AudioFlinger.cpp revision 5385b7b0f5d922ee38f8a54f11ee4462ef4b5e29
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 FastMixerState *state = sq->begin(); 2259 FastTrack *fastTrack = &state->mFastTracks[0]; 2260 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2261 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2262 fastTrack->mVolumeProvider = NULL; 2263 fastTrack->mGeneration++; 2264 state->mFastTracksGen++; 2265 state->mTrackMask = 1; 2266 // fast mixer will use the HAL output sink 2267 state->mOutputSink = mOutputSink.get(); 2268 state->mOutputSinkGen++; 2269 state->mFrameCount = mFrameCount; 2270 state->mCommand = FastMixerState::COLD_IDLE; 2271 // already done in constructor initialization list 2272 //mFastMixerFutex = 0; 2273 state->mColdFutexAddr = &mFastMixerFutex; 2274 state->mColdGen++; 2275 state->mDumpState = &mFastMixerDumpState; 2276 state->mTeeSink = mTeeSink.get(); 2277 sq->end(); 2278 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2279 2280 // start the fast mixer 2281 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2282#ifdef HAVE_REQUEST_PRIORITY 2283 pid_t tid = mFastMixer->getTid(); 2284 int err = requestPriority(getpid_cached, tid, 2); 2285 if (err != 0) { 2286 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2287 2, getpid_cached, tid, err); 2288 } 2289#endif 2290 2291 } else { 2292 mFastMixer = NULL; 2293 } 2294 2295 switch (kUseFastMixer) { 2296 case FastMixer_Never: 2297 case FastMixer_Dynamic: 2298 mNormalSink = mOutputSink; 2299 break; 2300 case FastMixer_Always: 2301 mNormalSink = mPipeSink; 2302 break; 2303 case FastMixer_Static: 2304 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2305 break; 2306 } 2307} 2308 2309AudioFlinger::MixerThread::~MixerThread() 2310{ 2311 if (mFastMixer != NULL) { 2312 FastMixerStateQueue *sq = mFastMixer->sq(); 2313 FastMixerState *state = sq->begin(); 2314 if (state->mCommand == FastMixerState::COLD_IDLE) { 2315 int32_t old = android_atomic_inc(&mFastMixerFutex); 2316 if (old == -1) { 2317 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2318 } 2319 } 2320 state->mCommand = FastMixerState::EXIT; 2321 sq->end(); 2322 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2323 mFastMixer->join(); 2324 // Though the fast mixer thread has exited, it's state queue is still valid. 2325 // We'll use that extract the final state which contains one remaining fast track 2326 // corresponding to our sub-mix. 2327 state = sq->begin(); 2328 ALOG_ASSERT(state->mTrackMask == 1); 2329 FastTrack *fastTrack = &state->mFastTracks[0]; 2330 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2331 delete fastTrack->mBufferProvider; 2332 sq->end(false /*didModify*/); 2333 delete mFastMixer; 2334#ifdef SOAKER 2335 if (mSoaker != NULL) { 2336 mSoaker->requestExitAndWait(); 2337 } 2338 delete mSoaker; 2339#endif 2340 } 2341 delete mAudioMixer; 2342} 2343 2344class CpuStats { 2345public: 2346 CpuStats(); 2347 void sample(const String8 &title); 2348#ifdef DEBUG_CPU_USAGE 2349private: 2350 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2351 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2352 2353 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2354 2355 int mCpuNum; // thread's current CPU number 2356 int mCpukHz; // frequency of thread's current CPU in kHz 2357#endif 2358}; 2359 2360CpuStats::CpuStats() 2361#ifdef DEBUG_CPU_USAGE 2362 : mCpuNum(-1), mCpukHz(-1) 2363#endif 2364{ 2365} 2366 2367void CpuStats::sample(const String8 &title) { 2368#ifdef DEBUG_CPU_USAGE 2369 // get current thread's delta CPU time in wall clock ns 2370 double wcNs; 2371 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2372 2373 // record sample for wall clock statistics 2374 if (valid) { 2375 mWcStats.sample(wcNs); 2376 } 2377 2378 // get the current CPU number 2379 int cpuNum = sched_getcpu(); 2380 2381 // get the current CPU frequency in kHz 2382 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2383 2384 // check if either CPU number or frequency changed 2385 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2386 mCpuNum = cpuNum; 2387 mCpukHz = cpukHz; 2388 // ignore sample for purposes of cycles 2389 valid = false; 2390 } 2391 2392 // if no change in CPU number or frequency, then record sample for cycle statistics 2393 if (valid && mCpukHz > 0) { 2394 double cycles = wcNs * cpukHz * 0.000001; 2395 mHzStats.sample(cycles); 2396 } 2397 2398 unsigned n = mWcStats.n(); 2399 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2400 if ((n & 127) == 1) { 2401 long long elapsed = mCpuUsage.elapsed(); 2402 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2403 double perLoop = elapsed / (double) n; 2404 double perLoop100 = perLoop * 0.01; 2405 double perLoop1k = perLoop * 0.001; 2406 double mean = mWcStats.mean(); 2407 double stddev = mWcStats.stddev(); 2408 double minimum = mWcStats.minimum(); 2409 double maximum = mWcStats.maximum(); 