AudioFlinger.cpp revision 24a2fd0113da60785ce5af5dd905f8aaf9e0f0a1
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 ALOGV("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 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2118 // decoding and transfer time. So sleeping for half of the latency would likely cause 2119 // underruns 2120 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2121 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2122 } else { 2123 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2124 } 2125} 2126 2127status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2128{ 2129 if (!isValidSyncEvent(event)) { 2130 return BAD_VALUE; 2131 } 2132 2133 Mutex::Autolock _l(mLock); 2134 2135 for (size_t i = 0; i < mTracks.size(); ++i) { 2136 sp<Track> track = mTracks[i]; 2137 if (event->triggerSession() == track->sessionId()) { 2138 track->setSyncEvent(event); 2139 return NO_ERROR; 2140 } 2141 } 2142 2143 return NAME_NOT_FOUND; 2144} 2145 2146bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2147{ 2148 switch (event->type()) { 2149 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2150 return true; 2151 default: 2152 break; 2153 } 2154 return false; 2155} 2156 2157void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2158{ 2159 size_t count = tracksToRemove.size(); 2160 if (CC_UNLIKELY(count)) { 2161 for (size_t i = 0 ; i < count ; i++) { 2162 const sp<Track>& track = tracksToRemove.itemAt(i); 2163 if ((track->sharedBuffer() != 0) && 2164 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2165 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2166 } 2167 } 2168 } 2169 2170} 2171 2172// ---------------------------------------------------------------------------- 2173 2174AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2175 audio_io_handle_t id, uint32_t device, type_t type) 2176 : PlaybackThread(audioFlinger, output, id, device, type), 2177 // mAudioMixer below 2178#ifdef SOAKER 2179 mSoaker(NULL), 2180#endif 2181 // mFastMixer below 2182 mFastMixerFutex(0) 2183 // mOutputSink below 2184 // mPipeSink below 2185 // mNormalSink below 2186{ 2187 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2188 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2189 "mFrameCount=%d, mNormalFrameCount=%d", 2190 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2191 mNormalFrameCount); 2192 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2193 2194 // FIXME - Current mixer implementation only supports stereo output 2195 if (mChannelCount == 1) { 2196 ALOGE("Invalid audio hardware channel count"); 2197 } 2198 2199 // create an NBAIO sink for the HAL output stream, and negotiate 2200 mOutputSink = new AudioStreamOutSink(output->stream); 2201 size_t numCounterOffers = 0; 2202 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2203 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2204 ALOG_ASSERT(index == 0); 2205 2206 // initialize fast mixer depending on configuration 2207 bool initFastMixer; 2208 switch (kUseFastMixer) { 2209 case FastMixer_Never: 2210 initFastMixer = false; 2211 break; 2212 case FastMixer_Always: 2213 initFastMixer = true; 2214 break; 2215 case FastMixer_Static: 2216 case FastMixer_Dynamic: 2217 initFastMixer = mFrameCount < mNormalFrameCount; 2218 break; 2219 } 2220 if (initFastMixer) { 2221 2222 // create a MonoPipe to connect our submix to FastMixer 2223 NBAIO_Format format = mOutputSink->format(); 2224 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2225 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2226 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2227 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2228 const NBAIO_Format offers[1] = {format}; 2229 size_t numCounterOffers = 0; 2230 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2231 ALOG_ASSERT(index == 0); 2232 mPipeSink = monoPipe; 2233 2234#ifdef TEE_SINK_FRAMES 2235 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2236 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2237 numCounterOffers = 0; 2238 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2239 ALOG_ASSERT(index == 0); 2240 mTeeSink = teeSink; 2241 PipeReader *teeSource = new PipeReader(*teeSink); 2242 numCounterOffers = 0; 2243 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2244 ALOG_ASSERT(index == 0); 2245 mTeeSource = teeSource; 2246#endif 2247 2248#ifdef SOAKER 2249 // create a soaker as workaround for governor issues 2250 mSoaker = new Soaker(); 2251 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2252 mSoaker->run("Soaker", PRIORITY_LOWEST); 2253#endif 2254 2255 // create fast mixer and configure it initially with just one fast track for our submix 2256 mFastMixer = new FastMixer(); 2257 FastMixerStateQueue *sq = mFastMixer->sq(); 2258#ifdef STATE_QUEUE_DUMP 2259 sq->setObserverDump(&mStateQueueObserverDump); 2260 sq->setMutatorDump(&mStateQueueMutatorDump); 2261#endif 2262 FastMixerState *state = sq->begin(); 2263 FastTrack *fastTrack = &state->mFastTracks[0]; 2264 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2265 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2266 fastTrack->mVolumeProvider = NULL; 2267 fastTrack->mGeneration++; 2268 state->mFastTracksGen++; 2269 state->mTrackMask = 1; 2270 // fast mixer will use the HAL output sink 2271 state->mOutputSink = mOutputSink.get(); 2272 state->mOutputSinkGen++; 2273 state->mFrameCount = mFrameCount; 2274 state->mCommand = FastMixerState::COLD_IDLE; 2275 // already done in constructor initialization list 2276 //mFastMixerFutex = 0; 2277 state->mColdFutexAddr = &mFastMixerFutex; 2278 state->mColdGen++; 2279 state->mDumpState = &mFastMixerDumpState; 2280 state->mTeeSink = mTeeSink.get(); 2281 sq->end(); 2282 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2283 2284 // start the fast mixer 2285 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2286#ifdef HAVE_REQUEST_PRIORITY 2287 pid_t tid = mFastMixer->getTid(); 2288 int err = requestPriority(getpid_cached, tid, 2); 2289 if (err != 0) { 2290 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2291 2, getpid_cached, tid, err); 2292 } 2293#endif 2294 2295 } else { 2296 mFastMixer = NULL; 2297 } 2298 2299 switch (kUseFastMixer) { 2300 case FastMixer_Never: 2301 case FastMixer_Dynamic: 2302 mNormalSink = mOutputSink; 2303 break; 2304 case FastMixer_Always: 2305 mNormalSink = mPipeSink; 2306 break; 2307 case FastMixer_Static: 2308 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2309 break; 2310 } 2311} 2312 2313AudioFlinger::MixerThread::~MixerThread() 2314{ 2315 if (mFastMixer != NULL) { 2316 FastMixerStateQueue *sq = mFastMixer->sq(); 2317 FastMixerState *state = sq->begin(); 2318 if (state->mCommand == FastMixerState::COLD_IDLE) { 2319 int32_t old = android_atomic_inc(&mFastMixerFutex); 2320 if (old == -1) { 2321 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2322 } 2323 } 2324 state->mCommand = FastMixerState::EXIT; 2325 sq->end(); 2326 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2327 mFastMixer->join(); 2328 // Though the fast mixer thread has exited, it's state queue is still valid. 2329 // We'll use that extract the final state which contains one remaining fast track 2330 // corresponding to our sub-mix. 2331 state = sq->begin(); 2332 ALOG_ASSERT(state->mTrackMask == 1); 2333 FastTrack *fastTrack = &state->mFastTracks[0]; 2334 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2335 delete fastTrack->mBufferProvider; 2336 sq->end(false /*didModify*/); 2337 delete mFastMixer; 2338#ifdef SOAKER 2339 if (mSoaker != NULL) { 2340 mSoaker->requestExitAndWait(); 2341 } 2342 delete mSoaker; 2343#endif 2344 } 2345 delete mAudioMixer; 2346} 2347 2348class CpuStats { 2349public: 2350 CpuStats(); 2351 void sample(const String8 &title); 2352#ifdef DEBUG_CPU_USAGE 2353private: 2354 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2355 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2356 2357 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2358 2359 int mCpuNum; // thread's current CPU number 2360 int mCpukHz; // frequency of thread's current CPU in kHz 2361#endif 2362}; 2363 2364CpuStats::CpuStats() 2365#ifdef DEBUG_CPU_USAGE 2366 : mCpuNum(-1), mCpukHz(-1) 2367#endif 2368{ 2369} 2370 2371void CpuStats::sample(const String8 &title) { 2372#ifdef DEBUG_CPU_USAGE 2373 // get current thread's delta CPU time in wall clock ns 2374 double wcNs; 2375 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2376 2377 // record sample for wall clock statistics 2378 if (valid) { 2379 mWcStats.sample(wcNs); 2380 } 2381 2382 // get the current CPU number 2383 int cpuNum = sched_getcpu(); 2384 2385 // get the current CPU frequency in kHz 2386 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2387 2388 // check if either CPU number or frequency changed 2389 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2390 mCpuNum = cpuNum; 2391 mCpukHz = cpukHz; 2392 // ignore sample for purposes of cycles 2393 valid = false; 2394 } 2395 2396 // if no change in CPU number or frequency, then record sample for cycle statistics 2397 if (valid && mCpukHz > 0) { 2398 double cycles = wcNs * cpukHz * 0.000001; 2399 mHzStats.sample(cycles); 2400 } 2401 2402 unsigned n = mWcStats.n(); 2403 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2404 if ((n & 127) == 1) { 2405 long long elapsed = mCpuUsage.elapsed(); 2406 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2407 double perLoop = elapsed / (double) n; 2408 double perLoop100 = perLoop * 0.01; 2409 double perLoop1k = perLoop * 0.001; 2410 double mean = mWcStats.mean(); 2411 double stddev = mWcStats.stddev(); 2412 double minimum = mWcStats.minimum(); 2413 double maximum = mWcStats.maximum(); 2414 double meanCycles = mHzStats.mean(); 2415 double stddevCycles = mHzStats.stddev(); 2416 double minCycles = mHzStats.minimum(); 2417 double maxCycles = mHzStats.maximum(); 2418 mCpuUsage.resetElapsed(); 2419 mWcStats.reset(); 2420 mHzStats.reset(); 2421 ALOGD("CPU usage for %s over past %.1f secs\n" 2422 " (%u mixer loops at %.1f mean ms per loop):\n" 2423 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2424 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2425 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2426 title.string(), 2427 elapsed * .000000001, n, perLoop * .000001, 2428 mean * .001, 2429 stddev * .001, 2430 minimum * .001, 2431 maximum * .001, 2432 mean / perLoop100, 2433 stddev / perLoop100, 2434 minimum / perLoop100, 2435 maximum / perLoop100, 2436 meanCycles / perLoop1k, 2437 stddevCycles / perLoop1k, 2438 minCycles / perLoop1k, 2439 maxCycles / perLoop1k); 2440 2441 } 2442 } 2443#endif 2444}; 2445 2446void AudioFlinger::PlaybackThread::checkSilentMode_l() 2447{ 2448 if (!mMasterMute) { 2449 char value[PROPERTY_VALUE_MAX]; 2450 if (property_get("ro.audio.silent", value, "0") > 0) { 2451 char *endptr; 2452 unsigned long ul = strtoul(value, &endptr, 0); 2453 if (*endptr == '\0' && ul != 0) { 2454 ALOGD("Silence is golden"); 2455 // The setprop command will not allow a property to be changed after 2456 // the first time it is set, so we don't have to worry about un-muting. 2457 setMasterMute_l(true); 2458 } 2459 } 2460 } 2461} 2462 2463bool AudioFlinger::PlaybackThread::threadLoop() 2464{ 2465 Vector< sp<Track> > tracksToRemove; 2466 2467 standbyTime = systemTime(); 2468 2469 // MIXER 2470 nsecs_t lastWarning = 0; 2471if (mType == MIXER) { 2472 longStandbyExit = false; 2473} 2474 2475 // DUPLICATING 2476 // FIXME could this be made local to while loop? 2477 writeFrames = 0; 2478 2479 cacheParameters_l(); 2480 sleepTime = idleSleepTime; 2481 2482if (mType == MIXER) { 2483 sleepTimeShift = 0; 2484} 2485 2486 CpuStats cpuStats; 2487 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2488 2489 acquireWakeLock(); 2490 2491 while (!exitPending()) 2492 { 2493 cpuStats.sample(myName); 2494 2495 Vector< sp<EffectChain> > effectChains; 2496 2497 processConfigEvents(); 2498 2499 { // scope for mLock 2500 2501 Mutex::Autolock _l(mLock); 2502 2503 if (checkForNewParameters_l()) { 2504 cacheParameters_l(); 2505 } 2506 2507 saveOutputTracks(); 2508 2509 // put audio hardware into standby after short delay 2510 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2511 mSuspended > 0)) { 2512 if (!mStandby) { 2513 2514 threadLoop_standby(); 2515 2516 mStandby = true; 2517 mBytesWritten = 0; 2518 } 2519 2520 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2521 // we're about to wait, flush the binder command buffer 2522 IPCThreadState::self()->flushCommands(); 2523 2524 clearOutputTracks(); 2525 2526 if (exitPending()) break; 2527 2528 releaseWakeLock_l(); 2529 // wait until we have something to do... 2530 ALOGV("%s going to sleep", myName.string()); 2531 mWaitWorkCV.wait(mLock); 2532 ALOGV("%s waking up", myName.string()); 2533 acquireWakeLock_l(); 2534 2535 mMixerStatus = MIXER_IDLE; 2536 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2537 2538 checkSilentMode_l(); 2539 2540 standbyTime = systemTime() + standbyDelay; 2541 sleepTime = idleSleepTime; 2542 if (mType == MIXER) { 2543 sleepTimeShift = 0; 2544 } 2545 2546 continue; 2547 } 2548 } 2549 2550 // mMixerStatusIgnoringFastTracks is also updated internally 2551 mMixerStatus = prepareTracks_l(&tracksToRemove); 2552 2553 // prevent any changes in effect chain list and in each effect chain 2554 // during mixing and effect process as the audio buffers could be deleted 2555 // or modified if an effect is created or deleted 2556 lockEffectChains_l(effectChains); 2557 } 2558 2559 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2560 threadLoop_mix(); 2561 } else { 2562 threadLoop_sleepTime(); 2563 } 2564 2565 if (mSuspended > 0) { 2566 sleepTime = suspendSleepTimeUs(); 2567 } 2568 2569 // only process effects if we're going to write 2570 if (sleepTime == 0) { 2571 for (size_t i = 0; i < effectChains.size(); i ++) { 2572 effectChains[i]->process_l(); 2573 } 2574 } 2575 2576 // enable changes in effect chain 2577 unlockEffectChains(effectChains); 2578 2579 // sleepTime == 0 means we must write to audio hardware 2580 if (sleepTime == 0) { 2581 2582 threadLoop_write(); 2583 2584if (mType == MIXER) { 2585 // write blocked detection 2586 nsecs_t now = systemTime(); 2587 nsecs_t delta = now - mLastWriteTime; 2588 if (!mStandby && delta > maxPeriod) { 2589 mNumDelayedWrites++; 2590 if ((now - lastWarning) > kWarningThrottleNs) { 2591#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2592 ScopedTrace st(ATRACE_TAG, "underrun"); 2593#endif 2594 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2595 ns2ms(delta), mNumDelayedWrites, this); 2596 lastWarning = now; 2597 } 2598 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2599 // a different threshold. Or completely removed for what it is worth anyway... 2600 if (mStandby) { 2601 longStandbyExit = true; 2602 } 2603 } 2604} 2605 2606 mStandby = false; 2607 } else { 2608 usleep(sleepTime); 2609 } 2610 2611 // Finally let go of removed track(s), without the lock held 2612 // since we can't guarantee the destructors won't acquire that 2613 // same lock. This will also mutate and push a new fast mixer state. 2614 threadLoop_removeTracks(tracksToRemove); 2615 tracksToRemove.clear(); 2616 2617 // FIXME I don't understand the need for this here; 2618 // it was in the original code but maybe the 2619 // assignment in saveOutputTracks() makes this unnecessary? 2620 clearOutputTracks(); 2621 2622 // Effect chains will be actually deleted here if they were removed from 2623 // mEffectChains list during mixing or effects processing 2624 effectChains.clear(); 2625 2626 // FIXME Note that the above .clear() is no longer necessary since effectChains 2627 // is now local to this block, but will keep it for now (at least until merge done). 2628 } 2629 2630if (mType == MIXER || mType == DIRECT) { 2631 // put output stream into standby mode 2632 if (!mStandby) { 2633 mOutput->stream->common.standby(&mOutput->stream->common); 2634 } 2635} 2636if (mType == DUPLICATING) { 2637 // for DuplicatingThread, standby mode is handled by the outputTracks 2638} 2639 2640 releaseWakeLock(); 2641 2642 ALOGV("Thread %p type %d exiting", this, mType); 2643 return false; 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2647{ 2648 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2649} 2650 2651void AudioFlinger::MixerThread::threadLoop_write() 2652{ 2653 // FIXME we should only do one push per cycle; confirm this is true 2654 // Start the fast mixer if it's not already running 2655 if (mFastMixer != NULL) { 2656 FastMixerStateQueue *sq = mFastMixer->sq(); 2657 FastMixerState *state = sq->begin(); 2658 if (state->mCommand != FastMixerState::MIX_WRITE && 2659 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2660 if (state->mCommand == FastMixerState::COLD_IDLE) { 2661 int32_t old = android_atomic_inc(&mFastMixerFutex); 2662 if (old == -1) { 2663 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2664 } 2665 } 2666 state->mCommand = FastMixerState::MIX_WRITE; 2667 sq->end(); 2668 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2669 if (kUseFastMixer == FastMixer_Dynamic) { 2670 mNormalSink = mPipeSink; 2671 } 2672 } else { 2673 sq->end(false /*didModify*/); 2674 } 2675 } 2676 PlaybackThread::threadLoop_write(); 2677} 2678 2679// shared by MIXER and DIRECT, overridden by DUPLICATING 2680void AudioFlinger::PlaybackThread::threadLoop_write() 2681{ 2682 // FIXME rewrite to reduce number of system calls 2683 mLastWriteTime = systemTime(); 2684 mInWrite = true; 2685 2686#define mBitShift 2 // FIXME 2687 size_t count = mixBufferSize >> mBitShift; 2688#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2689 Tracer::traceBegin(ATRACE_TAG, "write"); 2690#endif 2691 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2692#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2693 Tracer::traceEnd(ATRACE_TAG); 2694#endif 2695 if (framesWritten > 0) { 2696 size_t bytesWritten = framesWritten << mBitShift; 2697 mBytesWritten += bytesWritten; 2698 } 2699 2700 mNumWrites++; 2701 mInWrite = false; 2702} 2703 2704void AudioFlinger::MixerThread::threadLoop_standby() 2705{ 2706 // Idle the fast mixer if it's currently running 2707 if (mFastMixer != NULL) { 2708 FastMixerStateQueue *sq = mFastMixer->sq(); 2709 FastMixerState *state = sq->begin(); 2710 if (!(state->mCommand & FastMixerState::IDLE)) { 2711 state->mCommand = FastMixerState::COLD_IDLE; 2712 state->mColdFutexAddr = &mFastMixerFutex; 2713 state->mColdGen++; 2714 mFastMixerFutex = 0; 2715 sq->end(); 2716 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2717 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2718 if (kUseFastMixer == FastMixer_Dynamic) { 2719 mNormalSink = mOutputSink; 2720 } 2721 } else { 2722 sq->end(false /*didModify*/); 2723 } 2724 } 2725 PlaybackThread::threadLoop_standby(); 2726} 2727 2728// shared by MIXER and DIRECT, overridden by DUPLICATING 2729void AudioFlinger::PlaybackThread::threadLoop_standby() 2730{ 2731 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2732 mOutput->stream->common.standby(&mOutput->stream->common); 2733} 2734 2735void AudioFlinger::MixerThread::threadLoop_mix() 2736{ 2737 // obtain the presentation timestamp of the next output buffer 2738 int64_t pts; 2739 status_t status = INVALID_OPERATION; 2740 2741 if (NULL != mOutput->stream->get_next_write_timestamp) { 2742 status = mOutput->stream->get_next_write_timestamp( 2743 mOutput->stream, &pts); 2744 } 2745 2746 if (status != NO_ERROR) { 2747 pts = AudioBufferProvider::kInvalidPTS; 2748 } 2749 2750 // mix buffers... 2751 mAudioMixer->process(pts); 2752 // increase sleep time progressively when application underrun condition clears. 2753 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2754 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2755 // such that we would underrun the audio HAL. 2756 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2757 sleepTimeShift--; 2758 } 2759 sleepTime = 0; 2760 standbyTime = systemTime() + standbyDelay; 2761 //TODO: delay standby when effects have a tail 2762} 2763 2764void AudioFlinger::MixerThread::threadLoop_sleepTime() 2765{ 2766 // If no tracks are ready, sleep once for the duration of an output 2767 // buffer size, then write 0s to the output 2768 if (sleepTime == 0) { 2769 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2770 sleepTime = activeSleepTime >> sleepTimeShift; 2771 if (sleepTime < kMinThreadSleepTimeUs) { 2772 sleepTime = kMinThreadSleepTimeUs; 2773 } 2774 // reduce sleep time in case of consecutive application underruns to avoid 2775 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2776 // duration we would end up writing less data than needed by the audio HAL if 2777 // the condition persists. 2778 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2779 sleepTimeShift++; 2780 } 2781 } else { 2782 sleepTime = idleSleepTime; 2783 } 2784 } else if (mBytesWritten != 0 || 2785 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2786 memset (mMixBuffer, 0, mixBufferSize); 2787 sleepTime = 0; 2788 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2789 } 2790 // TODO add standby time extension fct of effect tail 2791} 2792 2793// prepareTracks_l() must be called with ThreadBase::mLock held 2794AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2795 Vector< sp<Track> > *tracksToRemove) 2796{ 2797 2798 mixer_state mixerStatus = MIXER_IDLE; 2799 // find out which tracks need to be processed 2800 size_t count = mActiveTracks.size(); 2801 size_t mixedTracks = 0; 2802 size_t tracksWithEffect = 0; 2803 // counts only _active_ fast tracks 2804 size_t fastTracks = 0; 2805 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2806 2807 float masterVolume = mMasterVolume; 2808 bool masterMute = mMasterMute; 2809 2810 if (masterMute) { 2811 masterVolume = 0; 2812 } 2813 // Delegate master volume control to effect in output mix effect chain if needed 2814 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2815 if (chain != 0) { 2816 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2817 chain->setVolume_l(&v, &v); 2818 masterVolume = (float)((v + (1 << 23)) >> 24); 2819 chain.clear(); 2820 } 2821 2822 // prepare a new state to push 2823 FastMixerStateQueue *sq = NULL; 2824 FastMixerState *state = NULL; 2825 bool didModify = false; 2826 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2827 if (mFastMixer != NULL) { 2828 sq = mFastMixer->sq(); 2829 state = sq->begin(); 2830 } 2831 2832 for (size_t i=0 ; i<count ; i++) { 2833 sp<Track> t = mActiveTracks[i].promote(); 2834 if (t == 0) continue; 2835 2836 // this const just means the local variable doesn't change 2837 Track* const track = t.get(); 2838 2839 // process fast tracks 2840 if (track->isFastTrack()) { 2841 2842 // It's theoretically possible (though unlikely) for a fast track to be created 2843 // and then removed within the same normal mix cycle. This is not a problem, as 2844 // the track never becomes active so it's fast mixer slot is never touched. 2845 // The converse, of removing an (active) track and then creating a new track 2846 // at the identical fast mixer slot within the same normal mix cycle, 2847 // is impossible because the slot isn't marked available until the end of each cycle. 2848 int j = track->mFastIndex; 2849 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2850 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2851 FastTrack *fastTrack = &state->mFastTracks[j]; 2852 2853 // Determine whether the track is currently in underrun condition, 2854 // and whether it had a recent underrun. 2855 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2856 FastTrackUnderruns underruns = ftDump->mUnderruns; 2857 uint32_t recentFull = (underruns.mBitFields.mFull - 2858 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2859 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2860 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2861 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2862 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2863 uint32_t recentUnderruns = recentPartial + recentEmpty; 2864 track->mObservedUnderruns = underruns; 2865 // don't count underruns that occur while stopping or pausing 2866 // or stopped which can occur when flush() is called while active 2867 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2868 track->mUnderrunCount += recentUnderruns; 2869 } 2870 2871 // This is similar to the state machine for normal tracks, 2872 // with a few modifications for fast tracks. 2873 bool isActive = true; 2874 switch (track->mState) { 2875 case TrackBase::STOPPING_1: 2876 // track stays active in STOPPING_1 state until first underrun 2877 if (recentUnderruns > 0) { 2878 track->mState = TrackBase::STOPPING_2; 2879 } 2880 break; 2881 case TrackBase::PAUSING: 2882 // ramp down is not yet implemented 2883 track->setPaused(); 2884 break; 2885 case TrackBase::RESUMING: 2886 // ramp up is not yet implemented 2887 track->mState = TrackBase::ACTIVE; 2888 break; 2889 case TrackBase::ACTIVE: 2890 if (recentFull > 0 || recentPartial > 0) { 2891 // track has provided at least some frames recently: reset retry count 2892 track->mRetryCount = kMaxTrackRetries; 2893 } 2894 if (recentUnderruns == 0) { 2895 // no recent underruns: stay active 2896 break; 2897 } 2898 // there has recently been an underrun of some kind 2899 if (track->sharedBuffer() == 0) { 2900 // were any of the recent underruns "empty" (no frames available)? 2901 if (recentEmpty == 0) { 2902 // no, then ignore the partial underruns as they are allowed indefinitely 2903 break; 2904 } 2905 // there has recently been an "empty" underrun: decrement the retry counter 2906 if (--(track->mRetryCount) > 0) { 2907 break; 2908 } 2909 // indicate to client process that the track was disabled because of underrun; 2910 // it will then automatically call start() when data is available 2911 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2912 // remove from active list, but state remains ACTIVE [confusing but true] 2913 isActive = false; 2914 break; 2915 } 2916 // fall through 2917 case TrackBase::STOPPING_2: 2918 case TrackBase::PAUSED: 2919 case TrackBase::TERMINATED: 2920 case TrackBase::STOPPED: 2921 case TrackBase::FLUSHED: // flush() while active 2922 // Check for presentation complete if track is inactive 2923 // We have consumed all the buffers of this track. 2924 // This would be incomplete if we auto-paused on underrun 2925 { 2926 size_t audioHALFrames = 2927 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2928 size_t framesWritten = 2929 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2930 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2931 // track stays in active list until presentation is complete 2932 break; 2933 } 2934 } 2935 if (track->isStopping_2()) { 2936 track->mState = TrackBase::STOPPED; 2937 } 2938 if (track->isStopped()) { 2939 // Can't reset directly, as fast mixer is still polling this track 2940 // track->reset(); 2941 // So instead mark this track as needing to be reset after push with ack 2942 resetMask |= 1 << i; 2943 } 2944 isActive = false; 2945 break; 2946 case TrackBase::IDLE: 2947 default: 2948 LOG_FATAL("unexpected track state %d", track->mState); 2949 } 2950 2951 if (isActive) { 2952 // was it previously inactive? 2953 if (!(state->mTrackMask & (1 << j))) { 2954 ExtendedAudioBufferProvider *eabp = track; 2955 VolumeProvider *vp = track; 2956 fastTrack->mBufferProvider = eabp; 2957 fastTrack->mVolumeProvider = vp; 2958 fastTrack->mSampleRate = track->mSampleRate; 2959 fastTrack->mChannelMask = track->mChannelMask; 2960 fastTrack->mGeneration++; 2961 state->mTrackMask |= 1 << j; 2962 didModify = true; 2963 // no acknowledgement required for newly active tracks 2964 } 2965 // cache the combined master volume and stream type volume for fast mixer; this 2966 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2967 track->mCachedVolume = track->isMuted() ? 2968 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2969 ++fastTracks; 2970 } else { 2971 // was it previously active? 2972 if (state->mTrackMask & (1 << j)) { 2973 fastTrack->mBufferProvider = NULL; 2974 fastTrack->mGeneration++; 2975 state->mTrackMask &= ~(1 << j); 2976 didModify = true; 2977 // If any fast tracks were removed, we must wait for acknowledgement 2978 // because we're about to decrement the last sp<> on those tracks. 2979 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2980 } else { 2981 LOG_FATAL("fast track %d should have been active", j); 2982 } 2983 tracksToRemove->add(track); 2984 // Avoids a misleading display in dumpsys 2985 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2986 } 2987 continue; 2988 } 2989 2990 { // local variable scope to avoid goto warning 2991 2992 audio_track_cblk_t* cblk = track->cblk(); 2993 2994 // The first time a track is added we wait 2995 // for all its buffers to be filled before processing it 2996 int name = track->name(); 2997 // make sure that we have enough frames to mix one full buffer. 