6702 } else { 6703 thread = new MixerThread(this, output, id, *pDevices); 6704 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6705 } 6706 mPlaybackThreads.add(id, thread); 6707 6708 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6709 if (pFormat != NULL) *pFormat = config.format; 6710 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6711 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6712 6713 // notify client processes of the new output creation 6714 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6715 6716 // the first primary output opened designates the primary hw device 6717 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6718 ALOGI("Using module %d has the primary audio interface", module); 6719 mPrimaryHardwareDev = outHwDev; 6720 6721 AutoMutex lock(mHardwareLock); 6722 mHardwareStatus = AUDIO_HW_SET_MODE; 6723 outHwDev->set_mode(outHwDev, mMode); 6724 6725 // Determine the level of master volume support the primary audio HAL has, 6726 // and set the initial master volume at the same time. 6727 float initialVolume = 1.0; 6728 mMasterVolumeSupportLvl = MVS_NONE; 6729 6730 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6731 if ((NULL != outHwDev->get_master_volume) && 6732 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6733 mMasterVolumeSupportLvl = MVS_FULL; 6734 } else { 6735 mMasterVolumeSupportLvl = MVS_SETONLY; 6736 initialVolume = 1.0; 6737 } 6738 6739 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6740 if ((NULL == outHwDev->set_master_volume) || 6741 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6742 mMasterVolumeSupportLvl = MVS_NONE; 6743 } 6744 // now that we have a primary device, initialize master volume on other devices 6745 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6746 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6747 6748 if ((dev != mPrimaryHardwareDev) && 6749 (NULL != dev->set_master_volume)) { 6750 dev->set_master_volume(dev, initialVolume); 6751 } 6752 } 6753 mHardwareStatus = AUDIO_HW_IDLE; 6754 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6755 ? initialVolume 6756 : 1.0; 6757 mMasterVolume = initialVolume; 6758 } 6759 return id; 6760 } 6761 6762 return 0; 6763} 6764 6765audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6766 audio_io_handle_t output2) 6767{ 6768 Mutex::Autolock _l(mLock); 6769 MixerThread *thread1 = checkMixerThread_l(output1); 6770 MixerThread *thread2 = checkMixerThread_l(output2); 6771 6772 if (thread1 == NULL || thread2 == NULL) { 6773 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6774 return 0; 6775 } 6776 6777 audio_io_handle_t id = nextUniqueId(); 6778 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6779 thread->addOutputTrack(thread2); 6780 mPlaybackThreads.add(id, thread); 6781 // notify client processes of the new output creation 6782 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6783 return id; 6784} 6785 6786status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6787{ 6788 // keep strong reference on the playback thread so that 6789 // it is not destroyed while exit() is executed 6790 sp<PlaybackThread> thread; 6791 { 6792 Mutex::Autolock _l(mLock); 6793 thread = checkPlaybackThread_l(output); 6794 if (thread == NULL) { 6795 return BAD_VALUE; 6796 } 6797 6798 ALOGV("closeOutput() %d", output); 6799 6800 if (thread->type() == ThreadBase::MIXER) { 6801 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6802 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6803 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6804 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6805 } 6806 } 6807 } 6808 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6809 mPlaybackThreads.removeItem(output); 6810 } 6811 thread->exit(); 6812 // The thread entity (active unit of execution) is no longer running here, 6813 // but the ThreadBase container still exists. 6814 6815 if (thread->type() != ThreadBase::DUPLICATING) { 6816 AudioStreamOut *out = thread->clearOutput(); 6817 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6818 // from now on thread->mOutput is NULL 6819 out->hwDev->close_output_stream(out->hwDev, out->stream); 6820 delete out; 6821 } 6822 return NO_ERROR; 6823} 6824 6825status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6826{ 6827 Mutex::Autolock _l(mLock); 6828 PlaybackThread *thread = checkPlaybackThread_l(output); 6829 6830 if (thread == NULL) { 6831 return BAD_VALUE; 6832 } 6833 6834 ALOGV("suspendOutput() %d", output); 6835 thread->suspend(); 6836 6837 return NO_ERROR; 6838} 6839 6840status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6841{ 6842 Mutex::Autolock _l(mLock); 6843 PlaybackThread *thread = checkPlaybackThread_l(output); 6844 6845 if (thread == NULL) { 6846 return BAD_VALUE; 6847 } 6848 6849 ALOGV("restoreOutput() %d", output); 6850 6851 thread->restore(); 6852 6853 return NO_ERROR; 6854} 6855 6856audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6857 audio_devices_t *pDevices, 6858 uint32_t *pSamplingRate, 6859 audio_format_t *pFormat, 6860 uint32_t *pChannelMask) 6861{ 6862 status_t status; 6863 RecordThread *thread = NULL; 6864 struct audio_config config = { 6865 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6866 channel_mask: pChannelMask ? *pChannelMask : 0, 6867 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6868 }; 6869 uint32_t reqSamplingRate = config.sample_rate; 6870 audio_format_t reqFormat = config.format; 6871 audio_channel_mask_t reqChannels = config.channel_mask; 6872 audio_stream_in_t *inStream = NULL; 6873 audio_hw_device_t *inHwDev; 6874 6875 if (pDevices == NULL || *pDevices == 0) { 6876 return 0; 6877 } 6878 6879 Mutex::Autolock _l(mLock); 6880 6881 inHwDev = findSuitableHwDev_l(module, *pDevices); 6882 if (inHwDev == NULL) 6883 return 0; 6884 6885 audio_io_handle_t id = nextUniqueId(); 6886 6887 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6888 &inStream); 6889 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6890 inStream, 6891 config.sample_rate, 6892 config.format, 6893 config.channel_mask, 6894 status); 6895 6896 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6897 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6898 // or stereo to mono conversions on 16 bit PCM inputs. 6899 if (status == BAD_VALUE && 6900 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6901 (config.sample_rate <= 2 * reqSamplingRate) && 6902 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6903 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6904 inStream = NULL; 6905 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6906 } 6907 6908 if (status == NO_ERROR && inStream != NULL) { 6909 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6910 6911 // Start record thread 6912 // RecorThread require both input and output device indication to forward to audio 6913 // pre processing modules 6914 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6915 thread = new RecordThread(this, 6916 input, 6917 reqSamplingRate, 6918 reqChannels, 6919 id, 6920 device); 6921 mRecordThreads.add(id, thread); 6922 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6923 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6924 if (pFormat != NULL) *pFormat = config.format; 6925 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6926 6927 input->stream->common.standby(&input->stream->common); 6928 6929 // notify client processes of the new input creation 6930 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6931 return id; 6932 } 6933 6934 return 0; 6935} 6936 6937status_t AudioFlinger::closeInput(audio_io_handle_t input) 6938{ 6939 // keep strong reference on the record thread so that 6940 // it is not destroyed while exit() is executed 6941 sp<RecordThread> thread; 6942 { 6943 Mutex::Autolock _l(mLock); 6944 thread = checkRecordThread_l(input); 6945 if (thread == NULL) { 6946 return BAD_VALUE; 6947 } 6948 6949 ALOGV("closeInput() %d", input); 6950 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6951 mRecordThreads.removeItem(input); 6952 } 6953 thread->exit(); 6954 // The thread entity (active unit of execution) is no longer running here, 6955 // but the ThreadBase container still exists. 6956 6957 AudioStreamIn *in = thread->clearInput(); 6958 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6959 // from now on thread->mInput is NULL 6960 in->hwDev->close_input_stream(in->hwDev, in->stream); 6961 delete in; 6962 6963 return NO_ERROR; 6964} 6965 6966status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6967{ 6968 Mutex::Autolock _l(mLock); 6969 MixerThread *dstThread = checkMixerThread_l(output); 6970 if (dstThread == NULL) { 6971 ALOGW("setStreamOutput() bad output id %d", output); 6972 return BAD_VALUE; 6973 } 6974 6975 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6976 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6977 6978 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6979 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6980 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6981 MixerThread *srcThread = (MixerThread *)thread; 6982 srcThread->invalidateTracks(stream); 6983 } 6984 } 6985 6986 return NO_ERROR; 6987} 6988 6989 6990int AudioFlinger::newAudioSessionId() 6991{ 6992 return nextUniqueId(); 6993} 6994 6995void AudioFlinger::acquireAudioSessionId(int audioSession) 6996{ 6997 Mutex::Autolock _l(mLock); 6998 pid_t caller = IPCThreadState::self()->getCallingPid(); 6999 ALOGV("acquiring %d from %d", audioSession, caller); 7000 size_t num = mAudioSessionRefs.size(); 7001 for (size_t i = 0; i< num; i++) { 7002 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7003 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7004 ref->mCnt++; 7005 ALOGV(" incremented refcount to %d", ref->mCnt); 7006 return; 7007 } 7008 } 7009 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7010 ALOGV(" added new entry for %d", audioSession); 7011} 7012 7013void AudioFlinger::releaseAudioSessionId(int audioSession) 7014{ 7015 Mutex::Autolock _l(mLock); 7016 pid_t caller = IPCThreadState::self()->getCallingPid(); 7017 ALOGV("releasing %d from %d", audioSession, caller); 7018 size_t num = mAudioSessionRefs.size(); 7019 for (size_t i = 0; i< num; i++) { 7020 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7021 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7022 ref->mCnt--; 7023 ALOGV(" decremented refcount to %d", ref->mCnt); 7024 if (ref->mCnt == 0) { 7025 