AudioFlinger.cpp revision ab9071b8d1b375418eb797c9a790da71de644344
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167// ---------------------------------------------------------------------------- 168 169#ifdef ADD_BATTERY_DATA 170// To collect the amplifier usage 171static void addBatteryData(uint32_t params) { 172 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 173 if (service == NULL) { 174 // it already logged 175 return; 176 } 177 178 service->addBatteryData(params); 179} 180#endif 181 182static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183{ 184 const hw_module_t *mod; 185 int rc; 186 187 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190 if (rc) { 191 goto out; 192 } 193 rc = audio_hw_device_open(mod, dev); 194 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 195 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196 if (rc) { 197 goto out; 198 } 199 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201 rc = BAD_VALUE; 202 goto out; 203 } 204 return 0; 205 206out: 207 *dev = NULL; 208 return rc; 209} 210 211// ---------------------------------------------------------------------------- 212 213AudioFlinger::AudioFlinger() 214 : BnAudioFlinger(), 215 mPrimaryHardwareDev(NULL), 216 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217 mMasterVolume(1.0f), 218 mMasterVolumeSupportLvl(MVS_NONE), 219 mMasterMute(false), 220 mNextUniqueId(1), 221 mMode(AUDIO_MODE_INVALID), 222 mBtNrecIsOff(false) 223{ 224} 225 226void AudioFlinger::onFirstRef() 227{ 228 int rc = 0; 229 230 Mutex::Autolock _l(mLock); 231 232 /* TODO: move all this work into an Init() function */ 233 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 235 uint32_t int_val; 236 if (1 == sscanf(val_str, "%u", &int_val)) { 237 mStandbyTimeInNsecs = milliseconds(int_val); 238 ALOGI("Using %u mSec as standby time.", int_val); 239 } else { 240 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241 ALOGI("Using default %u mSec as standby time.", 242 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243 } 244 } 245 246 mMode = AUDIO_MODE_NORMAL; 247 mMasterVolumeSW = 1.0; 248 mMasterVolume = 1.0; 249 mHardwareStatus = AUDIO_HW_IDLE; 250} 251 252AudioFlinger::~AudioFlinger() 253{ 254 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 // check if an effect chain with the same session ID is present on another 478 // output thread and move it here. 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 uint32_t sessions = t->hasAudioSession(*sessionId); 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 break; 486 } 487 } 488 } 489 lSessionId = *sessionId; 490 } else { 491 // if no audio session id is provided, create one here 492 lSessionId = nextUniqueId(); 493 if (sessionId != NULL) { 494 *sessionId = lSessionId; 495 } 496 } 497 ALOGV("createTrack() lSessionId: %d", lSessionId); 498 499 track = thread->createTrack_l(client, streamType, sampleRate, format, 500 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 501 502 // move effect chain to this output thread if an effect on same session was waiting 503 // for a track to be created 504 if (lStatus == NO_ERROR && effectThread != NULL) { 505 Mutex::Autolock _dl(thread->mLock); 506 Mutex::Autolock _sl(effectThread->mLock); 507 moveEffectChain_l(lSessionId, effectThread, thread, true); 508 } 509 510 // Look for sync events awaiting for a session to be used. 511 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 512 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 513 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 514 if (lStatus == NO_ERROR) { 515 track->setSyncEvent(mPendingSyncEvents[i]); 516 } else { 517 mPendingSyncEvents[i]->cancel(); 518 } 519 mPendingSyncEvents.removeAt(i); 520 i--; 521 } 522 } 523 } 524 } 525 if (lStatus == NO_ERROR) { 526 trackHandle = new TrackHandle(track); 527 } else { 528 // remove local strong reference to Client before deleting the Track so that the Client 529 // destructor is called by the TrackBase destructor with mLock held 530 client.clear(); 531 track.clear(); 532 } 533 534Exit: 535 if (status != NULL) { 536 *status = lStatus; 537 } 538 return trackHandle; 539} 540 541uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("sampleRate() unknown thread %d", output); 547 return 0; 548 } 549 return thread->sampleRate(); 550} 551 552int AudioFlinger::channelCount(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("channelCount() unknown thread %d", output); 558 return 0; 559 } 560 return thread->channelCount(); 561} 562 563audio_format_t AudioFlinger::format(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("format() unknown thread %d", output); 569 return AUDIO_FORMAT_INVALID; 570 } 571 return thread->format(); 572} 573 574size_t AudioFlinger::frameCount(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("frameCount() unknown thread %d", output); 580 return 0; 581 } 582 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 583 // should examine all callers and fix them to handle smaller counts 584 return thread->frameCount(); 585} 586 587uint32_t AudioFlinger::latency(audio_io_handle_t output) const 588{ 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGW("latency() unknown thread %d", output); 593 return 0; 594 } 595 return thread->latency(); 596} 597 598status_t AudioFlinger::setMasterVolume(float value) 599{ 600 status_t ret = initCheck(); 601 if (ret != NO_ERROR) { 602 return ret; 603 } 604 605 // check calling permissions 606 if (!settingsAllowed()) { 607 return PERMISSION_DENIED; 608 } 609 610 float swmv = value; 611 612 Mutex::Autolock _l(mLock); 613 614 // when hw supports master volume, don't scale in sw mixer 615 if (MVS_NONE != mMasterVolumeSupportLvl) { 616 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 617 AutoMutex lock(mHardwareLock); 618 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 619 620 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 621 if (NULL != dev->set_master_volume) { 622 dev->set_master_volume(dev, value); 623 } 624 mHardwareStatus = AUDIO_HW_IDLE; 625 } 626 627 swmv = 1.0; 628 } 629 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 Mutex::Autolock _l(mLock); 857 status_t final_result = NO_ERROR; 858 { 859 AutoMutex lock(mHardwareLock); 860 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 861 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 862 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 863 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 864 final_result = result ?: final_result; 865 } 866 mHardwareStatus = AUDIO_HW_IDLE; 867 } 868 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 869 AudioParameter param = AudioParameter(keyValuePairs); 870 String8 value; 871 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 Mutex::Autolock _l(mLock); 927 928 if (ioHandle == 0) { 929 String8 out_s8; 930 931 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 932 char *s; 933 { 934 AutoMutex lock(mHardwareLock); 935 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 936 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 937 s = dev->get_parameters(dev, keys.string()); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 } 940 out_s8 += String8(s ? s : ""); 941 free(s); 942 } 943 return out_s8; 944 } 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 struct audio_config config = { 967 sample_rate: sampleRate, 968 channel_mask: audio_channel_in_mask_from_count(channelCount), 969 format: format, 970 }; 971 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 return size; 974} 975 976unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 977{ 978 if (ioHandle == 0) { 979 return 0; 980 } 981 982 Mutex::Autolock _l(mLock); 983 984 RecordThread *recordThread = checkRecordThread_l(ioHandle); 985 if (recordThread != NULL) { 986 return recordThread->getInputFramesLost(); 987 } 988 return 0; 989} 990 991status_t AudioFlinger::setVoiceVolume(float value) 992{ 993 status_t ret = initCheck(); 994 if (ret != NO_ERROR) { 995 return ret; 996 } 997 998 // check calling permissions 999 if (!settingsAllowed()) { 1000 return PERMISSION_DENIED; 1001 } 1002 1003 AutoMutex lock(mHardwareLock); 1004 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1005 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1006 mHardwareStatus = AUDIO_HW_IDLE; 1007 1008 return ret; 1009} 1010 1011status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1012 audio_io_handle_t output) const 1013{ 1014 status_t status; 1015 1016 Mutex::Autolock _l(mLock); 1017 1018 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1019 if (playbackThread != NULL) { 1020 return playbackThread->getRenderPosition(halFrames, dspFrames); 1021 } 1022 1023 return BAD_VALUE; 1024} 1025 1026void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1027{ 1028 1029 Mutex::Autolock _l(mLock); 1030 1031 pid_t pid = IPCThreadState::self()->getCallingPid(); 1032 if (mNotificationClients.indexOfKey(pid) < 0) { 1033 sp<NotificationClient> notificationClient = new NotificationClient(this, 1034 client, 1035 pid); 1036 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1037 1038 mNotificationClients.add(pid, notificationClient); 1039 1040 sp<IBinder> binder = client->asBinder(); 1041 binder->linkToDeath(notificationClient); 1042 1043 // the config change is always sent from playback or record threads to avoid deadlock 1044 // with AudioSystem::gLock 1045 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1046 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1047 } 1048 1049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1050 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1051 } 1052 } 1053} 1054 1055void AudioFlinger::removeNotificationClient(pid_t pid) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 1059 mNotificationClients.removeItem(pid); 1060 1061 ALOGV("%d died, releasing its sessions", pid); 1062 size_t num = mAudioSessionRefs.size(); 1063 bool removed = false; 1064 for (size_t i = 0; i< num; ) { 1065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1066 ALOGV(" pid %d @ %d", ref->mPid, i); 1067 if (ref->mPid == pid) { 1068 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1069 mAudioSessionRefs.removeAt(i); 1070 delete ref; 1071 removed = true; 1072 num--; 1073 } else { 1074 i++; 1075 } 1076 } 1077 if (removed) { 1078 purgeStaleEffects_l(); 1079 } 1080} 1081 1082// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1083void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1084{ 1085 size_t size = mNotificationClients.size(); 1086 for (size_t i = 0; i < size; i++) { 1087 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1088 param2); 1089 } 1090} 1091 1092// removeClient_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::removeClient_l(pid_t pid) 1094{ 1095 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1096 mClients.removeItem(pid); 1097} 1098 1099 1100// ---------------------------------------------------------------------------- 1101 1102AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1103 uint32_t device, type_t type) 1104 : Thread(false), 1105 mType(type), 1106 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1107 // mChannelMask 1108 mChannelCount(0), 1109 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1110 mParamStatus(NO_ERROR), 1111 mStandby(false), mId(id), 1112 mDevice(device), 1113 mDeathRecipient(new PMDeathRecipient(this)) 1114{ 1115} 1116 1117AudioFlinger::ThreadBase::~ThreadBase() 1118{ 1119 mParamCond.broadcast(); 1120 // do not lock the mutex in destructor 1121 releaseWakeLock_l(); 1122 if (mPowerManager != 0) { 1123 sp<IBinder> binder = mPowerManager->asBinder(); 1124 binder->unlinkToDeath(mDeathRecipient); 1125 } 1126} 1127 1128void AudioFlinger::ThreadBase::exit() 1129{ 1130 ALOGV("ThreadBase::exit"); 1131 { 1132 // This lock prevents the following race in thread (uniprocessor for illustration): 1133 // if (!exitPending()) { 1134 // // context switch from here to exit() 1135 // // exit() calls requestExit(), what exitPending() observes 1136 // // exit() calls signal(), which is dropped since no waiters 1137 // // context switch back from exit() to here 1138 // mWaitWorkCV.wait(...); 1139 // // now thread is hung 1140 // } 1141 AutoMutex lock(mLock); 1142 requestExit(); 1143 mWaitWorkCV.signal(); 1144 } 1145 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1146 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1147 requestExitAndWait(); 1148} 1149 1150status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1151{ 1152 status_t status; 1153 1154 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1155 Mutex::Autolock _l(mLock); 1156 1157 mNewParameters.add(keyValuePairs); 1158 mWaitWorkCV.signal(); 1159 // wait condition with timeout in case the thread loop has exited 1160 // before the request could be processed 1161 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1162 status = mParamStatus; 1163 mWaitWorkCV.signal(); 1164 } else { 1165 status = TIMED_OUT; 1166 } 1167 return status; 1168} 1169 1170void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1171{ 1172 Mutex::Autolock _l(mLock); 1173 sendConfigEvent_l(event, param); 1174} 1175 1176// sendConfigEvent_l() must be called with ThreadBase::mLock held 1177void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1178{ 1179 ConfigEvent configEvent; 1180 configEvent.mEvent = event; 1181 configEvent.mParam = param; 1182 mConfigEvents.add(configEvent); 1183 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1184 mWaitWorkCV.signal(); 1185} 1186 1187void AudioFlinger::ThreadBase::processConfigEvents() 1188{ 1189 mLock.lock(); 1190 while (!mConfigEvents.isEmpty()) { 1191 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1192 ConfigEvent configEvent = mConfigEvents[0]; 1193 mConfigEvents.removeAt(0); 1194 // release mLock before locking AudioFlinger mLock: lock order is always 1195 // AudioFlinger then ThreadBase to avoid cross deadlock 1196 mLock.unlock(); 1197 mAudioFlinger->mLock.lock(); 1198 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1199 mAudioFlinger->mLock.unlock(); 1200 mLock.lock(); 1201 } 1202 mLock.unlock(); 1203} 1204 1205status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1206{ 1207 const size_t SIZE = 256; 1208 char buffer[SIZE]; 1209 String8 result; 1210 1211 bool locked = tryLock(mLock); 1212 if (!locked) { 1213 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1214 write(fd, buffer, strlen(buffer)); 1215 } 1216 1217 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1218 result.append(buffer); 1219 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1236 result.append(buffer); 1237 1238 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1239 result.append(buffer); 1240 result.append(" Index Command"); 1241 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1242 snprintf(buffer, SIZE, "\n %02d ", i); 1243 result.append(buffer); 1244 result.append(mNewParameters[i]); 1245 } 1246 1247 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, " Index event param\n"); 1250 result.append(buffer); 1251 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1252 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1253 result.append(buffer); 1254 } 1255 result.append("\n"); 1256 1257 write(fd, result.string(), result.size()); 1258 1259 if (locked) { 1260 mLock.unlock(); 1261 } 1262 return NO_ERROR; 1263} 1264 1265status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1266{ 1267 const size_t SIZE = 256; 1268 char buffer[SIZE]; 1269 String8 result; 1270 1271 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1272 write(fd, buffer, strlen(buffer)); 1273 1274 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1275 sp<EffectChain> chain = mEffectChains[i]; 1276 if (chain != 0) { 1277 chain->dump(fd, args); 1278 } 1279 } 1280 return NO_ERROR; 1281} 1282 1283void AudioFlinger::ThreadBase::acquireWakeLock() 1284{ 1285 Mutex::Autolock _l(mLock); 1286 acquireWakeLock_l(); 1287} 1288 1289void AudioFlinger::ThreadBase::acquireWakeLock_l() 1290{ 1291 if (mPowerManager == 0) { 1292 // use checkService() to avoid blocking if power service is not up yet 1293 sp<IBinder> binder = 1294 defaultServiceManager()->checkService(String16("power")); 1295 if (binder == 0) { 1296 ALOGW("Thread %s cannot connect to the power manager service", mName); 1297 } else { 1298 mPowerManager = interface_cast<IPowerManager>(binder); 1299 binder->linkToDeath(mDeathRecipient); 1300 } 1301 } 1302 if (mPowerManager != 0) { 1303 sp<IBinder> binder = new BBinder(); 1304 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1305 binder, 1306 String16(mName)); 1307 if (status == NO_ERROR) { 1308 mWakeLockToken = binder; 1309 } 1310 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::releaseWakeLock() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 releaseWakeLock_l(); 1318} 1319 1320void AudioFlinger::ThreadBase::releaseWakeLock_l() 1321{ 1322 if (mWakeLockToken != 0) { 1323 ALOGV("releaseWakeLock_l() %s", mName); 1324 if (mPowerManager != 0) { 1325 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1326 } 1327 mWakeLockToken.clear(); 1328 } 1329} 1330 1331void AudioFlinger::ThreadBase::clearPowerManager() 1332{ 1333 Mutex::Autolock _l(mLock); 1334 releaseWakeLock_l(); 1335 mPowerManager.clear(); 1336} 1337 1338void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1339{ 1340 sp<ThreadBase> thread = mThread.promote(); 1341 if (thread != 0) { 1342 thread->clearPowerManager(); 1343 } 1344 ALOGW("power manager service died !!!"); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 setEffectSuspended_l(type, suspend, sessionId); 1352} 1353 1354void AudioFlinger::ThreadBase::setEffectSuspended_l( 1355 const effect_uuid_t *type, bool suspend, int sessionId) 1356{ 1357 sp<EffectChain> chain = getEffectChain_l(sessionId); 1358 if (chain != 0) { 1359 if (type != NULL) { 1360 chain->setEffectSuspended_l(type, suspend); 1361 } else { 1362 chain->setEffectSuspendedAll_l(suspend); 1363 } 1364 } 1365 1366 updateSuspendedSessions_l(type, suspend, sessionId); 1367} 1368 1369void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1370{ 1371 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1372 if (index < 0) { 1373 return; 1374 } 1375 1376 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1377 mSuspendedSessions.editValueAt(index); 1378 1379 for (size_t i = 0; i < sessionEffects.size(); i++) { 1380 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1381 for (int j = 0; j < desc->mRefCount; j++) { 1382 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1383 chain->setEffectSuspendedAll_l(true); 1384 } else { 1385 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1386 desc->mType.timeLow); 1387 chain->setEffectSuspended_l(&desc->mType, true); 1388 } 1389 } 1390 } 1391} 1392 1393void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1394 bool suspend, 1395 int sessionId) 1396{ 1397 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1398 1399 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1400 1401 if (suspend) { 1402 if (index >= 0) { 1403 sessionEffects = mSuspendedSessions.editValueAt(index); 1404 } else { 1405 mSuspendedSessions.add(sessionId, sessionEffects); 1406 } 1407 } else { 1408 if (index < 0) { 1409 return; 1410 } 1411 sessionEffects = mSuspendedSessions.editValueAt(index); 1412 } 1413 1414 1415 int key = EffectChain::kKeyForSuspendAll; 1416 if (type != NULL) { 1417 key = type->timeLow; 1418 } 1419 index = sessionEffects.indexOfKey(key); 1420 1421 sp<SuspendedSessionDesc> desc; 1422 if (suspend) { 1423 if (index >= 0) { 1424 desc = sessionEffects.valueAt(index); 1425 } else { 1426 desc = new SuspendedSessionDesc(); 1427 if (type != NULL) { 1428 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1429 } 1430 sessionEffects.add(key, desc); 1431 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1432 } 1433 desc->mRefCount++; 1434 } else { 1435 if (index < 0) { 1436 return; 1437 } 1438 desc = sessionEffects.valueAt(index); 1439 if (--desc->mRefCount == 0) { 1440 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1441 sessionEffects.removeItemsAt(index); 1442 if (sessionEffects.isEmpty()) { 1443 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1444 sessionId); 1445 mSuspendedSessions.removeItem(sessionId); 1446 } 1447 } 1448 } 1449 if (!sessionEffects.isEmpty()) { 1450 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1451 } 1452} 1453 1454void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1455 bool enabled, 1456 int sessionId) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1460} 1461 1462void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1463 bool enabled, 1464 int sessionId) 1465{ 1466 if (mType != RECORD) { 1467 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1468 // another session. This gives the priority to well behaved effect control panels 1469 // and applications not using global effects. 1470 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1471 // global effects 1472 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1473 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1474 } 1475 } 1476 1477 sp<EffectChain> chain = getEffectChain_l(sessionId); 1478 if (chain != 0) { 1479 chain->checkSuspendOnEffectEnabled(effect, enabled); 1480 } 1481} 1482 1483// ---------------------------------------------------------------------------- 1484 1485AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1486 AudioStreamOut* output, 1487 audio_io_handle_t id, 1488 uint32_t device, 1489 type_t type) 1490 : ThreadBase(audioFlinger, id, device, type), 1491 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1492 // Assumes constructor is called by AudioFlinger with it's mLock held, 1493 // but it would be safer to explicitly pass initial masterMute as parameter 1494 mMasterMute(audioFlinger->masterMute_l()), 1495 // mStreamTypes[] initialized in constructor body 1496 mOutput(output), 1497 // Assumes constructor is called by AudioFlinger with it's mLock held, 1498 // but it would be safer to explicitly pass initial masterVolume as parameter 1499 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1500 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1501 mMixerStatus(MIXER_IDLE), 1502 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1503 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1504 // index 0 is reserved for normal mixer's submix 1505 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1506{ 1507 snprintf(mName, kNameLength, "AudioOut_%X", id); 1508 1509 readOutputParameters(); 1510 1511 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1512 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1513 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1514 stream = (audio_stream_type_t) (stream + 1)) { 1515 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1516 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1517 } 1518 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1519 // because mAudioFlinger doesn't have one to copy from 1520} 1521 1522AudioFlinger::PlaybackThread::~PlaybackThread() 1523{ 1524 delete [] mMixBuffer; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1528{ 1529 dumpInternals(fd, args); 1530 dumpTracks(fd, args); 1531 dumpEffectChains(fd, args); 1532 return NO_ERROR; 1533} 1534 1535status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1536{ 1537 const size_t SIZE = 256; 1538 char buffer[SIZE]; 1539 String8 result; 1540 1541 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1542 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1543 const stream_type_t *st = &mStreamTypes[i]; 1544 if (i > 0) { 1545 result.appendFormat(", "); 1546 } 1547 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1548 if (st->mute) { 1549 result.append("M"); 1550 } 1551 } 1552 result.append("\n"); 1553 write(fd, result.string(), result.length()); 1554 result.clear(); 1555 1556 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1557 result.append(buffer); 1558 Track::appendDumpHeader(result); 1559 for (size_t i = 0; i < mTracks.size(); ++i) { 1560 sp<Track> track = mTracks[i]; 1561 if (track != 0) { 1562 track->dump(buffer, SIZE); 1563 result.append(buffer); 1564 } 1565 } 1566 1567 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1571 sp<Track> track = mActiveTracks[i].promote(); 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 write(fd, result.string(), result.size()); 1578 1579 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1580 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1581 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1582 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1583 1584 return NO_ERROR; 1585} 1586 1587status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1588{ 1589 const size_t SIZE = 256; 1590 char buffer[SIZE]; 1591 String8 result; 1592 1593 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1606 result.append(buffer); 1607 write(fd, result.string(), result.size()); 1608 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1609 1610 dumpBase(fd, args); 1611 1612 return NO_ERROR; 1613} 1614 1615// Thread virtuals 1616status_t AudioFlinger::PlaybackThread::readyToRun() 1617{ 1618 status_t status = initCheck(); 1619 if (status == NO_ERROR) { 1620 ALOGI("AudioFlinger's thread %p ready to run", this); 1621 } else { 1622 ALOGE("No working audio driver found."); 1623 } 1624 return status; 1625} 1626 1627void AudioFlinger::PlaybackThread::onFirstRef() 1628{ 1629 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1630} 1631 1632// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1633sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1634 const sp<AudioFlinger::Client>& client, 1635 audio_stream_type_t streamType, 1636 uint32_t sampleRate, 1637 audio_format_t format, 1638 uint32_t channelMask, 1639 int frameCount, 1640 const sp<IMemory>& sharedBuffer, 1641 int sessionId, 1642 IAudioFlinger::track_flags_t flags, 1643 pid_t tid, 1644 status_t *status) 1645{ 1646 sp<Track> track; 1647 status_t lStatus; 1648 1649 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1650 1651 // client expresses a preference for FAST, but we get the final say 1652 if (flags & IAudioFlinger::TRACK_FAST) { 1653 if ( 1654 // not timed 1655 (!isTimed) && 1656 // either of these use cases: 1657 ( 1658 // use case 1: shared buffer with any frame count 1659 ( 1660 (sharedBuffer != 0) 1661 ) || 1662 // use case 2: callback handler and frame count is default or at least as large as HAL 1663 ( 1664 (tid != -1) && 1665 ((frameCount == 0) || 1666 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1667 ) 1668 ) && 1669 // PCM data 1670 audio_is_linear_pcm(format) && 1671 // mono or stereo 1672 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1673 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1674#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1675 // hardware sample rate 1676 (sampleRate == mSampleRate) && 1677#endif 1678 // normal mixer has an associated fast mixer 1679 hasFastMixer() && 1680 // there are sufficient fast track slots available 1681 (mFastTrackAvailMask != 0) 1682 // FIXME test that MixerThread for this fast track has a capable output HAL 1683 // FIXME add a permission test also? 1684 ) { 1685 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1686 if (frameCount == 0) { 1687 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1688 } 1689 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1690 frameCount, mFrameCount); 1691 } else { 1692 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1693 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1694 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1695 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1696 audio_is_linear_pcm(format), 1697 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1698 flags &= ~IAudioFlinger::TRACK_FAST; 1699 // For compatibility with AudioTrack calculation, buffer depth is forced 1700 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1701 // This is probably too conservative, but legacy application code may depend on it. 1702 // If you change this calculation, also review the start threshold which is related. 1703 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1704 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1705 if (minBufCount < 2) { 1706 minBufCount = 2; 1707 } 1708 int minFrameCount = mNormalFrameCount * minBufCount; 1709 if (frameCount < minFrameCount) { 1710 frameCount = minFrameCount; 1711 } 1712 } 1713 } 1714 1715 if (mType == DIRECT) { 1716 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1717 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1718 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1719 "for output %p with format %d", 1720 sampleRate, format, channelMask, mOutput, mFormat); 1721 lStatus = BAD_VALUE; 1722 goto Exit; 1723 } 1724 } 1725 } else { 1726 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1727 if (sampleRate > mSampleRate*2) { 1728 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1729 lStatus = BAD_VALUE; 1730 goto Exit; 1731 } 1732 } 1733 1734 lStatus = initCheck(); 1735 if (lStatus != NO_ERROR) { 1736 ALOGE("Audio driver not initialized."); 1737 goto Exit; 1738 } 1739 1740 { // scope for mLock 1741 Mutex::Autolock _l(mLock); 1742 1743 // all tracks in same audio session must share the same routing strategy otherwise 1744 // conflicts will happen when tracks are moved from one output to another by audio policy 1745 // manager 1746 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1747 for (size_t i = 0; i < mTracks.size(); ++i) { 1748 sp<Track> t = mTracks[i]; 1749 if (t != 0 && !t->isOutputTrack()) { 1750 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1751 if (sessionId == t->sessionId() && strategy != actual) { 1752 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1753 strategy, actual); 1754 lStatus = BAD_VALUE; 1755 goto Exit; 1756 } 1757 } 1758 } 1759 1760 if (!isTimed) { 1761 track = new Track(this, client, streamType, sampleRate, format, 1762 channelMask, frameCount, sharedBuffer, sessionId, flags); 1763 } else { 1764 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1765 channelMask, frameCount, sharedBuffer, sessionId); 1766 } 1767 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1768 lStatus = NO_MEMORY; 1769 goto Exit; 1770 } 1771 mTracks.add(track); 1772 1773 sp<EffectChain> chain = getEffectChain_l(sessionId); 1774 if (chain != 0) { 1775 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1776 track->setMainBuffer(chain->inBuffer()); 1777 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1778 chain->incTrackCnt(); 1779 } 1780 } 1781 1782#ifdef HAVE_REQUEST_PRIORITY 1783 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1784 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1785 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1786 // so ask activity manager to do this on our behalf 1787 int err = requestPriority(callingPid, tid, 1); 1788 if (err != 0) { 1789 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1790 1, callingPid, tid, err); 1791 } 1792 } 1793#endif 1794 1795 lStatus = NO_ERROR; 1796 1797Exit: 1798 if (status) { 1799 *status = lStatus; 1800 } 1801 return track; 1802} 1803 1804uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1805{ 1806 if (mFastMixer != NULL) { 1807 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1808 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1809 } 1810 return latency; 1811} 1812 1813uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1814{ 1815 return latency; 1816} 1817 1818uint32_t AudioFlinger::PlaybackThread::latency() const 1819{ 1820 Mutex::Autolock _l(mLock); 1821 if (initCheck() == NO_ERROR) { 1822 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1823 } else { 1824 return 0; 1825 } 1826} 1827 1828void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1829{ 1830 Mutex::Autolock _l(mLock); 1831 mMasterVolume = value; 1832} 1833 1834void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1835{ 1836 Mutex::Autolock _l(mLock); 1837 setMasterMute_l(muted); 1838} 1839 1840void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1841{ 1842 Mutex::Autolock _l(mLock); 1843 mStreamTypes[stream].volume = value; 1844} 1845 1846void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1847{ 1848 Mutex::Autolock _l(mLock); 1849 mStreamTypes[stream].mute = muted; 1850} 1851 1852float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1853{ 1854 Mutex::Autolock _l(mLock); 1855 return mStreamTypes[stream].volume; 1856} 1857 1858// addTrack_l() must be called with ThreadBase::mLock held 1859status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1860{ 1861 status_t status = ALREADY_EXISTS; 1862 1863 // set retry count for buffer fill 1864 track->mRetryCount = kMaxTrackStartupRetries; 1865 if (mActiveTracks.indexOf(track) < 0) { 1866 // the track is newly added, make sure it fills up all its 1867 // buffers before playing. This is to ensure the client will 1868 // effectively get the latency it requested. 