2998 // enforce this condition only once to enable draining the buffer in case the client 2999 // app does not call stop() and relies on underrun to stop: 3000 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3001 // during last round 3002 uint32_t minFrames = 1; 3003 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3004 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3005 if (t->sampleRate() == (int)mSampleRate) { 3006 minFrames = mNormalFrameCount; 3007 } else { 3008 // +1 for rounding and +1 for additional sample needed for interpolation 3009 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3010 // add frames already consumed but not yet released by the resampler 3011 // because cblk->framesReady() will include these frames 3012 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3013 // the minimum track buffer size is normally twice the number of frames necessary 3014 // to fill one buffer and the resampler should not leave more than one buffer worth 3015 // of unreleased frames after each pass, but just in case... 3016 ALOG_ASSERT(minFrames <= cblk->frameCount); 3017 } 3018 } 3019 if ((track->framesReady() >= minFrames) && track->isReady() && 3020 !track->isPaused() && !track->isTerminated()) 3021 { 3022 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3023 3024 mixedTracks++; 3025 3026 // track->mainBuffer() != mMixBuffer means there is an effect chain 3027 // connected to the track 3028 chain.clear(); 3029 if (track->mainBuffer() != mMixBuffer) { 3030 chain = getEffectChain_l(track->sessionId()); 3031 // Delegate volume control to effect in track effect chain if needed 3032 if (chain != 0) { 3033 tracksWithEffect++; 3034 } else { 3035 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3036 name, track->sessionId()); 3037 } 3038 } 3039 3040 3041 int param = AudioMixer::VOLUME; 3042 if (track->mFillingUpStatus == Track::FS_FILLED) { 3043 // no ramp for the first volume setting 3044 track->mFillingUpStatus = Track::FS_ACTIVE; 3045 if (track->mState == TrackBase::RESUMING) { 3046 track->mState = TrackBase::ACTIVE; 3047 param = AudioMixer::RAMP_VOLUME; 3048 } 3049 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3050 } else if (cblk->server != 0) { 3051 // If the track is stopped before the first frame was mixed, 3052 // do not apply ramp 3053 param = AudioMixer::RAMP_VOLUME; 3054 } 3055 3056 // compute volume for this track 3057 uint32_t vl, vr, va; 3058 if (track->isMuted() || track->isPausing() || 3059 mStreamTypes[track->streamType()].mute) { 3060 vl = vr = va = 0; 3061 if (track->isPausing()) { 3062 track->setPaused(); 3063 } 3064 } else { 3065 3066 // read original volumes with volume control 3067 float typeVolume = mStreamTypes[track->streamType()].volume; 3068 float v = masterVolume * typeVolume; 3069 uint32_t vlr = cblk->getVolumeLR(); 3070 vl = vlr & 0xFFFF; 3071 vr = vlr >> 16; 3072 // track volumes come from shared memory, so can't be trusted and must be clamped 3073 if (vl > MAX_GAIN_INT) { 3074 ALOGV("Track left volume out of range: %04X", vl); 3075 vl = MAX_GAIN_INT; 3076 } 3077 if (vr > MAX_GAIN_INT) { 3078 ALOGV("Track right volume out of range: %04X", vr); 3079 vr = MAX_GAIN_INT; 3080 } 3081 // now apply the master volume and stream type volume 3082 vl = (uint32_t)(v * vl) << 12; 3083 vr = (uint32_t)(v * vr) << 12; 3084 // assuming master volume and stream type volume each go up to 1.0, 3085 // vl and vr are now in 8.24 format 3086 3087 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3088 // send level comes from shared memory and so may be corrupt 3089 if (sendLevel > MAX_GAIN_INT) { 3090 ALOGV("Track send level out of range: %04X", sendLevel); 3091 sendLevel = MAX_GAIN_INT; 3092 } 3093 va = (uint32_t)(v * sendLevel); 3094 } 3095 // Delegate volume control to effect in track effect chain if needed 3096 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3097 // Do not ramp volume if volume is controlled by effect 3098 param = AudioMixer::VOLUME; 3099 track->mHasVolumeController = true; 3100 } else { 3101 // force no volume ramp when volume controller was just disabled or removed 3102 // from effect chain to avoid volume spike 3103 if (track->mHasVolumeController) { 3104 param = AudioMixer::VOLUME; 3105 } 3106 track->mHasVolumeController = false; 3107 } 3108 3109 // Convert volumes from 8.24 to 4.12 format 3110 // This additional clamping is needed in case chain->setVolume_l() overshot 3111 vl = (vl + (1 << 11)) >> 12; 3112 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3113 vr = (vr + (1 << 11)) >> 12; 3114 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3115 3116 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3117 3118 // XXX: these things DON'T need to be done each time 3119 mAudioMixer->setBufferProvider(name, track); 3120 mAudioMixer->enable(name); 3121 3122 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3123 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3124 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3125 mAudioMixer->setParameter( 3126 name, 3127 AudioMixer::TRACK, 3128 AudioMixer::FORMAT, (void *)track->format()); 3129 mAudioMixer->setParameter( 3130 name, 3131 AudioMixer::TRACK, 3132 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3133 mAudioMixer->setParameter( 3134 name, 3135 AudioMixer::RESAMPLE, 3136 AudioMixer::SAMPLE_RATE, 3137 (void *)(cblk->sampleRate)); 3138 mAudioMixer->setParameter( 3139 name, 3140 AudioMixer::TRACK, 3141 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3142 mAudioMixer->setParameter( 3143 name, 3144 AudioMixer::TRACK, 3145 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3146 3147 // reset retry count 3148 track->mRetryCount = kMaxTrackRetries; 3149 3150 // If one track is ready, set the mixer ready if: 3151 // - the mixer was not ready during previous round OR 3152 // - no other track is not ready 3153 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3154 mixerStatus != MIXER_TRACKS_ENABLED) { 3155 mixerStatus = MIXER_TRACKS_READY; 3156 } 3157 } else { 3158 // clear effect chain input buffer if an active track underruns to avoid sending 3159 // previous audio buffer again to effects 3160 chain = getEffectChain_l(track->sessionId()); 3161 if (chain != 0) { 3162 chain->clearInputBuffer(); 3163 } 3164 3165 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3166 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3167 track->isStopped() || track->isPaused()) { 3168 // We have consumed all the buffers of this track. 3169 // Remove it from the list of active tracks. 3170 // TODO: use actual buffer filling status instead of latency when available from 3171 // audio HAL 3172 size_t audioHALFrames = 3173 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3174 size_t framesWritten = 3175 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3176 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3177 if (track->isStopped()) { 3178 track->reset(); 3179 } 3180 tracksToRemove->add(track); 3181 } 3182 } else { 3183 track->mUnderrunCount++; 3184 // No buffers for this track. Give it a few chances to 3185 // fill a buffer, then remove it from active list. 3186 if (--(track->mRetryCount) <= 0) { 3187 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3188 tracksToRemove->add(track); 3189 // indicate to client process that the track was disabled because of underrun; 3190 // it will then automatically call start() when data is available 3191 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3192 // If one track is not ready, mark the mixer also not ready if: 3193 // - the mixer was ready during previous round OR 3194 // - no other track is ready 3195 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3196 mixerStatus != MIXER_TRACKS_READY) { 3197 mixerStatus = MIXER_TRACKS_ENABLED; 3198 } 3199 } 3200 mAudioMixer->disable(name); 3201 } 3202 3203 } // local variable scope to avoid goto warning 3204track_is_ready: ; 3205 3206 } 3207 3208 // Push the new FastMixer state if necessary 3209 if (didModify) { 3210 state->mFastTracksGen++; 3211 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3212 if (kUseFastMixer == FastMixer_Dynamic && 3213 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3214 state->mCommand = FastMixerState::COLD_IDLE; 3215 state->mColdFutexAddr = &mFastMixerFutex; 3216 state->mColdGen++; 3217 mFastMixerFutex = 0; 3218 if (kUseFastMixer == FastMixer_Dynamic) { 3219 mNormalSink = mOutputSink; 3220 } 3221 // If we go into cold idle, need to wait for acknowledgement 3222 // so that fast mixer stops doing I/O. 3223 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3224 } 3225 sq->end(); 3226 } 3227 if (sq != NULL) { 3228 sq->end(didModify); 3229 sq->push(block); 3230 } 3231 3232 // Now perform the deferred reset on fast tracks that have stopped 3233 while (resetMask != 0) { 3234 size_t i = __builtin_ctz(resetMask); 3235 ALOG_ASSERT(i < count); 3236 resetMask &= ~(1 << i); 3237 sp<Track> t = mActiveTracks[i].promote(); 3238 if (t == 0) continue; 3239 Track* track = t.get(); 3240 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3241 track->reset(); 3242 } 3243 3244 // remove all the tracks that need to be... 3245 count = tracksToRemove->size(); 3246 if (CC_UNLIKELY(count)) { 3247 for (size_t i=0 ; i<count ; i++) { 3248 const sp<Track>& track = tracksToRemove->itemAt(i); 3249 mActiveTracks.remove(track); 3250 if (track->mainBuffer() != mMixBuffer) { 3251 chain = getEffectChain_l(track->sessionId()); 3252 if (chain != 0) { 3253 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3254 chain->decActiveTrackCnt(); 3255 } 3256 } 3257 if (track->isTerminated()) { 3258 removeTrack_l(track); 3259 } 3260 } 3261 } 3262 3263 // mix buffer must be cleared if all tracks are connected to an 3264 // effect chain as in this case the mixer will not write to 3265 // mix buffer and track effects will accumulate into it 3266 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3267 // FIXME as a performance optimization, should remember previous zero status 3268 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3269 } 3270 3271 // if any fast tracks, then status is ready 3272 mMixerStatusIgnoringFastTracks = mixerStatus; 3273 if (fastTracks > 0) { 3274 mixerStatus = MIXER_TRACKS_READY; 3275 } 3276 return mixerStatus; 3277} 3278 3279/* 3280The derived values that are cached: 3281 - mixBufferSize from frame count * frame size 3282 - activeSleepTime from activeSleepTimeUs() 3283 - idleSleepTime from idleSleepTimeUs() 3284 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3285 - maxPeriod from frame count and sample rate (MIXER only) 3286 3287The parameters that affect these derived values are: 3288 - frame count 3289 - frame size 3290 - sample rate 3291 - device type: A2DP or not 3292 - device latency 3293 - format: PCM or not 3294 - active sleep time 3295 - idle sleep time 3296*/ 3297 3298void AudioFlinger::PlaybackThread::cacheParameters_l() 3299{ 3300 mixBufferSize = mNormalFrameCount * mFrameSize; 3301 activeSleepTime = activeSleepTimeUs(); 3302 idleSleepTime = idleSleepTimeUs(); 3303} 3304 3305void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3306{ 3307 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3308 this, streamType, mTracks.size()); 3309 Mutex::Autolock _l(mLock); 3310 3311 size_t size = mTracks.size(); 3312 for (size_t i = 0; i < size; i++) { 3313 sp<Track> t = mTracks[i]; 3314 if (t->streamType() == streamType) { 3315 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3316 t->mCblk->cv.signal(); 3317 } 3318 } 3319} 3320 3321// getTrackName_l() must be called with ThreadBase::mLock held 3322int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3323{ 3324 return mAudioMixer->getTrackName(channelMask); 3325} 3326 3327// deleteTrackName_l() must be called with ThreadBase::mLock held 3328void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3329{ 3330 ALOGV("remove track (%d) and delete from mixer", name); 3331 mAudioMixer->deleteTrackName(name); 3332} 3333 3334// checkForNewParameters_l() must be called with ThreadBase::mLock held 3335bool AudioFlinger::MixerThread::checkForNewParameters_l() 3336{ 3337 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3338 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3339 bool reconfig = false; 3340 3341 while (!mNewParameters.isEmpty()) { 3342 3343 if (mFastMixer != NULL) { 3344 FastMixerStateQueue *sq = mFastMixer->sq(); 3345 FastMixerState *state = sq->begin(); 3346 if (!(state->mCommand & FastMixerState::IDLE)) { 3347 previousCommand = state->mCommand; 3348 state->mCommand = FastMixerState::HOT_IDLE; 3349 sq->end(); 3350 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3351 } else { 3352 sq->end(false /*didModify*/); 3353 } 3354 } 3355 3356 status_t status = NO_ERROR; 3357 String8 keyValuePair = mNewParameters[0]; 3358 AudioParameter param = AudioParameter(keyValuePair); 3359 int value; 3360 3361 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3362 reconfig = true; 3363 } 3364 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3365 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3366 status = BAD_VALUE; 3367 } else { 3368 reconfig = true; 3369 } 3370 } 3371 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3372 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3373 status = BAD_VALUE; 3374 } else { 3375 reconfig = true; 3376 } 3377 } 3378 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3379 // do not accept frame count changes if tracks are open as the track buffer 3380 // size depends on frame count and correct behavior would not be guaranteed 3381 // if frame count is changed after track creation 3382 if (!mTracks.isEmpty()) { 3383 status = INVALID_OPERATION; 3384 } else { 3385 reconfig = true; 3386 } 3387 } 3388 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3389#ifdef ADD_BATTERY_DATA 3390 // when changing the audio output device, call addBatteryData to notify 3391 // the change 3392 if ((int)mDevice != value) { 3393 uint32_t params = 0; 3394 // check whether speaker is on 3395 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3396 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3397 } 3398 3399 int deviceWithoutSpeaker 3400 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3401 // check if any other device (except speaker) is on 3402 if (value & deviceWithoutSpeaker ) { 3403 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3404 } 3405 3406 if (params != 0) { 3407 addBatteryData(params); 3408 } 3409 } 3410#endif 3411 3412 // forward device change to effects that have requested to be 3413 // aware of attached audio device. 3414 mDevice = (uint32_t)value; 3415 for (size_t i = 0; i < mEffectChains.size(); i++) { 3416 mEffectChains[i]->setDevice_l(mDevice); 3417 } 3418 } 3419 3420 if (status == NO_ERROR) { 3421 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3422 keyValuePair.string()); 3423 if (!mStandby && status == INVALID_OPERATION) { 3424 mOutput->stream->common.standby(&mOutput->stream->common); 3425 mStandby = true; 3426 mBytesWritten = 0; 3427 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3428 keyValuePair.string()); 3429 } 3430 if (status == NO_ERROR && reconfig) { 3431 delete mAudioMixer; 3432 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3433 mAudioMixer = NULL; 3434 readOutputParameters(); 3435 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3436 for (size_t i = 0; i < mTracks.size() ; i++) { 3437 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3438 if (name < 0) break; 3439 mTracks[i]->mName = name; 3440 // limit track sample rate to 2 x new output sample rate 3441 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3442 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3443 } 3444 } 3445 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3446 } 3447 } 3448 3449 mNewParameters.removeAt(0); 3450 3451 mParamStatus = status; 3452 mParamCond.signal(); 3453 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3454 // already timed out waiting for the status and will never signal the condition. 3455 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3456 } 3457 3458 if (!(previousCommand & FastMixerState::IDLE)) { 3459 ALOG_ASSERT(mFastMixer != NULL); 3460 FastMixerStateQueue *sq = mFastMixer->sq(); 3461 FastMixerState *state = sq->begin(); 3462 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3463 state->mCommand = previousCommand; 3464 sq->end(); 3465 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3466 } 3467 3468 return reconfig; 3469} 3470 3471status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3472{ 3473 const size_t SIZE = 256; 3474 char buffer[SIZE]; 3475 String8 result; 3476 3477 PlaybackThread::dumpInternals(fd, args); 3478 3479 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3480 result.append(buffer); 3481 write(fd, result.string(), result.size()); 3482 3483 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3484 FastMixerDumpState copy = mFastMixerDumpState; 3485 copy.dump(fd); 3486 3487#ifdef STATE_QUEUE_DUMP 3488 // Similar for state queue 3489 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3490 observerCopy.dump(fd); 3491 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3492 mutatorCopy.dump(fd); 3493#endif 3494 3495 // Write the tee output to a .wav file 3496 NBAIO_Source *teeSource = mTeeSource.get(); 3497 if (teeSource != NULL) { 3498 char teePath[64]; 3499 struct timeval tv; 3500 gettimeofday(&tv, NULL); 3501 struct tm tm; 3502 localtime_r(&tv.tv_sec, &tm); 3503 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3504 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3505 if (teeFd >= 0) { 3506 char wavHeader[44]; 3507 memcpy(wavHeader, 3508 "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", 3509 sizeof(wavHeader)); 3510 NBAIO_Format format = teeSource->format(); 3511 unsigned channelCount = Format_channelCount(format); 3512 ALOG_ASSERT(channelCount <= FCC_2); 3513 unsigned sampleRate = Format_sampleRate(format); 3514 wavHeader[22] = channelCount; // number of channels 3515 wavHeader[24] = sampleRate; // sample rate 3516 wavHeader[25] = sampleRate >> 8; 3517 wavHeader[32] = channelCount * 2; // block alignment 3518 write(teeFd, wavHeader, sizeof(wavHeader)); 3519 size_t total = 0; 3520 bool firstRead = true; 3521 for (;;) { 3522#define TEE_SINK_READ 1024 3523 short buffer[TEE_SINK_READ * FCC_2]; 3524 size_t count = TEE_SINK_READ; 3525 ssize_t actual = teeSource->read(buffer, count); 3526 bool wasFirstRead = firstRead; 3527 firstRead = false; 3528 if (actual <= 0) { 3529 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3530 continue; 3531 } 3532 break; 3533 } 3534 ALOG_ASSERT(actual <= count); 3535 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3536 total += actual; 3537 } 3538 lseek(teeFd, (off_t) 4, SEEK_SET); 3539 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3540 write(teeFd, &temp, sizeof(temp)); 3541 lseek(teeFd, (off_t) 40, SEEK_SET); 3542 temp = total * channelCount * sizeof(short); 3543 write(teeFd, &temp, sizeof(temp)); 3544 close(teeFd); 3545 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3546 } else { 3547 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3548 } 3549 } 3550 3551 return NO_ERROR; 3552} 3553 3554uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3555{ 3556 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3557} 3558 3559uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3560{ 3561 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3562} 3563 3564void AudioFlinger::MixerThread::cacheParameters_l() 3565{ 3566 PlaybackThread::cacheParameters_l(); 3567 3568 // FIXME: Relaxed timing because of a certain device that can't meet latency 3569 // Should be reduced to 2x after the vendor fixes the driver issue 3570 // increase threshold again due to low power audio mode. The way this warning 3571 // threshold is calculated and its usefulness should be reconsidered anyway. 3572 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3573} 3574 3575// ---------------------------------------------------------------------------- 3576AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3577 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3578 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3579 // mLeftVolFloat, mRightVolFloat 3580 // mLeftVolShort, mRightVolShort 3581{ 3582} 3583 3584AudioFlinger::DirectOutputThread::~DirectOutputThread() 3585{ 3586} 3587 3588AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3589 Vector< sp<Track> > *tracksToRemove 3590) 3591{ 3592 sp<Track> trackToRemove; 3593 3594 mixer_state mixerStatus = MIXER_IDLE; 3595 3596 // find out which tracks need to be processed 3597 if (mActiveTracks.size() != 0) { 3598 sp<Track> t = mActiveTracks[0].promote(); 3599 // The track died recently 3600 if (t == 0) return MIXER_IDLE; 3601 3602 Track* const track = t.get(); 3603 audio_track_cblk_t* cblk = track->cblk(); 3604 3605 // The first time a track is added we wait 3606 // for all its buffers to be filled before processing it 3607 if (cblk->framesReady() && track->isReady() && 3608 !track->isPaused() && !track->isTerminated()) 3609 { 3610 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3611 3612 if (track->mFillingUpStatus == Track::FS_FILLED) { 3613 track->mFillingUpStatus = Track::FS_ACTIVE; 3614 mLeftVolFloat = mRightVolFloat = 0; 3615 mLeftVolShort = mRightVolShort = 0; 3616 if (track->mState == TrackBase::RESUMING) { 3617 track->mState = TrackBase::ACTIVE; 3618 rampVolume = true; 3619 } 3620 } else if (cblk->server != 0) { 3621 // If the track is stopped before the first frame was mixed, 3622 // do not apply ramp 3623 rampVolume = true; 3624 } 3625 // compute volume for this track 3626 float left, right; 3627 if (track->isMuted() || mMasterMute || track->isPausing() || 3628 mStreamTypes[track->streamType()].mute) { 3629 left = right = 0; 3630 if (track->isPausing()) { 3631 track->setPaused(); 3632 } 3633 } else { 3634 float typeVolume = mStreamTypes[track->streamType()].volume; 3635 float v = mMasterVolume * typeVolume; 3636 uint32_t vlr = cblk->getVolumeLR(); 3637 float v_clamped = v * (vlr & 0xFFFF); 3638 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3639 left = v_clamped/MAX_GAIN; 3640 v_clamped = v * (vlr >> 16); 3641 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3642 right = v_clamped/MAX_GAIN; 3643 } 3644 3645 if (left != mLeftVolFloat || right != mRightVolFloat) { 3646 mLeftVolFloat = left; 3647 mRightVolFloat = right; 3648 3649 // If audio HAL implements volume control, 3650 // force software volume to nominal value 3651 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3652 left = 1.0f; 3653 right = 1.0f; 3654 } 3655 3656 // Convert volumes from float to 8.24 3657 uint32_t vl = (uint32_t)(left * (1 << 24)); 3658 uint32_t vr = (uint32_t)(right * (1 << 24)); 3659 3660 // Delegate volume control to effect in track effect chain if needed 3661 // only one effect chain can be present on DirectOutputThread, so if 3662 // there is one, the track is connected to it 3663 if (!mEffectChains.isEmpty()) { 3664 // Do not ramp volume if volume is controlled by effect 3665 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3666 rampVolume = false; 3667 } 3668 } 3669 3670 // Convert volumes from 8.24 to 4.12 format 3671 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3672 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3673 leftVol = (uint16_t)v_clamped; 3674 v_clamped = (vr + (1 << 11)) >> 12; 3675 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3676 rightVol = (uint16_t)v_clamped; 3677 } else { 3678 leftVol = mLeftVolShort; 3679 rightVol = mRightVolShort; 3680 rampVolume = false; 3681 } 3682 3683 // reset retry count 3684 track->mRetryCount = kMaxTrackRetriesDirect; 3685 mActiveTrack = t; 3686 mixerStatus = MIXER_TRACKS_READY; 3687 } else { 3688 // clear effect chain input buffer if an active track underruns to avoid sending 3689 // previous audio buffer again to effects 3690 if (!mEffectChains.isEmpty()) { 3691 mEffectChains[0]->clearInputBuffer(); 3692 } 3693 3694 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3695 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3696 // We have consumed all the buffers of this track. 3697 // Remove it from the list of active tracks. 3698 // TODO: implement behavior for compressed audio 3699 size_t audioHALFrames = 3700 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3701 size_t framesWritten = 3702 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3703 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3704 if (track->isStopped()) { 3705 track->reset(); 3706 } 3707 trackToRemove = track; 3708 } 3709 } else { 3710 // No buffers for this track. Give it a few chances to 3711 // fill a buffer, then remove it from active list. 3712 if (--(track->mRetryCount) <= 0) { 3713 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3714 trackToRemove = track; 3715 } else { 3716 mixerStatus = MIXER_TRACKS_ENABLED; 3717 } 3718 } 3719 } 3720 } 3721 3722 // FIXME merge this with similar code for removing multiple tracks 3723 // remove all the tracks that need to be... 3724 if (CC_UNLIKELY(trackToRemove != 0)) { 3725 tracksToRemove->add(trackToRemove); 3726 mActiveTracks.remove(trackToRemove); 3727 if (!mEffectChains.isEmpty()) { 3728 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3729 trackToRemove->sessionId()); 3730 mEffectChains[0]->decActiveTrackCnt(); 3731 } 3732 if (trackToRemove->isTerminated()) { 3733 removeTrack_l(trackToRemove); 3734 } 3735 } 3736 3737 return mixerStatus; 3738} 3739 3740void AudioFlinger::DirectOutputThread::threadLoop_mix() 3741{ 3742 AudioBufferProvider::Buffer buffer; 3743 size_t frameCount = mFrameCount; 3744 int8_t *curBuf = (int8_t *)mMixBuffer; 3745 // output audio to hardware 3746 while (frameCount) { 3747 buffer.frameCount = frameCount; 3748 mActiveTrack->getNextBuffer(&buffer); 3749 if (CC_UNLIKELY(buffer.raw == NULL)) { 3750 memset(curBuf, 0, frameCount * mFrameSize); 3751 break; 3752 } 3753 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3754 frameCount -= buffer.frameCount; 3755 curBuf += buffer.frameCount * mFrameSize; 3756 mActiveTrack->releaseBuffer(&buffer); 3757 } 3758 sleepTime = 0; 3759 standbyTime = systemTime() + standbyDelay; 3760 mActiveTrack.clear(); 3761 3762 // apply volume 3763 3764 // Do not apply volume on compressed audio 3765 if (!audio_is_linear_pcm(mFormat)) { 3766 return; 3767 } 3768 3769 // convert to signed 16 bit before volume calculation 3770 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3771 size_t count = mFrameCount * mChannelCount; 3772 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3773 int16_t *dst = mMixBuffer + count-1; 3774 while (count--) { 3775 *dst-- = (int16_t)(*src--^0x80) << 8; 3776 } 3777 } 3778 3779 frameCount = mFrameCount; 3780 int16_t *out = mMixBuffer; 3781 if (rampVolume) { 3782 if (mChannelCount == 1) { 3783 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3784 int32_t vlInc = d / (int32_t)frameCount; 3785 int32_t vl = ((int32_t)mLeftVolShort << 16); 3786 do { 3787 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3788 out++; 3789 vl += vlInc; 3790 } while (--frameCount); 3791 3792 } else { 3793 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3794 int32_t vlInc = d / (int32_t)frameCount; 3795 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3796 int32_t vrInc = d / (int32_t)frameCount; 3797 int32_t vl = ((int32_t)mLeftVolShort << 16); 3798 int32_t vr = ((int32_t)mRightVolShort << 16); 3799 do { 3800 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3801 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3802 out += 2; 3803 vl += vlInc; 3804 vr += vrInc; 3805 } while (--frameCount); 3806 } 3807 } else { 3808 if (mChannelCount == 1) { 3809 do { 3810 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3811 out++; 3812 } while (--frameCount); 3813 } else { 3814 do { 3815 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3816 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3817 out += 2; 3818 } while (--frameCount); 3819 } 3820 } 3821 3822 // convert back to unsigned 8 bit after volume calculation 3823 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3824 size_t count = mFrameCount * mChannelCount; 3825 int16_t *src = mMixBuffer; 3826 uint8_t *dst = (uint8_t *)mMixBuffer; 3827 while (count--) { 3828 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3829 } 3830 } 3831 3832 mLeftVolShort = leftVol; 3833 mRightVolShort = rightVol; 3834} 3835 3836void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3837{ 3838 if (sleepTime == 0) { 3839 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3840 sleepTime = activeSleepTime; 3841 } else { 3842 sleepTime = idleSleepTime; 3843 } 3844 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3845 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3846 sleepTime = 0; 3847 } 3848} 3849 3850// getTrackName_l() must be called with ThreadBase::mLock held 3851int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3852{ 3853 return 0; 3854} 3855 3856// deleteTrackName_l() must be called with ThreadBase::mLock held 3857void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3858{ 3859} 3860 3861// checkForNewParameters_l() must be called with ThreadBase::mLock held 3862bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3863{ 3864 bool reconfig = false; 3865 3866 while (!mNewParameters.isEmpty()) { 3867 status_t status = NO_ERROR; 3868 String8 keyValuePair = mNewParameters[0]; 3869 AudioParameter param = AudioParameter(keyValuePair); 3870 int value; 3871 3872 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3873 // do not accept frame count changes if tracks are open as the track buffer 3874 // size depends on frame count and correct behavior would not be garantied 3875 // if frame count is changed after track creation 3876 if (!mTracks.isEmpty()) { 3877 status = INVALID_OPERATION; 3878 } else { 3879 reconfig = true; 3880 } 3881 } 3882 if (status == NO_ERROR) { 3883 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3884 keyValuePair.string()); 3885 if (!mStandby && status == INVALID_OPERATION) { 3886 mOutput->stream->common.standby(&mOutput->stream->common); 3887 mStandby = true; 3888 mBytesWritten = 0; 3889 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3890 keyValuePair.string()); 3891 } 3892 if (status == NO_ERROR && reconfig) { 3893 readOutputParameters(); 3894 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3895 } 3896 } 3897 3898 mNewParameters.removeAt(0); 3899 3900 mParamStatus = status; 3901 mParamCond.signal(); 3902 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3903 // already timed out waiting for the status and will never signal the condition. 