AudioFlinger.cpp revision 1d6573032ecde54a466ca32951e101b41a05c797
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 168 // AudioFlinger::setParameters() updates, other threads read w/o lock 169 170// ---------------------------------------------------------------------------- 171 172#ifdef ADD_BATTERY_DATA 173// To collect the amplifier usage 174static void addBatteryData(uint32_t params) { 175 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 176 if (service == NULL) { 177 // it already logged 178 return; 179 } 180 181 service->addBatteryData(params); 182} 183#endif 184 185static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 186{ 187 const hw_module_t *mod; 188 int rc; 189 190 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 191 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 rc = audio_hw_device_open(mod, dev); 197 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 198 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 199 if (rc) { 200 goto out; 201 } 202 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 203 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 204 rc = BAD_VALUE; 205 goto out; 206 } 207 return 0; 208 209out: 210 *dev = NULL; 211 return rc; 212} 213 214// ---------------------------------------------------------------------------- 215 216AudioFlinger::AudioFlinger() 217 : BnAudioFlinger(), 218 mPrimaryHardwareDev(NULL), 219 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 220 mMasterVolume(1.0f), 221 mMasterVolumeSupportLvl(MVS_NONE), 222 mMasterMute(false), 223 mNextUniqueId(1), 224 mMode(AUDIO_MODE_INVALID), 225 mBtNrecIsOff(false) 226{ 227} 228 229void AudioFlinger::onFirstRef() 230{ 231 int rc = 0; 232 233 Mutex::Autolock _l(mLock); 234 235 /* TODO: move all this work into an Init() function */ 236 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 237 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 238 uint32_t int_val; 239 if (1 == sscanf(val_str, "%u", &int_val)) { 240 mStandbyTimeInNsecs = milliseconds(int_val); 241 ALOGI("Using %u mSec as standby time.", int_val); 242 } else { 243 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 244 ALOGI("Using default %u mSec as standby time.", 245 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 246 } 247 } 248 249 mMode = AUDIO_MODE_NORMAL; 250 mMasterVolumeSW = 1.0; 251 mMasterVolume = 1.0; 252 mHardwareStatus = AUDIO_HW_IDLE; 253} 254 255AudioFlinger::~AudioFlinger() 256{ 257 258 while (!mRecordThreads.isEmpty()) { 259 // closeInput() will remove first entry from mRecordThreads 260 closeInput(mRecordThreads.keyAt(0)); 261 } 262 while (!mPlaybackThreads.isEmpty()) { 263 // closeOutput() will remove first entry from mPlaybackThreads 264 closeOutput(mPlaybackThreads.keyAt(0)); 265 } 266 267 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 268 // no mHardwareLock needed, as there are no other references to this 269 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 270 delete mAudioHwDevs.valueAt(i); 271 } 272} 273 274static const char * const audio_interfaces[] = { 275 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 276 AUDIO_HARDWARE_MODULE_ID_A2DP, 277 AUDIO_HARDWARE_MODULE_ID_USB, 278}; 279#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 280 281audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 282{ 283 // if module is 0, the request comes from an old policy manager and we should load 284 // well known modules 285 if (module == 0) { 286 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 287 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 288 loadHwModule_l(audio_interfaces[i]); 289 } 290 } else { 291 // check a match for the requested module handle 292 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 293 if (audioHwdevice != NULL) { 294 return audioHwdevice->hwDevice(); 295 } 296 } 297 // then try to find a module supporting the requested device. 298 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 299 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 300 if ((dev->get_supported_devices(dev) & devices) == devices) 301 return dev; 302 } 303 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 IAudioFlinger::track_flags_t flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 pid_t tid, 449 int *sessionId, 450 status_t *status) 451{ 452 sp<PlaybackThread::Track> track; 453 sp<TrackHandle> trackHandle; 454 sp<Client> client; 455 status_t lStatus; 456 int lSessionId; 457 458 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 459 // but if someone uses binder directly they could bypass that and cause us to crash 460 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 461 ALOGE("createTrack() invalid stream type %d", streamType); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 466 { 467 Mutex::Autolock _l(mLock); 468 PlaybackThread *thread = checkPlaybackThread_l(output); 469 PlaybackThread *effectThread = NULL; 470 if (thread == NULL) { 471 ALOGE("unknown output thread"); 472 lStatus = BAD_VALUE; 473 goto Exit; 474 } 475 476 client = registerPid_l(pid); 477 478 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 479 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 480 // check if an effect chain with the same session ID is present on another 481 // output thread and move it here. 482 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 483 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 484 if (mPlaybackThreads.keyAt(i) != output) { 485 uint32_t sessions = t->hasAudioSession(*sessionId); 486 if (sessions & PlaybackThread::EFFECT_SESSION) { 487 effectThread = t.get(); 488 break; 489 } 490 } 491 } 492 lSessionId = *sessionId; 493 } else { 494 // if no audio session id is provided, create one here 495 lSessionId = nextUniqueId(); 496 if (sessionId != NULL) { 497 *sessionId = lSessionId; 498 } 499 } 500 ALOGV("createTrack() lSessionId: %d", lSessionId); 501 502 track = thread->createTrack_l(client, streamType, sampleRate, format, 503 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 504 505 // move effect chain to this output thread if an effect on same session was waiting 506 // for a track to be created 507 if (lStatus == NO_ERROR && effectThread != NULL) { 508 Mutex::Autolock _dl(thread->mLock); 509 Mutex::Autolock _sl(effectThread->mLock); 510 moveEffectChain_l(lSessionId, effectThread, thread, true); 511 } 512 513 // Look for sync events awaiting for a session to be used. 514 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 515 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 516 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 517 if (lStatus == NO_ERROR) { 518 track->setSyncEvent(mPendingSyncEvents[i]); 519 } else { 520 mPendingSyncEvents[i]->cancel(); 521 } 522 mPendingSyncEvents.removeAt(i); 523 i--; 524 } 525 } 526 } 527 } 528 if (lStatus == NO_ERROR) { 529 trackHandle = new TrackHandle(track); 530 } else { 531 // remove local strong reference to Client before deleting the Track so that the Client 532 // destructor is called by the TrackBase destructor with mLock held 533 client.clear(); 534 track.clear(); 535 } 536 537Exit: 538 if (status != NULL) { 539 *status = lStatus; 540 } 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 586 // should examine all callers and fix them to handle smaller counts 587 return thread->frameCount(); 588} 589 590uint32_t AudioFlinger::latency(audio_io_handle_t output) const 591{ 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGW("latency() unknown thread %d", output); 596 return 0; 597 } 598 return thread->latency(); 599} 600 601status_t AudioFlinger::setMasterVolume(float value) 602{ 603 status_t ret = initCheck(); 604 if (ret != NO_ERROR) { 605 return ret; 606 } 607 608 // check calling permissions 609 if (!settingsAllowed()) { 610 return PERMISSION_DENIED; 611 } 612 613 float swmv = value; 614 615 Mutex::Autolock _l(mLock); 616 617 // when hw supports master volume, don't scale in sw mixer 618 if (MVS_NONE != mMasterVolumeSupportLvl) { 619 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 620 AutoMutex lock(mHardwareLock); 621 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 622 623 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 624 if (NULL != dev->set_master_volume) { 625 dev->set_master_volume(dev, value); 626 } 627 mHardwareStatus = AUDIO_HW_IDLE; 628 } 629 630 swmv = 1.0; 631 } 632 633 mMasterVolume = value; 634 mMasterVolumeSW = swmv; 635 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 636 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 637 638 return NO_ERROR; 639} 640 641status_t AudioFlinger::setMode(audio_mode_t mode) 642{ 643 status_t ret = initCheck(); 644 if (ret != NO_ERROR) { 645 return ret; 646 } 647 648 // check calling permissions 649 if (!settingsAllowed()) { 650 return PERMISSION_DENIED; 651 } 652 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 653 ALOGW("Illegal value: setMode(%d)", mode); 654 return BAD_VALUE; 655 } 656 657 { // scope for the lock 658 AutoMutex lock(mHardwareLock); 659 mHardwareStatus = AUDIO_HW_SET_MODE; 660 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 661 mHardwareStatus = AUDIO_HW_IDLE; 662 } 663 664 if (NO_ERROR == ret) { 665 Mutex::Autolock _l(mLock); 666 mMode = mode; 667 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 668 mPlaybackThreads.valueAt(i)->setMode(mode); 669 } 670 671 return ret; 672} 673 674status_t AudioFlinger::setMicMute(bool state) 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return ret; 679 } 680 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 AutoMutex lock(mHardwareLock); 687 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 688 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 689 mHardwareStatus = AUDIO_HW_IDLE; 690 return ret; 691} 692 693bool AudioFlinger::getMicMute() const 694{ 695 status_t ret = initCheck(); 696 if (ret != NO_ERROR) { 697 return false; 698 } 699 700 bool state = AUDIO_MODE_INVALID; 701 AutoMutex lock(mHardwareLock); 702 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 703 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 704 mHardwareStatus = AUDIO_HW_IDLE; 705 return state; 706} 707 708status_t AudioFlinger::setMasterMute(bool muted) 709{ 710 // check calling permissions 711 if (!settingsAllowed()) { 712 return PERMISSION_DENIED; 713 } 714 715 Mutex::Autolock _l(mLock); 716 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 717 mMasterMute = muted; 718 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 719 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 720 721 return NO_ERROR; 722} 723 724float AudioFlinger::masterVolume() const 725{ 726 Mutex::Autolock _l(mLock); 727 return masterVolume_l(); 728} 729 730float AudioFlinger::masterVolumeSW() const 731{ 732 Mutex::Autolock _l(mLock); 733 return masterVolumeSW_l(); 734} 735 736bool AudioFlinger::masterMute() const 737{ 738 Mutex::Autolock _l(mLock); 739 return masterMute_l(); 740} 741 742float AudioFlinger::masterVolume_l() const 743{ 744 if (MVS_FULL == mMasterVolumeSupportLvl) { 745 float ret_val; 746 AutoMutex lock(mHardwareLock); 747 748 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 749 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 750 (NULL != mPrimaryHardwareDev->get_master_volume), 751 "can't get master volume"); 752 753 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 754 mHardwareStatus = AUDIO_HW_IDLE; 755 return ret_val; 756 } 757 758 return mMasterVolume; 759} 760 761status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 762 audio_io_handle_t output) 763{ 764 // check calling permissions 765 if (!settingsAllowed()) { 766 return PERMISSION_DENIED; 767 } 768 769 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 770 ALOGE("setStreamVolume() invalid stream %d", stream); 771 return BAD_VALUE; 772 } 773 774 AutoMutex lock(mLock); 775 PlaybackThread *thread = NULL; 776 if (output) { 777 thread = checkPlaybackThread_l(output); 778 if (thread == NULL) { 779 return BAD_VALUE; 780 } 781 } 782 783 mStreamTypes[stream].volume = value; 784 785 if (thread == NULL) { 786 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 787 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 788 } 789 } else { 790 thread->setStreamVolume(stream, value); 791 } 792 793 return NO_ERROR; 794} 795 796status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 797{ 798 // check calling permissions 799 if (!settingsAllowed()) { 800 return PERMISSION_DENIED; 801 } 802 803 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 804 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 805 ALOGE("setStreamMute() invalid stream %d", stream); 806 return BAD_VALUE; 807 } 808 809 AutoMutex lock(mLock); 810 mStreamTypes[stream].mute = muted; 811 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 812 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 813 814 return NO_ERROR; 815} 816 817float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 818{ 819 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 820 return 0.0f; 821 } 822 823 AutoMutex lock(mLock); 824 float volume; 825 if (output) { 826 PlaybackThread *thread = checkPlaybackThread_l(output); 827 if (thread == NULL) { 828 return 0.0f; 829 } 830 volume = thread->streamVolume(stream); 831 } else { 832 volume = streamVolume_l(stream); 833 } 834 835 return volume; 836} 837 838bool AudioFlinger::streamMute(audio_stream_type_t stream) const 839{ 840 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 841 return true; 842 } 843 844 AutoMutex lock(mLock); 845 return streamMute_l(stream); 846} 847 848status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 849{ 850 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 851 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 852 // check calling permissions 853 if (!settingsAllowed()) { 854 return PERMISSION_DENIED; 855 } 856 857 // ioHandle == 0 means the parameters are global to the audio hardware interface 858 if (ioHandle == 0) { 859 Mutex::Autolock _l(mLock); 860 status_t final_result = NO_ERROR; 861 { 862 AutoMutex lock(mHardwareLock); 863 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 864 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 865 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 866 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 867 final_result = result ?: final_result; 868 } 869 mHardwareStatus = AUDIO_HW_IDLE; 870 } 871 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 872 AudioParameter param = AudioParameter(keyValuePairs); 873 String8 value; 874 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 875 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 876 if (mBtNrecIsOff != btNrecIsOff) { 877 for (size_t i = 0; i < mRecordThreads.size(); i++) { 878 sp<RecordThread> thread = mRecordThreads.valueAt(i); 879 RecordThread::RecordTrack *track = thread->track(); 880 if (track != NULL) { 881 audio_devices_t device = (audio_devices_t)( 882 thread->device() & AUDIO_DEVICE_IN_ALL); 883 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 884 thread->setEffectSuspended(FX_IID_AEC, 885 suspend, 886 track->sessionId()); 887 thread->setEffectSuspended(FX_IID_NS, 888 suspend, 889 track->sessionId()); 890 } 891 } 892 mBtNrecIsOff = btNrecIsOff; 893 } 894 } 895 String8 screenState; 896 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 897 bool isOff = screenState == "off"; 898 if (isOff != (gScreenState & 1)) { 899 gScreenState = ((gScreenState & ~1) + 2) | isOff; 900 } 901 } 902 return final_result; 903 } 904 905 // hold a strong ref on thread in case closeOutput() or closeInput() is called 906 // and the thread is exited once the lock is released 907 sp<ThreadBase> thread; 908 { 909 Mutex::Autolock _l(mLock); 910 thread = checkPlaybackThread_l(ioHandle); 911 if (thread == NULL) { 912 thread = checkRecordThread_l(ioHandle); 913 } else if (thread == primaryPlaybackThread_l()) { 914 // indicate output device change to all input threads for pre processing 915 AudioParameter param = AudioParameter(keyValuePairs); 916 int value; 917 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 918 (value != 0)) { 919 for (size_t i = 0; i < mRecordThreads.size(); i++) { 920 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 921 } 922 } 923 } 924 } 925 if (thread != 0) { 926 return thread->setParameters(keyValuePairs); 927 } 928 return BAD_VALUE; 929} 930 931String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 932{ 933// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 934// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 935 936 Mutex::Autolock _l(mLock); 937 938 if (ioHandle == 0) { 939 String8 out_s8; 940 941 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 942 char *s; 943 { 944 AutoMutex lock(mHardwareLock); 945 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 946 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 947 s = dev->get_parameters(dev, keys.string()); 948 mHardwareStatus = AUDIO_HW_IDLE; 949 } 950 out_s8 += String8(s ? s : ""); 951 free(s); 952 } 953 return out_s8; 954 } 955 956 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 957 if (playbackThread != NULL) { 958 return playbackThread->getParameters(keys); 959 } 960 RecordThread *recordThread = checkRecordThread_l(ioHandle); 961 if (recordThread != NULL) { 962 return recordThread->getParameters(keys); 963 } 964 return String8(""); 965} 966 967size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 968{ 969 status_t ret = initCheck(); 970 if (ret != NO_ERROR) { 971 return 0; 972 } 973 974 AutoMutex lock(mHardwareLock); 975 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 976 struct audio_config config = { 977 sample_rate: sampleRate, 978 channel_mask: audio_channel_in_mask_from_count(channelCount), 979 format: format, 980 }; 981 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 982 mHardwareStatus = AUDIO_HW_IDLE; 983 return size; 984} 985 986unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 987{ 988 if (ioHandle == 0) { 989 return 0; 990 } 991 992 Mutex::Autolock _l(mLock); 993 994 RecordThread *recordThread = checkRecordThread_l(ioHandle); 995 if (recordThread != NULL) { 996 return recordThread->getInputFramesLost(); 997 } 998 return 0; 999} 1000 1001status_t AudioFlinger::setVoiceVolume(float value) 1002{ 1003 status_t ret = initCheck(); 1004 if (ret != NO_ERROR) { 1005 return ret; 1006 } 1007 1008 // check calling permissions 1009 if (!settingsAllowed()) { 1010 return PERMISSION_DENIED; 1011 } 1012 1013 AutoMutex lock(mHardwareLock); 1014 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1015 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1016 mHardwareStatus = AUDIO_HW_IDLE; 1017 1018 return ret; 1019} 1020 1021status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1022 audio_io_handle_t output) const 1023{ 1024 status_t status; 1025 1026 Mutex::Autolock _l(mLock); 1027 1028 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1029 if (playbackThread != NULL) { 1030 return playbackThread->getRenderPosition(halFrames, dspFrames); 1031 } 1032 1033 return BAD_VALUE; 1034} 1035 1036void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1037{ 1038 1039 Mutex::Autolock _l(mLock); 1040 1041 pid_t pid = IPCThreadState::self()->getCallingPid(); 1042 if (mNotificationClients.indexOfKey(pid) < 0) { 1043 sp<NotificationClient> notificationClient = new NotificationClient(this, 1044 client, 1045 pid); 1046 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1047 1048 mNotificationClients.add(pid, notificationClient); 1049 1050 sp<IBinder> binder = client->asBinder(); 1051 binder->linkToDeath(notificationClient); 1052 1053 // the config change is always sent from playback or record threads to avoid deadlock 1054 // with AudioSystem::gLock 1055 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1056 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1057 } 1058 1059 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1060 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1061 } 1062 } 1063} 1064 1065void AudioFlinger::removeNotificationClient(pid_t pid) 1066{ 1067 Mutex::Autolock _l(mLock); 1068 1069 mNotificationClients.removeItem(pid); 1070 1071 ALOGV("%d died, releasing its sessions", pid); 1072 size_t num = mAudioSessionRefs.size(); 1073 bool removed = false; 1074 for (size_t i = 0; i< num; ) { 1075 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1076 ALOGV(" pid %d @ %d", ref->mPid, i); 1077 if (ref->mPid == pid) { 1078 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1079 mAudioSessionRefs.removeAt(i); 1080 delete ref; 1081 removed = true; 1082 num--; 1083 } else { 1084 i++; 1085 } 1086 } 1087 if (removed) { 1088 purgeStaleEffects_l(); 1089 } 1090} 1091 1092// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1094{ 1095 size_t size = mNotificationClients.size(); 1096 for (size_t i = 0; i < size; i++) { 1097 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1098 param2); 1099 } 1100} 1101 1102// removeClient_l() must be called with AudioFlinger::mLock held 1103void AudioFlinger::removeClient_l(pid_t pid) 1104{ 1105 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1106 mClients.removeItem(pid); 1107} 1108 1109 1110// ---------------------------------------------------------------------------- 1111 1112AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1113 uint32_t device, type_t type) 1114 : Thread(false), 1115 mType(type), 1116 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1117 // mChannelMask 1118 mChannelCount(0), 1119 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1120 mParamStatus(NO_ERROR), 1121 mStandby(false), mId(id), 1122 mDevice(device), 1123 mDeathRecipient(new PMDeathRecipient(this)) 1124{ 1125} 1126 1127AudioFlinger::ThreadBase::~ThreadBase() 1128{ 1129 mParamCond.broadcast(); 1130 // do not lock the mutex in destructor 1131 releaseWakeLock_l(); 1132 if (mPowerManager != 0) { 1133 sp<IBinder> binder = mPowerManager->asBinder(); 1134 binder->unlinkToDeath(mDeathRecipient); 1135 } 1136} 1137 1138void AudioFlinger::ThreadBase::exit() 1139{ 1140 ALOGV("ThreadBase::exit"); 1141 { 1142 // This lock prevents the following race in thread (uniprocessor for illustration): 1143 // if (!exitPending()) { 1144 // // context switch from here to exit() 1145 // // exit() calls requestExit(), what exitPending() observes 1146 // // exit() calls signal(), which is dropped since no waiters 1147 // // context switch back from exit() to here 1148 // mWaitWorkCV.wait(...); 1149 // // now thread is hung 1150 // } 1151 AutoMutex lock(mLock); 1152 requestExit(); 1153 mWaitWorkCV.signal(); 1154 } 1155 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1156 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1157 requestExitAndWait(); 1158} 1159 1160status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1161{ 1162 status_t status; 1163 1164 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1165 Mutex::Autolock _l(mLock); 1166 1167 mNewParameters.add(keyValuePairs); 1168 mWaitWorkCV.signal(); 1169 // wait condition with timeout in case the thread loop has exited 1170 // before the request could be processed 1171 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1172 status = mParamStatus; 1173 mWaitWorkCV.signal(); 1174 } else { 1175 status = TIMED_OUT; 1176 } 1177 return status; 1178} 1179 1180void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1181{ 1182 Mutex::Autolock _l(mLock); 1183 sendConfigEvent_l(event, param); 1184} 1185 1186// sendConfigEvent_l() must be called with ThreadBase::mLock held 1187void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1188{ 1189 ConfigEvent configEvent; 1190 configEvent.mEvent = event; 1191 configEvent.mParam = param; 1192 mConfigEvents.add(configEvent); 1193 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1194 mWaitWorkCV.signal(); 1195} 1196 1197void AudioFlinger::ThreadBase::processConfigEvents() 1198{ 1199 mLock.lock(); 1200 while (!mConfigEvents.isEmpty()) { 1201 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1202 ConfigEvent configEvent = mConfigEvents[0]; 1203 mConfigEvents.removeAt(0); 1204 // release mLock before locking AudioFlinger mLock: lock order is always 1205 // AudioFlinger then ThreadBase to avoid cross deadlock 1206 mLock.unlock(); 1207 mAudioFlinger->mLock.lock(); 1208 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1209 mAudioFlinger->mLock.unlock(); 1210 mLock.lock(); 1211 } 1212 mLock.unlock(); 1213} 1214 1215status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1216{ 1217 const size_t SIZE = 256; 1218 char buffer[SIZE]; 1219 String8 result; 1220 1221 bool locked = tryLock(mLock); 1222 if (!locked) { 1223 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1224 write(fd, buffer, strlen(buffer)); 1225 } 1226 1227 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1246 result.append(buffer); 1247 1248 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1249 result.append(buffer); 1250 result.append(" Index Command"); 1251 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1252 snprintf(buffer, SIZE, "\n %02d ", i); 1253 result.append(buffer); 1254 result.append(mNewParameters[i]); 1255 } 1256 1257 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1258 result.append(buffer); 1259 snprintf(buffer, SIZE, " Index event param\n"); 1260 result.append(buffer); 1261 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1262 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1263 result.append(buffer); 1264 } 1265 result.append("\n"); 1266 1267 write(fd, result.string(), result.size()); 1268 1269 if (locked) { 1270 mLock.unlock(); 1271 } 1272 return NO_ERROR; 1273} 1274 1275status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1276{ 1277 const size_t SIZE = 256; 1278 char buffer[SIZE]; 1279 String8 result; 1280 1281 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1282 write(fd, buffer, strlen(buffer)); 1283 1284 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1285 sp<EffectChain> chain = mEffectChains[i]; 1286 if (chain != 0) { 1287 chain->dump(fd, args); 1288 } 1289 } 1290 return NO_ERROR; 1291} 1292 1293void AudioFlinger::ThreadBase::acquireWakeLock() 1294{ 1295 Mutex::Autolock _l(mLock); 1296 acquireWakeLock_l(); 1297} 1298 1299void AudioFlinger::ThreadBase::acquireWakeLock_l() 1300{ 1301 if (mPowerManager == 0) { 1302 // use checkService() to avoid blocking if power service is not up yet 1303 sp<IBinder> binder = 1304 defaultServiceManager()->checkService(String16("power")); 1305 if (binder == 0) { 1306 ALOGW("Thread %s cannot connect to the power manager service", mName); 1307 } else { 1308 mPowerManager = interface_cast<IPowerManager>(binder); 1309 binder->linkToDeath(mDeathRecipient); 1310 } 1311 } 1312 if (mPowerManager != 0) { 1313 sp<IBinder> binder = new BBinder(); 1314 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1315 binder, 1316 String16(mName)); 1317 if (status == NO_ERROR) { 1318 mWakeLockToken = binder; 1319 } 1320 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1321 } 1322} 1323 1324void AudioFlinger::ThreadBase::releaseWakeLock() 1325{ 1326 Mutex::Autolock _l(mLock); 1327 releaseWakeLock_l(); 1328} 1329 1330void AudioFlinger::ThreadBase::releaseWakeLock_l() 1331{ 1332 if (mWakeLockToken != 0) { 1333 ALOGV("releaseWakeLock_l() %s", mName); 1334 if (mPowerManager != 0) { 1335 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1336 } 1337 mWakeLockToken.clear(); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::clearPowerManager() 1342{ 1343 Mutex::Autolock _l(mLock); 1344 releaseWakeLock_l(); 1345 mPowerManager.clear(); 1346} 1347 1348void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1349{ 1350 sp<ThreadBase> thread = mThread.promote(); 1351 if (thread != 0) { 1352 thread->clearPowerManager(); 1353 } 1354 ALOGW("power manager service died !!!"); 1355} 1356 1357void AudioFlinger::ThreadBase::setEffectSuspended( 1358 const effect_uuid_t *type, bool suspend, int sessionId) 1359{ 1360 Mutex::Autolock _l(mLock); 1361 setEffectSuspended_l(type, suspend, sessionId); 1362} 1363 1364void AudioFlinger::ThreadBase::setEffectSuspended_l( 1365 const effect_uuid_t *type, bool suspend, int sessionId) 1366{ 1367 sp<EffectChain> chain = getEffectChain_l(sessionId); 1368 if (chain != 0) { 1369 if (type != NULL) { 1370 chain->setEffectSuspended_l(type, suspend); 1371 } else { 1372 chain->setEffectSuspendedAll_l(suspend); 1373 } 1374 } 1375 1376 updateSuspendedSessions_l(type, suspend, sessionId); 1377} 1378 1379void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1380{ 1381 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1382 if (index < 0) { 1383 return; 1384 } 1385 1386 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1387 mSuspendedSessions.editValueAt(index); 1388 1389 for (size_t i = 0; i < sessionEffects.size(); i++) { 1390 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1391 for (int j = 0; j < desc->mRefCount; j++) { 1392 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1393 chain->setEffectSuspendedAll_l(true); 1394 } else { 1395 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1396 desc->mType.timeLow); 1397 chain->setEffectSuspended_l(&desc->mType, true); 1398 } 1399 } 1400 } 1401} 1402 1403void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1404 bool suspend, 1405 int sessionId) 1406{ 1407 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1408 1409 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1410 1411 if (suspend) { 1412 if (index >= 0) { 1413 sessionEffects = mSuspendedSessions.editValueAt(index); 1414 } else { 1415 mSuspendedSessions.add(sessionId, sessionEffects); 1416 } 1417 } else { 1418 if (index < 0) { 1419 return; 1420 } 1421 sessionEffects = mSuspendedSessions.editValueAt(index); 1422 } 1423 1424 1425 int key = EffectChain::kKeyForSuspendAll; 1426 if (type != NULL) { 1427 key = type->timeLow; 1428 } 1429 index = sessionEffects.indexOfKey(key); 1430 1431 sp<SuspendedSessionDesc> desc; 1432 if (suspend) { 1433 if (index >= 0) { 1434 desc = sessionEffects.valueAt(index); 1435 } else { 1436 desc = new SuspendedSessionDesc(); 1437 if (type != NULL) { 1438 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1439 } 1440 sessionEffects.add(key, desc); 1441 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1442 } 1443 desc->mRefCount++; 1444 } else { 1445 if (index < 0) { 1446 return; 1447 } 1448 desc = sessionEffects.valueAt(index); 1449 if (--desc->mRefCount == 0) { 1450 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1451 sessionEffects.removeItemsAt(index); 1452 if (sessionEffects.isEmpty()) { 1453 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1454 sessionId); 1455 mSuspendedSessions.removeItem(sessionId); 1456 } 1457 } 1458 } 1459 if (!sessionEffects.isEmpty()) { 1460 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1461 } 1462} 1463 1464void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1465 bool enabled, 1466 int sessionId) 1467{ 1468 Mutex::Autolock _l(mLock); 1469 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1470} 1471 1472void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1473 bool enabled, 1474 int sessionId) 1475{ 1476 if (mType != RECORD) { 1477 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1478 // another session. This gives the priority to well behaved effect control panels 1479 // and applications not using global effects. 1480 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1481 // global effects 1482 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1483 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1484 } 1485 } 1486 1487 sp<EffectChain> chain = getEffectChain_l(sessionId); 1488 if (chain != 0) { 1489 chain->checkSuspendOnEffectEnabled(effect, enabled); 1490 } 1491} 1492 1493// ---------------------------------------------------------------------------- 1494 1495AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1496 AudioStreamOut* output, 1497 audio_io_handle_t id, 1498 uint32_t device, 1499 type_t type) 1500 : ThreadBase(audioFlinger, id, device, type), 1501 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1502 // Assumes constructor is called by AudioFlinger with it's mLock held, 1503 // but it would be safer to explicitly pass initial masterMute as parameter 1504 mMasterMute(audioFlinger->masterMute_l()), 1505 // mStreamTypes[] initialized in constructor body 1506 mOutput(output), 1507 // Assumes constructor is called by AudioFlinger with it's mLock held, 1508 // but it would be safer to explicitly pass initial masterVolume as parameter 1509 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1510 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1511 mMixerStatus(MIXER_IDLE), 1512 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1513 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1514 mScreenState(gScreenState), 1515 // index 0 is reserved for normal mixer's submix 1516 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1517{ 1518 snprintf(mName, kNameLength, "AudioOut_%X", id); 1519 1520 readOutputParameters(); 1521 1522 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1523 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1524 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1525 stream = (audio_stream_type_t) (stream + 1)) { 1526 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1527 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1528 } 1529 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1530 // because mAudioFlinger doesn't have one to copy from 1531} 1532 1533AudioFlinger::PlaybackThread::~PlaybackThread() 1534{ 1535 delete [] mMixBuffer; 1536} 1537 1538status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1539{ 1540 dumpInternals(fd, args); 1541 dumpTracks(fd, args); 1542 dumpEffectChains(fd, args); 1543 return NO_ERROR; 1544} 1545 1546status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1547{ 1548 const size_t SIZE = 256; 1549 char buffer[SIZE]; 1550 String8 result; 1551 1552 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1553 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1554 const stream_type_t *st = &mStreamTypes[i]; 1555 if (i > 0) { 1556 result.appendFormat(", "); 1557 } 1558 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1559 if (st->mute) { 1560 result.append("M"); 1561 } 1562 } 1563 result.append("\n"); 1564 write(fd, result.string(), result.length()); 1565 result.clear(); 1566 1567 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mTracks.size(); ++i) { 1571 sp<Track> track = mTracks[i]; 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 1578 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1579 result.append(buffer); 1580 Track::appendDumpHeader(result); 1581 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1582 sp<Track> track = mActiveTracks[i].promote(); 1583 if (track != 0) { 1584 track->dump(buffer, SIZE); 1585 result.append(buffer); 1586 } 1587 } 1588 write(fd, result.string(), result.size()); 1589 1590 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1591 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1592 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1593 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1594 1595 return NO_ERROR; 1596} 1597 1598status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1599{ 1600 const size_t SIZE = 256; 1601 char buffer[SIZE]; 1602 String8 result; 1603 1604 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1605 result.append(buffer); 1606 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1607 result.append(buffer); 1608 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1609 result.append(buffer); 1610 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1611 result.append(buffer); 1612 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1613 result.append(buffer); 1614 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1615 result.append(buffer); 1616 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1617 result.append(buffer); 1618 write(fd, result.string(), result.size()); 1619 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1620 1621 dumpBase(fd, args); 1622 1623 return NO_ERROR; 1624} 1625 1626// Thread virtuals 1627status_t AudioFlinger::PlaybackThread::readyToRun() 1628{ 1629 status_t status = initCheck(); 1630 if (status == NO_ERROR) { 1631 ALOGI("AudioFlinger's thread %p ready to run", this); 1632 } else { 1633 ALOGE("No working audio driver found."); 1634 } 1635 return status; 1636} 1637 1638void AudioFlinger::PlaybackThread::onFirstRef() 1639{ 1640 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1641} 1642 1643// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1644sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1645 const sp<AudioFlinger::Client>& client, 1646 audio_stream_type_t streamType, 1647 uint32_t sampleRate, 1648 audio_format_t format, 1649 uint32_t channelMask, 1650 int frameCount, 1651 const sp<IMemory>& sharedBuffer, 1652 int sessionId, 1653 IAudioFlinger::track_flags_t flags, 1654 pid_t tid, 1655 status_t *status) 1656{ 1657 sp<Track> track; 1658 status_t lStatus; 1659 1660 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1661 1662 // client expresses a preference for FAST, but we get the final say 1663 if (flags & IAudioFlinger::TRACK_FAST) { 1664 if ( 1665 // not timed 1666 (!isTimed) && 1667 // either of these use cases: 1668 ( 1669 // use case 1: shared buffer with any frame count 1670 ( 1671 (sharedBuffer != 0) 1672 ) || 1673 // use case 2: callback handler and frame count is default or at least as large as HAL 1674 ( 1675 (tid != -1) && 1676 ((frameCount == 0) || 1677 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1678 ) 1679 ) && 1680 // PCM data 1681 audio_is_linear_pcm(format) && 1682 // mono or stereo 1683 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1684 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1685#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1686 // hardware sample rate 1687 (sampleRate == mSampleRate) && 1688#endif 1689 // normal mixer has an associated fast mixer 1690 hasFastMixer() && 1691 // there are sufficient fast track slots available 1692 (mFastTrackAvailMask != 0) 1693 // FIXME test that MixerThread for this fast track has a capable output HAL 1694 // FIXME add a permission test also? 1695 ) { 1696 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1697 if (frameCount == 0) { 1698 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1699 } 1700 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1701 frameCount, mFrameCount); 1702 } else { 1703 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1704 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1705 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1706 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1707 audio_is_linear_pcm(format), 1708 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1709 flags &= ~IAudioFlinger::TRACK_FAST; 1710 // For compatibility with AudioTrack calculation, buffer depth is forced 1711 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1712 // This is probably too conservative, but legacy application code may depend on it. 1713 // If you change this calculation, also review the start threshold which is related. 