1869 track->mFillingUpStatus = Track::FS_FILLING; 1870 track->mResetDone = false; 1871 track->mPresentationCompleteFrames = 0; 1872 mActiveTracks.add(track); 1873 if (track->mainBuffer() != mMixBuffer) { 1874 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1875 if (chain != 0) { 1876 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1877 chain->incActiveTrackCnt(); 1878 } 1879 } 1880 1881 status = NO_ERROR; 1882 } 1883 1884 ALOGV("mWaitWorkCV.broadcast"); 1885 mWaitWorkCV.broadcast(); 1886 1887 return status; 1888} 1889 1890// destroyTrack_l() must be called with ThreadBase::mLock held 1891void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1892{ 1893 track->mState = TrackBase::TERMINATED; 1894 // active tracks are removed by threadLoop() 1895 if (mActiveTracks.indexOf(track) < 0) { 1896 removeTrack_l(track); 1897 } 1898} 1899 1900void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1901{ 1902 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1903 mTracks.remove(track); 1904 deleteTrackName_l(track->name()); 1905 // redundant as track is about to be destroyed, for dumpsys only 1906 track->mName = -1; 1907 if (track->isFastTrack()) { 1908 int index = track->mFastIndex; 1909 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1910 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1911 mFastTrackAvailMask |= 1 << index; 1912 // redundant as track is about to be destroyed, for dumpsys only 1913 track->mFastIndex = -1; 1914 } 1915 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1916 if (chain != 0) { 1917 chain->decTrackCnt(); 1918 } 1919} 1920 1921String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1922{ 1923 String8 out_s8 = String8(""); 1924 char *s; 1925 1926 Mutex::Autolock _l(mLock); 1927 if (initCheck() != NO_ERROR) { 1928 return out_s8; 1929 } 1930 1931 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1932 out_s8 = String8(s); 1933 free(s); 1934 return out_s8; 1935} 1936 1937// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1938void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1939 AudioSystem::OutputDescriptor desc; 1940 void *param2 = NULL; 1941 1942 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1943 1944 switch (event) { 1945 case AudioSystem::OUTPUT_OPENED: 1946 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1947 desc.channels = mChannelMask; 1948 desc.samplingRate = mSampleRate; 1949 desc.format = mFormat; 1950 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1951 desc.latency = latency(); 1952 param2 = &desc; 1953 break; 1954 1955 case AudioSystem::STREAM_CONFIG_CHANGED: 1956 param2 = ¶m; 1957 case AudioSystem::OUTPUT_CLOSED: 1958 default: 1959 break; 1960 } 1961 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1962} 1963 1964void AudioFlinger::PlaybackThread::readOutputParameters() 1965{ 1966 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1967 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1968 mChannelCount = (uint16_t)popcount(mChannelMask); 1969 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1970 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1971 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1972 if (mFrameCount & 15) { 1973 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1974 mFrameCount); 1975 } 1976 1977 // Calculate size of normal mix buffer relative to the HAL output buffer size 1978 double multiplier = 1.0; 1979 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1980 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1981 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1982 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1983 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1984 maxNormalFrameCount = maxNormalFrameCount & ~15; 1985 if (maxNormalFrameCount < minNormalFrameCount) { 1986 maxNormalFrameCount = minNormalFrameCount; 1987 } 1988 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1989 if (multiplier <= 1.0) { 1990 multiplier = 1.0; 1991 } else if (multiplier <= 2.0) { 1992 if (2 * mFrameCount <= maxNormalFrameCount) { 1993 multiplier = 2.0; 1994 } else { 1995 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1996 } 1997 } else { 1998 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1999 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2000 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2001 // FIXME this rounding up should not be done if no HAL SRC 2002 uint32_t truncMult = (uint32_t) multiplier; 2003 if ((truncMult & 1)) { 2004 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2005 ++truncMult; 2006 } 2007 } 2008 multiplier = (double) truncMult; 2009 } 2010 } 2011 mNormalFrameCount = multiplier * mFrameCount; 2012 // round up to nearest 16 frames to satisfy AudioMixer 2013 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2014 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2015 2016 // FIXME - Current mixer implementation only supports stereo output: Always 2017 // Allocate a stereo buffer even if HW output is mono. 2018 delete[] mMixBuffer; 2019 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2020 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2021 2022 // force reconfiguration of effect chains and engines to take new buffer size and audio 2023 // parameters into account 2024 // Note that mLock is not held when readOutputParameters() is called from the constructor 2025 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2026 // matter. 2027 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2028 Vector< sp<EffectChain> > effectChains = mEffectChains; 2029 for (size_t i = 0; i < effectChains.size(); i ++) { 2030 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2031 } 2032} 2033 2034 2035status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2036{ 2037 if (halFrames == NULL || dspFrames == NULL) { 2038 return BAD_VALUE; 2039 } 2040 Mutex::Autolock _l(mLock); 2041 if (initCheck() != NO_ERROR) { 2042 return INVALID_OPERATION; 2043 } 2044 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2045 2046 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2047} 2048 2049uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2050{ 2051 Mutex::Autolock _l(mLock); 2052 uint32_t result = 0; 2053 if (getEffectChain_l(sessionId) != 0) { 2054 result = EFFECT_SESSION; 2055 } 2056 2057 for (size_t i = 0; i < mTracks.size(); ++i) { 2058 sp<Track> track = mTracks[i]; 2059 if (sessionId == track->sessionId() && 2060 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2061 result |= TRACK_SESSION; 2062 break; 2063 } 2064 } 2065 2066 return result; 2067} 2068 2069uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2070{ 2071 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2072 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2073 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2074 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2075 } 2076 for (size_t i = 0; i < mTracks.size(); i++) { 2077 sp<Track> track = mTracks[i]; 2078 if (sessionId == track->sessionId() && 2079 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2080 return AudioSystem::getStrategyForStream(track->streamType()); 2081 } 2082 } 2083 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2084} 2085 2086 2087AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2088{ 2089 Mutex::Autolock _l(mLock); 2090 return mOutput; 2091} 2092 2093AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2094{ 2095 Mutex::Autolock _l(mLock); 2096 AudioStreamOut *output = mOutput; 2097 mOutput = NULL; 2098 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2099 // must push a NULL and wait for ack 2100 mOutputSink.clear(); 2101 mPipeSink.clear(); 2102 mNormalSink.clear(); 2103 return output; 2104} 2105 2106// this method must always be called either with ThreadBase mLock held or inside the thread loop 2107audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2108{ 2109 if (mOutput == NULL) { 2110 return NULL; 2111 } 2112 return &mOutput->stream->common; 2113} 2114 2115uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2116{ 2117 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2118} 2119 2120status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2121{ 2122 if (!isValidSyncEvent(event)) { 2123 return BAD_VALUE; 2124 } 2125 2126 Mutex::Autolock _l(mLock); 2127 2128 for (size_t i = 0; i < mTracks.size(); ++i) { 2129 sp<Track> track = mTracks[i]; 2130 if (event->triggerSession() == track->sessionId()) { 2131 track->setSyncEvent(event); 2132 return NO_ERROR; 2133 } 2134 } 2135 2136 return NAME_NOT_FOUND; 2137} 2138 2139bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2140{ 2141 switch (event->type()) { 2142 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2143 return true; 2144 default: 2145 break; 2146 } 2147 return false; 2148} 2149 2150void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2151{ 2152 size_t count = tracksToRemove.size(); 2153 if (CC_UNLIKELY(count)) { 2154 for (size_t i = 0 ; i < count ; i++) { 2155 const sp<Track>& track = tracksToRemove.itemAt(i); 2156 if ((track->sharedBuffer() != 0) && 2157 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2158 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2159 } 2160 } 2161 } 2162 2163} 2164 2165// ---------------------------------------------------------------------------- 2166 2167AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2168 audio_io_handle_t id, uint32_t device, type_t type) 2169 : PlaybackThread(audioFlinger, output, id, device, type), 2170 // mAudioMixer below 2171#ifdef SOAKER 2172 mSoaker(NULL), 2173#endif 2174 // mFastMixer below 2175 mFastMixerFutex(0) 2176 // mOutputSink below 2177 // mPipeSink below 2178 // mNormalSink below 2179{ 2180 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2181 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2182 "mFrameCount=%d, mNormalFrameCount=%d", 2183 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2184 mNormalFrameCount); 2185 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2186 2187 // FIXME - Current mixer implementation only supports stereo output 2188 if (mChannelCount == 1) { 2189 ALOGE("Invalid audio hardware channel count"); 2190 } 2191 2192 // create an NBAIO sink for the HAL output stream, and negotiate 2193 mOutputSink = new AudioStreamOutSink(output->stream); 2194 size_t numCounterOffers = 0; 2195 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2196 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2197 ALOG_ASSERT(index == 0); 2198 2199 // initialize fast mixer depending on configuration 2200 bool initFastMixer; 2201 switch (kUseFastMixer) { 2202 case FastMixer_Never: 2203 initFastMixer = false; 2204 break; 2205 case FastMixer_Always: 2206 initFastMixer = true; 2207 break; 2208 case FastMixer_Static: 2209 case FastMixer_Dynamic: 2210 initFastMixer = mFrameCount < mNormalFrameCount; 2211 break; 2212 } 2213 if (initFastMixer) { 2214 2215 // create a MonoPipe to connect our submix to FastMixer 2216 NBAIO_Format format = mOutputSink->format(); 2217 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2218 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2219 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2220 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2221 const NBAIO_Format offers[1] = {format}; 2222 size_t numCounterOffers = 0; 2223 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2224 ALOG_ASSERT(index == 0); 2225 mPipeSink = monoPipe; 2226 2227#ifdef TEE_SINK_FRAMES 2228 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2229 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2230 numCounterOffers = 0; 2231 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2232 ALOG_ASSERT(index == 0); 2233 mTeeSink = teeSink; 2234 PipeReader *teeSource = new PipeReader(*teeSink); 2235 numCounterOffers = 0; 2236 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2237 ALOG_ASSERT(index == 0); 2238 mTeeSource = teeSource; 2239#endif 2240 2241#ifdef SOAKER 2242 // create a soaker as workaround for governor issues 2243 mSoaker = new Soaker(); 2244 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2245 mSoaker->run("Soaker", PRIORITY_LOWEST); 2246#endif 2247 2248 // create fast mixer and configure it initially with just one fast track for our submix 2249 mFastMixer = new FastMixer(); 2250 FastMixerStateQueue *sq = mFastMixer->sq(); 2251#ifdef STATE_QUEUE_DUMP 2252 sq->setObserverDump(&mStateQueueObserverDump); 2253 sq->setMutatorDump(&mStateQueueMutatorDump); 2254#endif 2255 FastMixerState *state = sq->begin(); 2256 FastTrack *fastTrack = &state->mFastTracks[0]; 2257 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2258 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2259 fastTrack->mVolumeProvider = NULL; 2260 fastTrack->mGeneration++; 2261 state->mFastTracksGen++; 2262 state->mTrackMask = 1; 2263 // fast mixer will use the HAL output sink 2264 state->mOutputSink = mOutputSink.get(); 2265 state->mOutputSinkGen++; 2266 state->mFrameCount = mFrameCount; 2267 state->mCommand = FastMixerState::COLD_IDLE; 2268 // already done in constructor initialization list 2269 //mFastMixerFutex = 0; 2270 state->mColdFutexAddr = &mFastMixerFutex; 2271 state->mColdGen++; 2272 state->mDumpState = &mFastMixerDumpState; 2273 state->mTeeSink = mTeeSink.get(); 2274 sq->end(); 2275 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2276 2277 // start the fast mixer 2278 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2279#ifdef HAVE_REQUEST_PRIORITY 2280 pid_t tid = mFastMixer->getTid(); 2281 int err = requestPriority(getpid_cached, tid, 2); 2282 if (err != 0) { 2283 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2284 2, getpid_cached, tid, err); 2285 } 2286#endif 2287 2288 } else { 2289 mFastMixer = NULL; 2290 } 2291 2292 switch (kUseFastMixer) { 2293 case FastMixer_Never: 2294 case FastMixer_Dynamic: 2295 mNormalSink = mOutputSink; 2296 break; 2297 case FastMixer_Always: 2298 mNormalSink = mPipeSink; 2299 break; 2300 case FastMixer_Static: 2301 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2302 break; 2303 } 2304} 2305 2306AudioFlinger::MixerThread::~MixerThread() 2307{ 2308 if (mFastMixer != NULL) { 2309 FastMixerStateQueue *sq = mFastMixer->sq(); 2310 FastMixerState *state = sq->begin(); 2311 if (state->mCommand == FastMixerState::COLD_IDLE) { 2312 int32_t old = android_atomic_inc(&mFastMixerFutex); 2313 if (old == -1) { 2314 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2315 } 2316 } 2317 state->mCommand = FastMixerState::EXIT; 2318 sq->end(); 2319 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2320 mFastMixer->join(); 2321 // Though the fast mixer thread has exited, it's state queue is still valid. 2322 // We'll use that extract the final state which contains one remaining fast track 2323 // corresponding to our sub-mix. 2324 state = sq->begin(); 2325 ALOG_ASSERT(state->mTrackMask == 1); 2326 FastTrack *fastTrack = &state->mFastTracks[0]; 2327 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2328 delete fastTrack->mBufferProvider; 2329 sq->end(false /*didModify*/); 2330 delete mFastMixer; 2331#ifdef SOAKER 2332 if (mSoaker != NULL) { 2333 mSoaker->requestExitAndWait(); 2334 } 2335 delete mSoaker; 2336#endif 2337 } 2338 delete mAudioMixer; 2339} 2340 2341class CpuStats { 2342public: 2343 CpuStats(); 2344 void sample(const String8 &title); 2345#ifdef DEBUG_CPU_USAGE 2346private: 2347 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2348 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2349 2350 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2351 2352 int mCpuNum; // thread's current CPU number 2353 int mCpukHz; // frequency of thread's current CPU in kHz 2354#endif 2355}; 2356 2357CpuStats::CpuStats() 2358#ifdef DEBUG_CPU_USAGE 2359 : mCpuNum(-1), mCpukHz(-1) 2360#endif 2361{ 2362} 2363 2364void CpuStats::sample(const String8 &title) { 2365#ifdef DEBUG_CPU_USAGE 2366 // get current thread's delta CPU time in wall clock ns 2367 double wcNs; 2368 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2369 2370 // record sample for wall clock statistics 2371 if (valid) { 2372 mWcStats.sample(wcNs); 2373 } 2374 2375 // get the current CPU number 2376 int cpuNum = sched_getcpu(); 2377 2378 // get the current CPU frequency in kHz 2379 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2380 2381 // check if either CPU number or frequency changed 2382 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2383 mCpuNum = cpuNum; 2384 mCpukHz = cpukHz; 2385 // ignore sample for purposes of cycles 2386 valid = false; 2387 } 2388 2389 // if no change in CPU number or frequency, then record sample for cycle statistics 2390 if (valid && mCpukHz > 0) { 2391 double cycles = wcNs * cpukHz * 0.000001; 2392 mHzStats.sample(cycles); 2393 } 2394 2395 unsigned n = mWcStats.n(); 2396 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2397 if ((n & 127) == 1) { 2398 long long elapsed = mCpuUsage.elapsed(); 2399 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2400 double perLoop = elapsed / (double) n; 2401 double perLoop100 = perLoop * 0.01; 2402 double perLoop1k = perLoop * 0.001; 2403 double mean = mWcStats.mean(); 2404 double stddev = mWcStats.stddev(); 2405 double minimum = mWcStats.minimum(); 2406 double maximum = mWcStats.maximum(); 2407 double meanCycles = mHzStats.mean(); 2408 double stddevCycles = mHzStats.stddev(); 2409 double minCycles = mHzStats.minimum(); 2410 double maxCycles = mHzStats.maximum(); 2411 mCpuUsage.resetElapsed(); 2412 mWcStats.reset(); 2413 mHzStats.reset(); 2414 ALOGD("CPU usage for %s over past %.1f secs\n" 2415 " (%u mixer loops at %.1f mean ms per loop):\n" 2416 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2417 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2418 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2419 title.string(), 2420 elapsed * .000000001, n, perLoop * .000001, 2421 mean * .001, 2422 stddev * .001, 2423 minimum * .001, 2424 maximum * .001, 2425 mean / perLoop100, 2426 stddev / perLoop100, 2427 minimum / perLoop100, 2428 maximum / perLoop100, 2429 meanCycles / perLoop1k, 2430 stddevCycles / perLoop1k, 2431 minCycles / perLoop1k, 2432 maxCycles / perLoop1k); 2433 2434 } 2435 } 2436#endif 2437}; 2438 2439void AudioFlinger::PlaybackThread::checkSilentMode_l() 2440{ 2441 if (!mMasterMute) { 2442 char value[PROPERTY_VALUE_MAX]; 2443 if (property_get("ro.audio.silent", value, "0") > 0) { 2444 char *endptr; 2445 unsigned long ul = strtoul(value, &endptr, 0); 2446 if (*endptr == '\0' && ul != 0) { 2447 ALOGD("Silence is golden"); 2448 // The setprop command will not allow a property to be changed after 2449 // the first time it is set, so we don't have to worry about un-muting. 2450 setMasterMute_l(true); 2451 } 2452 } 2453 } 2454} 2455 2456bool AudioFlinger::PlaybackThread::threadLoop() 2457{ 2458 Vector< sp<Track> > tracksToRemove; 2459 2460 standbyTime = systemTime(); 2461 2462 // MIXER 2463 nsecs_t lastWarning = 0; 2464if (mType == MIXER) { 2465 longStandbyExit = false; 2466} 2467 2468 // DUPLICATING 2469 // FIXME could this be made local to while loop? 2470 writeFrames = 0; 2471 2472 cacheParameters_l(); 2473 sleepTime = idleSleepTime; 2474 2475if (mType == MIXER) { 2476 sleepTimeShift = 0; 2477} 2478 2479 CpuStats cpuStats; 2480 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2481 2482 acquireWakeLock(); 2483 2484 while (!exitPending()) 2485 { 2486 cpuStats.sample(myName); 2487 2488 Vector< sp<EffectChain> > effectChains; 2489 2490 processConfigEvents(); 2491 2492 { // scope for mLock 2493 2494 Mutex::Autolock _l(mLock); 2495 2496 if (checkForNewParameters_l()) { 2497 cacheParameters_l(); 2498 } 2499 2500 saveOutputTracks(); 2501 2502 // put audio hardware into standby after short delay 2503 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2504 mSuspended > 0)) { 2505 if (!mStandby) { 2506 2507 threadLoop_standby(); 2508 2509 mStandby = true; 2510 mBytesWritten = 0; 2511 } 2512 2513 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2514 // we're about to wait, flush the binder command buffer 2515 IPCThreadState::self()->flushCommands(); 2516 2517 clearOutputTracks(); 2518 2519 if (exitPending()) break; 2520 2521 releaseWakeLock_l(); 2522 // wait until we have something to do... 2523 ALOGV("%s going to sleep", myName.string()); 2524 mWaitWorkCV.wait(mLock); 2525 ALOGV("%s waking up", myName.string()); 2526 acquireWakeLock_l(); 2527 2528 mMixerStatus = MIXER_IDLE; 2529 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2530 2531 checkSilentMode_l(); 2532 2533 standbyTime = systemTime() + standbyDelay; 2534 sleepTime = idleSleepTime; 2535 if (mType == MIXER) { 2536 sleepTimeShift = 0; 2537 } 2538 2539 continue; 2540 } 2541 } 2542 2543 // mMixerStatusIgnoringFastTracks is also updated internally 2544 mMixerStatus = prepareTracks_l(&tracksToRemove); 2545 2546 // prevent any changes in effect chain list and in each effect chain 2547 // during mixing and effect process as the audio buffers could be deleted 2548 // or modified if an effect is created or deleted 2549 lockEffectChains_l(effectChains); 2550 } 2551 2552 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2553 threadLoop_mix(); 2554 } else { 2555 threadLoop_sleepTime(); 2556 } 2557 2558 if (mSuspended > 0) { 2559 sleepTime = suspendSleepTimeUs(); 2560 } 2561 2562 // only process effects if we're going to write 2563 if (sleepTime == 0) { 2564 for (size_t i = 0; i < effectChains.size(); i ++) { 2565 effectChains[i]->process_l(); 2566 } 2567 } 2568 2569 // enable changes in effect chain 2570 unlockEffectChains(effectChains); 2571 2572 // sleepTime == 0 means we must write to audio hardware 2573 if (sleepTime == 0) { 2574 2575 threadLoop_write(); 2576 2577if (mType == MIXER) { 2578 // write blocked detection 2579 nsecs_t now = systemTime(); 2580 nsecs_t delta = now - mLastWriteTime; 2581 if (!mStandby && delta > maxPeriod) { 2582 mNumDelayedWrites++; 2583 if ((now - lastWarning) > kWarningThrottleNs) { 2584#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2585 ScopedTrace st(ATRACE_TAG, "underrun"); 2586#endif 2587 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2588 ns2ms(delta), mNumDelayedWrites, this); 2589 lastWarning = now; 2590 } 2591 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2592 // a different threshold. Or completely removed for what it is worth anyway... 2593 if (mStandby) { 2594 longStandbyExit = true; 2595 } 2596 } 2597} 2598 2599 mStandby = false; 2600 } else { 2601 usleep(sleepTime); 2602 } 2603 2604 // Finally let go of removed track(s), without the lock held 2605 // since we can't guarantee the destructors won't acquire that 2606 // same lock. This will also mutate and push a new fast mixer state. 2607 threadLoop_removeTracks(tracksToRemove); 2608 tracksToRemove.clear(); 2609 2610 // FIXME I don't understand the need for this here; 2611 // it was in the original code but maybe the 2612 // assignment in saveOutputTracks() makes this unnecessary? 2613 clearOutputTracks(); 2614 2615 // Effect chains will be actually deleted here if they were removed from 2616 // mEffectChains list during mixing or effects processing 2617 effectChains.clear(); 2618 2619 // FIXME Note that the above .clear() is no longer necessary since effectChains 2620 // is now local to this block, but will keep it for now (at least until merge done). 2621 } 2622 2623if (mType == MIXER || mType == DIRECT) { 2624 // put output stream into standby mode 2625 if (!mStandby) { 2626 mOutput->stream->common.standby(&mOutput->stream->common); 2627 } 2628} 2629if (mType == DUPLICATING) { 2630 // for DuplicatingThread, standby mode is handled by the outputTracks 2631} 2632 2633 releaseWakeLock(); 2634 2635 ALOGV("Thread %p type %d exiting", this, mType); 2636 return false; 2637} 2638 2639void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2640{ 2641 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2642} 2643 2644void AudioFlinger::MixerThread::threadLoop_write() 2645{ 2646 // FIXME we should only do one push per cycle; confirm this is true 2647 // Start the fast mixer if it's not already running 2648 if (mFastMixer != NULL) { 2649 FastMixerStateQueue *sq = mFastMixer->sq(); 2650 FastMixerState *state = sq->begin(); 2651 if (state->mCommand != FastMixerState::MIX_WRITE && 2652 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2653 if (state->mCommand == FastMixerState::COLD_IDLE) { 2654 int32_t old = android_atomic_inc(&mFastMixerFutex); 2655 if (old == -1) { 2656 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2657 } 2658 } 2659 state->mCommand = FastMixerState::MIX_WRITE; 2660 sq->end(); 2661 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2662 if (kUseFastMixer == FastMixer_Dynamic) { 2663 mNormalSink = mPipeSink; 2664 } 2665 } else { 2666 sq->end(false /*didModify*/); 2667 } 2668 } 2669 PlaybackThread::threadLoop_write(); 2670} 2671 2672// shared by MIXER and DIRECT, overridden by DUPLICATING 2673void AudioFlinger::PlaybackThread::threadLoop_write() 2674{ 2675 // FIXME rewrite to reduce number of system calls 2676 mLastWriteTime = systemTime(); 2677 mInWrite = true; 2678 2679#define mBitShift 2 // FIXME 2680 size_t count = mixBufferSize >> mBitShift; 2681#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2682 Tracer::traceBegin(ATRACE_TAG, "write"); 2683#endif 2684 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2685#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2686 Tracer::traceEnd(ATRACE_TAG); 2687#endif 2688 if (framesWritten > 0) { 2689 size_t bytesWritten = framesWritten << mBitShift; 2690 mBytesWritten += bytesWritten; 2691 } 2692 2693 mNumWrites++; 2694 mInWrite = false; 2695} 2696 2697void AudioFlinger::MixerThread::threadLoop_standby() 2698{ 2699 // Idle the fast mixer if it's currently running 2700 if (mFastMixer != NULL) { 2701 FastMixerStateQueue *sq = mFastMixer->sq(); 2702 FastMixerState *state = sq->begin(); 2703 if (!(state->mCommand & FastMixerState::IDLE)) { 2704 state->mCommand = FastMixerState::COLD_IDLE; 2705 state->mColdFutexAddr = &mFastMixerFutex; 2706 state->mColdGen++; 2707 mFastMixerFutex = 0; 2708 sq->end(); 2709 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2710 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2711 if (kUseFastMixer == FastMixer_Dynamic) { 2712 mNormalSink = mOutputSink; 2713 } 2714 } else { 2715 sq->end(false /*didModify*/); 2716 } 2717 } 2718 PlaybackThread::threadLoop_standby(); 2719} 2720 2721// shared by MIXER and DIRECT, overridden by DUPLICATING 2722void AudioFlinger::PlaybackThread::threadLoop_standby() 2723{ 2724 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2725 mOutput->stream->common.standby(&mOutput->stream->common); 2726} 2727 2728void AudioFlinger::MixerThread::threadLoop_mix() 2729{ 2730 // obtain the presentation timestamp of the next output buffer 2731 int64_t pts; 2732 status_t status = INVALID_OPERATION; 2733 2734 if (NULL != mOutput->stream->get_next_write_timestamp) { 2735 status = mOutput->stream->get_next_write_timestamp( 2736 mOutput->stream, &pts); 2737 } 2738 2739 if (status != NO_ERROR) { 2740 pts = AudioBufferProvider::kInvalidPTS; 2741 } 2742 2743 // mix buffers... 2744 mAudioMixer->process(pts); 2745 // increase sleep time progressively when application underrun condition clears. 2746 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2747 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2748 // such that we would underrun the audio HAL. 2749 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2750 sleepTimeShift--; 2751 } 2752 sleepTime = 0; 2753 standbyTime = systemTime() + standbyDelay; 2754 //TODO: delay standby when effects have a tail 2755} 2756 2757void AudioFlinger::MixerThread::threadLoop_sleepTime() 2758{ 2759 // If no tracks are ready, sleep once for the duration of an output 2760 // buffer size, then write 0s to the output 2761 if (sleepTime == 0) { 2762 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2763 sleepTime = activeSleepTime >> sleepTimeShift; 2764 if (sleepTime < kMinThreadSleepTimeUs) { 2765 sleepTime = kMinThreadSleepTimeUs; 2766 } 2767 // reduce sleep time in case of consecutive application underruns to avoid 2768 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2769 // duration we would end up writing less data than needed by the audio HAL if 2770 // the condition persists. 2771 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2772 sleepTimeShift++; 2773 } 2774 } else { 2775 sleepTime = idleSleepTime; 2776 } 2777 } else if (mBytesWritten != 0 || 2778 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2779 memset (mMixBuffer, 0, mixBufferSize); 2780 sleepTime = 0; 2781 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2782 } 2783 // TODO add standby time extension fct of effect tail 2784} 2785 2786// prepareTracks_l() must be called with ThreadBase::mLock held 2787AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2788 Vector< sp<Track> > *tracksToRemove) 2789{ 2790 2791 mixer_state mixerStatus = MIXER_IDLE; 2792 // find out which tracks need to be processed 2793 size_t count = mActiveTracks.size(); 2794 size_t mixedTracks = 0; 2795 size_t tracksWithEffect = 0; 2796 // counts only _active_ fast tracks 2797 size_t fastTracks = 0; 2798 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2799 2800 float masterVolume = mMasterVolume; 2801 bool masterMute = mMasterMute; 2802 2803 if (masterMute) { 2804 masterVolume = 0; 2805 } 2806 // Delegate master volume control to effect in output mix effect chain if needed 2807 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2808 if (chain != 0) { 2809 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2810 chain->setVolume_l(&v, &v); 2811 masterVolume = (float)((v + (1 << 23)) >> 24); 2812 chain.clear(); 2813 } 2814 2815 // prepare a new state to push 2816 FastMixerStateQueue *sq = NULL; 2817 FastMixerState *state = NULL; 2818 bool didModify = false; 2819 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2820 if (mFastMixer != NULL) { 2821 sq = mFastMixer->sq(); 2822 state = sq->begin(); 2823 } 2824 2825 for (size_t i=0 ; i<count ; i++) { 2826 sp<Track> t = mActiveTracks[i].promote(); 2827 if (t == 0) continue; 2828 2829 // this const just means the local variable doesn't change 2830 Track* const track = t.get(); 2831 2832 // process fast tracks 2833 if (track->isFastTrack()) { 2834 2835 // It's theoretically possible (though unlikely) for a fast track to be created 2836 // and then removed within the same normal mix cycle. This is not a problem, as 2837 // the track never becomes active so it's fast mixer slot is never touched. 2838 // The converse, of removing an (active) track and then creating a new track 2839 // at the identical fast mixer slot within the same normal mix cycle, 2840 // is impossible because the slot isn't marked available until the end of each cycle. 2841 int j = track->mFastIndex; 2842 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2843 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2844 FastTrack *fastTrack = &state->mFastTracks[j]; 2845 2846 // Determine whether the track is currently in underrun condition, 2847 // and whether it had a recent underrun. 2848 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2849 FastTrackUnderruns underruns = ftDump->mUnderruns; 2850 uint32_t recentFull = (underruns.mBitFields.mFull - 2851 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2852 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2853 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2854 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2855 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2856 uint32_t recentUnderruns = recentPartial + recentEmpty; 2857 track->mObservedUnderruns = underruns; 2858 // don't count underruns that occur while stopping or pausing 2859 // or stopped which can occur when flush() is called while active 2860 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2861 track->mUnderrunCount += recentUnderruns; 2862 } 2863 2864 // This is similar to the state machine for normal tracks, 2865 // with a few modifications for fast tracks. 2866 bool isActive = true; 2867 switch (track->mState) { 2868 case TrackBase::STOPPING_1: 2869 // track stays active in STOPPING_1 state until first underrun 2870 if (recentUnderruns > 0) { 2871 track->mState = TrackBase::STOPPING_2; 2872 } 2873 break; 2874 case TrackBase::PAUSING: 2875 // ramp down is not yet implemented 2876 track->setPaused(); 2877 break; 2878 case TrackBase::RESUMING: 2879 // ramp up is not yet implemented 2880 track->mState = TrackBase::ACTIVE; 2881 break; 2882 case TrackBase::ACTIVE: 2883 if (recentFull > 0 || recentPartial > 0) { 2884 // track has provided at least some frames recently: reset retry count 2885 track->mRetryCount = kMaxTrackRetries; 2886 } 2887 if (recentUnderruns == 0) { 2888 // no recent underruns: stay active 2889 break; 2890 } 2891 // there has recently been an underrun of some kind 2892 if (track->sharedBuffer() == 0) { 2893 // were any of the recent underruns "empty" (no frames available)? 2894 if (recentEmpty == 0) { 2895 // no, then ignore the partial underruns as they are allowed indefinitely 2896 break; 2897 } 2898 // there has recently been an "empty" underrun: decrement the retry counter 2899 if (--(track->mRetryCount) > 0) { 2900 break; 2901 } 2902 // indicate to client process that the track was disabled because of underrun; 2903 // it will then automatically call start() when data is available 2904 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2905 // remove from active list, but state remains ACTIVE [confusing but true] 2906 isActive = false; 2907 break; 2908 } 2909 // fall through 2910 case TrackBase::STOPPING_2: 2911 case TrackBase::PAUSED: 2912 case TrackBase::TERMINATED: 2913 case TrackBase::STOPPED: 2914 case TrackBase::FLUSHED: // flush() while active 2915 // Check for presentation complete if track is inactive 2916 // We have consumed all the buffers of this track. 2917 // This would be incomplete if we auto-paused on underrun 2918 { 2919 size_t audioHALFrames = 2920 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2921 size_t framesWritten = 2922 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2923 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2924 // track stays in active list until presentation is complete 2925 break; 2926 } 2927 } 2928 if (track->isStopping_2()) { 2929 track->mState = TrackBase::STOPPED; 2930 } 2931 if (track->isStopped()) { 2932 // Can't reset directly, as fast mixer is still polling this track 2933 // track->reset(); 2934 // So instead mark this track as needing to be reset after push with ack 2935 resetMask |= 1 << i; 2936 } 2937 isActive = false; 2938 break; 2939 case TrackBase::IDLE: 2940 default: 2941 LOG_FATAL("unexpected track state %d", track->mState); 2942 } 2943 2944 if (isActive) { 2945 // was it previously inactive? 2946 if (!