3904 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3905 } 3906 return reconfig; 3907} 3908 3909uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3910{ 3911 uint32_t time; 3912 if (audio_is_linear_pcm(mFormat)) { 3913 time = PlaybackThread::activeSleepTimeUs(); 3914 } else { 3915 time = 10000; 3916 } 3917 return time; 3918} 3919 3920uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3921{ 3922 uint32_t time; 3923 if (audio_is_linear_pcm(mFormat)) { 3924 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3925 } else { 3926 time = 10000; 3927 } 3928 return time; 3929} 3930 3931uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3932{ 3933 uint32_t time; 3934 if (audio_is_linear_pcm(mFormat)) { 3935 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3936 } else { 3937 time = 10000; 3938 } 3939 return time; 3940} 3941 3942void AudioFlinger::DirectOutputThread::cacheParameters_l() 3943{ 3944 PlaybackThread::cacheParameters_l(); 3945 3946 // use shorter standby delay as on normal output to release 3947 // hardware resources as soon as possible 3948 standbyDelay = microseconds(activeSleepTime*2); 3949} 3950 3951// ---------------------------------------------------------------------------- 3952 3953AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3954 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3955 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3956 mWaitTimeMs(UINT_MAX) 3957{ 3958 addOutputTrack(mainThread); 3959} 3960 3961AudioFlinger::DuplicatingThread::~DuplicatingThread() 3962{ 3963 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3964 mOutputTracks[i]->destroy(); 3965 } 3966} 3967 3968void AudioFlinger::DuplicatingThread::threadLoop_mix() 3969{ 3970 // mix buffers... 3971 if (outputsReady(outputTracks)) { 3972 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3973 } else { 3974 memset(mMixBuffer, 0, mixBufferSize); 3975 } 3976 sleepTime = 0; 3977 writeFrames = mNormalFrameCount; 3978} 3979 3980void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3981{ 3982 if (sleepTime == 0) { 3983 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3984 sleepTime = activeSleepTime; 3985 } else { 3986 sleepTime = idleSleepTime; 3987 } 3988 } else if (mBytesWritten != 0) { 3989 // flush remaining overflow buffers in output tracks 3990 for (size_t i = 0; i < outputTracks.size(); i++) { 3991 if (outputTracks[i]->isActive()) { 3992 sleepTime = 0; 3993 writeFrames = 0; 3994 memset(mMixBuffer, 0, mixBufferSize); 3995 break; 3996 } 3997 } 3998 } 3999} 4000 4001void AudioFlinger::DuplicatingThread::threadLoop_write() 4002{ 4003 standbyTime = systemTime() + standbyDelay; 4004 for (size_t i = 0; i < outputTracks.size(); i++) { 4005 outputTracks[i]->write(mMixBuffer, writeFrames); 4006 } 4007 mBytesWritten += mixBufferSize; 4008} 4009 4010void AudioFlinger::DuplicatingThread::threadLoop_standby() 4011{ 4012 // DuplicatingThread implements standby by stopping all tracks 4013 for (size_t i = 0; i < outputTracks.size(); i++) { 4014 outputTracks[i]->stop(); 4015 } 4016} 4017 4018void AudioFlinger::DuplicatingThread::saveOutputTracks() 4019{ 4020 outputTracks = mOutputTracks; 4021} 4022 4023void AudioFlinger::DuplicatingThread::clearOutputTracks() 4024{ 4025 outputTracks.clear(); 4026} 4027 4028void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4029{ 4030 Mutex::Autolock _l(mLock); 4031 // FIXME explain this formula 4032 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4033 OutputTrack *outputTrack = new OutputTrack(thread, 4034 this, 4035 mSampleRate, 4036 mFormat, 4037 mChannelMask, 4038 frameCount); 4039 if (outputTrack->cblk() != NULL) { 4040 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4041 mOutputTracks.add(outputTrack); 4042 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4043 updateWaitTime_l(); 4044 } 4045} 4046 4047void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4048{ 4049 Mutex::Autolock _l(mLock); 4050 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4051 if (mOutputTracks[i]->thread() == thread) { 4052 mOutputTracks[i]->destroy(); 4053 mOutputTracks.removeAt(i); 4054 updateWaitTime_l(); 4055 return; 4056 } 4057 } 4058 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4059} 4060 4061// caller must hold mLock 4062void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4063{ 4064 mWaitTimeMs = UINT_MAX; 4065 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4066 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4067 if (strong != 0) { 4068 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4069 if (waitTimeMs < mWaitTimeMs) { 4070 mWaitTimeMs = waitTimeMs; 4071 } 4072 } 4073 } 4074} 4075 4076 4077bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4078{ 4079 for (size_t i = 0; i < outputTracks.size(); i++) { 4080 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4081 if (thread == 0) { 4082 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4083 return false; 4084 } 4085 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4086 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4087 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4088 return false; 4089 } 4090 } 4091 return true; 4092} 4093 4094uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4095{ 4096 return (mWaitTimeMs * 1000) / 2; 4097} 4098 4099void AudioFlinger::DuplicatingThread::cacheParameters_l() 4100{ 4101 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4102 updateWaitTime_l(); 4103 4104 MixerThread::cacheParameters_l(); 4105} 4106 4107// ---------------------------------------------------------------------------- 4108 4109// TrackBase constructor must be called with AudioFlinger::mLock held 4110AudioFlinger::ThreadBase::TrackBase::TrackBase( 4111 ThreadBase *thread, 4112 const sp<Client>& client, 4113 uint32_t sampleRate, 4114 audio_format_t format, 4115 uint32_t channelMask, 4116 int frameCount, 4117 const sp<IMemory>& sharedBuffer, 4118 int sessionId) 4119 : RefBase(), 4120 mThread(thread), 4121 mClient(client), 4122 mCblk(NULL), 4123 // mBuffer 4124 // mBufferEnd 4125 mFrameCount(0), 4126 mState(IDLE), 4127 mSampleRate(sampleRate), 4128 mFormat(format), 4129 mStepServerFailed(false), 4130 mSessionId(sessionId) 4131 // mChannelCount 4132 // mChannelMask 4133{ 4134 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4135 4136 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4137 size_t size = sizeof(audio_track_cblk_t); 4138 uint8_t channelCount = popcount(channelMask); 4139 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4140 if (sharedBuffer == 0) { 4141 size += bufferSize; 4142 } 4143 4144 if (client != NULL) { 4145 mCblkMemory = client->heap()->allocate(size); 4146 if (mCblkMemory != 0) { 4147 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4148 if (mCblk != NULL) { // construct the shared structure in-place. 4149 new(mCblk) audio_track_cblk_t(); 4150 // clear all buffers 4151 mCblk->frameCount = frameCount; 4152 mCblk->sampleRate = sampleRate; 4153// uncomment the following lines to quickly test 32-bit wraparound 4154// mCblk->user = 0xffff0000; 4155// mCblk->server = 0xffff0000; 4156// mCblk->userBase = 0xffff0000; 4157// mCblk->serverBase = 0xffff0000; 4158 mChannelCount = channelCount; 4159 mChannelMask = channelMask; 4160 if (sharedBuffer == 0) { 4161 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4162 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4163 // Force underrun condition to avoid false underrun callback until first data is 4164 // written to buffer (other flags are cleared) 4165 mCblk->flags = CBLK_UNDERRUN_ON; 4166 } else { 4167 mBuffer = sharedBuffer->pointer(); 4168 } 4169 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4170 } 4171 } else { 4172 ALOGE("not enough memory for AudioTrack size=%u", size); 4173 client->heap()->dump("AudioTrack"); 4174 return; 4175 } 4176 } else { 4177 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4178 // construct the shared structure in-place. 4179 new(mCblk) audio_track_cblk_t(); 4180 // clear all buffers 4181 mCblk->frameCount = frameCount; 4182 mCblk->sampleRate = sampleRate; 4183// uncomment the following lines to quickly test 32-bit wraparound 4184// mCblk->user = 0xffff0000; 4185// mCblk->server = 0xffff0000; 4186// mCblk->userBase = 0xffff0000; 4187// mCblk->serverBase = 0xffff0000; 4188 mChannelCount = channelCount; 4189 mChannelMask = channelMask; 4190 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4191 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4192 // Force underrun condition to avoid false underrun callback until first data is 4193 // written to buffer (other flags are cleared) 4194 mCblk->flags = CBLK_UNDERRUN_ON; 4195 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4196 } 4197} 4198 4199AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4200{ 4201 if (mCblk != NULL) { 4202 if (mClient == 0) { 4203 delete mCblk; 4204 } else { 4205 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4206 } 4207 } 4208 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4209 if (mClient != 0) { 4210 // Client destructor must run with AudioFlinger mutex locked 4211 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4212 // If the client's reference count drops to zero, the associated destructor 4213 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4214 // relying on the automatic clear() at end of scope. 4215 mClient.clear(); 4216 } 4217} 4218 4219// AudioBufferProvider interface 4220// getNextBuffer() = 0; 4221// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4222void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4223{ 4224 buffer->raw = NULL; 4225 mFrameCount = buffer->frameCount; 4226 // FIXME See note at getNextBuffer() 4227 (void) step(); // ignore return value of step() 4228 buffer->frameCount = 0; 4229} 4230 4231bool AudioFlinger::ThreadBase::TrackBase::step() { 4232 bool result; 4233 audio_track_cblk_t* cblk = this->cblk(); 4234 4235 result = cblk->stepServer(mFrameCount); 4236 if (!result) { 4237 ALOGV("stepServer failed acquiring cblk mutex"); 4238 mStepServerFailed = true; 4239 } 4240 return result; 4241} 4242 4243void AudioFlinger::ThreadBase::TrackBase::reset() { 4244 audio_track_cblk_t* cblk = this->cblk(); 4245 4246 cblk->user = 0; 4247 cblk->server = 0; 4248 cblk->userBase = 0; 4249 cblk->serverBase = 0; 4250 mStepServerFailed = false; 4251 ALOGV("TrackBase::reset"); 4252} 4253 4254int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4255 return (int)mCblk->sampleRate; 4256} 4257 4258void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4259 audio_track_cblk_t* cblk = this->cblk(); 4260 size_t frameSize = cblk->frameSize; 4261 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4262 int8_t *bufferEnd = bufferStart + frames * frameSize; 4263 4264 // Check validity of returned pointer in case the track control block would have been corrupted. 4265 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4266 "TrackBase::getBuffer buffer out of range:\n" 4267 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4268 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4269 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4270 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4271 4272 return bufferStart; 4273} 4274 4275status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4276{ 4277 mSyncEvents.add(event); 4278 return NO_ERROR; 4279} 4280 4281// ---------------------------------------------------------------------------- 4282 4283// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4284AudioFlinger::PlaybackThread::Track::Track( 4285 PlaybackThread *thread, 4286 const sp<Client>& client, 4287 audio_stream_type_t streamType, 4288 uint32_t sampleRate, 4289 audio_format_t format, 4290 uint32_t channelMask, 4291 int frameCount, 4292 const sp<IMemory>& sharedBuffer, 4293 int sessionId, 4294 IAudioFlinger::track_flags_t flags) 4295 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4296 mMute(false), 4297 mFillingUpStatus(FS_INVALID), 4298 // mRetryCount initialized later when needed 4299 mSharedBuffer(sharedBuffer), 4300 mStreamType(streamType), 4301 mName(-1), // see note below 4302 mMainBuffer(thread->mixBuffer()), 4303 mAuxBuffer(NULL), 4304 mAuxEffectId(0), mHasVolumeController(false), 4305 mPresentationCompleteFrames(0), 4306 mFlags(flags), 4307 mFastIndex(-1), 4308 mUnderrunCount(0), 4309 mCachedVolume(1.0) 4310{ 4311 if (mCblk != NULL) { 4312 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4313 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4314 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4315 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4316 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4317 if (mName < 0) { 4318 ALOGE("no more track names available"); 4319 return; 4320 } 4321 // only allocate a fast track index if we were able to allocate a normal track name 4322 if (flags & IAudioFlinger::TRACK_FAST) { 4323 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4324 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4325 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4326 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4327 // FIXME This is too eager. We allocate a fast track index before the 4328 // fast track becomes active. Since fast tracks are a scarce resource, 4329 // this means we are potentially denying other more important fast tracks from 4330 // being created. It would be better to allocate the index dynamically. 4331 mFastIndex = i; 4332 // Read the initial underruns because this field is never cleared by the fast mixer 4333 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4334 thread->mFastTrackAvailMask &= ~(1 << i); 4335 } 4336 } 4337 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4338} 4339 4340AudioFlinger::PlaybackThread::Track::~Track() 4341{ 4342 ALOGV("PlaybackThread::Track destructor"); 4343 sp<ThreadBase> thread = mThread.promote(); 4344 if (thread != 0) { 4345 Mutex::Autolock _l(thread->mLock); 4346 mState = TERMINATED; 4347 } 4348} 4349 4350void AudioFlinger::PlaybackThread::Track::destroy() 4351{ 4352 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4353 // by removing it from mTracks vector, so there is a risk that this Tracks's 4354 // destructor is called. As the destructor needs to lock mLock, 4355 // we must acquire a strong reference on this Track before locking mLock 4356 // here so that the destructor is called only when exiting this function. 4357 // On the other hand, as long as Track::destroy() is only called by 4358 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4359 // this Track with its member mTrack. 4360 sp<Track> keep(this); 4361 { // scope for mLock 4362 sp<ThreadBase> thread = mThread.promote(); 4363 if (thread != 0) { 4364 if (!isOutputTrack()) { 4365 if (mState == ACTIVE || mState == RESUMING) { 4366 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4367 4368#ifdef ADD_BATTERY_DATA 4369 // to track the speaker usage 4370 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4371#endif 4372 } 4373 AudioSystem::releaseOutput(thread->id()); 4374 } 4375 Mutex::Autolock _l(thread->mLock); 4376 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4377 playbackThread->destroyTrack_l(this); 4378 } 4379 } 4380} 4381 4382/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4383{ 4384 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4385 " Server User Main buf Aux Buf Flags Underruns\n"); 4386} 4387 4388void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4389{ 4390 uint32_t vlr = mCblk->getVolumeLR(); 4391 if (isFastTrack()) { 4392 sprintf(buffer, " F %2d", mFastIndex); 4393 } else { 4394 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4395 } 4396 track_state state = mState; 4397 char stateChar; 4398 switch (state) { 4399 case IDLE: 4400 stateChar = 'I'; 4401 break; 4402 case TERMINATED: 4403 stateChar = 'T'; 4404 break; 4405 case STOPPING_1: 4406 stateChar = 's'; 4407 break; 4408 case STOPPING_2: 4409 stateChar = '5'; 4410 break; 4411 case STOPPED: 4412 stateChar = 'S'; 4413 break; 4414 case RESUMING: 4415 stateChar = 'R'; 4416 break; 4417 case ACTIVE: 4418 stateChar = 'A'; 4419 break; 4420 case PAUSING: 4421 stateChar = 'p'; 4422 break; 4423 case PAUSED: 4424 stateChar = 'P'; 4425 break; 4426 case FLUSHED: 4427 stateChar = 'F'; 4428 break; 4429 default: 4430 stateChar = '?'; 4431 break; 4432 } 4433 char nowInUnderrun; 4434 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4435 case UNDERRUN_FULL: 4436 nowInUnderrun = ' '; 4437 break; 4438 case UNDERRUN_PARTIAL: 4439 nowInUnderrun = '<'; 4440 break; 4441 case UNDERRUN_EMPTY: 4442 nowInUnderrun = '*'; 4443 break; 4444 default: 4445 nowInUnderrun = '?'; 4446 break; 4447 } 4448 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4449 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4450 (mClient == 0) ? getpid_cached : mClient->pid(), 4451 mStreamType, 4452 mFormat, 4453 mChannelMask, 4454 mSessionId, 4455 mFrameCount, 4456 mCblk->frameCount, 4457 stateChar, 4458 mMute, 4459 mFillingUpStatus, 4460 mCblk->sampleRate, 4461 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4462 20.0 * log10((vlr >> 16) / 4096.0), 4463 mCblk->server, 4464 mCblk->user, 4465 (int)mMainBuffer, 4466 (int)mAuxBuffer, 4467 mCblk->flags, 4468 mUnderrunCount, 4469 nowInUnderrun); 4470} 4471 4472// AudioBufferProvider interface 4473status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4474 AudioBufferProvider::Buffer* buffer, int64_t pts) 4475{ 4476 audio_track_cblk_t* cblk = this->cblk(); 4477 uint32_t framesReady; 4478 uint32_t framesReq = buffer->frameCount; 4479 4480 // Check if last stepServer failed, try to step now 4481 if (mStepServerFailed) { 4482 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4483 // Since the fast mixer is higher priority than client callback thread, 4484 // it does not result in priority inversion for client. 4485 // But a non-blocking solution would be preferable to avoid 4486 // fast mixer being unable to tryLock(), and 4487 // to avoid the extra context switches if the client wakes up, 4488 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4489 if (!step()) goto getNextBuffer_exit; 4490 ALOGV("stepServer recovered"); 4491 mStepServerFailed = false; 4492 } 4493 4494 // FIXME Same as above 4495 framesReady = cblk->framesReady(); 4496 4497 if (CC_LIKELY(framesReady)) { 4498 uint32_t s = cblk->server; 4499 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4500 4501 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4502 if (framesReq > framesReady) { 4503 framesReq = framesReady; 4504 } 4505 if (framesReq > bufferEnd - s) { 4506 framesReq = bufferEnd - s; 4507 } 4508 4509 buffer->raw = getBuffer(s, framesReq); 4510 if (buffer->raw == NULL) goto getNextBuffer_exit; 4511 4512 buffer->frameCount = framesReq; 4513 return NO_ERROR; 4514 } 4515 4516getNextBuffer_exit: 4517 buffer->raw = NULL; 4518 buffer->frameCount = 0; 4519 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4520 return NOT_ENOUGH_DATA; 4521} 4522 4523// Note that framesReady() takes a mutex on the control block using tryLock(). 4524// This could result in priority inversion if framesReady() is called by the normal mixer, 4525// as the normal mixer thread runs at lower 4526// priority than the client's callback thread: there is a short window within framesReady() 4527// during which the normal mixer could be preempted, and the client callback would block. 4528// Another problem can occur if framesReady() is called by the fast mixer: 4529// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4530// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4531size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4532 return mCblk->framesReady(); 4533} 4534 4535// Don't call for fast tracks; the framesReady() could result in priority inversion 4536bool AudioFlinger::PlaybackThread::Track::isReady() const { 4537 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4538 4539 if (framesReady() >= mCblk->frameCount || 4540 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4541 mFillingUpStatus = FS_FILLED; 4542 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4543 return true; 4544 } 4545 return false; 4546} 4547 4548status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4549 int triggerSession) 4550{ 4551 status_t status = NO_ERROR; 4552 ALOGV("start(%d), calling pid %d session %d", 4553 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4554 4555 sp<ThreadBase> thread = mThread.promote(); 4556 if (thread != 0) { 4557 Mutex::Autolock _l(thread->mLock); 4558 track_state state = mState; 4559 // here the track could be either new, or restarted 4560 // in both cases "unstop" the track 4561 if (mState == PAUSED) { 4562 mState = TrackBase::RESUMING; 4563 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4564 } else { 4565 mState = TrackBase::ACTIVE; 4566 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4567 } 4568 4569 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4570 thread->mLock.unlock(); 4571 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4572 thread->mLock.lock(); 4573 4574#ifdef ADD_BATTERY_DATA 4575 // to track the speaker usage 4576 if (status == NO_ERROR) { 4577 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4578 } 4579#endif 4580 } 4581 if (status == NO_ERROR) { 4582 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4583 playbackThread->addTrack_l(this); 4584 } else { 4585 mState = state; 4586 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4587 } 4588 } else { 4589 status = BAD_VALUE; 4590 } 4591 return status; 4592} 4593 4594void AudioFlinger::PlaybackThread::Track::stop() 4595{ 4596 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4597 sp<ThreadBase> thread = mThread.promote(); 4598 if (thread != 0) { 4599 Mutex::Autolock _l(thread->mLock); 4600 track_state state = mState; 4601 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4602 // If the track is not active (PAUSED and buffers full), flush buffers 4603 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4604 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4605 reset(); 4606 mState = STOPPED; 4607 } else if (!isFastTrack()) { 4608 mState = STOPPED; 4609 } else { 4610 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4611 // and then to STOPPED and reset() when presentation is complete 4612 mState = STOPPING_1; 4613 } 4614 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4615 } 4616 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4617 thread->mLock.unlock(); 4618 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4619 thread->mLock.lock(); 4620 4621#ifdef ADD_BATTERY_DATA 4622 // to track the speaker usage 4623 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4624#endif 4625 } 4626 } 4627} 4628 4629void AudioFlinger::PlaybackThread::Track::pause() 4630{ 4631 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4632 sp<ThreadBase> thread = mThread.promote(); 4633 if (thread != 0) { 4634 Mutex::Autolock _l(thread->mLock); 4635 if (mState == ACTIVE || mState == RESUMING) { 4636 mState = PAUSING; 4637 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4638 if (!isOutputTrack()) { 4639 thread->mLock.unlock(); 4640 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4641 thread->mLock.lock(); 4642 4643#ifdef ADD_BATTERY_DATA 4644 // to track the speaker usage 4645 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4646#endif 4647 } 4648 } 4649 } 4650} 4651 4652void AudioFlinger::PlaybackThread::Track::flush() 4653{ 4654 ALOGV("flush(%d)", mName); 4655 sp<ThreadBase> thread = mThread.promote(); 4656 if (thread != 0) { 4657 Mutex::Autolock _l(thread->mLock); 4658 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4659 mState != PAUSING) { 4660 return; 4661 } 4662 // No point remaining in PAUSED state after a flush => go to 4663 // FLUSHED state 4664 mState = FLUSHED; 4665 // do not reset the track if it is still in the process of being stopped or paused. 4666 // this will be done by prepareTracks_l() when the track is stopped. 4667 // prepareTracks_l() will see mState == FLUSHED, then 4668 // remove from active track list, reset(), and trigger presentation complete 4669 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4670 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4671 reset(); 4672 } 4673 } 4674} 4675 4676void AudioFlinger::PlaybackThread::Track::reset() 4677{ 4678 // Do not reset twice to avoid discarding data written just after a flush and before 4679 // the audioflinger thread detects the track is stopped. 