1714 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1715 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1716 if (minBufCount < 2) { 1717 minBufCount = 2; 1718 } 1719 int minFrameCount = mNormalFrameCount * minBufCount; 1720 if (frameCount < minFrameCount) { 1721 frameCount = minFrameCount; 1722 } 1723 } 1724 } 1725 1726 if (mType == DIRECT) { 1727 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1728 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1729 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1730 "for output %p with format %d", 1731 sampleRate, format, channelMask, mOutput, mFormat); 1732 lStatus = BAD_VALUE; 1733 goto Exit; 1734 } 1735 } 1736 } else { 1737 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1738 if (sampleRate > mSampleRate*2) { 1739 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1740 lStatus = BAD_VALUE; 1741 goto Exit; 1742 } 1743 } 1744 1745 lStatus = initCheck(); 1746 if (lStatus != NO_ERROR) { 1747 ALOGE("Audio driver not initialized."); 1748 goto Exit; 1749 } 1750 1751 { // scope for mLock 1752 Mutex::Autolock _l(mLock); 1753 1754 // all tracks in same audio session must share the same routing strategy otherwise 1755 // conflicts will happen when tracks are moved from one output to another by audio policy 1756 // manager 1757 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1758 for (size_t i = 0; i < mTracks.size(); ++i) { 1759 sp<Track> t = mTracks[i]; 1760 if (t != 0 && !t->isOutputTrack()) { 1761 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1762 if (sessionId == t->sessionId() && strategy != actual) { 1763 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1764 strategy, actual); 1765 lStatus = BAD_VALUE; 1766 goto Exit; 1767 } 1768 } 1769 } 1770 1771 if (!isTimed) { 1772 track = new Track(this, client, streamType, sampleRate, format, 1773 channelMask, frameCount, sharedBuffer, sessionId, flags); 1774 } else { 1775 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1776 channelMask, frameCount, sharedBuffer, sessionId); 1777 } 1778 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1779 lStatus = NO_MEMORY; 1780 goto Exit; 1781 } 1782 mTracks.add(track); 1783 1784 sp<EffectChain> chain = getEffectChain_l(sessionId); 1785 if (chain != 0) { 1786 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1787 track->setMainBuffer(chain->inBuffer()); 1788 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1789 chain->incTrackCnt(); 1790 } 1791 } 1792 1793#ifdef HAVE_REQUEST_PRIORITY 1794 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1795 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1796 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1797 // so ask activity manager to do this on our behalf 1798 int err = requestPriority(callingPid, tid, 1); 1799 if (err != 0) { 1800 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1801 1, callingPid, tid, err); 1802 } 1803 } 1804#endif 1805 1806 lStatus = NO_ERROR; 1807 1808Exit: 1809 if (status) { 1810 *status = lStatus; 1811 } 1812 return track; 1813} 1814 1815uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1816{ 1817 if (mFastMixer != NULL) { 1818 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1819 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1820 } 1821 return latency; 1822} 1823 1824uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1825{ 1826 return latency; 1827} 1828 1829uint32_t AudioFlinger::PlaybackThread::latency() const 1830{ 1831 Mutex::Autolock _l(mLock); 1832 return latency_l(); 1833} 1834uint32_t AudioFlinger::PlaybackThread::latency_l() const 1835{ 1836 if (initCheck() == NO_ERROR) { 1837 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1838 } else { 1839 return 0; 1840 } 1841} 1842 1843void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 mMasterVolume = value; 1847} 1848 1849void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 setMasterMute_l(muted); 1853} 1854 1855void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1856{ 1857 Mutex::Autolock _l(mLock); 1858 mStreamTypes[stream].volume = value; 1859} 1860 1861void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1862{ 1863 Mutex::Autolock _l(mLock); 1864 mStreamTypes[stream].mute = muted; 1865} 1866 1867float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1868{ 1869 Mutex::Autolock _l(mLock); 1870 return mStreamTypes[stream].volume; 1871} 1872 1873// addTrack_l() must be called with ThreadBase::mLock held 1874status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1875{ 1876 status_t status = ALREADY_EXISTS; 1877 1878 // set retry count for buffer fill 1879 track->mRetryCount = kMaxTrackStartupRetries; 1880 if (mActiveTracks.indexOf(track) < 0) { 1881 // the track is newly added, make sure it fills up all its 1882 // buffers before playing. This is to ensure the client will 1883 // effectively get the latency it requested. 1884 track->mFillingUpStatus = Track::FS_FILLING; 1885 track->mResetDone = false; 1886 track->mPresentationCompleteFrames = 0; 1887 mActiveTracks.add(track); 1888 if (track->mainBuffer() != mMixBuffer) { 1889 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1890 if (chain != 0) { 1891 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1892 chain->incActiveTrackCnt(); 1893 } 1894 } 1895 1896 status = NO_ERROR; 1897 } 1898 1899 ALOGV("mWaitWorkCV.broadcast"); 1900 mWaitWorkCV.broadcast(); 1901 1902 return status; 1903} 1904 1905// destroyTrack_l() must be called with ThreadBase::mLock held 1906void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1907{ 1908 track->mState = TrackBase::TERMINATED; 1909 // active tracks are removed by threadLoop() 1910 if (mActiveTracks.indexOf(track) < 0) { 1911 removeTrack_l(track); 1912 } 1913} 1914 1915void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1916{ 1917 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1918 mTracks.remove(track); 1919 deleteTrackName_l(track->name()); 1920 // redundant as track is about to be destroyed, for dumpsys only 1921 track->mName = -1; 1922 if (track->isFastTrack()) { 1923 int index = track->mFastIndex; 1924 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1925 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1926 mFastTrackAvailMask |= 1 << index; 1927 // redundant as track is about to be destroyed, for dumpsys only 1928 track->mFastIndex = -1; 1929 } 1930 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1931 if (chain != 0) { 1932 chain->decTrackCnt(); 1933 } 1934} 1935 1936String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1937{ 1938 String8 out_s8 = String8(""); 1939 char *s; 1940 1941 Mutex::Autolock _l(mLock); 1942 if (initCheck() != NO_ERROR) { 1943 return out_s8; 1944 } 1945 1946 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1947 out_s8 = String8(s); 1948 free(s); 1949 return out_s8; 1950} 1951 1952// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1953void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1954 AudioSystem::OutputDescriptor desc; 1955 void *param2 = NULL; 1956 1957 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1958 1959 switch (event) { 1960 case AudioSystem::OUTPUT_OPENED: 1961 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1962 desc.channels = mChannelMask; 1963 desc.samplingRate = mSampleRate; 1964 desc.format = mFormat; 1965 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1966 desc.latency = latency(); 1967 param2 = &desc; 1968 break; 1969 1970 case AudioSystem::STREAM_CONFIG_CHANGED: 1971 param2 = ¶m; 1972 case AudioSystem::OUTPUT_CLOSED: 1973 default: 1974 break; 1975 } 1976 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1977} 1978 1979void AudioFlinger::PlaybackThread::readOutputParameters() 1980{ 1981 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1982 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1983 mChannelCount = (uint16_t)popcount(mChannelMask); 1984 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1985 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1986 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1987 if (mFrameCount & 15) { 1988 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1989 mFrameCount); 1990 } 1991 1992 // Calculate size of normal mix buffer relative to the HAL output buffer size 1993 double multiplier = 1.0; 1994 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1995 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1996 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1997 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1998 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1999 maxNormalFrameCount = maxNormalFrameCount & ~15; 2000 if (maxNormalFrameCount < minNormalFrameCount) { 2001 maxNormalFrameCount = minNormalFrameCount; 2002 } 2003 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2004 if (multiplier <= 1.0) { 2005 multiplier = 1.0; 2006 } else if (multiplier <= 2.0) { 2007 if (2 * mFrameCount <= maxNormalFrameCount) { 2008 multiplier = 2.0; 2009 } else { 2010 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2011 } 2012 } else { 2013 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2014 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2015 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2016 // FIXME this rounding up should not be done if no HAL SRC 2017 uint32_t truncMult = (uint32_t) multiplier; 2018 if ((truncMult & 1)) { 2019 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2020 ++truncMult; 2021 } 2022 } 2023 multiplier = (double) truncMult; 2024 } 2025 } 2026 mNormalFrameCount = multiplier * mFrameCount; 2027 // round up to nearest 16 frames to satisfy AudioMixer 2028 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2029 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2030 2031 delete[] mMixBuffer; 2032 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2033 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2034 2035 // force reconfiguration of effect chains and engines to take new buffer size and audio 2036 // parameters into account 2037 // Note that mLock is not held when readOutputParameters() is called from the constructor 2038 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2039 // matter. 2040 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2041 Vector< sp<EffectChain> > effectChains = mEffectChains; 2042 for (size_t i = 0; i < effectChains.size(); i ++) { 2043 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2044 } 2045} 2046 2047 2048status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2049{ 2050 if (halFrames == NULL || dspFrames == NULL) { 2051 return BAD_VALUE; 2052 } 2053 Mutex::Autolock _l(mLock); 2054 if (initCheck() != NO_ERROR) { 2055 return INVALID_OPERATION; 2056 } 2057 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2058 2059 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2060} 2061 2062uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2063{ 2064 Mutex::Autolock _l(mLock); 2065 uint32_t result = 0; 2066 if (getEffectChain_l(sessionId) != 0) { 2067 result = EFFECT_SESSION; 2068 } 2069 2070 for (size_t i = 0; i < mTracks.size(); ++i) { 2071 sp<Track> track = mTracks[i]; 2072 if (sessionId == track->sessionId() && 2073 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2074 result |= TRACK_SESSION; 2075 break; 2076 } 2077 } 2078 2079 return result; 2080} 2081 2082uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2083{ 2084 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2085 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2086 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2087 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2088 } 2089 for (size_t i = 0; i < mTracks.size(); i++) { 2090 sp<Track> track = mTracks[i]; 2091 if (sessionId == track->sessionId() && 2092 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2093 return AudioSystem::getStrategyForStream(track->streamType()); 2094 } 2095 } 2096 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2097} 2098 2099 2100AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2101{ 2102 Mutex::Autolock _l(mLock); 2103 return mOutput; 2104} 2105 2106AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2107{ 2108 Mutex::Autolock _l(mLock); 2109 AudioStreamOut *output = mOutput; 2110 mOutput = NULL; 2111 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2112 // must push a NULL and wait for ack 2113 mOutputSink.clear(); 2114 mPipeSink.clear(); 2115 mNormalSink.clear(); 2116 return output; 2117} 2118 2119// this method must always be called either with ThreadBase mLock held or inside the thread loop 2120audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2121{ 2122 if (mOutput == NULL) { 2123 return NULL; 2124 } 2125 return &mOutput->stream->common; 2126} 2127 2128uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2129{ 2130 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2131} 2132 2133status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2134{ 2135 if (!isValidSyncEvent(event)) { 2136 return BAD_VALUE; 2137 } 2138 2139 Mutex::Autolock _l(mLock); 2140 2141 for (size_t i = 0; i < mTracks.size(); ++i) { 2142 sp<Track> track = mTracks[i]; 2143 if (event->triggerSession() == track->sessionId()) { 2144 track->setSyncEvent(event); 2145 return NO_ERROR; 2146 } 2147 } 2148 2149 return NAME_NOT_FOUND; 2150} 2151 2152bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2153{ 2154 switch (event->type()) { 2155 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2156 return true; 2157 default: 2158 break; 2159 } 2160 return false; 2161} 2162 2163void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2164{ 2165 size_t count = tracksToRemove.size(); 2166 if (CC_UNLIKELY(count)) { 2167 for (size_t i = 0 ; i < count ; i++) { 2168 const sp<Track>& track = tracksToRemove.itemAt(i); 2169 if ((track->sharedBuffer() != 0) && 2170 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2171 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2172 } 2173 } 2174 } 2175 2176} 2177 2178// ---------------------------------------------------------------------------- 2179 2180AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2181 audio_io_handle_t id, uint32_t device, type_t type) 2182 : PlaybackThread(audioFlinger, output, id, device, type), 2183 // mAudioMixer below 2184#ifdef SOAKER 2185 mSoaker(NULL), 2186#endif 2187 // mFastMixer below 2188 mFastMixerFutex(0) 2189 // mOutputSink below 2190 // mPipeSink below 2191 // mNormalSink below 2192{ 2193 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2194 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2195 "mFrameCount=%d, mNormalFrameCount=%d", 2196 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2197 mNormalFrameCount); 2198 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2199 2200 // FIXME - Current mixer implementation only supports stereo output 2201 if (mChannelCount == 1) { 2202 ALOGE("Invalid audio hardware channel count"); 2203 } 2204 2205 // create an NBAIO sink for the HAL output stream, and negotiate 2206 mOutputSink = new AudioStreamOutSink(output->stream); 2207 size_t numCounterOffers = 0; 2208 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2209 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2210 ALOG_ASSERT(index == 0); 2211 2212 // initialize fast mixer depending on configuration 2213 bool initFastMixer; 2214 switch (kUseFastMixer) { 2215 case FastMixer_Never: 2216 initFastMixer = false; 2217 break; 2218 case FastMixer_Always: 2219 initFastMixer = true; 2220 break; 2221 case FastMixer_Static: 2222 case FastMixer_Dynamic: 2223 initFastMixer = mFrameCount < mNormalFrameCount; 2224 break; 2225 } 2226 if (initFastMixer) { 2227 2228 // create a MonoPipe to connect our submix to FastMixer 2229 NBAIO_Format format = mOutputSink->format(); 2230 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2231 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2232 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2233 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2234 const NBAIO_Format offers[1] = {format}; 2235 size_t numCounterOffers = 0; 2236 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2237 ALOG_ASSERT(index == 0); 2238 monoPipe->setAvgFrames((mScreenState & 1) ? 2239 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2240 mPipeSink = monoPipe; 2241 2242#ifdef TEE_SINK_FRAMES 2243 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2244 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2245 numCounterOffers = 0; 2246 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2247 ALOG_ASSERT(index == 0); 2248 mTeeSink = teeSink; 2249 PipeReader *teeSource = new PipeReader(*teeSink); 2250 numCounterOffers = 0; 2251 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2252 ALOG_ASSERT(index == 0); 2253 mTeeSource = teeSource; 2254#endif 2255 2256#ifdef SOAKER 2257 // create a soaker as workaround for governor issues 2258 mSoaker = new Soaker(); 2259 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2260 mSoaker->run("Soaker", PRIORITY_LOWEST); 2261#endif 2262 2263 // create fast mixer and configure it initially with just one fast track for our submix 2264 mFastMixer = new FastMixer(); 2265 FastMixerStateQueue *sq = mFastMixer->sq(); 2266#ifdef STATE_QUEUE_DUMP 2267 sq->setObserverDump(&mStateQueueObserverDump); 2268 sq->setMutatorDump(&mStateQueueMutatorDump); 2269#endif 2270 FastMixerState *state = sq->begin(); 2271 FastTrack *fastTrack = &state->mFastTracks[0]; 2272 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2273 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2274 fastTrack->mVolumeProvider = NULL; 2275 fastTrack->mGeneration++; 2276 state->mFastTracksGen++; 2277 state->mTrackMask = 1; 2278 // fast mixer will use the HAL output sink 2279 state->mOutputSink = mOutputSink.get(); 2280 state->mOutputSinkGen++; 2281 state->mFrameCount = mFrameCount; 2282 state->mCommand = FastMixerState::COLD_IDLE; 2283 // already done in constructor initialization list 2284 //mFastMixerFutex = 0; 2285 state->mColdFutexAddr = &mFastMixerFutex; 2286 state->mColdGen++; 2287 state->mDumpState = &mFastMixerDumpState; 2288 state->mTeeSink = mTeeSink.get(); 2289 sq->end(); 2290 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2291 2292 // start the fast mixer 2293 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2294#ifdef HAVE_REQUEST_PRIORITY 2295 pid_t tid = mFastMixer->getTid(); 2296 int err = requestPriority(getpid_cached, tid, 2); 2297 if (err != 0) { 2298 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2299 2, getpid_cached, tid, err); 2300 } 2301#endif 2302 2303#ifdef AUDIO_WATCHDOG 2304 // create and start the watchdog 2305 mAudioWatchdog = new AudioWatchdog(); 2306 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2307 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2308 tid = mAudioWatchdog->getTid(); 2309 err = requestPriority(getpid_cached, tid, 1); 2310 if (err != 0) { 2311 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2312 1, getpid_cached, tid, err); 2313 } 2314#endif 2315 2316 } else { 2317 mFastMixer = NULL; 2318 } 2319 2320 switch (kUseFastMixer) { 2321 case FastMixer_Never: 2322 case FastMixer_Dynamic: 2323 mNormalSink = mOutputSink; 2324 break; 2325 case FastMixer_Always: 2326 mNormalSink = mPipeSink; 2327 break; 2328 case FastMixer_Static: 2329 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2330 break; 2331 } 2332} 2333 2334AudioFlinger::MixerThread::~MixerThread() 2335{ 2336 if (mFastMixer != NULL) { 2337 FastMixerStateQueue *sq = mFastMixer->sq(); 2338 FastMixerState *state = sq->begin(); 2339 if (state->mCommand == FastMixerState::COLD_IDLE) { 2340 int32_t old = android_atomic_inc(&mFastMixerFutex); 2341 if (old == -1) { 2342 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2343 } 2344 } 2345 state->mCommand = FastMixerState::EXIT; 2346 sq->end(); 2347 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2348 mFastMixer->join(); 2349 // Though the fast mixer thread has exited, it's state queue is still valid. 2350 // We'll use that extract the final state which contains one remaining fast track 2351 // corresponding to our sub-mix. 2352 state = sq->begin(); 2353 ALOG_ASSERT(state->mTrackMask == 1); 2354 FastTrack *fastTrack = &state->mFastTracks[0]; 2355 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2356 delete fastTrack->mBufferProvider; 2357 sq->end(false /*didModify*/); 2358 delete mFastMixer; 2359#ifdef SOAKER 2360 if (mSoaker != NULL) { 2361 mSoaker->requestExitAndWait(); 2362 } 2363 delete mSoaker; 2364#endif 2365 if (mAudioWatchdog != 0) { 2366 mAudioWatchdog->requestExit(); 2367 mAudioWatchdog->requestExitAndWait(); 2368 mAudioWatchdog.clear(); 2369 } 2370 } 2371 delete mAudioMixer; 2372} 2373 2374class CpuStats { 2375public: 2376 CpuStats(); 2377 void sample(const String8 &title); 2378#ifdef DEBUG_CPU_USAGE 2379private: 2380 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2381 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2382 2383 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2384 2385 int mCpuNum; // thread's current CPU number 2386 int mCpukHz; // frequency of thread's current CPU in kHz 2387#endif 2388}; 2389 2390CpuStats::CpuStats() 2391#ifdef DEBUG_CPU_USAGE 2392 : mCpuNum(-1), mCpukHz(-1) 2393#endif 2394{ 2395} 2396 2397void CpuStats::sample(const String8 &title) { 2398#ifdef DEBUG_CPU_USAGE 2399 // get current thread's delta CPU time in wall clock ns 2400 double wcNs; 2401 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2402 2403 // record sample for wall clock statistics 2404 if (valid) { 2405 mWcStats.sample(wcNs); 2406 } 2407 2408 // get the current CPU number 2409 int cpuNum = sched_getcpu(); 2410 2411 // get the current CPU frequency in kHz 2412 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2413 2414 // check if either CPU number or frequency changed 2415 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2416 mCpuNum = cpuNum; 2417 mCpukHz = cpukHz; 2418 // ignore sample for purposes of cycles 2419 valid = false; 2420 } 2421 2422 // if no change in CPU number or frequency, then record sample for cycle statistics 2423 if (valid && mCpukHz > 0) { 2424 double cycles = wcNs * cpukHz * 0.000001; 2425 mHzStats.sample(cycles); 2426 } 2427 2428 unsigned n = mWcStats.n(); 2429 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2430 if ((n & 127) == 1) { 2431 long long elapsed = mCpuUsage.elapsed(); 2432 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2433 double perLoop = elapsed / (double) n; 2434 double perLoop100 = perLoop * 0.01; 2435 double perLoop1k = perLoop * 0.001; 2436 double mean = mWcStats.mean(); 2437 double stddev = mWcStats.stddev(); 2438 double minimum = mWcStats.minimum(); 2439 double maximum = mWcStats.maximum(); 2440 double meanCycles = mHzStats.mean(); 2441 double stddevCycles = mHzStats.stddev(); 2442 double minCycles = mHzStats.minimum(); 2443 double maxCycles = mHzStats.maximum(); 2444 mCpuUsage.resetElapsed(); 2445 mWcStats.reset(); 2446 mHzStats.reset(); 2447 ALOGD("CPU usage for %s over past %.1f secs\n" 2448 " (%u mixer loops at %.1f mean ms per loop):\n" 2449 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2450 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2451 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2452 title.string(), 2453 elapsed * .000000001, n, perLoop * .000001, 2454 mean * .001, 2455 stddev * .001, 2456 minimum * .001, 2457 maximum * .001, 2458 mean / perLoop100, 2459 stddev / perLoop100, 2460 minimum / perLoop100, 2461 maximum / perLoop100, 2462 meanCycles / perLoop1k, 2463 stddevCycles / perLoop1k, 2464 minCycles / perLoop1k, 2465 maxCycles / perLoop1k); 2466 2467 } 2468 } 2469#endif 2470}; 2471 2472void AudioFlinger::PlaybackThread::checkSilentMode_l() 2473{ 2474 if (!mMasterMute) { 2475 char value[PROPERTY_VALUE_MAX]; 2476 if (property_get("ro.audio.silent", value, "0") > 0) { 2477 char *endptr; 2478 unsigned long ul = strtoul(value, &endptr, 0); 2479 if (*endptr == '\0' && ul != 0) { 2480 ALOGD("Silence is golden"); 2481 // The setprop command will not allow a property to be changed after 2482 // the first time it is set, so we don't have to worry about un-muting. 2483 setMasterMute_l(true); 2484 } 2485 } 2486 } 2487} 2488 2489bool AudioFlinger::PlaybackThread::threadLoop() 2490{ 2491 Vector< sp<Track> > tracksToRemove; 2492 2493 standbyTime = systemTime(); 2494 2495 // MIXER 2496 nsecs_t lastWarning = 0; 2497if (mType == MIXER) { 2498 longStandbyExit = false; 2499} 2500 2501 // DUPLICATING 2502 // FIXME could this be made local to while loop? 2503 writeFrames = 0; 2504 2505 cacheParameters_l(); 2506 sleepTime = idleSleepTime; 2507 2508if (mType == MIXER) { 2509 sleepTimeShift = 0; 2510} 2511 2512 CpuStats cpuStats; 2513 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2514 2515 acquireWakeLock(); 2516 2517 while (!exitPending()) 2518 { 2519 cpuStats.sample(myName); 2520 2521 Vector< sp<EffectChain> > effectChains; 2522 2523 processConfigEvents(); 2524 2525 { // scope for mLock 2526 2527 Mutex::Autolock _l(mLock); 2528 2529 if (checkForNewParameters_l()) { 2530 cacheParameters_l(); 2531 } 2532 2533 saveOutputTracks(); 2534 2535 // put audio hardware into standby after short delay 2536 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2537 mSuspended > 0)) { 2538 if (!mStandby) { 2539 2540 threadLoop_standby(); 2541 2542 mStandby = true; 2543 mBytesWritten = 0; 2544 } 2545 2546 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2547 // we're about to wait, flush the binder command buffer 2548 IPCThreadState::self()->flushCommands(); 2549 2550 clearOutputTracks(); 2551 2552 if (exitPending()) break; 2553 2554 releaseWakeLock_l(); 2555 // wait until we have something to do... 2556 ALOGV("%s going to sleep", myName.string()); 2557 mWaitWorkCV.wait(mLock); 2558 ALOGV("%s waking up", myName.string()); 2559 acquireWakeLock_l(); 2560 2561 mMixerStatus = MIXER_IDLE; 2562 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2563 2564 checkSilentMode_l(); 2565 2566 standbyTime = systemTime() + standbyDelay; 2567 sleepTime = idleSleepTime; 2568 if (mType == MIXER) { 2569 sleepTimeShift = 0; 2570 } 2571 2572 continue; 2573 } 2574 } 2575 2576 // mMixerStatusIgnoringFastTracks is also updated internally 2577 mMixerStatus = prepareTracks_l(&tracksToRemove); 2578 2579 // prevent any changes in effect chain list and in each effect chain 2580 // during mixing and effect process as the audio buffers could be deleted 2581 // or modified if an effect is created or deleted 2582 lockEffectChains_l(effectChains); 2583 } 2584 2585 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2586 threadLoop_mix(); 2587 } else { 2588 threadLoop_sleepTime(); 2589 } 2590 2591 if (mSuspended > 0) { 2592 sleepTime = suspendSleepTimeUs(); 2593 } 2594 2595 // only process effects if we're going to write 2596 if (sleepTime == 0) { 2597 for (size_t i = 0; i < effectChains.size(); i ++) { 2598 effectChains[i]->process_l(); 2599 } 2600 } 2601 2602 // enable changes in effect chain 2603 unlockEffectChains(effectChains); 2604 2605 // sleepTime == 0 means we must write to audio hardware 2606 if (sleepTime == 0) { 2607 2608 threadLoop_write(); 2609 2610if (mType == MIXER) { 2611 // write blocked detection 2612 nsecs_t now = systemTime(); 2613 nsecs_t delta = now - mLastWriteTime; 2614 if (!mStandby && delta > maxPeriod) { 2615 mNumDelayedWrites++; 2616 if ((now - lastWarning) > kWarningThrottleNs) { 2617#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2618 ScopedTrace st(ATRACE_TAG, "underrun"); 2619#endif 2620 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2621 ns2ms(delta), mNumDelayedWrites, this); 2622 lastWarning = now; 2623 } 2624 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2625 // a different threshold. Or completely removed for what it is worth anyway... 2626 if (mStandby) { 2627 longStandbyExit = true; 2628 } 2629 } 2630} 2631 2632 mStandby = false; 2633 } else { 2634 usleep(sleepTime); 2635 } 2636 2637 // Finally let go of removed track(s), without the lock held 2638 // since we can't guarantee the destructors won't acquire that 2639 // same lock. This will also mutate and push a new fast mixer state. 2640 threadLoop_removeTracks(tracksToRemove); 2641 tracksToRemove.clear(); 2642 2643 // FIXME I don't understand the need for this here; 2644 // it was in the original code but maybe the 2645 // assignment in saveOutputTracks() makes this unnecessary? 2646 clearOutputTracks(); 2647 2648 // Effect chains will be actually deleted here if they were removed from 2649 // mEffectChains list during mixing or effects processing 2650 effectChains.clear(); 2651 2652 // FIXME Note that the above .clear() is no longer necessary since effectChains 2653 // is now local to this block, but will keep it for now (at least until merge done). 2654 } 2655 2656if (mType == MIXER || mType == DIRECT) { 2657 // put output stream into standby mode 2658 if (!mStandby) { 2659 mOutput->stream->common.standby(&mOutput->stream->common); 2660 } 2661} 2662if (mType == DUPLICATING) { 2663 // for DuplicatingThread, standby mode is handled by the outputTracks 2664} 2665 2666 releaseWakeLock(); 2667 2668 ALOGV("Thread %p type %d exiting", this, mType); 2669 return false; 2670} 2671 2672void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2673{ 2674 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2675} 2676 2677void AudioFlinger::MixerThread::threadLoop_write() 2678{ 2679 // FIXME we should only do one push per cycle; confirm this is true 2680 // Start the fast mixer if it's not already running 2681 if (mFastMixer != NULL) { 2682 FastMixerStateQueue *sq = mFastMixer->sq(); 2683 FastMixerState *state = sq->begin(); 2684 if (state->mCommand != FastMixerState::MIX_WRITE && 2685 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2686 if (state->mCommand == FastMixerState::COLD_IDLE) { 2687 int32_t old = android_atomic_inc(&mFastMixerFutex); 2688 if (old == -1) { 2689 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2690 } 2691 if (mAudioWatchdog != 0) { 2692 mAudioWatchdog->resume(); 2693 } 2694 } 2695 state->mCommand = FastMixerState::MIX_WRITE; 2696 sq->end(); 2697 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2698 if (kUseFastMixer == FastMixer_Dynamic) { 2699 mNormalSink = mPipeSink; 2700 } 2701 } else { 2702 sq->end(false /*didModify*/); 2703 } 2704 } 2705 PlaybackThread::threadLoop_write(); 2706} 2707 2708// shared by MIXER and DIRECT, overridden by DUPLICATING 2709void AudioFlinger::PlaybackThread::threadLoop_write() 2710{ 2711 // FIXME rewrite to reduce number of system calls 2712 mLastWriteTime = systemTime(); 2713 mInWrite = true; 2714 int bytesWritten; 2715 2716 // If an NBAIO sink is present, use it to write the normal mixer's submix 2717 if (mNormalSink != 0) { 2718#define mBitShift 2 // FIXME 2719 size_t count = mixBufferSize >> mBitShift; 2720#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2721 Tracer::traceBegin(ATRACE_TAG, "write"); 2722#endif 2723 // update the setpoint when gScreenState changes 2724 uint32_t screenState = gScreenState; 2725 if (screenState != mScreenState) { 2726 mScreenState = screenState; 2727 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2728 if (pipe != NULL) { 2729 pipe->setAvgFrames((mScreenState & 1) ? 2730 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2731 } 2732 } 2733 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2734#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2735 Tracer::traceEnd(ATRACE_TAG); 2736#endif 2737 if (framesWritten > 0) { 2738 bytesWritten = framesWritten << mBitShift; 2739 } else { 2740 bytesWritten = framesWritten; 2741 } 2742 // otherwise use the HAL / AudioStreamOut directly 2743 } else { 2744 // Direct output thread. 