2410 double meanCycles = mHzStats.mean(); 2411 double stddevCycles = mHzStats.stddev(); 2412 double minCycles = mHzStats.minimum(); 2413 double maxCycles = mHzStats.maximum(); 2414 mCpuUsage.resetElapsed(); 2415 mWcStats.reset(); 2416 mHzStats.reset(); 2417 ALOGD("CPU usage for %s over past %.1f secs\n" 2418 " (%u mixer loops at %.1f mean ms per loop):\n" 2419 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2420 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2421 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2422 title.string(), 2423 elapsed * .000000001, n, perLoop * .000001, 2424 mean * .001, 2425 stddev * .001, 2426 minimum * .001, 2427 maximum * .001, 2428 mean / perLoop100, 2429 stddev / perLoop100, 2430 minimum / perLoop100, 2431 maximum / perLoop100, 2432 meanCycles / perLoop1k, 2433 stddevCycles / perLoop1k, 2434 minCycles / perLoop1k, 2435 maxCycles / perLoop1k); 2436 2437 } 2438 } 2439#endif 2440}; 2441 2442void AudioFlinger::PlaybackThread::checkSilentMode_l() 2443{ 2444 if (!mMasterMute) { 2445 char value[PROPERTY_VALUE_MAX]; 2446 if (property_get("ro.audio.silent", value, "0") > 0) { 2447 char *endptr; 2448 unsigned long ul = strtoul(value, &endptr, 0); 2449 if (*endptr == '\0' && ul != 0) { 2450 ALOGD("Silence is golden"); 2451 // The setprop command will not allow a property to be changed after 2452 // the first time it is set, so we don't have to worry about un-muting. 2453 setMasterMute_l(true); 2454 } 2455 } 2456 } 2457} 2458 2459bool AudioFlinger::PlaybackThread::threadLoop() 2460{ 2461 Vector< sp<Track> > tracksToRemove; 2462 2463 standbyTime = systemTime(); 2464 2465 // MIXER 2466 nsecs_t lastWarning = 0; 2467if (mType == MIXER) { 2468 longStandbyExit = false; 2469} 2470 2471 // DUPLICATING 2472 // FIXME could this be made local to while loop? 2473 writeFrames = 0; 2474 2475 cacheParameters_l(); 2476 sleepTime = idleSleepTime; 2477 2478if (mType == MIXER) { 2479 sleepTimeShift = 0; 2480} 2481 2482 CpuStats cpuStats; 2483 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2484 2485 acquireWakeLock(); 2486 2487 while (!exitPending()) 2488 { 2489 cpuStats.sample(myName); 2490 2491 Vector< sp<EffectChain> > effectChains; 2492 2493 processConfigEvents(); 2494 2495 { // scope for mLock 2496 2497 Mutex::Autolock _l(mLock); 2498 2499 if (checkForNewParameters_l()) { 2500 cacheParameters_l(); 2501 } 2502 2503 saveOutputTracks(); 2504 2505 // put audio hardware into standby after short delay 2506 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2507 mSuspended > 0)) { 2508 if (!mStandby) { 2509 2510 threadLoop_standby(); 2511 2512 mStandby = true; 2513 mBytesWritten = 0; 2514 } 2515 2516 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2517 // we're about to wait, flush the binder command buffer 2518 IPCThreadState::self()->flushCommands(); 2519 2520 clearOutputTracks(); 2521 2522 if (exitPending()) break; 2523 2524 releaseWakeLock_l(); 2525 // wait until we have something to do... 2526 ALOGV("%s going to sleep", myName.string()); 2527 mWaitWorkCV.wait(mLock); 2528 ALOGV("%s waking up", myName.string()); 2529 acquireWakeLock_l(); 2530 2531 mMixerStatus = MIXER_IDLE; 2532 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2533 2534 checkSilentMode_l(); 2535 2536 standbyTime = systemTime() + standbyDelay; 2537 sleepTime = idleSleepTime; 2538 if (mType == MIXER) { 2539 sleepTimeShift = 0; 2540 } 2541 2542 continue; 2543 } 2544 } 2545 2546 // mMixerStatusIgnoringFastTracks is also updated internally 2547 mMixerStatus = prepareTracks_l(&tracksToRemove); 2548 2549 // prevent any changes in effect chain list and in each effect chain 2550 // during mixing and effect process as the audio buffers could be deleted 2551 // or modified if an effect is created or deleted 2552 lockEffectChains_l(effectChains); 2553 } 2554 2555 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2556 threadLoop_mix(); 2557 } else { 2558 threadLoop_sleepTime(); 2559 } 2560 2561 if (mSuspended > 0) { 2562 sleepTime = suspendSleepTimeUs(); 2563 } 2564 2565 // only process effects if we're going to write 2566 if (sleepTime == 0) { 2567 for (size_t i = 0; i < effectChains.size(); i ++) { 2568 effectChains[i]->process_l(); 2569 } 2570 } 2571 2572 // enable changes in effect chain 2573 unlockEffectChains(effectChains); 2574 2575 // sleepTime == 0 means we must write to audio hardware 2576 if (sleepTime == 0) { 2577 2578 threadLoop_write(); 2579 2580if (mType == MIXER) { 2581 // write blocked detection 2582 nsecs_t now = systemTime(); 2583 nsecs_t delta = now - mLastWriteTime; 2584 if (!mStandby && delta > maxPeriod) { 2585 mNumDelayedWrites++; 2586 if ((now - lastWarning) > kWarningThrottleNs) { 2587#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2588 ScopedTrace st(ATRACE_TAG, "underrun"); 2589#endif 2590 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2591 ns2ms(delta), mNumDelayedWrites, this); 2592 lastWarning = now; 2593 } 2594 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2595 // a different threshold. Or completely removed for what it is worth anyway... 