(state->mTrackMask & (1 << j))) { 2947 ExtendedAudioBufferProvider *eabp = track; 2948 VolumeProvider *vp = track; 2949 fastTrack->mBufferProvider = eabp; 2950 fastTrack->mVolumeProvider = vp; 2951 fastTrack->mSampleRate = track->mSampleRate; 2952 fastTrack->mChannelMask = track->mChannelMask; 2953 fastTrack->mGeneration++; 2954 state->mTrackMask |= 1 << j; 2955 didModify = true; 2956 // no acknowledgement required for newly active tracks 2957 } 2958 // cache the combined master volume and stream type volume for fast mixer; this 2959 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2960 track->mCachedVolume = track->isMuted() ? 2961 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2962 ++fastTracks; 2963 } else { 2964 // was it previously active? 2965 if (state->mTrackMask & (1 << j)) { 2966 fastTrack->mBufferProvider = NULL; 2967 fastTrack->mGeneration++; 2968 state->mTrackMask &= ~(1 << j); 2969 didModify = true; 2970 // If any fast tracks were removed, we must wait for acknowledgement 2971 // because we're about to decrement the last sp<> on those tracks. 2972 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2973 } else { 2974 LOG_FATAL("fast track %d should have been active", j); 2975 } 2976 tracksToRemove->add(track); 2977 // Avoids a misleading display in dumpsys 2978 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2979 } 2980 continue; 2981 } 2982 2983 { // local variable scope to avoid goto warning 2984 2985 audio_track_cblk_t* cblk = track->cblk(); 2986 2987 // The first time a track is added we wait 2988 // for all its buffers to be filled before processing it 2989 int name = track->name(); 2990 // make sure that we have enough frames to mix one full buffer. 2991 // enforce this condition only once to enable draining the buffer in case the client 2992 // app does not call stop() and relies on underrun to stop: 2993 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2994 // during last round 2995 uint32_t minFrames = 1; 2996 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2997 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2998 if (t->sampleRate() == (int)mSampleRate) { 2999 minFrames = mNormalFrameCount; 3000 } else { 3001 // +1 for rounding and +1 for additional sample needed for interpolation 3002 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3003 // add frames already consumed but not yet released by the resampler 3004 // because cblk->framesReady() will include these frames 3005 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3006 // the minimum track buffer size is normally twice the number of frames necessary 3007 // to fill one buffer and the resampler should not leave more than one buffer worth 3008 // of unreleased frames after each pass, but just in case... 3009 ALOG_ASSERT(minFrames <= cblk->frameCount); 3010 } 3011 } 3012 if ((track->framesReady() >= minFrames) && track->isReady() && 3013 !track->isPaused() && !track->isTerminated()) 3014 { 3015 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3016 3017 mixedTracks++; 3018 3019 // track->mainBuffer() != mMixBuffer means there is an effect chain 3020 // connected to the track 3021 chain.clear(); 3022 if (track->mainBuffer() != mMixBuffer) { 3023 chain = getEffectChain_l(track->sessionId()); 3024 // Delegate volume control to effect in track effect chain if needed 3025 if (chain != 0) { 3026 tracksWithEffect++; 3027 } else { 3028 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3029 name, track->sessionId()); 3030 } 3031 } 3032 3033 3034 int param = AudioMixer::VOLUME; 3035 if (track->mFillingUpStatus == Track::FS_FILLED) { 3036 // no ramp for the first volume setting 3037 track->mFillingUpStatus = Track::FS_ACTIVE; 3038 if (track->mState == TrackBase::RESUMING) { 3039 track->mState = TrackBase::ACTIVE; 3040 param = AudioMixer::RAMP_VOLUME; 3041 } 3042 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3043 } else if (cblk->server != 0) { 3044 // If the track is stopped before the first frame was mixed, 3045 // do not apply ramp 3046 param = AudioMixer::RAMP_VOLUME; 3047 } 3048 3049 // compute volume for this track 3050 uint32_t vl, vr, va; 3051 if (track->isMuted() || track->isPausing() || 3052 mStreamTypes[track->streamType()].mute) { 3053 vl = vr = va = 0; 3054 if (track->isPausing()) { 3055 track->setPaused(); 3056 } 3057 } else { 3058 3059 // read original volumes with volume control 3060 float typeVolume = mStreamTypes[track->streamType()].volume; 3061 float v = masterVolume * typeVolume; 3062 uint32_t vlr = cblk->getVolumeLR(); 3063 vl = vlr & 0xFFFF; 3064 vr = vlr >> 16; 3065 // track volumes come from shared memory, so can't be trusted and must be clamped 3066 if (vl > MAX_GAIN_INT) { 3067 ALOGV("Track left volume out of range: %04X", vl); 3068 vl = MAX_GAIN_INT; 3069 } 3070 if (vr > MAX_GAIN_INT) { 3071 ALOGV("Track right volume out of range: %04X", vr); 3072 vr = MAX_GAIN_INT; 3073 } 3074 // now apply the master volume and stream type volume 3075 vl = (uint32_t)(v * vl) << 12; 3076 vr = (uint32_t)(v * vr) << 12; 3077 // assuming master volume and stream type volume each go up to 1.0, 3078 // vl and vr are now in 8.24 format 3079 3080 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3081 // send level comes from shared memory and so may be corrupt 3082 if (sendLevel > MAX_GAIN_INT) { 3083 ALOGV("Track send level out of range: %04X", sendLevel); 3084 sendLevel = MAX_GAIN_INT; 3085 } 3086 va = (uint32_t)(v * sendLevel); 3087 } 3088 // Delegate volume control to effect in track effect chain if needed 3089 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3090 // Do not ramp volume if volume is controlled by effect 3091 param = AudioMixer::VOLUME; 3092 track->mHasVolumeController = true; 3093 } else { 3094 // force no volume ramp when volume controller was just disabled or removed 3095 // from effect chain to avoid volume spike 3096 if (track->mHasVolumeController) { 3097 param = AudioMixer::VOLUME; 3098 } 3099 track->mHasVolumeController = false; 3100 } 3101 3102 // Convert volumes from 8.24 to 4.12 format 3103 // This additional clamping is needed in case chain->setVolume_l() overshot 3104 vl = (vl + (1 << 11)) >> 12; 3105 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3106 vr = (vr + (1 << 11)) >> 12; 3107 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3108 3109 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3110 3111 // XXX: these things DON'T need to be done each time 3112 mAudioMixer->setBufferProvider(name, track); 3113 mAudioMixer->enable(name); 3114 3115 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3116 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3117 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3118 mAudioMixer->setParameter( 3119 name, 3120 AudioMixer::TRACK, 3121 AudioMixer::FORMAT, (void *)track->format()); 3122 mAudioMixer->setParameter( 3123 name, 3124 AudioMixer::TRACK, 3125 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3126 mAudioMixer->setParameter( 3127 name, 3128 AudioMixer::RESAMPLE, 3129 AudioMixer::SAMPLE_RATE, 3130 (void *)(cblk->sampleRate)); 3131 mAudioMixer->setParameter( 3132 name, 3133 AudioMixer::TRACK, 3134 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3135 mAudioMixer->setParameter( 3136 name, 3137 AudioMixer::TRACK, 3138 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3139 3140 // reset retry count 3141 track->mRetryCount = kMaxTrackRetries; 3142 3143 // If one track is ready, set the mixer ready if: 3144 // - the mixer was not ready during previous round OR 3145 // - no other track is not ready 3146 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3147 mixerStatus != MIXER_TRACKS_ENABLED) { 3148 mixerStatus = MIXER_TRACKS_READY; 3149 } 3150 } else { 3151 // clear effect chain input buffer if an active track underruns to avoid sending 3152 // previous audio buffer again to effects 3153 chain = getEffectChain_l(track->sessionId()); 3154 if (chain != 0) { 3155 chain->clearInputBuffer(); 3156 } 3157 3158 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3159 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3160 track->isStopped() || track->isPaused()) { 3161 // We have consumed all the buffers of this track. 3162 // Remove it from the list of active tracks. 3163 // TODO: use actual buffer filling status instead of latency when available from 3164 // audio HAL 3165 size_t audioHALFrames = 3166 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3167 size_t framesWritten = 3168 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3169 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3170 if (track->isStopped()) { 3171 track->reset(); 3172 } 3173 tracksToRemove->add(track); 3174 } 3175 } else { 3176 track->mUnderrunCount++; 3177 // No buffers for this track. Give it a few chances to 3178 // fill a buffer, then remove it from active list. 3179 if (--(track->mRetryCount) <= 0) { 3180 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3181 tracksToRemove->add(track); 3182 // indicate to client process that the track was disabled because of underrun; 3183 // it will then automatically call start() when data is available 3184 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3185 // If one track is not ready, mark the mixer also not ready if: 3186 // - the mixer was ready during previous round OR 3187 // - no other track is ready 3188 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3189 mixerStatus != MIXER_TRACKS_READY) { 3190 mixerStatus = MIXER_TRACKS_ENABLED; 3191 } 3192 } 3193 mAudioMixer->disable(name); 3194 } 3195 3196 } // local variable scope to avoid goto warning 3197track_is_ready: ; 3198 3199 } 3200 3201 // Push the new FastMixer state if necessary 3202 if (didModify) { 3203 state->mFastTracksGen++; 3204 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3205 if (kUseFastMixer == FastMixer_Dynamic && 3206 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3207 state->mCommand = FastMixerState::COLD_IDLE; 3208 state->mColdFutexAddr = &mFastMixerFutex; 3209 state->mColdGen++; 3210 mFastMixerFutex = 0; 3211 if (kUseFastMixer == FastMixer_Dynamic) { 3212 mNormalSink = mOutputSink; 3213 } 3214 // If we go into cold idle, need to wait for acknowledgement 3215 // so that fast mixer stops doing I/O. 3216 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3217 } 3218 sq->end(); 3219 } 3220 if (sq != NULL) { 3221 sq->end(didModify); 3222 sq->push(block); 3223 } 3224 3225 // Now perform the deferred reset on fast tracks that have stopped 3226 while (resetMask != 0) { 3227 size_t i = __builtin_ctz(resetMask); 3228 ALOG_ASSERT(i < count); 3229 resetMask &= ~(1 << i); 3230 sp<Track> t = mActiveTracks[i].promote(); 3231 if (t == 0) continue; 3232 Track* track = t.get(); 3233 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3234 track->reset(); 3235 } 3236 3237 // remove all the tracks that need to be... 3238 count = tracksToRemove->size(); 3239 if (CC_UNLIKELY(count)) { 3240 for (size_t i=0 ; i<count ; i++) { 3241 const sp<Track>& track = tracksToRemove->itemAt(i); 3242 mActiveTracks.remove(track); 3243 if (track->mainBuffer() != mMixBuffer) { 3244 chain = getEffectChain_l(track->sessionId()); 3245 if (chain != 0) { 3246 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3247 chain->decActiveTrackCnt(); 3248 } 3249 } 3250 if (track->isTerminated()) { 3251 removeTrack_l(track); 3252 } 3253 } 3254 } 3255 3256 // mix buffer must be cleared if all tracks are connected to an 3257 // effect chain as in this case the mixer will not write to 3258 // mix buffer and track effects will accumulate into it 3259 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3260 // FIXME as a performance optimization, should remember previous zero status 3261 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3262 } 3263 3264 // if any fast tracks, then status is ready 3265 mMixerStatusIgnoringFastTracks = mixerStatus; 3266 if (fastTracks > 0) { 3267 mixerStatus = MIXER_TRACKS_READY; 3268 } 3269 return mixerStatus; 3270} 3271 3272/* 3273The derived values that are cached: 3274 - mixBufferSize from frame count * frame size 3275 - activeSleepTime from activeSleepTimeUs() 3276 - idleSleepTime from idleSleepTimeUs() 3277 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3278 - maxPeriod from frame count and sample rate (MIXER only) 3279 3280The parameters that affect these derived values are: 3281 - frame count 3282 - frame size 3283 - sample rate 3284 - device type: A2DP or not 3285 - device latency 3286 - format: PCM or not 3287 - active sleep time 3288 - idle sleep time 3289*/ 3290 3291void AudioFlinger::PlaybackThread::cacheParameters_l() 3292{ 3293 mixBufferSize = mNormalFrameCount * mFrameSize; 3294 activeSleepTime = activeSleepTimeUs(); 3295 idleSleepTime = idleSleepTimeUs(); 3296} 3297 3298void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3299{ 3300 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3301 this, streamType, mTracks.size()); 3302 Mutex::Autolock _l(mLock); 3303 3304 size_t size = mTracks.size(); 3305 for (size_t i = 0; i < size; i++) { 3306 sp<Track> t = mTracks[i]; 3307 if (t->streamType() == streamType) { 3308 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3309 t->mCblk->cv.signal(); 3310 } 3311 } 3312} 3313 3314// getTrackName_l() must be called with ThreadBase::mLock held 3315int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3316{ 3317 return mAudioMixer->getTrackName(channelMask); 3318} 3319 3320// deleteTrackName_l() must be called with ThreadBase::mLock held 3321void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3322{ 3323 ALOGV("remove track (%d) and delete from mixer", name); 3324 mAudioMixer->deleteTrackName(name); 3325} 3326 3327// checkForNewParameters_l() must be called with ThreadBase::mLock held 3328bool AudioFlinger::MixerThread::checkForNewParameters_l() 3329{ 3330 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3331 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3332 bool reconfig = false; 3333 3334 while (!mNewParameters.isEmpty()) { 3335 3336 if (mFastMixer != NULL) { 3337 FastMixerStateQueue *sq = mFastMixer->sq(); 3338 FastMixerState *state = sq->begin(); 3339 if (!(state->mCommand & FastMixerState::IDLE)) { 3340 previousCommand = state->mCommand; 3341 state->mCommand = FastMixerState::HOT_IDLE; 3342 sq->end(); 3343 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3344 } else { 3345 sq->end(false /*didModify*/); 3346 } 3347 } 3348 3349 status_t status = NO_ERROR; 3350 String8 keyValuePair = mNewParameters[0]; 3351 AudioParameter param = AudioParameter(keyValuePair); 3352 int value; 3353 3354 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3355 reconfig = true; 3356 } 3357 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3358 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3359 status = BAD_VALUE; 3360 } else { 3361 reconfig = true; 3362 } 3363 } 3364 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3365 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3366 status = BAD_VALUE; 3367 } else { 3368 reconfig = true; 3369 } 3370 } 3371 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3372 // do not accept frame count changes if tracks are open as the track buffer 3373 // size depends on frame count and correct behavior would not be guaranteed 3374 // if frame count is changed after track creation 3375 if (!mTracks.isEmpty()) { 3376 status = INVALID_OPERATION; 3377 } else { 3378 reconfig = true; 3379 } 3380 } 3381 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3382#ifdef ADD_BATTERY_DATA 3383 // when changing the audio output device, call addBatteryData to notify 3384 // the change 3385 if ((int)mDevice != value) { 3386 uint32_t params = 0; 3387 // check whether speaker is on 3388 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3389 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3390 } 3391 3392 int deviceWithoutSpeaker 3393 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3394 // check if any other device (except speaker) is on 3395 if (value & deviceWithoutSpeaker ) { 3396 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3397 } 3398 3399 if (params != 0) { 3400 addBatteryData(params); 3401 } 3402 } 3403#endif 3404 3405 // forward device change to effects that have requested to be 3406 // aware of attached audio device. 3407 mDevice = (uint32_t)value; 3408 for (size_t i = 0; i < mEffectChains.size(); i++) { 3409 mEffectChains[i]->setDevice_l(mDevice); 3410 } 3411 } 3412 3413 if (status == NO_ERROR) { 3414 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3415 keyValuePair.string()); 3416 if (!mStandby && status == INVALID_OPERATION) { 3417 mOutput->stream->common.standby(&mOutput->stream->common); 3418 mStandby = true; 3419 mBytesWritten = 0; 3420 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3421 keyValuePair.string()); 3422 } 3423 if (status == NO_ERROR && reconfig) { 3424 delete mAudioMixer; 3425 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3426 mAudioMixer = NULL; 3427 readOutputParameters(); 3428 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3429 for (size_t i = 0; i < mTracks.size() ; i++) { 3430 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3431 if (name < 0) break; 3432 mTracks[i]->mName = name; 3433 // limit track sample rate to 2 x new output sample rate 3434 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3435 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3436 } 3437 } 3438 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3439 } 3440 } 3441 3442 mNewParameters.removeAt(0); 3443 3444 mParamStatus = status; 3445 mParamCond.signal(); 3446 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3447 // already timed out waiting for the status and will never signal the condition. 3448 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3449 } 3450 3451 if (!(previousCommand & FastMixerState::IDLE)) { 3452 ALOG_ASSERT(mFastMixer != NULL); 3453 FastMixerStateQueue *sq = mFastMixer->sq(); 3454 FastMixerState *state = sq->begin(); 3455 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3456 state->mCommand = previousCommand; 3457 sq->end(); 3458 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3459 } 3460 3461 return reconfig; 3462} 3463 3464status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3465{ 3466 const size_t SIZE = 256; 3467 char buffer[SIZE]; 3468 String8 result; 3469 3470 PlaybackThread::dumpInternals(fd, args); 3471 3472 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3473 result.append(buffer); 3474 write(fd, result.string(), result.size()); 3475 3476 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3477 FastMixerDumpState copy = mFastMixerDumpState; 3478 copy.dump(fd); 3479 3480#ifdef STATE_QUEUE_DUMP 3481 // Similar for state queue 3482 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3483 observerCopy.dump(fd); 3484 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3485 mutatorCopy.dump(fd); 3486#endif 3487 3488 // Write the tee output to a .wav file 3489 NBAIO_Source *teeSource = mTeeSource.get(); 3490 if (teeSource != NULL) { 3491 char teePath[64]; 3492 struct timeval tv; 3493 gettimeofday(&tv, NULL); 3494 struct tm tm; 3495 localtime_r(&tv.tv_sec, &tm); 3496 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3497 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3498 if (teeFd >= 0) { 3499 char wavHeader[44]; 3500 memcpy(wavHeader, 3501 "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", 3502 sizeof(wavHeader)); 3503 NBAIO_Format format = teeSource->format(); 3504 unsigned channelCount = Format_channelCount(format); 3505 ALOG_ASSERT(channelCount <= FCC_2); 3506 unsigned sampleRate = Format_sampleRate(format); 3507 wavHeader[22] = channelCount; // number of channels 3508 wavHeader[24] = sampleRate; // sample rate 3509 wavHeader[25] = sampleRate >> 8; 3510 wavHeader[32] = channelCount * 2; // block alignment 3511 write(teeFd, wavHeader, sizeof(wavHeader)); 3512 size_t total = 0; 3513 bool firstRead = true; 3514 for (;;) { 3515#define TEE_SINK_READ 1024 3516 short buffer[TEE_SINK_READ * FCC_2]; 3517 size_t count = TEE_SINK_READ; 3518 ssize_t actual = teeSource->read(buffer, count); 3519 bool wasFirstRead = firstRead; 3520 firstRead = false; 3521 if (actual <= 0) { 3522 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3523 continue; 3524 } 3525 break; 3526 } 3527 ALOG_ASSERT(actual <= count); 3528 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3529 total += actual; 3530 } 3531 lseek(teeFd, (off_t) 4, SEEK_SET); 3532 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3533 write(teeFd, &temp, sizeof(temp)); 3534 lseek(teeFd, (off_t) 40, SEEK_SET); 3535 temp = total * channelCount * sizeof(short); 3536 write(teeFd, &temp, sizeof(temp)); 3537 close(teeFd); 3538 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3539 } else { 3540 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3541 } 3542 } 3543 3544 return NO_ERROR; 3545} 3546 3547uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3548{ 3549 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3550} 3551 3552uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3553{ 3554 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3555} 3556 3557void AudioFlinger::MixerThread::cacheParameters_l() 3558{ 3559 PlaybackThread::cacheParameters_l(); 3560 3561 // FIXME: Relaxed timing because of a certain device that can't meet latency 3562 // Should be reduced to 2x after the vendor fixes the driver issue 3563 // increase threshold again due to low power audio mode. The way this warning 3564 // threshold is calculated and its usefulness should be reconsidered anyway. 3565 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3566} 3567 3568// ---------------------------------------------------------------------------- 3569AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3570 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3571 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3572 // mLeftVolFloat, mRightVolFloat 3573 // mLeftVolShort, mRightVolShort 3574{ 3575} 3576 3577AudioFlinger::DirectOutputThread::~DirectOutputThread() 3578{ 3579} 3580 3581AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3582 Vector< sp<Track> > *tracksToRemove 3583) 3584{ 3585 sp<Track> trackToRemove; 3586 3587 mixer_state mixerStatus = MIXER_IDLE; 3588 3589 // find out which tracks need to be processed 3590 if (mActiveTracks.size() != 0) { 3591 sp<Track> t = mActiveTracks[0].promote(); 3592 // The track died recently 3593 if (t == 0) return MIXER_IDLE; 3594 3595 Track* const track = t.get(); 3596 audio_track_cblk_t* cblk = track->cblk(); 3597 3598 // The first time a track is added we wait 3599 // for all its buffers to be filled before processing it 3600 if (cblk->framesReady() && track->isReady() && 3601 !track->isPaused() && !track->isTerminated()) 3602 { 3603 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3604 3605 if (track->mFillingUpStatus == Track::FS_FILLED) { 3606 track->mFillingUpStatus = Track::FS_ACTIVE; 3607 mLeftVolFloat = mRightVolFloat = 0; 3608 mLeftVolShort = mRightVolShort = 0; 3609 if (track->mState == TrackBase::RESUMING) { 3610 track->mState = TrackBase::ACTIVE; 3611 rampVolume = true; 3612 } 3613 } else if (cblk->server != 0) { 3614 // If the track is stopped before the first frame was mixed, 3615 // do not apply ramp 3616 rampVolume = true; 3617 } 3618 // compute volume for this track 3619 float left, right; 3620 if (track->isMuted() || mMasterMute || track->isPausing() || 3621 mStreamTypes[track->streamType()].mute) { 3622 left = right = 0; 3623 if (track->isPausing()) { 3624 track->setPaused(); 3625 } 3626 } else { 3627 float typeVolume = mStreamTypes[track->streamType()].volume; 3628 float v = mMasterVolume * typeVolume; 3629 uint32_t vlr = cblk->getVolumeLR(); 3630 float v_clamped = v * (vlr & 0xFFFF); 3631 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3632 left = v_clamped/MAX_GAIN; 3633 v_clamped = v * (vlr >> 16); 3634 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3635 right = v_clamped/MAX_GAIN; 3636 } 3637 3638 if (left != mLeftVolFloat || right != mRightVolFloat) { 3639 mLeftVolFloat = left; 3640 mRightVolFloat = right; 3641 3642 // If audio HAL implements volume control, 3643 // force software volume to nominal value 3644 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3645 left = 1.0f; 3646 right = 1.0f; 3647 } 3648 3649 // Convert volumes from float to 8.24 3650 uint32_t vl = (uint32_t)(left * (1 << 24)); 3651 uint32_t vr = (uint32_t)(right * (1 << 24)); 3652 3653 // Delegate volume control to effect in track effect chain if needed 3654 // only one effect chain can be present on DirectOutputThread, so if 3655 // there is one, the track is connected to it 3656 if (!mEffectChains.isEmpty()) { 3657 // Do not ramp volume if volume is controlled by effect 3658 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3659 rampVolume = false; 3660 } 3661 } 3662 3663 // Convert volumes from 8.24 to 4.12 format 3664 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3665 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3666 leftVol = (uint16_t)v_clamped; 3667 v_clamped = (vr + (1 << 11)) >> 12; 3668 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3669 rightVol = (uint16_t)v_clamped; 3670 } else { 3671 leftVol = mLeftVolShort; 3672 rightVol = mRightVolShort; 3673 rampVolume = false; 3674 } 3675 3676 // reset retry count 3677 track->mRetryCount = kMaxTrackRetriesDirect; 3678 mActiveTrack = t; 3679 mixerStatus = MIXER_TRACKS_READY; 3680 } else { 3681 // clear effect chain input buffer if an active track underruns to avoid sending 3682 // previous audio buffer again to effects 3683 if (!mEffectChains.isEmpty()) { 3684 mEffectChains[0]->clearInputBuffer(); 3685 } 3686 3687 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3688 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3689 // We have consumed all the buffers of this track. 3690 // Remove it from the list of active tracks. 3691 // TODO: implement behavior for compressed audio 3692 size_t audioHALFrames = 3693 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3694 size_t framesWritten = 3695 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3696 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3697 if (track->isStopped()) { 3698 track->reset(); 3699 } 3700 trackToRemove = track; 3701 } 3702 } else { 3703 // No buffers for this track. Give it a few chances to 3704 // fill a buffer, then remove it from active list. 3705 if (--(track->mRetryCount) <= 0) { 3706 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3707 trackToRemove = track; 3708 } else { 3709 mixerStatus = MIXER_TRACKS_ENABLED; 3710 } 3711 } 3712 } 3713 } 3714 3715 // FIXME merge this with similar code for removing multiple tracks 3716 // remove all the tracks that need to be... 3717 if (CC_UNLIKELY(trackToRemove != 0)) { 3718 tracksToRemove->add(trackToRemove); 3719 mActiveTracks.remove(trackToRemove); 3720 if (!mEffectChains.isEmpty()) { 3721 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3722 trackToRemove->sessionId()); 3723 mEffectChains[0]->decActiveTrackCnt(); 3724 } 3725 if (trackToRemove->isTerminated()) { 3726 removeTrack_l(trackToRemove); 3727 } 3728 } 3729 3730 return mixerStatus; 3731} 3732 3733void AudioFlinger::DirectOutputThread::threadLoop_mix() 3734{ 3735 AudioBufferProvider::Buffer buffer; 3736 size_t frameCount = mFrameCount; 3737 int8_t *curBuf = (int8_t *)mMixBuffer; 3738 // output audio to hardware 3739 while (frameCount) { 3740 buffer.frameCount = frameCount; 3741 mActiveTrack->getNextBuffer(&buffer); 3742 if (CC_UNLIKELY(buffer.raw == NULL)) { 3743 memset(curBuf, 0, frameCount * mFrameSize); 3744 break; 3745 } 3746 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3747 frameCount -= buffer.frameCount; 3748 curBuf += buffer.frameCount * mFrameSize; 3749 mActiveTrack->releaseBuffer(&buffer); 3750 } 3751 sleepTime = 0; 3752 standbyTime = systemTime() + standbyDelay; 3753 mActiveTrack.clear(); 3754 3755 // apply volume 3756 3757 // Do not apply volume on compressed audio 3758 if (!audio_is_linear_pcm(mFormat)) { 3759 return; 3760 } 3761 3762 // convert to signed 16 bit before volume calculation 3763 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3764 size_t count = mFrameCount * mChannelCount; 3765 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3766 int16_t *dst = mMixBuffer + count-1; 3767 while (count--) { 3768 *dst-- = (int16_t)(*src--^0x80) << 8; 3769 } 3770 } 3771 3772 frameCount = mFrameCount; 3773 int16_t *out = mMixBuffer; 3774 if (rampVolume) { 3775 if (mChannelCount == 1) { 3776 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3777 int32_t vlInc = d / (int32_t)frameCount; 3778 int32_t vl = ((int32_t)mLeftVolShort << 16); 3779 do { 3780 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3781 out++; 3782 vl += vlInc; 3783 } while (--frameCount); 3784 3785 } else { 3786 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3787 int32_t vlInc = d / (int32_t)frameCount; 3788 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3789 int32_t vrInc = d / (int32_t)frameCount; 3790 int32_t vl = ((int32_t)mLeftVolShort << 16); 3791 int32_t vr = ((int32_t)mRightVolShort << 16); 3792 do { 3793 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3794 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3795 out += 2; 3796 vl += vlInc; 3797 vr += vrInc; 3798 } while (--frameCount); 3799 } 3800 } else { 3801 if (mChannelCount == 1) { 3802 do { 3803 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3804 out++; 3805 } while (--frameCount); 3806 } else { 3807 do { 3808 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3809 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3810 out += 2; 3811 } while (--frameCount); 3812 } 3813 } 3814 3815 // convert back to unsigned 8 bit after volume calculation 3816 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3817 size_t count = mFrameCount * mChannelCount; 3818 int16_t *src = mMixBuffer; 3819 uint8_t *dst = (uint8_t *)mMixBuffer; 3820 while (count--) { 3821 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3822 } 3823 } 3824 3825 mLeftVolShort = leftVol; 3826 mRightVolShort = rightVol; 3827} 3828 3829void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3830{ 3831 if (sleepTime == 0) { 3832 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3833 sleepTime = activeSleepTime; 3834 } else { 3835 sleepTime = idleSleepTime; 3836 } 3837 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3838 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3839 sleepTime = 0; 3840 } 3841} 3842 3843// getTrackName_l() must be called with ThreadBase::mLock held 3844int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3845{ 3846 return 0; 3847} 3848 3849// deleteTrackName_l() must be called with ThreadBase::mLock held 3850void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3851{ 3852} 3853 3854// checkForNewParameters_l() must be called with ThreadBase::mLock held 3855bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3856{ 3857 bool reconfig = false; 3858 3859 while (!mNewParameters.isEmpty()) { 3860 status_t status = NO_ERROR; 3861 String8 keyValuePair = mNewParameters[0]; 3862 AudioParameter param = AudioParameter(keyValuePair); 3863 int value; 3864 3865 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3866 // do not accept frame count changes if tracks are open as the track buffer 3867 // size depends on frame count and correct behavior would not be garantied 3868 // if frame count is changed after track creation 3869 if (!mTracks.isEmpty()) { 3870 status = INVALID_OPERATION; 3871 } else { 3872 reconfig = true; 3873 } 3874 } 3875 if (status == NO_ERROR) { 3876 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3877 keyValuePair.string()); 3878 if (!mStandby && status == INVALID_OPERATION) { 3879 mOutput->stream->common.standby(&mOutput->stream->common); 3880 mStandby = true; 3881 mBytesWritten = 0; 3882 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3883 keyValuePair.string()); 3884 } 3885 if (status == NO_ERROR && reconfig) { 3886 readOutputParameters(); 3887 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3888 } 3889 } 3890 3891 mNewParameters.removeAt(0); 3892 3893 mParamStatus = status; 3894 mParamCond.signal(); 3895 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3896 // already timed out waiting for the status and will never signal the condition. 3897 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3898 } 3899 return reconfig; 3900} 3901 3902uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3903{ 3904 uint32_t time; 3905 if (audio_is_linear_pcm(mFormat)) { 3906 time = PlaybackThread::activeSleepTimeUs(); 3907 } else { 3908 time = 10000; 3909 } 3910 return time; 3911} 3912 3913uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3914{ 3915 uint32_t time; 3916 if (audio_is_linear_pcm(mFormat)) { 3917 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3918 } else { 3919 time = 10000; 3920 } 3921 return time; 3922} 3923 3924uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3925{ 3926 uint32_t time; 3927 if (audio_is_linear_pcm(mFormat)) { 3928 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3929 } else { 3930 time = 10000; 3931 } 3932 return time; 3933} 3934 3935void AudioFlinger::DirectOutputThread::cacheParameters_l() 3936{ 3937 PlaybackThread::cacheParameters_l(); 3938 3939 // use shorter standby delay as on normal output to release 3940 // hardware resources as soon as possible 3941 standbyDelay = microseconds(activeSleepTime*2); 3942} 3943 3944// ---------------------------------------------------------------------------- 3945 3946AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3947 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3948 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3949 mWaitTimeMs(UINT_MAX) 3950{ 3951 addOutputTrack(mainThread); 3952} 3953 3954AudioFlinger::DuplicatingThread::~DuplicatingThread() 3955{ 3956 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3957 mOutputTracks[i]->destroy(); 3958 } 3959} 3960 3961void AudioFlinger::DuplicatingThread::threadLoop_mix() 3962{ 3963 // mix buffers... 