2745 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2746 } 2747 2748 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2749 mNumWrites++; 2750 mInWrite = false; 2751} 2752 2753void AudioFlinger::MixerThread::threadLoop_standby() 2754{ 2755 // Idle the fast mixer if it's currently running 2756 if (mFastMixer != NULL) { 2757 FastMixerStateQueue *sq = mFastMixer->sq(); 2758 FastMixerState *state = sq->begin(); 2759 if (!(state->mCommand & FastMixerState::IDLE)) { 2760 state->mCommand = FastMixerState::COLD_IDLE; 2761 state->mColdFutexAddr = &mFastMixerFutex; 2762 state->mColdGen++; 2763 mFastMixerFutex = 0; 2764 sq->end(); 2765 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2766 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2767 if (kUseFastMixer == FastMixer_Dynamic) { 2768 mNormalSink = mOutputSink; 2769 } 2770 if (mAudioWatchdog != 0) { 2771 mAudioWatchdog->pause(); 2772 } 2773 } else { 2774 sq->end(false /*didModify*/); 2775 } 2776 } 2777 PlaybackThread::threadLoop_standby(); 2778} 2779 2780// shared by MIXER and DIRECT, overridden by DUPLICATING 2781void AudioFlinger::PlaybackThread::threadLoop_standby() 2782{ 2783 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2784 mOutput->stream->common.standby(&mOutput->stream->common); 2785} 2786 2787void AudioFlinger::MixerThread::threadLoop_mix() 2788{ 2789 // obtain the presentation timestamp of the next output buffer 2790 int64_t pts; 2791 status_t status = INVALID_OPERATION; 2792 2793 if (NULL != mOutput->stream->get_next_write_timestamp) { 2794 status = mOutput->stream->get_next_write_timestamp( 2795 mOutput->stream, &pts); 2796 } 2797 2798 if (status != NO_ERROR) { 2799 pts = AudioBufferProvider::kInvalidPTS; 2800 } 2801 2802 // mix buffers... 2803 mAudioMixer->process(pts); 2804 // increase sleep time progressively when application underrun condition clears. 2805 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2806 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2807 // such that we would underrun the audio HAL. 2808 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2809 sleepTimeShift--; 2810 } 2811 sleepTime = 0; 2812 standbyTime = systemTime() + standbyDelay; 2813 //TODO: delay standby when effects have a tail 2814} 2815 2816void AudioFlinger::MixerThread::threadLoop_sleepTime() 2817{ 2818 // If no tracks are ready, sleep once for the duration of an output 2819 // buffer size, then write 0s to the output 2820 if (sleepTime == 0) { 2821 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2822 sleepTime = activeSleepTime >> sleepTimeShift; 2823 if (sleepTime < kMinThreadSleepTimeUs) { 2824 sleepTime = kMinThreadSleepTimeUs; 2825 } 2826 // reduce sleep time in case of consecutive application underruns to avoid 2827 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2828 // duration we would end up writing less data than needed by the audio HAL if 2829 // the condition persists. 2830 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2831 sleepTimeShift++; 2832 } 2833 } else { 2834 sleepTime = idleSleepTime; 2835 } 2836 } else if (mBytesWritten != 0 || 2837 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2838 memset (mMixBuffer, 0, mixBufferSize); 2839 sleepTime = 0; 2840 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2841 } 2842 // TODO add standby time extension fct of effect tail 2843} 2844 2845// prepareTracks_l() must be called with ThreadBase::mLock held 2846AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2847 Vector< sp<Track> > *tracksToRemove) 2848{ 2849 2850 mixer_state mixerStatus = MIXER_IDLE; 2851 // find out which tracks need to be processed 2852 size_t count = mActiveTracks.size(); 2853 size_t mixedTracks = 0; 2854 size_t tracksWithEffect = 0; 2855 // counts only _active_ fast tracks 2856 size_t fastTracks = 0; 2857 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2858 2859 float masterVolume = mMasterVolume; 2860 bool masterMute = mMasterMute; 2861 2862 if (masterMute) { 2863 masterVolume = 0; 2864 } 2865 // Delegate master volume control to effect in output mix effect chain if needed 2866 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2867 if (chain != 0) { 2868 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2869 chain->setVolume_l(&v, &v); 2870 masterVolume = (float)((v + (1 << 23)) >> 24); 2871 chain.clear(); 2872 } 2873 2874 // prepare a new state to push 2875 FastMixerStateQueue *sq = NULL; 2876 FastMixerState *state = NULL; 2877 bool didModify = false; 2878 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2879 if (mFastMixer != NULL) { 2880 sq = mFastMixer->sq(); 2881 state = sq->begin(); 2882 } 2883 2884 for (size_t i=0 ; i<count ; i++) { 2885 sp<Track> t = mActiveTracks[i].promote(); 2886 if (t == 0) continue; 2887 2888 // this const just means the local variable doesn't change 2889 Track* const track = t.get(); 2890 2891 // process fast tracks 2892 if (track->isFastTrack()) { 2893 2894 // It's theoretically possible (though unlikely) for a fast track to be created 2895 // and then removed within the same normal mix cycle. This is not a problem, as 2896 // the track never becomes active so it's fast mixer slot is never touched. 2897 // The converse, of removing an (active) track and then creating a new track 2898 // at the identical fast mixer slot within the same normal mix cycle, 2899 // is impossible because the slot isn't marked available until the end of each cycle. 2900 int j = track->mFastIndex; 2901 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2902 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2903 FastTrack *fastTrack = &state->mFastTracks[j]; 2904 2905 // Determine whether the track is currently in underrun condition, 2906 // and whether it had a recent underrun. 2907 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2908 FastTrackUnderruns underruns = ftDump->mUnderruns; 2909 uint32_t recentFull = (underruns.mBitFields.mFull - 2910 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2911 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2912 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2913 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2914 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2915 uint32_t recentUnderruns = recentPartial + recentEmpty; 2916 track->mObservedUnderruns = underruns; 2917 // don't count underruns that occur while stopping or pausing 2918 // or stopped which can occur when flush() is called while active 2919 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2920 track->mUnderrunCount += recentUnderruns; 2921 } 2922 2923 // This is similar to the state machine for normal tracks, 2924 // with a few modifications for fast tracks. 2925 bool isActive = true; 2926 switch (track->mState) { 2927 case TrackBase::STOPPING_1: 2928 // track stays active in STOPPING_1 state until first underrun 2929 if (recentUnderruns > 0) { 2930 track->mState = TrackBase::STOPPING_2; 2931 } 2932 break; 2933 case TrackBase::PAUSING: 2934 // ramp down is not yet implemented 2935 track->setPaused(); 2936 break; 2937 case TrackBase::RESUMING: 2938 // ramp up is not yet implemented 2939 track->mState = TrackBase::ACTIVE; 2940 break; 2941 case TrackBase::ACTIVE: 2942 if (recentFull > 0 || recentPartial > 0) { 2943 // track has provided at least some frames recently: reset retry count 2944 track->mRetryCount = kMaxTrackRetries; 2945 } 2946 if (recentUnderruns == 0) { 2947 // no recent underruns: stay active 2948 break; 2949 } 2950 // there has recently been an underrun of some kind 2951 if (track->sharedBuffer() == 0) { 2952 // were any of the recent underruns "empty" (no frames available)? 2953 if (recentEmpty == 0) { 2954 // no, then ignore the partial underruns as they are allowed indefinitely 2955 break; 2956 } 2957 // there has recently been an "empty" underrun: decrement the retry counter 2958 if (--(track->mRetryCount) > 0) { 2959 break; 2960 } 2961 // indicate to client process that the track was disabled because of underrun; 2962 // it will then automatically call start() when data is available 2963 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2964 // remove from active list, but state remains ACTIVE [confusing but true] 2965 isActive = false; 2966 break; 2967 } 2968 // fall through 2969 case TrackBase::STOPPING_2: 2970 case TrackBase::PAUSED: 2971 case TrackBase::TERMINATED: 2972 case TrackBase::STOPPED: 2973 case TrackBase::FLUSHED: // flush() while active 2974 // Check for presentation complete if track is inactive 2975 // We have consumed all the buffers of this track. 2976 // This would be incomplete if we auto-paused on underrun 2977 { 2978 size_t audioHALFrames = 2979 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2980 size_t framesWritten = 2981 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2982 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2983 // track stays in active list until presentation is complete 2984 break; 2985 } 2986 } 2987 if (track->isStopping_2()) { 2988 track->mState = TrackBase::STOPPED; 2989 } 2990 if (track->isStopped()) { 2991 // Can't reset directly, as fast mixer is still polling this track 2992 // track->reset(); 2993 // So instead mark this track as needing to be reset after push with ack 2994 resetMask |= 1 << i; 2995 } 2996 isActive = false; 2997 break; 2998 case TrackBase::IDLE: 2999 default: 3000 LOG_FATAL("unexpected track state %d", track->mState); 3001 } 3002 3003 if (isActive) { 3004 // was it previously inactive? 3005 if (!(state->mTrackMask & (1 << j))) { 3006 ExtendedAudioBufferProvider *eabp = track; 3007 VolumeProvider *vp = track; 3008 fastTrack->mBufferProvider = eabp; 3009 fastTrack->mVolumeProvider = vp; 3010 fastTrack->mSampleRate = track->mSampleRate; 3011 fastTrack->mChannelMask = track->mChannelMask; 3012 fastTrack->mGeneration++; 3013 state->mTrackMask |= 1 << j; 3014 didModify = true; 3015 // no acknowledgement required for newly active tracks 3016 } 3017 // cache the combined master volume and stream type volume for fast mixer; this 3018 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3019 track->mCachedVolume = track->isMuted() ? 3020 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3021 ++fastTracks; 3022 } else { 3023 // was it previously active? 3024 if (state->mTrackMask & (1 << j)) { 3025 fastTrack->mBufferProvider = NULL; 3026 fastTrack->mGeneration++; 3027 state->mTrackMask &= ~(1 << j); 3028 didModify = true; 3029 // If any fast tracks were removed, we must wait for acknowledgement 3030 // because we're about to decrement the last sp<> on those tracks. 3031 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3032 } else { 3033 LOG_FATAL("fast track %d should have been active", j); 3034 } 3035 tracksToRemove->add(track); 3036 // Avoids a misleading display in dumpsys 3037 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3038 } 3039 continue; 3040 } 3041 3042 { // local variable scope to avoid goto warning 3043 3044 audio_track_cblk_t* cblk = track->cblk(); 3045 3046 // The first time a track is added we wait 3047 // for all its buffers to be filled before processing it 3048 int name = track->name(); 3049 // make sure that we have enough frames to mix one full buffer. 3050 // enforce this condition only once to enable draining the buffer in case the client 3051 // app does not call stop() and relies on underrun to stop: 3052 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3053 // during last round 3054 uint32_t minFrames = 1; 3055 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3056 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3057 if (t->sampleRate() == (int)mSampleRate) { 3058 minFrames = mNormalFrameCount; 3059 } else { 3060 // +1 for rounding and +1 for additional sample needed for interpolation 3061 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3062 // add frames already consumed but not yet released by the resampler 3063 // because cblk->framesReady() will include these frames 3064 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3065 // the minimum track buffer size is normally twice the number of frames necessary 3066 // to fill one buffer and the resampler should not leave more than one buffer worth 3067 // of unreleased frames after each pass, but just in case... 3068 ALOG_ASSERT(minFrames <= cblk->frameCount); 3069 } 3070 } 3071 if ((track->framesReady() >= minFrames) && track->isReady() && 3072 !track->isPaused() && !track->isTerminated()) 3073 { 3074 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3075 3076 mixedTracks++; 3077 3078 // track->mainBuffer() != mMixBuffer means there is an effect chain 3079 // connected to the track 3080 chain.clear(); 3081 if (track->mainBuffer() != mMixBuffer) { 3082 chain = getEffectChain_l(track->sessionId()); 3083 // Delegate volume control to effect in track effect chain if needed 3084 if (chain != 0) { 3085 tracksWithEffect++; 3086 } else { 3087 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3088 name, track->sessionId()); 3089 } 3090 } 3091 3092 3093 int param = AudioMixer::VOLUME; 3094 if (track->mFillingUpStatus == Track::FS_FILLED) { 3095 // no ramp for the first volume setting 3096 track->mFillingUpStatus = Track::FS_ACTIVE; 3097 if (track->mState == TrackBase::RESUMING) { 3098 track->mState = TrackBase::ACTIVE; 3099 param = AudioMixer::RAMP_VOLUME; 3100 } 3101 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3102 } else if (cblk->server != 0) { 3103 // If the track is stopped before the first frame was mixed, 3104 // do not apply ramp 3105 param = AudioMixer::RAMP_VOLUME; 3106 } 3107 3108 // compute volume for this track 3109 uint32_t vl, vr, va; 3110 if (track->isMuted() || track->isPausing() || 3111 mStreamTypes[track->streamType()].mute) { 3112 vl = vr = va = 0; 3113 if (track->isPausing()) { 3114 track->setPaused(); 3115 } 3116 } else { 3117 3118 // read original volumes with volume control 3119 float typeVolume = mStreamTypes[track->streamType()].volume; 3120 float v = masterVolume * typeVolume; 3121 uint32_t vlr = cblk->getVolumeLR(); 3122 vl = vlr & 0xFFFF; 3123 vr = vlr >> 16; 3124 // track volumes come from shared memory, so can't be trusted and must be clamped 3125 if (vl > MAX_GAIN_INT) { 3126 ALOGV("Track left volume out of range: %04X", vl); 3127 vl = MAX_GAIN_INT; 3128 } 3129 if (vr > MAX_GAIN_INT) { 3130 ALOGV("Track right volume out of range: %04X", vr); 3131 vr = MAX_GAIN_INT; 3132 } 3133 // now apply the master volume and stream type volume 3134 vl = (uint32_t)(v * vl) << 12; 3135 vr = (uint32_t)(v * vr) << 12; 3136 // assuming master volume and stream type volume each go up to 1.0, 3137 // vl and vr are now in 8.24 format 3138 3139 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3140 // send level comes from shared memory and so may be corrupt 3141 if (sendLevel > MAX_GAIN_INT) { 3142 ALOGV("Track send level out of range: %04X", sendLevel); 3143 sendLevel = MAX_GAIN_INT; 3144 } 3145 va = (uint32_t)(v * sendLevel); 3146 } 3147 // Delegate volume control to effect in track effect chain if needed 3148 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3149 // Do not ramp volume if volume is controlled by effect 3150 param = AudioMixer::VOLUME; 3151 track->mHasVolumeController = true; 3152 } else { 3153 // force no volume ramp when volume controller was just disabled or removed 3154 // from effect chain to avoid volume spike 3155 if (track->mHasVolumeController) { 3156 param = AudioMixer::VOLUME; 3157 } 3158 track->mHasVolumeController = false; 3159 } 3160 3161 // Convert volumes from 8.24 to 4.12 format 3162 // This additional clamping is needed in case chain->setVolume_l() overshot 3163 vl = (vl + (1 << 11)) >> 12; 3164 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3165 vr = (vr + (1 << 11)) >> 12; 3166 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3167 3168 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3169 3170 // XXX: these things DON'T need to be done each time 3171 mAudioMixer->setBufferProvider(name, track); 3172 mAudioMixer->enable(name); 3173 3174 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3175 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3176 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3177 mAudioMixer->setParameter( 3178 name, 3179 AudioMixer::TRACK, 3180 AudioMixer::FORMAT, (void *)track->format()); 3181 mAudioMixer->setParameter( 3182 name, 3183 AudioMixer::TRACK, 3184 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3185 mAudioMixer->setParameter( 3186 name, 3187 AudioMixer::RESAMPLE, 3188 AudioMixer::SAMPLE_RATE, 3189 (void *)(cblk->sampleRate)); 3190 mAudioMixer->setParameter( 3191 name, 3192 AudioMixer::TRACK, 3193 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3194 mAudioMixer->setParameter( 3195 name, 3196 AudioMixer::TRACK, 3197 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3198 3199 // reset retry count 3200 track->mRetryCount = kMaxTrackRetries; 3201 3202 // If one track is ready, set the mixer ready if: 3203 // - the mixer was not ready during previous round OR 3204 // - no other track is not ready 3205 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3206 mixerStatus != MIXER_TRACKS_ENABLED) { 3207 mixerStatus = MIXER_TRACKS_READY; 3208 } 3209 } else { 3210 // clear effect chain input buffer if an active track underruns to avoid sending 3211 // previous audio buffer again to effects 3212 chain = getEffectChain_l(track->sessionId()); 3213 if (chain != 0) { 3214 chain->clearInputBuffer(); 3215 } 3216 3217 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3218 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3219 track->isStopped() || track->isPaused()) { 3220 // We have consumed all the buffers of this track. 3221 // Remove it from the list of active tracks. 3222 // TODO: use actual buffer filling status instead of latency when available from 3223 // audio HAL 3224 size_t audioHALFrames = 3225 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3226 size_t framesWritten = 3227 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3228 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3229 if (track->isStopped()) { 3230 track->reset(); 3231 } 3232 tracksToRemove->add(track); 3233 } 3234 } else { 3235 track->mUnderrunCount++; 3236 // No buffers for this track. Give it a few chances to 3237 // fill a buffer, then remove it from active list. 3238 if (--(track->mRetryCount) <= 0) { 3239 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3240 tracksToRemove->add(track); 3241 // indicate to client process that the track was disabled because of underrun; 3242 // it will then automatically call start() when data is available 3243 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3244 // If one track is not ready, mark the mixer also not ready if: 3245 // - the mixer was ready during previous round OR 3246 // - no other track is ready 3247 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3248 mixerStatus != MIXER_TRACKS_READY) { 3249 mixerStatus = MIXER_TRACKS_ENABLED; 3250 } 3251 } 3252 mAudioMixer->disable(name); 3253 } 3254 3255 } // local variable scope to avoid goto warning 3256track_is_ready: ; 3257 3258 } 3259 3260 // Push the new FastMixer state if necessary 3261 bool pauseAudioWatchdog = false; 3262 if (didModify) { 3263 state->mFastTracksGen++; 3264 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3265 if (kUseFastMixer == FastMixer_Dynamic && 3266 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3267 state->mCommand = FastMixerState::COLD_IDLE; 3268 state->mColdFutexAddr = &mFastMixerFutex; 3269 state->mColdGen++; 3270 mFastMixerFutex = 0; 3271 if (kUseFastMixer == FastMixer_Dynamic) { 3272 mNormalSink = mOutputSink; 3273 } 3274 // If we go into cold idle, need to wait for acknowledgement 3275 // so that fast mixer stops doing I/O. 3276 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3277 pauseAudioWatchdog = true; 3278 } 3279 sq->end(); 3280 } 3281 if (sq != NULL) { 3282 sq->end(didModify); 3283 sq->push(block); 3284 } 3285 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3286 mAudioWatchdog->pause(); 3287 } 3288 3289 // Now perform the deferred reset on fast tracks that have stopped 3290 while (resetMask != 0) { 3291 size_t i = __builtin_ctz(resetMask); 3292 ALOG_ASSERT(i < count); 3293 resetMask &= ~(1 << i); 3294 sp<Track> t = mActiveTracks[i].promote(); 3295 if (t == 0) continue; 3296 Track* track = t.get(); 3297 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3298 track->reset(); 3299 } 3300 3301 // remove all the tracks that need to be... 3302 count = tracksToRemove->size(); 3303 if (CC_UNLIKELY(count)) { 3304 for (size_t i=0 ; i<count ; i++) { 3305 const sp<Track>& track = tracksToRemove->itemAt(i); 3306 mActiveTracks.remove(track); 3307 if (track->mainBuffer() != mMixBuffer) { 3308 chain = getEffectChain_l(track->sessionId()); 3309 if (chain != 0) { 3310 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3311 chain->decActiveTrackCnt(); 3312 } 3313 } 3314 if (track->isTerminated()) { 3315 removeTrack_l(track); 3316 } 3317 } 3318 } 3319 3320 // mix buffer must be cleared if all tracks are connected to an 3321 // effect chain as in this case the mixer will not write to 3322 // mix buffer and track effects will accumulate into it 3323 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3324 // FIXME as a performance optimization, should remember previous zero status 3325 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3326 } 3327 3328 // if any fast tracks, then status is ready 3329 mMixerStatusIgnoringFastTracks = mixerStatus; 3330 if (fastTracks > 0) { 3331 mixerStatus = MIXER_TRACKS_READY; 3332 } 3333 return mixerStatus; 3334} 3335 3336/* 3337The derived values that are cached: 3338 - mixBufferSize from frame count * frame size 3339 - activeSleepTime from activeSleepTimeUs() 3340 - idleSleepTime from idleSleepTimeUs() 3341 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3342 - maxPeriod from frame count and sample rate (MIXER only) 3343 3344The parameters that affect these derived values are: 3345 - frame count 3346 - frame size 3347 - sample rate 3348 - device type: A2DP or not 3349 - device latency 3350 - format: PCM or not 3351 - active sleep time 3352 - idle sleep time 3353*/ 3354 3355void AudioFlinger::PlaybackThread::cacheParameters_l() 3356{ 3357 mixBufferSize = mNormalFrameCount * mFrameSize; 3358 activeSleepTime = activeSleepTimeUs(); 3359 idleSleepTime = idleSleepTimeUs(); 3360} 3361 3362void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3363{ 3364 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3365 this, streamType, mTracks.size()); 3366 Mutex::Autolock _l(mLock); 3367 3368 size_t size = mTracks.size(); 3369 for (size_t i = 0; i < size; i++) { 3370 sp<Track> t = mTracks[i]; 3371 if (t->streamType() == streamType) { 3372 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3373 t->mCblk->cv.signal(); 3374 } 3375 } 3376} 3377 3378// getTrackName_l() must be called with ThreadBase::mLock held 3379int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3380{ 3381 return mAudioMixer->getTrackName(channelMask); 3382} 3383 3384// deleteTrackName_l() must be called with ThreadBase::mLock held 3385void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3386{ 3387 ALOGV("remove track (%d) and delete from mixer", name); 3388 mAudioMixer->deleteTrackName(name); 3389} 3390 3391// checkForNewParameters_l() must be called with ThreadBase::mLock held 3392bool AudioFlinger::MixerThread::checkForNewParameters_l() 3393{ 3394 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3395 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3396 bool reconfig = false; 3397 3398 while (!mNewParameters.isEmpty()) { 3399 3400 if (mFastMixer != NULL) { 3401 FastMixerStateQueue *sq = mFastMixer->sq(); 3402 FastMixerState *state = sq->begin(); 3403 if (!(state->mCommand & FastMixerState::IDLE)) { 3404 previousCommand = state->mCommand; 3405 state->mCommand = FastMixerState::HOT_IDLE; 3406 sq->end(); 3407 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3408 } else { 3409 sq->end(false /*didModify*/); 3410 } 3411 } 3412 3413 status_t status = NO_ERROR; 3414 String8 keyValuePair = mNewParameters[0]; 3415 AudioParameter param = AudioParameter(keyValuePair); 3416 int value; 3417 3418 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3419 reconfig = true; 3420 } 3421 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3422 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3423 status = BAD_VALUE; 3424 } else { 3425 reconfig = true; 3426 } 3427 } 3428 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3429 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3430 status = BAD_VALUE; 3431 } else { 3432 reconfig = true; 3433 } 3434 } 3435 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3436 // do not accept frame count changes if tracks are open as the track buffer 3437 // size depends on frame count and correct behavior would not be guaranteed 3438 // if frame count is changed after track creation 3439 if (!mTracks.isEmpty()) { 3440 status = INVALID_OPERATION; 3441 } else { 3442 reconfig = true; 3443 } 3444 } 3445 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3446#ifdef ADD_BATTERY_DATA 3447 // when changing the audio output device, call addBatteryData to notify 3448 // the change 3449 if ((int)mDevice != value) { 3450 uint32_t params = 0; 3451 // check whether speaker is on 3452 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3453 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3454 } 3455 3456 int deviceWithoutSpeaker 3457 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3458 // check if any other device (except speaker) is on 3459 if (value & deviceWithoutSpeaker ) { 3460 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3461 } 3462 3463 if (params != 0) { 3464 addBatteryData(params); 3465 } 3466 } 3467#endif 3468 3469 // forward device change to effects that have requested to be 3470 // aware of attached audio device. 3471 mDevice = (uint32_t)value; 3472 for (size_t i = 0; i < mEffectChains.size(); i++) { 3473 mEffectChains[i]->setDevice_l(mDevice); 3474 } 3475 } 3476 3477 if (status == NO_ERROR) { 3478 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3479 keyValuePair.string()); 3480 if (!mStandby && status == INVALID_OPERATION) { 3481 mOutput->stream->common.standby(&mOutput->stream->common); 3482 mStandby = true; 3483 mBytesWritten = 0; 3484 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3485 keyValuePair.string()); 3486 } 3487 if (status == NO_ERROR && reconfig) { 3488 delete mAudioMixer; 3489 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3490 mAudioMixer = NULL; 3491 readOutputParameters(); 3492 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3493 for (size_t i = 0; i < mTracks.size() ; i++) { 3494 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3495 if (name < 0) break; 3496 mTracks[i]->mName = name; 3497 // limit track sample rate to 2 x new output sample rate 3498 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3499 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3500 } 3501 } 3502 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3503 } 3504 } 3505 3506 mNewParameters.removeAt(0); 3507 3508 mParamStatus = status; 3509 mParamCond.signal(); 3510 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3511 // already timed out waiting for the status and will never signal the condition. 3512 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3513 } 3514 3515 if (!(previousCommand & FastMixerState::IDLE)) { 3516 ALOG_ASSERT(mFastMixer != NULL); 3517 FastMixerStateQueue *sq = mFastMixer->sq(); 3518 FastMixerState *state = sq->begin(); 3519 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3520 state->mCommand = previousCommand; 3521 sq->end(); 3522 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3523 } 3524 3525 return reconfig; 3526} 3527 3528status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3529{ 3530 const size_t SIZE = 256; 3531 char buffer[SIZE]; 3532 String8 result; 3533 3534 PlaybackThread::dumpInternals(fd, args); 3535 3536 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3537 result.append(buffer); 3538 write(fd, result.string(), result.size()); 3539 3540 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3541 FastMixerDumpState copy = mFastMixerDumpState; 3542 copy.dump(fd); 3543 3544#ifdef STATE_QUEUE_DUMP 3545 // Similar for state queue 3546 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3547 observerCopy.dump(fd); 3548 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3549 mutatorCopy.dump(fd); 3550#endif 3551 3552 // Write the tee output to a .wav file 3553 NBAIO_Source *teeSource = mTeeSource.get(); 3554 if (teeSource != NULL) { 3555 char teePath[64]; 3556 struct timeval tv; 3557 gettimeofday(&tv, NULL); 3558 struct tm tm; 3559 localtime_r(&tv.tv_sec, &tm); 3560 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3561 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3562 if (teeFd >= 0) { 3563 char wavHeader[44]; 3564 memcpy(wavHeader, 3565 "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", 3566 sizeof(wavHeader)); 3567 NBAIO_Format format = teeSource->format(); 3568 unsigned channelCount = Format_channelCount(format); 3569 ALOG_ASSERT(channelCount <= FCC_2); 3570 unsigned sampleRate = Format_sampleRate(format); 3571 wavHeader[22] = channelCount; // number of channels 3572 wavHeader[24] = sampleRate; // sample rate 3573 wavHeader[25] = sampleRate >> 8; 3574 wavHeader[32] = channelCount * 2; // block alignment 3575 write(teeFd, wavHeader, sizeof(wavHeader)); 3576 size_t total = 0; 3577 bool firstRead = true; 3578 for (;;) { 3579#define TEE_SINK_READ 1024 3580 short buffer[TEE_SINK_READ * FCC_2]; 3581 size_t count = TEE_SINK_READ; 3582 ssize_t actual = teeSource->read(buffer, count); 3583 bool wasFirstRead = firstRead; 3584 firstRead = false; 3585 if (actual <= 0) { 3586 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3587 continue; 3588 } 3589 break; 3590 } 3591 ALOG_ASSERT(actual <= count); 3592 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3593 total += actual; 3594 } 3595 lseek(teeFd, (off_t) 4, SEEK_SET); 3596 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3597 write(teeFd, &temp, sizeof(temp)); 3598 lseek(teeFd, (off_t) 40, SEEK_SET); 3599 temp = total * channelCount * sizeof(short); 3600 write(teeFd, &temp, sizeof(temp)); 3601 close(teeFd); 3602 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3603 } else { 3604 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3605 } 3606 } 3607 3608 if (mAudioWatchdog != 0) { 3609 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3610 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3611 wdCopy.dump(fd); 3612 } 3613 3614 return NO_ERROR; 3615} 3616 3617uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3618{ 3619 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3620} 3621 3622uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3623{ 3624 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3625} 3626 3627void AudioFlinger::MixerThread::cacheParameters_l() 3628{ 3629 PlaybackThread::cacheParameters_l(); 3630 3631 // FIXME: Relaxed timing because of a certain device that can't meet latency 3632 // Should be reduced to 2x after the vendor fixes the driver issue 3633 // increase threshold again due to low power audio mode. The way this warning 3634 // threshold is calculated and its usefulness should be reconsidered anyway. 3635 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3636} 3637 3638// ---------------------------------------------------------------------------- 3639AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3640 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3641 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3642 // mLeftVolFloat, mRightVolFloat 3643{ 3644} 3645 3646AudioFlinger::DirectOutputThread::~DirectOutputThread() 3647{ 3648} 3649 3650AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3651 Vector< sp<Track> > *tracksToRemove 3652) 3653{ 3654 sp<Track> trackToRemove; 3655 3656 mixer_state mixerStatus = MIXER_IDLE; 3657 3658 // find out which tracks need to be processed 3659 if (mActiveTracks.size() != 0) { 3660 sp<Track> t = mActiveTracks[0].promote(); 3661 // The track died recently 3662 if (t == 0) return MIXER_IDLE; 3663 3664 Track* const track = t.get(); 3665 audio_track_cblk_t* cblk = track->cblk(); 3666 3667 // The first time a track is added we wait 3668 // for all its buffers to be filled before processing it 3669 uint32_t minFrames; 3670 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3671 minFrames = mNormalFrameCount; 3672 } else { 3673 minFrames = 1; 3674 } 3675 if ((track->framesReady() >= minFrames) && track->isReady() && 3676 !track->isPaused() && !track->isTerminated()) 3677 { 3678 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3679 3680 if (track->mFillingUpStatus == Track::FS_FILLED) { 3681 track->mFillingUpStatus = Track::FS_ACTIVE; 3682 mLeftVolFloat = mRightVolFloat = 0; 3683 if (track->mState == TrackBase::RESUMING) { 3684 track->mState = TrackBase::ACTIVE; 3685 } 3686 } 3687 3688 // compute volume for this track 3689 float left, right; 3690 if (track->isMuted() || mMasterMute || track->isPausing() || 3691 mStreamTypes[track->streamType()].mute) { 3692 left = right = 0; 3693 if (track->isPausing()) { 3694 track->setPaused(); 3695 } 3696 } else { 3697 float typeVolume = mStreamTypes[track->streamType()].volume; 3698 float v = mMasterVolume * typeVolume; 3699 uint32_t vlr = cblk->getVolumeLR(); 3700 float v_clamped = v * (vlr & 0xFFFF); 3701 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3702 left = v_clamped/MAX_GAIN; 3703 v_clamped = v * (vlr >> 16); 3704 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3705 right = v_clamped/MAX_GAIN; 3706 } 3707 3708 if (left != mLeftVolFloat || right != mRightVolFloat) { 3709 mLeftVolFloat = left; 3710 mRightVolFloat = right; 3711 3712 // Convert volumes from float to 8.24 3713 uint32_t vl = (uint32_t)(left * (1 << 24)); 3714 uint32_t vr = (uint32_t)(right * (1 << 24)); 3715 3716 // Delegate volume control to effect in track effect chain if needed 3717 // only one effect chain can be present on DirectOutputThread, so if 3718 // there is one, the track is connected to it 3719 if (!mEffectChains.isEmpty()) { 3720 // Do not ramp volume if volume is controlled by effect 3721 mEffectChains[0]->setVolume_l(&vl, &vr); 3722 left = (float)vl / (1 << 24); 3723 right = (float)vr / (1 << 24); 3724 } 3725 mOutput->stream->set_volume(mOutput->stream, left, right); 3726 } 3727 3728 // reset retry count 3729 track->mRetryCount = kMaxTrackRetriesDirect; 3730 mActiveTrack = t; 3731 mixerStatus = MIXER_TRACKS_READY; 3732 } else { 3733 // clear effect chain input buffer if an active track underruns to avoid sending 3734 // previous audio buffer again to effects 3735 if (!mEffectChains.isEmpty()) { 3736 mEffectChains[0]->clearInputBuffer(); 3737 } 3738 3739 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3740 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3741 track->isStopped() || track->isPaused()) { 3742 // We have consumed all the buffers of this track. 3743 // Remove it from the list of active tracks. 