2596 if (mStandby) { 2597 longStandbyExit = true; 2598 } 2599 } 2600} 2601 2602 mStandby = false; 2603 } else { 2604 usleep(sleepTime); 2605 } 2606 2607 // Finally let go of removed track(s), without the lock held 2608 // since we can't guarantee the destructors won't acquire that 2609 // same lock. This will also mutate and push a new fast mixer state. 2610 threadLoop_removeTracks(tracksToRemove); 2611 tracksToRemove.clear(); 2612 2613 // FIXME I don't understand the need for this here; 2614 // it was in the original code but maybe the 2615 // assignment in saveOutputTracks() makes this unnecessary? 2616 clearOutputTracks(); 2617 2618 // Effect chains will be actually deleted here if they were removed from 2619 // mEffectChains list during mixing or effects processing 2620 effectChains.clear(); 2621 2622 // FIXME Note that the above .clear() is no longer necessary since effectChains 2623 // is now local to this block, but will keep it for now (at least until merge done). 2624 } 2625 2626if (mType == MIXER || mType == DIRECT) { 2627 // put output stream into standby mode 2628 if (!mStandby) { 2629 mOutput->stream->common.standby(&mOutput->stream->common); 2630 } 2631} 2632if (mType == DUPLICATING) { 2633 // for DuplicatingThread, standby mode is handled by the outputTracks 2634} 2635 2636 releaseWakeLock(); 2637 2638 ALOGV("Thread %p type %d exiting", this, mType); 2639 return false; 2640} 2641 2642void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2643{ 2644 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2645} 2646 2647void AudioFlinger::MixerThread::threadLoop_write() 2648{ 2649 // FIXME we should only do one push per cycle; confirm this is true 2650 // Start the fast mixer if it's not already running 2651 if (mFastMixer != NULL) { 2652 FastMixerStateQueue *sq = mFastMixer->sq(); 2653 FastMixerState *state = sq->begin(); 2654 if (state->mCommand != FastMixerState::MIX_WRITE && 2655 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2656 if (state->mCommand == FastMixerState::COLD_IDLE) { 2657 int32_t old = android_atomic_inc(&mFastMixerFutex); 2658 if (old == -1) { 2659 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2660 } 2661 } 2662 state->mCommand = FastMixerState::MIX_WRITE; 2663 sq->end(); 2664 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2665 if (kUseFastMixer == FastMixer_Dynamic) { 2666 mNormalSink = mPipeSink; 2667 } 2668 } else { 2669 sq->end(false /*didModify*/); 2670 } 2671 } 2672 PlaybackThread::threadLoop_write(); 2673} 2674 2675// shared by MIXER and DIRECT, overridden by DUPLICATING 2676void AudioFlinger::PlaybackThread::threadLoop_write() 2677{ 2678 // FIXME rewrite to reduce number of system calls 2679 mLastWriteTime = systemTime(); 2680 mInWrite = true; 2681 2682#define mBitShift 2 // FIXME 2683 size_t count = mixBufferSize >> mBitShift; 2684#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2685 Tracer::traceBegin(ATRACE_TAG, "write"); 2686#endif 2687 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2688#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2689 Tracer::traceEnd(ATRACE_TAG); 2690#endif 2691 if (framesWritten > 0) { 2692 size_t bytesWritten = framesWritten << mBitShift; 2693 mBytesWritten += bytesWritten; 2694 } 2695 2696 mNumWrites++; 2697 mInWrite = false; 2698} 2699 2700void AudioFlinger::MixerThread::threadLoop_standby() 2701{ 2702 // Idle the fast mixer if it's currently running 2703 if (mFastMixer != NULL) { 2704 FastMixerStateQueue *sq = mFastMixer->sq(); 2705 FastMixerState *state = sq->begin(); 2706 if (!(state->mCommand & FastMixerState::IDLE)) { 2707 state->mCommand = FastMixerState::COLD_IDLE; 2708 state->mColdFutexAddr = &mFastMixerFutex; 2709 state->mColdGen++; 2710 mFastMixerFutex = 0; 2711 sq->end(); 2712 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2713 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2714 if (kUseFastMixer == FastMixer_Dynamic) { 2715 mNormalSink = mOutputSink; 2716 } 2717 } else { 2718 sq->end(false /*didModify*/); 2719 } 2720 } 2721 PlaybackThread::threadLoop_standby(); 2722} 2723 2724// shared by MIXER and DIRECT, overridden by DUPLICATING 2725void AudioFlinger::PlaybackThread::threadLoop_standby() 2726{ 2727 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2728 mOutput->stream->common.standby(&mOutput->stream->common); 2729} 2730 2731void AudioFlinger::MixerThread::threadLoop_mix() 2732{ 2733 // obtain the presentation timestamp of the next output buffer 2734 int64_t pts; 2735 status_t status = INVALID_OPERATION; 2736 2737 if (NULL != mOutput->stream->get_next_write_timestamp) { 2738 status = mOutput->stream->get_next_write_timestamp( 2739 mOutput->stream, &pts); 2740 } 2741 2742 if (status != NO_ERROR) { 2743 pts = AudioBufferProvider::kInvalidPTS; 2744 } 2745 2746 // mix buffers... 