3964 if (outputsReady(outputTracks)) { 3965 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3966 } else { 3967 memset(mMixBuffer, 0, mixBufferSize); 3968 } 3969 sleepTime = 0; 3970 writeFrames = mNormalFrameCount; 3971} 3972 3973void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3974{ 3975 if (sleepTime == 0) { 3976 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3977 sleepTime = activeSleepTime; 3978 } else { 3979 sleepTime = idleSleepTime; 3980 } 3981 } else if (mBytesWritten != 0) { 3982 // flush remaining overflow buffers in output tracks 3983 for (size_t i = 0; i < outputTracks.size(); i++) { 3984 if (outputTracks[i]->isActive()) { 3985 sleepTime = 0; 3986 writeFrames = 0; 3987 memset(mMixBuffer, 0, mixBufferSize); 3988 break; 3989 } 3990 } 3991 } 3992} 3993 3994void AudioFlinger::DuplicatingThread::threadLoop_write() 3995{ 3996 standbyTime = systemTime() + standbyDelay; 3997 for (size_t i = 0; i < outputTracks.size(); i++) { 3998 outputTracks[i]->write(mMixBuffer, writeFrames); 3999 } 4000 mBytesWritten += mixBufferSize; 4001} 4002 4003void AudioFlinger::DuplicatingThread::threadLoop_standby() 4004{ 4005 // DuplicatingThread implements standby by stopping all tracks 4006 for (size_t i = 0; i < outputTracks.size(); i++) { 4007 outputTracks[i]->stop(); 4008 } 4009} 4010 4011void AudioFlinger::DuplicatingThread::saveOutputTracks() 4012{ 4013 outputTracks = mOutputTracks; 4014} 4015 4016void AudioFlinger::DuplicatingThread::clearOutputTracks() 4017{ 4018 outputTracks.clear(); 4019} 4020 4021void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4022{ 4023 Mutex::Autolock _l(mLock); 4024 // FIXME explain this formula 4025 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4026 OutputTrack *outputTrack = new OutputTrack(thread, 4027 this, 4028 mSampleRate, 4029 mFormat, 4030 mChannelMask, 4031 frameCount); 4032 if (outputTrack->cblk() != NULL) { 4033 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4034 mOutputTracks.add(outputTrack); 4035 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4036 updateWaitTime_l(); 4037 } 4038} 4039 4040void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4041{ 4042 Mutex::Autolock _l(mLock); 4043 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4044 if (mOutputTracks[i]->thread() == thread) { 4045 mOutputTracks[i]->destroy(); 4046 mOutputTracks.removeAt(i); 4047 updateWaitTime_l(); 4048 return; 4049 } 4050 } 4051 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4052} 4053 4054// caller must hold mLock 4055void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4056{ 4057 mWaitTimeMs = UINT_MAX; 4058 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4059 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4060 if (strong != 0) { 4061 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4062 if (waitTimeMs < mWaitTimeMs) { 4063 mWaitTimeMs = waitTimeMs; 4064 } 4065 } 4066 } 4067} 4068 4069 4070bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4071{ 4072 for (size_t i = 0; i < outputTracks.size(); i++) { 4073 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4074 if (thread == 0) { 4075 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4076 return false; 4077 } 4078 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4079 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4080 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4081 return false; 4082 } 4083 } 4084 return true; 4085} 4086 4087uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4088{ 4089 return (mWaitTimeMs * 1000) / 2; 4090} 4091 4092void AudioFlinger::DuplicatingThread::cacheParameters_l() 4093{ 4094 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4095 updateWaitTime_l(); 4096 4097 MixerThread::cacheParameters_l(); 4098} 4099 4100// ---------------------------------------------------------------------------- 4101 4102// TrackBase constructor must be called with AudioFlinger::mLock held 4103AudioFlinger::ThreadBase::TrackBase::TrackBase( 4104 ThreadBase *thread, 4105 const sp<Client>& client, 4106 uint32_t sampleRate, 4107 audio_format_t format, 4108 uint32_t channelMask, 4109 int frameCount, 4110 const sp<IMemory>& sharedBuffer, 4111 int sessionId) 4112 : RefBase(), 4113 mThread(thread), 4114 mClient(client), 4115 mCblk(NULL), 4116 // mBuffer 4117 // mBufferEnd 4118 mFrameCount(0), 4119 mState(IDLE), 4120 mSampleRate(sampleRate), 4121 mFormat(format), 4122 mStepServerFailed(false), 4123 mSessionId(sessionId) 4124 // mChannelCount 4125 // mChannelMask 4126{ 4127 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4128 4129 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4130 size_t size = sizeof(audio_track_cblk_t); 4131 uint8_t channelCount = popcount(channelMask); 4132 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4133 if (sharedBuffer == 0) { 4134 size += bufferSize; 4135 } 4136 4137 if (client != NULL) { 4138 mCblkMemory = client->heap()->allocate(size); 4139 if (mCblkMemory != 0) { 4140 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4141 if (mCblk != NULL) { // construct the shared structure in-place. 4142 new(mCblk) audio_track_cblk_t(); 4143 // clear all buffers 4144 mCblk->frameCount = frameCount; 4145 mCblk->sampleRate = sampleRate; 4146// uncomment the following lines to quickly test 32-bit wraparound 4147// mCblk->user = 0xffff0000; 4148// mCblk->server = 0xffff0000; 4149// mCblk->userBase = 0xffff0000; 4150// mCblk->serverBase = 0xffff0000; 4151 mChannelCount = channelCount; 4152 mChannelMask = channelMask; 4153 if (sharedBuffer == 0) { 4154 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4155 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4156 // Force underrun condition to avoid false underrun callback until first data is 4157 // written to buffer (other flags are cleared) 4158 mCblk->flags = CBLK_UNDERRUN_ON; 4159 } else { 4160 mBuffer = sharedBuffer->pointer(); 4161 } 4162 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4163 } 4164 } else { 4165 ALOGE("not enough memory for AudioTrack size=%u", size); 4166 client->heap()->dump("AudioTrack"); 4167 return; 4168 } 4169 } else { 4170 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4171 // construct the shared structure in-place. 4172 new(mCblk) audio_track_cblk_t(); 4173 // clear all buffers 4174 mCblk->frameCount = frameCount; 4175 mCblk->sampleRate = sampleRate; 4176// uncomment the following lines to quickly test 32-bit wraparound 4177// mCblk->user = 0xffff0000; 4178// mCblk->server = 0xffff0000; 4179// mCblk->userBase = 0xffff0000; 4180// mCblk->serverBase = 0xffff0000; 4181 mChannelCount = channelCount; 4182 mChannelMask = channelMask; 4183 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4184 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4185 // Force underrun condition to avoid false underrun callback until first data is 4186 // written to buffer (other flags are cleared) 4187 mCblk->flags = CBLK_UNDERRUN_ON; 4188 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4189 } 4190} 4191 4192AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4193{ 4194 if (mCblk != NULL) { 4195 if (mClient == 0) { 4196 delete mCblk; 4197 } else { 4198 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4199 } 4200 } 4201 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4202 if (mClient != 0) { 4203 // Client destructor must run with AudioFlinger mutex locked 4204 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4205 // If the client's reference count drops to zero, the associated destructor 4206 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4207 // relying on the automatic clear() at end of scope. 4208 mClient.clear(); 4209 } 4210} 4211 4212// AudioBufferProvider interface 4213// getNextBuffer() = 0; 4214// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4215void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4216{ 4217 buffer->raw = NULL; 4218 mFrameCount = buffer->frameCount; 4219 // FIXME See note at getNextBuffer() 4220 (void) step(); // ignore return value of step() 4221 buffer->frameCount = 0; 4222} 4223 4224bool AudioFlinger::ThreadBase::TrackBase::step() { 4225 bool result; 4226 audio_track_cblk_t* cblk = this->cblk(); 4227 4228 result = cblk->stepServer(mFrameCount); 4229 if (!result) { 4230 ALOGV("stepServer failed acquiring cblk mutex"); 4231 mStepServerFailed = true; 4232 } 4233 return result; 4234} 4235 4236void AudioFlinger::ThreadBase::TrackBase::reset() { 4237 audio_track_cblk_t* cblk = this->cblk(); 4238 4239 cblk->user = 0; 4240 cblk->server = 0; 4241 cblk->userBase = 0; 4242 cblk->serverBase = 0; 4243 mStepServerFailed = false; 4244 ALOGV("TrackBase::reset"); 4245} 4246 4247int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4248 return (int)mCblk->sampleRate; 4249} 4250 4251void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4252 audio_track_cblk_t* cblk = this->cblk(); 4253 size_t frameSize = cblk->frameSize; 4254 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4255 int8_t *bufferEnd = bufferStart + frames * frameSize; 4256 4257 // Check validity of returned pointer in case the track control block would have been corrupted. 4258 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4259 "TrackBase::getBuffer buffer out of range:\n" 4260 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4261 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4262 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4263 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4264 4265 return bufferStart; 4266} 4267 4268status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4269{ 4270 mSyncEvents.add(event); 4271 return NO_ERROR; 4272} 4273 4274// ---------------------------------------------------------------------------- 4275 4276// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4277AudioFlinger::PlaybackThread::Track::Track( 4278 PlaybackThread *thread, 4279 const sp<Client>& client, 4280 audio_stream_type_t streamType, 4281 uint32_t sampleRate, 4282 audio_format_t format, 4283 uint32_t channelMask, 4284 int frameCount, 4285 const sp<IMemory>& sharedBuffer, 4286 int sessionId, 4287 IAudioFlinger::track_flags_t flags) 4288 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4289 mMute(false), 4290 mFillingUpStatus(FS_INVALID), 4291 // mRetryCount initialized later when needed 4292 mSharedBuffer(sharedBuffer), 4293 mStreamType(streamType), 4294 mName(-1), // see note below 4295 mMainBuffer(thread->mixBuffer()), 4296 mAuxBuffer(NULL), 4297 mAuxEffectId(0), mHasVolumeController(false), 4298 mPresentationCompleteFrames(0), 4299 mFlags(flags), 4300 mFastIndex(-1), 4301 mUnderrunCount(0), 4302 mCachedVolume(1.0) 4303{ 4304 if (mCblk != NULL) { 4305 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4306 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4307 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4308 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4309 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4310 if (mName < 0) { 4311 ALOGE("no more track names available"); 4312 return; 4313 } 4314 // only allocate a fast track index if we were able to allocate a normal track name 4315 if (flags & IAudioFlinger::TRACK_FAST) { 4316 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4317 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4318 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4319 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4320 // FIXME This is too eager. We allocate a fast track index before the 4321 // fast track becomes active. Since fast tracks are a scarce resource, 4322 // this means we are potentially denying other more important fast tracks from 4323 // being created. It would be better to allocate the index dynamically. 4324 mFastIndex = i; 4325 // Read the initial underruns because this field is never cleared by the fast mixer 4326 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4327 thread->mFastTrackAvailMask &= ~(1 << i); 4328 } 4329 } 4330 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4331} 4332 4333AudioFlinger::PlaybackThread::Track::~Track() 4334{ 4335 ALOGV("PlaybackThread::Track destructor"); 4336 sp<ThreadBase> thread = mThread.promote(); 4337 if (thread != 0) { 4338 Mutex::Autolock _l(thread->mLock); 4339 mState = TERMINATED; 4340 } 4341} 4342 4343void AudioFlinger::PlaybackThread::Track::destroy() 4344{ 4345 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4346 // by removing it from mTracks vector, so there is a risk that this Tracks's 4347 // destructor is called. As the destructor needs to lock mLock, 4348 // we must acquire a strong reference on this Track before locking mLock 4349 // here so that the destructor is called only when exiting this function. 4350 // On the other hand, as long as Track::destroy() is only called by 4351 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4352 // this Track with its member mTrack. 4353 sp<Track> keep(this); 4354 { // scope for mLock 4355 sp<ThreadBase> thread = mThread.promote(); 4356 if (thread != 0) { 4357 if (!isOutputTrack()) { 4358 if (mState == ACTIVE || mState == RESUMING) { 4359 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4360 4361#ifdef ADD_BATTERY_DATA 4362 // to track the speaker usage 4363 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4364#endif 4365 } 4366 AudioSystem::releaseOutput(thread->id()); 4367 } 4368 Mutex::Autolock _l(thread->mLock); 4369 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4370 playbackThread->destroyTrack_l(this); 4371 } 4372 } 4373} 4374 4375/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4376{ 4377 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4378 " Server User Main buf Aux Buf Flags Underruns\n"); 4379} 4380 4381void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4382{ 4383 uint32_t vlr = mCblk->getVolumeLR(); 4384 if (isFastTrack()) { 4385 sprintf(buffer, " F %2d", mFastIndex); 4386 } else { 4387 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4388 } 4389 track_state state = mState; 4390 char stateChar; 4391 switch (state) { 4392 case IDLE: 4393 stateChar = 'I'; 4394 break; 4395 case TERMINATED: 4396 stateChar = 'T'; 4397 break; 4398 case STOPPING_1: 4399 stateChar = 's'; 4400 break; 4401 case STOPPING_2: 4402 stateChar = '5'; 4403 break; 4404 case STOPPED: 4405 stateChar = 'S'; 4406 break; 4407 case RESUMING: 4408 stateChar = 'R'; 4409 break; 4410 case ACTIVE: 4411 stateChar = 'A'; 4412 break; 4413 case PAUSING: 4414 stateChar = 'p'; 4415 break; 4416 case PAUSED: 4417 stateChar = 'P'; 4418 break; 4419 case FLUSHED: 4420 stateChar = 'F'; 4421 break; 4422 default: 4423 stateChar = '?'; 4424 break; 4425 } 4426 char nowInUnderrun; 4427 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4428 case UNDERRUN_FULL: 4429 nowInUnderrun = ' '; 4430 break; 4431 case UNDERRUN_PARTIAL: 4432 nowInUnderrun = '<'; 4433 break; 4434 case UNDERRUN_EMPTY: 4435 nowInUnderrun = '*'; 4436 break; 4437 default: 4438 nowInUnderrun = '?'; 4439 break; 4440 } 4441 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4442 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4443 (mClient == 0) ? getpid_cached : mClient->pid(), 4444 mStreamType, 4445 mFormat, 4446 mChannelMask, 4447 mSessionId, 4448 mFrameCount, 4449 mCblk->frameCount, 4450 stateChar, 4451 mMute, 4452 mFillingUpStatus, 4453 mCblk->sampleRate, 4454 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4455 20.0 * log10((vlr >> 16) / 4096.0), 4456 mCblk->server, 4457 mCblk->user, 4458 (int)mMainBuffer, 4459 (int)mAuxBuffer, 4460 mCblk->flags, 4461 mUnderrunCount, 4462 nowInUnderrun); 4463} 4464 4465// AudioBufferProvider interface 4466status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4467 AudioBufferProvider::Buffer* buffer, int64_t pts) 4468{ 4469 audio_track_cblk_t* cblk = this->cblk(); 4470 uint32_t framesReady; 4471 uint32_t framesReq = buffer->frameCount; 4472 4473 // Check if last stepServer failed, try to step now 4474 if (mStepServerFailed) { 4475 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4476 // Since the fast mixer is higher priority than client callback thread, 4477 // it does not result in priority inversion for client. 4478 // But a non-blocking solution would be preferable to avoid 4479 // fast mixer being unable to tryLock(), and 4480 // to avoid the extra context switches if the client wakes up, 4481 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4482 if (!step()) goto getNextBuffer_exit; 4483 ALOGV("stepServer recovered"); 4484 mStepServerFailed = false; 4485 } 4486 4487 // FIXME Same as above 4488 framesReady = cblk->framesReady(); 4489 4490 if (CC_LIKELY(framesReady)) { 4491 uint32_t s = cblk->server; 4492 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4493 4494 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4495 if (framesReq > framesReady) { 4496 framesReq = framesReady; 4497 } 4498 if (framesReq > bufferEnd - s) { 4499 framesReq = bufferEnd - s; 4500 } 4501 4502 buffer->raw = getBuffer(s, framesReq); 4503 if (buffer->raw == NULL) goto getNextBuffer_exit; 4504 4505 buffer->frameCount = framesReq; 4506 return NO_ERROR; 4507 } 4508 4509getNextBuffer_exit: 4510 buffer->raw = NULL; 4511 buffer->frameCount = 0; 4512 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4513 return NOT_ENOUGH_DATA; 4514} 4515 4516// Note that framesReady() takes a mutex on the control block using tryLock(). 4517// This could result in priority inversion if framesReady() is called by the normal mixer, 4518// as the normal mixer thread runs at lower 4519// priority than the client's callback thread: there is a short window within framesReady() 4520// during which the normal mixer could be preempted, and the client callback would block. 4521// Another problem can occur if framesReady() is called by the fast mixer: 4522// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4523// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4524size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4525 return mCblk->framesReady(); 4526} 4527 4528// Don't call for fast tracks; the framesReady() could result in priority inversion 4529bool AudioFlinger::PlaybackThread::Track::isReady() const { 4530 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4531 4532 if (framesReady() >= mCblk->frameCount || 4533 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4534 mFillingUpStatus = FS_FILLED; 4535 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4536 return true; 4537 } 4538 return false; 4539} 4540 4541status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4542 int triggerSession) 4543{ 4544 status_t status = NO_ERROR; 4545 ALOGV("start(%d), calling pid %d session %d", 4546 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4547 4548 sp<ThreadBase> thread = mThread.promote(); 4549 if (thread != 0) { 4550 Mutex::Autolock _l(thread->mLock); 4551 track_state state = mState; 4552 // here the track could be either new, or restarted 4553 // in both cases "unstop" the track 4554 if (mState == PAUSED) { 4555 mState = TrackBase::RESUMING; 4556 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4557 } else { 4558 mState = TrackBase::ACTIVE; 4559 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4560 } 4561 4562 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4563 thread->mLock.unlock(); 4564 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4565 thread->mLock.lock(); 4566 4567#ifdef ADD_BATTERY_DATA 4568 // to track the speaker usage 4569 if (status == NO_ERROR) { 4570 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4571 } 4572#endif 4573 } 4574 if (status == NO_ERROR) { 4575 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4576 playbackThread->addTrack_l(this); 4577 } else { 4578 mState = state; 4579 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4580 } 4581 } else { 4582 status = BAD_VALUE; 4583 } 4584 return status; 4585} 4586 4587void AudioFlinger::PlaybackThread::Track::stop() 4588{ 4589 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4590 sp<ThreadBase> thread = mThread.promote(); 4591 if (thread != 0) { 4592 Mutex::Autolock _l(thread->mLock); 4593 track_state state = mState; 4594 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4595 // If the track is not active (PAUSED and buffers full), flush buffers 4596 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4597 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4598 reset(); 4599 mState = STOPPED; 4600 } else if (!isFastTrack()) { 4601 mState = STOPPED; 4602 } else { 4603 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4604 // and then to STOPPED and reset() when presentation is complete 4605 mState = STOPPING_1; 4606 } 4607 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4608 } 4609 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4610 thread->mLock.unlock(); 4611 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4612 thread->mLock.lock(); 4613 4614#ifdef ADD_BATTERY_DATA 4615 // to track the speaker usage 4616 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4617#endif 4618 } 4619 } 4620} 4621 4622void AudioFlinger::PlaybackThread::Track::pause() 4623{ 4624 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4625 sp<ThreadBase> thread = mThread.promote(); 4626 if (thread != 0) { 4627 Mutex::Autolock _l(thread->mLock); 4628 if (mState == ACTIVE || mState == RESUMING) { 4629 mState = PAUSING; 4630 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4631 if (!isOutputTrack()) { 4632 thread->mLock.unlock(); 4633 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4634 thread->mLock.lock(); 4635 4636#ifdef ADD_BATTERY_DATA 4637 // to track the speaker usage 4638 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4639#endif 4640 } 4641 } 4642 } 4643} 4644 4645void AudioFlinger::PlaybackThread::Track::flush() 4646{ 4647 ALOGV("flush(%d)", mName); 4648 sp<ThreadBase> thread = mThread.promote(); 4649 if (thread != 0) { 4650 Mutex::Autolock _l(thread->mLock); 4651 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4652 mState != PAUSING) { 4653 return; 4654 } 4655 // No point remaining in PAUSED state after a flush => go to 4656 // FLUSHED state 4657 mState = FLUSHED; 4658 // do not reset the track if it is still in the process of being stopped or paused. 4659 // this will be done by prepareTracks_l() when the track is stopped. 4660 // prepareTracks_l() will see mState == FLUSHED, then 4661 // remove from active track list, reset(), and trigger presentation complete 4662 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4663 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4664 reset(); 4665 } 4666 } 4667} 4668 4669void AudioFlinger::PlaybackThread::Track::reset() 4670{ 4671 // Do not reset twice to avoid discarding data written just after a flush and before 4672 // the audioflinger thread detects the track is stopped. 4673 if (!mResetDone) { 4674 TrackBase::reset(); 4675 // Force underrun condition to avoid false underrun callback until first data is 4676 // written to buffer 4677 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4678 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4679 mFillingUpStatus = FS_FILLING; 4680 mResetDone = true; 4681 if (mState == FLUSHED) { 4682 mState = IDLE; 4683 } 4684 } 4685} 4686 4687void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4688{ 4689 mMute = muted; 4690} 4691 4692status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4693{ 4694 status_t status = DEAD_OBJECT; 4695 sp<ThreadBase> thread = mThread.promote(); 4696 if (thread != 0) { 4697 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4698 status = playbackThread->attachAuxEffect(this, EffectId); 4699 } 4700 return status; 4701} 4702 4703void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4704{ 4705 mAuxEffectId = EffectId; 4706 mAuxBuffer = buffer; 4707} 4708 4709bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4710 size_t audioHalFrames) 4711{ 4712 // a track is considered presented when the total number of frames written to audio HAL 4713 // corresponds to the number of frames written when presentationComplete() is called for the 4714 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4715 if (mPresentationCompleteFrames == 0) { 4716 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4717 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4718 mPresentationCompleteFrames, audioHalFrames); 4719 } 4720 if (framesWritten >= mPresentationCompleteFrames) { 4721 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4722 mSessionId, framesWritten); 4723 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4724 return true; 4725 } 4726 return false; 4727} 4728 4729void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4730{ 4731 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4732 if (mSyncEvents[i]->type() == type) { 4733 mSyncEvents[i]->trigger(); 4734 mSyncEvents.removeAt(i); 4735 i--; 4736 } 4737 } 4738} 4739 4740// implement VolumeBufferProvider interface 4741 4742uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4743{ 4744 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4745 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4746 uint32_t vlr = mCblk->getVolumeLR(); 4747 uint32_t vl = vlr & 0xFFFF; 4748 uint32_t vr = vlr >> 16; 4749 // track volumes come from shared memory, so can't be trusted and must be clamped 4750 if (vl > MAX_GAIN_INT) { 4751 vl = MAX_GAIN_INT; 4752 } 4753 if (vr > MAX_GAIN_INT) { 4754 vr = MAX_GAIN_INT; 4755 } 4756 // now apply the cached master volume and stream type volume; 4757 // this is trusted but lacks any synchronization or barrier so may be stale 4758 float v = mCachedVolume; 4759 vl *= v; 4760 vr *= v; 4761 // re-combine into U4.16 4762 vlr = (vr << 16) | (vl & 0xFFFF); 4763 // FIXME look at mute, pause, and stop flags 4764 return vlr; 4765} 4766 4767status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4768{ 4769 if (mState == TERMINATED || mState == PAUSED || 4770 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4771 (mState == STOPPED)))) { 4772 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4773 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4774 event->cancel(); 4775 return INVALID_OPERATION; 4776 } 4777 TrackBase::setSyncEvent(event); 4778 return NO_ERROR; 4779} 4780 4781// timed audio tracks 4782 4783sp<AudioFlinger::PlaybackThread::TimedTrack> 4784AudioFlinger::PlaybackThread::TimedTrack::create( 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 if (!client->reserveTimedTrack()) 4795 return NULL; 4796 4797 return new TimedTrack( 4798 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4799 sharedBuffer, sessionId); 4800} 4801 4802AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4803 PlaybackThread *thread, 4804 const sp<Client>& client, 4805 audio_stream_type_t streamType, 4806 uint32_t sampleRate, 4807 audio_format_t format, 4808 uint32_t channelMask, 4809 int frameCount, 4810 const sp<IMemory>& sharedBuffer, 4811 int sessionId) 4812 : Track(thread, client, streamType, sampleRate, format, channelMask, 4813 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4814 mQueueHeadInFlight(false), 4815 mTrimQueueHeadOnRelease(false), 4816 mFramesPendingInQueue(0), 4817 mTimedSilenceBuffer(NULL), 4818 mTimedSilenceBufferSize(0), 4819 mTimedAudioOutputOnTime(false), 4820 mMediaTimeTransformValid(false) 4821{ 4822 LocalClock lc; 4823 mLocalTimeFreq = lc.getLocalFreq(); 4824 4825 mLocalTimeToSampleTransform.a_zero = 0; 4826 mLocalTimeToSampleTransform.b_zero = 0; 4827 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4828 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4829 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4830 &mLocalTimeToSampleTransform.a_to_b_denom); 4831 4832 mMediaTimeToSampleTransform.a_zero = 0; 4833 mMediaTimeToSampleTransform.b_zero = 0; 4834 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4835 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4836 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4837 &mMediaTimeToSampleTransform.a_to_b_denom); 4838} 4839 4840AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4841 mClient->releaseTimedTrack(); 4842 delete [] mTimedSilenceBuffer; 4843} 4844 4845status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4846 size_t size, sp<IMemory>* buffer) { 4847 4848 Mutex::Autolock _l(mTimedBufferQueueLock); 4849 4850 trimTimedBufferQueue_l(); 4851 4852 // lazily initialize the shared memory heap for timed buffers 4853 if (mTimedMemoryDealer == NULL) { 4854 const int kTimedBufferHeapSize = 512 << 10; 4855 4856 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4857 "AudioFlingerTimed"); 4858 if (mTimedMemoryDealer == NULL) 4859 return NO_MEMORY; 4860 } 4861 4862 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4863 if (newBuffer == NULL) { 4864 newBuffer = mTimedMemoryDealer->allocate(size); 4865 if (newBuffer == NULL) 4866 return NO_MEMORY; 4867 } 4868 4869 *buffer = newBuffer; 4870 return NO_ERROR; 4871} 4872 4873// caller must hold mTimedBufferQueueLock 4874void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4875 int64_t mediaTimeNow; 4876 { 4877 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4878 if (!mMediaTimeTransformValid) 4879 return; 4880 4881 int64_t targetTimeNow; 4882 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4883 ? mCCHelper.getCommonTime(&targetTimeNow) 4884 : mCCHelper.getLocalTime(&targetTimeNow); 4885 4886 if (OK != res) 4887 return; 4888 4889 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4890 &mediaTimeNow)) { 4891 return; 4892 } 4893 } 4894 4895 size_t trimEnd; 4896 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4897 int64_t bufEnd; 4898 4899 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4900 // We have a next buffer. Just use its PTS as the PTS of the frame 4901 // following the last frame in this buffer. If the stream is sparse 4902 // (ie, there are deliberate gaps left in the stream which should be 4903 // filled with silence by the TimedAudioTrack), then this can result 4904 // in one extra buffer being left un-trimmed when it could have 4905 // been. In general, this is not typical, and we would rather 4906 // optimized away the TS calculation below for the more common case 4907 // where PTSes are contiguous. 4908 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4909 } else { 4910 // We have no next buffer. Compute the PTS of the frame following 4911 // the last frame in this buffer by computing the duration of of 4912 // this frame in media time units and adding it to the PTS of the 4913 // buffer. 4914 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4915 / mCblk->frameSize; 4916 4917 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4918 &bufEnd)) { 4919 ALOGE("Failed to convert frame count of %lld to media time" 4920 " duration" " (scale factor %d/%u) in %s", 4921 frameCount, 4922 mMediaTimeToSampleTransform.a_to_b_numer, 4923 mMediaTimeToSampleTransform.a_to_b_denom, 4924 __PRETTY_FUNCTION__); 4925 break; 4926 } 4927 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4928 } 4929 4930 if (bufEnd > mediaTimeNow) 4931 break; 4932 4933 // Is the buffer we want to use in the middle of a mix operation right 4934 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4935 // from the mixer which should be coming back shortly. 4936 if (!trimEnd && mQueueHeadInFlight) { 4937 mTrimQueueHeadOnRelease = true; 4938 } 4939 } 4940 4941 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4942 if (trimStart < trimEnd) { 4943 // Update the bookkeeping for framesReady() 4944 for (size_t i = trimStart; i < trimEnd; ++i) { 4945 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4946 } 4947 4948 // Now actually remove the buffers from the queue. 4949 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4950 } 4951} 4952 4953void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4954 const char* logTag) { 4955 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4956 "%s called (reason \"%s\"), but timed buffer queue has no" 4957 " elements to trim.", __FUNCTION__, logTag); 4958 4959 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4960 mTimedBufferQueue.removeAt(0); 4961} 4962 4963void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4964 const TimedBuffer& buf, 4965 const char* logTag) { 4966 uint32_t bufBytes = buf.buffer()->size(); 4967 uint32_t consumedAlready = buf.position(); 4968 4969 ALOG_ASSERT(consumedAlready <= bufBytes, 4970 "Bad bookkeeping while updating frames pending. Timed buffer is" 4971 " only %u bytes long, but claims to have consumed %u" 4972 " bytes. (update reason: \"%s\")", 4973 bufBytes, consumedAlready, logTag); 4974 4975 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4976 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4977 "Bad bookkeeping while updating frames pending. Should have at" 4978 " least %u queued frames, but we think we have only %u. (update" 4979 " reason: \"%s\")", 4980 bufFrames, mFramesPendingInQueue, logTag); 4981 4982 mFramesPendingInQueue -= bufFrames; 4983} 4984 4985status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4986 const sp<IMemory>& buffer, int64_t pts) { 4987 4988 { 4989 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4990 if (!mMediaTimeTransformValid) 4991 return INVALID_OPERATION; 4992 } 4993 4994 Mutex::Autolock _l(mTimedBufferQueueLock); 4995 4996 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4997 mFramesPendingInQueue += bufFrames; 4998 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4999 5000 return NO_ERROR; 5001} 5002 5003status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5004 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5005 5006 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5007 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5008 target); 5009 5010 if (!