3744 // TODO: implement behavior for compressed audio 3745 size_t audioHALFrames = 3746 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3747 size_t framesWritten = 3748 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3749 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3750 if (track->isStopped()) { 3751 track->reset(); 3752 } 3753 trackToRemove = track; 3754 } 3755 } else { 3756 // No buffers for this track. Give it a few chances to 3757 // fill a buffer, then remove it from active list. 3758 if (--(track->mRetryCount) <= 0) { 3759 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3760 trackToRemove = track; 3761 } else { 3762 mixerStatus = MIXER_TRACKS_ENABLED; 3763 } 3764 } 3765 } 3766 } 3767 3768 // FIXME merge this with similar code for removing multiple tracks 3769 // remove all the tracks that need to be... 3770 if (CC_UNLIKELY(trackToRemove != 0)) { 3771 tracksToRemove->add(trackToRemove); 3772 mActiveTracks.remove(trackToRemove); 3773 if (!mEffectChains.isEmpty()) { 3774 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3775 trackToRemove->sessionId()); 3776 mEffectChains[0]->decActiveTrackCnt(); 3777 } 3778 if (trackToRemove->isTerminated()) { 3779 removeTrack_l(trackToRemove); 3780 } 3781 } 3782 3783 return mixerStatus; 3784} 3785 3786void AudioFlinger::DirectOutputThread::threadLoop_mix() 3787{ 3788 AudioBufferProvider::Buffer buffer; 3789 size_t frameCount = mFrameCount; 3790 int8_t *curBuf = (int8_t *)mMixBuffer; 3791 // output audio to hardware 3792 while (frameCount) { 3793 buffer.frameCount = frameCount; 3794 mActiveTrack->getNextBuffer(&buffer); 3795 if (CC_UNLIKELY(buffer.raw == NULL)) { 3796 memset(curBuf, 0, frameCount * mFrameSize); 3797 break; 3798 } 3799 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3800 frameCount -= buffer.frameCount; 3801 curBuf += buffer.frameCount * mFrameSize; 3802 mActiveTrack->releaseBuffer(&buffer); 3803 } 3804 sleepTime = 0; 3805 standbyTime = systemTime() + standbyDelay; 3806 mActiveTrack.clear(); 3807 3808} 3809 3810void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3811{ 3812 if (sleepTime == 0) { 3813 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3814 sleepTime = activeSleepTime; 3815 } else { 3816 sleepTime = idleSleepTime; 3817 } 3818 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3819 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3820 sleepTime = 0; 3821 } 3822} 3823 3824// getTrackName_l() must be called with ThreadBase::mLock held 3825int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3826{ 3827 return 0; 3828} 3829 3830// deleteTrackName_l() must be called with ThreadBase::mLock held 3831void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3832{ 3833} 3834 3835// checkForNewParameters_l() must be called with ThreadBase::mLock held 3836bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3837{ 3838 bool reconfig = false; 3839 3840 while (!mNewParameters.isEmpty()) { 3841 status_t status = NO_ERROR; 3842 String8 keyValuePair = mNewParameters[0]; 3843 AudioParameter param = AudioParameter(keyValuePair); 3844 int value; 3845 3846 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3847 // do not accept frame count changes if tracks are open as the track buffer 3848 // size depends on frame count and correct behavior would not be garantied 3849 // if frame count is changed after track creation 3850 if (!mTracks.isEmpty()) { 3851 status = INVALID_OPERATION; 3852 } else { 3853 reconfig = true; 3854 } 3855 } 3856 if (status == NO_ERROR) { 3857 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3858 keyValuePair.string()); 3859 if (!mStandby && status == INVALID_OPERATION) { 3860 mOutput->stream->common.standby(&mOutput->stream->common); 3861 mStandby = true; 3862 mBytesWritten = 0; 3863 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3864 keyValuePair.string()); 3865 } 3866 if (status == NO_ERROR && reconfig) { 3867 readOutputParameters(); 3868 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3869 } 3870 } 3871 3872 mNewParameters.removeAt(0); 3873 3874 mParamStatus = status; 3875 mParamCond.signal(); 3876 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3877 // already timed out waiting for the status and will never signal the condition. 3878 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3879 } 3880 return reconfig; 3881} 3882 3883uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3884{ 3885 uint32_t time; 3886 if (audio_is_linear_pcm(mFormat)) { 3887 time = PlaybackThread::activeSleepTimeUs(); 3888 } else { 3889 time = 10000; 3890 } 3891 return time; 3892} 3893 3894uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3895{ 3896 uint32_t time; 3897 if (audio_is_linear_pcm(mFormat)) { 3898 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3899 } else { 3900 time = 10000; 3901 } 3902 return time; 3903} 3904 3905uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3906{ 3907 uint32_t time; 3908 if (audio_is_linear_pcm(mFormat)) { 3909 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3910 } else { 3911 time = 10000; 3912 } 3913 return time; 3914} 3915 3916void AudioFlinger::DirectOutputThread::cacheParameters_l() 3917{ 3918 PlaybackThread::cacheParameters_l(); 3919 3920 // use shorter standby delay as on normal output to release 3921 // hardware resources as soon as possible 3922 standbyDelay = microseconds(activeSleepTime*2); 3923} 3924 3925// ---------------------------------------------------------------------------- 3926 3927AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3928 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3929 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3930 mWaitTimeMs(UINT_MAX) 3931{ 3932 addOutputTrack(mainThread); 3933} 3934 3935AudioFlinger::DuplicatingThread::~DuplicatingThread() 3936{ 3937 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3938 mOutputTracks[i]->destroy(); 3939 } 3940} 3941 3942void AudioFlinger::DuplicatingThread::threadLoop_mix() 3943{ 3944 // mix buffers... 3945 if (outputsReady(outputTracks)) { 3946 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3947 } else { 3948 memset(mMixBuffer, 0, mixBufferSize); 3949 } 3950 sleepTime = 0; 3951 writeFrames = mNormalFrameCount; 3952 standbyTime = systemTime() + standbyDelay; 3953} 3954 3955void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3956{ 3957 if (sleepTime == 0) { 3958 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3959 sleepTime = activeSleepTime; 3960 } else { 3961 sleepTime = idleSleepTime; 3962 } 3963 } else if (mBytesWritten != 0) { 3964 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3965 writeFrames = mNormalFrameCount; 3966 memset(mMixBuffer, 0, mixBufferSize); 3967 } else { 3968 // flush remaining overflow buffers in output tracks 3969 writeFrames = 0; 3970 } 3971 sleepTime = 0; 3972 } 3973} 3974 3975void AudioFlinger::DuplicatingThread::threadLoop_write() 3976{ 3977 for (size_t i = 0; i < outputTracks.size(); i++) { 3978 outputTracks[i]->write(mMixBuffer, writeFrames); 3979 } 3980 mBytesWritten += mixBufferSize; 3981} 3982 3983void AudioFlinger::DuplicatingThread::threadLoop_standby() 3984{ 3985 // DuplicatingThread implements standby by stopping all tracks 3986 for (size_t i = 0; i < outputTracks.size(); i++) { 3987 outputTracks[i]->stop(); 3988 } 3989} 3990 3991void AudioFlinger::DuplicatingThread::saveOutputTracks() 3992{ 3993 outputTracks = mOutputTracks; 3994} 3995 3996void AudioFlinger::DuplicatingThread::clearOutputTracks() 3997{ 3998 outputTracks.clear(); 3999} 4000 4001void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4002{ 4003 Mutex::Autolock _l(mLock); 4004 // FIXME explain this formula 4005 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4006 OutputTrack *outputTrack = new OutputTrack(thread, 4007 this, 4008 mSampleRate, 4009 mFormat, 4010 mChannelMask, 4011 frameCount); 4012 if (outputTrack->cblk() != NULL) { 4013 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4014 mOutputTracks.add(outputTrack); 4015 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4016 updateWaitTime_l(); 4017 } 4018} 4019 4020void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4021{ 4022 Mutex::Autolock _l(mLock); 4023 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4024 if (mOutputTracks[i]->thread() == thread) { 4025 mOutputTracks[i]->destroy(); 4026 mOutputTracks.removeAt(i); 4027 updateWaitTime_l(); 4028 return; 4029 } 4030 } 4031 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4032} 4033 4034// caller must hold mLock 4035void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4036{ 4037 mWaitTimeMs = UINT_MAX; 4038 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4039 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4040 if (strong != 0) { 4041 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4042 if (waitTimeMs < mWaitTimeMs) { 4043 mWaitTimeMs = waitTimeMs; 4044 } 4045 } 4046 } 4047} 4048 4049 4050bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4051{ 4052 for (size_t i = 0; i < outputTracks.size(); i++) { 4053 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4054 if (thread == 0) { 4055 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4056 return false; 4057 } 4058 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4059 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4060 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4061 return false; 4062 } 4063 } 4064 return true; 4065} 4066 4067uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4068{ 4069 return (mWaitTimeMs * 1000) / 2; 4070} 4071 4072void AudioFlinger::DuplicatingThread::cacheParameters_l() 4073{ 4074 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4075 updateWaitTime_l(); 4076 4077 MixerThread::cacheParameters_l(); 4078} 4079 4080// ---------------------------------------------------------------------------- 4081 4082// TrackBase constructor must be called with AudioFlinger::mLock held 4083AudioFlinger::ThreadBase::TrackBase::TrackBase( 4084 ThreadBase *thread, 4085 const sp<Client>& client, 4086 uint32_t sampleRate, 4087 audio_format_t format, 4088 uint32_t channelMask, 4089 int frameCount, 4090 const sp<IMemory>& sharedBuffer, 4091 int sessionId) 4092 : RefBase(), 4093 mThread(thread), 4094 mClient(client), 4095 mCblk(NULL), 4096 // mBuffer 4097 // mBufferEnd 4098 mFrameCount(0), 4099 mState(IDLE), 4100 mSampleRate(sampleRate), 4101 mFormat(format), 4102 mStepServerFailed(false), 4103 mSessionId(sessionId) 4104 // mChannelCount 4105 // mChannelMask 4106{ 4107 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4108 4109 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4110 size_t size = sizeof(audio_track_cblk_t); 4111 uint8_t channelCount = popcount(channelMask); 4112 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4113 if (sharedBuffer == 0) { 4114 size += bufferSize; 4115 } 4116 4117 if (client != NULL) { 4118 mCblkMemory = client->heap()->allocate(size); 4119 if (mCblkMemory != 0) { 4120 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4121 if (mCblk != NULL) { // construct the shared structure in-place. 4122 new(mCblk) audio_track_cblk_t(); 4123 // clear all buffers 4124 mCblk->frameCount = frameCount; 4125 mCblk->sampleRate = sampleRate; 4126// uncomment the following lines to quickly test 32-bit wraparound 4127// mCblk->user = 0xffff0000; 4128// mCblk->server = 0xffff0000; 4129// mCblk->userBase = 0xffff0000; 4130// mCblk->serverBase = 0xffff0000; 4131 mChannelCount = channelCount; 4132 mChannelMask = channelMask; 4133 if (sharedBuffer == 0) { 4134 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4135 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4136 // Force underrun condition to avoid false underrun callback until first data is 4137 // written to buffer (other flags are cleared) 4138 mCblk->flags = CBLK_UNDERRUN_ON; 4139 } else { 4140 mBuffer = sharedBuffer->pointer(); 4141 } 4142 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4143 } 4144 } else { 4145 ALOGE("not enough memory for AudioTrack size=%u", size); 4146 client->heap()->dump("AudioTrack"); 4147 return; 4148 } 4149 } else { 4150 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4151 // construct the shared structure in-place. 4152 new(mCblk) audio_track_cblk_t(); 4153 // clear all buffers 4154 mCblk->frameCount = frameCount; 4155 mCblk->sampleRate = sampleRate; 4156// uncomment the following lines to quickly test 32-bit wraparound 4157// mCblk->user = 0xffff0000; 4158// mCblk->server = 0xffff0000; 4159// mCblk->userBase = 0xffff0000; 4160// mCblk->serverBase = 0xffff0000; 4161 mChannelCount = channelCount; 4162 mChannelMask = channelMask; 4163 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4164 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4165 // Force underrun condition to avoid false underrun callback until first data is 4166 // written to buffer (other flags are cleared) 4167 mCblk->flags = CBLK_UNDERRUN_ON; 4168 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4169 } 4170} 4171 4172AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4173{ 4174 if (mCblk != NULL) { 4175 if (mClient == 0) { 4176 delete mCblk; 4177 } else { 4178 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4179 } 4180 } 4181 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4182 if (mClient != 0) { 4183 // Client destructor must run with AudioFlinger mutex locked 4184 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4185 // If the client's reference count drops to zero, the associated destructor 4186 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4187 // relying on the automatic clear() at end of scope. 4188 mClient.clear(); 4189 } 4190} 4191 4192// AudioBufferProvider interface 4193// getNextBuffer() = 0; 4194// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4195void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4196{ 4197 buffer->raw = NULL; 4198 mFrameCount = buffer->frameCount; 4199 // FIXME See note at getNextBuffer() 4200 (void) step(); // ignore return value of step() 4201 buffer->frameCount = 0; 4202} 4203 4204bool AudioFlinger::ThreadBase::TrackBase::step() { 4205 bool result; 4206 audio_track_cblk_t* cblk = this->cblk(); 4207 4208 result = cblk->stepServer(mFrameCount); 4209 if (!result) { 4210 ALOGV("stepServer failed acquiring cblk mutex"); 4211 mStepServerFailed = true; 4212 } 4213 return result; 4214} 4215 4216void AudioFlinger::ThreadBase::TrackBase::reset() { 4217 audio_track_cblk_t* cblk = this->cblk(); 4218 4219 cblk->user = 0; 4220 cblk->server = 0; 4221 cblk->userBase = 0; 4222 cblk->serverBase = 0; 4223 mStepServerFailed = false; 4224 ALOGV("TrackBase::reset"); 4225} 4226 4227int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4228 return (int)mCblk->sampleRate; 4229} 4230 4231void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4232 audio_track_cblk_t* cblk = this->cblk(); 4233 size_t frameSize = cblk->frameSize; 4234 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4235 int8_t *bufferEnd = bufferStart + frames * frameSize; 4236 4237 // Check validity of returned pointer in case the track control block would have been corrupted. 4238 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4239 "TrackBase::getBuffer buffer out of range:\n" 4240 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4241 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4242 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4243 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4244 4245 return bufferStart; 4246} 4247 4248status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4249{ 4250 mSyncEvents.add(event); 4251 return NO_ERROR; 4252} 4253 4254// ---------------------------------------------------------------------------- 4255 4256// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4257AudioFlinger::PlaybackThread::Track::Track( 4258 PlaybackThread *thread, 4259 const sp<Client>& client, 4260 audio_stream_type_t streamType, 4261 uint32_t sampleRate, 4262 audio_format_t format, 4263 uint32_t channelMask, 4264 int frameCount, 4265 const sp<IMemory>& sharedBuffer, 4266 int sessionId, 4267 IAudioFlinger::track_flags_t flags) 4268 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4269 mMute(false), 4270 mFillingUpStatus(FS_INVALID), 4271 // mRetryCount initialized later when needed 4272 mSharedBuffer(sharedBuffer), 4273 mStreamType(streamType), 4274 mName(-1), // see note below 4275 mMainBuffer(thread->mixBuffer()), 4276 mAuxBuffer(NULL), 4277 mAuxEffectId(0), mHasVolumeController(false), 4278 mPresentationCompleteFrames(0), 4279 mFlags(flags), 4280 mFastIndex(-1), 4281 mUnderrunCount(0), 4282 mCachedVolume(1.0) 4283{ 4284 if (mCblk != NULL) { 4285 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4286 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4287 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4288 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4289 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4290 mCblk->mName = mName; 4291 if (mName < 0) { 4292 ALOGE("no more track names available"); 4293 return; 4294 } 4295 // only allocate a fast track index if we were able to allocate a normal track name 4296 if (flags & IAudioFlinger::TRACK_FAST) { 4297 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4298 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4299 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4300 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4301 // FIXME This is too eager. We allocate a fast track index before the 4302 // fast track becomes active. Since fast tracks are a scarce resource, 4303 // this means we are potentially denying other more important fast tracks from 4304 // being created. It would be better to allocate the index dynamically. 4305 mFastIndex = i; 4306 mCblk->mName = i; 4307 // Read the initial underruns because this field is never cleared by the fast mixer 4308 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4309 thread->mFastTrackAvailMask &= ~(1 << i); 4310 } 4311 } 4312 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4313} 4314 4315AudioFlinger::PlaybackThread::Track::~Track() 4316{ 4317 ALOGV("PlaybackThread::Track destructor"); 4318 sp<ThreadBase> thread = mThread.promote(); 4319 if (thread != 0) { 4320 Mutex::Autolock _l(thread->mLock); 4321 mState = TERMINATED; 4322 } 4323} 4324 4325void AudioFlinger::PlaybackThread::Track::destroy() 4326{ 4327 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4328 // by removing it from mTracks vector, so there is a risk that this Tracks's 4329 // destructor is called. As the destructor needs to lock mLock, 4330 // we must acquire a strong reference on this Track before locking mLock 4331 // here so that the destructor is called only when exiting this function. 4332 // On the other hand, as long as Track::destroy() is only called by 4333 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4334 // this Track with its member mTrack. 4335 sp<Track> keep(this); 4336 { // scope for mLock 4337 sp<ThreadBase> thread = mThread.promote(); 4338 if (thread != 0) { 4339 if (!isOutputTrack()) { 4340 if (mState == ACTIVE || mState == RESUMING) { 4341 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4342 4343#ifdef ADD_BATTERY_DATA 4344 // to track the speaker usage 4345 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4346#endif 4347 } 4348 AudioSystem::releaseOutput(thread->id()); 4349 } 4350 Mutex::Autolock _l(thread->mLock); 4351 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4352 playbackThread->destroyTrack_l(this); 4353 } 4354 } 4355} 4356 4357/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4358{ 4359 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4360 " Server User Main buf Aux Buf Flags Underruns\n"); 4361} 4362 4363void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4364{ 4365 uint32_t vlr = mCblk->getVolumeLR(); 4366 if (isFastTrack()) { 4367 sprintf(buffer, " F %2d", mFastIndex); 4368 } else { 4369 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4370 } 4371 track_state state = mState; 4372 char stateChar; 4373 switch (state) { 4374 case IDLE: 4375 stateChar = 'I'; 4376 break; 4377 case TERMINATED: 4378 stateChar = 'T'; 4379 break; 4380 case STOPPING_1: 4381 stateChar = 's'; 4382 break; 4383 case STOPPING_2: 4384 stateChar = '5'; 4385 break; 4386 case STOPPED: 4387 stateChar = 'S'; 4388 break; 4389 case RESUMING: 4390 stateChar = 'R'; 4391 break; 4392 case ACTIVE: 4393 stateChar = 'A'; 4394 break; 4395 case PAUSING: 4396 stateChar = 'p'; 4397 break; 4398 case PAUSED: 4399 stateChar = 'P'; 4400 break; 4401 case FLUSHED: 4402 stateChar = 'F'; 4403 break; 4404 default: 4405 stateChar = '?'; 4406 break; 4407 } 4408 char nowInUnderrun; 4409 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4410 case UNDERRUN_FULL: 4411 nowInUnderrun = ' '; 4412 break; 4413 case UNDERRUN_PARTIAL: 4414 nowInUnderrun = '<'; 4415 break; 4416 case UNDERRUN_EMPTY: 4417 nowInUnderrun = '*'; 4418 break; 4419 default: 4420 nowInUnderrun = '?'; 4421 break; 4422 } 4423 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4424 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4425 (mClient == 0) ? getpid_cached : mClient->pid(), 4426 mStreamType, 4427 mFormat, 4428 mChannelMask, 4429 mSessionId, 4430 mFrameCount, 4431 mCblk->frameCount, 4432 stateChar, 4433 mMute, 4434 mFillingUpStatus, 4435 mCblk->sampleRate, 4436 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4437 20.0 * log10((vlr >> 16) / 4096.0), 4438 mCblk->server, 4439 mCblk->user, 4440 (int)mMainBuffer, 4441 (int)mAuxBuffer, 4442 mCblk->flags, 4443 mUnderrunCount, 4444 nowInUnderrun); 4445} 4446 4447// AudioBufferProvider interface 4448status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4449 AudioBufferProvider::Buffer* buffer, int64_t pts) 4450{ 4451 audio_track_cblk_t* cblk = this->cblk(); 4452 uint32_t framesReady; 4453 uint32_t framesReq = buffer->frameCount; 4454 4455 // Check if last stepServer failed, try to step now 4456 if (mStepServerFailed) { 4457 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4458 // Since the fast mixer is higher priority than client callback thread, 4459 // it does not result in priority inversion for client. 4460 // But a non-blocking solution would be preferable to avoid 4461 // fast mixer being unable to tryLock(), and 4462 // to avoid the extra context switches if the client wakes up, 4463 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4464 if (!step()) goto getNextBuffer_exit; 4465 ALOGV("stepServer recovered"); 4466 mStepServerFailed = false; 4467 } 4468 4469 // FIXME Same as above 4470 framesReady = cblk->framesReady(); 4471 4472 if (CC_LIKELY(framesReady)) { 4473 uint32_t s = cblk->server; 4474 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4475 4476 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4477 if (framesReq > framesReady) { 4478 framesReq = framesReady; 4479 } 4480 if (framesReq > bufferEnd - s) { 4481 framesReq = bufferEnd - s; 4482 } 4483 4484 buffer->raw = getBuffer(s, framesReq); 4485 if (buffer->raw == NULL) goto getNextBuffer_exit; 4486 4487 buffer->frameCount = framesReq; 4488 return NO_ERROR; 4489 } 4490 4491getNextBuffer_exit: 4492 buffer->raw = NULL; 4493 buffer->frameCount = 0; 4494 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4495 return NOT_ENOUGH_DATA; 4496} 4497 4498// Note that framesReady() takes a mutex on the control block using tryLock(). 4499// This could result in priority inversion if framesReady() is called by the normal mixer, 4500// as the normal mixer thread runs at lower 4501// priority than the client's callback thread: there is a short window within framesReady() 4502// during which the normal mixer could be preempted, and the client callback would block. 4503// Another problem can occur if framesReady() is called by the fast mixer: 4504// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4505// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4506size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4507 return mCblk->framesReady(); 4508} 4509 4510// Don't call for fast tracks; the framesReady() could result in priority inversion 4511bool AudioFlinger::PlaybackThread::Track::isReady() const { 4512 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4513 4514 if (framesReady() >= mCblk->frameCount || 4515 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4516 mFillingUpStatus = FS_FILLED; 4517 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4518 return true; 4519 } 4520 return false; 4521} 4522 4523status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4524 int triggerSession) 4525{ 4526 status_t status = NO_ERROR; 4527 ALOGV("start(%d), calling pid %d session %d", 4528 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4529 4530 sp<ThreadBase> thread = mThread.promote(); 4531 if (thread != 0) { 4532 Mutex::Autolock _l(thread->mLock); 4533 track_state state = mState; 4534 // here the track could be either new, or restarted 4535 // in both cases "unstop" the track 4536 if (mState == PAUSED) { 4537 mState = TrackBase::RESUMING; 4538 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4539 } else { 4540 mState = TrackBase::ACTIVE; 4541 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4542 } 4543 4544 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4545 thread->mLock.unlock(); 4546 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4547 thread->mLock.lock(); 4548 4549#ifdef ADD_BATTERY_DATA 4550 // to track the speaker usage 4551 if (status == NO_ERROR) { 4552 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4553 } 4554#endif 4555 } 4556 if (status == NO_ERROR) { 4557 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4558 playbackThread->addTrack_l(this); 4559 } else { 4560 mState = state; 4561 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4562 } 4563 } else { 4564 status = BAD_VALUE; 4565 } 4566 return status; 4567} 4568 4569void AudioFlinger::PlaybackThread::Track::stop() 4570{ 4571 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4572 sp<ThreadBase> thread = mThread.promote(); 4573 if (thread != 0) { 4574 Mutex::Autolock _l(thread->mLock); 4575 track_state state = mState; 4576 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4577 // If the track is not active (PAUSED and buffers full), flush buffers 4578 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4579 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4580 reset(); 4581 mState = STOPPED; 4582 } else if (!isFastTrack()) { 4583 mState = STOPPED; 4584 } else { 4585 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4586 // and then to STOPPED and reset() when presentation is complete 4587 mState = STOPPING_1; 4588 } 4589 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4590 } 4591 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4592 thread->mLock.unlock(); 4593 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4594 thread->mLock.lock(); 4595 4596#ifdef ADD_BATTERY_DATA 4597 // to track the speaker usage 4598 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4599#endif 4600 } 4601 } 4602} 4603 4604void AudioFlinger::PlaybackThread::Track::pause() 4605{ 4606 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4607 sp<ThreadBase> thread = mThread.promote(); 4608 if (thread != 0) { 4609 Mutex::Autolock _l(thread->mLock); 4610 if (mState == ACTIVE || mState == RESUMING) { 4611 mState = PAUSING; 4612 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4613 if (!isOutputTrack()) { 4614 thread->mLock.unlock(); 4615 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4616 thread->mLock.lock(); 4617 4618#ifdef ADD_BATTERY_DATA 4619 // to track the speaker usage 4620 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4621#endif 4622 } 4623 } 4624 } 4625} 4626 4627void AudioFlinger::PlaybackThread::Track::flush() 4628{ 4629 ALOGV("flush(%d)", mName); 4630 sp<ThreadBase> thread = mThread.promote(); 4631 if (thread != 0) { 4632 Mutex::Autolock _l(thread->mLock); 4633 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4634 mState != PAUSING) { 4635 return; 4636 } 4637 // No point remaining in PAUSED state after a flush => go to 4638 // FLUSHED state 4639 mState = FLUSHED; 4640 // do not reset the track if it is still in the process of being stopped or paused. 4641 // this will be done by prepareTracks_l() when the track is stopped. 4642 // prepareTracks_l() will see mState == FLUSHED, then 4643 // remove from active track list, reset(), and trigger presentation complete 4644 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4645 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4646 reset(); 4647 } 4648 } 4649} 4650 4651void AudioFlinger::PlaybackThread::Track::reset() 4652{ 4653 // Do not reset twice to avoid discarding data written just after a flush and before 4654 // the audioflinger thread detects the track is stopped. 4655 if (!mResetDone) { 4656 TrackBase::reset(); 4657 // Force underrun condition to avoid false underrun callback until first data is 4658 // written to buffer 4659 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4660 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4661 mFillingUpStatus = FS_FILLING; 4662 mResetDone = true; 4663 if (mState == FLUSHED) { 4664 mState = IDLE; 4665 } 4666 } 4667} 4668 4669void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4670{ 4671 mMute = muted; 4672} 4673 4674status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4675{ 4676 status_t status = DEAD_OBJECT; 4677 sp<ThreadBase> thread = mThread.promote(); 4678 if (thread != 0) { 4679 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4680 status = playbackThread->attachAuxEffect(this, EffectId); 4681 } 4682 return status; 4683} 4684 4685void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4686{ 4687 mAuxEffectId = EffectId; 4688 mAuxBuffer = buffer; 4689} 4690 4691bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4692 size_t audioHalFrames) 4693{ 4694 // a track is considered presented when the total number of frames written to audio HAL 4695 // corresponds to the number of frames written when presentationComplete() is called for the 4696 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4697 if (mPresentationCompleteFrames == 0) { 4698 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4699 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4700 mPresentationCompleteFrames, audioHalFrames); 4701 } 4702 if (framesWritten >= mPresentationCompleteFrames) { 4703 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4704 mSessionId, framesWritten); 4705 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4706 return true; 4707 } 4708 return false; 4709} 4710 4711void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4712{ 4713 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4714 if (mSyncEvents[i]->type() == type) { 4715 mSyncEvents[i]->trigger(); 4716 mSyncEvents.removeAt(i); 4717 i--; 4718 } 4719 } 4720} 4721 4722// implement VolumeBufferProvider interface 4723 4724uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4725{ 4726 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4727 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4728 uint32_t vlr = mCblk->getVolumeLR(); 4729 uint32_t vl = vlr & 0xFFFF; 4730 uint32_t vr = vlr >> 16; 4731 // track volumes come from shared memory, so can't be trusted and must be clamped 4732 if (vl > MAX_GAIN_INT) { 4733 vl = MAX_GAIN_INT; 4734 } 4735 if (vr > MAX_GAIN_INT) { 4736 vr = MAX_GAIN_INT; 4737 } 4738 // now apply the cached master volume and stream type volume; 4739 // this is trusted but lacks any synchronization or barrier so may be stale 4740 float v = mCachedVolume; 4741 vl *= v; 4742 vr *= v; 4743 // re-combine into U4.16 4744 vlr = (vr << 16) | (vl & 0xFFFF); 4745 // FIXME look at mute, pause, and stop flags 4746 return vlr; 4747} 4748 4749status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4750{ 4751 if (mState == TERMINATED || mState == PAUSED || 4752 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4753 (mState == STOPPED)))) { 4754 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4755 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4756 event->cancel(); 4757 return INVALID_OPERATION; 4758 } 4759 TrackBase::setSyncEvent(event); 4760 return NO_ERROR; 4761} 4762 4763// timed audio tracks 4764 4765sp<AudioFlinger::PlaybackThread::TimedTrack> 4766AudioFlinger::PlaybackThread::TimedTrack::create( 4767 PlaybackThread *thread, 4768 const sp<Client>& client, 4769 audio_stream_type_t streamType, 4770 uint32_t sampleRate, 4771 audio_format_t format, 4772 uint32_t channelMask, 4773 int frameCount, 4774 const sp<IMemory>& sharedBuffer, 4775 int sessionId) { 4776 if (!client->reserveTimedTrack()) 4777 return NULL; 4778 4779 return new TimedTrack( 4780 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4781 sharedBuffer, sessionId); 4782} 4783 4784AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4785 PlaybackThread *thread, 4786 const sp<Client>& client, 4787 audio_stream_type_t streamType, 4788 uint32_t sampleRate, 4789 audio_format_t format, 4790 uint32_t channelMask, 4791 int frameCount, 4792 const sp<IMemory>& sharedBuffer, 4793 int sessionId) 4794 : Track(thread, client, streamType, sampleRate, format, channelMask, 4795 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4796 mQueueHeadInFlight(false), 4797 mTrimQueueHeadOnRelease(false), 4798 mFramesPendingInQueue(0), 4799 mTimedSilenceBuffer(NULL), 4800 mTimedSilenceBufferSize(0), 4801 mTimedAudioOutputOnTime(false), 4802 mMediaTimeTransformValid(false) 4803{ 4804 LocalClock lc; 4805 mLocalTimeFreq = lc.getLocalFreq(); 4806 4807 mLocalTimeToSampleTransform.a_zero = 0; 4808 mLocalTimeToSampleTransform.b_zero = 0; 4809 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4810 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4811 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4812 &mLocalTimeToSampleTransform.a_to_b_denom); 4813 4814 mMediaTimeToSampleTransform.a_zero = 0; 4815 mMediaTimeToSampleTransform.b_zero = 0; 4816 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4817 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4818 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4819 &mMediaTimeToSampleTransform.a_to_b_denom); 4820} 4821 4822AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4823 mClient->releaseTimedTrack(); 4824 delete [] mTimedSilenceBuffer; 4825} 4826 4827status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4828 size_t size, sp<IMemory>* buffer) { 4829 4830 Mutex::Autolock _l(mTimedBufferQueueLock); 4831 4832 trimTimedBufferQueue_l(); 4833 4834 // lazily initialize the shared memory heap for timed buffers 4835 if (mTimedMemoryDealer == NULL) { 4836 const int kTimedBufferHeapSize = 512 << 10; 4837 4838 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4839 "AudioFlingerTimed"); 4840 if (mTimedMemoryDealer == NULL) 4841 return NO_MEMORY; 4842 } 4843 4844 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4845 if (newBuffer == NULL) { 4846 newBuffer = mTimedMemoryDealer->allocate(size); 4847 if (newBuffer == NULL) 4848 return NO_MEMORY; 4849 } 4850 4851 *buffer = newBuffer; 4852 return NO_ERROR; 4853} 4854 4855// caller must hold mTimedBufferQueueLock 4856void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4857 int64_t mediaTimeNow; 4858 { 4859 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4860 if (!mMediaTimeTransformValid) 4861 return; 4862 4863 int64_t targetTimeNow; 4864 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4865 ? mCCHelper.getCommonTime(&targetTimeNow) 4866 : mCCHelper.getLocalTime(&targetTimeNow); 4867 4868 if (OK != res) 4869 return; 4870 4871 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4872 &mediaTimeNow)) { 4873 return; 4874 } 4875 } 4876 4877 size_t trimEnd; 4878 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4879 int64_t bufEnd; 4880 4881 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4882 // We have a next buffer. Just use its PTS as the PTS of the frame 4883 // following the last frame in this buffer. If the stream is sparse 4884 // (ie, there are deliberate gaps left in the stream which should be 4885 // filled with silence by the TimedAudioTrack), then this can result 4886 // in one extra buffer being left un-trimmed when it could have 4887 // been. In general, this is not typical, and we would rather 4888 // optimized away the TS calculation below for the more common case 4889 // where PTSes are contiguous. 