mAudioSessionRefs.removeAt(i); 7026 delete ref; 7027 purgeStaleEffects_l(); 7028 } 7029 return; 7030 } 7031 } 7032 ALOGW("session id %d not found for pid %d", audioSession, caller); 7033} 7034 7035void AudioFlinger::purgeStaleEffects_l() { 7036 7037 ALOGV("purging stale effects"); 7038 7039 Vector< sp<EffectChain> > chains; 7040 7041 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7042 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7043 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7044 sp<EffectChain> ec = t->mEffectChains[j]; 7045 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7046 chains.push(ec); 7047 } 7048 } 7049 } 7050 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7051 sp<RecordThread> t = mRecordThreads.valueAt(i); 7052 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7053 sp<EffectChain> ec = t->mEffectChains[j]; 7054 chains.push(ec); 7055 } 7056 } 7057 7058 for (size_t i = 0; i < chains.size(); i++) { 7059 sp<EffectChain> ec = chains[i]; 7060 int sessionid = ec->sessionId(); 7061 sp<ThreadBase> t = ec->mThread.promote(); 7062 if (t == 0) { 7063 continue; 7064 } 7065 size_t numsessionrefs = mAudioSessionRefs.size(); 7066 bool found = false; 7067 for (size_t k = 0; k < numsessionrefs; k++) { 7068 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7069 if (ref->mSessionid == sessionid) { 7070 ALOGV(" session %d still exists for %d with %d refs", 7071 sessionid, ref->mPid, ref->mCnt); 7072 found = true; 7073 break; 7074 } 7075 } 7076 if (!found) { 7077 // remove all effects from the chain 7078 while (ec->mEffects.size()) { 7079 sp<EffectModule> effect = ec->mEffects[0]; 7080 effect->unPin(); 7081 Mutex::Autolock _l (t->mLock); 7082 t->removeEffect_l(effect); 7083 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7084 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7085 if (handle != 0) { 7086 handle->mEffect.clear(); 7087 if (handle->mHasControl && handle->mEnabled) { 7088 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7089 } 7090 } 7091 } 7092 AudioSystem::unregisterEffect(effect->id()); 7093 } 7094 } 7095 } 7096 return; 7097} 7098 7099// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7100AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7101{ 7102 return mPlaybackThreads.valueFor(output).get(); 7103} 7104 7105// checkMixerThread_l() must be called with AudioFlinger::mLock held 7106AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7107{ 7108 PlaybackThread *thread = checkPlaybackThread_l(output); 7109 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7110} 7111 7112// checkRecordThread_l() must be called with AudioFlinger::mLock held 7113AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7114{ 7115 return mRecordThreads.valueFor(input).get(); 7116} 7117 7118uint32_t AudioFlinger::nextUniqueId() 7119{ 7120 return android_atomic_inc(&mNextUniqueId); 7121} 7122 7123AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7124{ 7125 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7126 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7127 AudioStreamOut *output = thread->getOutput(); 7128 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7129 return thread; 7130 } 7131 } 7132 return NULL; 7133} 7134 7135uint32_t AudioFlinger::primaryOutputDevice_l() const 7136{ 7137 PlaybackThread *thread = primaryPlaybackThread_l(); 7138 7139 if (thread == NULL) { 7140 return 0; 7141 } 7142 7143 return thread->device(); 7144} 7145 7146sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7147 int triggerSession, 7148 int listenerSession, 7149 sync_event_callback_t callBack, 7150 void *cookie) 7151{ 7152 Mutex::Autolock _l(mLock); 7153 7154 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7155 status_t playStatus = NAME_NOT_FOUND; 7156 status_t recStatus = NAME_NOT_FOUND; 7157 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7158 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7159 if (playStatus == NO_ERROR) { 7160 return event; 7161 } 7162 } 7163 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7164 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7165 if (recStatus == NO_ERROR) { 7166 return event; 7167 } 7168 } 7169 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7170 mPendingSyncEvents.add(event); 7171 } else { 7172 ALOGV("createSyncEvent() invalid event %d", event->type()); 7173 event.clear(); 7174 } 7175 return event; 7176} 7177 7178// ---------------------------------------------------------------------------- 7179// Effect management 7180// ---------------------------------------------------------------------------- 7181 7182 7183status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7184{ 7185 Mutex::Autolock _l(mLock); 7186 return EffectQueryNumberEffects(numEffects); 7187} 7188 7189status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7190{ 7191 Mutex::Autolock _l(mLock); 7192 return EffectQueryEffect(index, descriptor); 7193} 7194 7195status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7196 effect_descriptor_t *descriptor) const 7197{ 7198 Mutex::Autolock _l(mLock); 7199 return EffectGetDescriptor(pUuid, descriptor); 7200} 7201 7202 7203sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7204 effect_descriptor_t *pDesc, 7205 const sp<IEffectClient>& effectClient, 7206 int32_t priority, 7207 audio_io_handle_t io, 7208 int sessionId, 7209 status_t *status, 7210 int *id, 7211 int *enabled) 7212{ 7213 status_t lStatus = NO_ERROR; 7214 sp<EffectHandle> handle; 7215 effect_descriptor_t desc; 7216 7217 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7218 pid, effectClient.get(), priority, sessionId, io); 7219 7220 if (pDesc == NULL) { 7221 lStatus = BAD_VALUE; 7222 goto Exit; 7223 } 7224 7225 // check audio settings permission for global effects 7226 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7227 lStatus = PERMISSION_DENIED; 7228 goto Exit; 7229 } 7230 7231 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7232 // that can only be created by audio policy manager (running in same process) 7233 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7234 lStatus = PERMISSION_DENIED; 7235 goto Exit; 7236 } 7237 7238 if (io == 0) { 7239 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7240 // output must be specified by AudioPolicyManager when using session 7241 // AUDIO_SESSION_OUTPUT_STAGE 7242 lStatus = BAD_VALUE; 7243 goto Exit; 7244 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7245 // if the output returned by getOutputForEffect() is removed before we lock the 7246 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7247 // and we will exit safely 7248 io = AudioSystem::getOutputForEffect(&desc); 7249 } 7250 } 7251 7252 { 7253 Mutex::Autolock _l(mLock); 7254 7255 7256 if (!EffectIsNullUuid(&pDesc->uuid)) { 7257 // if uuid is specified, request effect descriptor 7258 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7259 if (lStatus < 0) { 7260 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7261 goto Exit; 7262 } 7263 } else { 7264 // if uuid is not specified, look for an available implementation 7265 // of the required type in effect factory 7266 if (EffectIsNullUuid(&pDesc->type)) { 7267 ALOGW("createEffect() no effect type"); 7268 lStatus = BAD_VALUE; 7269 goto Exit; 7270 } 7271 uint32_t numEffects = 0; 7272 effect_descriptor_t d; 7273 d.flags = 0; // prevent compiler warning 7274 bool found = false; 7275 7276 lStatus = EffectQueryNumberEffects(&numEffects); 7277 if (lStatus < 0) { 7278 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7279 goto Exit; 7280 } 7281 for (uint32_t i = 0; i < numEffects; i++) { 7282 lStatus = EffectQueryEffect(i, &desc); 7283 if (lStatus < 0) { 7284 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7285 continue; 7286 } 7287 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7288 // If matching type found save effect descriptor. If the session is 7289 // 0 and the effect is not auxiliary, continue enumeration in case 7290 // an auxiliary version of this effect type is available 7291 found = true; 7292 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7293 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7294 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7295 break; 7296 } 7297 } 7298 } 7299 if (!found) { 7300 lStatus = BAD_VALUE; 7301 ALOGW("createEffect() effect not found"); 7302 goto Exit; 7303 } 7304 // For same effect type, chose auxiliary version over insert version if 7305 // connect to output mix (Compliance to OpenSL ES) 7306 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7307 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7308 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7309 } 7310 } 7311 7312 // Do not allow auxiliary effects on a session different from 0 (output mix) 7313 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7314 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7315 lStatus = INVALID_OPERATION; 7316 goto Exit; 7317 } 7318 7319 // check recording permission for visualizer 7320 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7321 !recordingAllowed()) { 7322 lStatus = PERMISSION_DENIED; 7323 goto Exit; 7324 } 7325 7326 // return effect descriptor 7327 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7328 7329 // If output is not specified try to find a matching audio session ID in one of the 7330 // output threads. 7331 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7332 // because of code checking output when entering the function. 