4680 if (!mResetDone) { 4681 TrackBase::reset(); 4682 // Force underrun condition to avoid false underrun callback until first data is 4683 // written to buffer 4684 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4685 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4686 mFillingUpStatus = FS_FILLING; 4687 mResetDone = true; 4688 if (mState == FLUSHED) { 4689 mState = IDLE; 4690 } 4691 } 4692} 4693 4694void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4695{ 4696 mMute = muted; 4697} 4698 4699status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4700{ 4701 status_t status = DEAD_OBJECT; 4702 sp<ThreadBase> thread = mThread.promote(); 4703 if (thread != 0) { 4704 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4705 status = playbackThread->attachAuxEffect(this, EffectId); 4706 } 4707 return status; 4708} 4709 4710void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4711{ 4712 mAuxEffectId = EffectId; 4713 mAuxBuffer = buffer; 4714} 4715 4716bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4717 size_t audioHalFrames) 4718{ 4719 // a track is considered presented when the total number of frames written to audio HAL 4720 // corresponds to the number of frames written when presentationComplete() is called for the 4721 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4722 if (mPresentationCompleteFrames == 0) { 4723 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4724 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4725 mPresentationCompleteFrames, audioHalFrames); 4726 } 4727 if (framesWritten >= mPresentationCompleteFrames) { 4728 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4729 mSessionId, framesWritten); 4730 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4731 return true; 4732 } 4733 return false; 4734} 4735 4736void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4737{ 4738 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4739 if (mSyncEvents[i]->type() == type) { 4740 mSyncEvents[i]->trigger(); 4741 mSyncEvents.removeAt(i); 4742 i--; 4743 } 4744 } 4745} 4746 4747// implement VolumeBufferProvider interface 4748 4749uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4750{ 4751 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4752 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4753 uint32_t vlr = mCblk->getVolumeLR(); 4754 uint32_t vl = vlr & 0xFFFF; 4755 uint32_t vr = vlr >> 16; 4756 // track volumes come from shared memory, so can't be trusted and must be clamped 4757 if (vl > MAX_GAIN_INT) { 4758 vl = MAX_GAIN_INT; 4759 } 4760 if (vr > MAX_GAIN_INT) { 4761 vr = MAX_GAIN_INT; 4762 } 4763 // now apply the cached master volume and stream type volume; 4764 // this is trusted but lacks any synchronization or barrier so may be stale 4765 float v = mCachedVolume; 4766 vl *= v; 4767 vr *= v; 4768 // re-combine into U4.16 4769 vlr = (vr << 16) | (vl & 0xFFFF); 4770 // FIXME look at mute, pause, and stop flags 4771 return vlr; 4772} 4773 4774status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4775{ 4776 if (mState == TERMINATED || mState == PAUSED || 4777 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4778 (mState == STOPPED)))) { 4779 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4780 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4781 event->cancel(); 4782 return INVALID_OPERATION; 4783 } 4784 TrackBase::setSyncEvent(event); 4785 return NO_ERROR; 4786} 4787 4788// timed audio tracks 4789 4790sp<AudioFlinger::PlaybackThread::TimedTrack> 4791AudioFlinger::PlaybackThread::TimedTrack::create( 4792 PlaybackThread *thread, 4793 const sp<Client>& client, 4794 audio_stream_type_t streamType, 4795 uint32_t sampleRate, 4796 audio_format_t format, 4797 uint32_t channelMask, 4798 int frameCount, 4799 const sp<IMemory>& sharedBuffer, 4800 int sessionId) { 4801 if (!client->reserveTimedTrack()) 4802 return NULL; 4803 4804 return new TimedTrack( 4805 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4806 sharedBuffer, sessionId); 4807} 4808 4809AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4810 PlaybackThread *thread, 4811 const sp<Client>& client, 4812 audio_stream_type_t streamType, 4813 uint32_t sampleRate, 4814 audio_format_t format, 4815 uint32_t channelMask, 4816 int frameCount, 4817 const sp<IMemory>& sharedBuffer, 4818 int sessionId) 4819 : Track(thread, client, streamType, sampleRate, format, channelMask, 4820 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4821 mQueueHeadInFlight(false), 4822 mTrimQueueHeadOnRelease(false), 4823 mFramesPendingInQueue(0), 4824 mTimedSilenceBuffer(NULL), 4825 mTimedSilenceBufferSize(0), 4826 mTimedAudioOutputOnTime(false), 4827 mMediaTimeTransformValid(false) 4828{ 4829 LocalClock lc; 4830 mLocalTimeFreq = lc.getLocalFreq(); 4831 4832 mLocalTimeToSampleTransform.a_zero = 0; 4833 mLocalTimeToSampleTransform.b_zero = 0; 4834 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4835 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4836 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4837 &mLocalTimeToSampleTransform.a_to_b_denom); 4838 4839 mMediaTimeToSampleTransform.a_zero = 0; 4840 mMediaTimeToSampleTransform.b_zero = 0; 4841 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4842 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4843 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4844 &mMediaTimeToSampleTransform.a_to_b_denom); 4845} 4846 4847AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4848 mClient->releaseTimedTrack(); 4849 delete [] mTimedSilenceBuffer; 4850} 4851 4852status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4853 size_t size, sp<IMemory>* buffer) { 4854 4855 Mutex::Autolock _l(mTimedBufferQueueLock); 4856 4857 trimTimedBufferQueue_l(); 4858 4859 // lazily initialize the shared memory heap for timed buffers 4860 if (mTimedMemoryDealer == NULL) { 4861 const int kTimedBufferHeapSize = 512 << 10; 4862 4863 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4864 "AudioFlingerTimed"); 4865 if (mTimedMemoryDealer == NULL) 4866 return NO_MEMORY; 4867 } 4868 4869 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4870 if (newBuffer == NULL) { 4871 newBuffer = mTimedMemoryDealer->allocate(size); 4872 if (newBuffer == NULL) 4873 return NO_MEMORY; 4874 } 4875 4876 *buffer = newBuffer; 4877 return NO_ERROR; 4878} 4879 4880// caller must hold mTimedBufferQueueLock 4881void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4882 int64_t mediaTimeNow; 4883 { 4884 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4885 if (!mMediaTimeTransformValid) 4886 return; 4887 4888 int64_t targetTimeNow; 4889 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4890 ? mCCHelper.getCommonTime(&targetTimeNow) 4891 : mCCHelper.getLocalTime(&targetTimeNow); 4892 4893 if (OK != res) 4894 return; 4895 4896 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4897 &mediaTimeNow)) { 4898 return; 4899 } 4900 } 4901 4902 size_t trimEnd; 4903 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4904 int64_t bufEnd; 4905 4906 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4907 // We have a next buffer. Just use its PTS as the PTS of the frame 4908 // following the last frame in this buffer. If the stream is sparse 4909 // (ie, there are deliberate gaps left in the stream which should be 4910 // filled with silence by the TimedAudioTrack), then this can result 4911 // in one extra buffer being left un-trimmed when it could have 4912 // been. In general, this is not typical, and we would rather 4913 // optimized away the TS calculation below for the more common case 4914 // where PTSes are contiguous. 4915 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4916 } else { 4917 // We have no next buffer. Compute the PTS of the frame following 4918 // the last frame in this buffer by computing the duration of of 4919 // this frame in media time units and adding it to the PTS of the 4920 // buffer. 4921 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4922 / mCblk->frameSize; 4923 4924 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4925 &bufEnd)) { 4926 ALOGE("Failed to convert frame count of %lld to media time" 4927 " duration" " (scale factor %d/%u) in %s", 4928 frameCount, 4929 mMediaTimeToSampleTransform.a_to_b_numer, 4930 mMediaTimeToSampleTransform.a_to_b_denom, 4931 __PRETTY_FUNCTION__); 4932 break; 4933 } 4934 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4935 } 4936 4937 if (bufEnd > mediaTimeNow) 4938 break; 4939 4940 // Is the buffer we want to use in the middle of a mix operation right 4941 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4942 // from the mixer which should be coming back shortly. 4943 if (!trimEnd && mQueueHeadInFlight) { 4944 mTrimQueueHeadOnRelease = true; 4945 } 4946 } 4947 4948 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4949 if (trimStart < trimEnd) { 4950 // Update the bookkeeping for framesReady() 4951 for (size_t i = trimStart; i < trimEnd; ++i) { 4952 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4953 } 4954 4955 // Now actually remove the buffers from the queue. 4956 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4957 } 4958} 4959 4960void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4961 const char* logTag) { 4962 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4963 "%s called (reason \"%s\"), but timed buffer queue has no" 4964 " elements to trim.", __FUNCTION__, logTag); 4965 4966 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4967 mTimedBufferQueue.removeAt(0); 4968} 4969 4970void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4971 const TimedBuffer& buf, 4972 const char* logTag) { 4973 uint32_t bufBytes = buf.buffer()->size(); 4974 uint32_t consumedAlready = buf.position(); 4975 4976 ALOG_ASSERT(consumedAlready <= bufBytes, 4977 "Bad bookkeeping while updating frames pending. Timed buffer is" 4978 " only %u bytes long, but claims to have consumed %u" 4979 " bytes. (update reason: \"%s\")", 4980 bufBytes, consumedAlready, logTag); 4981 4982 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4983 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4984 "Bad bookkeeping while updating frames pending. Should have at" 4985 " least %u queued frames, but we think we have only %u. (update" 4986 " reason: \"%s\")", 4987 bufFrames, mFramesPendingInQueue, logTag); 4988 4989 mFramesPendingInQueue -= bufFrames; 4990} 4991 4992status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4993 const sp<IMemory>& buffer, int64_t pts) { 4994 4995 { 4996 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4997 if (!mMediaTimeTransformValid) 4998 return INVALID_OPERATION; 4999 } 5000 5001 Mutex::Autolock _l(mTimedBufferQueueLock); 5002 5003 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5004 mFramesPendingInQueue += bufFrames; 5005 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5006 5007 return NO_ERROR; 5008} 5009 5010status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5011 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5012 5013 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5014 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5015 target); 5016 5017 if (!(target == TimedAudioTrack::LOCAL_TIME || 5018 target == TimedAudioTrack::COMMON_TIME)) { 5019 return BAD_VALUE; 5020 } 5021 5022 Mutex::Autolock lock(mMediaTimeTransformLock); 5023 mMediaTimeTransform = xform; 5024 mMediaTimeTransformTarget = target; 5025 mMediaTimeTransformValid = true; 5026 5027 return NO_ERROR; 5028} 5029 5030#define min(a, b) ((a) < (b) ? (a) : (b)) 5031 5032// implementation of getNextBuffer for tracks whose buffers have timestamps 5033status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5034 AudioBufferProvider::Buffer* buffer, int64_t pts) 5035{ 5036 if (pts == AudioBufferProvider::kInvalidPTS) { 5037 buffer->raw = 0; 5038 buffer->frameCount = 0; 5039 mTimedAudioOutputOnTime = false; 5040 return INVALID_OPERATION; 5041 } 5042 5043 Mutex::Autolock _l(mTimedBufferQueueLock); 5044 5045 ALOG_ASSERT(!mQueueHeadInFlight, 5046 "getNextBuffer called without releaseBuffer!"); 5047 5048 while (true) { 5049 5050 // if we have no timed buffers, then fail 5051 if (mTimedBufferQueue.isEmpty()) { 5052 buffer->raw = 0; 5053 buffer->frameCount = 0; 5054 return NOT_ENOUGH_DATA; 5055 } 5056 5057 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5058 5059 // calculate the PTS of the head of the timed buffer queue expressed in 5060 // local time 5061 int64_t headLocalPTS; 5062 { 5063 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5064 5065 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5066 5067 if (mMediaTimeTransform.a_to_b_denom == 0) { 5068 // the transform represents a pause, so yield silence 5069 timedYieldSilence_l(buffer->frameCount, buffer); 5070 return NO_ERROR; 5071 } 5072 5073 int64_t transformedPTS; 5074 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5075 &transformedPTS)) { 5076 // the transform failed. this shouldn't happen, but if it does 5077 // then just drop this buffer 5078 ALOGW("timedGetNextBuffer transform failed"); 5079 buffer->raw = 0; 5080 buffer->frameCount = 0; 5081 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5082 return NO_ERROR; 5083 } 5084 5085 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5086 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5087 &headLocalPTS)) { 5088 buffer->raw = 0; 5089 buffer->frameCount = 0; 5090 return INVALID_OPERATION; 5091 } 5092 } else { 5093 headLocalPTS = transformedPTS; 5094 } 5095 } 5096 5097 // adjust the head buffer's PTS to reflect the portion of the head buffer 5098 // that has already been consumed 5099 int64_t effectivePTS = headLocalPTS + 5100 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5101 5102 // Calculate the delta in samples between the head of the input buffer 5103 // queue and the start of the next output buffer that will be written. 5104 // If the transformation fails because of over or underflow, it means 5105 // that the sample's position in the output stream is so far out of 5106 // whack that it should just be dropped. 5107 int64_t sampleDelta; 5108 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5109 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5110 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5111 " mix"); 5112 continue; 5113 } 5114 if (!mLocalTimeToSampleTransform.doForwardTransform( 5115 (effectivePTS - pts) << 32, &sampleDelta)) { 5116 ALOGV("*** too late during sample rate transform: dropped buffer"); 5117 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5118 continue; 5119 } 5120 5121 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5122 " sampleDelta=[%d.%08x]", 5123 head.pts(), head.position(), pts, 5124 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5125 + (sampleDelta >> 32)), 5126 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5127 5128 // if the delta between the ideal placement for the next input sample and 5129 // the current output position is within this threshold, then we will 5130 // concatenate the next input samples to the previous output 5131 const int64_t kSampleContinuityThreshold = 5132 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5133 5134 // if this is the first buffer of audio that we're emitting from this track 5135 // then it should be almost exactly on time. 5136 const int64_t kSampleStartupThreshold = 1LL << 32; 5137 5138 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5139 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5140 // the next input is close enough to being on time, so concatenate it 5141 // with the last output 5142 timedYieldSamples_l(buffer); 5143 5144 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5145 head.position(), buffer->frameCount); 5146 return NO_ERROR; 5147 } 5148 5149 // Looks like our output is not on time. Reset our on timed status. 5150 // Next time we mix samples from our input queue, then should be within 5151 // the StartupThreshold. 5152 mTimedAudioOutputOnTime = false; 5153 if (sampleDelta > 0) { 5154 // the gap between the current output position and the proper start of 5155 // the next input sample is too big, so fill it with silence 5156 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5157 5158 timedYieldSilence_l(framesUntilNextInput, buffer); 5159 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5160 return NO_ERROR; 5161 } else { 5162 // the next input sample is late 5163 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5164 size_t onTimeSamplePosition = 5165 head.position() + lateFrames * mCblk->frameSize; 5166 5167 if (onTimeSamplePosition > head.buffer()->size()) { 5168 // all the remaining samples in the head are too late, so 5169 // drop it and move on 5170 ALOGV("*** too late: dropped buffer"); 5171 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5172 continue; 5173 } else { 5174 // skip over the late samples 5175 head.setPosition(onTimeSamplePosition); 5176 5177 // yield the available samples 5178 timedYieldSamples_l(buffer); 5179 5180 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5181 return NO_ERROR; 5182 } 5183 } 5184 } 5185} 5186 5187// Yield samples from the timed buffer queue head up to the given output 5188// buffer's capacity. 5189// 5190// Caller must hold mTimedBufferQueueLock 5191void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5192 AudioBufferProvider::Buffer* buffer) { 5193 5194 const TimedBuffer& head = mTimedBufferQueue[0]; 5195 5196 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5197 head.position()); 5198 5199 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5200 mCblk->frameSize); 5201 size_t framesRequested = buffer->frameCount; 5202 buffer->frameCount = min(framesLeftInHead, framesRequested); 5203 5204 mQueueHeadInFlight = true; 5205 mTimedAudioOutputOnTime = true; 5206} 5207 5208// Yield samples of silence up to the given output buffer's capacity 5209// 5210// Caller must hold mTimedBufferQueueLock 5211void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5212 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5213 5214 // lazily allocate a buffer filled with silence 5215 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5216 delete [] mTimedSilenceBuffer; 5217 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5218 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5219 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5220 } 5221 5222 buffer->raw = mTimedSilenceBuffer; 5223 size_t framesRequested = buffer->frameCount; 5224 buffer->frameCount = min(numFrames, framesRequested); 5225 5226 mTimedAudioOutputOnTime = false; 5227} 5228 5229// AudioBufferProvider interface 5230void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5231 AudioBufferProvider::Buffer* buffer) { 5232 5233 Mutex::Autolock _l(mTimedBufferQueueLock); 5234 5235 // If the buffer which was just released is part of the buffer at the head 5236 // of the queue, be sure to update the amt of the buffer which has been 5237 // consumed. If the buffer being returned is not part of the head of the 5238 // queue, its either because the buffer is part of the silence buffer, or 5239 // because the head of the timed queue was trimmed after the mixer called 5240 // getNextBuffer but before the mixer called releaseBuffer. 5241 if (buffer->raw == mTimedSilenceBuffer) { 5242 ALOG_ASSERT(!mQueueHeadInFlight, 5243 "Queue head in flight during release of silence buffer!"); 5244 goto done; 5245 } 5246 5247 ALOG_ASSERT(mQueueHeadInFlight, 5248 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5249 " head in flight."); 5250 5251 if (mTimedBufferQueue.size()) { 5252 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5253 5254 void* start = head.buffer()->pointer(); 5255 void* end = reinterpret_cast<void*>( 5256 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5257 + head.buffer()->size()); 5258 5259 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5260 "released buffer not within the head of the timed buffer" 5261 " queue; qHead = [%p, %p], released buffer = %p", 5262 start, end, buffer->raw); 5263 5264 head.setPosition(head.position() + 5265 (buffer->frameCount * mCblk->frameSize)); 5266 mQueueHeadInFlight = false; 5267 5268 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5269 "Bad bookkeeping during releaseBuffer! Should have at" 5270 " least %u queued frames, but we think we have only %u", 5271 buffer->frameCount, mFramesPendingInQueue); 5272 5273 mFramesPendingInQueue -= buffer->frameCount; 5274 5275 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5276 || mTrimQueueHeadOnRelease) { 5277 trimTimedBufferQueueHead_l("releaseBuffer"); 5278 mTrimQueueHeadOnRelease = false; 5279 } 5280 } else { 5281 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5282 " buffers in the timed buffer queue"); 5283 } 5284 5285done: 5286 buffer->raw = 0; 5287 buffer->frameCount = 0; 5288} 5289 5290size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5291 Mutex::Autolock _l(mTimedBufferQueueLock); 5292 return mFramesPendingInQueue; 5293} 5294 5295AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5296 : mPTS(0), mPosition(0) {} 5297 5298AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5299 const sp<IMemory>& buffer, int64_t pts) 5300 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5301 5302// ---------------------------------------------------------------------------- 5303 5304// RecordTrack constructor must be called with AudioFlinger::mLock held 5305AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5306 RecordThread *thread, 5307 const sp<Client>& client, 5308 uint32_t sampleRate, 5309 audio_format_t format, 5310 uint32_t channelMask, 5311 int frameCount, 5312 int sessionId) 5313 : TrackBase(thread, client, sampleRate, format, 5314 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5315 mOverflow(false) 5316{ 5317 if (mCblk != NULL) { 5318 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5319 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5320 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5321 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5322 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5323 } else { 5324 mCblk->frameSize = sizeof(int8_t); 5325 } 5326 } 5327} 5328 5329AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5330{ 5331 sp<ThreadBase> thread = mThread.promote(); 5332 if (thread != 0) { 5333 AudioSystem::releaseInput(thread->id()); 5334 } 5335} 5336 5337// AudioBufferProvider interface 5338status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5339{ 5340 audio_track_cblk_t* cblk = this->cblk(); 5341 uint32_t framesAvail; 5342 uint32_t framesReq = buffer->frameCount; 5343 5344 // Check if last stepServer failed, try to step now 5345 if (mStepServerFailed) { 5346 if (!step()) goto getNextBuffer_exit; 5347 ALOGV("stepServer recovered"); 5348 mStepServerFailed = false; 5349 } 5350 5351 framesAvail = cblk->framesAvailable_l(); 5352 5353 if (CC_LIKELY(framesAvail)) { 5354 uint32_t s = cblk->server; 5355 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5356 5357 if (framesReq > framesAvail) { 5358 framesReq = framesAvail; 5359 } 5360 if (framesReq > bufferEnd - s) { 5361 framesReq = bufferEnd - s; 5362 } 5363 5364 buffer->raw = getBuffer(s, framesReq); 5365 if (buffer->raw == NULL) goto getNextBuffer_exit; 5366 5367 buffer->frameCount = framesReq; 5368 return NO_ERROR; 5369 } 5370 5371getNextBuffer_exit: 5372 buffer->raw = NULL; 5373 buffer->frameCount = 0; 5374 return NOT_ENOUGH_DATA; 5375} 5376 5377status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5378 int triggerSession) 5379{ 5380 sp<ThreadBase> thread = mThread.promote(); 5381 if (thread != 0) { 5382 RecordThread *recordThread = (RecordThread *)thread.get(); 5383 return recordThread->start(this, event, triggerSession); 5384 } else { 5385 return BAD_VALUE; 5386 } 5387} 5388 5389void AudioFlinger::RecordThread::RecordTrack::stop() 5390{ 5391 sp<ThreadBase> thread = mThread.promote(); 5392 if (thread != 0) { 5393 RecordThread *recordThread = (RecordThread *)thread.get(); 5394 recordThread->stop(this); 5395 TrackBase::reset(); 5396 // Force overrun condition to avoid false overrun callback until first data is 5397 // read from buffer 5398 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5399 } 5400} 5401 5402void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5403{ 5404 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5405 (mClient == 0) ? getpid_cached : mClient->pid(), 5406 mFormat, 5407 mChannelMask, 5408 mSessionId, 5409 mFrameCount, 5410 mState, 5411 mCblk->sampleRate, 5412 mCblk->server, 5413 mCblk->user); 5414} 5415 5416 5417// ---------------------------------------------------------------------------- 5418 5419AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5420 PlaybackThread *playbackThread, 5421 DuplicatingThread *sourceThread, 5422 uint32_t sampleRate, 5423 audio_format_t format, 5424 uint32_t channelMask, 5425 int frameCount) 5426 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5427 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5428 mActive(false), mSourceThread(sourceThread) 5429{ 5430 5431 if (mCblk != NULL) { 5432 mCblk->flags |= CBLK_DIRECTION_OUT; 5433 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5434 mOutBuffer.frameCount = 0; 5435 playbackThread->mTracks.add(this); 5436 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5437 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5438 mCblk, mBuffer, mCblk->buffers, 5439 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5440 } else { 5441 ALOGW("Error creating output track on thread %p", playbackThread); 5442 } 5443} 5444 5445AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5446{ 5447 clearBufferQueue(); 5448} 5449 5450status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5451 int triggerSession) 5452{ 5453 status_t status = Track::start(event, triggerSession); 5454 if (status != NO_ERROR) { 5455 return status; 5456 } 5457 5458 mActive = true; 5459 mRetryCount = 127; 5460 return status; 5461} 5462 5463void AudioFlinger::PlaybackThread::OutputTrack::stop() 5464{ 5465 Track::stop(); 5466 clearBufferQueue(); 5467 mOutBuffer.frameCount = 0; 5468 mActive = false; 5469} 5470 5471bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5472{ 5473 Buffer *pInBuffer; 5474 Buffer inBuffer; 5475 uint32_t channelCount = mChannelCount; 5476 bool outputBufferFull = false; 5477 inBuffer.frameCount = frames; 5478 inBuffer.i16 = data; 5479 5480 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5481 5482 if (!mActive && frames != 0) { 5483 start(); 5484 sp<ThreadBase> thread = mThread.promote(); 5485 if (thread != 0) { 5486 MixerThread *mixerThread = (MixerThread *)thread.get(); 5487 if (mCblk->frameCount > frames){ 5488 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5489 uint32_t startFrames = (mCblk->frameCount - frames); 5490 pInBuffer = new Buffer; 5491 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5492 pInBuffer->frameCount = startFrames; 5493 pInBuffer->i16 = pInBuffer->mBuffer; 5494 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5495 mBufferQueue.add(pInBuffer); 5496 } else { 5497 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5498 } 5499 } 5500 } 5501 } 5502 5503 while (waitTimeLeftMs) { 5504 // First write pending buffers, then new data 5505 if (mBufferQueue.size()) { 5506 pInBuffer = mBufferQueue.itemAt(0); 5507 } else { 5508 pInBuffer = &inBuffer; 5509 } 5510 5511 if (pInBuffer->frameCount == 0) { 5512 break; 5513 } 5514 5515 if (mOutBuffer.frameCount == 0) { 5516 mOutBuffer.frameCount = pInBuffer->frameCount; 5517 nsecs_t startTime = systemTime(); 5518 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5519 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5520 outputBufferFull = true; 5521 break; 5522 } 5523 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5524 if (waitTimeLeftMs >= waitTimeMs) { 5525 waitTimeLeftMs -= waitTimeMs; 5526 } else { 5527 waitTimeLeftMs = 0; 5528 } 5529 } 5530 5531 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5532 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5533 mCblk->stepUser(outFrames); 5534 pInBuffer->frameCount -= outFrames; 5535 pInBuffer->i16 += outFrames * channelCount; 5536 mOutBuffer.frameCount -= outFrames; 5537 mOutBuffer.i16 += outFrames * channelCount; 5538 5539 if (pInBuffer->frameCount == 0) { 5540 if (mBufferQueue.size()) { 5541 mBufferQueue.removeAt(0); 5542 delete [] pInBuffer->mBuffer; 5543 delete pInBuffer; 5544 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5545 } else { 5546 break; 5547 } 5548 } 5549 } 5550 5551 // If we could not write all frames, allocate a buffer and queue it for next time. 5552 if (inBuffer.frameCount) { 5553 sp<ThreadBase> thread = mThread.promote(); 5554 if (thread != 0 && !thread->standby()) { 5555 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5556 pInBuffer = new Buffer; 5557 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5558 pInBuffer->frameCount = inBuffer.frameCount; 5559 pInBuffer->i16 = pInBuffer->mBuffer; 5560 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5561 mBufferQueue.add(pInBuffer); 5562 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5563 } else { 5564 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5565 } 5566 } 5567 } 5568 5569 // Calling write() with a 0 length buffer, means that no more data will be written: 5570 // If no more buffers are pending, fill output track buffer to make sure it is started 5571 // by output mixer. 