4890 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4891 } else { 4892 // We have no next buffer. Compute the PTS of the frame following 4893 // the last frame in this buffer by computing the duration of of 4894 // this frame in media time units and adding it to the PTS of the 4895 // buffer. 4896 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4897 / mCblk->frameSize; 4898 4899 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4900 &bufEnd)) { 4901 ALOGE("Failed to convert frame count of %lld to media time" 4902 " duration" " (scale factor %d/%u) in %s", 4903 frameCount, 4904 mMediaTimeToSampleTransform.a_to_b_numer, 4905 mMediaTimeToSampleTransform.a_to_b_denom, 4906 __PRETTY_FUNCTION__); 4907 break; 4908 } 4909 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4910 } 4911 4912 if (bufEnd > mediaTimeNow) 4913 break; 4914 4915 // Is the buffer we want to use in the middle of a mix operation right 4916 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4917 // from the mixer which should be coming back shortly. 4918 if (!trimEnd && mQueueHeadInFlight) { 4919 mTrimQueueHeadOnRelease = true; 4920 } 4921 } 4922 4923 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4924 if (trimStart < trimEnd) { 4925 // Update the bookkeeping for framesReady() 4926 for (size_t i = trimStart; i < trimEnd; ++i) { 4927 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4928 } 4929 4930 // Now actually remove the buffers from the queue. 4931 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4932 } 4933} 4934 4935void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4936 const char* logTag) { 4937 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4938 "%s called (reason \"%s\"), but timed buffer queue has no" 4939 " elements to trim.", __FUNCTION__, logTag); 4940 4941 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4942 mTimedBufferQueue.removeAt(0); 4943} 4944 4945void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4946 const TimedBuffer& buf, 4947 const char* logTag) { 4948 uint32_t bufBytes = buf.buffer()->size(); 4949 uint32_t consumedAlready = buf.position(); 4950 4951 ALOG_ASSERT(consumedAlready <= bufBytes, 4952 "Bad bookkeeping while updating frames pending. Timed buffer is" 4953 " only %u bytes long, but claims to have consumed %u" 4954 " bytes. (update reason: \"%s\")", 4955 bufBytes, consumedAlready, logTag); 4956 4957 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4958 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4959 "Bad bookkeeping while updating frames pending. Should have at" 4960 " least %u queued frames, but we think we have only %u. (update" 4961 " reason: \"%s\")", 4962 bufFrames, mFramesPendingInQueue, logTag); 4963 4964 mFramesPendingInQueue -= bufFrames; 4965} 4966 4967status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4968 const sp<IMemory>& buffer, int64_t pts) { 4969 4970 { 4971 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4972 if (!mMediaTimeTransformValid) 4973 return INVALID_OPERATION; 4974 } 4975 4976 Mutex::Autolock _l(mTimedBufferQueueLock); 4977 4978 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4979 mFramesPendingInQueue += bufFrames; 4980 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4981 4982 return NO_ERROR; 4983} 4984 4985status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4986 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4987 4988 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4989 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4990 target); 4991 4992 if (!(target == TimedAudioTrack::LOCAL_TIME || 4993 target == TimedAudioTrack::COMMON_TIME)) { 4994 return BAD_VALUE; 4995 } 4996 4997 Mutex::Autolock lock(mMediaTimeTransformLock); 4998 mMediaTimeTransform = xform; 4999 mMediaTimeTransformTarget = target; 5000 mMediaTimeTransformValid = true; 5001 5002 return NO_ERROR; 5003} 5004 5005#define min(a, b) ((a) < (b) ? (a) : (b)) 5006 5007// implementation of getNextBuffer for tracks whose buffers have timestamps 5008status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5009 AudioBufferProvider::Buffer* buffer, int64_t pts) 5010{ 5011 if (pts == AudioBufferProvider::kInvalidPTS) { 5012 buffer->raw = 0; 5013 buffer->frameCount = 0; 5014 mTimedAudioOutputOnTime = false; 5015 return INVALID_OPERATION; 5016 } 5017 5018 Mutex::Autolock _l(mTimedBufferQueueLock); 5019 5020 ALOG_ASSERT(!mQueueHeadInFlight, 5021 "getNextBuffer called without releaseBuffer!"); 5022 5023 while (true) { 5024 5025 // if we have no timed buffers, then fail 5026 if (mTimedBufferQueue.isEmpty()) { 5027 buffer->raw = 0; 5028 buffer->frameCount = 0; 5029 return NOT_ENOUGH_DATA; 5030 } 5031 5032 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5033 5034 // calculate the PTS of the head of the timed buffer queue expressed in 5035 // local time 5036 int64_t headLocalPTS; 5037 { 5038 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5039 5040 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5041 5042 if (mMediaTimeTransform.a_to_b_denom == 0) { 5043 // the transform represents a pause, so yield silence 5044 timedYieldSilence_l(buffer->frameCount, buffer); 5045 return NO_ERROR; 5046 } 5047 5048 int64_t transformedPTS; 5049 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5050 &transformedPTS)) { 5051 // the transform failed. this shouldn't happen, but if it does 5052 // then just drop this buffer 5053 ALOGW("timedGetNextBuffer transform failed"); 5054 buffer->raw = 0; 5055 buffer->frameCount = 0; 5056 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5057 return NO_ERROR; 5058 } 5059 5060 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5061 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5062 &headLocalPTS)) { 5063 buffer->raw = 0; 5064 buffer->frameCount = 0; 5065 return INVALID_OPERATION; 5066 } 5067 } else { 5068 headLocalPTS = transformedPTS; 5069 } 5070 } 5071 5072 // adjust the head buffer's PTS to reflect the portion of the head buffer 5073 // that has already been consumed 5074 int64_t effectivePTS = headLocalPTS + 5075 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5076 5077 // Calculate the delta in samples between the head of the input buffer 5078 // queue and the start of the next output buffer that will be written. 5079 // If the transformation fails because of over or underflow, it means 5080 // that the sample's position in the output stream is so far out of 5081 // whack that it should just be dropped. 5082 int64_t sampleDelta; 5083 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5084 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5085 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5086 " mix"); 5087 continue; 5088 } 5089 if (!mLocalTimeToSampleTransform.doForwardTransform( 5090 (effectivePTS - pts) << 32, &sampleDelta)) { 5091 ALOGV("*** too late during sample rate transform: dropped buffer"); 5092 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5093 continue; 5094 } 5095 5096 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5097 " sampleDelta=[%d.%08x]", 5098 head.pts(), head.position(), pts, 5099 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5100 + (sampleDelta >> 32)), 5101 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5102 5103 // if the delta between the ideal placement for the next input sample and 5104 // the current output position is within this threshold, then we will 5105 // concatenate the next input samples to the previous output 5106 const int64_t kSampleContinuityThreshold = 5107 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5108 5109 // if this is the first buffer of audio that we're emitting from this track 5110 // then it should be almost exactly on time. 5111 const int64_t kSampleStartupThreshold = 1LL << 32; 5112 5113 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5114 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5115 // the next input is close enough to being on time, so concatenate it 5116 // with the last output 5117 timedYieldSamples_l(buffer); 5118 5119 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5120 head.position(), buffer->frameCount); 5121 return NO_ERROR; 5122 } 5123 5124 // Looks like our output is not on time. Reset our on timed status. 5125 // Next time we mix samples from our input queue, then should be within 5126 // the StartupThreshold. 5127 mTimedAudioOutputOnTime = false; 5128 if (sampleDelta > 0) { 5129 // the gap between the current output position and the proper start of 5130 // the next input sample is too big, so fill it with silence 5131 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5132 5133 timedYieldSilence_l(framesUntilNextInput, buffer); 5134 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5135 return NO_ERROR; 5136 } else { 5137 // the next input sample is late 5138 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5139 size_t onTimeSamplePosition = 5140 head.position() + lateFrames * mCblk->frameSize; 5141 5142 if (onTimeSamplePosition > head.buffer()->size()) { 5143 // all the remaining samples in the head are too late, so 5144 // drop it and move on 5145 ALOGV("*** too late: dropped buffer"); 5146 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5147 continue; 5148 } else { 5149 // skip over the late samples 5150 head.setPosition(onTimeSamplePosition); 5151 5152 // yield the available samples 5153 timedYieldSamples_l(buffer); 5154 5155 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5156 return NO_ERROR; 5157 } 5158 } 5159 } 5160} 5161 5162// Yield samples from the timed buffer queue head up to the given output 5163// buffer's capacity. 5164// 5165// Caller must hold mTimedBufferQueueLock 5166void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5167 AudioBufferProvider::Buffer* buffer) { 5168 5169 const TimedBuffer& head = mTimedBufferQueue[0]; 5170 5171 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5172 head.position()); 5173 5174 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5175 mCblk->frameSize); 5176 size_t framesRequested = buffer->frameCount; 5177 buffer->frameCount = min(framesLeftInHead, framesRequested); 5178 5179 mQueueHeadInFlight = true; 5180 mTimedAudioOutputOnTime = true; 5181} 5182 5183// Yield samples of silence up to the given output buffer's capacity 5184// 5185// Caller must hold mTimedBufferQueueLock 5186void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5187 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5188 5189 // lazily allocate a buffer filled with silence 5190 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5191 delete [] mTimedSilenceBuffer; 5192 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5193 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5194 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5195 } 5196 5197 buffer->raw = mTimedSilenceBuffer; 5198 size_t framesRequested = buffer->frameCount; 5199 buffer->frameCount = min(numFrames, framesRequested); 5200 5201 mTimedAudioOutputOnTime = false; 5202} 5203 5204// AudioBufferProvider interface 5205void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5206 AudioBufferProvider::Buffer* buffer) { 5207 5208 Mutex::Autolock _l(mTimedBufferQueueLock); 5209 5210 // If the buffer which was just released is part of the buffer at the head 5211 // of the queue, be sure to update the amt of the buffer which has been 5212 // consumed. If the buffer being returned is not part of the head of the 5213 // queue, its either because the buffer is part of the silence buffer, or 5214 // because the head of the timed queue was trimmed after the mixer called 5215 // getNextBuffer but before the mixer called releaseBuffer. 5216 if (buffer->raw == mTimedSilenceBuffer) { 5217 ALOG_ASSERT(!mQueueHeadInFlight, 5218 "Queue head in flight during release of silence buffer!"); 5219 goto done; 5220 } 5221 5222 ALOG_ASSERT(mQueueHeadInFlight, 5223 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5224 " head in flight."); 5225 5226 if (mTimedBufferQueue.size()) { 5227 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5228 5229 void* start = head.buffer()->pointer(); 5230 void* end = reinterpret_cast<void*>( 5231 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5232 + head.buffer()->size()); 5233 5234 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5235 "released buffer not within the head of the timed buffer" 5236 " queue; qHead = [%p, %p], released buffer = %p", 5237 start, end, buffer->raw); 5238 5239 head.setPosition(head.position() + 5240 (buffer->frameCount * mCblk->frameSize)); 5241 mQueueHeadInFlight = false; 5242 5243 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5244 "Bad bookkeeping during releaseBuffer! Should have at" 5245 " least %u queued frames, but we think we have only %u", 5246 buffer->frameCount, mFramesPendingInQueue); 5247 5248 mFramesPendingInQueue -= buffer->frameCount; 5249 5250 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5251 || mTrimQueueHeadOnRelease) { 5252 trimTimedBufferQueueHead_l("releaseBuffer"); 5253 mTrimQueueHeadOnRelease = false; 5254 } 5255 } else { 5256 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5257 " buffers in the timed buffer queue"); 5258 } 5259 5260done: 5261 buffer->raw = 0; 5262 buffer->frameCount = 0; 5263} 5264 5265size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5266 Mutex::Autolock _l(mTimedBufferQueueLock); 5267 return mFramesPendingInQueue; 5268} 5269 5270AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5271 : mPTS(0), mPosition(0) {} 5272 5273AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5274 const sp<IMemory>& buffer, int64_t pts) 5275 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5276 5277// ---------------------------------------------------------------------------- 5278 5279// RecordTrack constructor must be called with AudioFlinger::mLock held 5280AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5281 RecordThread *thread, 5282 const sp<Client>& client, 5283 uint32_t sampleRate, 5284 audio_format_t format, 5285 uint32_t channelMask, 5286 int frameCount, 5287 int sessionId) 5288 : TrackBase(thread, client, sampleRate, format, 5289 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5290 mOverflow(false) 5291{ 5292 if (mCblk != NULL) { 5293 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5294 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5295 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5296 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5297 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5298 } else { 5299 mCblk->frameSize = sizeof(int8_t); 5300 } 5301 } 5302} 5303 5304AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5305{ 5306 sp<ThreadBase> thread = mThread.promote(); 5307 if (thread != 0) { 5308 AudioSystem::releaseInput(thread->id()); 5309 } 5310} 5311 5312// AudioBufferProvider interface 5313status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5314{ 5315 audio_track_cblk_t* cblk = this->cblk(); 5316 uint32_t framesAvail; 5317 uint32_t framesReq = buffer->frameCount; 5318 5319 // Check if last stepServer failed, try to step now 5320 if (mStepServerFailed) { 5321 if (!step()) goto getNextBuffer_exit; 5322 ALOGV("stepServer recovered"); 5323 mStepServerFailed = false; 5324 } 5325 5326 framesAvail = cblk->framesAvailable_l(); 5327 5328 if (CC_LIKELY(framesAvail)) { 5329 uint32_t s = cblk->server; 5330 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5331 5332 if (framesReq > framesAvail) { 5333 framesReq = framesAvail; 5334 } 5335 if (framesReq > bufferEnd - s) { 5336 framesReq = bufferEnd - s; 5337 } 5338 5339 buffer->raw = getBuffer(s, framesReq); 5340 if (buffer->raw == NULL) goto getNextBuffer_exit; 5341 5342 buffer->frameCount = framesReq; 5343 return NO_ERROR; 5344 } 5345 5346getNextBuffer_exit: 5347 buffer->raw = NULL; 5348 buffer->frameCount = 0; 5349 return NOT_ENOUGH_DATA; 5350} 5351 5352status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5353 int triggerSession) 5354{ 5355 sp<ThreadBase> thread = mThread.promote(); 5356 if (thread != 0) { 5357 RecordThread *recordThread = (RecordThread *)thread.get(); 5358 return recordThread->start(this, event, triggerSession); 5359 } else { 5360 return BAD_VALUE; 5361 } 5362} 5363 5364void AudioFlinger::RecordThread::RecordTrack::stop() 5365{ 5366 sp<ThreadBase> thread = mThread.promote(); 5367 if (thread != 0) { 5368 RecordThread *recordThread = (RecordThread *)thread.get(); 5369 recordThread->stop(this); 5370 TrackBase::reset(); 5371 // Force overrun condition to avoid false overrun callback until first data is 5372 // read from buffer 5373 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5374 } 5375} 5376 5377void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5378{ 5379 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5380 (mClient == 0) ? getpid_cached : mClient->pid(), 5381 mFormat, 5382 mChannelMask, 5383 mSessionId, 5384 mFrameCount, 5385 mState, 5386 mCblk->sampleRate, 5387 mCblk->server, 5388 mCblk->user); 5389} 5390 5391 5392// ---------------------------------------------------------------------------- 5393 5394AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5395 PlaybackThread *playbackThread, 5396 DuplicatingThread *sourceThread, 5397 uint32_t sampleRate, 5398 audio_format_t format, 5399 uint32_t channelMask, 5400 int frameCount) 5401 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5402 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5403 mActive(false), mSourceThread(sourceThread) 5404{ 5405 5406 if (mCblk != NULL) { 5407 mCblk->flags |= CBLK_DIRECTION_OUT; 5408 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5409 mOutBuffer.frameCount = 0; 5410 playbackThread->mTracks.add(this); 5411 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5412 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5413 mCblk, mBuffer, mCblk->buffers, 5414 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5415 } else { 5416 ALOGW("Error creating output track on thread %p", playbackThread); 5417 } 5418} 5419 5420AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5421{ 5422 clearBufferQueue(); 5423} 5424 5425status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5426 int triggerSession) 5427{ 5428 status_t status = Track::start(event, triggerSession); 5429 if (status != NO_ERROR) { 5430 return status; 5431 } 5432 5433 mActive = true; 5434 mRetryCount = 127; 5435 return status; 5436} 5437 5438void AudioFlinger::PlaybackThread::OutputTrack::stop() 5439{ 5440 Track::stop(); 5441 clearBufferQueue(); 5442 mOutBuffer.frameCount = 0; 5443 mActive = false; 5444} 5445 5446bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5447{ 5448 Buffer *pInBuffer; 5449 Buffer inBuffer; 5450 uint32_t channelCount = mChannelCount; 5451 bool outputBufferFull = false; 5452 inBuffer.frameCount = frames; 5453 inBuffer.i16 = data; 5454 5455 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5456 5457 if (!mActive && frames != 0) { 5458 start(); 5459 sp<ThreadBase> thread = mThread.promote(); 5460 if (thread != 0) { 5461 MixerThread *mixerThread = (MixerThread *)thread.get(); 5462 if (mCblk->frameCount > frames){ 5463 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5464 uint32_t startFrames = (mCblk->frameCount - frames); 5465 pInBuffer = new Buffer; 5466 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5467 pInBuffer->frameCount = startFrames; 5468 pInBuffer->i16 = pInBuffer->mBuffer; 5469 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5470 mBufferQueue.add(pInBuffer); 5471 } else { 5472 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5473 } 5474 } 5475 } 5476 } 5477 5478 while (waitTimeLeftMs) { 5479 // First write pending buffers, then new data 5480 if (mBufferQueue.size()) { 5481 pInBuffer = mBufferQueue.itemAt(0); 5482 } else { 5483 pInBuffer = &inBuffer; 5484 } 5485 5486 if (pInBuffer->frameCount == 0) { 5487 break; 5488 } 5489 5490 if (mOutBuffer.frameCount == 0) { 5491 mOutBuffer.frameCount = pInBuffer->frameCount; 5492 nsecs_t startTime = systemTime(); 5493 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5494 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5495 outputBufferFull = true; 5496 break; 5497 } 5498 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5499 if (waitTimeLeftMs >= waitTimeMs) { 5500 waitTimeLeftMs -= waitTimeMs; 5501 } else { 5502 waitTimeLeftMs = 0; 5503 } 5504 } 5505 5506 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5507 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5508 mCblk->stepUser(outFrames); 5509 pInBuffer->frameCount -= outFrames; 5510 pInBuffer->i16 += outFrames * channelCount; 5511 mOutBuffer.frameCount -= outFrames; 5512 mOutBuffer.i16 += outFrames * channelCount; 5513 5514 if (pInBuffer->frameCount == 0) { 5515 if (mBufferQueue.size()) { 5516 mBufferQueue.removeAt(0); 5517 delete [] pInBuffer->mBuffer; 5518 delete pInBuffer; 5519 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5520 } else { 5521 break; 5522 } 5523 } 5524 } 5525 5526 // If we could not write all frames, allocate a buffer and queue it for next time. 5527 if (inBuffer.frameCount) { 5528 sp<ThreadBase> thread = mThread.promote(); 5529 if (thread != 0 && !thread->standby()) { 5530 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5531 pInBuffer = new Buffer; 5532 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5533 pInBuffer->frameCount = inBuffer.frameCount; 5534 pInBuffer->i16 = pInBuffer->mBuffer; 5535 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5536 mBufferQueue.add(pInBuffer); 5537 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5538 } else { 5539 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5540 } 5541 } 5542 } 5543 5544 // Calling write() with a 0 length buffer, means that no more data will be written: 5545 // If no more buffers are pending, fill output track buffer to make sure it is started 5546 // by output mixer. 5547 if (frames == 0 && mBufferQueue.size() == 0) { 5548 if (mCblk->user < mCblk->frameCount) { 5549 frames = mCblk->frameCount - mCblk->user; 5550 pInBuffer = new Buffer; 5551 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5552 pInBuffer->frameCount = frames; 5553 pInBuffer->i16 = pInBuffer->mBuffer; 5554 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5555 mBufferQueue.add(pInBuffer); 5556 } else if (mActive) { 5557 stop(); 5558 } 5559 } 5560 5561 return outputBufferFull; 5562} 5563 5564status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5565{ 5566 int active; 5567 status_t result; 5568 audio_track_cblk_t* cblk = mCblk; 5569 uint32_t framesReq = buffer->frameCount; 5570 5571// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5572 buffer->frameCount = 0; 5573 5574 uint32_t framesAvail = cblk->framesAvailable(); 5575 5576 5577 if (framesAvail == 0) { 5578 Mutex::Autolock _l(cblk->lock); 5579 goto start_loop_here; 5580 while (framesAvail == 0) { 5581 active = mActive; 5582 if (CC_UNLIKELY(!active)) { 5583 ALOGV("Not active and NO_MORE_BUFFERS"); 5584 return NO_MORE_BUFFERS; 5585 } 5586 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5587 if (result != NO_ERROR) { 5588 return NO_MORE_BUFFERS; 5589 } 5590 // read the server count again 5591 start_loop_here: 5592 framesAvail = cblk->framesAvailable_l(); 5593 } 5594 } 5595 5596// if (framesAvail < framesReq) { 5597// return NO_MORE_BUFFERS; 5598// } 5599 5600 if (framesReq > framesAvail) { 5601 framesReq = framesAvail; 5602 } 5603 5604 uint32_t u = cblk->user; 5605 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5606 5607 if (framesReq > bufferEnd - u) { 5608 framesReq = bufferEnd - u; 5609 } 5610 5611 buffer->frameCount = framesReq; 5612 buffer->raw = (void *)cblk->buffer(u); 5613 return NO_ERROR; 5614} 5615 5616 5617void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5618{ 5619 size_t size = mBufferQueue.size(); 5620 5621 for (size_t i = 0; i < size; i++) { 5622 Buffer *pBuffer = mBufferQueue.itemAt(i); 5623 delete [] pBuffer->mBuffer; 5624 delete pBuffer; 5625 } 5626 mBufferQueue.clear(); 5627} 5628 5629// ---------------------------------------------------------------------------- 5630 5631AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5632 : RefBase(), 5633 mAudioFlinger(audioFlinger), 5634 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5635 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5636 mPid(pid), 5637 mTimedTrackCount(0) 5638{ 5639 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5640} 5641 5642// Client destructor must be called with AudioFlinger::mLock held 5643AudioFlinger::Client::~Client() 5644{ 5645 mAudioFlinger->removeClient_l(mPid); 5646} 5647 5648sp<MemoryDealer> AudioFlinger::Client::heap() const 5649{ 5650 return mMemoryDealer; 5651} 5652 5653// Reserve one of the limited slots for a timed audio track associated 5654// with this client 5655bool AudioFlinger::Client::reserveTimedTrack() 5656{ 5657 const int kMaxTimedTracksPerClient = 4; 5658 5659 Mutex::Autolock _l(mTimedTrackLock); 5660 5661 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5662 ALOGW("can not create timed track - pid %d has exceeded the limit", 5663 mPid); 5664 return false; 5665 } 5666 5667 mTimedTrackCount++; 5668 return true; 5669} 5670 5671// Release a slot for a timed audio track 5672void AudioFlinger::Client::releaseTimedTrack() 5673{ 5674 Mutex::Autolock _l(mTimedTrackLock); 5675 mTimedTrackCount--; 5676} 5677 5678// ---------------------------------------------------------------------------- 5679 5680AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5681 const sp<IAudioFlingerClient>& client, 5682 pid_t pid) 5683 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5684{ 5685} 5686 5687AudioFlinger::NotificationClient::~NotificationClient() 5688{ 5689} 5690 5691void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5692{ 5693 sp<NotificationClient> keep(this); 5694 mAudioFlinger->removeNotificationClient(mPid); 5695} 5696 5697// ---------------------------------------------------------------------------- 5698 5699AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5700 : BnAudioTrack(), 5701 mTrack(track) 5702{ 5703} 5704 5705AudioFlinger::TrackHandle::~TrackHandle() { 5706 // just stop the track on deletion, associated resources 5707 // will be freed from the main thread once all pending buffers have 5708 // been played. Unless it's not in the active track list, in which 5709 // case we free everything now... 5710 mTrack->destroy(); 5711} 5712 5713sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5714 return mTrack->getCblk(); 5715} 5716 5717status_t AudioFlinger::TrackHandle::start() { 5718 return mTrack->start(); 5719} 5720 5721void AudioFlinger::TrackHandle::stop() { 5722 mTrack->stop(); 5723} 5724 5725void AudioFlinger::TrackHandle::flush() { 5726 mTrack->flush(); 5727} 5728 5729void AudioFlinger::TrackHandle::mute(bool e) { 5730 mTrack->mute(e); 5731} 5732 5733void AudioFlinger::TrackHandle::pause() { 5734 mTrack->pause(); 5735} 5736 5737status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5738{ 5739 return mTrack->attachAuxEffect(EffectId); 5740} 5741 5742status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5743 sp<IMemory>* buffer) { 5744 if (!mTrack->isTimedTrack()) 5745 return INVALID_OPERATION; 5746 5747 PlaybackThread::TimedTrack* tt = 5748 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5749 return tt->allocateTimedBuffer(size, buffer); 5750} 5751 5752status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5753 int64_t pts) { 5754 if (!mTrack->isTimedTrack()) 5755 return INVALID_OPERATION; 5756 5757 PlaybackThread::TimedTrack* tt = 5758 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5759 return tt->queueTimedBuffer(buffer, pts); 5760} 5761 5762status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5763 const LinearTransform& xform, int target) { 5764 5765 if (!mTrack->isTimedTrack()) 5766 return INVALID_OPERATION; 5767 5768 PlaybackThread::TimedTrack* tt = 5769 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5770 return tt->setMediaTimeTransform( 5771 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5772} 5773 5774status_t AudioFlinger::TrackHandle::onTransact( 5775 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5776{ 5777 return BnAudioTrack::onTransact(code, data, reply, flags); 5778} 5779 5780// ---------------------------------------------------------------------------- 5781 5782sp<IAudioRecord> AudioFlinger::openRecord( 5783 pid_t pid, 5784 audio_io_handle_t input, 5785 uint32_t sampleRate, 5786 audio_format_t format, 5787 uint32_t channelMask, 5788 int frameCount, 5789 IAudioFlinger::track_flags_t flags, 5790 int *sessionId, 5791 status_t *status) 5792{ 5793 sp<RecordThread::RecordTrack> recordTrack; 5794 sp<RecordHandle> recordHandle; 5795 sp<Client> client; 5796 status_t lStatus; 5797 RecordThread *thread; 5798 size_t inFrameCount; 5799 int lSessionId; 5800 5801 // check calling permissions 5802 if (!recordingAllowed()) { 5803 lStatus = PERMISSION_DENIED; 5804 goto Exit; 5805 } 5806 5807 // add client to list 5808 { // scope for mLock 5809 Mutex::Autolock _l(mLock); 5810 thread = checkRecordThread_l(input); 5811 if (thread == NULL) { 5812 lStatus = BAD_VALUE; 5813 goto Exit; 5814 } 5815 5816 client = registerPid_l(pid); 5817 5818 // If no audio session id is provided, create one here 5819 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5820 lSessionId = *sessionId; 5821 } else { 5822 lSessionId = nextUniqueId(); 5823 if (sessionId != NULL) { 5824 *sessionId = lSessionId; 5825 } 5826 } 5827 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5828 recordTrack = thread->createRecordTrack_l(client, 5829 sampleRate, 5830 format, 5831 channelMask, 5832 frameCount, 5833 lSessionId, 5834 &lStatus); 5835 } 5836 if (lStatus != NO_ERROR) { 5837 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5838 // destructor is called by the TrackBase destructor with mLock held 5839 client.clear(); 5840 recordTrack.clear(); 5841 goto Exit; 5842 } 5843 5844 // return to handle to client 5845 recordHandle = new RecordHandle(recordTrack); 5846 lStatus = NO_ERROR; 5847 5848Exit: 5849 if (status) { 5850 *status = lStatus; 5851 } 5852 return recordHandle; 5853} 5854 5855// ---------------------------------------------------------------------------- 5856 5857AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5858 : BnAudioRecord(), 5859 mRecordTrack(recordTrack) 5860{ 5861} 5862 5863AudioFlinger::RecordHandle::~RecordHandle() { 5864 stop(); 5865} 5866 5867sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5868 return mRecordTrack->getCblk(); 5869} 5870 5871status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5872 ALOGV("RecordHandle::start()"); 5873 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5874} 5875 5876void AudioFlinger::RecordHandle::stop() { 5877 ALOGV("RecordHandle::stop()"); 5878 mRecordTrack->stop(); 5879} 5880 5881status_t AudioFlinger::RecordHandle::onTransact( 5882 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5883{ 5884 return BnAudioRecord::onTransact(code, data, reply, flags); 5885} 5886 5887// ---------------------------------------------------------------------------- 5888 5889AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5890 AudioStreamIn *input, 5891 uint32_t sampleRate, 5892 uint32_t channels, 5893 audio_io_handle_t id, 5894 uint32_t device) : 5895 ThreadBase(audioFlinger, id, device, RECORD), 5896 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5897 // mRsmpInIndex and mInputBytes set by readInputParameters() 5898 mReqChannelCount(popcount(channels)), 5899 mReqSampleRate(sampleRate) 5900 // mBytesRead is only meaningful while active, and so is cleared in start() 5901 // (but might be better to also clear here for dump?) 5902{ 5903 snprintf(mName, kNameLength, "AudioIn_%X", id); 5904 5905 readInputParameters(); 5906} 5907 5908 5909AudioFlinger::RecordThread::~RecordThread() 5910{ 5911 delete[] mRsmpInBuffer; 5912 delete mResampler; 5913 delete[] mRsmpOutBuffer; 5914} 5915 5916void AudioFlinger::RecordThread::onFirstRef() 5917{ 5918 run(mName, PRIORITY_URGENT_AUDIO); 5919} 5920 5921status_t AudioFlinger::RecordThread::readyToRun() 5922{ 5923 status_t status = initCheck(); 5924 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5925 return status; 5926} 5927 5928bool AudioFlinger::RecordThread::threadLoop() 5929{ 5930 AudioBufferProvider::Buffer buffer; 5931 sp<RecordTrack> activeTrack; 5932 Vector< sp<EffectChain> > effectChains; 5933 5934 nsecs_t lastWarning = 0; 5935 5936 acquireWakeLock(); 5937 5938 // start recording 5939 while (!exitPending()) { 5940 5941 processConfigEvents(); 5942 5943 { // scope for mLock 5944 Mutex::Autolock _l(mLock); 5945 checkForNewParameters_l(); 5946 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5947 if (!mStandby) { 5948 mInput->stream->common.standby(&mInput->stream->common); 5949 mStandby = true; 5950 } 5951 5952 if (exitPending()) break; 5953 5954 releaseWakeLock_l(); 5955 ALOGV("RecordThread: loop stopping"); 5956 // go to sleep 5957 mWaitWorkCV.wait(mLock); 5958 ALOGV("RecordThread: loop starting"); 5959 acquireWakeLock_l(); 5960 continue; 5961 } 5962 if (mActiveTrack != 0) { 5963 if (mActiveTrack->mState == TrackBase::PAUSING) { 5964 if (!mStandby) { 5965 mInput->stream->common.standby(&mInput->stream->common); 5966 mStandby = true; 5967 } 5968 mActiveTrack.clear(); 5969 mStartStopCond.broadcast(); 5970 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5971 if (mReqChannelCount != mActiveTrack->channelCount()) { 5972 mActiveTrack.clear(); 5973 mStartStopCond.broadcast(); 5974 } else if (mBytesRead != 0) { 5975 // record start succeeds only if first read from audio input 5976 // succeeds 5977 if (mBytesRead > 0) { 5978 mActiveTrack->mState = TrackBase::ACTIVE; 5979 } else { 5980 mActiveTrack.clear(); 5981 } 5982 mStartStopCond.broadcast(); 5983 } 5984 mStandby = false; 5985 } 5986 } 5987 lockEffectChains_l(effectChains); 5988 } 5989 5990 if (mActiveTrack != 0) { 5991 if (mActiveTrack->mState != TrackBase::ACTIVE && 5992 mActiveTrack->mState != TrackBase::RESUMING) { 5993 unlockEffectChains(effectChains); 5994 usleep(kRecordThreadSleepUs); 5995 continue; 5996 } 5997 for (size_t i = 0; i < effectChains.size(); i ++) { 5998 effectChains[i]->process_l(); 5999 } 6000 6001 buffer.frameCount = mFrameCount; 6002 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6003 size_t framesOut = buffer.frameCount; 6004 if (mResampler == NULL) { 6005 // no resampling 6006 while (framesOut) { 6007 size_t framesIn = mFrameCount - mRsmpInIndex; 6008 if (framesIn) { 6009 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6010 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6011 if (framesIn > framesOut) 6012 framesIn = framesOut; 6013 mRsmpInIndex += framesIn; 6014 framesOut -= framesIn; 6015 if ((int)mChannelCount == mReqChannelCount || 6016 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6017 memcpy(dst, src, framesIn * mFrameSize); 6018 } else { 6019 int16_t *src16 = (int16_t *)src; 6020 int16_t *dst16 = (int16_t *)dst; 6021 if (mChannelCount == 1) { 6022 while (framesIn--) { 6023 *dst16++ = *src16; 6024 *dst16++ = *src16++; 6025 } 6026 } else { 6027 while (framesIn--) { 6028 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6029 src16 += 2; 6030 } 6031 } 6032 } 6033 } 6034 if (framesOut && mFrameCount == mRsmpInIndex) { 6035 if (framesOut == mFrameCount && 6036 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6037 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6038 framesOut = 0; 6039 } else { 6040 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6041 mRsmpInIndex = 0; 6042 } 6043 if (mBytesRead < 0) { 6044 ALOGE("Error reading audio input"); 6045 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6046 // Force input into standby so that it tries to 6047 // recover at next read attempt 6048 mInput->stream->common.standby(&mInput->stream->common); 6049 usleep(kRecordThreadSleepUs); 6050 } 6051 mRsmpInIndex = mFrameCount; 6052 framesOut = 0; 6053 buffer.frameCount = 0; 6054 } 6055 } 6056 } 6057 } else { 6058 // resampling 6059 6060 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6061 // alter output frame count as if we were expecting stereo samples 6062 if (mChannelCount == 1 && mReqChannelCount == 1) { 6063 framesOut >>= 1; 6064 } 6065 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6066 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6067 // are 32 bit aligned which should be always true. 6068 if (mChannelCount == 2 && mReqChannelCount == 1) { 6069 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6070 // the resampler always outputs stereo samples: do post stereo to mono conversion 6071 int16_t *src = (int16_t *)mRsmpOutBuffer; 6072 int16_t *dst = buffer.i16; 6073 while (framesOut--) { 6074 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6075 src += 2; 6076 } 6077 } else { 6078 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6079 } 6080 6081 } 6082 if (mFramestoDrop == 0) { 6083 mActiveTrack->releaseBuffer(&buffer); 6084 } else { 6085 if (mFramestoDrop > 0) { 6086 mFramestoDrop -= buffer.frameCount; 6087 if (mFramestoDrop <= 0) { 6088 clearSyncStartEvent(); 6089 } 6090 } else { 6091 mFramestoDrop += buffer.frameCount; 6092 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6093 mSyncStartEvent->isCancelled()) { 6094 ALOGW("Synced record %s, session %d, trigger session %d", 6095 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6096 mActiveTrack->sessionId(), 6097 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6098 clearSyncStartEvent(); 6099 } 6100 } 6101 } 6102 mActiveTrack->overflow(); 6103 } 6104 // client isn't retrieving buffers fast enough 6105 else { 6106 if (!mActiveTrack->setOverflow()) { 6107 nsecs_t now = systemTime(); 6108 if ((now - lastWarning) > kWarningThrottleNs) { 6109 ALOGW("RecordThread: buffer overflow"); 6110 lastWarning = now; 6111 } 6112 } 6113 // Release the processor for a while before asking for a new buffer. 