7333 // Note: io is never 0 when creating an effect on an input 7334 if (io == 0) { 7335 // look for the thread where the specified audio session is present 7336 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7337 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7338 io = mPlaybackThreads.keyAt(i); 7339 break; 7340 } 7341 } 7342 if (io == 0) { 7343 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7344 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7345 io = mRecordThreads.keyAt(i); 7346 break; 7347 } 7348 } 7349 } 7350 // If no output thread contains the requested session ID, default to 7351 // first output. The effect chain will be moved to the correct output 7352 // thread when a track with the same session ID is created 7353 if (io == 0 && mPlaybackThreads.size()) { 7354 io = mPlaybackThreads.keyAt(0); 7355 } 7356 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7357 } 7358 ThreadBase *thread = checkRecordThread_l(io); 7359 if (thread == NULL) { 7360 thread = checkPlaybackThread_l(io); 7361 if (thread == NULL) { 7362 ALOGE("createEffect() unknown output thread"); 7363 lStatus = BAD_VALUE; 7364 goto Exit; 7365 } 7366 } 7367 7368 sp<Client> client = registerPid_l(pid); 7369 7370 // create effect on selected output thread 7371 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7372 &desc, enabled, &lStatus); 7373 if (handle != 0 && id != NULL) { 7374 *id = handle->id(); 7375 } 7376 } 7377 7378Exit: 7379 if (status != NULL) { 7380 *status = lStatus; 7381 } 7382 return handle; 7383} 7384 7385status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7386 audio_io_handle_t dstOutput) 7387{ 7388 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7389 sessionId, srcOutput, dstOutput); 7390 Mutex::Autolock _l(mLock); 7391 if (srcOutput == dstOutput) { 7392 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7393 return NO_ERROR; 7394 } 7395 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7396 if (srcThread == NULL) { 7397 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7398 return BAD_VALUE; 7399 } 7400 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7401 if (dstThread == NULL) { 7402 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7403 return BAD_VALUE; 7404 } 7405 7406 Mutex::Autolock _dl(dstThread->mLock); 7407 Mutex::Autolock _sl(srcThread->mLock); 7408 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7409 7410 return NO_ERROR; 7411} 7412 7413// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7414status_t AudioFlinger::moveEffectChain_l(int sessionId, 7415 AudioFlinger::PlaybackThread *srcThread, 7416 AudioFlinger::PlaybackThread *dstThread, 7417 bool reRegister) 7418{ 7419 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7420 sessionId, srcThread, dstThread); 7421 7422 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7423 if (chain == 0) { 7424 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7425 sessionId, srcThread); 7426 return INVALID_OPERATION; 7427 } 7428 7429 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7430 // so that a new chain is created with correct parameters when first effect is added. This is 7431 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7432 // removed. 7433 srcThread->removeEffectChain_l(chain); 7434 7435 // transfer all effects one by one so that new effect chain is created on new thread with 7436 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7437 audio_io_handle_t dstOutput = dstThread->id(); 7438 sp<EffectChain> dstChain; 7439 uint32_t strategy = 0; // prevent compiler warning 7440 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7441 while (effect != 0) { 7442 srcThread->removeEffect_l(effect); 7443 dstThread->addEffect_l(effect); 7444 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7445 if (effect->state() == EffectModule::ACTIVE || 7446 effect->state() == EffectModule::STOPPING) { 7447 effect->start(); 7448 } 7449 // if the move request is not received from audio policy manager, the effect must be 7450 // re-registered with the new strategy and output 7451 if (dstChain == 0) { 7452 dstChain = effect->chain().promote(); 7453 if (dstChain == 0) { 7454 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7455 srcThread->addEffect_l(effect); 7456 return NO_INIT; 7457 } 7458 strategy = dstChain->strategy(); 7459 } 7460 if (reRegister) { 7461 AudioSystem::unregisterEffect(effect->id()); 7462 AudioSystem::registerEffect(&effect->desc(), 7463 dstOutput, 7464 strategy, 7465 sessionId, 7466 effect->id()); 7467 } 7468 effect = chain->getEffectFromId_l(0); 7469 } 7470 7471 return NO_ERROR; 7472} 7473 7474 7475// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7476sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7477 const sp<AudioFlinger::Client>& client, 7478 const sp<IEffectClient>& effectClient, 7479 int32_t priority, 7480 int sessionId, 7481 effect_descriptor_t *desc, 7482 int *enabled, 7483 status_t *status 7484 ) 7485{ 7486 sp<EffectModule> effect; 7487 sp<EffectHandle> handle; 7488 status_t lStatus; 7489 sp<EffectChain> chain; 7490 bool chainCreated = false; 7491 bool effectCreated = false; 7492 bool effectRegistered = false; 7493 7494 lStatus = initCheck(); 7495 if (lStatus != NO_ERROR) { 7496 ALOGW("createEffect_l() Audio driver not initialized."); 7497 goto Exit; 7498 } 7499 7500 // Do not allow effects with session ID 0 on direct output or duplicating threads 7501 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7502 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7503 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7504 desc->name, sessionId); 7505 lStatus = BAD_VALUE; 7506 goto Exit; 7507 } 7508 // Only Pre processor effects are allowed on input threads and only on input threads 7509 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7510 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7511 desc->name, desc->flags, mType); 7512 lStatus = BAD_VALUE; 7513 goto Exit; 7514 } 7515 7516 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7517 7518 { // scope for mLock 7519 Mutex::Autolock _l(mLock); 7520 7521 // check for existing effect chain with the requested audio session 7522 chain = getEffectChain_l(sessionId); 7523 if (chain == 0) { 7524 // create a new chain for this session 7525 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7526 chain = new EffectChain(this, sessionId); 7527 addEffectChain_l(chain); 7528 chain->setStrategy(getStrategyForSession_l(sessionId)); 7529 chainCreated = true; 7530 } else { 7531 effect = chain->getEffectFromDesc_l(desc); 7532 } 7533 7534 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7535 7536 if (effect == 0) { 7537 int id = mAudioFlinger->nextUniqueId(); 7538 // Check CPU and memory usage 7539 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7540 if (lStatus != NO_ERROR) { 7541 goto Exit; 7542 } 7543 effectRegistered = true; 7544 // create a new effect module if none present in the chain 7545 effect = new EffectModule(this, chain, desc, id, sessionId); 7546 lStatus = effect->status(); 7547 if (lStatus != NO_ERROR) { 7548 goto Exit; 7549 } 7550 lStatus = chain->addEffect_l(effect); 7551 if (lStatus != NO_ERROR) { 7552 goto Exit; 7553 } 7554 effectCreated = true; 7555 7556 effect->setDevice(mDevice); 7557 effect->setMode(mAudioFlinger->getMode()); 7558 } 7559 // create effect handle and connect it to effect module 7560 handle = new EffectHandle(effect, client, effectClient, priority); 7561 lStatus = effect->addHandle(handle); 7562 if (enabled != NULL) { 7563 *enabled = (int)effect->isEnabled(); 7564 } 7565 } 7566 7567Exit: 7568 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7569 Mutex::Autolock _l(mLock); 7570 if (effectCreated) { 7571 chain->removeEffect_l(effect); 7572 } 7573 if (effectRegistered) { 7574 AudioSystem::unregisterEffect(effect->id()); 7575 } 7576 if (chainCreated) { 7577 removeEffectChain_l(chain); 7578 } 7579 handle.clear(); 7580 } 7581 7582 if (status != NULL) { 7583 *status = lStatus; 7584 } 7585 return handle; 7586} 7587 7588sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7589{ 7590 sp<EffectChain> chain = getEffectChain_l(sessionId); 7591 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7592} 7593 7594// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7595// PlaybackThread::mLock held 7596status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7597{ 7598 // check for existing effect chain with the requested audio session 7599 int sessionId = effect->sessionId(); 7600 sp<EffectChain> chain = getEffectChain_l(sessionId); 7601 bool chainCreated = false; 7602 7603 if (chain == 0) { 7604 // create a new chain for this session 7605 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7606 chain = new EffectChain(this, sessionId); 7607 addEffectChain_l(chain); 7608 chain->setStrategy(getStrategyForSession_l(sessionId)); 7609 chainCreated = true; 7610 } 7611 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7612 7613 if (chain->getEffectFromId_l(effect->id()) != 0) { 7614 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7615 this, effect->desc().name, chain.get()); 7616 return BAD_VALUE; 7617 } 7618 7619 status_t status = chain->addEffect_l(effect); 7620 if (status != NO_ERROR) { 7621 if (chainCreated) { 7622 removeEffectChain_l(chain); 7623 } 7624 return status; 7625 } 7626 7627 effect->setDevice(mDevice); 7628 effect->setMode(mAudioFlinger->getMode()); 7629 return NO_ERROR; 7630} 7631 7632void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7633 7634 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7635 effect_descriptor_t desc = effect->desc(); 7636 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7637 detachAuxEffect_l(effect->id()); 7638 } 7639 7640 sp<EffectChain> chain = effect->chain().promote(); 7641 if (chain != 0) { 7642 // remove effect chain if removing last effect 7643 if (chain->removeEffect_l(effect) == 0) { 7644 removeEffectChain_l(chain); 7645 } 7646 } else { 7647 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7648 } 7649} 7650 7651void AudioFlinger::ThreadBase::lockEffectChains_l( 7652 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7653{ 7654 effectChains = mEffectChains; 7655 for (size_t i = 0; i < mEffectChains.size(); i++) { 7656 mEffectChains[i]->lock(); 7657 } 7658} 7659 7660void AudioFlinger::ThreadBase::unlockEffectChains( 7661 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7662{ 7663 for (size_t i = 0; i < effectChains.size(); i++) { 7664 effectChains[i]->unlock(); 7665 } 7666} 7667 7668sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7669{ 7670 Mutex::Autolock _l(mLock); 7671 return getEffectChain_l(sessionId); 7672} 7673 7674sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7675{ 7676 size_t size = mEffectChains.size(); 7677 for (size_t i = 0; i < size; i++) { 7678 if (mEffectChains[i]->sessionId() == sessionId) { 7679 return mEffectChains[i]; 7680 } 7681 } 7682 return 0; 7683} 7684 7685void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7686{ 7687 Mutex::Autolock _l(mLock); 7688 size_t size = mEffectChains.size(); 7689 for (size_t i = 0; i < size; i++) { 7690 mEffectChains[i]->setMode_l(mode); 7691 } 7692} 7693 7694void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7695 const wp<EffectHandle>& handle, 7696 bool unpinIfLast) { 7697 7698 Mutex::Autolock _l(mLock); 7699 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7700 // delete the effect module if removing last handle on it 7701 if (effect->removeHandle(handle) == 0) { 7702 if (!effect->isPinned() || unpinIfLast) { 7703 removeEffect_l(effect); 7704 AudioSystem::unregisterEffect(effect->id()); 7705 } 7706 } 7707} 7708 7709status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7710{ 7711 int session = chain->sessionId(); 7712 int16_t *buffer = mMixBuffer; 7713 bool ownsBuffer = false; 7714 7715 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7716 if (session > 0) { 7717 // Only one effect chain can be present in direct output thread and it uses 7718 // the mix buffer as input 7719 if (mType != DIRECT) { 7720 size_t numSamples = mNormalFrameCount * mChannelCount; 7721 buffer = new int16_t[numSamples]; 7722 memset(buffer, 0, numSamples * sizeof(int16_t)); 7723 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7724 ownsBuffer = true; 7725 } 7726 7727 // Attach all tracks with same session ID to this chain. 