5572 if (frames == 0 && mBufferQueue.size() == 0) { 5573 if (mCblk->user < mCblk->frameCount) { 5574 frames = mCblk->frameCount - mCblk->user; 5575 pInBuffer = new Buffer; 5576 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5577 pInBuffer->frameCount = frames; 5578 pInBuffer->i16 = pInBuffer->mBuffer; 5579 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5580 mBufferQueue.add(pInBuffer); 5581 } else if (mActive) { 5582 stop(); 5583 } 5584 } 5585 5586 return outputBufferFull; 5587} 5588 5589status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5590{ 5591 int active; 5592 status_t result; 5593 audio_track_cblk_t* cblk = mCblk; 5594 uint32_t framesReq = buffer->frameCount; 5595 5596// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5597 buffer->frameCount = 0; 5598 5599 uint32_t framesAvail = cblk->framesAvailable(); 5600 5601 5602 if (framesAvail == 0) { 5603 Mutex::Autolock _l(cblk->lock); 5604 goto start_loop_here; 5605 while (framesAvail == 0) { 5606 active = mActive; 5607 if (CC_UNLIKELY(!active)) { 5608 ALOGV("Not active and NO_MORE_BUFFERS"); 5609 return NO_MORE_BUFFERS; 5610 } 5611 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5612 if (result != NO_ERROR) { 5613 return NO_MORE_BUFFERS; 5614 } 5615 // read the server count again 5616 start_loop_here: 5617 framesAvail = cblk->framesAvailable_l(); 5618 } 5619 } 5620 5621// if (framesAvail < framesReq) { 5622// return NO_MORE_BUFFERS; 5623// } 5624 5625 if (framesReq > framesAvail) { 5626 framesReq = framesAvail; 5627 } 5628 5629 uint32_t u = cblk->user; 5630 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5631 5632 if (framesReq > bufferEnd - u) { 5633 framesReq = bufferEnd - u; 5634 } 5635 5636 buffer->frameCount = framesReq; 5637 buffer->raw = (void *)cblk->buffer(u); 5638 return NO_ERROR; 5639} 5640 5641 5642void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5643{ 5644 size_t size = mBufferQueue.size(); 5645 5646 for (size_t i = 0; i < size; i++) { 5647 Buffer *pBuffer = mBufferQueue.itemAt(i); 5648 delete [] pBuffer->mBuffer; 5649 delete pBuffer; 5650 } 5651 mBufferQueue.clear(); 5652} 5653 5654// ---------------------------------------------------------------------------- 5655 5656AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5657 : RefBase(), 5658 mAudioFlinger(audioFlinger), 5659 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5660 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5661 mPid(pid), 5662 mTimedTrackCount(0) 5663{ 5664 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5665} 5666 5667// Client destructor must be called with AudioFlinger::mLock held 5668AudioFlinger::Client::~Client() 5669{ 5670 mAudioFlinger->removeClient_l(mPid); 5671} 5672 5673sp<MemoryDealer> AudioFlinger::Client::heap() const 5674{ 5675 return mMemoryDealer; 5676} 5677 5678// Reserve one of the limited slots for a timed audio track associated 5679// with this client 5680bool AudioFlinger::Client::reserveTimedTrack() 5681{ 5682 const int kMaxTimedTracksPerClient = 4; 5683 5684 Mutex::Autolock _l(mTimedTrackLock); 5685 5686 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5687 ALOGW("can not create timed track - pid %d has exceeded the limit", 5688 mPid); 5689 return false; 5690 } 5691 5692 mTimedTrackCount++; 5693 return true; 5694} 5695 5696// Release a slot for a timed audio track 5697void AudioFlinger::Client::releaseTimedTrack() 5698{ 5699 Mutex::Autolock _l(mTimedTrackLock); 5700 mTimedTrackCount--; 5701} 5702 5703// ---------------------------------------------------------------------------- 5704 5705AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5706 const sp<IAudioFlingerClient>& client, 5707 pid_t pid) 5708 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5709{ 5710} 5711 5712AudioFlinger::NotificationClient::~NotificationClient() 5713{ 5714} 5715 5716void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5717{ 5718 sp<NotificationClient> keep(this); 5719 mAudioFlinger->removeNotificationClient(mPid); 5720} 5721 5722// ---------------------------------------------------------------------------- 5723 5724AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5725 : BnAudioTrack(), 5726 mTrack(track) 5727{ 5728} 5729 5730AudioFlinger::TrackHandle::~TrackHandle() { 5731 // just stop the track on deletion, associated resources 5732 // will be freed from the main thread once all pending buffers have 5733 // been played. Unless it's not in the active track list, in which 5734 // case we free everything now... 5735 mTrack->destroy(); 5736} 5737 5738sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5739 return mTrack->getCblk(); 5740} 5741 5742status_t AudioFlinger::TrackHandle::start() { 5743 return mTrack->start(); 5744} 5745 5746void AudioFlinger::TrackHandle::stop() { 5747 mTrack->stop(); 5748} 5749 5750void AudioFlinger::TrackHandle::flush() { 5751 mTrack->flush(); 5752} 5753 5754void AudioFlinger::TrackHandle::mute(bool e) { 5755 mTrack->mute(e); 5756} 5757 5758void AudioFlinger::TrackHandle::pause() { 5759 mTrack->pause(); 5760} 5761 5762status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5763{ 5764 return mTrack->attachAuxEffect(EffectId); 5765} 5766 5767status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5768 sp<IMemory>* buffer) { 5769 if (!mTrack->isTimedTrack()) 5770 return INVALID_OPERATION; 5771 5772 PlaybackThread::TimedTrack* tt = 5773 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5774 return tt->allocateTimedBuffer(size, buffer); 5775} 5776 5777status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5778 int64_t pts) { 5779 if (!mTrack->isTimedTrack()) 5780 return INVALID_OPERATION; 5781 5782 PlaybackThread::TimedTrack* tt = 5783 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5784 return tt->queueTimedBuffer(buffer, pts); 5785} 5786 5787status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5788 const LinearTransform& xform, int target) { 5789 5790 if (!mTrack->isTimedTrack()) 5791 return INVALID_OPERATION; 5792 5793 PlaybackThread::TimedTrack* tt = 5794 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5795 return tt->setMediaTimeTransform( 5796 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5797} 5798 5799status_t AudioFlinger::TrackHandle::onTransact( 5800 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5801{ 5802 return BnAudioTrack::onTransact(code, data, reply, flags); 5803} 5804 5805// ---------------------------------------------------------------------------- 5806 5807sp<IAudioRecord> AudioFlinger::openRecord( 5808 pid_t pid, 5809 audio_io_handle_t input, 5810 uint32_t sampleRate, 5811 audio_format_t format, 5812 uint32_t channelMask, 5813 int frameCount, 5814 IAudioFlinger::track_flags_t flags, 5815 int *sessionId, 5816 status_t *status) 5817{ 5818 sp<RecordThread::RecordTrack> recordTrack; 5819 sp<RecordHandle> recordHandle; 5820 sp<Client> client; 5821 status_t lStatus; 5822 RecordThread *thread; 5823 size_t inFrameCount; 5824 int lSessionId; 5825 5826 // check calling permissions 5827 if (!recordingAllowed()) { 5828 lStatus = PERMISSION_DENIED; 5829 goto Exit; 5830 } 5831 5832 // add client to list 5833 { // scope for mLock 5834 Mutex::Autolock _l(mLock); 5835 thread = checkRecordThread_l(input); 5836 if (thread == NULL) { 5837 lStatus = BAD_VALUE; 5838 goto Exit; 5839 } 5840 5841 client = registerPid_l(pid); 5842 5843 // If no audio session id is provided, create one here 5844 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5845 lSessionId = *sessionId; 5846 } else { 5847 lSessionId = nextUniqueId(); 5848 if (sessionId != NULL) { 5849 *sessionId = lSessionId; 5850 } 5851 } 5852 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5853 recordTrack = thread->createRecordTrack_l(client, 5854 sampleRate, 5855 format, 5856 channelMask, 5857 frameCount, 5858 lSessionId, 5859 &lStatus); 5860 } 5861 if (lStatus != NO_ERROR) { 5862 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5863 // destructor is called by the TrackBase destructor with mLock held 5864 client.clear(); 5865 recordTrack.clear(); 5866 goto Exit; 5867 } 5868 5869 // return to handle to client 5870 recordHandle = new RecordHandle(recordTrack); 5871 lStatus = NO_ERROR; 5872 5873Exit: 5874 if (status) { 5875 *status = lStatus; 5876 } 5877 return recordHandle; 5878} 5879 5880// ---------------------------------------------------------------------------- 5881 5882AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5883 : BnAudioRecord(), 5884 mRecordTrack(recordTrack) 5885{ 5886} 5887 5888AudioFlinger::RecordHandle::~RecordHandle() { 5889 stop(); 5890} 5891 5892sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5893 return mRecordTrack->getCblk(); 5894} 5895 5896status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5897 ALOGV("RecordHandle::start()"); 5898 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5899} 5900 5901void AudioFlinger::RecordHandle::stop() { 5902 ALOGV("RecordHandle::stop()"); 5903 mRecordTrack->stop(); 5904} 5905 5906status_t AudioFlinger::RecordHandle::onTransact( 5907 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5908{ 5909 return BnAudioRecord::onTransact(code, data, reply, flags); 5910} 5911 5912// ---------------------------------------------------------------------------- 5913 5914AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5915 AudioStreamIn *input, 5916 uint32_t sampleRate, 5917 uint32_t channels, 5918 audio_io_handle_t id, 5919 uint32_t device) : 5920 ThreadBase(audioFlinger, id, device, RECORD), 5921 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5922 // mRsmpInIndex and mInputBytes set by readInputParameters() 5923 mReqChannelCount(popcount(channels)), 5924 mReqSampleRate(sampleRate) 5925 // mBytesRead is only meaningful while active, and so is cleared in start() 5926 // (but might be better to also clear here for dump?) 5927{ 5928 snprintf(mName, kNameLength, "AudioIn_%X", id); 5929 5930 readInputParameters(); 5931} 5932 5933 5934AudioFlinger::RecordThread::~RecordThread() 5935{ 5936 delete[] mRsmpInBuffer; 5937 delete mResampler; 5938 delete[] mRsmpOutBuffer; 5939} 5940 5941void AudioFlinger::RecordThread::onFirstRef() 5942{ 5943 run(mName, PRIORITY_URGENT_AUDIO); 5944} 5945 5946status_t AudioFlinger::RecordThread::readyToRun() 5947{ 5948 status_t status = initCheck(); 5949 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5950 return status; 5951} 5952 5953bool AudioFlinger::RecordThread::threadLoop() 5954{ 5955 AudioBufferProvider::Buffer buffer; 5956 sp<RecordTrack> activeTrack; 5957 Vector< sp<EffectChain> > effectChains; 5958 5959 nsecs_t lastWarning = 0; 5960 5961 acquireWakeLock(); 5962 5963 // start recording 5964 while (!exitPending()) { 5965 5966 processConfigEvents(); 5967 5968 { // scope for mLock 5969 Mutex::Autolock _l(mLock); 5970 checkForNewParameters_l(); 5971 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5972 if (!mStandby) { 5973 mInput->stream->common.standby(&mInput->stream->common); 5974 mStandby = true; 5975 } 5976 5977 if (exitPending()) break; 5978 5979 releaseWakeLock_l(); 5980 ALOGV("RecordThread: loop stopping"); 5981 // go to sleep 5982 mWaitWorkCV.wait(mLock); 5983 ALOGV("RecordThread: loop starting"); 5984 acquireWakeLock_l(); 5985 continue; 5986 } 5987 if (mActiveTrack != 0) { 5988 if (mActiveTrack->mState == TrackBase::PAUSING) { 5989 if (!mStandby) { 5990 mInput->stream->common.standby(&mInput->stream->common); 5991 mStandby = true; 5992 } 5993 mActiveTrack.clear(); 5994 mStartStopCond.broadcast(); 5995 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5996 if (mReqChannelCount != mActiveTrack->channelCount()) { 5997 mActiveTrack.clear(); 5998 mStartStopCond.broadcast(); 5999 } else if (mBytesRead != 0) { 6000 // record start succeeds only if first read from audio input 6001 // succeeds 6002 if (mBytesRead > 0) { 6003 mActiveTrack->mState = TrackBase::ACTIVE; 6004 } else { 6005 mActiveTrack.clear(); 6006 } 6007 mStartStopCond.broadcast(); 6008 } 6009 mStandby = false; 6010 } 6011 } 6012 lockEffectChains_l(effectChains); 6013 } 6014 6015 if (mActiveTrack != 0) { 6016 if (mActiveTrack->mState != TrackBase::ACTIVE && 6017 mActiveTrack->mState != TrackBase::RESUMING) { 6018 unlockEffectChains(effectChains); 6019 usleep(kRecordThreadSleepUs); 6020 continue; 6021 } 6022 for (size_t i = 0; i < effectChains.size(); i ++) { 6023 effectChains[i]->process_l(); 6024 } 6025 6026 buffer.frameCount = mFrameCount; 6027 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6028 size_t framesOut = buffer.frameCount; 6029 if (mResampler == NULL) { 6030 // no resampling 6031 while (framesOut) { 6032 size_t framesIn = mFrameCount - mRsmpInIndex; 6033 if (framesIn) { 6034 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6035 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6036 if (framesIn > framesOut) 6037 framesIn = framesOut; 6038 mRsmpInIndex += framesIn; 6039 framesOut -= framesIn; 6040 if ((int)mChannelCount == mReqChannelCount || 6041 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6042 memcpy(dst, src, framesIn * mFrameSize); 6043 } else { 6044 int16_t *src16 = (int16_t *)src; 6045 int16_t *dst16 = (int16_t *)dst; 6046 if (mChannelCount == 1) { 6047 while (framesIn--) { 6048 *dst16++ = *src16; 6049 *dst16++ = *src16++; 6050 } 6051 } else { 6052 while (framesIn--) { 6053 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6054 src16 += 2; 6055 } 6056 } 6057 } 6058 } 6059 if (framesOut && mFrameCount == mRsmpInIndex) { 6060 if (framesOut == mFrameCount && 6061 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6062 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6063 framesOut = 0; 6064 } else { 6065 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6066 mRsmpInIndex = 0; 6067 } 6068 if (mBytesRead < 0) { 6069 ALOGE("Error reading audio input"); 6070 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6071 // Force input into standby so that it tries to 6072 // recover at next read attempt 6073 mInput->stream->common.standby(&mInput->stream->common); 6074 usleep(kRecordThreadSleepUs); 6075 } 6076 mRsmpInIndex = mFrameCount; 6077 framesOut = 0; 6078 buffer.frameCount = 0; 6079 } 6080 } 6081 } 6082 } else { 6083 // resampling 6084 6085 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6086 // alter output frame count as if we were expecting stereo samples 6087 if (mChannelCount == 1 && mReqChannelCount == 1) { 6088 framesOut >>= 1; 6089 } 6090 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6091 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6092 // are 32 bit aligned which should be always true. 6093 if (mChannelCount == 2 && mReqChannelCount == 1) { 6094 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6095 // the resampler always outputs stereo samples: do post stereo to mono conversion 6096 int16_t *src = (int16_t *)mRsmpOutBuffer; 6097 int16_t *dst = buffer.i16; 6098 while (framesOut--) { 6099 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6100 src += 2; 6101 } 6102 } else { 6103 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6104 } 6105 6106 } 6107 if (mFramestoDrop == 0) { 6108 mActiveTrack->releaseBuffer(&buffer); 6109 } else { 6110 if (mFramestoDrop > 0) { 6111 mFramestoDrop -= buffer.frameCount; 6112 if (mFramestoDrop <= 0) { 6113 clearSyncStartEvent(); 6114 } 6115 } else { 6116 mFramestoDrop += buffer.frameCount; 6117 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6118 mSyncStartEvent->isCancelled()) { 6119 ALOGW("Synced record %s, session %d, trigger session %d", 6120 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6121 mActiveTrack->sessionId(), 6122 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6123 clearSyncStartEvent(); 6124 } 6125 } 6126 } 6127 mActiveTrack->overflow(); 6128 } 6129 // client isn't retrieving buffers fast enough 6130 else { 6131 if (!mActiveTrack->setOverflow()) { 6132 nsecs_t now = systemTime(); 6133 if ((now - lastWarning) > kWarningThrottleNs) { 6134 ALOGW("RecordThread: buffer overflow"); 6135 lastWarning = now; 6136 } 6137 } 6138 // Release the processor for a while before asking for a new buffer. 6139 // This will give the application more chance to read from the buffer and 6140 // clear the overflow. 6141 usleep(kRecordThreadSleepUs); 6142 } 6143 } 6144 // enable changes in effect chain 6145 unlockEffectChains(effectChains); 6146 effectChains.clear(); 6147 } 6148 6149 if (!mStandby) { 6150 mInput->stream->common.standby(&mInput->stream->common); 6151 } 6152 mActiveTrack.clear(); 6153 6154 mStartStopCond.broadcast(); 6155 6156 releaseWakeLock(); 6157 6158 ALOGV("RecordThread %p exiting", this); 6159 return false; 6160} 6161 6162 6163sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6164 const sp<AudioFlinger::Client>& client, 6165 uint32_t sampleRate, 6166 audio_format_t format, 6167 int channelMask, 6168 int frameCount, 6169 int sessionId, 6170 status_t *status) 6171{ 6172 sp<RecordTrack> track; 6173 status_t lStatus; 6174 6175 lStatus = initCheck(); 6176 if (lStatus != NO_ERROR) { 6177 ALOGE("Audio driver not initialized."); 6178 goto Exit; 6179 } 6180 6181 { // scope for mLock 6182 Mutex::Autolock _l(mLock); 6183 6184 track = new RecordTrack(this, client, sampleRate, 6185 format, channelMask, frameCount, sessionId); 6186 6187 if (track->getCblk() == 0) { 6188 lStatus = NO_MEMORY; 6189 goto Exit; 6190 } 6191 6192 mTrack = track.get(); 6193 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6194 bool suspend = audio_is_bluetooth_sco_device( 6195 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6196 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6197 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6198 } 6199 lStatus = NO_ERROR; 6200 6201Exit: 6202 if (status) { 6203 *status = lStatus; 6204 } 6205 return track; 6206} 6207 6208status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6209 AudioSystem::sync_event_t event, 6210 int triggerSession) 6211{ 6212 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6213 sp<ThreadBase> strongMe = this; 6214 status_t status = NO_ERROR; 6215 6216 if (event == AudioSystem::SYNC_EVENT_NONE) { 6217 clearSyncStartEvent(); 6218 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6219 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6220 triggerSession, 6221 recordTrack->sessionId(), 6222 syncStartEventCallback, 6223 this); 6224 // Sync event can be cancelled by the trigger session if the track is not in a 6225 // compatible state in which case we start record immediately 6226 if (mSyncStartEvent->isCancelled()) { 6227 clearSyncStartEvent(); 6228 } else { 6229 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6230 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6231 } 6232 } 6233 6234 { 6235 AutoMutex lock(mLock); 6236 if (mActiveTrack != 0) { 6237 if (recordTrack != mActiveTrack.get()) { 6238 status = -EBUSY; 6239 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6240 mActiveTrack->mState = TrackBase::ACTIVE; 6241 } 6242 return status; 6243 } 6244 6245 recordTrack->mState = TrackBase::IDLE; 6246 mActiveTrack = recordTrack; 6247 mLock.unlock(); 6248 status_t status = AudioSystem::startInput(mId); 6249 mLock.lock(); 6250 if (status != NO_ERROR) { 6251 mActiveTrack.clear(); 6252 clearSyncStartEvent(); 6253 return status; 6254 } 6255 mRsmpInIndex = mFrameCount; 6256 mBytesRead = 0; 6257 if (mResampler != NULL) { 6258 mResampler->reset(); 6259 } 6260 mActiveTrack->mState = TrackBase::RESUMING; 6261 // signal thread to start 6262 ALOGV("Signal record thread"); 6263 mWaitWorkCV.signal(); 6264 // do not wait for mStartStopCond if exiting 6265 if (exitPending()) { 6266 mActiveTrack.clear(); 6267 status = INVALID_OPERATION; 6268 goto startError; 6269 } 6270 mStartStopCond.wait(mLock); 6271 if (mActiveTrack == 0) { 6272 ALOGV("Record failed to start"); 6273 status = BAD_VALUE; 6274 goto startError; 6275 } 6276 ALOGV("Record started OK"); 6277 return status; 6278 } 6279startError: 6280 AudioSystem::stopInput(mId); 6281 clearSyncStartEvent(); 6282 return status; 6283} 6284 6285void AudioFlinger::RecordThread::clearSyncStartEvent() 6286{ 6287 if (mSyncStartEvent != 0) { 6288 mSyncStartEvent->cancel(); 6289 } 6290 mSyncStartEvent.clear(); 6291 mFramestoDrop = 0; 6292} 6293 6294void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6295{ 6296 sp<SyncEvent> strongEvent = event.promote(); 6297 6298 if (strongEvent != 0) { 6299 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6300 me->handleSyncStartEvent(strongEvent); 6301 } 6302} 6303 6304void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6305{ 6306 if (event == mSyncStartEvent) { 6307 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6308 // from audio HAL 6309 mFramestoDrop = mFrameCount * 2; 6310 } 6311} 6312 6313void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6314 ALOGV("RecordThread::stop"); 6315 sp<ThreadBase> strongMe = this; 6316 { 6317 AutoMutex lock(mLock); 6318 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6319 mActiveTrack->mState = TrackBase::PAUSING; 6320 // do not wait for mStartStopCond if exiting 6321 if (exitPending()) { 6322 return; 6323 } 6324 mStartStopCond.wait(mLock); 6325 // if we have been restarted, recordTrack == mActiveTrack.get() here 6326 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6327 mLock.unlock(); 6328 AudioSystem::stopInput(mId); 6329 mLock.lock(); 6330 ALOGV("Record stopped OK"); 6331 } 6332 } 6333 } 6334} 6335 6336bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6337{ 6338 return false; 6339} 6340 6341status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6342{ 6343 if (!isValidSyncEvent(event)) { 6344 return BAD_VALUE; 6345 } 6346 6347 Mutex::Autolock _l(mLock); 6348 6349 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6350 mTrack->setSyncEvent(event); 6351 return NO_ERROR; 6352 } 6353 return NAME_NOT_FOUND; 6354} 6355 6356status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6357{ 6358 const size_t SIZE = 256; 6359 char buffer[SIZE]; 6360 String8 result; 6361 6362 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6363 result.append(buffer); 6364 6365 if (mActiveTrack != 0) { 6366 result.append("Active Track:\n"); 6367 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6368 mActiveTrack->dump(buffer, SIZE); 6369 result.append(buffer); 6370 6371 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6372 result.append(buffer); 6373 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6374 result.append(buffer); 6375 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6376 result.append(buffer); 6377 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6378 result.append(buffer); 6379 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6380 result.append(buffer); 6381 6382 6383 } else { 6384 result.append("No record client\n"); 6385 } 6386 write(fd, result.string(), result.size()); 6387 6388 dumpBase(fd, args); 6389 dumpEffectChains(fd, args); 6390 6391 return NO_ERROR; 6392} 6393 6394// AudioBufferProvider interface 6395status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6396{ 6397 size_t framesReq = buffer->frameCount; 6398 size_t framesReady = mFrameCount - mRsmpInIndex; 6399 int channelCount; 6400 6401 if (framesReady == 0) { 6402 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6403 if (mBytesRead < 0) { 6404 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6405 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6406 // Force input into standby so that it tries to 6407 // recover at next read attempt 6408 mInput->stream->common.standby(&mInput->stream->common); 6409 usleep(kRecordThreadSleepUs); 6410 } 6411 buffer->raw = NULL; 6412 buffer->frameCount = 0; 6413 return NOT_ENOUGH_DATA; 6414 } 6415 mRsmpInIndex = 0; 6416 framesReady = mFrameCount; 6417 } 6418 6419 if (framesReq > framesReady) { 6420 framesReq = framesReady; 6421 } 6422 6423 if (mChannelCount == 1 && mReqChannelCount == 2) { 6424 channelCount = 1; 6425 } else { 6426 channelCount = 2; 6427 } 6428 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6429 buffer->frameCount = framesReq; 6430 return NO_ERROR; 6431} 6432 6433// AudioBufferProvider interface 6434void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6435{ 6436 mRsmpInIndex += buffer->frameCount; 6437 buffer->frameCount = 0; 6438} 6439 6440bool AudioFlinger::RecordThread::checkForNewParameters_l() 6441{ 6442 bool reconfig = false; 6443 6444 while (!mNewParameters.isEmpty()) { 6445 status_t status = NO_ERROR; 6446 String8 keyValuePair = mNewParameters[0]; 6447 AudioParameter param = AudioParameter(keyValuePair); 6448 int value; 6449 audio_format_t reqFormat = mFormat; 6450 int reqSamplingRate = mReqSampleRate; 6451 int reqChannelCount = mReqChannelCount; 6452 6453 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6454 reqSamplingRate = value; 6455 reconfig = true; 6456 } 6457 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6458 reqFormat = (audio_format_t) value; 6459 reconfig = true; 6460 } 6461 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6462 reqChannelCount = popcount(value); 6463 reconfig = true; 6464 } 6465 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6466 // do not accept frame count changes if tracks are open as the track buffer 6467 // size depends on frame count and correct behavior would not be guaranteed 6468 // if frame count is changed after track creation 6469 if (mActiveTrack != 0) { 6470 status = INVALID_OPERATION; 6471 } else { 6472 reconfig = true; 6473 } 6474 } 6475 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6476 // forward device change to effects that have requested to be 6477 // aware of attached audio device. 6478 for (size_t i = 0; i < mEffectChains.size(); i++) { 6479 mEffectChains[i]->setDevice_l(value); 6480 } 6481 // store input device and output device but do not forward output device to audio HAL. 6482 // Note that status is ignored by the caller for output device 6483 // (see AudioFlinger::setParameters() 6484 if (value & AUDIO_DEVICE_OUT_ALL) { 6485 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6486 status = BAD_VALUE; 6487 } else { 6488 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6489 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6490 if (mTrack != NULL) { 6491 bool suspend = audio_is_bluetooth_sco_device( 6492 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6493 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6494 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6495 } 6496 } 6497 mDevice |= (uint32_t)value; 6498 } 6499 if (status == NO_ERROR) { 6500 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6501 if (status == INVALID_OPERATION) { 6502 mInput->stream->common.standby(&mInput->stream->common); 6503 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6504 keyValuePair.string()); 6505 } 6506 if (reconfig) { 6507 if (status == BAD_VALUE && 6508 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6509 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6510 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6511 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6512 (reqChannelCount <= FCC_2)) { 6513 status = NO_ERROR; 6514 } 6515 if (status == NO_ERROR) { 6516 readInputParameters(); 6517 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6518 } 6519 } 6520 } 6521 6522 mNewParameters.removeAt(0); 6523 6524 mParamStatus = status; 6525 mParamCond.signal(); 6526 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6527 // already timed out waiting for the status and will never signal the condition. 6528 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6529 } 6530 return reconfig; 6531} 6532 6533String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6534{ 6535 char *s; 6536 String8 out_s8 = String8(); 6537 6538 Mutex::Autolock _l(mLock); 6539 if (initCheck() != NO_ERROR) { 6540 return out_s8; 6541 } 6542 6543 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6544 out_s8 = String8(s); 6545 free(s); 6546 return out_s8; 6547} 6548 6549void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6550 AudioSystem::OutputDescriptor desc; 6551 void *param2 = NULL; 6552 6553 switch (event) { 6554 case AudioSystem::INPUT_OPENED: 6555 case AudioSystem::INPUT_CONFIG_CHANGED: 6556 desc.channels = mChannelMask; 6557 desc.samplingRate = mSampleRate; 6558 desc.format = mFormat; 6559 desc.frameCount = mFrameCount; 6560 desc.latency = 0; 6561 param2 = &desc; 6562 break; 6563 6564 case AudioSystem::INPUT_CLOSED: 6565 default: 6566 break; 6567 } 6568 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6569} 6570 6571void AudioFlinger::RecordThread::readInputParameters() 6572{ 6573 delete mRsmpInBuffer; 6574 // mRsmpInBuffer is always assigned a new[] below 6575 delete mRsmpOutBuffer; 6576 mRsmpOutBuffer = NULL; 6577 delete mResampler; 6578 mResampler = NULL; 6579 6580 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6581 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6582 mChannelCount = (uint16_t)popcount(mChannelMask); 6583 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6584 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6585 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6586 mFrameCount = mInputBytes / mFrameSize; 6587 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6588 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6589 6590 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6591 { 6592 int channelCount; 6593 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6594 // stereo to mono post process as the resampler always outputs stereo. 