6114 // This will give the application more chance to read from the buffer and 6115 // clear the overflow. 6116 usleep(kRecordThreadSleepUs); 6117 } 6118 } 6119 // enable changes in effect chain 6120 unlockEffectChains(effectChains); 6121 effectChains.clear(); 6122 } 6123 6124 if (!mStandby) { 6125 mInput->stream->common.standby(&mInput->stream->common); 6126 } 6127 mActiveTrack.clear(); 6128 6129 mStartStopCond.broadcast(); 6130 6131 releaseWakeLock(); 6132 6133 ALOGV("RecordThread %p exiting", this); 6134 return false; 6135} 6136 6137 6138sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6139 const sp<AudioFlinger::Client>& client, 6140 uint32_t sampleRate, 6141 audio_format_t format, 6142 int channelMask, 6143 int frameCount, 6144 int sessionId, 6145 status_t *status) 6146{ 6147 sp<RecordTrack> track; 6148 status_t lStatus; 6149 6150 lStatus = initCheck(); 6151 if (lStatus != NO_ERROR) { 6152 ALOGE("Audio driver not initialized."); 6153 goto Exit; 6154 } 6155 6156 { // scope for mLock 6157 Mutex::Autolock _l(mLock); 6158 6159 track = new RecordTrack(this, client, sampleRate, 6160 format, channelMask, frameCount, sessionId); 6161 6162 if (track->getCblk() == 0) { 6163 lStatus = NO_MEMORY; 6164 goto Exit; 6165 } 6166 6167 mTrack = track.get(); 6168 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6169 bool suspend = audio_is_bluetooth_sco_device( 6170 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6171 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6172 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6173 } 6174 lStatus = NO_ERROR; 6175 6176Exit: 6177 if (status) { 6178 *status = lStatus; 6179 } 6180 return track; 6181} 6182 6183status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6184 AudioSystem::sync_event_t event, 6185 int triggerSession) 6186{ 6187 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6188 sp<ThreadBase> strongMe = this; 6189 status_t status = NO_ERROR; 6190 6191 if (event == AudioSystem::SYNC_EVENT_NONE) { 6192 clearSyncStartEvent(); 6193 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6194 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6195 triggerSession, 6196 recordTrack->sessionId(), 6197 syncStartEventCallback, 6198 this); 6199 // Sync event can be cancelled by the trigger session if the track is not in a 6200 // compatible state in which case we start record immediately 6201 if (mSyncStartEvent->isCancelled()) { 6202 clearSyncStartEvent(); 6203 } else { 6204 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6205 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6206 } 6207 } 6208 6209 { 6210 AutoMutex lock(mLock); 6211 if (mActiveTrack != 0) { 6212 if (recordTrack != mActiveTrack.get()) { 6213 status = -EBUSY; 6214 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6215 mActiveTrack->mState = TrackBase::ACTIVE; 6216 } 6217 return status; 6218 } 6219 6220 recordTrack->mState = TrackBase::IDLE; 6221 mActiveTrack = recordTrack; 6222 mLock.unlock(); 6223 status_t status = AudioSystem::startInput(mId); 6224 mLock.lock(); 6225 if (status != NO_ERROR) { 6226 mActiveTrack.clear(); 6227 clearSyncStartEvent(); 6228 return status; 6229 } 6230 mRsmpInIndex = mFrameCount; 6231 mBytesRead = 0; 6232 if (mResampler != NULL) { 6233 mResampler->reset(); 6234 } 6235 mActiveTrack->mState = TrackBase::RESUMING; 6236 // signal thread to start 6237 ALOGV("Signal record thread"); 6238 mWaitWorkCV.signal(); 6239 // do not wait for mStartStopCond if exiting 6240 if (exitPending()) { 6241 mActiveTrack.clear(); 6242 status = INVALID_OPERATION; 6243 goto startError; 6244 } 6245 mStartStopCond.wait(mLock); 6246 if (mActiveTrack == 0) { 6247 ALOGV("Record failed to start"); 6248 status = BAD_VALUE; 6249 goto startError; 6250 } 6251 ALOGV("Record started OK"); 6252 return status; 6253 } 6254startError: 6255 AudioSystem::stopInput(mId); 6256 clearSyncStartEvent(); 6257 return status; 6258} 6259 6260void AudioFlinger::RecordThread::clearSyncStartEvent() 6261{ 6262 if (mSyncStartEvent != 0) { 6263 mSyncStartEvent->cancel(); 6264 } 6265 mSyncStartEvent.clear(); 6266 mFramestoDrop = 0; 6267} 6268 6269void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6270{ 6271 sp<SyncEvent> strongEvent = event.promote(); 6272 6273 if (strongEvent != 0) { 6274 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6275 me->handleSyncStartEvent(strongEvent); 6276 } 6277} 6278 6279void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6280{ 6281 if (event == mSyncStartEvent) { 6282 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6283 // from audio HAL 6284 mFramestoDrop = mFrameCount * 2; 6285 } 6286} 6287 6288void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6289 ALOGV("RecordThread::stop"); 6290 sp<ThreadBase> strongMe = this; 6291 { 6292 AutoMutex lock(mLock); 6293 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6294 mActiveTrack->mState = TrackBase::PAUSING; 6295 // do not wait for mStartStopCond if exiting 6296 if (exitPending()) { 6297 return; 6298 } 6299 mStartStopCond.wait(mLock); 6300 // if we have been restarted, recordTrack == mActiveTrack.get() here 6301 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6302 mLock.unlock(); 6303 AudioSystem::stopInput(mId); 6304 mLock.lock(); 6305 ALOGV("Record stopped OK"); 6306 } 6307 } 6308 } 6309} 6310 6311bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6312{ 6313 return false; 6314} 6315 6316status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6317{ 6318 if (!isValidSyncEvent(event)) { 6319 return BAD_VALUE; 6320 } 6321 6322 Mutex::Autolock _l(mLock); 6323 6324 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6325 mTrack->setSyncEvent(event); 6326 return NO_ERROR; 6327 } 6328 return NAME_NOT_FOUND; 6329} 6330 6331status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6332{ 6333 const size_t SIZE = 256; 6334 char buffer[SIZE]; 6335 String8 result; 6336 6337 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6338 result.append(buffer); 6339 6340 if (mActiveTrack != 0) { 6341 result.append("Active Track:\n"); 6342 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6343 mActiveTrack->dump(buffer, SIZE); 6344 result.append(buffer); 6345 6346 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6347 result.append(buffer); 6348 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6349 result.append(buffer); 6350 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6351 result.append(buffer); 6352 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6353 result.append(buffer); 6354 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6355 result.append(buffer); 6356 6357 6358 } else { 6359 result.append("No record client\n"); 6360 } 6361 write(fd, result.string(), result.size()); 6362 6363 dumpBase(fd, args); 6364 dumpEffectChains(fd, args); 6365 6366 return NO_ERROR; 6367} 6368 6369// AudioBufferProvider interface 6370status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6371{ 6372 size_t framesReq = buffer->frameCount; 6373 size_t framesReady = mFrameCount - mRsmpInIndex; 6374 int channelCount; 6375 6376 if (framesReady == 0) { 6377 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6378 if (mBytesRead < 0) { 6379 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6380 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6381 // Force input into standby so that it tries to 6382 // recover at next read attempt 6383 mInput->stream->common.standby(&mInput->stream->common); 6384 usleep(kRecordThreadSleepUs); 6385 } 6386 buffer->raw = NULL; 6387 buffer->frameCount = 0; 6388 return NOT_ENOUGH_DATA; 6389 } 6390 mRsmpInIndex = 0; 6391 framesReady = mFrameCount; 6392 } 6393 6394 if (framesReq > framesReady) { 6395 framesReq = framesReady; 6396 } 6397 6398 if (mChannelCount == 1 && mReqChannelCount == 2) { 6399 channelCount = 1; 6400 } else { 6401 channelCount = 2; 6402 } 6403 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6404 buffer->frameCount = framesReq; 6405 return NO_ERROR; 6406} 6407 6408// AudioBufferProvider interface 6409void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6410{ 6411 mRsmpInIndex += buffer->frameCount; 6412 buffer->frameCount = 0; 6413} 6414 6415bool AudioFlinger::RecordThread::checkForNewParameters_l() 6416{ 6417 bool reconfig = false; 6418 6419 while (!mNewParameters.isEmpty()) { 6420 status_t status = NO_ERROR; 6421 String8 keyValuePair = mNewParameters[0]; 6422 AudioParameter param = AudioParameter(keyValuePair); 6423 int value; 6424 audio_format_t reqFormat = mFormat; 6425 int reqSamplingRate = mReqSampleRate; 6426 int reqChannelCount = mReqChannelCount; 6427 6428 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6429 reqSamplingRate = value; 6430 reconfig = true; 6431 } 6432 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6433 reqFormat = (audio_format_t) value; 6434 reconfig = true; 6435 } 6436 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6437 reqChannelCount = popcount(value); 6438 reconfig = true; 6439 } 6440 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6441 // do not accept frame count changes if tracks are open as the track buffer 6442 // size depends on frame count and correct behavior would not be guaranteed 6443 // if frame count is changed after track creation 6444 if (mActiveTrack != 0) { 6445 status = INVALID_OPERATION; 6446 } else { 6447 reconfig = true; 6448 } 6449 } 6450 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6451 // forward device change to effects that have requested to be 6452 // aware of attached audio device. 6453 for (size_t i = 0; i < mEffectChains.size(); i++) { 6454 mEffectChains[i]->setDevice_l(value); 6455 } 6456 // store input device and output device but do not forward output device to audio HAL. 6457 // Note that status is ignored by the caller for output device 6458 // (see AudioFlinger::setParameters() 6459 if (value & AUDIO_DEVICE_OUT_ALL) { 6460 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6461 status = BAD_VALUE; 6462 } else { 6463 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6464 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6465 if (mTrack != NULL) { 6466 bool suspend = audio_is_bluetooth_sco_device( 6467 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6468 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6469 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6470 } 6471 } 6472 mDevice |= (uint32_t)value; 6473 } 6474 if (status == NO_ERROR) { 6475 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6476 if (status == INVALID_OPERATION) { 6477 mInput->stream->common.standby(&mInput->stream->common); 6478 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6479 keyValuePair.string()); 6480 } 6481 if (reconfig) { 6482 if (status == BAD_VALUE && 6483 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6484 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6485 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6486 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6487 (reqChannelCount <= FCC_2)) { 6488 status = NO_ERROR; 6489 } 6490 if (status == NO_ERROR) { 6491 readInputParameters(); 6492 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6493 } 6494 } 6495 } 6496 6497 mNewParameters.removeAt(0); 6498 6499 mParamStatus = status; 6500 mParamCond.signal(); 6501 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6502 // already timed out waiting for the status and will never signal the condition. 6503 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6504 } 6505 return reconfig; 6506} 6507 6508String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6509{ 6510 char *s; 6511 String8 out_s8 = String8(); 6512 6513 Mutex::Autolock _l(mLock); 6514 if (initCheck() != NO_ERROR) { 6515 return out_s8; 6516 } 6517 6518 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6519 out_s8 = String8(s); 6520 free(s); 6521 return out_s8; 6522} 6523 6524void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6525 AudioSystem::OutputDescriptor desc; 6526 void *param2 = NULL; 6527 6528 switch (event) { 6529 case AudioSystem::INPUT_OPENED: 6530 case AudioSystem::INPUT_CONFIG_CHANGED: 6531 desc.channels = mChannelMask; 6532 desc.samplingRate = mSampleRate; 6533 desc.format = mFormat; 6534 desc.frameCount = mFrameCount; 6535 desc.latency = 0; 6536 param2 = &desc; 6537 break; 6538 6539 case AudioSystem::INPUT_CLOSED: 6540 default: 6541 break; 6542 } 6543 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6544} 6545 6546void AudioFlinger::RecordThread::readInputParameters() 6547{ 6548 delete mRsmpInBuffer; 6549 // mRsmpInBuffer is always assigned a new[] below 6550 delete mRsmpOutBuffer; 6551 mRsmpOutBuffer = NULL; 6552 delete mResampler; 6553 mResampler = NULL; 6554 6555 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6556 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6557 mChannelCount = (uint16_t)popcount(mChannelMask); 6558 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6559 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6560 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6561 mFrameCount = mInputBytes / mFrameSize; 6562 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6563 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6564 6565 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6566 { 6567 int channelCount; 6568 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6569 // stereo to mono post process as the resampler always outputs stereo. 6570 if (mChannelCount == 1 && mReqChannelCount == 2) { 6571 channelCount = 1; 6572 } else { 6573 channelCount = 2; 6574 } 6575 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6576 mResampler->setSampleRate(mSampleRate); 6577 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6578 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6579 6580 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6581 if (mChannelCount == 1 && mReqChannelCount == 1) { 6582 mFrameCount >>= 1; 6583 } 6584 6585 } 6586 mRsmpInIndex = mFrameCount; 6587} 6588 6589unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6590{ 6591 Mutex::Autolock _l(mLock); 6592 if (initCheck() != NO_ERROR) { 6593 return 0; 6594 } 6595 6596 return mInput->stream->get_input_frames_lost(mInput->stream); 6597} 6598 6599uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6600{ 6601 Mutex::Autolock _l(mLock); 6602 uint32_t result = 0; 6603 if (getEffectChain_l(sessionId) != 0) { 6604 result = EFFECT_SESSION; 6605 } 6606 6607 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6608 result |= TRACK_SESSION; 6609 } 6610 6611 return result; 6612} 6613 6614AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6615{ 6616 Mutex::Autolock _l(mLock); 6617 return mTrack; 6618} 6619 6620AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6621{ 6622 Mutex::Autolock _l(mLock); 6623 return mInput; 6624} 6625 6626AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6627{ 6628 Mutex::Autolock _l(mLock); 6629 AudioStreamIn *input = mInput; 6630 mInput = NULL; 6631 return input; 6632} 6633 6634// this method must always be called either with ThreadBase mLock held or inside the thread loop 6635audio_stream_t* AudioFlinger::RecordThread::stream() const 6636{ 6637 if (mInput == NULL) { 6638 return NULL; 6639 } 6640 return &mInput->stream->common; 6641} 6642 6643 6644// ---------------------------------------------------------------------------- 6645 6646audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6647{ 6648 if (!settingsAllowed()) { 6649 return 0; 6650 } 6651 Mutex::Autolock _l(mLock); 6652 return loadHwModule_l(name); 6653} 6654 6655// loadHwModule_l() must be called with AudioFlinger::mLock held 6656audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6657{ 6658 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6659 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6660 ALOGW("loadHwModule() module %s already loaded", name); 6661 return mAudioHwDevs.keyAt(i); 6662 } 6663 } 6664 6665 audio_hw_device_t *dev; 6666 6667 int rc = load_audio_interface(name, &dev); 6668 if (rc) { 6669 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6670 return 0; 6671 } 6672 6673 mHardwareStatus = AUDIO_HW_INIT; 6674 rc = dev->init_check(dev); 6675 mHardwareStatus = AUDIO_HW_IDLE; 6676 if (rc) { 6677 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6678 return 0; 6679 } 6680 6681 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6682 (NULL != dev->set_master_volume)) { 6683 AutoMutex lock(mHardwareLock); 6684 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6685 dev->set_master_volume(dev, mMasterVolume); 6686 mHardwareStatus = AUDIO_HW_IDLE; 6687 } 6688 6689 audio_module_handle_t handle = nextUniqueId(); 6690 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6691 6692 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6693 name, dev->common.module->name, dev->common.module->id, handle); 6694 6695 return handle; 6696 6697} 6698 6699audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6700 audio_devices_t *pDevices, 6701 uint32_t *pSamplingRate, 6702 audio_format_t *pFormat, 6703 audio_channel_mask_t *pChannelMask, 6704 uint32_t *pLatencyMs, 6705 audio_output_flags_t flags) 6706{ 6707 status_t status; 6708 PlaybackThread *thread = NULL; 6709 struct audio_config config = { 6710 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6711 channel_mask: pChannelMask ? *pChannelMask : 0, 6712 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6713 }; 6714 audio_stream_out_t *outStream = NULL; 6715 audio_hw_device_t *outHwDev; 6716 6717 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6718 module, 6719 (pDevices != NULL) ? (int)*pDevices : 0, 6720 config.sample_rate, 6721 config.format, 6722 config.channel_mask, 6723 flags); 6724 6725 if (pDevices == NULL || *pDevices == 0) { 6726 return 0; 6727 } 6728 6729 Mutex::Autolock _l(mLock); 6730 6731 outHwDev = findSuitableHwDev_l(module, *pDevices); 6732 if (outHwDev == NULL) 6733 return 0; 6734 6735 audio_io_handle_t id = nextUniqueId(); 6736 6737 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6738 6739 status = outHwDev->open_output_stream(outHwDev, 6740 id, 6741 *pDevices, 6742 (audio_output_flags_t)flags, 6743 &config, 6744 &outStream); 6745 6746 mHardwareStatus = AUDIO_HW_IDLE; 6747 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6748 outStream, 6749 config.sample_rate, 6750 config.format, 6751 config.channel_mask, 6752 status); 6753 6754 if (status == NO_ERROR && outStream != NULL) { 6755 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6756 6757 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6758 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6759 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6760 thread = new DirectOutputThread(this, output, id, *pDevices); 6761 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6762 } else { 6763 thread = new MixerThread(this, output, id, *pDevices); 6764 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6765 } 6766 mPlaybackThreads.add(id, thread); 6767 6768 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6769 if (pFormat != NULL) *pFormat = config.format; 6770 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6771 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6772 6773 // notify client processes of the new output creation 6774 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6775 6776 // the first primary output opened designates the primary hw device 6777 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6778 ALOGI("Using module %d has the primary audio interface", module); 6779 mPrimaryHardwareDev = outHwDev; 6780 6781 AutoMutex lock(mHardwareLock); 6782 mHardwareStatus = AUDIO_HW_SET_MODE; 6783 outHwDev->set_mode(outHwDev, mMode); 6784 6785 // Determine the level of master volume support the primary audio HAL has, 6786 // and set the initial master volume at the same time. 6787 float initialVolume = 1.0; 6788 mMasterVolumeSupportLvl = MVS_NONE; 6789 6790 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6791 if ((NULL != outHwDev->get_master_volume) && 6792 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6793 mMasterVolumeSupportLvl = MVS_FULL; 6794 } else { 6795 mMasterVolumeSupportLvl = MVS_SETONLY; 6796 initialVolume = 1.0; 6797 } 6798 6799 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6800 if ((NULL == outHwDev->set_master_volume) || 6801 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6802 mMasterVolumeSupportLvl = MVS_NONE; 6803 } 6804 // now that we have a primary device, initialize master volume on other devices 6805 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6806 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6807 6808 if ((dev != mPrimaryHardwareDev) && 6809 (NULL != dev->set_master_volume)) { 6810 dev->set_master_volume(dev, initialVolume); 6811 } 6812 } 6813 mHardwareStatus = AUDIO_HW_IDLE; 6814 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6815 ? initialVolume 6816 : 1.0; 6817 mMasterVolume = initialVolume; 6818 } 6819 return id; 6820 } 6821 6822 return 0; 6823} 6824 6825audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6826 audio_io_handle_t output2) 6827{ 6828 Mutex::Autolock _l(mLock); 6829 MixerThread *thread1 = checkMixerThread_l(output1); 6830 MixerThread *thread2 = checkMixerThread_l(output2); 6831 6832 if (thread1 == NULL || thread2 == NULL) { 6833 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6834 return 0; 6835 } 6836 6837 audio_io_handle_t id = nextUniqueId(); 6838 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6839 thread->addOutputTrack(thread2); 6840 mPlaybackThreads.add(id, thread); 6841 // notify client processes of the new output creation 6842 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6843 return id; 6844} 6845 6846status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6847{ 6848 // keep strong reference on the playback thread so that 6849 // it is not destroyed while exit() is executed 6850 sp<PlaybackThread> thread; 6851 { 6852 Mutex::Autolock _l(mLock); 6853 thread = checkPlaybackThread_l(output); 6854 if (thread == NULL) { 6855 return BAD_VALUE; 6856 } 6857 6858 ALOGV("closeOutput() %d", output); 6859 6860 if (thread->type() == ThreadBase::MIXER) { 6861 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6862 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6863 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6864 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6865 } 6866 } 6867 } 6868 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6869 mPlaybackThreads.removeItem(output); 6870 } 6871 thread->exit(); 6872 // The thread entity (active unit of execution) is no longer running here, 6873 // but the ThreadBase container still exists. 6874 6875 if (thread->type() != ThreadBase::DUPLICATING) { 6876 AudioStreamOut *out = thread->clearOutput(); 6877 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6878 // from now on thread->mOutput is NULL 6879 out->hwDev->close_output_stream(out->hwDev, out->stream); 6880 delete out; 6881 } 6882 return NO_ERROR; 6883} 6884 6885status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6886{ 6887 Mutex::Autolock _l(mLock); 6888 PlaybackThread *thread = checkPlaybackThread_l(output); 6889 6890 if (thread == NULL) { 6891 return BAD_VALUE; 6892 } 6893 6894 ALOGV("suspendOutput() %d", output); 6895 thread->suspend(); 6896 6897 return NO_ERROR; 6898} 6899 6900status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6901{ 6902 Mutex::Autolock _l(mLock); 6903 PlaybackThread *thread = checkPlaybackThread_l(output); 6904 6905 if (thread == NULL) { 6906 return BAD_VALUE; 6907 } 6908 6909 ALOGV("restoreOutput() %d", output); 6910 6911 thread->restore(); 6912 6913 return NO_ERROR; 6914} 6915 6916audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6917 audio_devices_t *pDevices, 6918 uint32_t *pSamplingRate, 6919 audio_format_t *pFormat, 6920 uint32_t *pChannelMask) 6921{ 6922 status_t status; 6923 RecordThread *thread = NULL; 6924 struct audio_config config = { 6925 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6926 channel_mask: pChannelMask ? *pChannelMask : 0, 6927 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6928 }; 6929 uint32_t reqSamplingRate = config.sample_rate; 6930 audio_format_t reqFormat = config.format; 6931 audio_channel_mask_t reqChannels = config.channel_mask; 6932 audio_stream_in_t *inStream = NULL; 6933 audio_hw_device_t *inHwDev; 6934 6935 if (pDevices == NULL || *pDevices == 0) { 6936 return 0; 6937 } 6938 6939 Mutex::Autolock _l(mLock); 6940 6941 inHwDev = findSuitableHwDev_l(module, *pDevices); 6942 if (inHwDev == NULL) 6943 return 0; 6944 6945 audio_io_handle_t id = nextUniqueId(); 6946 6947 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6948 &inStream); 6949 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6950 inStream, 6951 config.sample_rate, 6952 config.format, 6953 config.channel_mask, 6954 status); 6955 6956 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6957 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6958 // or stereo to mono conversions on 16 bit PCM inputs. 6959 if (status == BAD_VALUE && 6960 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6961 (config.sample_rate <= 2 * reqSamplingRate) && 6962 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6963 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6964 inStream = NULL; 6965 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6966 } 6967 6968 if (status == NO_ERROR && inStream != NULL) { 6969 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6970 6971 // Start record thread 6972 // RecorThread require both input and output device indication to forward to audio 6973 // pre processing modules 6974 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6975 thread = new RecordThread(this, 6976 input, 6977 reqSamplingRate, 6978 reqChannels, 6979 id, 6980 device); 6981 mRecordThreads.add(id, thread); 6982 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6983 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6984 if (pFormat != NULL) *pFormat = config.format; 6985 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6986 6987 input->stream->common.standby(&input->stream->common); 6988 6989 // notify client processes of the new input creation 6990 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6991 return id; 6992 } 6993 6994 return 0; 6995} 6996 6997status_t AudioFlinger::closeInput(audio_io_handle_t input) 6998{ 6999 // keep strong reference on the record thread so that 7000 // it is not destroyed while exit() is executed 7001 sp<RecordThread> thread; 7002 { 7003 Mutex::Autolock _l(mLock); 7004 thread = checkRecordThread_l(input); 7005 if (thread == NULL) { 7006 return BAD_VALUE; 7007 } 7008 7009 ALOGV("closeInput() %d", input); 7010 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7011 mRecordThreads.removeItem(input); 7012 } 7013 thread->exit(); 7014 // The thread entity (active unit of execution) is no longer running here, 7015 // but the ThreadBase container still exists. 