(target == TimedAudioTrack::LOCAL_TIME || 5011 target == TimedAudioTrack::COMMON_TIME)) { 5012 return BAD_VALUE; 5013 } 5014 5015 Mutex::Autolock lock(mMediaTimeTransformLock); 5016 mMediaTimeTransform = xform; 5017 mMediaTimeTransformTarget = target; 5018 mMediaTimeTransformValid = true; 5019 5020 return NO_ERROR; 5021} 5022 5023#define min(a, b) ((a) < (b) ? (a) : (b)) 5024 5025// implementation of getNextBuffer for tracks whose buffers have timestamps 5026status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5027 AudioBufferProvider::Buffer* buffer, int64_t pts) 5028{ 5029 if (pts == AudioBufferProvider::kInvalidPTS) { 5030 buffer->raw = 0; 5031 buffer->frameCount = 0; 5032 mTimedAudioOutputOnTime = false; 5033 return INVALID_OPERATION; 5034 } 5035 5036 Mutex::Autolock _l(mTimedBufferQueueLock); 5037 5038 ALOG_ASSERT(!mQueueHeadInFlight, 5039 "getNextBuffer called without releaseBuffer!"); 5040 5041 while (true) { 5042 5043 // if we have no timed buffers, then fail 5044 if (mTimedBufferQueue.isEmpty()) { 5045 buffer->raw = 0; 5046 buffer->frameCount = 0; 5047 return NOT_ENOUGH_DATA; 5048 } 5049 5050 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5051 5052 // calculate the PTS of the head of the timed buffer queue expressed in 5053 // local time 5054 int64_t headLocalPTS; 5055 { 5056 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5057 5058 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5059 5060 if (mMediaTimeTransform.a_to_b_denom == 0) { 5061 // the transform represents a pause, so yield silence 5062 timedYieldSilence_l(buffer->frameCount, buffer); 5063 return NO_ERROR; 5064 } 5065 5066 int64_t transformedPTS; 5067 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5068 &transformedPTS)) { 5069 // the transform failed. this shouldn't happen, but if it does 5070 // then just drop this buffer 5071 ALOGW("timedGetNextBuffer transform failed"); 5072 buffer->raw = 0; 5073 buffer->frameCount = 0; 5074 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5075 return NO_ERROR; 5076 } 5077 5078 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5079 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5080 &headLocalPTS)) { 5081 buffer->raw = 0; 5082 buffer->frameCount = 0; 5083 return INVALID_OPERATION; 5084 } 5085 } else { 5086 headLocalPTS = transformedPTS; 5087 } 5088 } 5089 5090 // adjust the head buffer's PTS to reflect the portion of the head buffer 5091 // that has already been consumed 5092 int64_t effectivePTS = headLocalPTS + 5093 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5094 5095 // Calculate the delta in samples between the head of the input buffer 5096 // queue and the start of the next output buffer that will be written. 5097 // If the transformation fails because of over or underflow, it means 5098 // that the sample's position in the output stream is so far out of 5099 // whack that it should just be dropped. 5100 int64_t sampleDelta; 5101 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5102 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5103 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5104 " mix"); 5105 continue; 5106 } 5107 if (!mLocalTimeToSampleTransform.doForwardTransform( 5108 (effectivePTS - pts) << 32, &sampleDelta)) { 5109 ALOGV("*** too late during sample rate transform: dropped buffer"); 5110 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5111 continue; 5112 } 5113 5114 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5115 " sampleDelta=[%d.%08x]", 5116 head.pts(), head.position(), pts, 5117 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5118 + (sampleDelta >> 32)), 5119 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5120 5121 // if the delta between the ideal placement for the next input sample and 5122 // the current output position is within this threshold, then we will 5123 // concatenate the next input samples to the previous output 5124 const int64_t kSampleContinuityThreshold = 5125 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5126 5127 // if this is the first buffer of audio that we're emitting from this track 5128 // then it should be almost exactly on time. 5129 const int64_t kSampleStartupThreshold = 1LL << 32; 5130 5131 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5132 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5133 // the next input is close enough to being on time, so concatenate it 5134 // with the last output 5135 timedYieldSamples_l(buffer); 5136 5137 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5138 head.position(), buffer->frameCount); 5139 return NO_ERROR; 5140 } 5141 5142 // Looks like our output is not on time. Reset our on timed status. 5143 // Next time we mix samples from our input queue, then should be within 5144 // the StartupThreshold. 5145 mTimedAudioOutputOnTime = false; 5146 if (sampleDelta > 0) { 5147 // the gap between the current output position and the proper start of 5148 // the next input sample is too big, so fill it with silence 5149 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5150 5151 timedYieldSilence_l(framesUntilNextInput, buffer); 5152 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5153 return NO_ERROR; 5154 } else { 5155 // the next input sample is late 5156 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5157 size_t onTimeSamplePosition = 5158 head.position() + lateFrames * mCblk->frameSize; 5159 5160 if (onTimeSamplePosition > head.buffer()->size()) { 5161 // all the remaining samples in the head are too late, so 5162 // drop it and move on 5163 ALOGV("*** too late: dropped buffer"); 5164 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5165 continue; 5166 } else { 5167 // skip over the late samples 5168 head.setPosition(onTimeSamplePosition); 5169 5170 // yield the available samples 5171 timedYieldSamples_l(buffer); 5172 5173 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5174 return NO_ERROR; 5175 } 5176 } 5177 } 5178} 5179 5180// Yield samples from the timed buffer queue head up to the given output 5181// buffer's capacity. 5182// 5183// Caller must hold mTimedBufferQueueLock 5184void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5185 AudioBufferProvider::Buffer* buffer) { 5186 5187 const TimedBuffer& head = mTimedBufferQueue[0]; 5188 5189 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5190 head.position()); 5191 5192 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5193 mCblk->frameSize); 5194 size_t framesRequested = buffer->frameCount; 5195 buffer->frameCount = min(framesLeftInHead, framesRequested); 5196 5197 mQueueHeadInFlight = true; 5198 mTimedAudioOutputOnTime = true; 5199} 5200 5201// Yield samples of silence up to the given output buffer's capacity 5202// 5203// Caller must hold mTimedBufferQueueLock 5204void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5205 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5206 5207 // lazily allocate a buffer filled with silence 5208 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5209 delete [] mTimedSilenceBuffer; 5210 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5211 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5212 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5213 } 5214 5215 buffer->raw = mTimedSilenceBuffer; 5216 size_t framesRequested = buffer->frameCount; 5217 buffer->frameCount = min(numFrames, framesRequested); 5218 5219 mTimedAudioOutputOnTime = false; 5220} 5221 5222// AudioBufferProvider interface 5223void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5224 AudioBufferProvider::Buffer* buffer) { 5225 5226 Mutex::Autolock _l(mTimedBufferQueueLock); 5227 5228 // If the buffer which was just released is part of the buffer at the head 5229 // of the queue, be sure to update the amt of the buffer which has been 5230 // consumed. If the buffer being returned is not part of the head of the 5231 // queue, its either because the buffer is part of the silence buffer, or 5232 // because the head of the timed queue was trimmed after the mixer called 5233 // getNextBuffer but before the mixer called releaseBuffer. 5234 if (buffer->raw == mTimedSilenceBuffer) { 5235 ALOG_ASSERT(!mQueueHeadInFlight, 5236 "Queue head in flight during release of silence buffer!"); 5237 goto done; 5238 } 5239 5240 ALOG_ASSERT(mQueueHeadInFlight, 5241 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5242 " head in flight."); 5243 5244 if (mTimedBufferQueue.size()) { 5245 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5246 5247 void* start = head.buffer()->pointer(); 5248 void* end = reinterpret_cast<void*>( 5249 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5250 + head.buffer()->size()); 5251 5252 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5253 "released buffer not within the head of the timed buffer" 5254 " queue; qHead = [%p, %p], released buffer = %p", 5255 start, end, buffer->raw); 5256 5257 head.setPosition(head.position() + 5258 (buffer->frameCount * mCblk->frameSize)); 5259 mQueueHeadInFlight = false; 5260 5261 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5262 "Bad bookkeeping during releaseBuffer! Should have at" 5263 " least %u queued frames, but we think we have only %u", 5264 buffer->frameCount, mFramesPendingInQueue); 5265 5266 mFramesPendingInQueue -= buffer->frameCount; 5267 5268 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5269 || mTrimQueueHeadOnRelease) { 5270 trimTimedBufferQueueHead_l("releaseBuffer"); 5271 mTrimQueueHeadOnRelease = false; 5272 } 5273 } else { 5274 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5275 " buffers in the timed buffer queue"); 5276 } 5277 5278done: 5279 buffer->raw = 0; 5280 buffer->frameCount = 0; 5281} 5282 5283size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5284 Mutex::Autolock _l(mTimedBufferQueueLock); 5285 return mFramesPendingInQueue; 5286} 5287 5288AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5289 : mPTS(0), mPosition(0) {} 5290 5291AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5292 const sp<IMemory>& buffer, int64_t pts) 5293 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5294 5295// ---------------------------------------------------------------------------- 5296 5297// RecordTrack constructor must be called with AudioFlinger::mLock held 5298AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5299 RecordThread *thread, 5300 const sp<Client>& client, 5301 uint32_t sampleRate, 5302 audio_format_t format, 5303 uint32_t channelMask, 5304 int frameCount, 5305 int sessionId) 5306 : TrackBase(thread, client, sampleRate, format, 5307 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5308 mOverflow(false) 5309{ 5310 if (mCblk != NULL) { 5311 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5312 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5313 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5314 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5315 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5316 } else { 5317 mCblk->frameSize = sizeof(int8_t); 5318 } 5319 } 5320} 5321 5322AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5323{ 5324 sp<ThreadBase> thread = mThread.promote(); 5325 if (thread != 0) { 5326 AudioSystem::releaseInput(thread->id()); 5327 } 5328} 5329 5330// AudioBufferProvider interface 5331status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5332{ 5333 audio_track_cblk_t* cblk = this->cblk(); 5334 uint32_t framesAvail; 5335 uint32_t framesReq = buffer->frameCount; 5336 5337 // Check if last stepServer failed, try to step now 5338 if (mStepServerFailed) { 5339 if (!step()) goto getNextBuffer_exit; 5340 ALOGV("stepServer recovered"); 5341 mStepServerFailed = false; 5342 } 5343 5344 framesAvail = cblk->framesAvailable_l(); 5345 5346 if (CC_LIKELY(framesAvail)) { 5347 uint32_t s = cblk->server; 5348 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5349 5350 if (framesReq > framesAvail) { 5351 framesReq = framesAvail; 5352 } 5353 if (framesReq > bufferEnd - s) { 5354 framesReq = bufferEnd - s; 5355 } 5356 5357 buffer->raw = getBuffer(s, framesReq); 5358 if (buffer->raw == NULL) goto getNextBuffer_exit; 5359 5360 buffer->frameCount = framesReq; 5361 return NO_ERROR; 5362 } 5363 5364getNextBuffer_exit: 5365 buffer->raw = NULL; 5366 buffer->frameCount = 0; 5367 return NOT_ENOUGH_DATA; 5368} 5369 5370status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5371 int triggerSession) 5372{ 5373 sp<ThreadBase> thread = mThread.promote(); 5374 if (thread != 0) { 5375 RecordThread *recordThread = (RecordThread *)thread.get(); 5376 return recordThread->start(this, event, triggerSession); 5377 } else { 5378 return BAD_VALUE; 5379 } 5380} 5381 5382void AudioFlinger::RecordThread::RecordTrack::stop() 5383{ 5384 sp<ThreadBase> thread = mThread.promote(); 5385 if (thread != 0) { 5386 RecordThread *recordThread = (RecordThread *)thread.get(); 5387 recordThread->stop(this); 5388 TrackBase::reset(); 5389 // Force overrun condition to avoid false overrun callback until first data is 5390 // read from buffer 5391 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5392 } 5393} 5394 5395void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5396{ 5397 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5398 (mClient == 0) ? getpid_cached : mClient->pid(), 5399 mFormat, 5400 mChannelMask, 5401 mSessionId, 5402 mFrameCount, 5403 mState, 5404 mCblk->sampleRate, 5405 mCblk->server, 5406 mCblk->user); 5407} 5408 5409 5410// ---------------------------------------------------------------------------- 5411 5412AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5413 PlaybackThread *playbackThread, 5414 DuplicatingThread *sourceThread, 5415 uint32_t sampleRate, 5416 audio_format_t format, 5417 uint32_t channelMask, 5418 int frameCount) 5419 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5420 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5421 mActive(false), mSourceThread(sourceThread) 5422{ 5423 5424 if (mCblk != NULL) { 5425 mCblk->flags |= CBLK_DIRECTION_OUT; 5426 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5427 mOutBuffer.frameCount = 0; 5428 playbackThread->mTracks.add(this); 5429 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5430 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5431 mCblk, mBuffer, mCblk->buffers, 5432 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5433 } else { 5434 ALOGW("Error creating output track on thread %p", playbackThread); 5435 } 5436} 5437 5438AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5439{ 5440 clearBufferQueue(); 5441} 5442 5443status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5444 int triggerSession) 5445{ 5446 status_t status = Track::start(event, triggerSession); 5447 if (status != NO_ERROR) { 5448 return status; 5449 } 5450 5451 mActive = true; 5452 mRetryCount = 127; 5453 return status; 5454} 5455 5456void AudioFlinger::PlaybackThread::OutputTrack::stop() 5457{ 5458 Track::stop(); 5459 clearBufferQueue(); 5460 mOutBuffer.frameCount = 0; 5461 mActive = false; 5462} 5463 5464bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5465{ 5466 Buffer *pInBuffer; 5467 Buffer inBuffer; 5468 uint32_t channelCount = mChannelCount; 5469 bool outputBufferFull = false; 5470 inBuffer.frameCount = frames; 5471 inBuffer.i16 = data; 5472 5473 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5474 5475 if (!mActive && frames != 0) { 5476 start(); 5477 sp<ThreadBase> thread = mThread.promote(); 5478 if (thread != 0) { 5479 MixerThread *mixerThread = (MixerThread *)thread.get(); 5480 if (mCblk->frameCount > frames){ 5481 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5482 uint32_t startFrames = (mCblk->frameCount - frames); 5483 pInBuffer = new Buffer; 5484 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5485 pInBuffer->frameCount = startFrames; 5486 pInBuffer->i16 = pInBuffer->mBuffer; 5487 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5488 mBufferQueue.add(pInBuffer); 5489 } else { 5490 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5491 } 5492 } 5493 } 5494 } 5495 5496 while (waitTimeLeftMs) { 5497 // First write pending buffers, then new data 5498 if (mBufferQueue.size()) { 5499 pInBuffer = mBufferQueue.itemAt(0); 5500 } else { 5501 pInBuffer = &inBuffer; 5502 } 5503 5504 if (pInBuffer->frameCount == 0) { 5505 break; 5506 } 5507 5508 if (mOutBuffer.frameCount == 0) { 5509 mOutBuffer.frameCount = pInBuffer->frameCount; 5510 nsecs_t startTime = systemTime(); 5511 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5512 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5513 outputBufferFull = true; 5514 break; 5515 } 5516 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5517 if (waitTimeLeftMs >= waitTimeMs) { 5518 waitTimeLeftMs -= waitTimeMs; 5519 } else { 5520 waitTimeLeftMs = 0; 5521 } 5522 } 5523 5524 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5525 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5526 mCblk->stepUser(outFrames); 5527 pInBuffer->frameCount -= outFrames; 5528 pInBuffer->i16 += outFrames * channelCount; 5529 mOutBuffer.frameCount -= outFrames; 5530 mOutBuffer.i16 += outFrames * channelCount; 5531 5532 if (pInBuffer->frameCount == 0) { 5533 if (mBufferQueue.size()) { 5534 mBufferQueue.removeAt(0); 5535 delete [] pInBuffer->mBuffer; 5536 delete pInBuffer; 5537 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5538 } else { 5539 break; 5540 } 5541 } 5542 } 5543 5544 // If we could not write all frames, allocate a buffer and queue it for next time. 5545 if (inBuffer.frameCount) { 5546 sp<ThreadBase> thread = mThread.promote(); 5547 if (thread != 0 && !thread->standby()) { 5548 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5549 pInBuffer = new Buffer; 5550 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5551 pInBuffer->frameCount = inBuffer.frameCount; 5552 pInBuffer->i16 = pInBuffer->mBuffer; 5553 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5554 mBufferQueue.add(pInBuffer); 5555 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5556 } else { 5557 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5558 } 5559 } 5560 } 5561 5562 // Calling write() with a 0 length buffer, means that no more data will be written: 5563 // If no more buffers are pending, fill output track buffer to make sure it is started 5564 // by output mixer. 5565 if (frames == 0 && mBufferQueue.size() == 0) { 5566 if (mCblk->user < mCblk->frameCount) { 5567 frames = mCblk->frameCount - mCblk->user; 5568 pInBuffer = new Buffer; 5569 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5570 pInBuffer->frameCount = frames; 5571 pInBuffer->i16 = pInBuffer->mBuffer; 5572 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5573 mBufferQueue.add(pInBuffer); 5574 } else if (mActive) { 5575 stop(); 5576 } 5577 } 5578 5579 return outputBufferFull; 5580} 5581 5582status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5583{ 5584 int active; 5585 status_t result; 5586 audio_track_cblk_t* cblk = mCblk; 5587 uint32_t framesReq = buffer->frameCount; 5588 5589// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5590 buffer->frameCount = 0; 5591 5592 uint32_t framesAvail = cblk->framesAvailable(); 5593 5594 5595 if (framesAvail == 0) { 5596 Mutex::Autolock _l(cblk->lock); 5597 goto start_loop_here; 5598 while (framesAvail == 0) { 5599 active = mActive; 5600 if (CC_UNLIKELY(!active)) { 5601 ALOGV("Not active and NO_MORE_BUFFERS"); 5602 return NO_MORE_BUFFERS; 5603 } 5604 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5605 if (result != NO_ERROR) { 5606 return NO_MORE_BUFFERS; 5607 } 5608 // read the server count again 5609 start_loop_here: 5610 framesAvail = cblk->framesAvailable_l(); 5611 } 5612 } 5613 5614// if (framesAvail < framesReq) { 5615// return NO_MORE_BUFFERS; 5616// } 5617 5618 if (framesReq > framesAvail) { 5619 framesReq = framesAvail; 5620 } 5621 5622 uint32_t u = cblk->user; 5623 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5624 5625 if (framesReq > bufferEnd - u) { 5626 framesReq = bufferEnd - u; 5627 } 5628 5629 buffer->frameCount = framesReq; 5630 buffer->raw = (void *)cblk->buffer(u); 5631 return NO_ERROR; 5632} 5633 5634 5635void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5636{ 5637 size_t size = mBufferQueue.size(); 5638 5639 for (size_t i = 0; i < size; i++) { 5640 Buffer *pBuffer = mBufferQueue.itemAt(i); 5641 delete [] pBuffer->mBuffer; 5642 delete pBuffer; 5643 } 5644 mBufferQueue.clear(); 5645} 5646 5647// ---------------------------------------------------------------------------- 5648 5649AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5650 : RefBase(), 5651 mAudioFlinger(audioFlinger), 5652 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5653 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5654 mPid(pid), 5655 mTimedTrackCount(0) 5656{ 5657 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5658} 5659 5660// Client destructor must be called with AudioFlinger::mLock held 5661AudioFlinger::Client::~Client() 5662{ 5663 mAudioFlinger->removeClient_l(mPid); 5664} 5665 5666sp<MemoryDealer> AudioFlinger::Client::heap() const 5667{ 5668 return mMemoryDealer; 5669} 5670 5671// Reserve one of the limited slots for a timed audio track associated 5672// with this client 5673bool AudioFlinger::Client::reserveTimedTrack() 5674{ 5675 const int kMaxTimedTracksPerClient = 4; 5676 5677 Mutex::Autolock _l(mTimedTrackLock); 5678 5679 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5680 ALOGW("can not create timed track - pid %d has exceeded the limit", 5681 mPid); 5682 return false; 5683 } 5684 5685 mTimedTrackCount++; 5686 return true; 5687} 5688 5689// Release a slot for a timed audio track 5690void AudioFlinger::Client::releaseTimedTrack() 5691{ 5692 Mutex::Autolock _l(mTimedTrackLock); 5693 mTimedTrackCount--; 5694} 5695 5696// ---------------------------------------------------------------------------- 5697 5698AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5699 const sp<IAudioFlingerClient>& client, 5700 pid_t pid) 5701 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5702{ 5703} 5704 5705AudioFlinger::NotificationClient::~NotificationClient() 5706{ 5707} 5708 5709void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5710{ 5711 sp<NotificationClient> keep(this); 5712 mAudioFlinger->removeNotificationClient(mPid); 5713} 5714 5715// ---------------------------------------------------------------------------- 5716 5717AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5718 : BnAudioTrack(), 5719 mTrack(track) 5720{ 5721} 5722 5723AudioFlinger::TrackHandle::~TrackHandle() { 5724 // just stop the track on deletion, associated resources 5725 // will be freed from the main thread once all pending buffers have 5726 // been played. Unless it's not in the active track list, in which 5727 // case we free everything now... 5728 mTrack->destroy(); 5729} 5730 5731sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5732 return mTrack->getCblk(); 5733} 5734 5735status_t AudioFlinger::TrackHandle::start() { 5736 return mTrack->start(); 5737} 5738 5739void AudioFlinger::TrackHandle::stop() { 5740 mTrack->stop(); 5741} 5742 5743void AudioFlinger::TrackHandle::flush() { 5744 mTrack->flush(); 5745} 5746 5747void AudioFlinger::TrackHandle::mute(bool e) { 5748 mTrack->mute(e); 5749} 5750 5751void AudioFlinger::TrackHandle::pause() { 5752 mTrack->pause(); 5753} 5754 5755status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5756{ 5757 return mTrack->attachAuxEffect(EffectId); 5758} 5759 5760status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5761 sp<IMemory>* buffer) { 5762 if (!mTrack->isTimedTrack()) 5763 return INVALID_OPERATION; 5764 5765 PlaybackThread::TimedTrack* tt = 5766 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5767 return tt->allocateTimedBuffer(size, buffer); 5768} 5769 5770status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5771 int64_t pts) { 5772 if (!mTrack->isTimedTrack()) 5773 return INVALID_OPERATION; 5774 5775 PlaybackThread::TimedTrack* tt = 5776 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5777 return tt->queueTimedBuffer(buffer, pts); 5778} 5779 5780status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5781 const LinearTransform& xform, int target) { 5782 5783 if (!mTrack->isTimedTrack()) 5784 return INVALID_OPERATION; 5785 5786 PlaybackThread::TimedTrack* tt = 5787 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5788 return tt->setMediaTimeTransform( 5789 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5790} 5791 5792status_t AudioFlinger::TrackHandle::onTransact( 5793 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5794{ 5795 return BnAudioTrack::onTransact(code, data, reply, flags); 5796} 5797 5798// ---------------------------------------------------------------------------- 5799 5800sp<IAudioRecord> AudioFlinger::openRecord( 5801 pid_t pid, 5802 audio_io_handle_t input, 5803 uint32_t sampleRate, 5804 audio_format_t format, 5805 uint32_t channelMask, 5806 int frameCount, 5807 IAudioFlinger::track_flags_t flags, 5808 int *sessionId, 5809 status_t *status) 5810{ 5811 sp<RecordThread::RecordTrack> recordTrack; 5812 sp<RecordHandle> recordHandle; 5813 sp<Client> client; 5814 status_t lStatus; 5815 RecordThread *thread; 5816 size_t inFrameCount; 5817 int lSessionId; 5818 5819 // check calling permissions 5820 if (!recordingAllowed()) { 5821 lStatus = PERMISSION_DENIED; 5822 goto Exit; 5823 } 5824 5825 // add client to list 5826 { // scope for mLock 5827 Mutex::Autolock _l(mLock); 5828 thread = checkRecordThread_l(input); 5829 if (thread == NULL) { 5830 lStatus = BAD_VALUE; 5831 goto Exit; 5832 } 5833 5834 client = registerPid_l(pid); 5835 5836 // If no audio session id is provided, create one here 5837 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5838 lSessionId = *sessionId; 5839 } else { 5840 lSessionId = nextUniqueId(); 5841 if (sessionId != NULL) { 5842 *sessionId = lSessionId; 5843 } 5844 } 5845 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5846 recordTrack = thread->createRecordTrack_l(client, 5847 sampleRate, 5848 format, 5849 channelMask, 5850 frameCount, 5851 lSessionId, 5852 &lStatus); 5853 } 5854 if (lStatus != NO_ERROR) { 5855 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5856 // destructor is called by the TrackBase destructor with mLock held 5857 client.clear(); 5858 recordTrack.clear(); 5859 goto Exit; 5860 } 5861 5862 // return to handle to client 5863 recordHandle = new RecordHandle(recordTrack); 5864 lStatus = NO_ERROR; 5865 5866Exit: 5867 if (status) { 5868 *status = lStatus; 5869 } 5870 return recordHandle; 5871} 5872 5873// ---------------------------------------------------------------------------- 5874 5875AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5876 : BnAudioRecord(), 5877 mRecordTrack(recordTrack) 5878{ 5879} 5880 5881AudioFlinger::RecordHandle::~RecordHandle() { 5882 stop(); 5883} 5884 5885sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5886 return mRecordTrack->getCblk(); 5887} 5888 5889status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5890 ALOGV("RecordHandle::start()"); 5891 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5892} 5893 5894void AudioFlinger::RecordHandle::stop() { 5895 ALOGV("RecordHandle::stop()"); 5896 mRecordTrack->stop(); 5897} 5898 5899status_t AudioFlinger::RecordHandle::onTransact( 5900 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5901{ 5902 return BnAudioRecord::onTransact(code, data, reply, flags); 5903} 5904 5905// ---------------------------------------------------------------------------- 5906 5907AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5908 AudioStreamIn *input, 5909 uint32_t sampleRate, 5910 uint32_t channels, 5911 audio_io_handle_t id, 5912 uint32_t device) : 5913 ThreadBase(audioFlinger, id, device, RECORD), 5914 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5915 // mRsmpInIndex and mInputBytes set by readInputParameters() 5916 mReqChannelCount(popcount(channels)), 5917 mReqSampleRate(sampleRate) 5918 // mBytesRead is only meaningful while active, and so is cleared in start() 5919 // (but might be better to also clear here for dump?) 5920{ 5921 snprintf(mName, kNameLength, "AudioIn_%X", id); 5922 5923 readInputParameters(); 5924} 5925 5926 5927AudioFlinger::RecordThread::~RecordThread() 5928{ 5929 delete[] mRsmpInBuffer; 5930 delete mResampler; 5931 delete[] mRsmpOutBuffer; 5932} 5933 5934void AudioFlinger::RecordThread::onFirstRef() 5935{ 5936 run(mName, PRIORITY_URGENT_AUDIO); 5937} 5938 5939status_t AudioFlinger::RecordThread::readyToRun() 5940{ 5941 status_t status = initCheck(); 5942 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5943 return status; 5944} 5945 5946bool AudioFlinger::RecordThread::threadLoop() 5947{ 5948 AudioBufferProvider::Buffer buffer; 5949 sp<RecordTrack> activeTrack; 5950 Vector< sp<EffectChain> > effectChains; 5951 5952 nsecs_t lastWarning = 0; 5953 5954 acquireWakeLock(); 5955 5956 // start recording 5957 while (!exitPending()) { 5958 5959 processConfigEvents(); 5960 5961 { // scope for mLock 5962 Mutex::Autolock _l(mLock); 5963 checkForNewParameters_l(); 5964 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5965 if (!mStandby) { 5966 mInput->stream->common.standby(&mInput->stream->common); 5967 mStandby = true; 5968 } 5969 5970 if (exitPending()) break; 5971 5972 releaseWakeLock_l(); 5973 ALOGV("RecordThread: loop stopping"); 5974 // go to sleep 5975 mWaitWorkCV.wait(mLock); 5976 ALOGV("RecordThread: loop starting"); 5977 acquireWakeLock_l(); 5978 continue; 5979 } 5980 if (mActiveTrack != 0) { 5981 if (mActiveTrack->mState == TrackBase::PAUSING) { 5982 if (!mStandby) { 5983 mInput->stream->common.standby(&mInput->stream->common); 5984 mStandby = true; 5985 } 5986 mActiveTrack.clear(); 5987 mStartStopCond.broadcast(); 5988 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5989 if (mReqChannelCount != mActiveTrack->channelCount()) { 5990 mActiveTrack.clear(); 5991 mStartStopCond.broadcast(); 5992 } else if (mBytesRead != 0) { 5993 // record start succeeds only if first read from audio input 5994 // succeeds 5995 if (mBytesRead > 0) { 5996 mActiveTrack->mState = TrackBase::ACTIVE; 5997 } else { 5998 mActiveTrack.clear(); 5999 } 6000 mStartStopCond.broadcast(); 6001 } 6002 mStandby = false; 6003 } 6004 } 6005 lockEffectChains_l(effectChains); 6006 } 6007 6008 if (mActiveTrack != 0) { 6009 if (mActiveTrack->mState != TrackBase::ACTIVE && 6010 mActiveTrack->mState != TrackBase::RESUMING) { 6011 unlockEffectChains(effectChains); 6012 usleep(kRecordThreadSleepUs); 6013 continue; 6014 } 6015 for (size_t i = 0; i < effectChains.size(); i ++) { 6016 effectChains[i]->process_l(); 6017 } 6018 6019 buffer.frameCount = mFrameCount; 6020 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6021 size_t framesOut = buffer.frameCount; 6022 if (mResampler == NULL) { 6023 // no resampling 6024 while (framesOut) { 6025 size_t framesIn = mFrameCount - mRsmpInIndex; 6026 if (framesIn) { 6027 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6028 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6029 if (framesIn > framesOut) 6030 framesIn = framesOut; 6031 mRsmpInIndex += framesIn; 6032 framesOut -= framesIn; 6033 if ((int)mChannelCount == mReqChannelCount || 6034 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6035 memcpy(dst, src, framesIn * mFrameSize); 6036 } else { 6037 int16_t *src16 = (int16_t *)src; 6038 int16_t *dst16 = (int16_t *)dst; 6039 if (mChannelCount == 1) { 6040 while (framesIn--) { 6041 *dst16++ = *src16; 6042 *dst16++ = *src16++; 6043 } 6044 } else { 6045 while (framesIn--) { 6046 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6047 src16 += 2; 6048 } 6049 } 6050 } 6051 } 6052 if (framesOut && mFrameCount == mRsmpInIndex) { 6053 if (framesOut == mFrameCount && 6054 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6055 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6056 framesOut = 0; 6057 } else { 6058 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6059 mRsmpInIndex = 0; 6060 } 6061 if (mBytesRead < 0) { 6062 ALOGE("Error reading audio input"); 6063 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6064 // Force input into standby so that it tries to 6065 // recover at next read attempt 6066 mInput->stream->common.standby(&mInput->stream->common); 6067 usleep(kRecordThreadSleepUs); 6068 } 6069 mRsmpInIndex = mFrameCount; 6070 framesOut = 0; 6071 buffer.frameCount = 0; 6072 } 6073 } 6074 } 6075 } else { 6076 // resampling 6077 6078 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6079 // alter output frame count as if we were expecting stereo samples 6080 if (mChannelCount == 1 && mReqChannelCount == 1) { 6081 framesOut >>= 1; 6082 } 6083 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6084 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6085 // are 32 bit aligned which should be always true. 6086 if (mChannelCount == 2 && mReqChannelCount == 1) { 6087 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6088 // the resampler always outputs stereo samples: do post stereo to mono conversion 6089 int16_t *src = (int16_t *)mRsmpOutBuffer; 6090 int16_t *dst = buffer.i16; 6091 while (framesOut--) { 6092 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6093 src += 2; 6094 } 6095 } else { 6096 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6097 } 6098 6099 } 6100 if (mFramestoDrop == 0) { 6101 mActiveTrack->releaseBuffer(&buffer); 6102 } else { 6103 if (mFramestoDrop > 0) { 6104 mFramestoDrop -= buffer.frameCount; 6105 if (mFramestoDrop <= 0) { 6106 clearSyncStartEvent(); 6107 } 6108 } else { 6109 mFramestoDrop += buffer.frameCount; 6110 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6111 mSyncStartEvent->isCancelled()) { 6112 ALOGW("Synced record %s, session %d, trigger session %d", 6113 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6114 mActiveTrack->sessionId(), 6115 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6116 clearSyncStartEvent(); 6117 } 6118 } 6119 } 6120 mActiveTrack->overflow(); 6121 } 6122 // client isn't retrieving buffers fast enough 6123 else { 6124 if (!mActiveTrack->setOverflow()) { 6125 nsecs_t now = systemTime(); 6126 if ((now - lastWarning) > kWarningThrottleNs) { 6127 ALOGW("RecordThread: buffer overflow"); 6128 lastWarning = now; 6129 } 6130 } 6131 // Release the processor for a while before asking for a new buffer. 