2747 mAudioMixer->process(pts); 2748 // increase sleep time progressively when application underrun condition clears. 2749 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2750 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2751 // such that we would underrun the audio HAL. 2752 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2753 sleepTimeShift--; 2754 } 2755 sleepTime = 0; 2756 standbyTime = systemTime() + standbyDelay; 2757 //TODO: delay standby when effects have a tail 2758} 2759 2760void AudioFlinger::MixerThread::threadLoop_sleepTime() 2761{ 2762 // If no tracks are ready, sleep once for the duration of an output 2763 // buffer size, then write 0s to the output 2764 if (sleepTime == 0) { 2765 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2766 sleepTime = activeSleepTime >> sleepTimeShift; 2767 if (sleepTime < kMinThreadSleepTimeUs) { 2768 sleepTime = kMinThreadSleepTimeUs; 2769 } 2770 // reduce sleep time in case of consecutive application underruns to avoid 2771 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2772 // duration we would end up writing less data than needed by the audio HAL if 2773 // the condition persists. 2774 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2775 sleepTimeShift++; 2776 } 2777 } else { 2778 sleepTime = idleSleepTime; 2779 } 2780 } else if (mBytesWritten != 0 || 2781 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2782 memset (mMixBuffer, 0, mixBufferSize); 2783 sleepTime = 0; 2784 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2785 } 2786 // TODO add standby time extension fct of effect tail 2787} 2788 2789// prepareTracks_l() must be called with ThreadBase::mLock held 2790AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2791 Vector< sp<Track> > *tracksToRemove) 2792{ 2793 2794 mixer_state mixerStatus = MIXER_IDLE; 2795 // find out which tracks need to be processed 2796 size_t count = mActiveTracks.size(); 2797 size_t mixedTracks = 0; 2798 size_t tracksWithEffect = 0; 2799 // counts only _active_ fast tracks 2800 size_t fastTracks = 0; 2801 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2802 2803 float masterVolume = mMasterVolume; 2804 bool masterMute = mMasterMute; 2805 2806 if (masterMute) { 2807 masterVolume = 0; 2808 } 2809 // Delegate master volume control to effect in output mix effect chain if needed 2810 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2811 if (chain != 0) { 2812 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2813 chain->setVolume_l(&v, &v); 2814 masterVolume = (float)((v + (1 << 23)) >> 24); 2815 chain.clear(); 2816 } 2817 2818 // prepare a new state to push 2819 FastMixerStateQueue *sq = NULL; 2820 FastMixerState *state = NULL; 2821 bool didModify = false; 2822 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2823 if (mFastMixer != NULL) { 2824 sq = mFastMixer->sq(); 2825 state = sq->begin(); 2826 } 2827 2828 for (size_t i=0 ; i<count ; i++) { 2829 sp<Track> t = mActiveTracks[i].promote(); 2830 if (t == 0) continue; 2831 2832 // this const just means the local variable doesn't change 2833 Track* const track = t.get(); 2834 2835 // process fast tracks 2836 if (track->isFastTrack()) { 2837 2838 // It's theoretically possible (though unlikely) for a fast track to be created 2839 // and then removed within the same normal mix cycle. This is not a problem, as 2840 // the track never becomes active so it's fast mixer slot is never touched. 2841 // The converse, of removing an (active) track and then creating a new track 2842 // at the identical fast mixer slot within the same normal mix cycle, 2843 // is impossible because the slot isn't marked available until the end of each cycle. 2844 int j = track->mFastIndex; 2845 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2846 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2847 FastTrack *fastTrack = &state->mFastTracks[j]; 2848 2849 // Determine whether the track is currently in underrun condition, 2850 // and whether it had a recent underrun. 2851 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2852 FastTrackUnderruns underruns = ftDump->mUnderruns; 2853 uint32_t recentFull = (underruns.mBitFields.mFull - 2854 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2855 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2856 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2857 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2858 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2859 uint32_t recentUnderruns = recentPartial + recentEmpty; 2860 track->mObservedUnderruns = underruns; 2861 // don't count underruns that occur while stopping or pausing 2862 // or stopped which can occur when flush() is called while active 2863 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2864 track->mUnderrunCount += recentUnderruns; 2865 } 2866 2867 // This is similar to the state machine for normal tracks, 2868 // with a few modifications for fast tracks. 