7016 7017 AudioStreamIn *in = thread->clearInput(); 7018 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7019 // from now on thread->mInput is NULL 7020 in->hwDev->close_input_stream(in->hwDev, in->stream); 7021 delete in; 7022 7023 return NO_ERROR; 7024} 7025 7026status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7027{ 7028 Mutex::Autolock _l(mLock); 7029 MixerThread *dstThread = checkMixerThread_l(output); 7030 if (dstThread == NULL) { 7031 ALOGW("setStreamOutput() bad output id %d", output); 7032 return BAD_VALUE; 7033 } 7034 7035 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7036 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7037 7038 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7039 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7040 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7041 MixerThread *srcThread = (MixerThread *)thread; 7042 srcThread->invalidateTracks(stream); 7043 } 7044 } 7045 7046 return NO_ERROR; 7047} 7048 7049 7050int AudioFlinger::newAudioSessionId() 7051{ 7052 return nextUniqueId(); 7053} 7054 7055void AudioFlinger::acquireAudioSessionId(int audioSession) 7056{ 7057 Mutex::Autolock _l(mLock); 7058 pid_t caller = IPCThreadState::self()->getCallingPid(); 7059 ALOGV("acquiring %d from %d", audioSession, caller); 7060 size_t num = mAudioSessionRefs.size(); 7061 for (size_t i = 0; i< num; i++) { 7062 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7063 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7064 ref->mCnt++; 7065 ALOGV(" incremented refcount to %d", ref->mCnt); 7066 return; 7067 } 7068 } 7069 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7070 ALOGV(" added new entry for %d", audioSession); 7071} 7072 7073void AudioFlinger::releaseAudioSessionId(int audioSession) 7074{ 7075 Mutex::Autolock _l(mLock); 7076 pid_t caller = IPCThreadState::self()->getCallingPid(); 7077 ALOGV("releasing %d from %d", audioSession, caller); 7078 size_t num = mAudioSessionRefs.size(); 7079 for (size_t i = 0; i< num; i++) { 7080 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7081 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7082 ref->mCnt--; 7083 ALOGV(" decremented refcount to %d", ref->mCnt); 7084 if (ref->mCnt == 0) { 7085 mAudioSessionRefs.removeAt(i); 7086 delete ref; 7087 purgeStaleEffects_l(); 7088 } 7089 return; 7090 } 7091 } 7092 ALOGW("session id %d not found for pid %d", audioSession, caller); 7093} 7094 7095void AudioFlinger::purgeStaleEffects_l() { 7096 7097 ALOGV("purging stale effects"); 7098 7099 Vector< sp<EffectChain> > chains; 7100 7101 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7102 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7103 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7104 sp<EffectChain> ec = t->mEffectChains[j]; 7105 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7106 chains.push(ec); 7107 } 7108 } 7109 } 7110 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7111 sp<RecordThread> t = mRecordThreads.valueAt(i); 7112 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7113 sp<EffectChain> ec = t->mEffectChains[j]; 7114 chains.push(ec); 7115 } 7116 } 7117 7118 for (size_t i = 0; i < chains.size(); i++) { 7119 sp<EffectChain> ec = chains[i]; 7120 int sessionid = ec->sessionId(); 7121 sp<ThreadBase> t = ec->mThread.promote(); 7122 if (t == 0) { 7123 continue; 7124 } 7125 size_t numsessionrefs = mAudioSessionRefs.size(); 7126 bool found = false; 7127 for (size_t k = 0; k < numsessionrefs; k++) { 7128 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7129 if (ref->mSessionid == sessionid) { 7130 ALOGV(" session %d still exists for %d with %d refs", 7131 sessionid, ref->mPid, ref->mCnt); 7132 found = true; 7133 break; 7134 } 7135 } 7136 if (!found) { 7137 // remove all effects from the chain 7138 while (ec->mEffects.size()) { 7139 sp<EffectModule> effect = ec->mEffects[0]; 7140 effect->unPin(); 7141 Mutex::Autolock _l (t->mLock); 7142 t->removeEffect_l(effect); 7143 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7144 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7145 if (handle != 0) { 7146 handle->mEffect.clear(); 7147 if (handle->mHasControl && handle->mEnabled) { 7148 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7149 } 7150 } 7151 } 7152 AudioSystem::unregisterEffect(effect->id()); 7153 } 7154 } 7155 } 7156 return; 7157} 7158 7159// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7160AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7161{ 7162 return mPlaybackThreads.valueFor(output).get(); 7163} 7164 7165// checkMixerThread_l() must be called with AudioFlinger::mLock held 7166AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7167{ 7168 PlaybackThread *thread = checkPlaybackThread_l(output); 7169 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7170} 7171 7172// checkRecordThread_l() must be called with AudioFlinger::mLock held 7173AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7174{ 7175 return mRecordThreads.valueFor(input).get(); 7176} 7177 7178uint32_t AudioFlinger::nextUniqueId() 7179{ 7180 return android_atomic_inc(&mNextUniqueId); 7181} 7182 7183AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7184{ 7185 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7186 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7187 AudioStreamOut *output = thread->getOutput(); 7188 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7189 return thread; 7190 } 7191 } 7192 return NULL; 7193} 7194 7195uint32_t AudioFlinger::primaryOutputDevice_l() const 7196{ 7197 PlaybackThread *thread = primaryPlaybackThread_l(); 7198 7199 if (thread == NULL) { 7200 return 0; 7201 } 7202 7203 return thread->device(); 7204} 7205 7206sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7207 int triggerSession, 7208 int listenerSession, 7209 sync_event_callback_t callBack, 7210 void *cookie) 7211{ 7212 Mutex::Autolock _l(mLock); 7213 7214 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7215 status_t playStatus = NAME_NOT_FOUND; 7216 status_t recStatus = NAME_NOT_FOUND; 7217 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7218 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7219 if (playStatus == NO_ERROR) { 7220 return event; 7221 } 7222 } 7223 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7224 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7225 if (recStatus == NO_ERROR) { 7226 return event; 7227 } 7228 } 7229 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7230 mPendingSyncEvents.add(event); 7231 } else { 7232 ALOGV("createSyncEvent() invalid event %d", event->type()); 7233 event.clear(); 7234 } 7235 return event; 7236} 7237 7238// ---------------------------------------------------------------------------- 7239// Effect management 7240// ---------------------------------------------------------------------------- 7241 7242 7243status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7244{ 7245 Mutex::Autolock _l(mLock); 7246 return EffectQueryNumberEffects(numEffects); 7247} 7248 7249status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7250{ 7251 Mutex::Autolock _l(mLock); 7252 return EffectQueryEffect(index, descriptor); 7253} 7254 7255status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7256 effect_descriptor_t *descriptor) const 7257{ 7258 Mutex::Autolock _l(mLock); 7259 return EffectGetDescriptor(pUuid, descriptor); 7260} 7261 7262 7263sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7264 effect_descriptor_t *pDesc, 7265 const sp<IEffectClient>& effectClient, 7266 int32_t priority, 7267 audio_io_handle_t io, 7268 int sessionId, 7269 status_t *status, 7270 int *id, 7271 int *enabled) 7272{ 7273 status_t lStatus = NO_ERROR; 7274 sp<EffectHandle> handle; 7275 effect_descriptor_t desc; 7276 7277 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7278 pid, effectClient.get(), priority, sessionId, io); 7279 7280 if (pDesc == NULL) { 7281 lStatus = BAD_VALUE; 7282 goto Exit; 7283 } 7284 7285 // check audio settings permission for global effects 7286 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7287 lStatus = PERMISSION_DENIED; 7288 goto Exit; 7289 } 7290 7291 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7292 // that can only be created by audio policy manager (running in same process) 7293 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7294 lStatus = PERMISSION_DENIED; 7295 goto Exit; 7296 } 7297 7298 if (io == 0) { 7299 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7300 // output must be specified by AudioPolicyManager when using session 7301 // AUDIO_SESSION_OUTPUT_STAGE 7302 lStatus = BAD_VALUE; 7303 goto Exit; 7304 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7305 // if the output returned by getOutputForEffect() is removed before we lock the 7306 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7307 // and we will exit safely 7308 io = AudioSystem::getOutputForEffect(&desc); 7309 } 7310 } 7311 7312 { 7313 Mutex::Autolock _l(mLock); 7314 7315 7316 if (!EffectIsNullUuid(&pDesc->uuid)) { 7317 // if uuid is specified, request effect descriptor 7318 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7319 if (lStatus < 0) { 7320 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7321 goto Exit; 7322 } 7323 } else { 7324 // if uuid is not specified, look for an available implementation 7325 // of the required type in effect factory 7326 if (EffectIsNullUuid(&pDesc->type)) { 7327 ALOGW("createEffect() no effect type"); 7328 lStatus = BAD_VALUE; 7329 goto Exit; 7330 } 7331 uint32_t numEffects = 0; 7332 effect_descriptor_t d; 7333 d.flags = 0; // prevent compiler warning 7334 bool found = false; 7335 7336 lStatus = EffectQueryNumberEffects(&numEffects); 7337 if (lStatus < 0) { 7338 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7339 goto Exit; 7340 } 7341 for (uint32_t i = 0; i < numEffects; i++) { 7342 lStatus = EffectQueryEffect(i, &desc); 7343 if (lStatus < 0) { 7344 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7345 continue; 7346 } 7347 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7348 // If matching type found save effect descriptor. If the session is 7349 // 0 and the effect is not auxiliary, continue enumeration in case 7350 // an auxiliary version of this effect type is available 7351 found = true; 7352 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7353 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7354 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7355 break; 7356 } 7357 } 7358 } 7359 if (!found) { 7360 lStatus = BAD_VALUE; 7361 ALOGW("createEffect() effect not found"); 7362 goto Exit; 7363 } 7364 // For same effect type, chose auxiliary version over insert version if 7365 // connect to output mix (Compliance to OpenSL ES) 7366 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7367 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7368 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7369 } 7370 } 7371 7372 // Do not allow auxiliary effects on a session different from 0 (output mix) 7373 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7374 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7375 lStatus = INVALID_OPERATION; 7376 goto Exit; 7377 } 7378 7379 // check recording permission for visualizer 7380 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7381 !recordingAllowed()) { 7382 lStatus = PERMISSION_DENIED; 7383 goto Exit; 7384 } 7385 7386 // return effect descriptor 7387 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7388 7389 // If output is not specified try to find a matching audio session ID in one of the 7390 // output threads. 7391 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7392 // because of code checking output when entering the function. 7393 // Note: io is never 0 when creating an effect on an input 7394 if (io == 0) { 7395 // look for the thread where the specified audio session is present 7396 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7397 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7398 io = mPlaybackThreads.keyAt(i); 7399 break; 7400 } 7401 } 7402 if (io == 0) { 7403 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7404 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7405 io = mRecordThreads.keyAt(i); 7406 break; 7407 } 7408 } 7409 } 7410 // If no output thread contains the requested session ID, default to 7411 // first output. The effect chain will be moved to the correct output 7412 // thread when a track with the same session ID is created 7413 if (io == 0 && mPlaybackThreads.size()) { 7414 io = mPlaybackThreads.keyAt(0); 7415 } 7416 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7417 } 7418 ThreadBase *thread = checkRecordThread_l(io); 7419 if (thread == NULL) { 7420 thread = checkPlaybackThread_l(io); 7421 if (thread == NULL) { 7422 ALOGE("createEffect() unknown output thread"); 7423 lStatus = BAD_VALUE; 7424 goto Exit; 7425 } 7426 } 7427 7428 sp<Client> client = registerPid_l(pid); 7429 7430 // create effect on selected output thread 7431 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7432 &desc, enabled, &lStatus); 7433 if (handle != 0 && id != NULL) { 7434 *id = handle->id(); 7435 } 7436 } 7437 7438Exit: 7439 if (status != NULL) { 7440 *status = lStatus; 7441 } 7442 return handle; 7443} 7444 7445status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7446 audio_io_handle_t dstOutput) 7447{ 7448 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7449 sessionId, srcOutput, dstOutput); 7450 Mutex::Autolock _l(mLock); 7451 if (srcOutput == dstOutput) { 7452 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7453 return NO_ERROR; 7454 } 7455 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7456 if (srcThread == NULL) { 7457 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7458 return BAD_VALUE; 7459 } 7460 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7461 if (dstThread == NULL) { 7462 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7463 return BAD_VALUE; 7464 } 7465 7466 Mutex::Autolock _dl(dstThread->mLock); 7467 Mutex::Autolock _sl(srcThread->mLock); 7468 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7469 7470 return NO_ERROR; 7471} 7472 7473// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7474status_t AudioFlinger::moveEffectChain_l(int sessionId, 7475 AudioFlinger::PlaybackThread *srcThread, 7476 AudioFlinger::PlaybackThread *dstThread, 7477 bool reRegister) 7478{ 7479 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7480 sessionId, srcThread, dstThread); 7481 7482 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7483 if (chain == 0) { 7484 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7485 sessionId, srcThread); 7486 return INVALID_OPERATION; 7487 } 7488 7489 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7490 // so that a new chain is created with correct parameters when first effect is added. This is 7491 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7492 // removed. 7493 srcThread->removeEffectChain_l(chain); 7494 7495 // transfer all effects one by one so that new effect chain is created on new thread with 7496 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7497 audio_io_handle_t dstOutput = dstThread->id(); 7498 sp<EffectChain> dstChain; 7499 uint32_t strategy = 0; // prevent compiler warning 7500 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7501 while (effect != 0) { 7502 srcThread->removeEffect_l(effect); 7503 dstThread->addEffect_l(effect); 7504 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7505 if (effect->state() == EffectModule::ACTIVE || 7506 effect->state() == EffectModule::STOPPING) { 7507 effect->start(); 7508 } 7509 // if the move request is not received from audio policy manager, the effect must be 7510 // re-registered with the new strategy and output 7511 if (dstChain == 0) { 7512 dstChain = effect->chain().promote(); 7513 if (dstChain == 0) { 7514 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7515 srcThread->addEffect_l(effect); 7516 return NO_INIT; 7517 } 7518 strategy = dstChain->strategy(); 7519 } 7520 if (reRegister) { 7521 AudioSystem::unregisterEffect(effect->id()); 7522 AudioSystem::registerEffect(&effect->desc(), 7523 dstOutput, 7524 strategy, 7525 sessionId, 7526 effect->id()); 7527 } 7528 effect = chain->getEffectFromId_l(0); 7529 } 7530 7531 return NO_ERROR; 7532} 7533 7534 7535// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7536sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7537 const sp<AudioFlinger::Client>& client, 7538 const sp<IEffectClient>& effectClient, 7539 int32_t priority, 7540 int sessionId, 7541 effect_descriptor_t *desc, 7542 int *enabled, 7543 status_t *status 7544 ) 7545{ 7546 sp<EffectModule> effect; 7547 sp<EffectHandle> handle; 7548 status_t lStatus; 7549 sp<EffectChain> chain; 7550 bool chainCreated = false; 7551 bool effectCreated = false; 7552 bool effectRegistered = false; 7553 7554 lStatus = initCheck(); 7555 if (lStatus != NO_ERROR) { 7556 ALOGW("createEffect_l() Audio driver not initialized."); 7557 goto Exit; 7558 } 7559 7560 // Do not allow effects with session ID 0 on direct output or duplicating threads 7561 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7562 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7563 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7564 desc->name, sessionId); 7565 lStatus = BAD_VALUE; 7566 goto Exit; 7567 } 7568 // Only Pre processor effects are allowed on input threads and only on input threads 7569 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7570 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7571 desc->name, desc->flags, mType); 7572 lStatus = BAD_VALUE; 7573 goto Exit; 7574 } 7575 7576 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7577 7578 { // scope for mLock 7579 Mutex::Autolock _l(mLock); 7580 7581 // check for existing effect chain with the requested audio session 7582 chain = getEffectChain_l(sessionId); 7583 if (chain == 0) { 7584 // create a new chain for this session 7585 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7586 chain = new EffectChain(this, sessionId); 7587 addEffectChain_l(chain); 7588 chain->setStrategy(getStrategyForSession_l(sessionId)); 7589 chainCreated = true; 7590 } else { 7591 effect = chain->getEffectFromDesc_l(desc); 7592 } 7593 7594 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7595 7596 if (effect == 0) { 7597 int id = mAudioFlinger->nextUniqueId(); 7598 // Check CPU and memory usage 7599 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7600 if (lStatus != NO_ERROR) { 7601 goto Exit; 7602 } 7603 effectRegistered = true; 7604 // create a new effect module if none present in the chain 7605 effect = new EffectModule(this, chain, desc, id, sessionId); 7606 lStatus = effect->status(); 7607 if (lStatus != NO_ERROR) { 7608 goto Exit; 7609 } 7610 lStatus = chain->addEffect_l(effect); 7611 if (lStatus != NO_ERROR) { 7612 goto Exit; 7613 } 7614 effectCreated = true; 7615 7616 effect->setDevice(mDevice); 7617 effect->setMode(mAudioFlinger->getMode()); 7618 } 7619 // create effect handle and connect it to effect module 7620 handle = new EffectHandle(effect, client, effectClient, priority); 7621 lStatus = effect->addHandle(handle); 7622 if (enabled != NULL) { 7623 *enabled = (int)effect->isEnabled(); 7624 } 7625 } 7626 7627Exit: 7628 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7629 Mutex::Autolock _l(mLock); 7630 if (effectCreated) { 7631 chain->removeEffect_l(effect); 7632 } 7633 if (effectRegistered) { 7634 AudioSystem::unregisterEffect(effect->id()); 7635 } 7636 if (chainCreated) { 7637 removeEffectChain_l(chain); 7638 } 7639 handle.clear(); 7640 } 7641 7642 if (status != NULL) { 7643 *status = lStatus; 7644 } 7645 return handle; 7646} 7647 7648sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7649{ 7650 sp<EffectChain> chain = getEffectChain_l(sessionId); 7651 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7652} 7653 7654// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7655// PlaybackThread::mLock held 7656status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7657{ 7658 // check for existing effect chain with the requested audio session 7659 int sessionId = effect->sessionId(); 7660 sp<EffectChain> chain = getEffectChain_l(sessionId); 7661 bool chainCreated = false; 7662 7663 if (chain == 0) { 7664 // create a new chain for this session 7665 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7666 chain = new EffectChain(this, sessionId); 7667 addEffectChain_l(chain); 7668 chain->setStrategy(getStrategyForSession_l(sessionId)); 7669 chainCreated = true; 7670 } 7671 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7672 7673 if (chain->getEffectFromId_l(effect->id()) != 0) { 7674 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7675 this, effect->desc().name, chain.get()); 7676 return BAD_VALUE; 7677 } 7678 7679 status_t status = chain->addEffect_l(effect); 7680 if (status != NO_ERROR) { 7681 if (chainCreated) { 7682 removeEffectChain_l(chain); 7683 } 7684 return status; 7685 } 7686 7687 effect->setDevice(mDevice); 7688 effect->setMode(mAudioFlinger->getMode()); 7689 return NO_ERROR; 7690} 7691 7692void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7693 7694 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7695 effect_descriptor_t desc = effect->desc(); 7696 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7697 detachAuxEffect_l(effect->id()); 7698 } 7699 7700 sp<EffectChain> chain = effect->chain().promote(); 7701 if (chain != 0) { 7702 // remove effect chain if removing last effect 7703 if (chain->removeEffect_l(effect) == 0) { 7704 removeEffectChain_l(chain); 7705 } 7706 } else { 7707 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7708 } 7709} 7710 7711void AudioFlinger::ThreadBase::lockEffectChains_l( 7712 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7713{ 7714 effectChains = mEffectChains; 7715 for (size_t i = 0; i < mEffectChains.size(); i++) { 7716 mEffectChains[i]->lock(); 7717 } 7718} 7719 7720void AudioFlinger::ThreadBase::unlockEffectChains( 7721 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7722{ 7723 for (size_t i = 0; i < effectChains.size(); i++) { 7724 effectChains[i]->unlock(); 7725 } 7726} 7727 7728sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7729{ 7730 Mutex::Autolock _l(mLock); 7731 return getEffectChain_l(sessionId); 7732} 7733 7734sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7735{ 7736 size_t size = mEffectChains.size(); 7737 for (size_t i = 0; i < size; i++) { 7738 if (mEffectChains[i]->sessionId() == sessionId) { 7739 return mEffectChains[i]; 7740 } 7741 } 7742 return 0; 7743} 7744 7745void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7746{ 7747 Mutex::Autolock _l(mLock); 7748 size_t size = mEffectChains.size(); 7749 for (size_t i = 0; i < size; i++) { 7750 mEffectChains[i]->setMode_l(mode); 7751 } 7752} 7753 7754void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7755 const wp<EffectHandle>& handle, 7756 bool unpinIfLast) { 7757 7758 Mutex::Autolock _l(mLock); 7759 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7760 // delete the effect module if removing last handle on it 7761 if (effect->removeHandle(handle) == 0) { 7762 if (!effect->isPinned() || unpinIfLast) { 7763 removeEffect_l(effect); 7764 AudioSystem::unregisterEffect(effect->id()); 7765 } 7766 } 7767} 7768 7769status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7770{ 7771 int session = chain->sessionId(); 7772 int16_t *buffer = mMixBuffer; 7773 bool ownsBuffer = false; 7774 7775 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7776 if (session > 0) { 7777 // Only one effect chain can be present in direct output thread and it uses 7778 // the mix buffer as input 7779 if (mType != DIRECT) { 7780 size_t numSamples = mNormalFrameCount * mChannelCount; 7781 buffer = new int16_t[numSamples]; 7782 memset(buffer, 0, numSamples * sizeof(int16_t)); 7783 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7784 ownsBuffer = true; 7785 } 7786 7787 // Attach all tracks with same session ID to this chain. 7788 for (size_t i = 0; i < mTracks.size(); ++i) { 7789 sp<Track> track = mTracks[i]; 7790 if (session == track->sessionId()) { 7791 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7792 track->setMainBuffer(buffer); 7793 chain->incTrackCnt(); 7794 } 7795 } 7796 7797 // indicate all active tracks in the chain 7798 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7799 sp<Track> track = mActiveTracks[i].promote(); 7800 if (track == 0) continue; 7801 if (session == track->sessionId()) { 7802 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7803 chain->incActiveTrackCnt(); 7804 } 7805 } 7806 } 7807 7808 chain->setInBuffer(buffer, ownsBuffer); 7809 chain->setOutBuffer(mMixBuffer); 7810 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7811 // chains list in order to be processed last as it contains output stage effects 7812 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7813 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7814 // after track specific effects and before output stage 7815 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7816 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7817 // Effect chain for other sessions are inserted at beginning of effect 7818 // chains list to be processed before output mix effects. Relative order between other 7819 // sessions is not important 7820 size_t size = mEffectChains.size(); 7821 size_t i = 0; 7822 for (i = 0; i < size; i++) { 7823 if (mEffectChains[i]->sessionId() < session) break; 7824 } 7825 mEffectChains.insertAt(chain, i); 7826 checkSuspendOnAddEffectChain_l(chain); 7827 7828 return NO_ERROR; 7829} 7830 7831size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7832{ 7833 int session = chain->sessionId(); 7834 7835 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7836 7837 for (size_t i = 0; i < mEffectChains.size(); i++) { 7838 if (chain == mEffectChains[i]) { 7839 mEffectChains.removeAt(i); 7840 // detach all active tracks from the chain 7841 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7842 sp<Track> track = mActiveTracks[i].promote(); 7843 if (track == 0) continue; 7844 if (session == track->sessionId()) { 7845 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7846 chain.get(), session); 7847 chain->decActiveTrackCnt(); 7848 } 7849 } 7850 7851 // detach all tracks with same session ID from this chain 7852 for (size_t i = 0; i < mTracks.size(); ++i) { 7853 sp<Track> track = mTracks[i]; 7854 if (session == track->sessionId()) { 7855 track->setMainBuffer(mMixBuffer); 7856 chain->decTrackCnt(); 7857 } 7858 } 7859 break; 7860 } 7861 } 7862 return mEffectChains.size(); 7863} 7864 7865status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7866 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7867{ 7868 Mutex::Autolock _l(mLock); 7869 return attachAuxEffect_l(track, EffectId); 7870} 7871 7872status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7873 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7874{ 7875 status_t status = NO_ERROR; 7876 7877 if (EffectId == 0) { 7878 track->setAuxBuffer(0, NULL); 7879 } else { 7880 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7881 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7882 if (effect != 0) { 7883 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7884 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7885 } else { 7886 status = INVALID_OPERATION; 7887 } 7888 } else { 7889 status = BAD_VALUE; 7890 } 7891 } 7892 return status; 7893} 7894 7895void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7896{ 7897 for (size_t i = 0; i < mTracks.size(); ++i) { 7898 sp<Track> track = mTracks[i]; 7899 if (track->auxEffectId() == effectId) { 7900 attachAuxEffect_l(track, 0); 7901 } 7902 } 7903} 7904 7905status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7906{ 7907 // only one chain per input thread 7908 if (mEffectChains.size() != 0) { 7909 return INVALID_OPERATION; 7910 } 7911 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7912 7913 chain->setInBuffer(NULL); 7914 chain->setOutBuffer(NULL); 7915 7916 checkSuspendOnAddEffectChain_l(chain); 7917 7918 mEffectChains.add(chain); 7919 7920 return NO_ERROR; 7921} 7922 7923size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7924{ 7925 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7926 ALOGW_IF(mEffectChains.size() != 1, 7927 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7928 chain.get(), mEffectChains.size(), this); 7929 if (mEffectChains.size() == 1) { 7930 mEffectChains.removeAt(0); 7931 } 7932 return 0; 7933} 7934 7935// ---------------------------------------------------------------------------- 7936// EffectModule implementation 7937// ---------------------------------------------------------------------------- 7938 7939#undef LOG_TAG 7940#define LOG_TAG "AudioFlinger::EffectModule" 7941 7942AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7943 const wp<AudioFlinger::EffectChain>& chain, 7944 effect_descriptor_t *desc, 7945 int id, 7946 int sessionId) 7947 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7948 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7949{ 7950 ALOGV("Constructor %p", this); 7951 int lStatus; 7952 if (thread == NULL) { 7953 return; 7954 } 7955 7956 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7957 7958 // create effect engine from effect factory 7959 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7960 7961 if (mStatus != NO_ERROR) { 7962 return; 7963 } 7964 lStatus = init(); 7965 if (lStatus < 0) { 7966 mStatus = lStatus; 7967 goto Error; 7968 } 7969 7970 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7971 mPinned = true; 7972 } 7973 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7974 return; 7975Error: 7976 EffectRelease(mEffectInterface); 7977 mEffectInterface = NULL; 7978 ALOGV("Constructor Error %d", mStatus); 7979} 7980 7981AudioFlinger::EffectModule::~EffectModule() 7982{ 7983 ALOGV("Destructor %p", this); 7984 if (mEffectInterface != NULL) { 7985 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7986 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7987 sp<ThreadBase> thread = mThread.promote(); 7988 if (thread != 0) { 7989 audio_stream_t *stream = thread->stream(); 7990 if (stream != NULL) { 7991 stream->remove_audio_effect(stream, mEffectInterface); 7992 } 7993 } 7994 } 7995 // release effect engine 7996 EffectRelease(mEffectInterface); 7997 } 7998} 7999 8000status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8001{ 8002 status_t status; 8003 8004 Mutex::Autolock _l(mLock); 8005 int priority = handle->priority(); 8006 size_t size = mHandles.size(); 8007 sp<EffectHandle> h; 8008 size_t i; 8009 for (i = 0; i < size; i++) { 8010 h = mHandles[i].promote(); 8011 if (h == 0) continue; 8012 if (h->priority() <= priority) break; 8013 } 8014 // if inserted in first place, move effect control from previous owner to this handle 8015 if (i == 0) { 8016 bool enabled = false; 8017 if (h != 0) { 8018 enabled = h->enabled(); 8019 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8020 } 8021 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8022 status = NO_ERROR; 8023 } else { 8024 status = ALREADY_EXISTS; 8025 } 8026 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8027 mHandles.insertAt(handle, i); 8028 return status; 8029} 8030 8031size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8032{ 8033 Mutex::Autolock _l(mLock); 8034 size_t size = mHandles.size(); 8035 size_t i; 8036 for (i = 0; i < size; i++) { 8037 if (mHandles[i] == handle) break; 8038 } 8039 if (i == size) { 8040 return size; 8041 } 8042 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8043 8044 bool enabled = false; 8045 EffectHandle *hdl = handle.unsafe_get(); 8046 if (hdl != NULL) { 8047 ALOGV("removeHandle() unsafe_get OK"); 8048 enabled = hdl->enabled(); 8049 } 8050 mHandles.removeAt(i); 8051 size = mHandles.size(); 8052 // if removed from first place, move effect control from this handle to next in line 8053 if (i == 0 && size != 0) { 8054 sp<EffectHandle> h = mHandles[0].promote(); 8055 if (h != 0) { 8056 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8057 } 8058 } 8059 8060 // Prevent calls to process() and other functions on effect interface from now on. 8061 // The effect engine will be released by the destructor when the last strong reference on 8062 // this object is released which can happen after next process is called. 8063 if (size == 0 && !mPinned) { 8064 mState = DESTROYED; 8065 } 8066 8067 return size; 8068} 8069 8070sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8071{ 8072 Mutex::Autolock _l(mLock); 8073 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8074} 8075 8076void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8077{ 8078 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8079 // keep a strong reference on this EffectModule to avoid calling the 8080 // destructor before we exit 8081 sp<EffectModule> keep(this); 8082 { 8083 sp<ThreadBase> thread = mThread.promote(); 8084 if (thread != 0) { 8085 thread->disconnectEffect(keep, handle, unpinIfLast); 8086 } 8087 } 8088} 8089 8090void AudioFlinger::EffectModule::updateState() { 8091 Mutex::Autolock _l(mLock); 8092 8093 switch (mState) { 8094 case RESTART: 8095 reset_l(); 8096 // FALL THROUGH 8097 8098 case STARTING: 8099 // clear auxiliary effect input buffer for next accumulation 8100 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8101 memset(mConfig.inputCfg.buffer.raw, 8102 0, 8103 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8104 } 8105 start_l(); 8106 mState = ACTIVE; 8107 break; 8108 case STOPPING: 8109 stop_l(); 8110 mDisableWaitCnt = mMaxDisableWaitCnt; 8111 mState = STOPPED; 8112 break; 8113 case STOPPED: 8114 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8115 // turn off sequence. 8116 if (--mDisableWaitCnt == 0) { 8117 reset_l(); 8118 mState = IDLE; 8119 } 8120 break; 8121 default: //IDLE , ACTIVE, DESTROYED 8122 break; 8123 } 8124} 8125 8126void AudioFlinger::EffectModule::process() 8127{ 8128 Mutex::Autolock _l(mLock); 8129 8130 if (mState == DESTROYED || mEffectInterface == NULL || 8131 mConfig.inputCfg.buffer.raw == NULL || 8132 mConfig.outputCfg.buffer.raw == NULL) { 8133 return; 8134 } 8135 8136 if (isProcessEnabled()) { 8137 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8138 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8139 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8140 mConfig.inputCfg.buffer.s32, 8141 mConfig.inputCfg.buffer.frameCount/2); 8142 } 8143 8144 // do the actual processing in the effect engine 8145 int ret = (*mEffectInterface)->process(mEffectInterface, 8146 &mConfig.inputCfg.buffer, 8147 &mConfig.outputCfg.buffer); 8148 8149 // force transition to IDLE state when engine is ready 8150 if (mState == STOPPED && ret == -ENODATA) { 8151 mDisableWaitCnt = 1; 8152 } 8153 8154 // clear auxiliary effect input buffer for next accumulation 8155 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8156 memset(mConfig.inputCfg.buffer.raw, 0, 8157 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8158 } 8159 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8160 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8161 // If an insert effect is idle and input buffer is different from output buffer, 8162 // accumulate input onto output 8163 sp<EffectChain> chain = mChain.promote(); 8164 if (chain != 0 && chain->activeTrackCnt() != 0) { 8165 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8166 int16_t *in = mConfig.inputCfg.buffer.s16; 8167 int16_t *out = mConfig.outputCfg.buffer.s16; 8168 for (size_t i = 0; i < frameCnt; i++) { 8169 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8170 } 8171 } 8172 } 8173} 8174 8175void AudioFlinger::EffectModule::reset_l() 8176{ 8177 if (mEffectInterface == NULL) { 8178 return; 8179 } 8180 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8181} 8182 8183status_t AudioFlinger::EffectModule::configure() 8184{ 8185 uint32_t channels; 8186 if (mEffectInterface == NULL) { 8187 return NO_INIT; 8188 } 8189 8190 sp<ThreadBase> thread = mThread.promote(); 8191 if (thread == 0) { 8192 return DEAD_OBJECT; 8193 } 8194 8195 // TODO: handle configuration of effects replacing track process 8196 if (thread->channelCount() == 1) { 8197 channels = AUDIO_CHANNEL_OUT_MONO; 8198 } else { 8199 channels = AUDIO_CHANNEL_OUT_STEREO; 8200 } 8201 8202 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8203 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8204 } else { 8205 mConfig.inputCfg.channels = channels; 8206 } 8207 mConfig.outputCfg.channels = channels; 8208 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8209 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8210 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8211 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8212 mConfig.inputCfg.bufferProvider.cookie = NULL; 8213 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8214 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8215 mConfig.outputCfg.bufferProvider.cookie = NULL; 8216 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8217 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8218 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8219 // Insert effect: 8220 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8221 // always overwrites output buffer: input buffer == output buffer 8222 // - in other sessions: 8223 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8224 // other effect: overwrites output buffer: input buffer == output buffer 8225 // Auxiliary effect: 8226 // accumulates in output buffer: input buffer != output buffer 8227 // Therefore: accumulate <=> input buffer != output buffer 8228 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8229 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8230 } else { 8231 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8232 } 8233 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8234 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8235 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8236 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8237 8238 ALOGV("configure() %p thread %p buffer %p framecount %d", 8239 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8240 8241 status_t cmdStatus; 8242 uint32_t size = sizeof(int); 8243 status_t status = (*mEffectInterface)->command(mEffectInterface, 8244 EFFECT_CMD_SET_CONFIG, 8245 sizeof(effect_config_t), 8246 &mConfig, 8247 &size, 8248 &cmdStatus); 8249 if (status == 0) { 8250 status = cmdStatus; 8251 } 8252 8253 if (status == 0 && 8254 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8255 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8256 effect_param_t *p = (effect_param_t *)buf32; 8257 8258 p->psize = sizeof(uint32_t); 8259 p->vsize = sizeof(uint32_t); 8260 size = sizeof(int); 8261 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8262 8263 uint32_t latency = 0; 8264 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8265 if (pbt != NULL) { 8266 latency = pbt->latency_l(); 8267 } 8268 8269 *((int32_t *)p->data + 1)= latency; 8270 (*mEffectInterface)->command(mEffectInterface, 8271 EFFECT_CMD_SET_PARAM, 8272 sizeof(effect_param_t) + 8, 8273 &buf32, 8274 &size, 8275 &cmdStatus); 8276 } 8277 8278 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8279 (1000 * mConfig.outputCfg.buffer.frameCount); 8280 8281 return status; 8282} 8283 8284status_t AudioFlinger::EffectModule::init() 8285{ 8286 Mutex::Autolock _l(mLock); 8287 if (mEffectInterface == NULL) { 8288 return NO_INIT; 8289 } 8290 status_t cmdStatus; 8291 uint32_t size = sizeof(status_t); 8292 status_t status = (*mEffectInterface)->command(mEffectInterface, 8293 EFFECT_CMD_INIT, 8294 0, 8295 NULL, 8296 &size, 8297 &cmdStatus); 8298 if (status == 0) { 8299 status = cmdStatus; 8300 } 8301 return status; 8302} 8303 8304status_t AudioFlinger::EffectModule::start() 8305{ 8306 Mutex::Autolock _l(mLock); 8307 return start_l(); 8308} 8309 8310status_t