6132 // This will give the application more chance to read from the buffer and 6133 // clear the overflow. 6134 usleep(kRecordThreadSleepUs); 6135 } 6136 } 6137 // enable changes in effect chain 6138 unlockEffectChains(effectChains); 6139 effectChains.clear(); 6140 } 6141 6142 if (!mStandby) { 6143 mInput->stream->common.standby(&mInput->stream->common); 6144 } 6145 mActiveTrack.clear(); 6146 6147 mStartStopCond.broadcast(); 6148 6149 releaseWakeLock(); 6150 6151 ALOGV("RecordThread %p exiting", this); 6152 return false; 6153} 6154 6155 6156sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6157 const sp<AudioFlinger::Client>& client, 6158 uint32_t sampleRate, 6159 audio_format_t format, 6160 int channelMask, 6161 int frameCount, 6162 int sessionId, 6163 status_t *status) 6164{ 6165 sp<RecordTrack> track; 6166 status_t lStatus; 6167 6168 lStatus = initCheck(); 6169 if (lStatus != NO_ERROR) { 6170 ALOGE("Audio driver not initialized."); 6171 goto Exit; 6172 } 6173 6174 { // scope for mLock 6175 Mutex::Autolock _l(mLock); 6176 6177 track = new RecordTrack(this, client, sampleRate, 6178 format, channelMask, frameCount, sessionId); 6179 6180 if (track->getCblk() == 0) { 6181 lStatus = NO_MEMORY; 6182 goto Exit; 6183 } 6184 6185 mTrack = track.get(); 6186 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6187 bool suspend = audio_is_bluetooth_sco_device( 6188 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6189 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6190 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6191 } 6192 lStatus = NO_ERROR; 6193 6194Exit: 6195 if (status) { 6196 *status = lStatus; 6197 } 6198 return track; 6199} 6200 6201status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6202 AudioSystem::sync_event_t event, 6203 int triggerSession) 6204{ 6205 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6206 sp<ThreadBase> strongMe = this; 6207 status_t status = NO_ERROR; 6208 6209 if (event == AudioSystem::SYNC_EVENT_NONE) { 6210 clearSyncStartEvent(); 6211 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6212 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6213 triggerSession, 6214 recordTrack->sessionId(), 6215 syncStartEventCallback, 6216 this); 6217 // Sync event can be cancelled by the trigger session if the track is not in a 6218 // compatible state in which case we start record immediately 6219 if (mSyncStartEvent->isCancelled()) { 6220 clearSyncStartEvent(); 6221 } else { 6222 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6223 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6224 } 6225 } 6226 6227 { 6228 AutoMutex lock(mLock); 6229 if (mActiveTrack != 0) { 6230 if (recordTrack != mActiveTrack.get()) { 6231 status = -EBUSY; 6232 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6233 mActiveTrack->mState = TrackBase::ACTIVE; 6234 } 6235 return status; 6236 } 6237 6238 recordTrack->mState = TrackBase::IDLE; 6239 mActiveTrack = recordTrack; 6240 mLock.unlock(); 6241 status_t status = AudioSystem::startInput(mId); 6242 mLock.lock(); 6243 if (status != NO_ERROR) { 6244 mActiveTrack.clear(); 6245 clearSyncStartEvent(); 6246 return status; 6247 } 6248 mRsmpInIndex = mFrameCount; 6249 mBytesRead = 0; 6250 if (mResampler != NULL) { 6251 mResampler->reset(); 6252 } 6253 mActiveTrack->mState = TrackBase::RESUMING; 6254 // signal thread to start 6255 ALOGV("Signal record thread"); 6256 mWaitWorkCV.signal(); 6257 // do not wait for mStartStopCond if exiting 6258 if (exitPending()) { 6259 mActiveTrack.clear(); 6260 status = INVALID_OPERATION; 6261 goto startError; 6262 } 6263 mStartStopCond.wait(mLock); 6264 if (mActiveTrack == 0) { 6265 ALOGV("Record failed to start"); 6266 status = BAD_VALUE; 6267 goto startError; 6268 } 6269 ALOGV("Record started OK"); 6270 return status; 6271 } 6272startError: 6273 AudioSystem::stopInput(mId); 6274 clearSyncStartEvent(); 6275 return status; 6276} 6277 6278void AudioFlinger::RecordThread::clearSyncStartEvent() 6279{ 6280 if (mSyncStartEvent != 0) { 6281 mSyncStartEvent->cancel(); 6282 } 6283 mSyncStartEvent.clear(); 6284 mFramestoDrop = 0; 6285} 6286 6287void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6288{ 6289 sp<SyncEvent> strongEvent = event.promote(); 6290 6291 if (strongEvent != 0) { 6292 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6293 me->handleSyncStartEvent(strongEvent); 6294 } 6295} 6296 6297void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6298{ 6299 if (event == mSyncStartEvent) { 6300 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6301 // from audio HAL 6302 mFramestoDrop = mFrameCount * 2; 6303 } 6304} 6305 6306void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6307 ALOGV("RecordThread::stop"); 6308 sp<ThreadBase> strongMe = this; 6309 { 6310 AutoMutex lock(mLock); 6311 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6312 mActiveTrack->mState = TrackBase::PAUSING; 6313 // do not wait for mStartStopCond if exiting 6314 if (exitPending()) { 6315 return; 6316 } 6317 mStartStopCond.wait(mLock); 6318 // if we have been restarted, recordTrack == mActiveTrack.get() here 6319 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6320 mLock.unlock(); 6321 AudioSystem::stopInput(mId); 6322 mLock.lock(); 6323 ALOGV("Record stopped OK"); 6324 } 6325 } 6326 } 6327} 6328 6329bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6330{ 6331 return false; 6332} 6333 6334status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6335{ 6336 if (!isValidSyncEvent(event)) { 6337 return BAD_VALUE; 6338 } 6339 6340 Mutex::Autolock _l(mLock); 6341 6342 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6343 mTrack->setSyncEvent(event); 6344 return NO_ERROR; 6345 } 6346 return NAME_NOT_FOUND; 6347} 6348 6349status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6350{ 6351 const size_t SIZE = 256; 6352 char buffer[SIZE]; 6353 String8 result; 6354 6355 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6356 result.append(buffer); 6357 6358 if (mActiveTrack != 0) { 6359 result.append("Active Track:\n"); 6360 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6361 mActiveTrack->dump(buffer, SIZE); 6362 result.append(buffer); 6363 6364 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6365 result.append(buffer); 6366 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6367 result.append(buffer); 6368 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6369 result.append(buffer); 6370 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6371 result.append(buffer); 6372 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6373 result.append(buffer); 6374 6375 6376 } else { 6377 result.append("No record client\n"); 6378 } 6379 write(fd, result.string(), result.size()); 6380 6381 dumpBase(fd, args); 6382 dumpEffectChains(fd, args); 6383 6384 return NO_ERROR; 6385} 6386 6387// AudioBufferProvider interface 6388status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6389{ 6390 size_t framesReq = buffer->frameCount; 6391 size_t framesReady = mFrameCount - mRsmpInIndex; 6392 int channelCount; 6393 6394 if (framesReady == 0) { 6395 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6396 if (mBytesRead < 0) { 6397 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6398 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6399 // Force input into standby so that it tries to 6400 // recover at next read attempt 6401 mInput->stream->common.standby(&mInput->stream->common); 6402 usleep(kRecordThreadSleepUs); 6403 } 6404 buffer->raw = NULL; 6405 buffer->frameCount = 0; 6406 return NOT_ENOUGH_DATA; 6407 } 6408 mRsmpInIndex = 0; 6409 framesReady = mFrameCount; 6410 } 6411 6412 if (framesReq > framesReady) { 6413 framesReq = framesReady; 6414 } 6415 6416 if (mChannelCount == 1 && mReqChannelCount == 2) { 6417 channelCount = 1; 6418 } else { 6419 channelCount = 2; 6420 } 6421 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6422 buffer->frameCount = framesReq; 6423 return NO_ERROR; 6424} 6425 6426// AudioBufferProvider interface 6427void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6428{ 6429 mRsmpInIndex += buffer->frameCount; 6430 buffer->frameCount = 0; 6431} 6432 6433bool AudioFlinger::RecordThread::checkForNewParameters_l() 6434{ 6435 bool reconfig = false; 6436 6437 while (!mNewParameters.isEmpty()) { 6438 status_t status = NO_ERROR; 6439 String8 keyValuePair = mNewParameters[0]; 6440 AudioParameter param = AudioParameter(keyValuePair); 6441 int value; 6442 audio_format_t reqFormat = mFormat; 6443 int reqSamplingRate = mReqSampleRate; 6444 int reqChannelCount = mReqChannelCount; 6445 6446 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6447 reqSamplingRate = value; 6448 reconfig = true; 6449 } 6450 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6451 reqFormat = (audio_format_t) value; 6452 reconfig = true; 6453 } 6454 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6455 reqChannelCount = popcount(value); 6456 reconfig = true; 6457 } 6458 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6459 // do not accept frame count changes if tracks are open as the track buffer 6460 // size depends on frame count and correct behavior would not be guaranteed 6461 // if frame count is changed after track creation 6462 if (mActiveTrack != 0) { 6463 status = INVALID_OPERATION; 6464 } else { 6465 reconfig = true; 6466 } 6467 } 6468 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6469 // forward device change to effects that have requested to be 6470 // aware of attached audio device. 6471 for (size_t i = 0; i < mEffectChains.size(); i++) { 6472 mEffectChains[i]->setDevice_l(value); 6473 } 6474 // store input device and output device but do not forward output device to audio HAL. 6475 // Note that status is ignored by the caller for output device 6476 // (see AudioFlinger::setParameters() 6477 if (value & AUDIO_DEVICE_OUT_ALL) { 6478 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6479 status = BAD_VALUE; 6480 } else { 6481 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6482 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6483 if (mTrack != NULL) { 6484 bool suspend = audio_is_bluetooth_sco_device( 6485 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6486 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6487 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6488 } 6489 } 6490 mDevice |= (uint32_t)value; 6491 } 6492 if (status == NO_ERROR) { 6493 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6494 if (status == INVALID_OPERATION) { 6495 mInput->stream->common.standby(&mInput->stream->common); 6496 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6497 keyValuePair.string()); 6498 } 6499 if (reconfig) { 6500 if (status == BAD_VALUE && 6501 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6502 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6503 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6504 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6505 (reqChannelCount <= FCC_2)) { 6506 status = NO_ERROR; 6507 } 6508 if (status == NO_ERROR) { 6509 readInputParameters(); 6510 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6511 } 6512 } 6513 } 6514 6515 mNewParameters.removeAt(0); 6516 6517 mParamStatus = status; 6518 mParamCond.signal(); 6519 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6520 // already timed out waiting for the status and will never signal the condition. 6521 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6522 } 6523 return reconfig; 6524} 6525 6526String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6527{ 6528 char *s; 6529 String8 out_s8 = String8(); 6530 6531 Mutex::Autolock _l(mLock); 6532 if (initCheck() != NO_ERROR) { 6533 return out_s8; 6534 } 6535 6536 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6537 out_s8 = String8(s); 6538 free(s); 6539 return out_s8; 6540} 6541 6542void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6543 AudioSystem::OutputDescriptor desc; 6544 void *param2 = NULL; 6545 6546 switch (event) { 6547 case AudioSystem::INPUT_OPENED: 6548 case AudioSystem::INPUT_CONFIG_CHANGED: 6549 desc.channels = mChannelMask; 6550 desc.samplingRate = mSampleRate; 6551 desc.format = mFormat; 6552 desc.frameCount = mFrameCount; 6553 desc.latency = 0; 6554 param2 = &desc; 6555 break; 6556 6557 case AudioSystem::INPUT_CLOSED: 6558 default: 6559 break; 6560 } 6561 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6562} 6563 6564void AudioFlinger::RecordThread::readInputParameters() 6565{ 6566 delete mRsmpInBuffer; 6567 // mRsmpInBuffer is always assigned a new[] below 6568 delete mRsmpOutBuffer; 6569 mRsmpOutBuffer = NULL; 6570 delete mResampler; 6571 mResampler = NULL; 6572 6573 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6574 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6575 mChannelCount = (uint16_t)popcount(mChannelMask); 6576 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6577 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6578 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6579 mFrameCount = mInputBytes / mFrameSize; 6580 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6581 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6582 6583 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6584 { 6585 int channelCount; 6586 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6587 // stereo to mono post process as the resampler always outputs stereo. 6588 if (mChannelCount == 1 && mReqChannelCount == 2) { 6589 channelCount = 1; 6590 } else { 6591 channelCount = 2; 6592 } 6593 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6594 mResampler->setSampleRate(mSampleRate); 6595 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6596 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6597 6598 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6599 if (mChannelCount == 1 && mReqChannelCount == 1) { 6600 mFrameCount >>= 1; 6601 } 6602 6603 } 6604 mRsmpInIndex = mFrameCount; 6605} 6606 6607unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6608{ 6609 Mutex::Autolock _l(mLock); 6610 if (initCheck() != NO_ERROR) { 6611 return 0; 6612 } 6613 6614 return mInput->stream->get_input_frames_lost(mInput->stream); 6615} 6616 6617uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6618{ 6619 Mutex::Autolock _l(mLock); 6620 uint32_t result = 0; 6621 if (getEffectChain_l(sessionId) != 0) { 6622 result = EFFECT_SESSION; 6623 } 6624 6625 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6626 result |= TRACK_SESSION; 6627 } 6628 6629 return result; 6630} 6631 6632AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6633{ 6634 Mutex::Autolock _l(mLock); 6635 return mTrack; 6636} 6637 6638AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6639{ 6640 Mutex::Autolock _l(mLock); 6641 return mInput; 6642} 6643 6644AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6645{ 6646 Mutex::Autolock _l(mLock); 6647 AudioStreamIn *input = mInput; 6648 mInput = NULL; 6649 return input; 6650} 6651 6652// this method must always be called either with ThreadBase mLock held or inside the thread loop 6653audio_stream_t* AudioFlinger::RecordThread::stream() const 6654{ 6655 if (mInput == NULL) { 6656 return NULL; 6657 } 6658 return &mInput->stream->common; 6659} 6660 6661 6662// ---------------------------------------------------------------------------- 6663 6664audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6665{ 6666 if (!settingsAllowed()) { 6667 return 0; 6668 } 6669 Mutex::Autolock _l(mLock); 6670 return loadHwModule_l(name); 6671} 6672 6673// loadHwModule_l() must be called with AudioFlinger::mLock held 6674audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6675{ 6676 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6677 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6678 ALOGW("loadHwModule() module %s already loaded", name); 6679 return mAudioHwDevs.keyAt(i); 6680 } 6681 } 6682 6683 audio_hw_device_t *dev; 6684 6685 int rc = load_audio_interface(name, &dev); 6686 if (rc) { 6687 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6688 return 0; 6689 } 6690 6691 mHardwareStatus = AUDIO_HW_INIT; 6692 rc = dev->init_check(dev); 6693 mHardwareStatus = AUDIO_HW_IDLE; 6694 if (rc) { 6695 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6696 return 0; 6697 } 6698 6699 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6700 (NULL != dev->set_master_volume)) { 6701 AutoMutex lock(mHardwareLock); 6702 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6703 dev->set_master_volume(dev, mMasterVolume); 6704 mHardwareStatus = AUDIO_HW_IDLE; 6705 } 6706 6707 audio_module_handle_t handle = nextUniqueId(); 6708 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6709 6710 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6711 name, dev->common.module->name, dev->common.module->id, handle); 6712 6713 return handle; 6714 6715} 6716 6717audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6718 audio_devices_t *pDevices, 6719 uint32_t *pSamplingRate, 6720 audio_format_t *pFormat, 6721 audio_channel_mask_t *pChannelMask, 6722 uint32_t *pLatencyMs, 6723 audio_output_flags_t flags) 6724{ 6725 status_t status; 6726 PlaybackThread *thread = NULL; 6727 struct audio_config config = { 6728 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6729 channel_mask: pChannelMask ? *pChannelMask : 0, 6730 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6731 }; 6732 audio_stream_out_t *outStream = NULL; 6733 audio_hw_device_t *outHwDev; 6734 6735 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6736 module, 6737 (pDevices != NULL) ? (int)*pDevices : 0, 6738 config.sample_rate, 6739 config.format, 6740 config.channel_mask, 6741 flags); 6742 6743 if (pDevices == NULL || *pDevices == 0) { 6744 return 0; 6745 } 6746 6747 Mutex::Autolock _l(mLock); 6748 6749 outHwDev = findSuitableHwDev_l(module, *pDevices); 6750 if (outHwDev == NULL) 6751 return 0; 6752 6753 audio_io_handle_t id = nextUniqueId(); 6754 6755 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6756 6757 status = outHwDev->open_output_stream(outHwDev, 6758 id, 6759 *pDevices, 6760 (audio_output_flags_t)flags, 6761 &config, 6762 &outStream); 6763 6764 mHardwareStatus = AUDIO_HW_IDLE; 6765 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6766 outStream, 6767 config.sample_rate, 6768 config.format, 6769 config.channel_mask, 6770 status); 6771 6772 if (status == NO_ERROR && outStream != NULL) { 6773 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6774 6775 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6776 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6777 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6778 thread = new DirectOutputThread(this, output, id, *pDevices); 6779 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6780 } else { 6781 thread = new MixerThread(this, output, id, *pDevices); 6782 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6783 } 6784 mPlaybackThreads.add(id, thread); 6785 6786 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6787 if (pFormat != NULL) *pFormat = config.format; 6788 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6789 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6790 6791 // notify client processes of the new output creation 6792 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6793 6794 // the first primary output opened designates the primary hw device 6795 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6796 ALOGI("Using module %d has the primary audio interface", module); 6797 mPrimaryHardwareDev = outHwDev; 6798 6799 AutoMutex lock(mHardwareLock); 6800 mHardwareStatus = AUDIO_HW_SET_MODE; 6801 outHwDev->set_mode(outHwDev, mMode); 6802 6803 // Determine the level of master volume support the primary audio HAL has, 6804 // and set the initial master volume at the same time. 6805 float initialVolume = 1.0; 6806 mMasterVolumeSupportLvl = MVS_NONE; 6807 6808 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6809 if ((NULL != outHwDev->get_master_volume) && 6810 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6811 mMasterVolumeSupportLvl = MVS_FULL; 6812 } else { 6813 mMasterVolumeSupportLvl = MVS_SETONLY; 6814 initialVolume = 1.0; 6815 } 6816 6817 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6818 if ((NULL == outHwDev->set_master_volume) || 6819 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6820 mMasterVolumeSupportLvl = MVS_NONE; 6821 } 6822 // now that we have a primary device, initialize master volume on other devices 6823 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6824 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6825 6826 if ((dev != mPrimaryHardwareDev) && 6827 (NULL != dev->set_master_volume)) { 6828 dev->set_master_volume(dev, initialVolume); 6829 } 6830 } 6831 mHardwareStatus = AUDIO_HW_IDLE; 6832 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6833 ? initialVolume 6834 : 1.0; 6835 mMasterVolume = initialVolume; 6836 } 6837 return id; 6838 } 6839 6840 return 0; 6841} 6842 6843audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6844 audio_io_handle_t output2) 6845{ 6846 Mutex::Autolock _l(mLock); 6847 MixerThread *thread1 = checkMixerThread_l(output1); 6848 MixerThread *thread2 = checkMixerThread_l(output2); 6849 6850 if (thread1 == NULL || thread2 == NULL) { 6851 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6852 return 0; 6853 } 6854 6855 audio_io_handle_t id = nextUniqueId(); 6856 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6857 thread->addOutputTrack(thread2); 6858 mPlaybackThreads.add(id, thread); 6859 // notify client processes of the new output creation 6860 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6861 return id; 6862} 6863 6864status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6865{ 6866 // keep strong reference on the playback thread so that 6867 // it is not destroyed while exit() is executed 6868 sp<PlaybackThread> thread; 6869 { 6870 Mutex::Autolock _l(mLock); 6871 thread = checkPlaybackThread_l(output); 6872 if (thread == NULL) { 6873 return BAD_VALUE; 6874 } 6875 6876 ALOGV("closeOutput() %d", output); 6877 6878 if (thread->type() == ThreadBase::MIXER) { 6879 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6880 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6881 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6882 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6883 } 6884 } 6885 } 6886 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6887 mPlaybackThreads.removeItem(output); 6888 } 6889 thread->exit(); 6890 // The thread entity (active unit of execution) is no longer running here, 6891 // but the ThreadBase container still exists. 6892 6893 if (thread->type() != ThreadBase::DUPLICATING) { 6894 AudioStreamOut *out = thread->clearOutput(); 6895 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6896 // from now on thread->mOutput is NULL 6897 out->hwDev->close_output_stream(out->hwDev, out->stream); 6898 delete out; 6899 } 6900 return NO_ERROR; 6901} 6902 6903status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6904{ 6905 Mutex::Autolock _l(mLock); 6906 PlaybackThread *thread = checkPlaybackThread_l(output); 6907 6908 if (thread == NULL) { 6909 return BAD_VALUE; 6910 } 6911 6912 ALOGV("suspendOutput() %d", output); 6913 thread->suspend(); 6914 6915 return NO_ERROR; 6916} 6917 6918status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6919{ 6920 Mutex::Autolock _l(mLock); 6921 PlaybackThread *thread = checkPlaybackThread_l(output); 6922 6923 if (thread == NULL) { 6924 return BAD_VALUE; 6925 } 6926 6927 ALOGV("restoreOutput() %d", output); 6928 6929 thread->restore(); 6930 6931 return NO_ERROR; 6932} 6933 6934audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6935 audio_devices_t *pDevices, 6936 uint32_t *pSamplingRate, 6937 audio_format_t *pFormat, 6938 uint32_t *pChannelMask) 6939{ 6940 status_t status; 6941 RecordThread *thread = NULL; 6942 struct audio_config config = { 6943 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6944 channel_mask: pChannelMask ? *pChannelMask : 0, 6945 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6946 }; 6947 uint32_t reqSamplingRate = config.sample_rate; 6948 audio_format_t reqFormat = config.format; 6949 audio_channel_mask_t reqChannels = config.channel_mask; 6950 audio_stream_in_t *inStream = NULL; 6951 audio_hw_device_t *inHwDev; 6952 6953 if (pDevices == NULL || *pDevices == 0) { 6954 return 0; 6955 } 6956 6957 Mutex::Autolock _l(mLock); 6958 6959 inHwDev = findSuitableHwDev_l(module, *pDevices); 6960 if (inHwDev == NULL) 6961 return 0; 6962 6963 audio_io_handle_t id = nextUniqueId(); 6964 6965 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6966 &inStream); 6967 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6968 inStream, 6969 config.sample_rate, 6970 config.format, 6971 config.channel_mask, 6972 status); 6973 6974 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6975 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6976 // or stereo to mono conversions on 16 bit PCM inputs. 6977 if (status == BAD_VALUE && 6978 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6979 (config.sample_rate <= 2 * reqSamplingRate) && 6980 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6981 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6982 inStream = NULL; 6983 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6984 } 6985 6986 if (status == NO_ERROR && inStream != NULL) { 6987 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6988 6989 // Start record thread 6990 // RecorThread require both input and output device indication to forward to audio 6991 // pre processing modules 6992 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6993 thread = new RecordThread(this, 6994 input, 6995 reqSamplingRate, 6996 reqChannels, 6997 id, 6998 device); 6999 mRecordThreads.add(id, thread); 7000 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7001 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7002 if (pFormat != NULL) *pFormat = config.format; 7003 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7004 7005 input->stream->common.standby(&input->stream->common); 7006 7007 // notify client processes of the new input creation 7008 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7009 return id; 7010 } 7011 7012 return 0; 7013} 7014 7015status_t AudioFlinger::closeInput(audio_io_handle_t input) 7016{ 7017 // keep strong reference on the record thread so that 7018 // it is not destroyed while exit() is executed 7019 sp<RecordThread> thread; 7020 { 7021 Mutex::Autolock _l(mLock); 7022 thread = checkRecordThread_l(input); 7023 if (thread == NULL) { 7024 return BAD_VALUE; 7025 } 7026 7027 ALOGV("closeInput() %d", input); 7028 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7029 mRecordThreads.removeItem(input); 7030 } 7031 thread->exit(); 7032 // The thread entity (active unit of execution) is no longer running here, 7033 // but the ThreadBase container still exists. 7034 7035 AudioStreamIn *in = thread->clearInput(); 7036 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7037 // from now on thread->mInput is NULL 7038 in->hwDev->close_input_stream(in->hwDev, in->stream); 7039 delete in; 7040 7041 return NO_ERROR; 7042} 7043 7044status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7045{ 7046 Mutex::Autolock _l(mLock); 7047 MixerThread *dstThread = checkMixerThread_l(output); 7048 if (dstThread == NULL) { 7049 ALOGW("setStreamOutput() bad output id %d", output); 7050 return BAD_VALUE; 7051 } 7052 7053 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7054 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7055 7056 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7057 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7058 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7059 MixerThread *srcThread = (MixerThread *)thread; 7060 srcThread->invalidateTracks(stream); 7061 } 7062 } 7063 7064 return NO_ERROR; 7065} 7066 7067 7068int AudioFlinger::newAudioSessionId() 7069{ 7070 return nextUniqueId(); 7071} 7072 7073void AudioFlinger::acquireAudioSessionId(int audioSession) 7074{ 7075 Mutex::Autolock _l(mLock); 7076 pid_t caller = IPCThreadState::self()->getCallingPid(); 7077 ALOGV("acquiring %d from %d", audioSession, caller); 7078 size_t num = mAudioSessionRefs.size(); 7079 for (size_t i = 0; i< num; i++) { 7080 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7081 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7082 ref->mCnt++; 7083 ALOGV(" incremented refcount to %d", ref->mCnt); 7084 return; 7085 } 7086 } 7087 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7088 ALOGV(" added new entry for %d", audioSession); 7089} 7090 7091void AudioFlinger::releaseAudioSessionId(int audioSession) 7092{ 7093 Mutex::Autolock _l(mLock); 7094 pid_t caller = IPCThreadState::self()->getCallingPid(); 7095 ALOGV("releasing %d from %d", audioSession, caller); 7096 size_t num = mAudioSessionRefs.size(); 7097 for (size_t i = 0; i< num; i++) { 7098 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7099 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7100 ref->mCnt--; 7101 ALOGV(" decremented refcount to %d", ref->mCnt); 7102 if (ref->mCnt == 0) { 7103 mAudioSessionRefs.removeAt(i); 7104 delete ref; 7105 purgeStaleEffects_l(); 7106 } 7107 return; 7108 } 7109 } 7110 ALOGW("session id %d not found for pid %d", audioSession, caller); 7111} 7112 7113void AudioFlinger::purgeStaleEffects_l() { 7114 7115 ALOGV("purging stale effects"); 7116 7117 Vector< sp<EffectChain> > chains; 7118 7119 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7120 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7121 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7122 sp<EffectChain> ec = t->mEffectChains[j]; 7123 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7124 chains.push(ec); 7125 } 7126 } 7127 } 7128 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7129 sp<RecordThread> t = mRecordThreads.valueAt(i); 7130 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7131 sp<EffectChain> ec = t->mEffectChains[j]; 7132 chains.push(ec); 7133 } 7134 } 7135 7136 for (size_t i = 0; i < chains.size(); i++) { 7137 sp<EffectChain> ec = chains[i]; 7138 int sessionid = ec->sessionId(); 7139 sp<ThreadBase> t = ec->mThread.promote(); 7140 if (t == 0) { 7141 continue; 7142 } 7143 size_t numsessionrefs = mAudioSessionRefs.size(); 7144 bool found = false; 7145 for (size_t k = 0; k < numsessionrefs; k++) { 7146 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7147 if (ref->mSessionid == sessionid) { 7148 ALOGV(" session %d still exists for %d with %d refs", 7149 sessionid, ref->mPid, ref->mCnt); 7150 found = true; 7151 break; 7152 } 7153 } 7154 if (!found) { 7155 // remove all effects from the chain 7156 while (ec->mEffects.size()) { 7157 sp<EffectModule> effect = ec->mEffects[0]; 7158 effect->unPin(); 7159 Mutex::Autolock _l (t->mLock); 7160 t->removeEffect_l(effect); 7161 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7162 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7163 if (handle != 0) { 7164 handle->mEffect.clear(); 7165 if (handle->mHasControl && handle->mEnabled) { 7166 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7167 } 7168 } 7169 } 7170 AudioSystem::unregisterEffect(effect->id()); 7171 } 7172 } 7173 } 7174 return; 7175} 7176 7177// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7178AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7179{ 7180 return mPlaybackThreads.valueFor(output).get(); 7181} 7182 7183// checkMixerThread_l() must be called with AudioFlinger::mLock held 7184AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7185{ 7186 PlaybackThread *thread = checkPlaybackThread_l(output); 7187 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7188} 7189 7190// checkRecordThread_l() must be called with AudioFlinger::mLock held 7191AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7192{ 7193 return mRecordThreads.valueFor(input).get(); 7194} 7195 7196uint32_t AudioFlinger::nextUniqueId() 7197{ 7198 return android_atomic_inc(&mNextUniqueId); 7199} 7200 7201AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7202{ 7203 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7204 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7205 AudioStreamOut *output = thread->getOutput(); 7206 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7207 return thread; 7208 } 7209 } 7210 return NULL; 7211} 7212 7213uint32_t AudioFlinger::primaryOutputDevice_l() const 7214{ 7215 PlaybackThread *thread = primaryPlaybackThread_l(); 7216 7217 if (thread == NULL) { 7218 return 0; 7219 } 7220 7221 return thread->device(); 7222} 7223 7224sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7225 int triggerSession, 7226 int listenerSession, 7227 sync_event_callback_t callBack, 7228 void *cookie) 7229{ 7230 Mutex::Autolock _l(mLock); 7231 7232 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7233 status_t playStatus = NAME_NOT_FOUND; 7234 status_t recStatus = NAME_NOT_FOUND; 7235 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7236 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7237 if (playStatus == NO_ERROR) { 7238 return event; 7239 } 7240 } 7241 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7242 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7243 if (recStatus == NO_ERROR) { 7244 return event; 7245 } 7246 } 7247 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7248 mPendingSyncEvents.add(event); 7249 } else { 7250 ALOGV("createSyncEvent() invalid event %d", event->type()); 7251 event.clear(); 7252 } 7253 return event; 7254} 7255 7256// ---------------------------------------------------------------------------- 7257// Effect management 7258// ---------------------------------------------------------------------------- 7259 7260 7261status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7262{ 7263 Mutex::Autolock _l(mLock); 7264 return EffectQueryNumberEffects(numEffects); 7265} 7266 7267status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7268{ 7269 Mutex::Autolock _l(mLock); 7270 return EffectQueryEffect(index, descriptor); 7271} 7272 7273status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7274 effect_descriptor_t *descriptor) const 7275{ 7276 Mutex::Autolock _l(mLock); 7277 return EffectGetDescriptor(pUuid, descriptor); 7278} 7279 7280 7281sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7282 effect_descriptor_t *pDesc, 7283 const sp<IEffectClient>& effectClient, 7284 int32_t priority, 7285 audio_io_handle_t io, 7286 int sessionId, 7287 status_t *status, 7288 int *id, 7289 int *enabled) 7290{ 7291 status_t lStatus = NO_ERROR; 7292 sp<EffectHandle> handle; 7293 effect_descriptor_t desc; 7294 7295 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7296 pid, effectClient.get(), priority, sessionId, io); 7297 7298 if (pDesc == NULL) { 7299 lStatus = BAD_VALUE; 7300 goto Exit; 7301 } 7302 7303 // check audio settings permission for global effects 7304 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7305 lStatus = PERMISSION_DENIED; 7306 goto Exit; 7307 } 7308 7309 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7310 // that can only be created by audio policy manager (running in same process) 7311 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7312 lStatus = PERMISSION_DENIED; 7313 goto Exit; 7314 } 7315 7316 if (io == 0) { 7317 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7318 // output must be specified by AudioPolicyManager when using session 7319 // AUDIO_SESSION_OUTPUT_STAGE 7320 lStatus = BAD_VALUE; 7321 goto Exit; 7322 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7323 // if the output returned by getOutputForEffect() is removed before we lock the 7324 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7325 // and we will exit safely 7326 io = AudioSystem::getOutputForEffect(&desc); 7327 } 7328 } 7329 7330 { 7331 Mutex::Autolock _l(mLock); 7332 7333 7334 if (!EffectIsNullUuid(&pDesc->uuid)) { 7335 // if uuid is specified, request effect descriptor 7336 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7337 if (lStatus < 0) { 7338 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7339 goto Exit; 7340 } 7341 } else { 7342 // if uuid is not specified, look for an available implementation 7343 // of the required type in effect factory 7344 if (EffectIsNullUuid(&pDesc->type)) { 7345 ALOGW("createEffect() no effect type"); 7346 lStatus = BAD_VALUE; 7347 goto Exit; 7348 } 7349 uint32_t numEffects = 0; 7350 effect_descriptor_t d; 7351 d.flags = 0; // prevent compiler warning 7352 bool found = false; 7353 7354 lStatus = EffectQueryNumberEffects(&numEffects); 7355 if (lStatus < 0) { 7356 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7357 goto Exit; 7358 } 7359 for (uint32_t i = 0; i < numEffects; i++) { 7360 lStatus = EffectQueryEffect(i, &desc); 7361 if (lStatus < 0) { 7362 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7363 continue; 7364 } 7365 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7366 // If matching type found save effect descriptor. If the session is 7367 // 0 and the effect is not auxiliary, continue enumeration in case 7368 // an auxiliary version of this effect type is available 7369 found = true; 7370 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7371 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7372 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7373 break; 7374 } 7375 } 7376 } 7377 if (!found) { 7378 lStatus = BAD_VALUE; 7379 ALOGW("createEffect() effect not found"); 7380 goto Exit; 7381 } 7382 // For same effect type, chose auxiliary version over insert version if 7383 // connect to output mix (Compliance to OpenSL ES) 7384 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7385 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7386 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7387 } 7388 } 7389 7390 // Do not allow auxiliary effects on a session different from 0 (output mix) 7391 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7392 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7393 lStatus = INVALID_OPERATION; 7394 goto Exit; 7395 } 7396 7397 // check recording permission for visualizer 7398 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7399 !recordingAllowed()) { 7400 lStatus = PERMISSION_DENIED; 7401 goto Exit; 7402 } 7403 7404 // return effect descriptor 7405 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7406 7407 // If output is not specified try to find a matching audio session ID in one of the 7408 // output threads. 