6595 if (mChannelCount == 1 && mReqChannelCount == 2) { 6596 channelCount = 1; 6597 } else { 6598 channelCount = 2; 6599 } 6600 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6601 mResampler->setSampleRate(mSampleRate); 6602 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6603 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6604 6605 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6606 if (mChannelCount == 1 && mReqChannelCount == 1) { 6607 mFrameCount >>= 1; 6608 } 6609 6610 } 6611 mRsmpInIndex = mFrameCount; 6612} 6613 6614unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6615{ 6616 Mutex::Autolock _l(mLock); 6617 if (initCheck() != NO_ERROR) { 6618 return 0; 6619 } 6620 6621 return mInput->stream->get_input_frames_lost(mInput->stream); 6622} 6623 6624uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6625{ 6626 Mutex::Autolock _l(mLock); 6627 uint32_t result = 0; 6628 if (getEffectChain_l(sessionId) != 0) { 6629 result = EFFECT_SESSION; 6630 } 6631 6632 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6633 result |= TRACK_SESSION; 6634 } 6635 6636 return result; 6637} 6638 6639AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6640{ 6641 Mutex::Autolock _l(mLock); 6642 return mTrack; 6643} 6644 6645AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6646{ 6647 Mutex::Autolock _l(mLock); 6648 return mInput; 6649} 6650 6651AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6652{ 6653 Mutex::Autolock _l(mLock); 6654 AudioStreamIn *input = mInput; 6655 mInput = NULL; 6656 return input; 6657} 6658 6659// this method must always be called either with ThreadBase mLock held or inside the thread loop 6660audio_stream_t* AudioFlinger::RecordThread::stream() const 6661{ 6662 if (mInput == NULL) { 6663 return NULL; 6664 } 6665 return &mInput->stream->common; 6666} 6667 6668 6669// ---------------------------------------------------------------------------- 6670 6671audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6672{ 6673 if (!settingsAllowed()) { 6674 return 0; 6675 } 6676 Mutex::Autolock _l(mLock); 6677 return loadHwModule_l(name); 6678} 6679 6680// loadHwModule_l() must be called with AudioFlinger::mLock held 6681audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6682{ 6683 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6684 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6685 ALOGW("loadHwModule() module %s already loaded", name); 6686 return mAudioHwDevs.keyAt(i); 6687 } 6688 } 6689 6690 audio_hw_device_t *dev; 6691 6692 int rc = load_audio_interface(name, &dev); 6693 if (rc) { 6694 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6695 return 0; 6696 } 6697 6698 mHardwareStatus = AUDIO_HW_INIT; 6699 rc = dev->init_check(dev); 6700 mHardwareStatus = AUDIO_HW_IDLE; 6701 if (rc) { 6702 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6703 return 0; 6704 } 6705 6706 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6707 (NULL != dev->set_master_volume)) { 6708 AutoMutex lock(mHardwareLock); 6709 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6710 dev->set_master_volume(dev, mMasterVolume); 6711 mHardwareStatus = AUDIO_HW_IDLE; 6712 } 6713 6714 audio_module_handle_t handle = nextUniqueId(); 6715 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6716 6717 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6718 name, dev->common.module->name, dev->common.module->id, handle); 6719 6720 return handle; 6721 6722} 6723 6724audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6725 audio_devices_t *pDevices, 6726 uint32_t *pSamplingRate, 6727 audio_format_t *pFormat, 6728 audio_channel_mask_t *pChannelMask, 6729 uint32_t *pLatencyMs, 6730 audio_output_flags_t flags) 6731{ 6732 status_t status; 6733 PlaybackThread *thread = NULL; 6734 struct audio_config config = { 6735 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6736 channel_mask: pChannelMask ? *pChannelMask : 0, 6737 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6738 }; 6739 audio_stream_out_t *outStream = NULL; 6740 audio_hw_device_t *outHwDev; 6741 6742 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6743 module, 6744 (pDevices != NULL) ? (int)*pDevices : 0, 6745 config.sample_rate, 6746 config.format, 6747 config.channel_mask, 6748 flags); 6749 6750 if (pDevices == NULL || *pDevices == 0) { 6751 return 0; 6752 } 6753 6754 Mutex::Autolock _l(mLock); 6755 6756 outHwDev = findSuitableHwDev_l(module, *pDevices); 6757 if (outHwDev == NULL) 6758 return 0; 6759 6760 audio_io_handle_t id = nextUniqueId(); 6761 6762 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6763 6764 status = outHwDev->open_output_stream(outHwDev, 6765 id, 6766 *pDevices, 6767 (audio_output_flags_t)flags, 6768 &config, 6769 &outStream); 6770 6771 mHardwareStatus = AUDIO_HW_IDLE; 6772 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6773 outStream, 6774 config.sample_rate, 6775 config.format, 6776 config.channel_mask, 6777 status); 6778 6779 if (status == NO_ERROR && outStream != NULL) { 6780 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6781 6782 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6783 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6784 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6785 thread = new DirectOutputThread(this, output, id, *pDevices); 6786 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6787 } else { 6788 thread = new MixerThread(this, output, id, *pDevices); 6789 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6790 } 6791 mPlaybackThreads.add(id, thread); 6792 6793 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6794 if (pFormat != NULL) *pFormat = config.format; 6795 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6796 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6797 6798 // notify client processes of the new output creation 6799 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6800 6801 // the first primary output opened designates the primary hw device 6802 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6803 ALOGI("Using module %d has the primary audio interface", module); 6804 mPrimaryHardwareDev = outHwDev; 6805 6806 AutoMutex lock(mHardwareLock); 6807 mHardwareStatus = AUDIO_HW_SET_MODE; 6808 outHwDev->set_mode(outHwDev, mMode); 6809 6810 // Determine the level of master volume support the primary audio HAL has, 6811 // and set the initial master volume at the same time. 6812 float initialVolume = 1.0; 6813 mMasterVolumeSupportLvl = MVS_NONE; 6814 6815 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6816 if ((NULL != outHwDev->get_master_volume) && 6817 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6818 mMasterVolumeSupportLvl = MVS_FULL; 6819 } else { 6820 mMasterVolumeSupportLvl = MVS_SETONLY; 6821 initialVolume = 1.0; 6822 } 6823 6824 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6825 if ((NULL == outHwDev->set_master_volume) || 6826 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6827 mMasterVolumeSupportLvl = MVS_NONE; 6828 } 6829 // now that we have a primary device, initialize master volume on other devices 6830 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6831 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6832 6833 if ((dev != mPrimaryHardwareDev) && 6834 (NULL != dev->set_master_volume)) { 6835 dev->set_master_volume(dev, initialVolume); 6836 } 6837 } 6838 mHardwareStatus = AUDIO_HW_IDLE; 6839 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6840 ? initialVolume 6841 : 1.0; 6842 mMasterVolume = initialVolume; 6843 } 6844 return id; 6845 } 6846 6847 return 0; 6848} 6849 6850audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6851 audio_io_handle_t output2) 6852{ 6853 Mutex::Autolock _l(mLock); 6854 MixerThread *thread1 = checkMixerThread_l(output1); 6855 MixerThread *thread2 = checkMixerThread_l(output2); 6856 6857 if (thread1 == NULL || thread2 == NULL) { 6858 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6859 return 0; 6860 } 6861 6862 audio_io_handle_t id = nextUniqueId(); 6863 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6864 thread->addOutputTrack(thread2); 6865 mPlaybackThreads.add(id, thread); 6866 // notify client processes of the new output creation 6867 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6868 return id; 6869} 6870 6871status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6872{ 6873 // keep strong reference on the playback thread so that 6874 // it is not destroyed while exit() is executed 6875 sp<PlaybackThread> thread; 6876 { 6877 Mutex::Autolock _l(mLock); 6878 thread = checkPlaybackThread_l(output); 6879 if (thread == NULL) { 6880 return BAD_VALUE; 6881 } 6882 6883 ALOGV("closeOutput() %d", output); 6884 6885 if (thread->type() == ThreadBase::MIXER) { 6886 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6887 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6888 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6889 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6890 } 6891 } 6892 } 6893 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6894 mPlaybackThreads.removeItem(output); 6895 } 6896 thread->exit(); 6897 // The thread entity (active unit of execution) is no longer running here, 6898 // but the ThreadBase container still exists. 6899 6900 if (thread->type() != ThreadBase::DUPLICATING) { 6901 AudioStreamOut *out = thread->clearOutput(); 6902 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6903 // from now on thread->mOutput is NULL 6904 out->hwDev->close_output_stream(out->hwDev, out->stream); 6905 delete out; 6906 } 6907 return NO_ERROR; 6908} 6909 6910status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6911{ 6912 Mutex::Autolock _l(mLock); 6913 PlaybackThread *thread = checkPlaybackThread_l(output); 6914 6915 if (thread == NULL) { 6916 return BAD_VALUE; 6917 } 6918 6919 ALOGV("suspendOutput() %d", output); 6920 thread->suspend(); 6921 6922 return NO_ERROR; 6923} 6924 6925status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6926{ 6927 Mutex::Autolock _l(mLock); 6928 PlaybackThread *thread = checkPlaybackThread_l(output); 6929 6930 if (thread == NULL) { 6931 return BAD_VALUE; 6932 } 6933 6934 ALOGV("restoreOutput() %d", output); 6935 6936 thread->restore(); 6937 6938 return NO_ERROR; 6939} 6940 6941audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6942 audio_devices_t *pDevices, 6943 uint32_t *pSamplingRate, 6944 audio_format_t *pFormat, 6945 uint32_t *pChannelMask) 6946{ 6947 status_t status; 6948 RecordThread *thread = NULL; 6949 struct audio_config config = { 6950 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6951 channel_mask: pChannelMask ? *pChannelMask : 0, 6952 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6953 }; 6954 uint32_t reqSamplingRate = config.sample_rate; 6955 audio_format_t reqFormat = config.format; 6956 audio_channel_mask_t reqChannels = config.channel_mask; 6957 audio_stream_in_t *inStream = NULL; 6958 audio_hw_device_t *inHwDev; 6959 6960 if (pDevices == NULL || *pDevices == 0) { 6961 return 0; 6962 } 6963 6964 Mutex::Autolock _l(mLock); 6965 6966 inHwDev = findSuitableHwDev_l(module, *pDevices); 6967 if (inHwDev == NULL) 6968 return 0; 6969 6970 audio_io_handle_t id = nextUniqueId(); 6971 6972 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6973 &inStream); 6974 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6975 inStream, 6976 config.sample_rate, 6977 config.format, 6978 config.channel_mask, 6979 status); 6980 6981 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6982 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6983 // or stereo to mono conversions on 16 bit PCM inputs. 6984 if (status == BAD_VALUE && 6985 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6986 (config.sample_rate <= 2 * reqSamplingRate) && 6987 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6988 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6989 inStream = NULL; 6990 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6991 } 6992 6993 if (status == NO_ERROR && inStream != NULL) { 6994 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6995 6996 // Start record thread 6997 // RecorThread require both input and output device indication to forward to audio 6998 // pre processing modules 6999 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 7000 thread = new RecordThread(this, 7001 input, 7002 reqSamplingRate, 7003 reqChannels, 7004 id, 7005 device); 7006 mRecordThreads.add(id, thread); 7007 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7008 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7009 if (pFormat != NULL) *pFormat = config.format; 7010 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7011 7012 input->stream->common.standby(&input->stream->common); 7013 7014 // notify client processes of the new input creation 7015 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7016 return id; 7017 } 7018 7019 return 0; 7020} 7021 7022status_t AudioFlinger::closeInput(audio_io_handle_t input) 7023{ 7024 // keep strong reference on the record thread so that 7025 // it is not destroyed while exit() is executed 7026 sp<RecordThread> thread; 7027 { 7028 Mutex::Autolock _l(mLock); 7029 thread = checkRecordThread_l(input); 7030 if (thread == NULL) { 7031 return BAD_VALUE; 7032 } 7033 7034 ALOGV("closeInput() %d", input); 7035 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7036 mRecordThreads.removeItem(input); 7037 } 7038 thread->exit(); 7039 // The thread entity (active unit of execution) is no longer running here, 7040 // but the ThreadBase container still exists. 7041 7042 AudioStreamIn *in = thread->clearInput(); 7043 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7044 // from now on thread->mInput is NULL 7045 in->hwDev->close_input_stream(in->hwDev, in->stream); 7046 delete in; 7047 7048 return NO_ERROR; 7049} 7050 7051status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7052{ 7053 Mutex::Autolock _l(mLock); 7054 MixerThread *dstThread = checkMixerThread_l(output); 7055 if (dstThread == NULL) { 7056 ALOGW("setStreamOutput() bad output id %d", output); 7057 return BAD_VALUE; 7058 } 7059 7060 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7061 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7062 7063 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7064 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7065 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7066 MixerThread *srcThread = (MixerThread *)thread; 7067 srcThread->invalidateTracks(stream); 7068 } 7069 } 7070 7071 return NO_ERROR; 7072} 7073 7074 7075int AudioFlinger::newAudioSessionId() 7076{ 7077 return nextUniqueId(); 7078} 7079 7080void AudioFlinger::acquireAudioSessionId(int audioSession) 7081{ 7082 Mutex::Autolock _l(mLock); 7083 pid_t caller = IPCThreadState::self()->getCallingPid(); 7084 ALOGV("acquiring %d from %d", audioSession, caller); 7085 size_t num = mAudioSessionRefs.size(); 7086 for (size_t i = 0; i< num; i++) { 7087 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7088 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7089 ref->mCnt++; 7090 ALOGV(" incremented refcount to %d", ref->mCnt); 7091 return; 7092 } 7093 } 7094 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7095 ALOGV(" added new entry for %d", audioSession); 7096} 7097 7098void AudioFlinger::releaseAudioSessionId(int audioSession) 7099{ 7100 Mutex::Autolock _l(mLock); 7101 pid_t caller = IPCThreadState::self()->getCallingPid(); 7102 ALOGV("releasing %d from %d", audioSession, caller); 7103 size_t num = mAudioSessionRefs.size(); 7104 for (size_t i = 0; i< num; i++) { 7105 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7106 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7107 ref->mCnt--; 7108 ALOGV(" decremented refcount to %d", ref->mCnt); 7109 if (ref->mCnt == 0) { 7110 mAudioSessionRefs.removeAt(i); 7111 delete ref; 7112 purgeStaleEffects_l(); 7113 } 7114 return; 7115 } 7116 } 7117 ALOGW("session id %d not found for pid %d", audioSession, caller); 7118} 7119 7120void AudioFlinger::purgeStaleEffects_l() { 7121 7122 ALOGV("purging stale effects"); 7123 7124 Vector< sp<EffectChain> > chains; 7125 7126 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7127 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7128 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7129 sp<EffectChain> ec = t->mEffectChains[j]; 7130 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7131 chains.push(ec); 7132 } 7133 } 7134 } 7135 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7136 sp<RecordThread> t = mRecordThreads.valueAt(i); 7137 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7138 sp<EffectChain> ec = t->mEffectChains[j]; 7139 chains.push(ec); 7140 } 7141 } 7142 7143 for (size_t i = 0; i < chains.size(); i++) { 7144 sp<EffectChain> ec = chains[i]; 7145 int sessionid = ec->sessionId(); 7146 sp<ThreadBase> t = ec->mThread.promote(); 7147 if (t == 0) { 7148 continue; 7149 } 7150 size_t numsessionrefs = mAudioSessionRefs.size(); 7151 bool found = false; 7152 for (size_t k = 0; k < numsessionrefs; k++) { 7153 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7154 if (ref->mSessionid == sessionid) { 7155 ALOGV(" session %d still exists for %d with %d refs", 7156 sessionid, ref->mPid, ref->mCnt); 7157 found = true; 7158 break; 7159 } 7160 } 7161 if (!found) { 7162 // remove all effects from the chain 7163 while (ec->mEffects.size()) { 7164 sp<EffectModule> effect = ec->mEffects[0]; 7165 effect->unPin(); 7166 Mutex::Autolock _l (t->mLock); 7167 t->removeEffect_l(effect); 7168 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7169 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7170 if (handle != 0) { 7171 handle->mEffect.clear(); 7172 if (handle->mHasControl && handle->mEnabled) { 7173 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7174 } 7175 } 7176 } 7177 AudioSystem::unregisterEffect(effect->id()); 7178 } 7179 } 7180 } 7181 return; 7182} 7183 7184// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7185AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7186{ 7187 return mPlaybackThreads.valueFor(output).get(); 7188} 7189 7190// checkMixerThread_l() must be called with AudioFlinger::mLock held 7191AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7192{ 7193 PlaybackThread *thread = checkPlaybackThread_l(output); 7194 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7195} 7196 7197// checkRecordThread_l() must be called with AudioFlinger::mLock held 7198AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7199{ 7200 return mRecordThreads.valueFor(input).get(); 7201} 7202 7203uint32_t AudioFlinger::nextUniqueId() 7204{ 7205 return android_atomic_inc(&mNextUniqueId); 7206} 7207 7208AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7209{ 7210 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7211 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7212 AudioStreamOut *output = thread->getOutput(); 7213 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7214 return thread; 7215 } 7216 } 7217 return NULL; 7218} 7219 7220uint32_t AudioFlinger::primaryOutputDevice_l() const 7221{ 7222 PlaybackThread *thread = primaryPlaybackThread_l(); 7223 7224 if (thread == NULL) { 7225 return 0; 7226 } 7227 7228 return thread->device(); 7229} 7230 7231sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7232 int triggerSession, 7233 int listenerSession, 7234 sync_event_callback_t callBack, 7235 void *cookie) 7236{ 7237 Mutex::Autolock _l(mLock); 7238 7239 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7240 status_t playStatus = NAME_NOT_FOUND; 7241 status_t recStatus = NAME_NOT_FOUND; 7242 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7243 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7244 if (playStatus == NO_ERROR) { 7245 return event; 7246 } 7247 } 7248 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7249 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7250 if (recStatus == NO_ERROR) { 7251 return event; 7252 } 7253 } 7254 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7255 mPendingSyncEvents.add(event); 7256 } else { 7257 ALOGV("createSyncEvent() invalid event %d", event->type()); 7258 event.clear(); 7259 } 7260 return event; 7261} 7262 7263// ---------------------------------------------------------------------------- 7264// Effect management 7265// ---------------------------------------------------------------------------- 7266 7267 7268status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7269{ 7270 Mutex::Autolock _l(mLock); 7271 return EffectQueryNumberEffects(numEffects); 7272} 7273 7274status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7275{ 7276 Mutex::Autolock _l(mLock); 7277 return EffectQueryEffect(index, descriptor); 7278} 7279 7280status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7281 effect_descriptor_t *descriptor) const 7282{ 7283 Mutex::Autolock _l(mLock); 7284 return EffectGetDescriptor(pUuid, descriptor); 7285} 7286 7287 7288sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7289 effect_descriptor_t *pDesc, 7290 const sp<IEffectClient>& effectClient, 7291 int32_t priority, 7292 audio_io_handle_t io, 7293 int sessionId, 7294 status_t *status, 7295 int *id, 7296 int *enabled) 7297{ 7298 status_t lStatus = NO_ERROR; 7299 sp<EffectHandle> handle; 7300 effect_descriptor_t desc; 7301 7302 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7303 pid, effectClient.get(), priority, sessionId, io); 7304 7305 if (pDesc == NULL) { 7306 lStatus = BAD_VALUE; 7307 goto Exit; 7308 } 7309 7310 // check audio settings permission for global effects 7311 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7312 lStatus = PERMISSION_DENIED; 7313 goto Exit; 7314 } 7315 7316 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7317 // that can only be created by audio policy manager (running in same process) 7318 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7319 lStatus = PERMISSION_DENIED; 7320 goto Exit; 7321 } 7322 7323 if (io == 0) { 7324 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7325 // output must be specified by AudioPolicyManager when using session 7326 // AUDIO_SESSION_OUTPUT_STAGE 7327 lStatus = BAD_VALUE; 7328 goto Exit; 7329 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7330 // if the output returned by getOutputForEffect() is removed before we lock the 7331 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7332 // and we will exit safely 7333 io = AudioSystem::getOutputForEffect(&desc); 7334 } 7335 } 7336 7337 { 7338 Mutex::Autolock _l(mLock); 7339 7340 7341 if (!EffectIsNullUuid(&pDesc->uuid)) { 7342 // if uuid is specified, request effect descriptor 7343 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7344 if (lStatus < 0) { 7345 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7346 goto Exit; 7347 } 7348 } else { 7349 // if uuid is not specified, look for an available implementation 7350 // of the required type in effect factory 7351 if (EffectIsNullUuid(&pDesc->type)) { 7352 ALOGW("createEffect() no effect type"); 7353 lStatus = BAD_VALUE; 7354 goto Exit; 7355 } 7356 uint32_t numEffects = 0; 7357 effect_descriptor_t d; 7358 d.flags = 0; // prevent compiler warning 7359 bool found = false; 7360 7361 lStatus = EffectQueryNumberEffects(&numEffects); 7362 if (lStatus < 0) { 7363 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7364 goto Exit; 7365 } 7366 for (uint32_t i = 0; i < numEffects; i++) { 7367 lStatus = EffectQueryEffect(i, &desc); 7368 if (lStatus < 0) { 7369 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7370 continue; 7371 } 7372 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7373 // If matching type found save effect descriptor. If the session is 7374 // 0 and the effect is not auxiliary, continue enumeration in case 7375 // an auxiliary version of this effect type is available 7376 found = true; 7377 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7378 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7379 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7380 break; 7381 } 7382 } 7383 } 7384 if (!found) { 7385 lStatus = BAD_VALUE; 7386 ALOGW("createEffect() effect not found"); 7387 goto Exit; 7388 } 7389 // For same effect type, chose auxiliary version over insert version if 7390 // connect to output mix (Compliance to OpenSL ES) 7391 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7392 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7393 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7394 } 7395 } 7396 7397 // Do not allow auxiliary effects on a session different from 0 (output mix) 7398 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7399 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7400 lStatus = INVALID_OPERATION; 7401 goto Exit; 7402 } 7403 7404 // check recording permission for visualizer 7405 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7406 !recordingAllowed()) { 7407 lStatus = PERMISSION_DENIED; 7408 goto Exit; 7409 } 7410 7411 // return effect descriptor 7412 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7413 7414 // If output is not specified try to find a matching audio session ID in one of the 7415 // output threads. 7416 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7417 // because of code checking output when entering the function. 7418 // Note: io is never 0 when creating an effect on an input 7419 if (io == 0) { 7420 // look for the thread where the specified audio session is present 7421 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7422 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7423 io = mPlaybackThreads.keyAt(i); 7424 break; 7425 } 7426 } 7427 if (io == 0) { 7428 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7429 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7430 io = mRecordThreads.keyAt(i); 7431 break; 7432 } 7433 } 7434 } 7435 // If no output thread contains the requested session ID, default to 7436 // first output. The effect chain will be moved to the correct output 7437 // thread when a track with the same session ID is created 7438 if (io == 0 && mPlaybackThreads.size()) { 7439 io = mPlaybackThreads.keyAt(0); 7440 } 7441 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7442 } 7443 ThreadBase *thread = checkRecordThread_l(io); 7444 if (thread == NULL) { 7445 thread = checkPlaybackThread_l(io); 7446 if (thread == NULL) { 7447 ALOGE("createEffect() unknown output thread"); 7448 lStatus = BAD_VALUE; 7449 goto Exit; 7450 } 7451 } 7452 7453 sp<Client> client = registerPid_l(pid); 7454 7455 // create effect on selected output thread 7456 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7457 &desc, enabled, &lStatus); 7458 if (handle != 0 && id != NULL) { 7459 *id = handle->id(); 7460 } 7461 } 7462 7463Exit: 7464 if (status != NULL) { 7465 *status = lStatus; 7466 } 7467 return handle; 7468} 7469 7470status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7471 audio_io_handle_t dstOutput) 7472{ 7473 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7474 sessionId, srcOutput, dstOutput); 7475 Mutex::Autolock _l(mLock); 7476 if (srcOutput == dstOutput) { 7477 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7478 return NO_ERROR; 7479 } 7480 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7481 if (srcThread == NULL) { 7482 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7483 return BAD_VALUE; 7484 } 7485 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7486 if (dstThread == NULL) { 7487 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7488 return BAD_VALUE; 7489 } 7490 7491 Mutex::Autolock _dl(dstThread->mLock); 7492 Mutex::Autolock _sl(srcThread->mLock); 7493 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7494 7495 return NO_ERROR; 7496} 7497 7498// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7499status_t AudioFlinger::moveEffectChain_l(int sessionId, 7500 AudioFlinger::PlaybackThread *srcThread, 7501 AudioFlinger::PlaybackThread *dstThread, 7502 bool reRegister) 7503{ 7504 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7505 sessionId, srcThread, dstThread); 7506 7507 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7508 if (chain == 0) { 7509 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7510 sessionId, srcThread); 7511 return INVALID_OPERATION; 7512 } 7513 7514 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7515 // so that a new chain is created with correct parameters when first effect is added. This is 7516 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7517 // removed. 