AudioFlinger::EffectModule::start_l() 8311{ 8312 if (mEffectInterface == NULL) { 8313 return NO_INIT; 8314 } 8315 status_t cmdStatus; 8316 uint32_t size = sizeof(status_t); 8317 status_t status = (*mEffectInterface)->command(mEffectInterface, 8318 EFFECT_CMD_ENABLE, 8319 0, 8320 NULL, 8321 &size, 8322 &cmdStatus); 8323 if (status == 0) { 8324 status = cmdStatus; 8325 } 8326 if (status == 0 && 8327 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8328 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8329 sp<ThreadBase> thread = mThread.promote(); 8330 if (thread != 0) { 8331 audio_stream_t *stream = thread->stream(); 8332 if (stream != NULL) { 8333 stream->add_audio_effect(stream, mEffectInterface); 8334 } 8335 } 8336 } 8337 return status; 8338} 8339 8340status_t AudioFlinger::EffectModule::stop() 8341{ 8342 Mutex::Autolock _l(mLock); 8343 return stop_l(); 8344} 8345 8346status_t AudioFlinger::EffectModule::stop_l() 8347{ 8348 if (mEffectInterface == NULL) { 8349 return NO_INIT; 8350 } 8351 status_t cmdStatus; 8352 uint32_t size = sizeof(status_t); 8353 status_t status = (*mEffectInterface)->command(mEffectInterface, 8354 EFFECT_CMD_DISABLE, 8355 0, 8356 NULL, 8357 &size, 8358 &cmdStatus); 8359 if (status == 0) { 8360 status = cmdStatus; 8361 } 8362 if (status == 0 && 8363 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8364 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8365 sp<ThreadBase> thread = mThread.promote(); 8366 if (thread != 0) { 8367 audio_stream_t *stream = thread->stream(); 8368 if (stream != NULL) { 8369 stream->remove_audio_effect(stream, mEffectInterface); 8370 } 8371 } 8372 } 8373 return status; 8374} 8375 8376status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8377 uint32_t cmdSize, 8378 void *pCmdData, 8379 uint32_t *replySize, 8380 void *pReplyData) 8381{ 8382 Mutex::Autolock _l(mLock); 8383// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8384 8385 if (mState == DESTROYED || mEffectInterface == NULL) { 8386 return NO_INIT; 8387 } 8388 status_t status = (*mEffectInterface)->command(mEffectInterface, 8389 cmdCode, 8390 cmdSize, 8391 pCmdData, 8392 replySize, 8393 pReplyData); 8394 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8395 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8396 for (size_t i = 1; i < mHandles.size(); i++) { 8397 sp<EffectHandle> h = mHandles[i].promote(); 8398 if (h != 0) { 8399 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8400 } 8401 } 8402 } 8403 return status; 8404} 8405 8406status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8407{ 8408 8409 Mutex::Autolock _l(mLock); 8410 ALOGV("setEnabled %p enabled %d", this, enabled); 8411 8412 if (enabled != isEnabled()) { 8413 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8414 if (enabled && status != NO_ERROR) { 8415 return status; 8416 } 8417 8418 switch (mState) { 8419 // going from disabled to enabled 8420 case IDLE: 8421 mState = STARTING; 8422 break; 8423 case STOPPED: 8424 mState = RESTART; 8425 break; 8426 case STOPPING: 8427 mState = ACTIVE; 8428 break; 8429 8430 // going from enabled to disabled 8431 case RESTART: 8432 mState = STOPPED; 8433 break; 8434 case STARTING: 8435 mState = IDLE; 8436 break; 8437 case ACTIVE: 8438 mState = STOPPING; 8439 break; 8440 case DESTROYED: 8441 return NO_ERROR; // simply ignore as we are being destroyed 8442 } 8443 for (size_t i = 1; i < mHandles.size(); i++) { 8444 sp<EffectHandle> h = mHandles[i].promote(); 8445 if (h != 0) { 8446 h->setEnabled(enabled); 8447 } 8448 } 8449 } 8450 return NO_ERROR; 8451} 8452 8453bool AudioFlinger::EffectModule::isEnabled() const 8454{ 8455 switch (mState) { 8456 case RESTART: 8457 case STARTING: 8458 case ACTIVE: 8459 return true; 8460 case IDLE: 8461 case STOPPING: 8462 case STOPPED: 8463 case DESTROYED: 8464 default: 8465 return false; 8466 } 8467} 8468 8469bool AudioFlinger::EffectModule::isProcessEnabled() const 8470{ 8471 switch (mState) { 8472 case RESTART: 8473 case ACTIVE: 8474 case STOPPING: 8475 case STOPPED: 8476 return true; 8477 case IDLE: 8478 case STARTING: 8479 case DESTROYED: 8480 default: 8481 return false; 8482 } 8483} 8484 8485status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8486{ 8487 Mutex::Autolock _l(mLock); 8488 status_t status = NO_ERROR; 8489 8490 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8491 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8492 if (isProcessEnabled() && 8493 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8494 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8495 status_t cmdStatus; 8496 uint32_t volume[2]; 8497 uint32_t *pVolume = NULL; 8498 uint32_t size = sizeof(volume); 8499 volume[0] = *left; 8500 volume[1] = *right; 8501 if (controller) { 8502 pVolume = volume; 8503 } 8504 status = (*mEffectInterface)->command(mEffectInterface, 8505 EFFECT_CMD_SET_VOLUME, 8506 size, 8507 volume, 8508 &size, 8509 pVolume); 8510 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8511 *left = volume[0]; 8512 *right = volume[1]; 8513 } 8514 } 8515 return status; 8516} 8517 8518status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8519{ 8520 Mutex::Autolock _l(mLock); 8521 status_t status = NO_ERROR; 8522 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8523 // audio pre processing modules on RecordThread can receive both output and 8524 // input device indication in the same call 8525 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8526 if (dev) { 8527 status_t cmdStatus; 8528 uint32_t size = sizeof(status_t); 8529 8530 status = (*mEffectInterface)->command(mEffectInterface, 8531 EFFECT_CMD_SET_DEVICE, 8532 sizeof(uint32_t), 8533 &dev, 8534 &size, 8535 &cmdStatus); 8536 if (status == NO_ERROR) { 8537 status = cmdStatus; 8538 } 8539 } 8540 dev = device & AUDIO_DEVICE_IN_ALL; 8541 if (dev) { 8542 status_t cmdStatus; 8543 uint32_t size = sizeof(status_t); 8544 8545 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8546 EFFECT_CMD_SET_INPUT_DEVICE, 8547 sizeof(uint32_t), 8548 &dev, 8549 &size, 8550 &cmdStatus); 8551 if (status2 == NO_ERROR) { 8552 status2 = cmdStatus; 8553 } 8554 if (status == NO_ERROR) { 8555 status = status2; 8556 } 8557 } 8558 } 8559 return status; 8560} 8561 8562status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8563{ 8564 Mutex::Autolock _l(mLock); 8565 status_t status = NO_ERROR; 8566 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8567 status_t cmdStatus; 8568 uint32_t size = sizeof(status_t); 8569 status = (*mEffectInterface)->command(mEffectInterface, 8570 EFFECT_CMD_SET_AUDIO_MODE, 8571 sizeof(audio_mode_t), 8572 &mode, 8573 &size, 8574 &cmdStatus); 8575 if (status == NO_ERROR) { 8576 status = cmdStatus; 8577 } 8578 } 8579 return status; 8580} 8581 8582void AudioFlinger::EffectModule::setSuspended(bool suspended) 8583{ 8584 Mutex::Autolock _l(mLock); 8585 mSuspended = suspended; 8586} 8587 8588bool AudioFlinger::EffectModule::suspended() const 8589{ 8590 Mutex::Autolock _l(mLock); 8591 return mSuspended; 8592} 8593 8594status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8595{ 8596 const size_t SIZE = 256; 8597 char buffer[SIZE]; 8598 String8 result; 8599 8600 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8601 result.append(buffer); 8602 8603 bool locked = tryLock(mLock); 8604 // failed to lock - AudioFlinger is probably deadlocked 8605 if (!locked) { 8606 result.append("\t\tCould not lock Fx mutex:\n"); 8607 } 8608 8609 result.append("\t\tSession Status State Engine:\n"); 8610 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8611 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8612 result.append(buffer); 8613 8614 result.append("\t\tDescriptor:\n"); 8615 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8616 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8617 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8618 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8619 result.append(buffer); 8620 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8621 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8622 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8623 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8624 result.append(buffer); 8625 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8626 mDescriptor.apiVersion, 8627 mDescriptor.flags); 8628 result.append(buffer); 8629 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8630 mDescriptor.name); 8631 result.append(buffer); 8632 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8633 mDescriptor.implementor); 8634 result.append(buffer); 8635 8636 result.append("\t\t- Input configuration:\n"); 8637 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8638 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8639 (uint32_t)mConfig.inputCfg.buffer.raw, 8640 mConfig.inputCfg.buffer.frameCount, 8641 mConfig.inputCfg.samplingRate, 8642 mConfig.inputCfg.channels, 8643 mConfig.inputCfg.format); 8644 result.append(buffer); 8645 8646 result.append("\t\t- Output configuration:\n"); 8647 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8648 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8649 (uint32_t)mConfig.outputCfg.buffer.raw, 8650 mConfig.outputCfg.buffer.frameCount, 8651 mConfig.outputCfg.samplingRate, 8652 mConfig.outputCfg.channels, 8653 mConfig.outputCfg.format); 8654 result.append(buffer); 8655 8656 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8657 result.append(buffer); 8658 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8659 for (size_t i = 0; i < mHandles.size(); ++i) { 8660 sp<EffectHandle> handle = mHandles[i].promote(); 8661 if (handle != 0) { 8662 handle->dump(buffer, SIZE); 8663 result.append(buffer); 8664 } 8665 } 8666 8667 result.append("\n"); 8668 8669 write(fd, result.string(), result.length()); 8670 8671 if (locked) { 8672 mLock.unlock(); 8673 } 8674 8675 return NO_ERROR; 8676} 8677 8678// ---------------------------------------------------------------------------- 8679// EffectHandle implementation 8680// ---------------------------------------------------------------------------- 8681 8682#undef LOG_TAG 8683#define LOG_TAG "AudioFlinger::EffectHandle" 8684 8685AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8686 const sp<AudioFlinger::Client>& client, 8687 const sp<IEffectClient>& effectClient, 8688 int32_t priority) 8689 : BnEffect(), 8690 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8691 mPriority(priority), mHasControl(false), mEnabled(false) 8692{ 8693 ALOGV("constructor %p", this); 8694 8695 if (client == 0) { 8696 return; 8697 } 8698 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8699 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8700 if (mCblkMemory != 0) { 8701 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8702 8703 if (mCblk != NULL) { 8704 new(mCblk) effect_param_cblk_t(); 8705 mBuffer = (uint8_t *)mCblk + bufOffset; 8706 } 8707 } else { 8708 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8709 return; 8710 } 8711} 8712 8713AudioFlinger::EffectHandle::~EffectHandle() 8714{ 8715 ALOGV("Destructor %p", this); 8716 disconnect(false); 8717 ALOGV("Destructor DONE %p", this); 8718} 8719 8720status_t AudioFlinger::EffectHandle::enable() 8721{ 8722 ALOGV("enable %p", this); 8723 if (!mHasControl) return INVALID_OPERATION; 8724 if (mEffect == 0) return DEAD_OBJECT; 8725 8726 if (mEnabled) { 8727 return NO_ERROR; 8728 } 8729 8730 mEnabled = true; 8731 8732 sp<ThreadBase> thread = mEffect->thread().promote(); 8733 if (thread != 0) { 8734 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8735 } 8736 8737 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8738 if (mEffect->suspended()) { 8739 return NO_ERROR; 8740 } 8741 8742 status_t status = mEffect->setEnabled(true); 8743 if (status != NO_ERROR) { 8744 if (thread != 0) { 8745 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8746 } 8747 mEnabled = false; 8748 } 8749 return status; 8750} 8751 8752status_t AudioFlinger::EffectHandle::disable() 8753{ 8754 ALOGV("disable %p", this); 8755 if (!mHasControl) return INVALID_OPERATION; 8756 if (mEffect == 0) return DEAD_OBJECT; 8757 8758 if (!mEnabled) { 8759 return NO_ERROR; 8760 } 8761 mEnabled = false; 8762 8763 if (mEffect->suspended()) { 8764 return NO_ERROR; 8765 } 8766 8767 status_t status = mEffect->setEnabled(false); 8768 8769 sp<ThreadBase> thread = mEffect->thread().promote(); 8770 if (thread != 0) { 8771 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8772 } 8773 8774 return status; 8775} 8776 8777void AudioFlinger::EffectHandle::disconnect() 8778{ 8779 disconnect(true); 8780} 8781 8782void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8783{ 8784 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8785 if (mEffect == 0) { 8786 return; 8787 } 8788 mEffect->disconnect(this, unpinIfLast); 8789 8790 if (mHasControl && mEnabled) { 8791 sp<ThreadBase> thread = mEffect->thread().promote(); 8792 if (thread != 0) { 8793 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8794 } 8795 } 8796 8797 // release sp on module => module destructor can be called now 8798 mEffect.clear(); 8799 if (mClient != 0) { 8800 if (mCblk != NULL) { 8801 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8802 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8803 } 8804 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8805 // Client destructor must run with AudioFlinger mutex locked 8806 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8807 mClient.clear(); 8808 } 8809} 8810 8811status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8812 uint32_t cmdSize, 8813 void *pCmdData, 8814 uint32_t *replySize, 8815 void *pReplyData) 8816{ 8817// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8818// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8819 8820 // only get parameter command is permitted for applications not controlling the effect 8821 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8822 return INVALID_OPERATION; 8823 } 8824 if (mEffect == 0) return DEAD_OBJECT; 8825 if (mClient == 0) return INVALID_OPERATION; 8826 8827 // handle commands that are not forwarded transparently to effect engine 8828 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8829 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8830 // no risk to block the whole media server process or mixer threads is we are stuck here 8831 Mutex::Autolock _l(mCblk->lock); 8832 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8833 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8834 mCblk->serverIndex = 0; 8835 mCblk->clientIndex = 0; 8836 return BAD_VALUE; 8837 } 8838 status_t status = NO_ERROR; 8839 while (mCblk->serverIndex < mCblk->clientIndex) { 8840 int reply; 8841 uint32_t rsize = sizeof(int); 8842 int *p = (int *)(mBuffer + mCblk->serverIndex); 8843 int size = *p++; 8844 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8845 ALOGW("command(): invalid parameter block size"); 8846 break; 8847 } 8848 effect_param_t *param = (effect_param_t *)p; 8849 if (param->psize == 0 || param->vsize == 0) { 8850 ALOGW("command(): null parameter or value size"); 8851 mCblk->serverIndex += size; 8852 continue; 8853 } 8854 uint32_t psize = sizeof(effect_param_t) + 8855 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8856 param->vsize; 8857 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8858 psize, 8859 p, 8860 &rsize, 8861 &reply); 8862 // stop at first error encountered 8863 if (ret != NO_ERROR) { 8864 status = ret; 8865 *(int *)pReplyData = reply; 8866 break; 8867 } else if (reply != NO_ERROR) { 8868 *(int *)pReplyData = reply; 8869 break; 8870 } 8871 mCblk->serverIndex += size; 8872 } 8873 mCblk->serverIndex = 0; 8874 mCblk->clientIndex = 0; 8875 return status; 8876 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8877 *(int *)pReplyData = NO_ERROR; 8878 return enable(); 8879 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8880 *(int *)pReplyData = NO_ERROR; 8881 return disable(); 8882 } 8883 8884 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8885} 8886 8887void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8888{ 8889 ALOGV("setControl %p control %d", this, hasControl); 8890 8891 mHasControl = hasControl; 8892 mEnabled = enabled; 8893 8894 if (signal && mEffectClient != 0) { 8895 mEffectClient->controlStatusChanged(hasControl); 8896 } 8897} 8898 8899void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8900 uint32_t cmdSize, 8901 void *pCmdData, 8902 uint32_t replySize, 8903 void *pReplyData) 8904{ 8905 if (mEffectClient != 0) { 8906 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8907 } 8908} 8909 8910 8911 8912void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8913{ 8914 if (mEffectClient != 0) { 8915 mEffectClient->enableStatusChanged(enabled); 8916 } 8917} 8918 8919status_t AudioFlinger::EffectHandle::onTransact( 8920 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8921{ 8922 return BnEffect::onTransact(code, data, reply, flags); 8923} 8924 8925 8926void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8927{ 8928 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8929 8930 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8931 (mClient == 0) ? getpid_cached : mClient->pid(), 8932 mPriority, 8933 mHasControl, 8934 !locked, 8935 mCblk ? mCblk->clientIndex : 0, 8936 mCblk ? mCblk->serverIndex : 0 8937 ); 8938 8939 if (locked) { 8940 mCblk->lock.unlock(); 8941 } 8942} 8943 8944#undef LOG_TAG 8945#define LOG_TAG "AudioFlinger::EffectChain" 8946 8947AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8948 int sessionId) 8949 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8950 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8951 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8952{ 8953 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8954 if (thread == NULL) { 8955 return; 8956 } 8957 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8958 thread->frameCount(); 8959} 8960 8961AudioFlinger::EffectChain::~EffectChain() 8962{ 8963 if (mOwnInBuffer) { 8964 delete mInBuffer; 8965 } 8966 8967} 8968 8969// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8970sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8971{ 8972 size_t size = mEffects.size(); 8973 8974 for (size_t i = 0; i < size; i++) { 8975 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8976 return mEffects[i]; 8977 } 8978 } 8979 return 0; 8980} 8981 8982// getEffectFromId_l() must be called with ThreadBase::mLock held 8983sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8984{ 8985 size_t size = mEffects.size(); 8986 8987 for (size_t i = 0; i < size; i++) { 8988 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8989 if (id == 0 || mEffects[i]->id() == id) { 8990 return mEffects[i]; 8991 } 8992 } 8993 return 0; 8994} 8995 8996// getEffectFromType_l() must be called with ThreadBase::mLock held 8997sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8998 const effect_uuid_t *type) 8999{ 9000 size_t size = mEffects.size(); 9001 9002 for (size_t i = 0; i < size; i++) { 9003 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9004 return mEffects[i]; 9005 } 9006 } 9007 return 0; 9008} 9009 9010void AudioFlinger::EffectChain::clearInputBuffer() 9011{ 9012 Mutex::Autolock _l(mLock); 9013 sp<ThreadBase> thread = mThread.promote(); 9014 if (thread == 0) { 9015 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9016 return; 9017 } 9018 clearInputBuffer_l(thread); 9019} 9020 9021// Must be called with EffectChain::mLock locked 9022void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9023{ 9024 size_t numSamples = thread->frameCount() * thread->channelCount(); 9025 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9026 9027} 9028 9029// Must be called with EffectChain::mLock locked 9030void AudioFlinger::EffectChain::process_l() 9031{ 9032 sp<ThreadBase> thread = mThread.promote(); 9033 if (thread == 0) { 9034 ALOGW("process_l(): cannot promote mixer thread"); 9035 return; 9036 } 9037 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9038 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9039 // always process effects unless no more tracks are on the session and the effect tail 9040 // has been rendered 9041 bool doProcess = true; 9042 if (!isGlobalSession) { 9043 bool tracksOnSession = (trackCnt() != 0); 9044 9045 if (!tracksOnSession && mTailBufferCount == 0) { 9046 doProcess = false; 9047 } 9048 9049 if (activeTrackCnt() == 0) { 9050 // if no track is active and the effect tail has not been rendered, 9051 // the input buffer must be cleared here as the mixer process will not do it 9052 if (tracksOnSession || mTailBufferCount > 0) { 9053 clearInputBuffer_l(thread); 9054 if (mTailBufferCount > 0) { 9055 mTailBufferCount--; 9056 } 9057 } 9058 } 9059 } 9060 9061 size_t size = mEffects.size(); 9062 if (doProcess) { 9063 for (size_t i = 0; i < size; i++) { 9064 mEffects[i]->process(); 9065 } 9066 } 9067 for (size_t i = 0; i < size; i++) { 9068 mEffects[i]->updateState(); 9069 } 9070} 9071 9072// addEffect_l() must be called with PlaybackThread::mLock held 9073status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9074{ 9075 effect_descriptor_t desc = effect->desc(); 9076 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9077 9078 Mutex::Autolock _l(mLock); 9079 effect->setChain(this); 9080 sp<ThreadBase> thread = mThread.promote(); 9081 if (thread == 0) { 9082 return NO_INIT; 9083 } 9084 effect->setThread(thread); 9085 9086 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9087 // Auxiliary effects are inserted at the beginning of mEffects vector as 9088 // they are processed first and accumulated in chain input buffer 9089 mEffects.insertAt(effect, 0); 9090 9091 // the input buffer for auxiliary effect contains mono samples in 9092 // 32 bit format. This is to avoid saturation in AudoMixer 9093 // accumulation stage. Saturation is done in EffectModule::process() before 9094 // calling the process in effect engine 9095 size_t numSamples = thread->frameCount(); 9096 int32_t *buffer = new int32_t[numSamples]; 9097 memset(buffer, 0, numSamples * sizeof(int32_t)); 9098 effect->setInBuffer((int16_t *)buffer); 9099 // auxiliary effects output samples to chain input buffer for further processing 9100 // by insert effects 9101 effect->setOutBuffer(mInBuffer); 9102 } else { 9103 // Insert effects are inserted at the end of mEffects vector as they are processed 9104 // after track and auxiliary effects. 9105 // Insert effect order as a function of indicated preference: 9106 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9107 // another effect is present 9108 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9109 // last effect claiming first position 9110 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9111 // first effect claiming last position 9112 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9113 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9114 // already present 9115 9116 size_t size = mEffects.size(); 9117 size_t idx_insert = size; 9118 ssize_t idx_insert_first = -1; 9119 ssize_t idx_insert_last = -1; 9120 9121 for (size_t i = 0; i < size; i++) { 9122 effect_descriptor_t d = mEffects[i]->desc(); 9123 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9124 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9125 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9126 // check invalid effect chaining combinations 9127 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9128 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9129 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9130 return INVALID_OPERATION; 9131 } 9132 // remember position of first insert effect and by default 9133 // select this as insert position for new effect 9134 if (idx_insert == size) { 9135 idx_insert = i; 9136 } 9137 // remember position of last insert effect claiming 9138 // first position 9139 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9140 idx_insert_first = i; 9141 } 9142 // remember position of first insert effect claiming 9143 // last position 9144 if (iPref == EFFECT_FLAG_INSERT_LAST && 9145 idx_insert_last == -1) { 9146 idx_insert_last = i; 9147 } 9148 } 9149 } 9150 9151 // modify idx_insert from first position if needed 9152 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9153 if (idx_insert_last != -1) { 9154 idx_insert = idx_insert_last; 9155 } else { 9156 idx_insert = size; 9157 } 9158 } else { 9159 if (idx_insert_first != -1) { 9160 idx_insert = idx_insert_first + 1; 9161 } 9162 } 9163 9164 // always read samples from chain input buffer 9165 effect->setInBuffer(mInBuffer); 9166 9167 // if last effect in the chain, output samples to chain 9168 // output buffer, otherwise to chain input buffer 9169 if (idx_insert == size) { 9170 if (idx_insert != 0) { 9171 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9172 mEffects[idx_insert-1]->configure(); 9173 } 9174 effect->setOutBuffer(mOutBuffer); 9175 } else { 9176 effect->setOutBuffer(mInBuffer); 9177 } 9178 mEffects.insertAt(effect, idx_insert); 9179 9180 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9181 } 9182 effect->configure(); 9183 return NO_ERROR; 9184} 9185 9186// removeEffect_l() must be called with PlaybackThread::mLock held 9187size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9188{ 9189 Mutex::Autolock _l(mLock); 9190 size_t size = mEffects.size(); 9191 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9192 9193 for (size_t i = 0; i < size; i++) { 9194 if (effect == mEffects[i]) { 9195 // calling stop here will remove pre-processing effect from the audio HAL. 9196 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9197 // the middle of a read from audio HAL 9198 if (mEffects[i]->state() == EffectModule::ACTIVE || 9199 mEffects[i]->state() == EffectModule::STOPPING) { 9200 mEffects[i]->stop(); 9201 } 9202 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9203 delete[] effect->inBuffer(); 9204 } else { 9205 if (i == size - 1 && i != 0) { 9206 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9207 mEffects[i - 1]->configure(); 9208 } 9209 } 9210 mEffects.removeAt(i); 9211 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9212 break; 9213 } 9214 } 9215 9216 return mEffects.size(); 9217} 9218 9219// setDevice_l() must be called with PlaybackThread::mLock held 9220void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9221{ 9222 size_t size = mEffects.size(); 9223 for (size_t i = 0; i < size; i++) { 9224 mEffects[i]->setDevice(device); 9225 } 9226} 9227 9228// setMode_l() must be called with PlaybackThread::mLock held 9229void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9230{ 9231 size_t size = mEffects.size(); 9232 for (size_t i = 0; i < size; i++) { 9233 mEffects[i]->setMode(mode); 9234 } 9235} 9236 9237// setVolume_l() must be called with PlaybackThread::mLock held 9238bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9239{ 9240 uint32_t newLeft = *left; 9241 uint32_t newRight = *right; 9242 bool hasControl = false; 9243 int ctrlIdx = -1; 9244 size_t size = mEffects.size(); 9245 9246 // first update volume controller 9247 for (size_t i = size; i > 0; i--) { 9248 if (mEffects[i - 1]->isProcessEnabled() && 9249 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9250 ctrlIdx = i - 1; 9251 hasControl = true; 9252 break; 9253 } 9254 } 9255 9256 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9257 if (hasControl) { 9258 *left = mNewLeftVolume; 9259 *right = mNewRightVolume; 9260 } 9261 return hasControl; 9262 } 9263 9264 mVolumeCtrlIdx = ctrlIdx; 9265 mLeftVolume = newLeft; 9266 mRightVolume = newRight; 9267 9268 // second get volume update from volume controller 9269 if (ctrlIdx >= 0) { 9270 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9271 mNewLeftVolume = newLeft; 9272 mNewRightVolume = newRight; 9273 } 9274 // then indicate volume to all other effects in chain. 9275 // Pass altered volume to effects before volume controller 9276 // and requested volume to effects after controller 9277 uint32_t lVol = newLeft; 9278 uint32_t rVol = newRight; 9279 9280 for (size_t i = 0; i < size; i++) { 9281 if ((int)i == ctrlIdx) continue; 9282 // this also works for ctrlIdx == -1 when there is no volume controller 9283 if ((int)i > ctrlIdx) { 9284 lVol = *left; 9285 rVol = *right; 9286 } 9287 mEffects[i]->setVolume(&lVol, &rVol, false); 9288 } 9289 *left = newLeft; 9290 *right = newRight; 9291 9292 return hasControl; 9293} 9294 9295status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9296{ 9297 const size_t SIZE = 256; 9298 char buffer[SIZE]; 9299 String8 result; 9300 9301 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9302 result.append(buffer); 9303 9304 bool locked = tryLock(mLock); 9305 // failed to lock - AudioFlinger is probably deadlocked 9306 if (!locked) { 9307 result.append("\tCould not lock mutex:\n"); 9308 } 9309 9310 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9311 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9312 mEffects.size(), 9313 (uint32_t)mInBuffer, 9314 (uint32_t)mOutBuffer, 9315 mActiveTrackCnt); 9316 result.append(buffer); 9317 write(fd, result.string(), result.size()); 9318 9319 for (size_t i = 0; i < mEffects.size(); ++i) { 9320 sp<EffectModule> effect = mEffects[i]; 9321 if (effect != 0) { 9322 effect->dump(fd, args); 9323 } 9324 } 9325 9326 if (locked) { 9327 mLock.unlock(); 9328 } 9329 9330 return NO_ERROR; 9331} 9332 9333// must be called with ThreadBase::mLock held 9334void AudioFlinger::EffectChain::setEffectSuspended_l( 9335 const effect_uuid_t *type, bool suspend) 9336{ 9337 sp<SuspendedEffectDesc> desc; 9338 // use effect type UUID timelow as key as there is no real risk of identical 9339 // timeLow fields among effect type UUIDs. 9340 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9341 if (suspend) { 9342 if (index >= 0) { 9343 desc = mSuspendedEffects.valueAt(index); 9344 } else { 9345 desc = new SuspendedEffectDesc(); 9346 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9347 mSuspendedEffects.add(type->timeLow, desc); 9348 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9349 } 9350 if (desc->mRefCount++ == 0) { 9351 sp<EffectModule> effect = getEffectIfEnabled(type); 9352 if (effect != 0) { 9353 desc->mEffect = effect; 9354 effect->setSuspended(true); 9355 effect->setEnabled(false); 9356 } 9357 } 9358 } else { 9359 if (index < 0) { 9360 return; 9361 } 9362 desc = mSuspendedEffects.valueAt(index); 9363 if (desc->mRefCount <= 0) { 9364 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9365 desc->mRefCount = 1; 9366 } 9367 if (--desc->mRefCount == 0) { 9368 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9369 if (desc->mEffect != 0) { 9370 sp<EffectModule> effect = desc->mEffect.promote(); 9371 if (effect != 0) { 9372 effect->setSuspended(false); 9373 sp<EffectHandle> handle = effect->controlHandle(); 9374 if (handle != 0) { 9375 effect->setEnabled(handle->enabled()); 9376 } 9377 } 9378 desc->mEffect.clear(); 9379 } 9380 mSuspendedEffects.removeItemsAt(index); 9381 } 9382 } 9383} 9384 9385// must be called with ThreadBase::mLock held 9386void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9387{ 9388 sp<SuspendedEffectDesc> desc; 9389 9390 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9391 if (suspend) { 9392 if (index >= 0) { 9393 desc = mSuspendedEffects.valueAt(index); 9394 } else { 9395 desc = new SuspendedEffectDesc(); 9396 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9397 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9398 } 9399 if (desc->mRefCount++ == 0) { 9400 Vector< sp<EffectModule> > effects; 9401 getSuspendEligibleEffects(effects); 9402 for (size_t i = 0; i < effects.size(); i++) { 9403 setEffectSuspended_l(&effects[i]->desc().type, true); 9404 } 9405 } 9406 } else { 9407 if (index < 0) { 9408 return; 9409 } 9410 desc = mSuspendedEffects.valueAt(index); 9411 if (desc->mRefCount <= 0) { 9412 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9413 desc->mRefCount = 1; 9414 } 9415 if (--desc->mRefCount == 0) { 9416 Vector<const effect_uuid_t *> types; 9417 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9418 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9419 continue; 9420 } 9421 types.add(&mSuspendedEffects.valueAt(i)->mType); 9422 } 9423 for (size_t i = 0; i < types.size(); i++) { 9424 setEffectSuspended_l(types[i], false); 9425 } 9426 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9427 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9428 } 9429 } 9430} 9431 9432 9433// The volume effect is used for automated tests only 9434#ifndef OPENSL_ES_H_ 9435static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9436 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9437const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9438#endif //OPENSL_ES_H_ 9439 9440bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9441{ 9442 // auxiliary effects and visualizer are never suspended on output mix 9443 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9444 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9445 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9446 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9447 return false; 9448 } 9449 return true; 9450} 9451 9452void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9453{ 9454 effects.clear(); 9455 for (size_t i = 0; i < mEffects.size(); i++) { 9456 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9457 effects.add(mEffects[i]); 9458 } 9459 } 9460} 9461 9462sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9463 const effect_uuid_t *type) 9464{ 9465 sp<EffectModule> effect = getEffectFromType_l(type); 9466 return effect != 0 && effect->isEnabled() ? effect : 0; 9467} 9468 9469void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9470 bool enabled) 9471{ 9472 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9473 if (enabled) { 9474 if (index < 0) { 9475 // if the effect is not suspend check if all effects are suspended 9476 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9477 if (index < 0) { 9478 return; 9479 } 9480 if (!isEffectEligibleForSuspend(effect->desc())) { 9481 return; 9482 } 9483 setEffectSuspended_l(&effect->desc().type, enabled); 9484 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9485 if (index < 0) { 9486 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9487 return; 9488 } 9489 } 9490 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9491 effect->desc().type.timeLow); 9492 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9493 // if effect is requested to suspended but was not yet enabled, supend it now. 9494 if (desc->mEffect == 0) { 9495 desc->mEffect = effect; 9496 effect->setEnabled(false); 9497 effect->setSuspended(true); 9498 } 9499 } else { 9500 if (index < 0) { 9501 return; 9502 } 9503 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9504 effect->desc().type.timeLow); 9505 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9506 desc->mEffect.clear(); 9507 effect->setSuspended(false); 9508 } 9509} 9510 9511#undef LOG_TAG 9512#define LOG_TAG "AudioFlinger" 9513 9514// ---------------------------------------------------------------------------- 9515 9516status_t AudioFlinger::onTransact( 9517 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9518{ 9519 return BnAudioFlinger::onTransact(code, data, reply, flags); 9520} 9521 9522}; // namespace android 9523