7409 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7410 // because of code checking output when entering the function. 7411 // Note: io is never 0 when creating an effect on an input 7412 if (io == 0) { 7413 // look for the thread where the specified audio session is present 7414 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7415 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7416 io = mPlaybackThreads.keyAt(i); 7417 break; 7418 } 7419 } 7420 if (io == 0) { 7421 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7422 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7423 io = mRecordThreads.keyAt(i); 7424 break; 7425 } 7426 } 7427 } 7428 // If no output thread contains the requested session ID, default to 7429 // first output. The effect chain will be moved to the correct output 7430 // thread when a track with the same session ID is created 7431 if (io == 0 && mPlaybackThreads.size()) { 7432 io = mPlaybackThreads.keyAt(0); 7433 } 7434 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7435 } 7436 ThreadBase *thread = checkRecordThread_l(io); 7437 if (thread == NULL) { 7438 thread = checkPlaybackThread_l(io); 7439 if (thread == NULL) { 7440 ALOGE("createEffect() unknown output thread"); 7441 lStatus = BAD_VALUE; 7442 goto Exit; 7443 } 7444 } 7445 7446 sp<Client> client = registerPid_l(pid); 7447 7448 // create effect on selected output thread 7449 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7450 &desc, enabled, &lStatus); 7451 if (handle != 0 && id != NULL) { 7452 *id = handle->id(); 7453 } 7454 } 7455 7456Exit: 7457 if (status != NULL) { 7458 *status = lStatus; 7459 } 7460 return handle; 7461} 7462 7463status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7464 audio_io_handle_t dstOutput) 7465{ 7466 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7467 sessionId, srcOutput, dstOutput); 7468 Mutex::Autolock _l(mLock); 7469 if (srcOutput == dstOutput) { 7470 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7471 return NO_ERROR; 7472 } 7473 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7474 if (srcThread == NULL) { 7475 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7476 return BAD_VALUE; 7477 } 7478 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7479 if (dstThread == NULL) { 7480 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7481 return BAD_VALUE; 7482 } 7483 7484 Mutex::Autolock _dl(dstThread->mLock); 7485 Mutex::Autolock _sl(srcThread->mLock); 7486 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7487 7488 return NO_ERROR; 7489} 7490 7491// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7492status_t AudioFlinger::moveEffectChain_l(int sessionId, 7493 AudioFlinger::PlaybackThread *srcThread, 7494 AudioFlinger::PlaybackThread *dstThread, 7495 bool reRegister) 7496{ 7497 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7498 sessionId, srcThread, dstThread); 7499 7500 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7501 if (chain == 0) { 7502 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7503 sessionId, srcThread); 7504 return INVALID_OPERATION; 7505 } 7506 7507 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7508 // so that a new chain is created with correct parameters when first effect is added. This is 7509 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7510 // removed. 7511 srcThread->removeEffectChain_l(chain); 7512 7513 // transfer all effects one by one so that new effect chain is created on new thread with 7514 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7515 audio_io_handle_t dstOutput = dstThread->id(); 7516 sp<EffectChain> dstChain; 7517 uint32_t strategy = 0; // prevent compiler warning 7518 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7519 while (effect != 0) { 7520 srcThread->removeEffect_l(effect); 7521 dstThread->addEffect_l(effect); 7522 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7523 if (effect->state() == EffectModule::ACTIVE || 7524 effect->state() == EffectModule::STOPPING) { 7525 effect->start(); 7526 } 7527 // if the move request is not received from audio policy manager, the effect must be 7528 // re-registered with the new strategy and output 7529 if (dstChain == 0) { 7530 dstChain = effect->chain().promote(); 7531 if (dstChain == 0) { 7532 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7533 srcThread->addEffect_l(effect); 7534 return NO_INIT; 7535 } 7536 strategy = dstChain->strategy(); 7537 } 7538 if (reRegister) { 7539 AudioSystem::unregisterEffect(effect->id()); 7540 AudioSystem::registerEffect(&effect->desc(), 7541 dstOutput, 7542 strategy, 7543 sessionId, 7544 effect->id()); 7545 } 7546 effect = chain->getEffectFromId_l(0); 7547 } 7548 7549 return NO_ERROR; 7550} 7551 7552 7553// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7554sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7555 const sp<AudioFlinger::Client>& client, 7556 const sp<IEffectClient>& effectClient, 7557 int32_t priority, 7558 int sessionId, 7559 effect_descriptor_t *desc, 7560 int *enabled, 7561 status_t *status 7562 ) 7563{ 7564 sp<EffectModule> effect; 7565 sp<EffectHandle> handle; 7566 status_t lStatus; 7567 sp<EffectChain> chain; 7568 bool chainCreated = false; 7569 bool effectCreated = false; 7570 bool effectRegistered = false; 7571 7572 lStatus = initCheck(); 7573 if (lStatus != NO_ERROR) { 7574 ALOGW("createEffect_l() Audio driver not initialized."); 7575 goto Exit; 7576 } 7577 7578 // Do not allow effects with session ID 0 on direct output or duplicating threads 7579 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7580 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7581 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7582 desc->name, sessionId); 7583 lStatus = BAD_VALUE; 7584 goto Exit; 7585 } 7586 // Only Pre processor effects are allowed on input threads and only on input threads 7587 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7588 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7589 desc->name, desc->flags, mType); 7590 lStatus = BAD_VALUE; 7591 goto Exit; 7592 } 7593 7594 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7595 7596 { // scope for mLock 7597 Mutex::Autolock _l(mLock); 7598 7599 // check for existing effect chain with the requested audio session 7600 chain = getEffectChain_l(sessionId); 7601 if (chain == 0) { 7602 // create a new chain for this session 7603 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7604 chain = new EffectChain(this, sessionId); 7605 addEffectChain_l(chain); 7606 chain->setStrategy(getStrategyForSession_l(sessionId)); 7607 chainCreated = true; 7608 } else { 7609 effect = chain->getEffectFromDesc_l(desc); 7610 } 7611 7612 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7613 7614 if (effect == 0) { 7615 int id = mAudioFlinger->nextUniqueId(); 7616 // Check CPU and memory usage 7617 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7618 if (lStatus != NO_ERROR) { 7619 goto Exit; 7620 } 7621 effectRegistered = true; 7622 // create a new effect module if none present in the chain 7623 effect = new EffectModule(this, chain, desc, id, sessionId); 7624 lStatus = effect->status(); 7625 if (lStatus != NO_ERROR) { 7626 goto Exit; 7627 } 7628 lStatus = chain->addEffect_l(effect); 7629 if (lStatus != NO_ERROR) { 7630 goto Exit; 7631 } 7632 effectCreated = true; 7633 7634 effect->setDevice(mDevice); 7635 effect->setMode(mAudioFlinger->getMode()); 7636 } 7637 // create effect handle and connect it to effect module 7638 handle = new EffectHandle(effect, client, effectClient, priority); 7639 lStatus = effect->addHandle(handle); 7640 if (enabled != NULL) { 7641 *enabled = (int)effect->isEnabled(); 7642 } 7643 } 7644 7645Exit: 7646 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7647 Mutex::Autolock _l(mLock); 7648 if (effectCreated) { 7649 chain->removeEffect_l(effect); 7650 } 7651 if (effectRegistered) { 7652 AudioSystem::unregisterEffect(effect->id()); 7653 } 7654 if (chainCreated) { 7655 removeEffectChain_l(chain); 7656 } 7657 handle.clear(); 7658 } 7659 7660 if (status != NULL) { 7661 *status = lStatus; 7662 } 7663 return handle; 7664} 7665 7666sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7667{ 7668 sp<EffectChain> chain = getEffectChain_l(sessionId); 7669 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7670} 7671 7672// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7673// PlaybackThread::mLock held 7674status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7675{ 7676 // check for existing effect chain with the requested audio session 7677 int sessionId = effect->sessionId(); 7678 sp<EffectChain> chain = getEffectChain_l(sessionId); 7679 bool chainCreated = false; 7680 7681 if (chain == 0) { 7682 // create a new chain for this session 7683 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7684 chain = new EffectChain(this, sessionId); 7685 addEffectChain_l(chain); 7686 chain->setStrategy(getStrategyForSession_l(sessionId)); 7687 chainCreated = true; 7688 } 7689 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7690 7691 if (chain->getEffectFromId_l(effect->id()) != 0) { 7692 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7693 this, effect->desc().name, chain.get()); 7694 return BAD_VALUE; 7695 } 7696 7697 status_t status = chain->addEffect_l(effect); 7698 if (status != NO_ERROR) { 7699 if (chainCreated) { 7700 removeEffectChain_l(chain); 7701 } 7702 return status; 7703 } 7704 7705 effect->setDevice(mDevice); 7706 effect->setMode(mAudioFlinger->getMode()); 7707 return NO_ERROR; 7708} 7709 7710void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7711 7712 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7713 effect_descriptor_t desc = effect->desc(); 7714 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7715 detachAuxEffect_l(effect->id()); 7716 } 7717 7718 sp<EffectChain> chain = effect->chain().promote(); 7719 if (chain != 0) { 7720 // remove effect chain if removing last effect 7721 if (chain->removeEffect_l(effect) == 0) { 7722 removeEffectChain_l(chain); 7723 } 7724 } else { 7725 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7726 } 7727} 7728 7729void AudioFlinger::ThreadBase::lockEffectChains_l( 7730 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7731{ 7732 effectChains = mEffectChains; 7733 for (size_t i = 0; i < mEffectChains.size(); i++) { 7734 mEffectChains[i]->lock(); 7735 } 7736} 7737 7738void AudioFlinger::ThreadBase::unlockEffectChains( 7739 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7740{ 7741 for (size_t i = 0; i < effectChains.size(); i++) { 7742 effectChains[i]->unlock(); 7743 } 7744} 7745 7746sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7747{ 7748 Mutex::Autolock _l(mLock); 7749 return getEffectChain_l(sessionId); 7750} 7751 7752sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7753{ 7754 size_t size = mEffectChains.size(); 7755 for (size_t i = 0; i < size; i++) { 7756 if (mEffectChains[i]->sessionId() == sessionId) { 7757 return mEffectChains[i]; 7758 } 7759 } 7760 return 0; 7761} 7762 7763void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7764{ 7765 Mutex::Autolock _l(mLock); 7766 size_t size = mEffectChains.size(); 7767 for (size_t i = 0; i < size; i++) { 7768 mEffectChains[i]->setMode_l(mode); 7769 } 7770} 7771 7772void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7773 const wp<EffectHandle>& handle, 7774 bool unpinIfLast) { 7775 7776 Mutex::Autolock _l(mLock); 7777 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7778 // delete the effect module if removing last handle on it 7779 if (effect->removeHandle(handle) == 0) { 7780 if (!effect->isPinned() || unpinIfLast) { 7781 removeEffect_l(effect); 7782 AudioSystem::unregisterEffect(effect->id()); 7783 } 7784 } 7785} 7786 7787status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7788{ 7789 int session = chain->sessionId(); 7790 int16_t *buffer = mMixBuffer; 7791 bool ownsBuffer = false; 7792 7793 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7794 if (session > 0) { 7795 // Only one effect chain can be present in direct output thread and it uses 7796 // the mix buffer as input 7797 if (mType != DIRECT) { 7798 size_t numSamples = mNormalFrameCount * mChannelCount; 7799 buffer = new int16_t[numSamples]; 7800 memset(buffer, 0, numSamples * sizeof(int16_t)); 7801 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7802 ownsBuffer = true; 7803 } 7804 7805 // Attach all tracks with same session ID to this chain. 7806 for (size_t i = 0; i < mTracks.size(); ++i) { 7807 sp<Track> track = mTracks[i]; 7808 if (session == track->sessionId()) { 7809 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7810 track->setMainBuffer(buffer); 7811 chain->incTrackCnt(); 7812 } 7813 } 7814 7815 // indicate all active tracks in the chain 7816 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7817 sp<Track> track = mActiveTracks[i].promote(); 7818 if (track == 0) continue; 7819 if (session == track->sessionId()) { 7820 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7821 chain->incActiveTrackCnt(); 7822 } 7823 } 7824 } 7825 7826 chain->setInBuffer(buffer, ownsBuffer); 7827 chain->setOutBuffer(mMixBuffer); 7828 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7829 // chains list in order to be processed last as it contains output stage effects 7830 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7831 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7832 // after track specific effects and before output stage 7833 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7834 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7835 // Effect chain for other sessions are inserted at beginning of effect 7836 // chains list to be processed before output mix effects. Relative order between other 7837 // sessions is not important 7838 size_t size = mEffectChains.size(); 7839 size_t i = 0; 7840 for (i = 0; i < size; i++) { 7841 if (mEffectChains[i]->sessionId() < session) break; 7842 } 7843 mEffectChains.insertAt(chain, i); 7844 checkSuspendOnAddEffectChain_l(chain); 7845 7846 return NO_ERROR; 7847} 7848 7849size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7850{ 7851 int session = chain->sessionId(); 7852 7853 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7854 7855 for (size_t i = 0; i < mEffectChains.size(); i++) { 7856 if (chain == mEffectChains[i]) { 7857 mEffectChains.removeAt(i); 7858 // detach all active tracks from the chain 7859 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7860 sp<Track> track = mActiveTracks[i].promote(); 7861 if (track == 0) continue; 7862 if (session == track->sessionId()) { 7863 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7864 chain.get(), session); 7865 chain->decActiveTrackCnt(); 7866 } 7867 } 7868 7869 // detach all tracks with same session ID from this chain 7870 for (size_t i = 0; i < mTracks.size(); ++i) { 7871 sp<Track> track = mTracks[i]; 7872 if (session == track->sessionId()) { 7873 track->setMainBuffer(mMixBuffer); 7874 chain->decTrackCnt(); 7875 } 7876 } 7877 break; 7878 } 7879 } 7880 return mEffectChains.size(); 7881} 7882 7883status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7884 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7885{ 7886 Mutex::Autolock _l(mLock); 7887 return attachAuxEffect_l(track, EffectId); 7888} 7889 7890status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7891 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7892{ 7893 status_t status = NO_ERROR; 7894 7895 if (EffectId == 0) { 7896 track->setAuxBuffer(0, NULL); 7897 } else { 7898 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7899 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7900 if (effect != 0) { 7901 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7902 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7903 } else { 7904 status = INVALID_OPERATION; 7905 } 7906 } else { 7907 status = BAD_VALUE; 7908 } 7909 } 7910 return status; 7911} 7912 7913void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7914{ 7915 for (size_t i = 0; i < mTracks.size(); ++i) { 7916 sp<Track> track = mTracks[i]; 7917 if (track->auxEffectId() == effectId) { 7918 attachAuxEffect_l(track, 0); 7919 } 7920 } 7921} 7922 7923status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7924{ 7925 // only one chain per input thread 7926 if (mEffectChains.size() != 0) { 7927 return INVALID_OPERATION; 7928 } 7929 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7930 7931 chain->setInBuffer(NULL); 7932 chain->setOutBuffer(NULL); 7933 7934 checkSuspendOnAddEffectChain_l(chain); 7935 7936 mEffectChains.add(chain); 7937 7938 return NO_ERROR; 7939} 7940 7941size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7942{ 7943 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7944 ALOGW_IF(mEffectChains.size() != 1, 7945 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7946 chain.get(), mEffectChains.size(), this); 7947 if (mEffectChains.size() == 1) { 7948 mEffectChains.removeAt(0); 7949 } 7950 return 0; 7951} 7952 7953// ---------------------------------------------------------------------------- 7954// EffectModule implementation 7955// ---------------------------------------------------------------------------- 7956 7957#undef LOG_TAG 7958#define LOG_TAG "AudioFlinger::EffectModule" 7959 7960AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7961 const wp<AudioFlinger::EffectChain>& chain, 7962 effect_descriptor_t *desc, 7963 int id, 7964 int sessionId) 7965 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7966 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7967{ 7968 ALOGV("Constructor %p", this); 7969 int lStatus; 7970 if (thread == NULL) { 7971 return; 7972 } 7973 7974 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7975 7976 // create effect engine from effect factory 7977 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7978 7979 if (mStatus != NO_ERROR) { 7980 return; 7981 } 7982 lStatus = init(); 7983 if (lStatus < 0) { 7984 mStatus = lStatus; 7985 goto Error; 7986 } 7987 7988 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7989 mPinned = true; 7990 } 7991 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7992 return; 7993Error: 7994 EffectRelease(mEffectInterface); 7995 mEffectInterface = NULL; 7996 ALOGV("Constructor Error %d", mStatus); 7997} 7998 7999AudioFlinger::EffectModule::~EffectModule() 8000{ 8001 ALOGV("Destructor %p", this); 8002 if (mEffectInterface != NULL) { 8003 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8004 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8005 sp<ThreadBase> thread = mThread.promote(); 8006 if (thread != 0) { 8007 audio_stream_t *stream = thread->stream(); 8008 if (stream != NULL) { 8009 stream->remove_audio_effect(stream, mEffectInterface); 8010 } 8011 } 8012 } 8013 // release effect engine 8014 EffectRelease(mEffectInterface); 8015 } 8016} 8017 8018status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8019{ 8020 status_t status; 8021 8022 Mutex::Autolock _l(mLock); 8023 int priority = handle->priority(); 8024 size_t size = mHandles.size(); 8025 sp<EffectHandle> h; 8026 size_t i; 8027 for (i = 0; i < size; i++) { 8028 h = mHandles[i].promote(); 8029 if (h == 0) continue; 8030 if (h->priority() <= priority) break; 8031 } 8032 // if inserted in first place, move effect control from previous owner to this handle 8033 if (i == 0) { 8034 bool enabled = false; 8035 if (h != 0) { 8036 enabled = h->enabled(); 8037 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8038 } 8039 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8040 status = NO_ERROR; 8041 } else { 8042 status = ALREADY_EXISTS; 8043 } 8044 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8045 mHandles.insertAt(handle, i); 8046 return status; 8047} 8048 8049size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8050{ 8051 Mutex::Autolock _l(mLock); 8052 size_t size = mHandles.size(); 8053 size_t i; 8054 for (i = 0; i < size; i++) { 8055 if (mHandles[i] == handle) break; 8056 } 8057 if (i == size) { 8058 return size; 8059 } 8060 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8061 8062 bool enabled = false; 8063 EffectHandle *hdl = handle.unsafe_get(); 8064 if (hdl != NULL) { 8065 ALOGV("removeHandle() unsafe_get OK"); 8066 enabled = hdl->enabled(); 8067 } 8068 mHandles.removeAt(i); 8069 size = mHandles.size(); 8070 // if removed from first place, move effect control from this handle to next in line 8071 if (i == 0 && size != 0) { 8072 sp<EffectHandle> h = mHandles[0].promote(); 8073 if (h != 0) { 8074 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8075 } 8076 } 8077 8078 // Prevent calls to process() and other functions on effect interface from now on. 8079 // The effect engine will be released by the destructor when the last strong reference on 8080 // this object is released which can happen after next process is called. 8081 if (size == 0 && !mPinned) { 8082 mState = DESTROYED; 8083 } 8084 8085 return size; 8086} 8087 8088sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8089{ 8090 Mutex::Autolock _l(mLock); 8091 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8092} 8093 8094void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8095{ 8096 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8097 // keep a strong reference on this EffectModule to avoid calling the 8098 // destructor before we exit 8099 sp<EffectModule> keep(this); 8100 { 8101 sp<ThreadBase> thread = mThread.promote(); 8102 if (thread != 0) { 8103 thread->disconnectEffect(keep, handle, unpinIfLast); 8104 } 8105 } 8106} 8107 8108void AudioFlinger::EffectModule::updateState() { 8109 Mutex::Autolock _l(mLock); 8110 8111 switch (mState) { 8112 case RESTART: 8113 reset_l(); 8114 // FALL THROUGH 8115 8116 case STARTING: 8117 // clear auxiliary effect input buffer for next accumulation 8118 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8119 memset(mConfig.inputCfg.buffer.raw, 8120 0, 8121 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8122 } 8123 start_l(); 8124 mState = ACTIVE; 8125 break; 8126 case STOPPING: 8127 stop_l(); 8128 mDisableWaitCnt = mMaxDisableWaitCnt; 8129 mState = STOPPED; 8130 break; 8131 case STOPPED: 8132 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8133 // turn off sequence. 8134 if (--mDisableWaitCnt == 0) { 8135 reset_l(); 8136 mState = IDLE; 8137 } 8138 break; 8139 default: //IDLE , ACTIVE, DESTROYED 8140 break; 8141 } 8142} 8143 8144void AudioFlinger::EffectModule::process() 8145{ 8146 Mutex::Autolock _l(mLock); 8147 8148 if (mState == DESTROYED || mEffectInterface == NULL || 8149 mConfig.inputCfg.buffer.raw == NULL || 8150 mConfig.outputCfg.buffer.raw == NULL) { 8151 return; 8152 } 8153 8154 if (isProcessEnabled()) { 8155 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8156 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8157 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8158 mConfig.inputCfg.buffer.s32, 8159 mConfig.inputCfg.buffer.frameCount/2); 8160 } 8161 8162 // do the actual processing in the effect engine 8163 int ret = (*mEffectInterface)->process(mEffectInterface, 8164 &mConfig.inputCfg.buffer, 8165 &mConfig.outputCfg.buffer); 8166 8167 // force transition to IDLE state when engine is ready 8168 if (mState == STOPPED && ret == -ENODATA) { 8169 mDisableWaitCnt = 1; 8170 } 8171 8172 // clear auxiliary effect input buffer for next accumulation 8173 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8174 memset(mConfig.inputCfg.buffer.raw, 0, 8175 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8176 } 8177 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8178 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8179 // If an insert effect is idle and input buffer is different from output buffer, 8180 // accumulate input onto output 8181 sp<EffectChain> chain = mChain.promote(); 8182 if (chain != 0 && chain->activeTrackCnt() != 0) { 8183 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8184 int16_t *in = mConfig.inputCfg.buffer.s16; 8185 int16_t *out = mConfig.outputCfg.buffer.s16; 8186 for (size_t i = 0; i < frameCnt; i++) { 8187 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8188 } 8189 } 8190 } 8191} 8192 8193void AudioFlinger::EffectModule::reset_l() 8194{ 8195 if (mEffectInterface == NULL) { 8196 return; 8197 } 8198 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8199} 8200 8201status_t AudioFlinger::EffectModule::configure() 8202{ 8203 uint32_t channels; 8204 if (mEffectInterface == NULL) { 8205 return NO_INIT; 8206 } 8207 8208 sp<ThreadBase> thread = mThread.promote(); 8209 if (thread == 0) { 8210 return DEAD_OBJECT; 8211 } 8212 8213 // TODO: handle configuration of effects replacing track process 8214 if (thread->channelCount() == 1) { 8215 channels = AUDIO_CHANNEL_OUT_MONO; 8216 } else { 8217 channels = AUDIO_CHANNEL_OUT_STEREO; 8218 } 8219 8220 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8221 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8222 } else { 8223 mConfig.inputCfg.channels = channels; 8224 } 8225 mConfig.outputCfg.channels = channels; 8226 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8227 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8228 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8229 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8230 mConfig.inputCfg.bufferProvider.cookie = NULL; 8231 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8232 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8233 mConfig.outputCfg.bufferProvider.cookie = NULL; 8234 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8235 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8236 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8237 // Insert effect: 8238 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8239 // always overwrites output buffer: input buffer == output buffer 8240 // - in other sessions: 8241 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8242 // other effect: overwrites output buffer: input buffer == output buffer 8243 // Auxiliary effect: 8244 // accumulates in output buffer: input buffer != output buffer 8245 // Therefore: accumulate <=> input buffer != output buffer 8246 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8247 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8248 } else { 8249 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8250 } 8251 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8252 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8253 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8254 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8255 8256 ALOGV("configure() %p thread %p buffer %p framecount %d", 8257 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8258 8259 status_t cmdStatus; 8260 uint32_t size = sizeof(int); 8261 status_t status = (*mEffectInterface)->command(mEffectInterface, 8262 EFFECT_CMD_SET_CONFIG, 8263 sizeof(effect_config_t), 8264 &mConfig, 8265 &size, 8266 &cmdStatus); 8267 if (status == 0) { 8268 status = cmdStatus; 8269 } 8270 8271 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8272 (1000 * mConfig.outputCfg.buffer.frameCount); 8273 8274 return status; 8275} 8276 8277status_t AudioFlinger::EffectModule::init() 8278{ 8279 Mutex::Autolock _l(mLock); 8280 if (mEffectInterface == NULL) { 8281 return NO_INIT; 8282 } 8283 status_t cmdStatus; 8284 uint32_t size = sizeof(status_t); 8285 status_t status = (*mEffectInterface)->command(mEffectInterface, 8286 EFFECT_CMD_INIT, 8287 0, 8288 NULL, 8289 &size, 8290 &cmdStatus); 8291 if (status == 0) { 8292 status = cmdStatus; 8293 } 8294 return status; 8295} 8296 8297status_t AudioFlinger::EffectModule::start() 8298{ 8299 Mutex::Autolock _l(mLock); 8300 return start_l(); 8301} 8302 8303status_t AudioFlinger::EffectModule::start_l() 8304{ 8305 if (mEffectInterface == NULL) { 8306 return NO_INIT; 8307 } 8308 status_t cmdStatus; 8309 uint32_t size = sizeof(status_t); 8310 status_t status = (*mEffectInterface)->command(mEffectInterface, 8311 EFFECT_CMD_ENABLE, 8312 0, 8313 NULL, 8314 &size, 8315 &cmdStatus); 8316 if (status == 0) { 8317 status = cmdStatus; 8318 } 8319 if (status == 0 && 8320 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8321 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8322 sp<ThreadBase> thread = mThread.promote(); 8323 if (thread != 0) { 8324 audio_stream_t *stream = thread->stream(); 8325 if (stream != NULL) { 8326 stream->add_audio_effect(stream, mEffectInterface); 8327 } 8328 } 8329 } 8330 return status; 8331} 8332 8333status_t AudioFlinger::EffectModule::stop() 8334{ 8335 Mutex::Autolock _l(mLock); 8336 return stop_l(); 8337} 8338 8339status_t AudioFlinger::EffectModule::stop_l() 8340{ 8341 if (mEffectInterface == NULL) { 8342 return NO_INIT; 8343 } 8344 status_t cmdStatus; 8345 uint32_t size = sizeof(status_t); 8346 status_t status = (*mEffectInterface)->command(mEffectInterface, 8347 EFFECT_CMD_DISABLE, 8348 0, 8349 NULL, 8350 &size, 8351 &cmdStatus); 8352 if (status == 0) { 8353 status = cmdStatus; 8354 } 8355 if (status == 0 && 8356 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8357 