2869 bool isActive = true; 2870 switch (track->mState) { 2871 case TrackBase::STOPPING_1: 2872 // track stays active in STOPPING_1 state until first underrun 2873 if (recentUnderruns > 0) { 2874 track->mState = TrackBase::STOPPING_2; 2875 } 2876 break; 2877 case TrackBase::PAUSING: 2878 // ramp down is not yet implemented 2879 track->setPaused(); 2880 break; 2881 case TrackBase::RESUMING: 2882 // ramp up is not yet implemented 2883 track->mState = TrackBase::ACTIVE; 2884 break; 2885 case TrackBase::ACTIVE: 2886 if (recentFull > 0 || recentPartial > 0) { 2887 // track has provided at least some frames recently: reset retry count 2888 track->mRetryCount = kMaxTrackRetries; 2889 } 2890 if (recentUnderruns == 0) { 2891 // no recent underruns: stay active 2892 break; 2893 } 2894 // there has recently been an underrun of some kind 2895 if (track->sharedBuffer() == 0) { 2896 // were any of the recent underruns "empty" (no frames available)? 2897 if (recentEmpty == 0) { 2898 // no, then ignore the partial underruns as they are allowed indefinitely 2899 break; 2900 } 2901 // there has recently been an "empty" underrun: decrement the retry counter 2902 if (--(track->mRetryCount) > 0) { 2903 break; 2904 } 2905 // indicate to client process that the track was disabled because of underrun; 2906 // it will then automatically call start() when data is available 2907 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2908 // remove from active list, but state remains ACTIVE [confusing but true] 2909 isActive = false; 2910 break; 2911 } 2912 // fall through 2913 case TrackBase::STOPPING_2: 2914 case TrackBase::PAUSED: 2915 case TrackBase::TERMINATED: 2916 case TrackBase::STOPPED: 2917 case TrackBase::FLUSHED: // flush() while active 2918 // Check for presentation complete if track is inactive 2919 // We have consumed all the buffers of this track. 2920 // This would be incomplete if we auto-paused on underrun 2921 { 2922 size_t audioHALFrames = 2923 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2924 size_t framesWritten = 2925 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2926 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2927 // track stays in active list until presentation is complete 2928 break; 2929 } 2930 } 2931 if (track->isStopping_2()) { 2932 track->mState = TrackBase::STOPPED; 2933 } 2934 if (track->isStopped()) { 2935 // Can't reset directly, as fast mixer is still polling this track 2936 // track->reset(); 2937 // So instead mark this track as needing to be reset after push with ack 2938 resetMask |= 1 << i; 2939 } 2940 isActive = false; 2941 break; 2942 case TrackBase::IDLE: 2943 default: 2944 LOG_FATAL("unexpected track state %d", track->mState); 2945 } 2946 2947 if (isActive) { 2948 // was it previously inactive? 2949 if (!(state->mTrackMask & (1 << j))) { 2950 ExtendedAudioBufferProvider *eabp = track; 2951 VolumeProvider *vp = track; 2952 fastTrack->mBufferProvider = eabp; 2953 fastTrack->mVolumeProvider = vp; 2954 fastTrack->mSampleRate = track->mSampleRate; 2955 fastTrack->mChannelMask = track->mChannelMask; 2956 fastTrack->mGeneration++; 2957 state->mTrackMask |= 1 << j; 2958 didModify = true; 2959 // no acknowledgement required for newly active tracks 2960 } 2961 // cache the combined master volume and stream type volume for fast mixer; this 2962 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2963 track->mCachedVolume = track->isMuted() ? 2964 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2965 ++fastTracks; 2966 } else { 2967 // was it previously active? 2968 if (state->mTrackMask & (1 << j)) { 2969 fastTrack->mBufferProvider = NULL; 2970 fastTrack->mGeneration++; 2971 state->mTrackMask &= ~(1 << j); 2972 didModify = true; 2973 // If any fast tracks were removed, we must wait for acknowledgement 2974 // because we're about to decrement the last sp<> on those tracks. 2975 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2976 } else { 2977 LOG_FATAL("fast track %d should have been active", j); 2978 } 2979 tracksToRemove->add(track); 2980 // Avoids a misleading display in dumpsys 2981 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2982 } 2983 continue; 2984 } 2985 2986 { // local variable scope to avoid goto warning 2987 2988 audio_track_cblk_t* cblk = track->cblk(); 2989 2990 // The first time a track is added we wait 2991 // for all its buffers to be filled before processing it 2992 int name = track->name(); 2993 // make sure that we have enough frames to mix one full buffer. 