7518 srcThread->removeEffectChain_l(chain); 7519 7520 // transfer all effects one by one so that new effect chain is created on new thread with 7521 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7522 audio_io_handle_t dstOutput = dstThread->id(); 7523 sp<EffectChain> dstChain; 7524 uint32_t strategy = 0; // prevent compiler warning 7525 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7526 while (effect != 0) { 7527 srcThread->removeEffect_l(effect); 7528 dstThread->addEffect_l(effect); 7529 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7530 if (effect->state() == EffectModule::ACTIVE || 7531 effect->state() == EffectModule::STOPPING) { 7532 effect->start(); 7533 } 7534 // if the move request is not received from audio policy manager, the effect must be 7535 // re-registered with the new strategy and output 7536 if (dstChain == 0) { 7537 dstChain = effect->chain().promote(); 7538 if (dstChain == 0) { 7539 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7540 srcThread->addEffect_l(effect); 7541 return NO_INIT; 7542 } 7543 strategy = dstChain->strategy(); 7544 } 7545 if (reRegister) { 7546 AudioSystem::unregisterEffect(effect->id()); 7547 AudioSystem::registerEffect(&effect->desc(), 7548 dstOutput, 7549 strategy, 7550 sessionId, 7551 effect->id()); 7552 } 7553 effect = chain->getEffectFromId_l(0); 7554 } 7555 7556 return NO_ERROR; 7557} 7558 7559 7560// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7561sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7562 const sp<AudioFlinger::Client>& client, 7563 const sp<IEffectClient>& effectClient, 7564 int32_t priority, 7565 int sessionId, 7566 effect_descriptor_t *desc, 7567 int *enabled, 7568 status_t *status 7569 ) 7570{ 7571 sp<EffectModule> effect; 7572 sp<EffectHandle> handle; 7573 status_t lStatus; 7574 sp<EffectChain> chain; 7575 bool chainCreated = false; 7576 bool effectCreated = false; 7577 bool effectRegistered = false; 7578 7579 lStatus = initCheck(); 7580 if (lStatus != NO_ERROR) { 7581 ALOGW("createEffect_l() Audio driver not initialized."); 7582 goto Exit; 7583 } 7584 7585 // Do not allow effects with session ID 0 on direct output or duplicating threads 7586 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7587 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7588 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7589 desc->name, sessionId); 7590 lStatus = BAD_VALUE; 7591 goto Exit; 7592 } 7593 // Only Pre processor effects are allowed on input threads and only on input threads 7594 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7595 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7596 desc->name, desc->flags, mType); 7597 lStatus = BAD_VALUE; 7598 goto Exit; 7599 } 7600 7601 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7602 7603 { // scope for mLock 7604 Mutex::Autolock _l(mLock); 7605 7606 // check for existing effect chain with the requested audio session 7607 chain = getEffectChain_l(sessionId); 7608 if (chain == 0) { 7609 // create a new chain for this session 7610 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7611 chain = new EffectChain(this, sessionId); 7612 addEffectChain_l(chain); 7613 chain->setStrategy(getStrategyForSession_l(sessionId)); 7614 chainCreated = true; 7615 } else { 7616 effect = chain->getEffectFromDesc_l(desc); 7617 } 7618 7619 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7620 7621 if (effect == 0) { 7622 int id = mAudioFlinger->nextUniqueId(); 7623 // Check CPU and memory usage 7624 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7625 if (lStatus != NO_ERROR) { 7626 goto Exit; 7627 } 7628 effectRegistered = true; 7629 // create a new effect module if none present in the chain 7630 effect = new EffectModule(this, chain, desc, id, sessionId); 7631 lStatus = effect->status(); 7632 if (lStatus != NO_ERROR) { 7633 goto Exit; 7634 } 7635 lStatus = chain->addEffect_l(effect); 7636 if (lStatus != NO_ERROR) { 7637 goto Exit; 7638 } 7639 effectCreated = true; 7640 7641 effect->setDevice(mDevice); 7642 effect->setMode(mAudioFlinger->getMode()); 7643 } 7644 // create effect handle and connect it to effect module 7645 handle = new EffectHandle(effect, client, effectClient, priority); 7646 lStatus = effect->addHandle(handle); 7647 if (enabled != NULL) { 7648 *enabled = (int)effect->isEnabled(); 7649 } 7650 } 7651 7652Exit: 7653 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7654 Mutex::Autolock _l(mLock); 7655 if (effectCreated) { 7656 chain->removeEffect_l(effect); 7657 } 7658 if (effectRegistered) { 7659 AudioSystem::unregisterEffect(effect->id()); 7660 } 7661 if (chainCreated) { 7662 removeEffectChain_l(chain); 7663 } 7664 handle.clear(); 7665 } 7666 7667 if (status != NULL) { 7668 *status = lStatus; 7669 } 7670 return handle; 7671} 7672 7673sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7674{ 7675 sp<EffectChain> chain = getEffectChain_l(sessionId); 7676 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7677} 7678 7679// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7680// PlaybackThread::mLock held 7681status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7682{ 7683 // check for existing effect chain with the requested audio session 7684 int sessionId = effect->sessionId(); 7685 sp<EffectChain> chain = getEffectChain_l(sessionId); 7686 bool chainCreated = false; 7687 7688 if (chain == 0) { 7689 // create a new chain for this session 7690 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7691 chain = new EffectChain(this, sessionId); 7692 addEffectChain_l(chain); 7693 chain->setStrategy(getStrategyForSession_l(sessionId)); 7694 chainCreated = true; 7695 } 7696 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7697 7698 if (chain->getEffectFromId_l(effect->id()) != 0) { 7699 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7700 this, effect->desc().name, chain.get()); 7701 return BAD_VALUE; 7702 } 7703 7704 status_t status = chain->addEffect_l(effect); 7705 if (status != NO_ERROR) { 7706 if (chainCreated) { 7707 removeEffectChain_l(chain); 7708 } 7709 return status; 7710 } 7711 7712 effect->setDevice(mDevice); 7713 effect->setMode(mAudioFlinger->getMode()); 7714 return NO_ERROR; 7715} 7716 7717void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7718 7719 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7720 effect_descriptor_t desc = effect->desc(); 7721 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7722 detachAuxEffect_l(effect->id()); 7723 } 7724 7725 sp<EffectChain> chain = effect->chain().promote(); 7726 if (chain != 0) { 7727 // remove effect chain if removing last effect 7728 if (chain->removeEffect_l(effect) == 0) { 7729 removeEffectChain_l(chain); 7730 } 7731 } else { 7732 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7733 } 7734} 7735 7736void AudioFlinger::ThreadBase::lockEffectChains_l( 7737 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7738{ 7739 effectChains = mEffectChains; 7740 for (size_t i = 0; i < mEffectChains.size(); i++) { 7741 mEffectChains[i]->lock(); 7742 } 7743} 7744 7745void AudioFlinger::ThreadBase::unlockEffectChains( 7746 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7747{ 7748 for (size_t i = 0; i < effectChains.size(); i++) { 7749 effectChains[i]->unlock(); 7750 } 7751} 7752 7753sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7754{ 7755 Mutex::Autolock _l(mLock); 7756 return getEffectChain_l(sessionId); 7757} 7758 7759sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7760{ 7761 size_t size = mEffectChains.size(); 7762 for (size_t i = 0; i < size; i++) { 7763 if (mEffectChains[i]->sessionId() == sessionId) { 7764 return mEffectChains[i]; 7765 } 7766 } 7767 return 0; 7768} 7769 7770void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7771{ 7772 Mutex::Autolock _l(mLock); 7773 size_t size = mEffectChains.size(); 7774 for (size_t i = 0; i < size; i++) { 7775 mEffectChains[i]->setMode_l(mode); 7776 } 7777} 7778 7779void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7780 const wp<EffectHandle>& handle, 7781 bool unpinIfLast) { 7782 7783 Mutex::Autolock _l(mLock); 7784 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7785 // delete the effect module if removing last handle on it 7786 if (effect->removeHandle(handle) == 0) { 7787 if (!effect->isPinned() || unpinIfLast) { 7788 removeEffect_l(effect); 7789 AudioSystem::unregisterEffect(effect->id()); 7790 } 7791 } 7792} 7793 7794status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7795{ 7796 int session = chain->sessionId(); 7797 int16_t *buffer = mMixBuffer; 7798 bool ownsBuffer = false; 7799 7800 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7801 if (session > 0) { 7802 // Only one effect chain can be present in direct output thread and it uses 7803 // the mix buffer as input 7804 if (mType != DIRECT) { 7805 size_t numSamples = mNormalFrameCount * mChannelCount; 7806 buffer = new int16_t[numSamples]; 7807 memset(buffer, 0, numSamples * sizeof(int16_t)); 7808 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7809 ownsBuffer = true; 7810 } 7811 7812 // Attach all tracks with same session ID to this chain. 7813 for (size_t i = 0; i < mTracks.size(); ++i) { 7814 sp<Track> track = mTracks[i]; 7815 if (session == track->sessionId()) { 7816 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7817 track->setMainBuffer(buffer); 7818 chain->incTrackCnt(); 7819 } 7820 } 7821 7822 // indicate all active tracks in the chain 7823 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7824 sp<Track> track = mActiveTracks[i].promote(); 7825 if (track == 0) continue; 7826 if (session == track->sessionId()) { 7827 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7828 chain->incActiveTrackCnt(); 7829 } 7830 } 7831 } 7832 7833 chain->setInBuffer(buffer, ownsBuffer); 7834 chain->setOutBuffer(mMixBuffer); 7835 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7836 // chains list in order to be processed last as it contains output stage effects 7837 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7838 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7839 // after track specific effects and before output stage 7840 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7841 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7842 // Effect chain for other sessions are inserted at beginning of effect 7843 // chains list to be processed before output mix effects. Relative order between other 7844 // sessions is not important 7845 size_t size = mEffectChains.size(); 7846 size_t i = 0; 7847 for (i = 0; i < size; i++) { 7848 if (mEffectChains[i]->sessionId() < session) break; 7849 } 7850 mEffectChains.insertAt(chain, i); 7851 checkSuspendOnAddEffectChain_l(chain); 7852 7853 return NO_ERROR; 7854} 7855 7856size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7857{ 7858 int session = chain->sessionId(); 7859 7860 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7861 7862 for (size_t i = 0; i < mEffectChains.size(); i++) { 7863 if (chain == mEffectChains[i]) { 7864 mEffectChains.removeAt(i); 7865 // detach all active tracks from the chain 7866 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7867 sp<Track> track = mActiveTracks[i].promote(); 7868 if (track == 0) continue; 7869 if (session == track->sessionId()) { 7870 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7871 chain.get(), session); 7872 chain->decActiveTrackCnt(); 7873 } 7874 } 7875 7876 // detach all tracks with same session ID from this chain 7877 for (size_t i = 0; i < mTracks.size(); ++i) { 7878 sp<Track> track = mTracks[i]; 7879 if (session == track->sessionId()) { 7880 track->setMainBuffer(mMixBuffer); 7881 chain->decTrackCnt(); 7882 } 7883 } 7884 break; 7885 } 7886 } 7887 return mEffectChains.size(); 7888} 7889 7890status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7891 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7892{ 7893 Mutex::Autolock _l(mLock); 7894 return attachAuxEffect_l(track, EffectId); 7895} 7896 7897status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7898 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7899{ 7900 status_t status = NO_ERROR; 7901 7902 if (EffectId == 0) { 7903 track->setAuxBuffer(0, NULL); 7904 } else { 7905 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7906 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7907 if (effect != 0) { 7908 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7909 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7910 } else { 7911 status = INVALID_OPERATION; 7912 } 7913 } else { 7914 status = BAD_VALUE; 7915 } 7916 } 7917 return status; 7918} 7919 7920void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7921{ 7922 for (size_t i = 0; i < mTracks.size(); ++i) { 7923 sp<Track> track = mTracks[i]; 7924 if (track->auxEffectId() == effectId) { 7925 attachAuxEffect_l(track, 0); 7926 } 7927 } 7928} 7929 7930status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7931{ 7932 // only one chain per input thread 7933 if (mEffectChains.size() != 0) { 7934 return INVALID_OPERATION; 7935 } 7936 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7937 7938 chain->setInBuffer(NULL); 7939 chain->setOutBuffer(NULL); 7940 7941 checkSuspendOnAddEffectChain_l(chain); 7942 7943 mEffectChains.add(chain); 7944 7945 return NO_ERROR; 7946} 7947 7948size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7949{ 7950 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7951 ALOGW_IF(mEffectChains.size() != 1, 7952 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7953 chain.get(), mEffectChains.size(), this); 7954 if (mEffectChains.size() == 1) { 7955 mEffectChains.removeAt(0); 7956 } 7957 return 0; 7958} 7959 7960// ---------------------------------------------------------------------------- 7961// EffectModule implementation 7962// ---------------------------------------------------------------------------- 7963 7964#undef LOG_TAG 7965#define LOG_TAG "AudioFlinger::EffectModule" 7966 7967AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7968 const wp<AudioFlinger::EffectChain>& chain, 7969 effect_descriptor_t *desc, 7970 int id, 7971 int sessionId) 7972 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7973 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7974{ 7975 ALOGV("Constructor %p", this); 7976 int lStatus; 7977 if (thread == NULL) { 7978 return; 7979 } 7980 7981 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7982 7983 // create effect engine from effect factory 7984 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7985 7986 if (mStatus != NO_ERROR) { 7987 return; 7988 } 7989 lStatus = init(); 7990 if (lStatus < 0) { 7991 mStatus = lStatus; 7992 goto Error; 7993 } 7994 7995 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7996 mPinned = true; 7997 } 7998 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7999 return; 8000Error: 8001 EffectRelease(mEffectInterface); 8002 mEffectInterface = NULL; 8003 ALOGV("Constructor Error %d", mStatus); 8004} 8005 8006AudioFlinger::EffectModule::~EffectModule() 8007{ 8008 ALOGV("Destructor %p", this); 8009 if (mEffectInterface != NULL) { 8010 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8011 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8012 sp<ThreadBase> thread = mThread.promote(); 8013 if (thread != 0) { 8014 audio_stream_t *stream = thread->stream(); 8015 if (stream != NULL) { 8016 stream->remove_audio_effect(stream, mEffectInterface); 8017 } 8018 } 8019 } 8020 // release effect engine 8021 EffectRelease(mEffectInterface); 8022 } 8023} 8024 8025status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8026{ 8027 status_t status; 8028 8029 Mutex::Autolock _l(mLock); 8030 int priority = handle->priority(); 8031 size_t size = mHandles.size(); 8032 sp<EffectHandle> h; 8033 size_t i; 8034 for (i = 0; i < size; i++) { 8035 h = mHandles[i].promote(); 8036 if (h == 0) continue; 8037 if (h->priority() <= priority) break; 8038 } 8039 // if inserted in first place, move effect control from previous owner to this handle 8040 if (i == 0) { 8041 bool enabled = false; 8042 if (h != 0) { 8043 enabled = h->enabled(); 8044 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8045 } 8046 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8047 status = NO_ERROR; 8048 } else { 8049 status = ALREADY_EXISTS; 8050 } 8051 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8052 mHandles.insertAt(handle, i); 8053 return status; 8054} 8055 8056size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8057{ 8058 Mutex::Autolock _l(mLock); 8059 size_t size = mHandles.size(); 8060 size_t i; 8061 for (i = 0; i < size; i++) { 8062 if (mHandles[i] == handle) break; 8063 } 8064 if (i == size) { 8065 return size; 8066 } 8067 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8068 8069 bool enabled = false; 8070 EffectHandle *hdl = handle.unsafe_get(); 8071 if (hdl != NULL) { 8072 ALOGV("removeHandle() unsafe_get OK"); 8073 enabled = hdl->enabled(); 8074 } 8075 mHandles.removeAt(i); 8076 size = mHandles.size(); 8077 // if removed from first place, move effect control from this handle to next in line 8078 if (i == 0 && size != 0) { 8079 sp<EffectHandle> h = mHandles[0].promote(); 8080 if (h != 0) { 8081 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8082 } 8083 } 8084 8085 // Prevent calls to process() and other functions on effect interface from now on. 8086 // The effect engine will be released by the destructor when the last strong reference on 8087 // this object is released which can happen after next process is called. 8088 if (size == 0 && !mPinned) { 8089 mState = DESTROYED; 8090 } 8091 8092 return size; 8093} 8094 8095sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8096{ 8097 Mutex::Autolock _l(mLock); 8098 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8099} 8100 8101void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8102{ 8103 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8104 // keep a strong reference on this EffectModule to avoid calling the 8105 // destructor before we exit 8106 sp<EffectModule> keep(this); 8107 { 8108 sp<ThreadBase> thread = mThread.promote(); 8109 if (thread != 0) { 8110 thread->disconnectEffect(keep, handle, unpinIfLast); 8111 } 8112 } 8113} 8114 8115void AudioFlinger::EffectModule::updateState() { 8116 Mutex::Autolock _l(mLock); 8117 8118 switch (mState) { 8119 case RESTART: 8120 reset_l(); 8121 // FALL THROUGH 8122 8123 case STARTING: 8124 // clear auxiliary effect input buffer for next accumulation 8125 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8126 memset(mConfig.inputCfg.buffer.raw, 8127 0, 8128 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8129 } 8130 start_l(); 8131 mState = ACTIVE; 8132 break; 8133 case STOPPING: 8134 stop_l(); 8135 mDisableWaitCnt = mMaxDisableWaitCnt; 8136 mState = STOPPED; 8137 break; 8138 case STOPPED: 8139 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8140 // turn off sequence. 8141 if (--mDisableWaitCnt == 0) { 8142 reset_l(); 8143 mState = IDLE; 8144 } 8145 break; 8146 default: //IDLE , ACTIVE, DESTROYED 8147 break; 8148 } 8149} 8150 8151void AudioFlinger::EffectModule::process() 8152{ 8153 Mutex::Autolock _l(mLock); 8154 8155 if (mState == DESTROYED || mEffectInterface == NULL || 8156 mConfig.inputCfg.buffer.raw == NULL || 8157 mConfig.outputCfg.buffer.raw == NULL) { 8158 return; 8159 } 8160 8161 if (isProcessEnabled()) { 8162 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8163 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8164 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8165 mConfig.inputCfg.buffer.s32, 8166 mConfig.inputCfg.buffer.frameCount/2); 8167 } 8168 8169 // do the actual processing in the effect engine 8170 int ret = (*mEffectInterface)->process(mEffectInterface, 8171 &mConfig.inputCfg.buffer, 8172 &mConfig.outputCfg.buffer); 8173 8174 // force transition to IDLE state when engine is ready 8175 if (mState == STOPPED && ret == -ENODATA) { 8176 mDisableWaitCnt = 1; 8177 } 8178 8179 // clear auxiliary effect input buffer for next accumulation 8180 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8181 memset(mConfig.inputCfg.buffer.raw, 0, 8182 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8183 } 8184 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8185 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8186 // If an insert effect is idle and input buffer is different from output buffer, 8187 // accumulate input onto output 8188 sp<EffectChain> chain = mChain.promote(); 8189 if (chain != 0 && chain->activeTrackCnt() != 0) { 8190 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8191 int16_t *in = mConfig.inputCfg.buffer.s16; 8192 int16_t *out = mConfig.outputCfg.buffer.s16; 8193 for (size_t i = 0; i < frameCnt; i++) { 8194 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8195 } 8196 } 8197 } 8198} 8199 8200void AudioFlinger::EffectModule::reset_l() 8201{ 8202 if (mEffectInterface == NULL) { 8203 return; 8204 } 8205 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8206} 8207 8208status_t AudioFlinger::EffectModule::configure() 8209{ 8210 uint32_t channels; 8211 if (mEffectInterface == NULL) { 8212 return NO_INIT; 8213 } 8214 8215 sp<ThreadBase> thread = mThread.promote(); 8216 if (thread == 0) { 8217 return DEAD_OBJECT; 8218 } 8219 8220 // TODO: handle configuration of effects replacing track process 8221 if (thread->channelCount() == 1) { 8222 channels = AUDIO_CHANNEL_OUT_MONO; 8223 } else { 8224 channels = AUDIO_CHANNEL_OUT_STEREO; 8225 } 8226 8227 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8228 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8229 } else { 8230 mConfig.inputCfg.channels = channels; 8231 } 8232 mConfig.outputCfg.channels = channels; 8233 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8234 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8235 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8236 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8237 mConfig.inputCfg.bufferProvider.cookie = NULL; 8238 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8239 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8240 mConfig.outputCfg.bufferProvider.cookie = NULL; 8241 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8242 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8243 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8244 // Insert effect: 8245 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8246 // always overwrites output buffer: input buffer == output buffer 8247 // - in other sessions: 8248 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8249 // other effect: overwrites output buffer: input buffer == output buffer 8250 // Auxiliary effect: 8251 // accumulates in output buffer: input buffer != output buffer 8252 // Therefore: accumulate <=> input buffer != output buffer 8253 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8254 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8255 } else { 8256 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8257 } 8258 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8259 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8260 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8261 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8262 8263 ALOGV("configure() %p thread %p buffer %p framecount %d", 8264 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8265 8266 status_t cmdStatus; 8267 uint32_t size = sizeof(int); 8268 status_t status = (*mEffectInterface)->command(mEffectInterface, 8269 EFFECT_CMD_SET_CONFIG, 8270 sizeof(effect_config_t), 8271 &mConfig, 8272 &size, 8273 &cmdStatus); 8274 if (status == 0) { 8275 status = cmdStatus; 8276 } 8277 8278 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8279 (1000 * mConfig.outputCfg.buffer.frameCount); 8280 8281 return status; 8282} 8283 8284status_t AudioFlinger::EffectModule::init() 8285{ 8286 Mutex::Autolock _l(mLock); 8287 if (mEffectInterface == NULL) { 8288 return NO_INIT; 8289 } 8290 status_t cmdStatus; 8291 uint32_t size = sizeof(status_t); 8292 status_t status = (*mEffectInterface)->command(mEffectInterface, 8293 EFFECT_CMD_INIT, 8294 0, 8295 NULL, 8296 &size, 8297 &cmdStatus); 8298 if (status == 0) { 8299 status = cmdStatus; 8300 } 8301 return status; 8302} 8303 8304status_t AudioFlinger::EffectModule::start() 8305{ 8306 Mutex::Autolock _l(mLock); 8307 return start_l(); 8308} 8309 8310status_t AudioFlinger::EffectModule::start_l() 8311{ 8312 if (mEffectInterface == NULL) { 8313 return NO_INIT; 8314 } 8315 status_t cmdStatus; 8316 uint32_t size = sizeof(status_t); 8317 status_t status = (*mEffectInterface)->command(mEffectInterface, 8318 EFFECT_CMD_ENABLE, 8319 0, 8320 NULL, 8321 &size, 8322 &cmdStatus); 8323 if (status == 0) { 8324 status = cmdStatus; 8325 } 8326 if (status == 0 && 8327 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8328 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8329 sp<ThreadBase> thread = mThread.promote(); 8330 if (thread != 0) { 8331 audio_stream_t *stream = thread->stream(); 8332 if (stream != NULL) { 8333 stream->add_audio_effect(stream, mEffectInterface); 8334 } 8335 } 8336 } 8337 return status; 8338} 8339 8340status_t AudioFlinger::EffectModule::stop() 8341{ 8342 Mutex::Autolock _l(mLock); 8343 return stop_l(); 8344} 8345 8346status_t AudioFlinger::EffectModule::stop_l() 8347{ 8348 if (mEffectInterface == NULL) { 8349 return NO_INIT; 8350 } 8351 status_t cmdStatus; 8352 uint32_t size = sizeof(status_t); 8353 status_t status = (*mEffectInterface)->command(mEffectInterface, 8354 EFFECT_CMD_DISABLE, 8355 0, 8356 NULL, 8357 &size, 8358 &cmdStatus); 8359 if (status == 0) { 8360 status = cmdStatus; 8361 } 8362 if (status == 0 && 8363 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8364 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8365 sp<ThreadBase> thread = mThread.promote(); 8366 if (thread != 0) { 8367 audio_stream_t *stream = thread->stream(); 8368 if (stream != NULL) { 8369 stream->remove_audio_effect(stream, mEffectInterface); 8370 } 8371 } 8372 } 8373 return status; 8374} 8375 8376status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8377 uint32_t cmdSize, 8378 void *pCmdData, 8379 uint32_t *replySize, 8380 void *pReplyData) 8381{ 8382 Mutex::Autolock _l(mLock); 8383// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8384 8385 if (mState == DESTROYED || mEffectInterface == NULL) { 8386 return NO_INIT; 8387 } 8388 status_t status = (*mEffectInterface)->command(mEffectInterface, 8389 cmdCode, 8390 cmdSize, 8391 pCmdData, 8392 replySize, 8393 pReplyData); 8394 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8395 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8396 for (size_t i = 1; i < mHandles.size(); i++) { 8397 sp<EffectHandle> h = mHandles[i].promote(); 8398 if (h != 0) { 8399 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8400 } 8401 } 8402 } 8403 return status; 8404} 8405 8406status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8407{ 8408 8409 Mutex::Autolock _l(mLock); 8410 ALOGV("setEnabled %p enabled %d", this, enabled); 8411 8412 if (enabled != isEnabled()) { 8413 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8414 if (enabled && status != NO_ERROR) { 8415 return status; 8416 } 8417 8418 switch (mState) { 8419 // going from disabled to enabled 8420 case IDLE: 8421 mState = STARTING; 8422 break; 8423 case STOPPED: 8424 mState = RESTART; 8425 break; 8426 case STOPPING: 8427 mState = ACTIVE; 8428 break; 8429 8430 // going from enabled to disabled 8431 case RESTART: 8432 mState = STOPPED; 8433 break; 8434 case STARTING: 8435 mState = IDLE; 8436 break; 8437 case ACTIVE: 8438 mState = STOPPING; 8439 break; 8440 case DESTROYED: 8441 return NO_ERROR; // simply ignore as we are being destroyed 8442 } 8443 for (size_t i = 1; i < mHandles.size(); i++) { 8444 sp<EffectHandle> h = mHandles[i].promote(); 8445 if (h != 0) { 8446 h->setEnabled(enabled); 8447 } 8448 } 8449 } 8450 return NO_ERROR; 8451} 8452 8453bool AudioFlinger::EffectModule::isEnabled() const 8454{ 8455 switch (mState) { 8456 case RESTART: 8457 case STARTING: 8458 case ACTIVE: 8459 return true; 8460 case IDLE: 8461 case STOPPING: 8462 case STOPPED: 8463 case DESTROYED: 8464 default: 8465 return false; 8466 } 8467} 8468 8469bool AudioFlinger::EffectModule::isProcessEnabled() const 8470{ 8471 switch (mState) { 8472 case RESTART: 8473 case ACTIVE: 8474 case STOPPING: 8475 case STOPPED: 8476 return true; 8477 case IDLE: 8478 case STARTING: 8479 case DESTROYED: 8480 default: 8481 return false; 8482 } 8483} 8484 8485status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8486{ 8487 Mutex::Autolock _l(mLock); 8488 status_t status = NO_ERROR; 8489 8490 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8491 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8492 if (isProcessEnabled() && 8493 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8494 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8495 status_t cmdStatus; 8496 uint32_t volume[2]; 8497 uint32_t *pVolume = NULL; 8498 uint32_t size = sizeof(volume); 8499 volume[0] = *left; 8500 volume[1] = *right; 8501 if (controller) { 8502 pVolume = volume; 8503 } 8504 status = (*mEffectInterface)->command(mEffectInterface, 8505 EFFECT_CMD_SET_VOLUME, 8506 size, 8507 volume, 8508 &size, 8509 pVolume); 8510 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8511 *left = volume[0]; 8512 *right = volume[1]; 8513 } 8514 } 8515 return status; 8516} 8517 8518status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8519{ 8520 Mutex::Autolock _l(mLock); 8521 status_t status = NO_ERROR; 8522 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8523 // audio pre processing modules on RecordThread can receive both output and 8524 // input device indication in the same call 8525 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8526 if (dev) { 8527 status_t cmdStatus; 8528 uint32_t size = sizeof(status_t); 8529 8530 status = (*mEffectInterface)->command(mEffectInterface, 8531 EFFECT_CMD_SET_DEVICE, 8532 sizeof(uint32_t), 8533 &dev, 8534 &size, 8535 &cmdStatus); 8536 if (status == NO_ERROR) { 8537 status = cmdStatus; 8538 } 8539 } 8540 dev = device & AUDIO_DEVICE_IN_ALL; 8541 if (dev) { 8542 status_t cmdStatus; 8543 uint32_t size = sizeof(status_t); 8544 8545 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8546 EFFECT_CMD_SET_INPUT_DEVICE, 8547 sizeof(uint32_t), 8548 &dev, 8549 &size, 8550 &cmdStatus); 8551 if (status2 == NO_ERROR) { 8552 status2 = cmdStatus; 8553 } 8554 if (status == NO_ERROR) { 8555 status = status2; 8556 } 8557 } 8558 } 8559 return status; 8560} 8561 8562status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8563{ 8564 Mutex::Autolock _l(mLock); 8565 status_t status = NO_ERROR; 8566 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8567 status_t cmdStatus; 