7728 for (size_t i = 0; i < mTracks.size(); ++i) { 7729 sp<Track> track = mTracks[i]; 7730 if (session == track->sessionId()) { 7731 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7732 track->setMainBuffer(buffer); 7733 chain->incTrackCnt(); 7734 } 7735 } 7736 7737 // indicate all active tracks in the chain 7738 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7739 sp<Track> track = mActiveTracks[i].promote(); 7740 if (track == 0) continue; 7741 if (session == track->sessionId()) { 7742 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7743 chain->incActiveTrackCnt(); 7744 } 7745 } 7746 } 7747 7748 chain->setInBuffer(buffer, ownsBuffer); 7749 chain->setOutBuffer(mMixBuffer); 7750 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7751 // chains list in order to be processed last as it contains output stage effects 7752 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7753 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7754 // after track specific effects and before output stage 7755 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7756 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7757 // Effect chain for other sessions are inserted at beginning of effect 7758 // chains list to be processed before output mix effects. Relative order between other 7759 // sessions is not important 7760 size_t size = mEffectChains.size(); 7761 size_t i = 0; 7762 for (i = 0; i < size; i++) { 7763 if (mEffectChains[i]->sessionId() < session) break; 7764 } 7765 mEffectChains.insertAt(chain, i); 7766 checkSuspendOnAddEffectChain_l(chain); 7767 7768 return NO_ERROR; 7769} 7770 7771size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7772{ 7773 int session = chain->sessionId(); 7774 7775 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7776 7777 for (size_t i = 0; i < mEffectChains.size(); i++) { 7778 if (chain == mEffectChains[i]) { 7779 mEffectChains.removeAt(i); 7780 // detach all active tracks from the chain 7781 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7782 sp<Track> track = mActiveTracks[i].promote(); 7783 if (track == 0) continue; 7784 if (session == track->sessionId()) { 7785 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7786 chain.get(), session); 7787 chain->decActiveTrackCnt(); 7788 } 7789 } 7790 7791 // detach all tracks with same session ID from this chain 7792 for (size_t i = 0; i < mTracks.size(); ++i) { 7793 sp<Track> track = mTracks[i]; 7794 if (session == track->sessionId()) { 7795 track->setMainBuffer(mMixBuffer); 7796 chain->decTrackCnt(); 7797 } 7798 } 7799 break; 7800 } 7801 } 7802 return mEffectChains.size(); 7803} 7804 7805status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7806 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7807{ 7808 Mutex::Autolock _l(mLock); 7809 return attachAuxEffect_l(track, EffectId); 7810} 7811 7812status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7813 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7814{ 7815 status_t status = NO_ERROR; 7816 7817 if (EffectId == 0) { 7818 track->setAuxBuffer(0, NULL); 7819 } else { 7820 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7821 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7822 if (effect != 0) { 7823 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7824 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7825 } else { 7826 status = INVALID_OPERATION; 7827 } 7828 } else { 7829 status = BAD_VALUE; 7830 } 7831 } 7832 return status; 7833} 7834 7835void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7836{ 7837 for (size_t i = 0; i < mTracks.size(); ++i) { 7838 sp<Track> track = mTracks[i]; 7839 if (track->auxEffectId() == effectId) { 7840 attachAuxEffect_l(track, 0); 7841 } 7842 } 7843} 7844 7845status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7846{ 7847 // only one chain per input thread 7848 if (mEffectChains.size() != 0) { 7849 return INVALID_OPERATION; 7850 } 7851 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7852 7853 chain->setInBuffer(NULL); 7854 chain->setOutBuffer(NULL); 7855 7856 checkSuspendOnAddEffectChain_l(chain); 7857 7858 mEffectChains.add(chain); 7859 7860 return NO_ERROR; 7861} 7862 7863size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7864{ 7865 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7866 ALOGW_IF(mEffectChains.size() != 1, 7867 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7868 chain.get(), mEffectChains.size(), this); 7869 if (mEffectChains.size() == 1) { 7870 mEffectChains.removeAt(0); 7871 } 7872 return 0; 7873} 7874 7875// ---------------------------------------------------------------------------- 7876// EffectModule implementation 7877// ---------------------------------------------------------------------------- 7878 7879#undef LOG_TAG 7880#define LOG_TAG "AudioFlinger::EffectModule" 7881 7882AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7883 const wp<AudioFlinger::EffectChain>& chain, 7884 effect_descriptor_t *desc, 7885 int id, 7886 int sessionId) 7887 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7888 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7889{ 7890 ALOGV("Constructor %p", this); 7891 int lStatus; 7892 if (thread == NULL) { 7893 return; 7894 } 7895 7896 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7897 7898 // create effect engine from effect factory 7899 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7900 7901 if (mStatus != NO_ERROR) { 7902 return; 7903 } 7904 lStatus = init(); 7905 if (lStatus < 0) { 7906 mStatus = lStatus; 7907 goto Error; 7908 } 7909 7910 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7911 mPinned = true; 7912 } 7913 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7914 return; 7915Error: 7916 EffectRelease(mEffectInterface); 7917 mEffectInterface = NULL; 7918 ALOGV("Constructor Error %d", mStatus); 7919} 7920 7921AudioFlinger::EffectModule::~EffectModule() 7922{ 7923 ALOGV("Destructor %p", this); 7924 if (mEffectInterface != NULL) { 7925 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7926 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7927 sp<ThreadBase> thread = mThread.promote(); 7928 if (thread != 0) { 7929 audio_stream_t *stream = thread->stream(); 7930 if (stream != NULL) { 7931 stream->remove_audio_effect(stream, mEffectInterface); 7932 } 7933 } 7934 } 7935 // release effect engine 7936 EffectRelease(mEffectInterface); 7937 } 7938} 7939 7940status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7941{ 7942 status_t status; 7943 7944 Mutex::Autolock _l(mLock); 7945 int priority = handle->priority(); 7946 size_t size = mHandles.size(); 7947 sp<EffectHandle> h; 7948 size_t i; 7949 for (i = 0; i < size; i++) { 7950 h = mHandles[i].promote(); 7951 if (h == 0) continue; 7952 if (h->priority() <= priority) break; 7953 } 7954 // if inserted in first place, move effect control from previous owner to this handle 7955 if (i == 0) { 7956 bool enabled = false; 7957 if (h != 0) { 7958 enabled = h->enabled(); 7959 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7960 } 7961 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7962 status = NO_ERROR; 7963 } else { 7964 status = ALREADY_EXISTS; 7965 } 7966 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7967 mHandles.insertAt(handle, i); 7968 return status; 7969} 7970 7971size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7972{ 7973 Mutex::Autolock _l(mLock); 7974 size_t size = mHandles.size(); 7975 size_t i; 7976 for (i = 0; i < size; i++) { 7977 if (mHandles[i] == handle) break; 7978 } 7979 if (i == size) { 7980 return size; 7981 } 7982 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7983 7984 bool enabled = false; 7985 EffectHandle *hdl = handle.unsafe_get(); 7986 if (hdl != NULL) { 7987 ALOGV("removeHandle() unsafe_get OK"); 7988 enabled = hdl->enabled(); 7989 } 7990 mHandles.removeAt(i); 7991 size = mHandles.size(); 7992 // if removed from first place, move effect control from this handle to next in line 7993 if (i == 0 && size != 0) { 7994 sp<EffectHandle> h = mHandles[0].promote(); 7995 if (h != 0) { 7996 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7997 } 7998 } 7999 8000 // Prevent calls to process() and other functions on effect interface from now on. 8001 // The effect engine will be released by the destructor when the last strong reference on 8002 // this object is released which can happen after next process is called. 8003 if (size == 0 && !mPinned) { 8004 mState = DESTROYED; 8005 } 8006 8007 return size; 8008} 8009 8010sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8011{ 8012 Mutex::Autolock _l(mLock); 8013 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8014} 8015 8016void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8017{ 8018 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8019 // keep a strong reference on this EffectModule to avoid calling the 8020 // destructor before we exit 8021 sp<EffectModule> keep(this); 8022 { 8023 sp<ThreadBase> thread = mThread.promote(); 8024 if (thread != 0) { 8025 thread->disconnectEffect(keep, handle, unpinIfLast); 8026 } 8027 } 8028} 8029 8030void AudioFlinger::EffectModule::updateState() { 8031 Mutex::Autolock _l(mLock); 8032 8033 switch (mState) { 8034 case RESTART: 8035 reset_l(); 8036 // FALL THROUGH 8037 8038 case STARTING: 8039 // clear auxiliary effect input buffer for next accumulation 8040 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8041 memset(mConfig.inputCfg.buffer.raw, 8042 0, 8043 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8044 } 8045 start_l(); 8046 mState = ACTIVE; 8047 break; 8048 case STOPPING: 8049 stop_l(); 8050 mDisableWaitCnt = mMaxDisableWaitCnt; 8051 mState = STOPPED; 8052 break; 8053 case STOPPED: 8054 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8055 // turn off sequence. 