2994 // enforce this condition only once to enable draining the buffer in case the client 2995 // app does not call stop() and relies on underrun to stop: 2996 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2997 // during last round 2998 uint32_t minFrames = 1; 2999 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3000 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3001 if (t->sampleRate() == (int)mSampleRate) { 3002 minFrames = mNormalFrameCount; 3003 } else { 3004 // +1 for rounding and +1 for additional sample needed for interpolation 3005 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3006 // add frames already consumed but not yet released by the resampler 3007 // because cblk->framesReady() will include these frames 3008 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3009 // the minimum track buffer size is normally twice the number of frames necessary 3010 // to fill one buffer and the resampler should not leave more than one buffer worth 3011 // of unreleased frames after each pass, but just in case... 3012 ALOG_ASSERT(minFrames <= cblk->frameCount); 3013 } 3014 } 3015 if ((track->framesReady() >= minFrames) && track->isReady() && 3016 !track->isPaused() && !track->isTerminated()) 3017 { 3018 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3019 3020 mixedTracks++; 3021 3022 // track->mainBuffer() != mMixBuffer means there is an effect chain 3023 // connected to the track 3024 chain.clear(); 3025 if (track->mainBuffer() != mMixBuffer) { 3026 chain = getEffectChain_l(track->sessionId()); 3027 // Delegate volume control to effect in track effect chain if needed 3028 if (chain != 0) { 3029 tracksWithEffect++; 3030 } else { 3031 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3032 name, track->sessionId()); 3033 } 3034 } 3035 3036 3037 int param = AudioMixer::VOLUME; 3038 if (track->mFillingUpStatus == Track::FS_FILLED) { 3039 // no ramp for the first volume setting 3040 track->mFillingUpStatus = Track::FS_ACTIVE; 3041 if (track->mState == TrackBase::RESUMING) { 3042 track->mState = TrackBase::ACTIVE; 3043 param = AudioMixer::RAMP_VOLUME; 3044 } 3045 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3046 } else if (cblk->server != 0) { 3047 // If the track is stopped before the first frame was mixed, 3048 // do not apply ramp 3049 param = AudioMixer::RAMP_VOLUME; 3050 } 3051 3052 // compute volume for this track 3053 uint32_t vl, vr, va; 3054 if (track->isMuted() || track->isPausing() || 3055 mStreamTypes[track->streamType()].mute) { 3056 vl = vr = va = 0; 3057 if (track->isPausing()) { 3058 track->setPaused(); 3059 } 3060 } else { 3061 3062 // read original volumes with volume control 3063 float typeVolume = mStreamTypes[track->streamType()].volume; 3064 float v = masterVolume * typeVolume; 3065 uint32_t vlr = cblk->getVolumeLR(); 3066 vl = vlr & 0xFFFF; 3067 vr = vlr >> 16; 3068 // track volumes come from shared memory, so can't be trusted and must be clamped 3069 if (vl > MAX_GAIN_INT) { 3070 ALOGV("Track left volume out of range: %04X", vl); 3071 vl = MAX_GAIN_INT; 3072 } 3073 if (vr > MAX_GAIN_INT) { 3074 ALOGV("Track right volume out of range: %04X", vr); 3075 vr = MAX_GAIN_INT; 3076 } 3077 // now apply the master volume and stream type volume 3078 vl = (uint32_t)(v * vl) << 12; 3079 vr = (uint32_t)(v * vr) << 12; 3080 // assuming master volume and stream type volume each go up to 1.0, 3081 // vl and vr are now in 8.24 format 3082 3083 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3084 // send level comes from shared memory and so may be corrupt 3085 if (sendLevel > MAX_GAIN_INT) { 3086 ALOGV("Track send level out of range: %04X", sendLevel); 3087 sendLevel = MAX_GAIN_INT; 3088 } 3089 va = (uint32_t)(v * sendLevel); 3090 } 3091 // Delegate volume control to effect in track effect chain if needed 3092 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3093 // Do not ramp volume if volume is controlled by effect 3094 param = AudioMixer::VOLUME; 3095 track->mHasVolumeController = true; 3096 } else { 3097 // force no volume ramp when volume controller was just disabled or removed 3098 // from effect chain to avoid volume spike 3099 if (track->mHasVolumeController) { 3100 param = AudioMixer::VOLUME; 3101 } 3102 track->mHasVolumeController = false; 3103 } 3104 3105 // Convert volumes from 8.24 to 4.12 format 3106 // This additional clamping is needed in case chain->setVolume_l() overshot 3107 vl = (vl + (1 << 11)) >> 12; 3108 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3109 vr = (vr + (1 << 11)) >> 12; 3110 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3111 3112 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3113 3114 // XXX: these things DON'T need to be done each time 3115 mAudioMixer->setBufferProvider(name, track); 3116 mAudioMixer->enable(name); 3117 3118 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3119 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3120 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3121 mAudioMixer->setParameter( 3122 name, 3123 AudioMixer::TRACK, 3124 AudioMixer::FORMAT, (void *)track->format()); 3125 