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8358 sp<ThreadBase> thread = mThread.promote(); 8359 if (thread != 0) { 8360 audio_stream_t *stream = thread->stream(); 8361 if (stream != NULL) { 8362 stream->remove_audio_effect(stream, mEffectInterface); 8363 } 8364 } 8365 } 8366 return status; 8367} 8368 8369status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8370 uint32_t cmdSize, 8371 void *pCmdData, 8372 uint32_t *replySize, 8373 void *pReplyData) 8374{ 8375 Mutex::Autolock _l(mLock); 8376// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8377 8378 if (mState == DESTROYED || mEffectInterface == NULL) { 8379 return NO_INIT; 8380 } 8381 status_t status = (*mEffectInterface)->command(mEffectInterface, 8382 cmdCode, 8383 cmdSize, 8384 pCmdData, 8385 replySize, 8386 pReplyData); 8387 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8388 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8389 for (size_t i = 1; i < mHandles.size(); i++) { 8390 sp<EffectHandle> h = mHandles[i].promote(); 8391 if (h != 0) { 8392 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8393 } 8394 } 8395 } 8396 return status; 8397} 8398 8399status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8400{ 8401 8402 Mutex::Autolock _l(mLock); 8403 ALOGV("setEnabled %p enabled %d", this, enabled); 8404 8405 if (enabled != isEnabled()) { 8406 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8407 if (enabled && status != NO_ERROR) { 8408 return status; 8409 } 8410 8411 switch (mState) { 8412 // going from disabled to enabled 8413 case IDLE: 8414 mState = STARTING; 8415 break; 8416 case STOPPED: 8417 mState = RESTART; 8418 break; 8419 case STOPPING: 8420 mState = ACTIVE; 8421 break; 8422 8423 // going from enabled to disabled 8424 case RESTART: 8425 mState = STOPPED; 8426 break; 8427 case STARTING: 8428 mState = IDLE; 8429 break; 8430 case ACTIVE: 8431 mState = STOPPING; 8432 break; 8433 case DESTROYED: 8434 return NO_ERROR; // simply ignore as we are being destroyed 8435 } 8436 for (size_t i = 1; i < mHandles.size(); i++) { 8437 sp<EffectHandle> h = mHandles[i].promote(); 8438 if (h != 0) { 8439 h->setEnabled(enabled); 8440 } 8441 } 8442 } 8443 return NO_ERROR; 8444} 8445 8446bool AudioFlinger::EffectModule::isEnabled() const 8447{ 8448 switch (mState) { 8449 case RESTART: 8450 case STARTING: 8451 case ACTIVE: 8452 return true; 8453 case IDLE: 8454 case STOPPING: 8455 case STOPPED: 8456 case DESTROYED: 8457 default: 8458 return false; 8459 } 8460} 8461 8462bool AudioFlinger::EffectModule::isProcessEnabled() const 8463{ 8464 switch (mState) { 8465 case RESTART: 8466 case ACTIVE: 8467 case STOPPING: 8468 case STOPPED: 8469 return true; 8470 case IDLE: 8471 case STARTING: 8472 case DESTROYED: 8473 default: 8474 return false; 8475 } 8476} 8477 8478status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8479{ 8480 Mutex::Autolock _l(mLock); 8481 status_t status = NO_ERROR; 8482 8483 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8484 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8485 if (isProcessEnabled() && 8486 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8487 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8488 status_t cmdStatus; 8489 uint32_t volume[2]; 8490 uint32_t *pVolume = NULL; 8491 uint32_t size = sizeof(volume); 8492 volume[0] = *left; 8493 volume[1] = *right; 8494 if (controller) { 8495 pVolume = volume; 8496 } 8497 status = (*mEffectInterface)->command(mEffectInterface, 8498 EFFECT_CMD_SET_VOLUME, 8499 size, 8500 volume, 8501 &size, 8502 pVolume); 8503 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8504 *left = volume[0]; 8505 *right = volume[1]; 8506 } 8507 } 8508 return status; 8509} 8510 8511status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8512{ 8513 Mutex::Autolock _l(mLock); 8514 status_t status = NO_ERROR; 8515 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8516 // audio pre processing modules on RecordThread can receive both output and 8517 // input device indication in the same call 8518 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8519 if (dev) { 8520 status_t cmdStatus; 8521 uint32_t size = sizeof(status_t); 8522 8523 status = (*mEffectInterface)->command(mEffectInterface, 8524 EFFECT_CMD_SET_DEVICE, 8525 sizeof(uint32_t), 8526 &dev, 8527 &size, 8528 &cmdStatus); 8529 if (status == NO_ERROR) { 8530 status = cmdStatus; 8531 } 8532 } 8533 dev = device & AUDIO_DEVICE_IN_ALL; 8534 if (dev) { 8535 status_t cmdStatus; 8536 uint32_t size = sizeof(status_t); 8537 8538 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8539 EFFECT_CMD_SET_INPUT_DEVICE, 8540 sizeof(uint32_t), 8541 &dev, 8542 &size, 8543 &cmdStatus); 8544 if (status2 == NO_ERROR) { 8545 status2 = cmdStatus; 8546 } 8547 if (status == NO_ERROR) { 8548 status = status2; 8549 } 8550 } 8551 } 8552 return status; 8553} 8554 8555status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8556{ 8557 Mutex::Autolock _l(mLock); 8558 status_t status = NO_ERROR; 8559 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8560 status_t cmdStatus; 8561 uint32_t size = sizeof(status_t); 8562 status = (*mEffectInterface)->command(mEffectInterface, 8563 EFFECT_CMD_SET_AUDIO_MODE, 8564 sizeof(audio_mode_t), 8565 &mode, 8566 &size, 8567 &cmdStatus); 8568 if (status == NO_ERROR) { 8569 status = cmdStatus; 8570 } 8571 } 8572 return status; 8573} 8574 8575void AudioFlinger::EffectModule::setSuspended(bool suspended) 8576{ 8577 Mutex::Autolock _l(mLock); 8578 mSuspended = suspended; 8579} 8580 8581bool AudioFlinger::EffectModule::suspended() const 8582{ 8583 Mutex::Autolock _l(mLock); 8584 return mSuspended; 8585} 8586 8587status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8588{ 8589 const size_t SIZE = 256; 8590 char buffer[SIZE]; 8591 String8 result; 8592 8593 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8594 result.append(buffer); 8595 8596 bool locked = tryLock(mLock); 8597 // failed to lock - AudioFlinger is probably deadlocked 8598 if (!locked) { 8599 result.append("\t\tCould not lock Fx mutex:\n"); 8600 } 8601 8602 result.append("\t\tSession Status State Engine:\n"); 8603 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8604 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8605 result.append(buffer); 8606 8607 result.append("\t\tDescriptor:\n"); 8608 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8609 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8610 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8611 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8612 result.append(buffer); 8613 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8614 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8615 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8616 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8617 result.append(buffer); 8618 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8619 mDescriptor.apiVersion, 8620 mDescriptor.flags); 8621 result.append(buffer); 8622 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8623 mDescriptor.name); 8624 result.append(buffer); 8625 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8626 mDescriptor.implementor); 8627 result.append(buffer); 8628 8629 result.append("\t\t- Input configuration:\n"); 8630 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8631 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8632 (uint32_t)mConfig.inputCfg.buffer.raw, 8633 mConfig.inputCfg.buffer.frameCount, 8634 mConfig.inputCfg.samplingRate, 8635 mConfig.inputCfg.channels, 8636 mConfig.inputCfg.format); 8637 result.append(buffer); 8638 8639 result.append("\t\t- Output configuration:\n"); 8640 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8641 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8642 (uint32_t)mConfig.outputCfg.buffer.raw, 8643 mConfig.outputCfg.buffer.frameCount, 8644 mConfig.outputCfg.samplingRate, 8645 mConfig.outputCfg.channels, 8646 mConfig.outputCfg.format); 8647 result.append(buffer); 8648 8649 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8650 result.append(buffer); 8651 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8652 for (size_t i = 0; i < mHandles.size(); ++i) { 8653 sp<EffectHandle> handle = mHandles[i].promote(); 8654 if (handle != 0) { 8655 handle->dump(buffer, SIZE); 8656 result.append(buffer); 8657 } 8658 } 8659 8660 result.append("\n"); 8661 8662 write(fd, result.string(), result.length()); 8663 8664 if (locked) { 8665 mLock.unlock(); 8666 } 8667 8668 return NO_ERROR; 8669} 8670 8671// ---------------------------------------------------------------------------- 8672// EffectHandle implementation 8673// ---------------------------------------------------------------------------- 8674 8675#undef LOG_TAG 8676#define LOG_TAG "AudioFlinger::EffectHandle" 8677 8678AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8679 const sp<AudioFlinger::Client>& client, 8680 const sp<IEffectClient>& effectClient, 8681 int32_t priority) 8682 : BnEffect(), 8683 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8684 mPriority(priority), mHasControl(false), mEnabled(false) 8685{ 8686 ALOGV("constructor %p", this); 8687 8688 if (client == 0) { 8689 return; 8690 } 8691 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8692 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8693 if (mCblkMemory != 0) { 8694 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8695 8696 if (mCblk != NULL) { 8697 new(mCblk) effect_param_cblk_t(); 8698 mBuffer = (uint8_t *)mCblk + bufOffset; 8699 } 8700 } else { 8701 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8702 return; 8703 } 8704} 8705 8706AudioFlinger::EffectHandle::~EffectHandle() 8707{ 8708 ALOGV("Destructor %p", this); 8709 disconnect(false); 8710 ALOGV("Destructor DONE %p", this); 8711} 8712 8713status_t AudioFlinger::EffectHandle::enable() 8714{ 8715 ALOGV("enable %p", this); 8716 if (!mHasControl) return INVALID_OPERATION; 8717 if (mEffect == 0) return DEAD_OBJECT; 8718 8719 if (mEnabled) { 8720 return NO_ERROR; 8721 } 8722 8723 mEnabled = true; 8724 8725 sp<ThreadBase> thread = mEffect->thread().promote(); 8726 if (thread != 0) { 8727 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8728 } 8729 8730 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8731 if (mEffect->suspended()) { 8732 return NO_ERROR; 8733 } 8734 8735 status_t status = mEffect->setEnabled(true); 8736 if (status != NO_ERROR) { 8737 if (thread != 0) { 8738 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8739 } 8740 mEnabled = false; 8741 } 8742 return status; 8743} 8744 8745status_t AudioFlinger::EffectHandle::disable() 8746{ 8747 ALOGV("disable %p", this); 8748 if (!mHasControl) return INVALID_OPERATION; 8749 if (mEffect == 0) return DEAD_OBJECT; 8750 8751 if (!mEnabled) { 8752 return NO_ERROR; 8753 } 8754 mEnabled = false; 8755 8756 if (mEffect->suspended()) { 8757 return NO_ERROR; 8758 } 8759 8760 status_t status = mEffect->setEnabled(false); 8761 8762 sp<ThreadBase> thread = mEffect->thread().promote(); 8763 if (thread != 0) { 8764 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8765 } 8766 8767 return status; 8768} 8769 8770void AudioFlinger::EffectHandle::disconnect() 8771{ 8772 disconnect(true); 8773} 8774 8775void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8776{ 8777 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8778 if (mEffect == 0) { 8779 return; 8780 } 8781 mEffect->disconnect(this, unpinIfLast); 8782 8783 if (mHasControl && mEnabled) { 8784 sp<ThreadBase> thread = mEffect->thread().promote(); 8785 if (thread != 0) { 8786 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8787 } 8788 } 8789 8790 // release sp on module => module destructor can be called now 8791 mEffect.clear(); 8792 if (mClient != 0) { 8793 if (mCblk != NULL) { 8794 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8795 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8796 } 8797 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8798 // Client destructor must run with AudioFlinger mutex locked 8799 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8800 mClient.clear(); 8801 } 8802} 8803 8804status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8805 uint32_t cmdSize, 8806 void *pCmdData, 8807 uint32_t *replySize, 8808 void *pReplyData) 8809{ 8810// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8811// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8812 8813 // only get parameter command is permitted for applications not controlling the effect 8814 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8815 return INVALID_OPERATION; 8816 } 8817 if (mEffect == 0) return DEAD_OBJECT; 8818 if (mClient == 0) return INVALID_OPERATION; 8819 8820 // handle commands that are not forwarded transparently to effect engine 8821 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8822 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8823 // no risk to block the whole media server process or mixer threads is we are stuck here 8824 Mutex::Autolock _l(mCblk->lock); 8825 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8826 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8827 mCblk->serverIndex = 0; 8828 mCblk->clientIndex = 0; 8829 return BAD_VALUE; 8830 } 8831 status_t status = NO_ERROR; 8832 while (mCblk->serverIndex < mCblk->clientIndex) { 8833 int reply; 8834 uint32_t rsize = sizeof(int); 8835 int *p = (int *)(mBuffer + mCblk->serverIndex); 8836 int size = *p++; 8837 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8838 ALOGW("command(): invalid parameter block size"); 8839 break; 8840 } 8841 effect_param_t *param = (effect_param_t *)p; 8842 if (param->psize == 0 || param->vsize == 0) { 8843 ALOGW("command(): null parameter or value size"); 8844 mCblk->serverIndex += size; 8845 continue; 8846 } 8847 uint32_t psize = sizeof(effect_param_t) + 8848 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8849 param->vsize; 8850 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8851 psize, 8852 p, 8853 &rsize, 8854 &reply); 8855 // stop at first error encountered 8856 if (ret != NO_ERROR) { 8857 status = ret; 8858 *(int *)pReplyData = reply; 8859 break; 8860 } else if (reply != NO_ERROR) { 8861 *(int *)pReplyData = reply; 8862 break; 8863 } 8864 mCblk->serverIndex += size; 8865 } 8866 mCblk->serverIndex = 0; 8867 mCblk->clientIndex = 0; 8868 return status; 8869 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8870 *(int *)pReplyData = NO_ERROR; 8871 return enable(); 8872 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8873 *(int *)pReplyData = NO_ERROR; 8874 return disable(); 8875 } 8876 8877 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8878} 8879 8880void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8881{ 8882 ALOGV("setControl %p control %d", this, hasControl); 8883 8884 mHasControl = hasControl; 8885 mEnabled = enabled; 8886 8887 if (signal && mEffectClient != 0) { 8888 mEffectClient->controlStatusChanged(hasControl); 8889 } 8890} 8891 8892void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8893 uint32_t cmdSize, 8894 void *pCmdData, 8895 uint32_t replySize, 8896 void *pReplyData) 8897{ 8898 if (mEffectClient != 0) { 8899 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8900 } 8901} 8902 8903 8904 8905void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8906{ 8907 if (mEffectClient != 0) { 8908 mEffectClient->enableStatusChanged(enabled); 8909 } 8910} 8911 8912status_t AudioFlinger::EffectHandle::onTransact( 8913 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8914{ 8915 return BnEffect::onTransact(code, data, reply, flags); 8916} 8917 8918 8919void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8920{ 8921 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8922 8923 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8924 (mClient == 0) ? getpid_cached : mClient->pid(), 8925 mPriority, 8926 mHasControl, 8927 !locked, 8928 mCblk ? mCblk->clientIndex : 0, 8929 mCblk ? mCblk->serverIndex : 0 8930 ); 8931 8932 if (locked) { 8933 mCblk->lock.unlock(); 8934 } 8935} 8936 8937#undef LOG_TAG 8938#define LOG_TAG "AudioFlinger::EffectChain" 8939 8940AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8941 int sessionId) 8942 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8943 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8944 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8945{ 8946 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8947 if (thread == NULL) { 8948 return; 8949 } 8950 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8951 thread->frameCount(); 8952} 8953 8954AudioFlinger::EffectChain::~EffectChain() 8955{ 8956 if (mOwnInBuffer) { 8957 delete mInBuffer; 8958 } 8959 8960} 8961 8962// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8963sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8964{ 8965 size_t size = mEffects.size(); 8966 8967 for (size_t i = 0; i < size; i++) { 8968 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8969 return mEffects[i]; 8970 } 8971 } 8972 return 0; 8973} 8974 8975// getEffectFromId_l() must be called with ThreadBase::mLock held 8976sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8977{ 8978 size_t size = mEffects.size(); 8979 8980 for (size_t i = 0; i < size; i++) { 8981 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8982 if (id == 0 || mEffects[i]->id() == id) { 8983 return mEffects[i]; 8984 } 8985 } 8986 return 0; 8987} 8988 8989// getEffectFromType_l() must be called with ThreadBase::mLock held 8990sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8991 const effect_uuid_t *type) 8992{ 8993 size_t size = mEffects.size(); 8994 8995 for (size_t i = 0; i < size; i++) { 8996 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8997 return mEffects[i]; 8998 } 8999 } 9000 return 0; 9001} 9002 9003void AudioFlinger::EffectChain::clearInputBuffer() 9004{ 9005 Mutex::Autolock _l(mLock); 9006 sp<ThreadBase> thread = mThread.promote(); 9007 if (thread == 0) { 9008 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9009 return; 9010 } 9011 clearInputBuffer_l(thread); 9012} 9013 9014// Must be called with EffectChain::mLock locked 9015void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9016{ 9017 size_t numSamples = thread->frameCount() * thread->channelCount(); 9018 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9019 9020} 9021 9022// Must be called with EffectChain::mLock locked 9023void AudioFlinger::EffectChain::process_l() 9024{ 9025 sp<ThreadBase> thread = mThread.promote(); 9026 if (thread == 0) { 9027 ALOGW("process_l(): cannot promote mixer thread"); 9028 return; 9029 } 9030 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9031 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9032 // always process effects unless no more tracks are on the session and the effect tail 9033 // has been rendered 9034 bool doProcess = true; 9035 if (!isGlobalSession) { 9036 bool tracksOnSession = (trackCnt() != 0); 9037 9038 if (!tracksOnSession && mTailBufferCount == 0) { 9039 doProcess = false; 9040 } 9041 9042 if (activeTrackCnt() == 0) { 9043 // if no track is active and the effect tail has not been rendered, 9044 // the input buffer must be cleared here as the mixer process will not do it 9045 if (tracksOnSession || mTailBufferCount > 0) { 9046 clearInputBuffer_l(thread); 9047 if (mTailBufferCount > 0) { 9048 mTailBufferCount--; 9049 } 9050 } 9051 } 9052 } 9053 9054 size_t size = mEffects.size(); 9055 if (doProcess) { 9056 for (size_t i = 0; i < size; i++) { 9057 mEffects[i]->process(); 9058 } 9059 } 9060 for (size_t i = 0; i < size; i++) { 9061 mEffects[i]->updateState(); 9062 } 9063} 9064 9065// addEffect_l() must be called with PlaybackThread::mLock held 9066status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9067{ 9068 effect_descriptor_t desc = effect->desc(); 9069 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9070 9071 Mutex::Autolock _l(mLock); 9072 effect->setChain(this); 9073 sp<ThreadBase> thread = mThread.promote(); 9074 if (thread == 0) { 9075 return NO_INIT; 9076 } 9077 effect->setThread(thread); 9078 9079 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9080 // Auxiliary effects are inserted at the beginning of mEffects vector as 9081 // they are processed first and accumulated in chain input buffer 9082 mEffects.insertAt(effect, 0); 9083 9084 // the input buffer for auxiliary effect contains mono samples in 9085 // 32 bit format. This is to avoid saturation in AudoMixer 9086 // accumulation stage. Saturation is done in EffectModule::process() before 9087 // calling the process in effect engine 9088 size_t numSamples = thread->frameCount(); 9089 int32_t *buffer = new int32_t[numSamples]; 9090 memset(buffer, 0, numSamples * sizeof(int32_t)); 9091 effect->setInBuffer((int16_t *)buffer); 9092 // auxiliary effects output samples to chain input buffer for further processing 9093 // by insert effects 9094 effect->setOutBuffer(mInBuffer); 9095 } else { 9096 // Insert effects are inserted at the end of mEffects vector as they are processed 9097 // after track and auxiliary effects. 9098 // Insert effect order as a function of indicated preference: 9099 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9100 // another effect is present 9101 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9102 // last effect claiming first position 9103 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9104 // first effect claiming last position 9105 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9106 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9107 // already present 9108 9109 size_t size = mEffects.size(); 9110 size_t idx_insert = size; 9111 ssize_t idx_insert_first = -1; 9112 ssize_t idx_insert_last = -1; 9113 9114 for (size_t i = 0; i < size; i++) { 9115 effect_descriptor_t d = mEffects[i]->desc(); 9116 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9117 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9118 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9119 // check invalid effect chaining combinations 9120 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9121 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9122 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9123 return INVALID_OPERATION; 9124 } 9125 // remember position of first insert effect and by default 9126 // select this as insert position for new effect 9127 if (idx_insert == size) { 9128 idx_insert = i; 9129 } 9130 // remember position of last insert effect claiming 9131 // first position 9132 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9133 idx_insert_first = i; 9134 } 9135 // remember position of first insert effect claiming 9136 // last position 9137 if (iPref == EFFECT_FLAG_INSERT_LAST && 9138 idx_insert_last == -1) { 9139 idx_insert_last = i; 9140 } 9141 } 9142 } 9143 9144 // modify idx_insert from first position if needed 9145 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9146 if (idx_insert_last != -1) { 9147 idx_insert = idx_insert_last; 9148 } else { 9149 idx_insert = size; 9150 } 9151 } else { 9152 if (idx_insert_first != -1) { 9153 idx_insert = idx_insert_first + 1; 9154 } 9155 } 9156 9157 // always read samples from chain input buffer 9158 effect->setInBuffer(mInBuffer); 9159 9160 // if last effect in the chain, output samples to chain 9161 // output buffer, otherwise to chain input buffer 9162 if (idx_insert == size) { 9163 if (idx_insert != 0) { 9164 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9165 mEffects[idx_insert-1]->configure(); 9166 } 9167 effect->setOutBuffer(mOutBuffer); 9168 } else { 9169 effect->setOutBuffer(mInBuffer); 9170 } 9171 mEffects.insertAt(effect, idx_insert); 9172 9173 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9174 } 9175 effect->configure(); 9176 return NO_ERROR; 9177} 9178 9179// removeEffect_l() must be called with PlaybackThread::mLock held 9180size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9181{ 9182 Mutex::Autolock _l(mLock); 9183 size_t size = mEffects.size(); 9184 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9185 9186 for (size_t i = 0; i < size; i++) { 9187 if (effect == mEffects[i]) { 9188 // calling stop here will remove pre-processing effect from the audio HAL. 9189 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9190 // the middle of a read from audio HAL 9191 if (mEffects[i]->state() == EffectModule::ACTIVE || 9192 mEffects[i]->state() == EffectModule::STOPPING) { 9193 mEffects[i]->stop(); 9194 } 9195 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9196 delete[] effect->inBuffer(); 9197 } else { 9198 if (i == size - 1 && i != 0) { 9199 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9200 mEffects[i - 1]->configure(); 9201 } 9202 } 9203 mEffects.removeAt(i); 9204 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9205 break; 9206 } 9207 } 9208 9209 return mEffects.size(); 9210} 9211 9212// setDevice_l() must be called with PlaybackThread::mLock held 9213void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9214{ 9215 size_t size = mEffects.size(); 9216 for (size_t i = 0; i < size; i++) { 9217 mEffects[i]->setDevice(device); 9218 } 9219} 9220 9221// setMode_l() must be called with PlaybackThread::mLock held 9222void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9223{ 9224 size_t size = mEffects.size(); 9225 for (size_t i = 0; i < size; i++) { 9226 mEffects[i]->setMode(mode); 9227 } 9228} 9229 9230// setVolume_l() must be called with PlaybackThread::mLock held 9231bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9232{ 9233 uint32_t newLeft = *left; 9234 uint32_t newRight = *right; 9235 bool hasControl = false; 9236 int ctrlIdx = -1; 9237 size_t size = mEffects.size(); 9238 9239 // first update volume controller 9240 for (size_t i = size; i > 0; i--) { 9241 if (mEffects[i - 1]->isProcessEnabled() && 9242 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9243 ctrlIdx = i - 1; 9244 hasControl = true; 9245 break; 9246 } 9247 } 9248 9249 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9250 if (hasControl) { 9251 *left = mNewLeftVolume; 9252 *right = mNewRightVolume; 9253 } 9254 return hasControl; 9255 } 9256 9257 mVolumeCtrlIdx = ctrlIdx; 9258 mLeftVolume = newLeft; 9259 mRightVolume = newRight; 9260 9261 // second get volume update from volume controller 9262 if (ctrlIdx >= 0) { 9263 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9264 mNewLeftVolume = newLeft; 9265 mNewRightVolume = newRight; 9266 } 9267 // then indicate volume to all other effects in chain. 9268 // Pass altered volume to effects before volume controller 9269 // and requested volume to effects after controller 9270 uint32_t lVol = newLeft; 9271 uint32_t rVol = newRight; 9272 9273 for (size_t i = 0; i < size; i++) { 9274 if ((int)i == ctrlIdx) continue; 9275 // this also works for ctrlIdx == -1 when there is no volume controller 9276 if ((int)i > ctrlIdx) { 9277 lVol = *left; 9278 rVol = *right; 9279 } 9280 mEffects[i]->setVolume(&lVol, &rVol, false); 9281 } 9282 *left = newLeft; 9283 *right = newRight; 9284 9285 return hasControl; 9286} 9287 9288status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9289{ 9290 const size_t SIZE = 256; 9291 char buffer[SIZE]; 9292 String8 result; 9293 9294 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9295 result.append(buffer); 9296 9297 bool locked = tryLock(mLock); 9298 // failed to lock - AudioFlinger is probably deadlocked 9299 if (!locked) { 9300 result.append("\tCould not lock mutex:\n"); 9301 } 9302 9303 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9304 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9305 mEffects.size(), 9306 (uint32_t)mInBuffer, 9307 (uint32_t)mOutBuffer, 9308 mActiveTrackCnt); 9309 result.append(buffer); 9310 write(fd, result.string(), result.size()); 9311 9312 for (size_t i = 0; i < mEffects.size(); ++i) { 9313 sp<EffectModule> effect = mEffects[i]; 9314 if (effect != 0) { 9315 effect->dump(fd, args); 9316 } 9317 } 9318 9319 if (locked) { 9320 mLock.unlock(); 9321 } 9322 9323 return NO_ERROR; 9324} 9325 9326// must be called with ThreadBase::mLock held 9327void AudioFlinger::EffectChain::setEffectSuspended_l( 9328 const effect_uuid_t *type, bool suspend) 9329{ 9330 sp<SuspendedEffectDesc> desc; 9331 // use effect type UUID timelow as key as there is no real risk of identical 9332 // timeLow fields among effect type UUIDs. 9333 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9334 if (suspend) { 9335 if (index >= 0) { 9336 desc = mSuspendedEffects.valueAt(index); 9337 } else { 9338 desc = new SuspendedEffectDesc(); 9339 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9340 mSuspendedEffects.add(type->timeLow, desc); 9341 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9342 } 9343 if (desc->mRefCount++ == 0) { 9344 sp<EffectModule> effect = getEffectIfEnabled(type); 9345 if (effect != 0) { 9346 desc->mEffect = effect; 9347 effect->setSuspended(true); 9348 effect->setEnabled(false); 9349 } 9350 } 9351 } else { 9352 if (index < 0) { 9353 return; 9354 } 9355 desc = mSuspendedEffects.valueAt(index); 9356 if (desc->mRefCount <= 0) { 9357 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9358 desc->mRefCount = 1; 9359 } 9360 if (--desc->mRefCount == 0) { 9361 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9362 if (desc->mEffect != 0) { 9363 sp<EffectModule> effect = desc->mEffect.promote(); 9364 if (effect != 0) { 9365 effect->setSuspended(false); 9366 sp<EffectHandle> handle = effect->controlHandle(); 9367 if (handle != 0) { 9368 effect->setEnabled(handle->enabled()); 9369 } 9370 } 9371 desc->mEffect.clear(); 9372 } 9373 mSuspendedEffects.removeItemsAt(index); 9374 } 9375 } 9376} 9377 9378// must be called with ThreadBase::mLock held 9379void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9380{ 9381 sp<SuspendedEffectDesc> desc; 9382 9383 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9384 if (suspend) { 9385 if (index >= 0) { 9386 desc = mSuspendedEffects.valueAt(index); 9387 } else { 9388 desc = new SuspendedEffectDesc(); 9389 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9390 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9391 } 9392 if (desc->mRefCount++ == 0) { 9393 Vector< sp<EffectModule> > effects; 9394 getSuspendEligibleEffects(effects); 9395 for (size_t i = 0; i < effects.size(); i++) { 9396 setEffectSuspended_l(&effects[i]->desc().type, true); 9397 } 9398 } 9399 } else { 9400 if (index < 0) { 9401 return; 9402 } 9403 desc = mSuspendedEffects.valueAt(index); 9404 if (desc->mRefCount <= 0) { 9405 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9406 desc->mRefCount = 1; 9407 } 9408 if (--desc->mRefCount == 0) { 9409 Vector<const effect_uuid_t *> types; 9410 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9411 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9412 continue; 9413 } 9414 types.add(&mSuspendedEffects.valueAt(i)->mType); 9415 } 9416 for (size_t i = 0; i < types.size(); i++) { 9417 setEffectSuspended_l(types[i], false); 9418 } 9419 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9420 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9421 } 9422 } 9423} 9424 9425 9426// The volume effect is used for automated tests only 9427#ifndef OPENSL_ES_H_ 9428static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9429 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9430const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9431#endif //OPENSL_ES_H_ 9432 9433bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9434{ 9435 // auxiliary effects and visualizer are never suspended on output mix 9436 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9437 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9438 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9439 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9440 return false; 9441 } 9442 return true; 9443} 9444 9445void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9446{ 9447 effects.clear(); 9448 for (size_t i = 0; i < mEffects.size(); i++) { 9449 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9450 effects.add(mEffects[i]); 9451 } 9452 } 9453} 9454 9455sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9456 const effect_uuid_t *type) 9457{ 9458 sp<EffectModule> effect = getEffectFromType_l(type); 9459 return effect != 0 && effect->isEnabled() ? effect : 0; 9460} 9461 9462void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9463 bool enabled) 9464{ 9465 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9466 if (enabled) { 9467 if (index < 0) { 9468 // if the effect is not suspend check if all effects are suspended 9469 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9470 if (index < 0) { 9471 return; 9472 } 9473 if (!isEffectEligibleForSuspend(effect->desc())) { 9474 return; 9475 } 9476 setEffectSuspended_l(&effect->desc().type, enabled); 9477 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9478 if (index < 0) { 9479 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9480 return; 9481 } 9482 } 9483 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9484 effect->desc().type.timeLow); 9485 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9486 // if effect is requested to suspended but was not yet enabled, supend it now. 9487 if (desc->mEffect == 0) { 9488 desc->mEffect = effect; 9489 effect->setEnabled(false); 9490 effect->setSuspended(true); 9491 } 9492 } else { 9493 if (index < 0) { 9494 return; 9495 } 9496 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9497 effect->desc().type.timeLow); 9498 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9499 desc->mEffect.clear(); 9500 effect->setSuspended(false); 9501 } 9502} 9503 9504#undef LOG_TAG 9505#define LOG_TAG "AudioFlinger" 9506 9507// ---------------------------------------------------------------------------- 9508 9509status_t AudioFlinger::onTransact( 9510 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9511{ 9512 return BnAudioFlinger::onTransact(code, data, reply, flags); 9513} 9514 9515}; // namespace android 9516