8056 if (--mDisableWaitCnt == 0) { 8057 reset_l(); 8058 mState = IDLE; 8059 } 8060 break; 8061 default: //IDLE , ACTIVE, DESTROYED 8062 break; 8063 } 8064} 8065 8066void AudioFlinger::EffectModule::process() 8067{ 8068 Mutex::Autolock _l(mLock); 8069 8070 if (mState == DESTROYED || mEffectInterface == NULL || 8071 mConfig.inputCfg.buffer.raw == NULL || 8072 mConfig.outputCfg.buffer.raw == NULL) { 8073 return; 8074 } 8075 8076 if (isProcessEnabled()) { 8077 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8078 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8079 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8080 mConfig.inputCfg.buffer.s32, 8081 mConfig.inputCfg.buffer.frameCount/2); 8082 } 8083 8084 // do the actual processing in the effect engine 8085 int ret = (*mEffectInterface)->process(mEffectInterface, 8086 &mConfig.inputCfg.buffer, 8087 &mConfig.outputCfg.buffer); 8088 8089 // force transition to IDLE state when engine is ready 8090 if (mState == STOPPED && ret == -ENODATA) { 8091 mDisableWaitCnt = 1; 8092 } 8093 8094 // clear auxiliary effect input buffer for next accumulation 8095 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8096 memset(mConfig.inputCfg.buffer.raw, 0, 8097 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8098 } 8099 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8100 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8101 // If an insert effect is idle and input buffer is different from output buffer, 8102 // accumulate input onto output 8103 sp<EffectChain> chain = mChain.promote(); 8104 if (chain != 0 && chain->activeTrackCnt() != 0) { 8105 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8106 int16_t *in = mConfig.inputCfg.buffer.s16; 8107 int16_t *out = mConfig.outputCfg.buffer.s16; 8108 for (size_t i = 0; i < frameCnt; i++) { 8109 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8110 } 8111 } 8112 } 8113} 8114 8115void AudioFlinger::EffectModule::reset_l() 8116{ 8117 if (mEffectInterface == NULL) { 8118 return; 8119 } 8120 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8121} 8122 8123status_t AudioFlinger::EffectModule::configure() 8124{ 8125 uint32_t channels; 8126 if (mEffectInterface == NULL) { 8127 return NO_INIT; 8128 } 8129 8130 sp<ThreadBase> thread = mThread.promote(); 8131 if (thread == 0) { 8132 return DEAD_OBJECT; 8133 } 8134 8135 // TODO: handle configuration of effects replacing track process 8136 if (thread->channelCount() == 1) { 8137 channels = AUDIO_CHANNEL_OUT_MONO; 8138 } else { 8139 channels = AUDIO_CHANNEL_OUT_STEREO; 8140 } 8141 8142 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8143 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8144 } else { 8145 mConfig.inputCfg.channels = channels; 8146 } 8147 mConfig.outputCfg.channels = channels; 8148 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8149 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8150 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8151 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8152 mConfig.inputCfg.bufferProvider.cookie = NULL; 8153 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8154 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8155 mConfig.outputCfg.bufferProvider.cookie = NULL; 8156 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8157 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8158 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8159 // Insert effect: 8160 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8161 // always overwrites output buffer: input buffer == output buffer 8162 // - in other sessions: 8163 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8164 // other effect: overwrites output buffer: input buffer == output buffer 8165 // Auxiliary effect: 8166 // accumulates in output buffer: input buffer != output buffer 8167 // Therefore: accumulate <=> input buffer != output buffer 8168 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8169 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8170 } else { 8171 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8172 } 8173 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8174 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8175 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8176 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8177 8178 ALOGV("configure() %p thread %p buffer %p framecount %d", 8179 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8180 8181 status_t cmdStatus; 8182 uint32_t size = sizeof(int); 8183 status_t status = (*mEffectInterface)->command(mEffectInterface, 8184 EFFECT_CMD_SET_CONFIG, 8185 sizeof(effect_config_t), 8186 &mConfig, 8187 &size, 8188 &cmdStatus); 8189 if (status == 0) { 8190 status = cmdStatus; 8191 } 8192 8193 if (status == 0 && 8194 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8195 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8196 effect_param_t *p = (effect_param_t *)buf32; 8197 8198 p->psize = sizeof(uint32_t); 8199 p->vsize = sizeof(uint32_t); 8200 size = sizeof(int); 8201 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8202 8203 uint32_t latency = 0; 8204 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8205 if (pbt != NULL) { 8206 latency = pbt->latency_l(); 8207 } 8208 8209 *((int32_t *)p->data + 1)= latency; 8210 (*mEffectInterface)->command(mEffectInterface, 8211 EFFECT_CMD_SET_PARAM, 8212 sizeof(effect_param_t) + 8, 8213 &buf32, 8214 &size, 8215 &cmdStatus); 8216 } 8217 8218 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8219 (1000 * mConfig.outputCfg.buffer.frameCount); 8220 8221 return status; 8222} 8223 8224status_t AudioFlinger::EffectModule::init() 8225{ 8226 Mutex::Autolock _l(mLock); 8227 if (mEffectInterface == NULL) { 8228 return NO_INIT; 8229 } 8230 status_t cmdStatus; 8231 uint32_t size = sizeof(status_t); 8232 status_t status = (*mEffectInterface)->command(mEffectInterface, 8233 EFFECT_CMD_INIT, 8234 0, 8235 NULL, 8236 &size, 8237 &cmdStatus); 8238 if (status == 0) { 8239 status = cmdStatus; 8240 } 8241 return status; 8242} 8243 8244status_t AudioFlinger::EffectModule::start() 8245{ 8246 Mutex::Autolock _l(mLock); 8247 return start_l(); 8248} 8249 8250status_t AudioFlinger::EffectModule::start_l() 8251{ 8252 if (mEffectInterface == NULL) { 8253 return NO_INIT; 8254 } 8255 status_t cmdStatus; 8256 uint32_t size = sizeof(status_t); 8257 status_t status = (*mEffectInterface)->command(mEffectInterface, 8258 EFFECT_CMD_ENABLE, 8259 0, 8260 NULL, 8261 &size, 8262 &cmdStatus); 8263 if (status == 0) { 8264 status = cmdStatus; 8265 } 8266 if (status == 0 && 8267 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8268 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8269 sp<ThreadBase> thread = mThread.promote(); 8270 if (thread != 0) { 8271 audio_stream_t *stream = thread->stream(); 8272 if (stream != NULL) { 8273 stream->add_audio_effect(stream, mEffectInterface); 8274 } 8275 } 8276 } 8277 return status; 8278} 8279 8280status_t AudioFlinger::EffectModule::stop() 8281{ 8282 Mutex::Autolock _l(mLock); 8283 return stop_l(); 8284} 8285 8286status_t AudioFlinger::EffectModule::stop_l() 8287{ 8288 if (mEffectInterface == NULL) { 8289 return NO_INIT; 8290 } 8291 status_t cmdStatus; 8292 uint32_t size = sizeof(status_t); 8293 status_t status = (*mEffectInterface)->command(mEffectInterface, 8294 EFFECT_CMD_DISABLE, 8295 0, 8296 NULL, 8297 &size, 8298 &cmdStatus); 8299 if (status == 0) { 8300 status = cmdStatus; 8301 } 8302 if (status == 0 && 8303 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8304 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8305 sp<ThreadBase> thread = mThread.promote(); 8306 if (thread != 0) { 8307 audio_stream_t *stream = thread->stream(); 8308 if (stream != NULL) { 8309 stream->remove_audio_effect(stream, mEffectInterface); 8310 } 8311 } 8312 } 8313 return status; 8314} 8315 8316status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8317 uint32_t cmdSize, 8318 void *pCmdData, 8319 uint32_t *replySize, 8320 void *pReplyData) 8321{ 8322 Mutex::Autolock _l(mLock); 8323// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8324 8325 if (mState == DESTROYED || mEffectInterface == NULL) { 8326 return NO_INIT; 8327 } 8328 status_t status = (*mEffectInterface)->command(mEffectInterface, 8329 cmdCode, 8330 cmdSize, 8331 pCmdData, 8332 replySize, 8333 pReplyData); 8334 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8335 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8336 for (size_t i = 1; i < mHandles.size(); i++) { 8337 sp<EffectHandle> h = mHandles[i].promote(); 8338 if (h != 0) { 8339 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8340 } 8341 } 8342 } 8343 return status; 8344} 8345 8346status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8347{ 8348 8349 Mutex::Autolock _l(mLock); 8350 ALOGV("setEnabled %p enabled %d", this, enabled); 8351 8352 if (enabled != isEnabled()) { 8353 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8354 if (enabled && status != NO_ERROR) { 8355 return status; 8356 } 8357 8358 switch (mState) { 8359 // going from disabled to enabled 8360 case IDLE: 8361 mState = STARTING; 8362 break; 8363 case STOPPED: 8364 mState = RESTART; 8365 break; 8366 case STOPPING: 8367 mState = ACTIVE; 8368 break; 8369 8370 // going from enabled to disabled 8371 case RESTART: 8372 mState = STOPPED; 8373 break; 8374 case STARTING: 8375 mState = IDLE; 8376 break; 8377 case ACTIVE: 8378 mState = STOPPING; 8379 break; 8380 case DESTROYED: 8381 return NO_ERROR; // simply ignore as we are being destroyed 8382 } 8383 for (size_t i = 1; i < mHandles.size(); i++) { 8384 sp<EffectHandle> h = mHandles[i].promote(); 8385 if (h != 0) { 8386 h->setEnabled(enabled); 8387 } 8388 } 8389 } 8390 return NO_ERROR; 8391} 8392 8393bool AudioFlinger::EffectModule::isEnabled() const 8394{ 8395 switch (mState) { 8396 case RESTART: 8397 case STARTING: 8398 case ACTIVE: 8399 return true; 8400 case IDLE: 8401 case STOPPING: 8402 case STOPPED: 8403 case DESTROYED: 8404 default: 8405 return false; 8406 } 8407} 8408 8409bool AudioFlinger::EffectModule::isProcessEnabled() const 8410{ 8411 switch (mState) { 8412 case RESTART: 8413 case ACTIVE: 8414 case STOPPING: 