mAudioMixer->setParameter( 3126 name, 3127 AudioMixer::TRACK, 3128 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3129 mAudioMixer->setParameter( 3130 name, 3131 AudioMixer::RESAMPLE, 3132 AudioMixer::SAMPLE_RATE, 3133 (void *)(cblk->sampleRate)); 3134 mAudioMixer->setParameter( 3135 name, 3136 AudioMixer::TRACK, 3137 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3138 mAudioMixer->setParameter( 3139 name, 3140 AudioMixer::TRACK, 3141 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3142 3143 // reset retry count 3144 track->mRetryCount = kMaxTrackRetries; 3145 3146 // If one track is ready, set the mixer ready if: 3147 // - the mixer was not ready during previous round OR 3148 // - no other track is not ready 3149 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3150 mixerStatus != MIXER_TRACKS_ENABLED) { 3151 mixerStatus = MIXER_TRACKS_READY; 3152 } 3153 } else { 3154 // clear effect chain input buffer if an active track underruns to avoid sending 3155 // previous audio buffer again to effects 3156 chain = getEffectChain_l(track->sessionId()); 3157 if (chain != 0) { 3158 chain->clearInputBuffer(); 3159 } 3160 3161 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3162 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3163 track->isStopped() || track->isPaused()) { 3164 // We have consumed all the buffers of this track. 3165 // Remove it from the list of active tracks. 3166 // TODO: use actual buffer filling status instead of latency when available from 3167 // audio HAL 3168 size_t audioHALFrames = 3169 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3170 size_t framesWritten = 3171 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3172 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3173 if (track->isStopped()) { 3174 track->reset(); 3175 } 3176 tracksToRemove->add(track); 3177 } 3178 } else { 3179 track->mUnderrunCount++; 3180 // No buffers for this track. Give it a few chances to 3181 // fill a buffer, then remove it from active list. 3182 if (--(track->mRetryCount) <= 0) { 3183 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3184 tracksToRemove->add(track); 3185 // indicate to client process that the track was disabled because of underrun; 3186 // it will then automatically call start() when data is available 3187 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3188 // If one track is not ready, mark the mixer also not ready if: 3189 // - the mixer was ready during previous round OR 3190 // - no other track is ready 3191 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3192 mixerStatus != MIXER_TRACKS_READY) { 3193 mixerStatus = MIXER_TRACKS_ENABLED; 3194 } 3195 } 3196 mAudioMixer->disable(name); 3197 } 3198 3199 } // local variable scope to avoid goto warning 3200track_is_ready: ; 3201 3202 } 3203 3204 // Push the new FastMixer state if necessary 3205 if (didModify) { 3206 state->mFastTracksGen++; 3207 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3208 if (kUseFastMixer == FastMixer_Dynamic && 3209 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3210 state->mCommand = FastMixerState::COLD_IDLE; 3211 state->mColdFutexAddr = &mFastMixerFutex; 3212 state->mColdGen++; 3213 mFastMixerFutex = 0; 3214 if (kUseFastMixer == FastMixer_Dynamic) { 3215 mNormalSink = mOutputSink; 3216 } 3217 // If we go into cold idle, need to wait for acknowledgement 3218 // so that fast mixer stops doing I/O. 3219 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3220 } 3221 sq->end(); 3222 } 3223 if (sq != NULL) { 3224 sq->end(didModify); 3225 sq->push(block); 3226 } 3227 3228 // Now perform the deferred reset on fast tracks that have stopped 3229 while (resetMask != 0) { 3230 size_t i = __builtin_ctz(resetMask); 3231 ALOG_ASSERT(i < count); 3232 resetMask &= ~(1 << i); 3233 sp<Track> t = mActiveTracks[i].promote(); 3234 if (t == 0) continue; 3235 Track* track = t.get(); 3236 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3237 track->reset(); 3238 } 3239 3240 // remove all the tracks that need to be... 3241 count = tracksToRemove->size(); 3242 if (CC_UNLIKELY(count)) { 3243 for (size_t i=0 ; i<count ; i++) { 3244 const sp<Track>& track = tracksToRemove->itemAt(i); 3245 mActiveTracks.remove(track); 3246 if (track->mainBuffer() != mMixBuffer) { 3247 chain = getEffectChain_l(track->sessionId()); 3248 if (chain != 0) { 3249 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3250 chain->decActiveTrackCnt(); 3251 } 3252 } 3253 if (track->isTerminated()) { 3254 removeTrack_l(track); 3255 } 3256 } 3257 } 3258 3259 // mix buffer must be cleared if all tracks are connected to an 3260 // effect chain as in this case the mixer will not write to 3261 // mix buffer and track effects will accumulate into it 3262 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3263 // FIXME as a performance optimization, should remember previous zero status 3264 