8568 uint32_t size = sizeof(status_t); 8569 status = (*mEffectInterface)->command(mEffectInterface, 8570 EFFECT_CMD_SET_AUDIO_MODE, 8571 sizeof(audio_mode_t), 8572 &mode, 8573 &size, 8574 &cmdStatus); 8575 if (status == NO_ERROR) { 8576 status = cmdStatus; 8577 } 8578 } 8579 return status; 8580} 8581 8582void AudioFlinger::EffectModule::setSuspended(bool suspended) 8583{ 8584 Mutex::Autolock _l(mLock); 8585 mSuspended = suspended; 8586} 8587 8588bool AudioFlinger::EffectModule::suspended() const 8589{ 8590 Mutex::Autolock _l(mLock); 8591 return mSuspended; 8592} 8593 8594status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8595{ 8596 const size_t SIZE = 256; 8597 char buffer[SIZE]; 8598 String8 result; 8599 8600 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8601 result.append(buffer); 8602 8603 bool locked = tryLock(mLock); 8604 // failed to lock - AudioFlinger is probably deadlocked 8605 if (!locked) { 8606 result.append("\t\tCould not lock Fx mutex:\n"); 8607 } 8608 8609 result.append("\t\tSession Status State Engine:\n"); 8610 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8611 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8612 result.append(buffer); 8613 8614 result.append("\t\tDescriptor:\n"); 8615 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8616 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8617 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8618 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8619 result.append(buffer); 8620 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8621 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8622 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8623 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8624 result.append(buffer); 8625 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8626 mDescriptor.apiVersion, 8627 mDescriptor.flags); 8628 result.append(buffer); 8629 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8630 mDescriptor.name); 8631 result.append(buffer); 8632 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8633 mDescriptor.implementor); 8634 result.append(buffer); 8635 8636 result.append("\t\t- Input configuration:\n"); 8637 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8638 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8639 (uint32_t)mConfig.inputCfg.buffer.raw, 8640 mConfig.inputCfg.buffer.frameCount, 8641 mConfig.inputCfg.samplingRate, 8642 mConfig.inputCfg.channels, 8643 mConfig.inputCfg.format); 8644 result.append(buffer); 8645 8646 result.append("\t\t- Output configuration:\n"); 8647 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8648 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8649 (uint32_t)mConfig.outputCfg.buffer.raw, 8650 mConfig.outputCfg.buffer.frameCount, 8651 mConfig.outputCfg.samplingRate, 8652 mConfig.outputCfg.channels, 8653 mConfig.outputCfg.format); 8654 result.append(buffer); 8655 8656 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8657 result.append(buffer); 8658 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8659 for (size_t i = 0; i < mHandles.size(); ++i) { 8660 sp<EffectHandle> handle = mHandles[i].promote(); 8661 if (handle != 0) { 8662 handle->dump(buffer, SIZE); 8663 result.append(buffer); 8664 } 8665 } 8666 8667 result.append("\n"); 8668 8669 write(fd, result.string(), result.length()); 8670 8671 if (locked) { 8672 mLock.unlock(); 8673 } 8674 8675 return NO_ERROR; 8676} 8677 8678// ---------------------------------------------------------------------------- 8679// EffectHandle implementation 8680// ---------------------------------------------------------------------------- 8681 8682#undef LOG_TAG 8683#define LOG_TAG "AudioFlinger::EffectHandle" 8684 8685AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8686 const sp<AudioFlinger::Client>& client, 8687 const sp<IEffectClient>& effectClient, 8688 int32_t priority) 8689 : BnEffect(), 8690 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8691 mPriority(priority), mHasControl(false), mEnabled(false) 8692{ 8693 ALOGV("constructor %p", this); 8694 8695 if (client == 0) { 8696 return; 8697 } 8698 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8699 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8700 if (mCblkMemory != 0) { 8701 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8702 8703 if (mCblk != NULL) { 8704 new(mCblk) effect_param_cblk_t(); 8705 mBuffer = (uint8_t *)mCblk + bufOffset; 8706 } 8707 } else { 8708 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8709 return; 8710 } 8711} 8712 8713AudioFlinger::EffectHandle::~EffectHandle() 8714{ 8715 ALOGV("Destructor %p", this); 8716 disconnect(false); 8717 ALOGV("Destructor DONE %p", this); 8718} 8719 8720status_t AudioFlinger::EffectHandle::enable() 8721{ 8722 ALOGV("enable %p", this); 8723 if (!mHasControl) return INVALID_OPERATION; 8724 if (mEffect == 0) return DEAD_OBJECT; 8725 8726 if (mEnabled) { 8727 return NO_ERROR; 8728 } 8729 8730 mEnabled = true; 8731 8732 sp<ThreadBase> thread = mEffect->thread().promote(); 8733 if (thread != 0) { 8734 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8735 } 8736 8737 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8738 if (mEffect->suspended()) { 8739 return NO_ERROR; 8740 } 8741 8742 status_t status = mEffect->setEnabled(true); 8743 if (status != NO_ERROR) { 8744 if (thread != 0) { 8745 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8746 } 8747 mEnabled = false; 8748 } 8749 return status; 8750} 8751 8752status_t AudioFlinger::EffectHandle::disable() 8753{ 8754 ALOGV("disable %p", this); 8755 if (!mHasControl) return INVALID_OPERATION; 8756 if (mEffect == 0) return DEAD_OBJECT; 8757 8758 if (!mEnabled) { 8759 return NO_ERROR; 8760 } 8761 mEnabled = false; 8762 8763 if (mEffect->suspended()) { 8764 return NO_ERROR; 8765 } 8766 8767 status_t status = mEffect->setEnabled(false); 8768 8769 sp<ThreadBase> thread = mEffect->thread().promote(); 8770 if (thread != 0) { 8771 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8772 } 8773 8774 return status; 8775} 8776 8777void AudioFlinger::EffectHandle::disconnect() 8778{ 8779 disconnect(true); 8780} 8781 8782void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8783{ 8784 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8785 if (mEffect == 0) { 8786 return; 8787 } 8788 mEffect->disconnect(this, unpinIfLast); 8789 8790 if (mHasControl && mEnabled) { 8791 sp<ThreadBase> thread = mEffect->thread().promote(); 8792 if (thread != 0) { 8793 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8794 } 8795 } 8796 8797 // release sp on module => module destructor can be called now 8798 mEffect.clear(); 8799 if (mClient != 0) { 8800 if (mCblk != NULL) { 8801 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8802 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8803 } 8804 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8805 // Client destructor must run with AudioFlinger mutex locked 8806 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8807 mClient.clear(); 8808 } 8809} 8810 8811status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8812 uint32_t cmdSize, 8813 void *pCmdData, 8814 uint32_t *replySize, 8815 void *pReplyData) 8816{ 8817// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8818// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8819 8820 // only get parameter command is permitted for applications not controlling the effect 8821 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8822 return INVALID_OPERATION; 8823 } 8824 if (mEffect == 0) return DEAD_OBJECT; 8825 if (mClient == 0) return INVALID_OPERATION; 8826 8827 // handle commands that are not forwarded transparently to effect engine 8828 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8829 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8830 // no risk to block the whole media server process or mixer threads is we are stuck here 8831 Mutex::Autolock _l(mCblk->lock); 8832 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8833 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8834 mCblk->serverIndex = 0; 8835 mCblk->clientIndex = 0; 8836 return BAD_VALUE; 8837 } 8838 status_t status = NO_ERROR; 8839 while (mCblk->serverIndex < mCblk->clientIndex) { 8840 int reply; 8841 uint32_t rsize = sizeof(int); 8842 int *p = (int *)(mBuffer + mCblk->serverIndex); 8843 int size = *p++; 8844 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8845 ALOGW("command(): invalid parameter block size"); 8846 break; 8847 } 8848 effect_param_t *param = (effect_param_t *)p; 8849 if (param->psize == 0 || param->vsize == 0) { 8850 ALOGW("command(): null parameter or value size"); 8851 mCblk->serverIndex += size; 8852 continue; 8853 } 8854 uint32_t psize = sizeof(effect_param_t) + 8855 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8856 param->vsize; 8857 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8858 psize, 8859 p, 8860 &rsize, 8861 &reply); 8862 // stop at first error encountered 8863 if (ret != NO_ERROR) { 8864 status = ret; 8865 *(int *)pReplyData = reply; 8866 break; 8867 } else if (reply != NO_ERROR) { 8868 *(int *)pReplyData = reply; 8869 break; 8870 } 8871 mCblk->serverIndex += size; 8872 } 8873 mCblk->serverIndex = 0; 8874 mCblk->clientIndex = 0; 8875 return status; 8876 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8877 *(int *)pReplyData = NO_ERROR; 8878 return enable(); 8879 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8880 *(int *)pReplyData = NO_ERROR; 8881 return disable(); 8882 } 8883 8884 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8885} 8886 8887void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8888{ 8889 ALOGV("setControl %p control %d", this, hasControl); 8890 8891 mHasControl = hasControl; 8892 mEnabled = enabled; 8893 8894 if (signal && mEffectClient != 0) { 8895 mEffectClient->controlStatusChanged(hasControl); 8896 } 8897} 8898 8899void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8900 uint32_t cmdSize, 8901 void *pCmdData, 8902 uint32_t replySize, 8903 void *pReplyData) 8904{ 8905 if (mEffectClient != 0) { 8906 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8907 } 8908} 8909 8910 8911 8912void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8913{ 8914 if (mEffectClient != 0) { 8915 mEffectClient->enableStatusChanged(enabled); 8916 } 8917} 8918 8919status_t AudioFlinger::EffectHandle::onTransact( 8920 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8921{ 8922 return BnEffect::onTransact(code, data, reply, flags); 8923} 8924 8925 8926void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8927{ 8928 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8929 8930 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8931 (mClient == 0) ? getpid_cached : mClient->pid(), 8932 mPriority, 8933 mHasControl, 8934 !locked, 8935 mCblk ? mCblk->clientIndex : 0, 8936 mCblk ? mCblk->serverIndex : 0 8937 ); 8938 8939 if (locked) { 8940 mCblk->lock.unlock(); 8941 } 8942} 8943 8944#undef LOG_TAG 8945#define LOG_TAG "AudioFlinger::EffectChain" 8946 8947AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8948 int sessionId) 8949 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8950 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8951 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8952{ 8953 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8954 if (thread == NULL) { 8955 return; 8956 } 8957 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8958 thread->frameCount(); 8959} 8960 8961AudioFlinger::EffectChain::~EffectChain() 8962{ 8963 if (mOwnInBuffer) { 8964 delete mInBuffer; 8965 } 8966 8967} 8968 8969// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8970sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8971{ 8972 size_t size = mEffects.size(); 8973 8974 for (size_t i = 0; i < size; i++) { 8975 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8976 return mEffects[i]; 8977 } 8978 } 8979 return 0; 8980} 8981 8982// getEffectFromId_l() must be called with ThreadBase::mLock held 8983sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8984{ 8985 size_t size = mEffects.size(); 8986 8987 for (size_t i = 0; i < size; i++) { 8988 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8989 if (id == 0 || mEffects[i]->id() == id) { 8990 return mEffects[i]; 8991 } 8992 } 8993 return 0; 8994} 8995 8996// getEffectFromType_l() must be called with ThreadBase::mLock held 8997sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8998 const effect_uuid_t *type) 8999{ 9000 size_t size = mEffects.size(); 9001 9002 for (size_t i = 0; i < size; i++) { 9003 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9004 return mEffects[i]; 9005 } 9006 } 9007 return 0; 9008} 9009 9010void AudioFlinger::EffectChain::clearInputBuffer() 9011{ 9012 Mutex::Autolock _l(mLock); 9013 sp<ThreadBase> thread = mThread.promote(); 9014 if (thread == 0) { 9015 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9016 return; 9017 } 9018 clearInputBuffer_l(thread); 9019} 9020 9021// Must be called with EffectChain::mLock locked 9022void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9023{ 9024 size_t numSamples = thread->frameCount() * thread->channelCount(); 9025 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9026 9027} 9028 9029// Must be called with EffectChain::mLock locked 9030void AudioFlinger::EffectChain::process_l() 9031{ 9032 sp<ThreadBase> thread = mThread.promote(); 9033 if (thread == 0) { 9034 ALOGW("process_l(): cannot promote mixer thread"); 9035 return; 9036 } 9037 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9038 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9039 // always process effects unless no more tracks are on the session and the effect tail 9040 // has been rendered 9041 bool doProcess = true; 9042 if (!isGlobalSession) { 9043 bool tracksOnSession = (trackCnt() != 0); 9044 9045 if (!tracksOnSession && mTailBufferCount == 0) { 9046 doProcess = false; 9047 } 9048 9049 if (activeTrackCnt() == 0) { 9050 // if no track is active and the effect tail has not been rendered, 9051 // the input buffer must be cleared here as the mixer process will not do it 9052 if (tracksOnSession || mTailBufferCount > 0) { 9053 clearInputBuffer_l(thread); 9054 if (mTailBufferCount > 0) { 9055 mTailBufferCount--; 9056 } 9057 } 9058 } 9059 } 9060 9061 size_t size = mEffects.size(); 9062 if (doProcess) { 9063 for (size_t i = 0; i < size; i++) { 9064 mEffects[i]->process(); 9065 } 9066 } 9067 for (size_t i = 0; i < size; i++) { 9068 mEffects[i]->updateState(); 9069 } 9070} 9071 9072// addEffect_l() must be called with PlaybackThread::mLock held 9073status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9074{ 9075 effect_descriptor_t desc = effect->desc(); 9076 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9077 9078 Mutex::Autolock _l(mLock); 9079 effect->setChain(this); 9080 sp<ThreadBase> thread = mThread.promote(); 9081 if (thread == 0) { 9082 return NO_INIT; 9083 } 9084 effect->setThread(thread); 9085 9086 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9087 // Auxiliary effects are inserted at the beginning of mEffects vector as 9088 // they are processed first and accumulated in chain input buffer 9089 mEffects.insertAt(effect, 0); 9090 9091 // the input buffer for auxiliary effect contains mono samples in 9092 // 32 bit format. This is to avoid saturation in AudoMixer 9093 // accumulation stage. Saturation is done in EffectModule::process() before 9094 // calling the process in effect engine 9095 size_t numSamples = thread->frameCount(); 9096 int32_t *buffer = new int32_t[numSamples]; 9097 memset(buffer, 0, numSamples * sizeof(int32_t)); 9098 effect->setInBuffer((int16_t *)buffer); 9099 // auxiliary effects output samples to chain input buffer for further processing 9100 // by insert effects 9101 effect->setOutBuffer(mInBuffer); 9102 } else { 9103 // Insert effects are inserted at the end of mEffects vector as they are processed 9104 // after track and auxiliary effects. 9105 // Insert effect order as a function of indicated preference: 9106 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9107 // another effect is present 9108 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9109 // last effect claiming first position 9110 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9111 // first effect claiming last position 9112 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9113 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9114 // already present 9115 9116 size_t size = mEffects.size(); 9117 size_t idx_insert = size; 9118 ssize_t idx_insert_first = -1; 9119 ssize_t idx_insert_last = -1; 9120 9121 for (size_t i = 0; i < size; i++) { 9122 effect_descriptor_t d = mEffects[i]->desc(); 9123 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9124 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9125 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9126 // check invalid effect chaining combinations 9127 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9128 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9129 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9130 return INVALID_OPERATION; 9131 } 9132 // remember position of first insert effect and by default 9133 // select this as insert position for new effect 9134 if (idx_insert == size) { 9135 idx_insert = i; 9136 } 9137 // remember position of last insert effect claiming 9138 // first position 9139 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9140 idx_insert_first = i; 9141 } 9142 // remember position of first insert effect claiming 9143 // last position 9144 if (iPref == EFFECT_FLAG_INSERT_LAST && 9145 idx_insert_last == -1) { 9146 idx_insert_last = i; 9147 } 9148 } 9149 } 9150 9151 // modify idx_insert from first position if needed 9152 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9153 if (idx_insert_last != -1) { 9154 idx_insert = idx_insert_last; 9155 } else { 9156 idx_insert = size; 9157 } 9158 } else { 9159 if (idx_insert_first != -1) { 9160 idx_insert = idx_insert_first + 1; 9161 } 9162 } 9163 9164 // always read samples from chain input buffer 9165 effect->setInBuffer(mInBuffer); 9166 9167 // if last effect in the chain, output samples to chain 9168 // output buffer, otherwise to chain input buffer 9169 if (idx_insert == size) { 9170 if (idx_insert != 0) { 9171 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9172 mEffects[idx_insert-1]->configure(); 9173 } 9174 effect->setOutBuffer(mOutBuffer); 9175 } else { 9176 effect->setOutBuffer(mInBuffer); 9177 } 9178 mEffects.insertAt(effect, idx_insert); 9179 9180 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9181 } 9182 effect->configure(); 9183 return NO_ERROR; 9184} 9185 9186// removeEffect_l() must be called with PlaybackThread::mLock held 9187size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9188{ 9189 Mutex::Autolock _l(mLock); 9190 size_t size = mEffects.size(); 9191 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9192 9193 for (size_t i = 0; i < size; i++) { 9194 if (effect == mEffects[i]) { 9195 // calling stop here will remove pre-processing effect from the audio HAL. 9196 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9197 // the middle of a read from audio HAL 9198 if (mEffects[i]->state() == EffectModule::ACTIVE || 9199 mEffects[i]->state() == EffectModule::STOPPING) { 9200 mEffects[i]->stop(); 9201 } 9202 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9203 delete[] effect->inBuffer(); 9204 } else { 9205 if (i == size - 1 && i != 0) { 9206 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9207 mEffects[i - 1]->configure(); 9208 } 9209 } 9210 mEffects.removeAt(i); 9211 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9212 break; 9213 } 9214 } 9215 9216 return mEffects.size(); 9217} 9218 9219// setDevice_l() must be called with PlaybackThread::mLock held 9220void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9221{ 9222 size_t size = mEffects.size(); 9223 for (size_t i = 0; i < size; i++) { 9224 mEffects[i]->setDevice(device); 9225 } 9226} 9227 9228// setMode_l() must be called with PlaybackThread::mLock held 9229void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9230{ 9231 size_t size = mEffects.size(); 9232 for (size_t i = 0; i < size; i++) { 9233 mEffects[i]->setMode(mode); 9234 } 9235} 9236 9237// setVolume_l() must be called with PlaybackThread::mLock held 9238bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9239{ 9240 uint32_t newLeft = *left; 9241 uint32_t newRight = *right; 9242 bool hasControl = false; 9243 int ctrlIdx = -1; 9244 size_t size = mEffects.size(); 9245 9246 // first update volume controller 9247 for (size_t i = size; i > 0; i--) { 9248 if (mEffects[i - 1]->isProcessEnabled() && 9249 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9250 ctrlIdx = i - 1; 9251 hasControl = true; 9252 break; 9253 } 9254 } 9255 9256 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9257 if (hasControl) { 9258 *left = mNewLeftVolume; 9259 *right = mNewRightVolume; 9260 } 9261 return hasControl; 9262 } 9263 9264 mVolumeCtrlIdx = ctrlIdx; 9265 mLeftVolume = newLeft; 9266 mRightVolume = newRight; 9267 9268 // second get volume update from volume controller 9269 if (ctrlIdx >= 0) { 9270 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9271 mNewLeftVolume = newLeft; 9272 mNewRightVolume = newRight; 9273 } 9274 // then indicate volume to all other effects in chain. 9275 // Pass altered volume to effects before volume controller 9276 // and requested volume to effects after controller 9277 uint32_t lVol = newLeft; 9278 uint32_t rVol = newRight; 9279 9280 for (size_t i = 0; i < size; i++) { 9281 if ((int)i == ctrlIdx) continue; 9282 // this also works for ctrlIdx == -1 when there is no volume controller 9283 if ((int)i > ctrlIdx) { 9284 lVol = *left; 9285 rVol = *right; 9286 } 9287 mEffects[i]->setVolume(&lVol, &rVol, false); 9288 } 9289 *left = newLeft; 9290 *right = newRight; 9291 9292 return hasControl; 9293} 9294 9295status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9296{ 9297 const size_t SIZE = 256; 9298 char buffer[SIZE]; 9299 String8 result; 9300 9301 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9302 result.append(buffer); 9303 9304 bool locked = tryLock(mLock); 9305 // failed to lock - AudioFlinger is probably deadlocked 9306 if (!locked) { 9307 result.append("\tCould not lock mutex:\n"); 9308 } 9309 9310 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9311 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9312 mEffects.size(), 9313 (uint32_t)mInBuffer, 9314 (uint32_t)mOutBuffer, 9315 mActiveTrackCnt); 9316 result.append(buffer); 9317 write(fd, result.string(), result.size()); 9318 9319 for (size_t i = 0; i < mEffects.size(); ++i) { 9320 sp<EffectModule> effect = mEffects[i]; 9321 if (effect != 0) { 9322 effect->dump(fd, args); 9323 } 9324 } 9325 9326 if (locked) { 9327 mLock.unlock(); 9328 } 9329 9330 return NO_ERROR; 9331} 9332 9333// must be called with ThreadBase::mLock held 9334void AudioFlinger::EffectChain::setEffectSuspended_l( 9335 const effect_uuid_t *type, bool suspend) 9336{ 9337 sp<SuspendedEffectDesc> desc; 9338 // use effect type UUID timelow as key as there is no real risk of identical 9339 // timeLow fields among effect type UUIDs. 9340 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9341 if (suspend) { 9342 if (index >= 0) { 9343 desc = mSuspendedEffects.valueAt(index); 9344 } else { 9345 desc = new SuspendedEffectDesc(); 9346 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9347 mSuspendedEffects.add(type->timeLow, desc); 9348 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9349 } 9350 if (desc->mRefCount++ == 0) { 9351 sp<EffectModule> effect = getEffectIfEnabled(type); 9352 if (effect != 0) { 9353 desc->mEffect = effect; 9354 effect->setSuspended(true); 9355 effect->setEnabled(false); 9356 } 9357 } 9358 } else { 9359 if (index < 0) { 9360 return; 9361 } 9362 desc = mSuspendedEffects.valueAt(index); 9363 if (desc->mRefCount <= 0) { 9364 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9365 desc->mRefCount = 1; 9366 } 9367 if (--desc->mRefCount == 0) { 9368 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9369 if (desc->mEffect != 0) { 9370 sp<EffectModule> effect = desc->mEffect.promote(); 9371 if (effect != 0) { 9372 effect->setSuspended(false); 9373 sp<EffectHandle> handle = effect->controlHandle(); 9374 if (handle != 0) { 9375 effect->setEnabled(handle->enabled()); 9376 } 9377 } 9378 desc->mEffect.clear(); 9379 } 9380 mSuspendedEffects.removeItemsAt(index); 9381 } 9382 } 9383} 9384 9385// must be called with ThreadBase::mLock held 9386void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9387{ 9388 sp<SuspendedEffectDesc> desc; 9389 9390 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9391 if (suspend) { 9392 if (index >= 0) { 9393 desc = mSuspendedEffects.valueAt(index); 9394 } else { 9395 desc = new SuspendedEffectDesc(); 9396 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9397 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9398 } 9399 if (desc->mRefCount++ == 0) { 9400 Vector< sp<EffectModule> > effects; 9401 getSuspendEligibleEffects(effects); 9402 for (size_t i = 0; i < effects.size(); i++) { 9403 setEffectSuspended_l(&effects[i]->desc().type, true); 9404 } 9405 } 9406 } else { 9407 if (index < 0) { 9408 return; 9409 } 9410 desc = mSuspendedEffects.valueAt(index); 9411 if (desc->mRefCount <= 0) { 9412 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9413 desc->mRefCount = 1; 9414 } 9415 if (--desc->mRefCount == 0) { 9416 Vector<const effect_uuid_t *> types; 9417 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9418 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9419 continue; 9420 } 9421 types.add(&mSuspendedEffects.valueAt(i)->mType); 9422 } 9423 for (size_t i = 0; i < types.size(); i++) { 9424 setEffectSuspended_l(types[i], false); 9425 } 9426 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9427 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9428 } 9429 } 9430} 9431 9432 9433// The volume effect is used for automated tests only 9434#ifndef OPENSL_ES_H_ 9435static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9436 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9437const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9438#endif //OPENSL_ES_H_ 9439 9440bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9441{ 9442 // auxiliary effects and visualizer are never suspended on output mix 9443 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9444 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9445 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9446 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9447 return false; 9448 } 9449 return true; 9450} 9451 9452void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9453{ 9454 effects.clear(); 9455 for (size_t i = 0; i < mEffects.size(); i++) { 9456 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9457 effects.add(mEffects[i]); 9458 } 9459 } 9460} 9461 9462sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9463 const effect_uuid_t *type) 9464{ 9465 sp<EffectModule> effect = getEffectFromType_l(type); 9466 return effect != 0 && effect->isEnabled() ? effect : 0; 9467} 9468 9469void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9470 bool enabled) 9471{ 9472 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9473 if (enabled) { 9474 if (index < 0) { 9475 // if the effect is not suspend check if all effects are suspended 9476 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9477 if (index < 0) { 9478 return; 9479 } 9480 if (!isEffectEligibleForSuspend(effect->desc())) { 9481 return; 9482 } 9483 setEffectSuspended_l(&effect->desc().type, enabled); 9484 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9485 if (index < 0) { 9486 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9487 return; 9488 } 9489 } 9490 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9491 effect->desc().type.timeLow); 9492 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9493 // if effect is requested to suspended but was not yet enabled, supend it now. 9494 if (desc->mEffect == 0) { 9495 desc->mEffect = effect; 9496 effect->setEnabled(false); 9497 effect->setSuspended(true); 9498 } 9499 } else { 9500 if (index < 0) { 9501 return; 9502 } 9503 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9504 effect->desc().type.timeLow); 9505 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9506 desc->mEffect.clear(); 9507 effect->setSuspended(false); 9508 } 9509} 9510 9511#undef LOG_TAG 9512#define LOG_TAG "AudioFlinger" 9513 9514// ---------------------------------------------------------------------------- 9515 9516status_t AudioFlinger::onTransact( 9517 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9518{ 9519 return BnAudioFlinger::onTransact(code, data, reply, flags); 9520} 9521 9522}; // namespace android 9523