8415 case STOPPED: 8416 return true; 8417 case IDLE: 8418 case STARTING: 8419 case DESTROYED: 8420 default: 8421 return false; 8422 } 8423} 8424 8425status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8426{ 8427 Mutex::Autolock _l(mLock); 8428 status_t status = NO_ERROR; 8429 8430 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8431 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8432 if (isProcessEnabled() && 8433 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8434 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8435 status_t cmdStatus; 8436 uint32_t volume[2]; 8437 uint32_t *pVolume = NULL; 8438 uint32_t size = sizeof(volume); 8439 volume[0] = *left; 8440 volume[1] = *right; 8441 if (controller) { 8442 pVolume = volume; 8443 } 8444 status = (*mEffectInterface)->command(mEffectInterface, 8445 EFFECT_CMD_SET_VOLUME, 8446 size, 8447 volume, 8448 &size, 8449 pVolume); 8450 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8451 *left = volume[0]; 8452 *right = volume[1]; 8453 } 8454 } 8455 return status; 8456} 8457 8458status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8459{ 8460 Mutex::Autolock _l(mLock); 8461 status_t status = NO_ERROR; 8462 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8463 // audio pre processing modules on RecordThread can receive both output and 8464 // input device indication in the same call 8465 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8466 if (dev) { 8467 status_t cmdStatus; 8468 uint32_t size = sizeof(status_t); 8469 8470 status = (*mEffectInterface)->command(mEffectInterface, 8471 EFFECT_CMD_SET_DEVICE, 8472 sizeof(uint32_t), 8473 &dev, 8474 &size, 8475 &cmdStatus); 8476 if (status == NO_ERROR) { 8477 status = cmdStatus; 8478 } 8479 } 8480 dev = device & AUDIO_DEVICE_IN_ALL; 8481 if (dev) { 8482 status_t cmdStatus; 8483 uint32_t size = sizeof(status_t); 8484 8485 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8486 EFFECT_CMD_SET_INPUT_DEVICE, 8487 sizeof(uint32_t), 8488 &dev, 8489 &size, 8490 &cmdStatus); 8491 if (status2 == NO_ERROR) { 8492 status2 = cmdStatus; 8493 } 8494 if (status == NO_ERROR) { 8495 status = status2; 8496 } 8497 } 8498 } 8499 return status; 8500} 8501 8502status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8503{ 8504 Mutex::Autolock _l(mLock); 8505 status_t status = NO_ERROR; 8506 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8507 status_t cmdStatus; 8508 uint32_t size = sizeof(status_t); 8509 status = (*mEffectInterface)->command(mEffectInterface, 8510 EFFECT_CMD_SET_AUDIO_MODE, 8511 sizeof(audio_mode_t), 8512 &mode, 8513 &size, 8514 &cmdStatus); 8515 if (status == NO_ERROR) { 8516 status = cmdStatus; 8517 } 8518 } 8519 return status; 8520} 8521 8522void AudioFlinger::EffectModule::setSuspended(bool suspended) 8523{ 8524 Mutex::Autolock _l(mLock); 8525 mSuspended = suspended; 8526} 8527 8528bool AudioFlinger::EffectModule::suspended() const 8529{ 8530 Mutex::Autolock _l(mLock); 8531 return mSuspended; 8532} 8533 8534status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8535{ 8536 const size_t SIZE = 256; 8537 char buffer[SIZE]; 8538 String8 result; 8539 8540 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8541 result.append(buffer); 8542 8543 bool locked = tryLock(mLock); 8544 // failed to lock - AudioFlinger is probably deadlocked 8545 if (!locked) { 8546 result.append("\t\tCould not lock Fx mutex:\n"); 8547 } 8548 8549 result.append("\t\tSession Status State Engine:\n"); 8550 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8551 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8552 result.append(buffer); 8553 8554 result.append("\t\tDescriptor:\n"); 8555 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8556 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8557 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8558 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8559 result.append(buffer); 8560 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8561 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8562 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8563 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8564 result.append(buffer); 8565 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8566 mDescriptor.apiVersion, 8567 mDescriptor.flags); 8568 result.append(buffer); 8569 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8570 mDescriptor.name); 8571 result.append(buffer); 8572 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8573 mDescriptor.implementor); 8574 result.append(buffer); 8575 8576 result.append("\t\t- Input configuration:\n"); 8577 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8578 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8579 (uint32_t)mConfig.inputCfg.buffer.raw, 8580 mConfig.inputCfg.buffer.frameCount, 8581 mConfig.inputCfg.samplingRate, 8582 mConfig.inputCfg.channels, 8583 mConfig.inputCfg.format); 8584 result.append(buffer); 8585 8586 result.append("\t\t- Output configuration:\n"); 8587 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8588 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8589 (uint32_t)mConfig.outputCfg.buffer.raw, 8590 mConfig.outputCfg.buffer.frameCount, 8591 mConfig.outputCfg.samplingRate, 8592 mConfig.outputCfg.channels, 8593 mConfig.outputCfg.format); 8594 result.append(buffer); 8595 8596 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8597 result.append(buffer); 8598 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8599 for (size_t i = 0; i < mHandles.size(); ++i) { 8600 sp<EffectHandle> handle = mHandles[i].promote(); 8601 if (handle != 0) { 8602 handle->dump(buffer, SIZE); 8603 result.append(buffer); 8604 } 8605 } 8606 8607 result.append("\n"); 8608 8609 write(fd, result.string(), result.length()); 8610 8611 if (locked) { 8612 mLock.unlock(); 8613 } 8614 8615 return NO_ERROR; 8616} 8617 8618// ---------------------------------------------------------------------------- 8619// EffectHandle implementation 8620// ---------------------------------------------------------------------------- 8621 8622#undef LOG_TAG 8623#define LOG_TAG "AudioFlinger::EffectHandle" 8624 8625AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8626 const sp<AudioFlinger::Client>& client, 8627 const sp<IEffectClient>& effectClient, 8628 int32_t priority) 8629 : BnEffect(), 8630 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8631 mPriority(priority), mHasControl(false), mEnabled(false) 8632{ 8633 ALOGV("constructor %p", this); 8634 8635 if (client == 0) { 8636 return; 8637 } 8638 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8639 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8640 if (mCblkMemory != 0) { 8641 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8642 8643 if (mCblk != NULL) { 8644 new(mCblk) effect_param_cblk_t(); 8645 mBuffer = (uint8_t *)mCblk + bufOffset; 8646 } 8647 } else { 8648 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8649 return; 8650 } 8651} 8652 8653AudioFlinger::EffectHandle::~EffectHandle() 8654{ 8655 ALOGV("Destructor %p", this); 8656 disconnect(false); 8657 ALOGV("Destructor DONE %p", this); 8658} 8659 8660status_t AudioFlinger::EffectHandle::enable() 8661{ 8662 ALOGV("enable %p", this); 8663 if (!mHasControl) return INVALID_OPERATION; 8664 if (mEffect == 0) return DEAD_OBJECT; 8665 8666 if (mEnabled) { 8667 return NO_ERROR; 8668 } 8669 8670 mEnabled = true; 8671 8672 sp<ThreadBase> thread = mEffect->thread().promote(); 8673 if (thread != 0) { 8674 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8675 } 8676 8677 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8678 if (mEffect->suspended()) { 8679 return NO_ERROR; 8680 } 8681 8682 status_t status = mEffect->setEnabled(true); 8683 if (status != NO_ERROR) { 8684 if (thread != 0) { 8685 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8686 } 8687 mEnabled = false; 8688 } 8689 return status; 8690} 8691 8692status_t AudioFlinger::EffectHandle::disable() 8693{ 8694 ALOGV("disable %p", this); 8695 if (!mHasControl) return INVALID_OPERATION; 8696 if (mEffect == 0) return DEAD_OBJECT; 8697 8698 if (!mEnabled) { 8699 return NO_ERROR; 8700 } 8701 mEnabled = false; 8702 8703 if (mEffect->suspended()) { 8704 return NO_ERROR; 8705 } 8706 8707 status_t status = mEffect->setEnabled(false); 8708 8709 sp<ThreadBase> thread = mEffect->thread().promote(); 8710 if (thread != 0) { 8711 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8712 } 8713 8714 return status; 8715} 8716 8717void AudioFlinger::EffectHandle::disconnect() 8718{ 8719 disconnect(true); 8720} 8721 8722void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8723{ 8724 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8725 if (mEffect == 0) { 8726 return; 8727 } 8728 mEffect->disconnect(this, unpinIfLast); 8729 8730 if (mHasControl && mEnabled) { 8731 sp<ThreadBase> thread = mEffect->thread().promote(); 8732 if (thread != 0) { 8733 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8734 } 8735 } 8736 8737 // release sp on module => module destructor can be called now 8738 mEffect.clear(); 8739 if (mClient != 0) { 8740 if (mCblk != NULL) { 8741 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8742 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8743 } 8744 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8745 // Client destructor must run with AudioFlinger mutex locked 8746 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8747 mClient.clear(); 8748 } 8749} 8750 8751status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8752 uint32_t cmdSize, 8753 void *pCmdData, 8754 uint32_t *replySize, 8755 void *pReplyData) 8756{ 8757// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8758// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8759 8760 // only get parameter command is permitted for applications not controlling the effect 8761 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8762 return INVALID_OPERATION; 8763 } 8764 if (mEffect == 0) return DEAD_OBJECT; 8765 if (mClient == 0) return INVALID_OPERATION; 8766 8767 // handle commands that are not forwarded transparently to effect engine 8768 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8769 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8770 // no risk to block the whole media server process or mixer threads is we are stuck here 8771 Mutex::Autolock _l(mCblk->lock); 8772 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8773 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8774 mCblk->serverIndex = 0; 8775 mCblk->clientIndex = 0; 8776 return BAD_VALUE; 8777 } 8778 status_t status = NO_ERROR; 8779 while (mCblk->serverIndex < mCblk->clientIndex) { 8780 int reply; 8781 uint32_t rsize = sizeof(int); 8782 int *p = (int *)(mBuffer + mCblk->serverIndex); 8783 int size = *p++; 8784 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8785 ALOGW("command(): invalid parameter block size"); 8786 break; 8787 } 8788 effect_param_t *param = (effect_param_t *)p; 8789 if (param->psize == 0 || param->vsize == 0) { 8790 ALOGW("command(): null parameter or value size"); 8791 mCblk->serverIndex += size; 8792 continue; 8793 } 8794 uint32_t psize = sizeof(effect_param_t) + 8795 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8796 param->vsize; 8797 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8798 psize, 8799 p, 8800 &rsize, 8801 &reply); 8802 // stop at first error encountered 8803 if (ret != NO_ERROR) { 8804 status = ret; 8805 *(int *)pReplyData = reply; 8806 break; 8807 } else if (reply != NO_ERROR) { 8808 *(int *)pReplyData = reply; 8809 break; 8810 } 8811 mCblk->serverIndex += size; 8812 } 8813 mCblk->serverIndex = 0; 8814 mCblk->clientIndex = 0; 8815 return status; 8816 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8817 *(int *)pReplyData = NO_ERROR; 8818 return enable(); 8819 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8820 *(int *)pReplyData = NO_ERROR; 8821 return disable(); 8822 } 8823 8824 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8825} 8826 8827void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8828{ 8829 ALOGV("setControl %p control %d", this, hasControl); 8830 8831 mHasControl = hasControl; 8832 mEnabled = enabled; 8833 8834 if (signal && mEffectClient != 0) { 8835 mEffectClient->controlStatusChanged(hasControl); 8836 } 8837} 8838 8839void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8840 uint32_t cmdSize, 8841 void *pCmdData, 8842 uint32_t replySize, 8843 void *pReplyData) 8844{ 8845 if (mEffectClient != 0) { 8846 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8847 } 8848} 8849 8850 8851 8852void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8853{ 8854 if (mEffectClient != 0) { 8855 mEffectClient->enableStatusChanged(enabled); 8856 } 8857} 8858 8859status_t AudioFlinger::EffectHandle::onTransact( 8860 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8861{ 8862 return BnEffect::onTransact(code, data, reply, flags); 8863} 8864 8865 8866void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8867{ 8868 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8869 8870 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8871 (mClient == 0) ? getpid_cached : mClient->pid(), 8872 mPriority, 8873 mHasControl, 8874 !locked, 8875 mCblk ? mCblk->clientIndex : 0, 8876 mCblk ? mCblk->serverIndex : 0 8877 ); 8878 8879 if (locked) { 8880 mCblk->lock.unlock(); 8881 } 8882} 8883 8884#undef LOG_TAG 8885#define LOG_TAG "AudioFlinger::EffectChain" 8886 8887AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8888 int sessionId) 8889 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8890 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8891 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8892{ 8893 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8894 if (thread == NULL) { 8895 return; 8896 } 8897 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8898 thread->frameCount(); 8899} 8900 8901AudioFlinger::EffectChain::~EffectChain() 8902{ 8903 if (mOwnInBuffer) { 8904 delete mInBuffer; 8905 } 8906 8907} 8908 8909// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8910sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8911{ 8912 size_t size = mEffects.size(); 8913 8914 for (size_t i = 0; i < size; i++) { 8915 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8916 return mEffects[i]; 8917 } 8918 } 8919 return 0; 8920} 8921 8922// getEffectFromId_l() must be called with ThreadBase::mLock held 8923sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8924{ 8925 size_t size = mEffects.size(); 8926 8927 for (size_t i = 0; i < size; i++) { 8928 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8929 if (id == 0 || mEffects[i]->id() == id) { 8930 return mEffects[i]; 8931 } 8932 } 8933 return 0; 8934} 8935 8936// getEffectFromType_l() must be called with ThreadBase::mLock held 8937sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8938 const effect_uuid_t *type) 8939{ 8940 size_t size = mEffects.size(); 8941 8942 for (size_t i = 0; i < size; i++) { 8943 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8944 return mEffects[i]; 8945 } 8946 } 8947 return 0; 8948} 8949 8950void AudioFlinger::EffectChain::clearInputBuffer() 8951{ 8952 Mutex::Autolock _l(mLock); 8953 sp<ThreadBase> thread = mThread.promote(); 8954 if (thread == 0) { 8955 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 8956 return; 8957 } 8958 clearInputBuffer_l(thread); 8959} 8960 8961// Must be called with EffectChain::mLock locked 8962void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 8963{ 8964 size_t numSamples = thread->frameCount() * thread->channelCount(); 8965 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8966 8967} 8968 8969// Must be called with EffectChain::mLock locked 8970void AudioFlinger::EffectChain::process_l() 8971{ 8972 sp<ThreadBase> thread = mThread.promote(); 8973 if (thread == 0) { 8974 ALOGW("process_l(): cannot promote mixer thread"); 8975 return; 8976 } 8977 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8978 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8979 // always process effects unless no more tracks are on the session and the effect tail 8980 // has been rendered 8981 bool doProcess = true; 8982 if (!isGlobalSession) { 8983 bool tracksOnSession = (trackCnt() != 0); 8984 8985 if (!tracksOnSession && mTailBufferCount == 0) { 8986 doProcess = false; 8987 } 8988 8989 if (activeTrackCnt() == 0) { 8990 // if no track is active and the effect tail has not been rendered, 8991 // the input buffer must be cleared here as the mixer process will not do it 8992 if (tracksOnSession || mTailBufferCount > 0) { 8993 clearInputBuffer_l(thread); 8994 if (mTailBufferCount > 0) { 8995 mTailBufferCount--; 8996 } 8997 } 8998 } 8999 } 9000 9001 size_t size = mEffects.size(); 9002 if (doProcess) { 9003 for (size_t i = 0; i < size; i++) { 9004 mEffects[i]->process(); 9005 } 9006 } 9007 for (size_t i = 0; i < size; i++) { 9008 mEffects[i]->updateState(); 9009 } 9010} 9011 9012// addEffect_l() must be called with PlaybackThread::mLock held 9013status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9014{ 9015 effect_descriptor_t desc = effect->desc(); 9016 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9017 9018 Mutex::Autolock _l(mLock); 9019 effect->setChain(this); 9020 sp<ThreadBase> thread = mThread.promote(); 9021 if (thread == 0) { 9022 return NO_INIT; 9023 } 9024 effect->setThread(thread); 9025 9026 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9027 // Auxiliary effects are inserted at the beginning of mEffects vector as 9028 // they are processed first and accumulated in chain input buffer 9029 mEffects.insertAt(effect, 0); 9030 9031 // the input buffer for auxiliary effect contains mono samples in 9032 // 32 bit format. This is to avoid saturation in AudoMixer 9033 // accumulation stage. Saturation is done in EffectModule::process() before 9034 // calling the process in effect engine 9035 size_t numSamples = thread->frameCount(); 9036 int32_t *buffer = new int32_t[numSamples]; 9037 memset(buffer, 0, numSamples * sizeof(int32_t)); 9038 effect->setInBuffer((int16_t *)buffer); 9039 // auxiliary effects output samples to chain input buffer for further processing 9040 // by insert effects 9041 effect->setOutBuffer(mInBuffer); 9042 } else { 9043 // Insert effects are inserted at the end of mEffects vector as they are processed 9044 // after track and auxiliary effects. 9045 // Insert effect order as a function of indicated preference: 9046 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9047 // another effect is present 9048 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9049 // last effect claiming first position 9050 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9051 // first effect claiming last position 9052 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9053 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9054 // already present 9055 9056 size_t size = mEffects.size(); 9057 size_t idx_insert = size; 9058 ssize_t idx_insert_first = -1; 9059 ssize_t idx_insert_last = -1; 9060 9061 for (size_t i = 0; i < size; i++) { 9062 effect_descriptor_t d = mEffects[i]->desc(); 9063 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9064 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9065 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9066 // check invalid effect chaining combinations 9067 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9068 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9069 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9070 return INVALID_OPERATION; 9071 } 9072 // remember position of first insert effect and by default 9073 // select this as insert position for new effect 9074 if (idx_insert == size) { 9075 idx_insert = i; 9076 } 9077 // remember position of last insert effect claiming 9078 // first position 9079 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9080 idx_insert_first = i; 9081 } 9082 // remember position of first insert effect claiming 9083 // last position 9084 if (iPref == EFFECT_FLAG_INSERT_LAST && 9085 idx_insert_last == -1) { 9086 idx_insert_last = i; 9087 } 9088 } 9089 } 9090 9091 // modify idx_insert from first position if needed 9092 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9093 if (idx_insert_last != -1) { 9094 idx_insert = idx_insert_last; 9095 } else { 9096 idx_insert = size; 9097 } 9098 } else { 9099 if (idx_insert_first != -1) { 9100 idx_insert = idx_insert_first + 1; 9101 } 9102 } 9103 9104 // always read samples from chain input buffer 9105 effect->setInBuffer(mInBuffer); 9106 9107 // if last effect in the chain, output samples to chain 9108 // output buffer, otherwise to chain input buffer 9109 if (idx_insert == size) { 9110 if (idx_insert != 0) { 9111 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9112 mEffects[idx_insert-1]->configure(); 9113 } 9114 effect->setOutBuffer(mOutBuffer); 9115 } else { 9116 effect->setOutBuffer(mInBuffer); 9117 } 9118 mEffects.insertAt(effect, idx_insert); 9119 9120 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9121 } 9122 effect->configure(); 9123 return NO_ERROR; 9124} 9125 9126// removeEffect_l() must be called with PlaybackThread::mLock held 9127size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9128{ 9129 Mutex::Autolock _l(mLock); 9130 size_t size = mEffects.size(); 9131 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9132 9133 for (size_t i = 0; i < size; i++) { 9134 if (effect == mEffects[i]) { 9135 // calling stop here will remove pre-processing effect from the audio HAL. 9136 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9137 // the middle of a read from audio HAL 9138 if (mEffects[i]->state() == EffectModule::ACTIVE || 9139 mEffects[i]->state() == EffectModule::STOPPING) { 9140 mEffects[i]->stop(); 9141 } 9142 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9143 delete[] effect->inBuffer(); 9144 } else { 9145 if (i == size - 1 && i != 0) { 9146 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9147 mEffects[i - 1]->configure(); 9148 } 9149 } 9150 mEffects.removeAt(i); 9151 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9152 break; 9153 } 9154 } 9155 9156 return mEffects.size(); 9157} 9158 9159// setDevice_l() must be called with PlaybackThread::mLock held 9160void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9161{ 9162 size_t size = mEffects.size(); 9163 for (size_t i = 0; i < size; i++) { 9164 mEffects[i]->setDevice(device); 9165 } 9166} 9167 9168// setMode_l() must be called with PlaybackThread::mLock held 9169void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9170{ 9171 size_t size = mEffects.size(); 9172 for (size_t i = 0; i < size; i++) { 9173 mEffects[i]->setMode(mode); 9174 } 9175} 9176 9177// setVolume_l() must be called with PlaybackThread::mLock held 9178bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9179{ 9180 uint32_t newLeft = *left; 9181 uint32_t newRight = *right; 9182 bool hasControl = false; 9183 int ctrlIdx = -1; 9184 size_t size = mEffects.size(); 9185 9186 // first update volume controller 9187 for (size_t i = size; i > 0; i--) { 9188 if (mEffects[i - 1]->isProcessEnabled() && 9189 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9190 ctrlIdx = i - 1; 9191 hasControl = true; 9192 break; 9193 } 9194 } 9195 9196 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9197 if (hasControl) { 9198 *left = mNewLeftVolume; 9199 *right = mNewRightVolume; 9200 } 9201 return hasControl; 9202 } 9203 9204 mVolumeCtrlIdx = ctrlIdx; 9205 mLeftVolume = newLeft; 9206 mRightVolume = newRight; 9207 9208 // second get volume update from volume controller 9209 if (ctrlIdx >= 0) { 9210 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9211 mNewLeftVolume = newLeft; 9212 mNewRightVolume = newRight; 9213 } 9214 // then indicate volume to all other effects in chain. 9215 // Pass altered volume to effects before volume controller 9216 // and requested volume to effects after controller 9217 uint32_t lVol = newLeft; 9218 uint32_t rVol = newRight; 9219 9220 for (size_t i = 0; i < size; i++) { 9221 if ((int)i == ctrlIdx) continue; 9222 // this also works for ctrlIdx == -1 when there is no volume controller 9223 if ((int)i > ctrlIdx) { 9224 lVol = *left; 9225 rVol = *right; 9226 } 9227 mEffects[i]->setVolume(&lVol, &rVol, false); 9228 } 9229 *left = newLeft; 9230 *right = newRight; 9231 9232 return hasControl; 9233} 9234 9235status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9236{ 9237 const size_t SIZE = 256; 9238 char buffer[SIZE]; 9239 String8 result; 9240 9241 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9242 result.append(buffer); 9243 9244 bool locked = tryLock(mLock); 9245 // failed to lock - AudioFlinger is probably deadlocked 9246 if (!locked) { 9247 result.append("\tCould not lock mutex:\n"); 9248 } 9249 9250 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9251 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9252 mEffects.size(), 9253 (uint32_t)mInBuffer, 9254 (uint32_t)mOutBuffer, 9255 mActiveTrackCnt); 9256 result.append(buffer); 9257 write(fd, result.string(), result.size()); 9258 9259 for (size_t i = 0; i < mEffects.size(); ++i) { 9260 sp<EffectModule> effect = mEffects[i]; 9261 if (effect != 0) { 9262 effect->dump(fd, args); 9263 } 9264 } 9265 9266 if (locked) { 9267 mLock.unlock(); 9268 } 9269 9270 return NO_ERROR; 9271} 9272 9273// must be called with ThreadBase::mLock held 9274void AudioFlinger::EffectChain::setEffectSuspended_l( 9275 const effect_uuid_t *type, bool suspend) 9276{ 9277 sp<SuspendedEffectDesc> desc; 9278 // use effect type UUID timelow as key as there is no real risk of identical 9279 // timeLow fields among effect type UUIDs. 9280 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9281 if (suspend) { 9282 if (index >= 0) { 9283 desc = mSuspendedEffects.valueAt(index); 9284 } else { 9285 desc = new SuspendedEffectDesc(); 9286 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9287 mSuspendedEffects.add(type->timeLow, desc); 9288 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9289 } 9290 if (desc->mRefCount++ == 0) { 9291 sp<EffectModule> effect = getEffectIfEnabled(type); 9292 if (effect != 0) { 9293 desc->mEffect = effect; 9294 effect->setSuspended(true); 9295 effect->setEnabled(false); 9296 } 9297 } 9298 } else { 9299 if (index < 0) { 9300 return; 9301 } 9302 desc = mSuspendedEffects.valueAt(index); 9303 if (desc->mRefCount <= 0) { 9304 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9305 desc->mRefCount = 1; 9306 } 9307 if (--desc->mRefCount == 0) { 9308 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9309 if (desc->mEffect != 0) { 9310 sp<EffectModule> effect = desc->mEffect.promote(); 9311 if (effect != 0) { 9312 effect->setSuspended(false); 9313 sp<EffectHandle> handle = effect->controlHandle(); 9314 if (handle != 0) { 9315 effect->setEnabled(handle->enabled()); 9316 } 9317 } 9318 desc->mEffect.clear(); 9319 } 9320 mSuspendedEffects.removeItemsAt(index); 9321 } 9322 } 9323} 9324 9325// must be called with ThreadBase::mLock held 9326void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9327{ 9328 sp<SuspendedEffectDesc> desc; 9329 9330 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9331 if (suspend) { 9332 if (index >= 0) { 9333 desc = mSuspendedEffects.valueAt(index); 9334 } else { 9335 desc = new SuspendedEffectDesc(); 9336 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9337 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9338 } 9339 if (desc->mRefCount++ == 0) { 9340 Vector< sp<EffectModule> > effects; 9341 getSuspendEligibleEffects(effects); 9342 for (size_t i = 0; i < effects.size(); i++) { 9343 setEffectSuspended_l(&effects[i]->desc().type, true); 9344 } 9345 } 9346 } else { 9347 if (index < 0) { 9348 return; 9349 } 9350 desc = mSuspendedEffects.valueAt(index); 9351 if (desc->mRefCount <= 0) { 9352 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9353 desc->mRefCount = 1; 9354 } 9355 if (--desc->mRefCount == 0) { 9356 Vector<const effect_uuid_t *> types; 9357 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9358 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9359 continue; 9360 } 9361 types.add(&mSuspendedEffects.valueAt(i)->mType); 9362 } 9363 for (size_t i = 0; i < types.size(); i++) { 9364 setEffectSuspended_l(types[i], false); 9365 } 9366 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9367 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9368 } 9369 } 9370} 9371 9372 9373// The volume effect is used for automated tests only 9374#ifndef OPENSL_ES_H_ 9375static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9376 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9377const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9378#endif //OPENSL_ES_H_ 9379 9380bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9381{ 9382 // auxiliary effects and visualizer are never suspended on output mix 9383 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9384 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9385 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9386 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9387 return false; 9388 } 9389 return true; 9390} 9391 9392void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9393{ 9394 effects.clear(); 9395 for (size_t i = 0; i < mEffects.size(); i++) { 9396 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9397 effects.add(mEffects[i]); 9398 } 9399 } 9400} 9401 9402sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9403 const effect_uuid_t *type) 9404{ 9405 sp<EffectModule> effect = getEffectFromType_l(type); 9406 return effect != 0 && effect->isEnabled() ? effect : 0; 9407} 9408 9409void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9410 bool enabled) 9411{ 9412 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9413 if (enabled) { 9414 if (index < 0) { 9415 // if the effect is not suspend check if all effects are suspended 9416 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9417 if (index < 0) { 9418 return; 9419 } 9420 if (!isEffectEligibleForSuspend(effect->desc())) { 9421 return; 9422 } 9423 setEffectSuspended_l(&effect->desc().type, enabled); 9424 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9425 if (index < 0) { 9426 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9427 return; 9428 } 9429 } 9430 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9431 effect->desc().type.timeLow); 9432 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9433 // if effect is requested to suspended but was not yet enabled, supend it now. 9434 if (desc->mEffect == 0) { 9435 desc->mEffect = effect; 9436 effect->setEnabled(false); 9437 effect->setSuspended(true); 9438 } 9439 } else { 9440 if (index < 0) { 9441 return; 9442 } 9443 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9444 effect->desc().type.timeLow); 9445 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9446 desc->mEffect.clear(); 9447 effect->setSuspended(false); 9448 } 9449} 9450 9451#undef LOG_TAG 9452#define LOG_TAG "AudioFlinger" 9453 9454// ---------------------------------------------------------------------------- 9455 9456status_t AudioFlinger::onTransact( 9457 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9458{ 9459 return BnAudioFlinger::onTransact(code, data, reply, flags); 9460} 9461 9462}; // namespace android 9463