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3265 } 3266 3267 // if any fast tracks, then status is ready 3268 mMixerStatusIgnoringFastTracks = mixerStatus; 3269 if (fastTracks > 0) { 3270 mixerStatus = MIXER_TRACKS_READY; 3271 } 3272 return mixerStatus; 3273} 3274 3275/* 3276The derived values that are cached: 3277 - mixBufferSize from frame count * frame size 3278 - activeSleepTime from activeSleepTimeUs() 3279 - idleSleepTime from idleSleepTimeUs() 3280 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3281 - maxPeriod from frame count and sample rate (MIXER only) 3282 3283The parameters that affect these derived values are: 3284 - frame count 3285 - frame size 3286 - sample rate 3287 - device type: A2DP or not 3288 - device latency 3289 - format: PCM or not 3290 - active sleep time 3291 - idle sleep time 3292*/ 3293 3294void AudioFlinger::PlaybackThread::cacheParameters_l() 3295{ 3296 mixBufferSize = mNormalFrameCount * mFrameSize; 3297 activeSleepTime = activeSleepTimeUs(); 3298 idleSleepTime = idleSleepTimeUs(); 3299} 3300 3301void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3302{ 3303 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3304 this, streamType, mTracks.size()); 3305 Mutex::Autolock _l(mLock); 3306 3307 size_t size = mTracks.size(); 3308 for (size_t i = 0; i < size; i++) { 3309 sp<Track> t = mTracks[i]; 3310 if (t->streamType() == streamType) { 3311 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3312 t->mCblk->cv.signal(); 3313 } 3314 } 3315} 3316 3317// getTrackName_l() must be called with ThreadBase::mLock held 3318int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3319{ 3320 return mAudioMixer->getTrackName(channelMask); 3321} 3322 3323// deleteTrackName_l() must be called with ThreadBase::mLock held 3324void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3325{ 3326 ALOGV("remove track (%d) and delete from mixer", name); 3327 mAudioMixer->deleteTrackName(name); 3328} 3329 3330// checkForNewParameters_l() must be called with ThreadBase::mLock held 3331bool AudioFlinger::MixerThread::checkForNewParameters_l() 3332{ 3333 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3334 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3335 bool reconfig = false; 3336 3337 while (!mNewParameters.isEmpty()) { 3338 3339 if (mFastMixer != NULL) { 3340 FastMixerStateQueue *sq = mFastMixer->sq(); 3341 FastMixerState *state = sq->begin(); 3342 if (!(state->mCommand & FastMixerState::IDLE)) { 3343 previousCommand = state->mCommand; 3344 state->mCommand = FastMixerState::HOT_IDLE; 3345 sq->end(); 3346 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3347 } else { 3348 sq->end(false /*didModify*/); 3349 } 3350 } 3351 3352 status_t status = NO_ERROR; 3353 String8 keyValuePair = mNewParameters[0]; 3354 AudioParameter param = AudioParameter(keyValuePair); 3355 int value; 3356 3357 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3358 reconfig = true; 3359 } 3360 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3361 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3362 status = BAD_VALUE; 3363 } else { 3364 reconfig = true; 3365 } 3366 } 3367 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3368 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3369 status = BAD_VALUE; 3370 } else { 3371 reconfig = true; 3372 } 3373 } 3374 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3375 // do not accept frame count changes if tracks are open as the track buffer 3376 // size depends on frame count and correct behavior would not be guaranteed 3377 // if frame count is changed after track creation 3378 if (!mTracks.isEmpty()) { 3379 status = INVALID_OPERATION; 3380 } else { 3381 reconfig = true; 3382 } 3383 } 3384 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3385#ifdef ADD_BATTERY_DATA 3386 // when changing the audio output device, call addBatteryData to notify 3387 // the change 3388 if ((int)mDevice != value) { 3389 uint32_t params = 0; 3390 // check whether speaker is on 3391 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3392 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3393 } 3394 3395 int deviceWithoutSpeaker 3396 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3397 // check if any other device (except speaker) is on 3398 if (value & deviceWithoutSpeaker ) { 3399 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3400 } 3401 3402 if (params != 0) { 3403 addBatteryData(params); 3404 } 3405 } 3406#endif 3407 3408 // forward device change to effects that have requested to be 3409 // aware of attached audio device. 3410 mDevice = (uint32_t)value; 3411 for (size_t i = 0; i < mEffectChains.size(); i++) { 3412 mEffectChains[i]->setDevice_l(mDevice); 3413 } 3414 } 3415 3416 if (status == NO_ERROR) { 3417 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3418 keyValuePair.string()); 3419 if (!mStandby && status == INVALID_OPERATION) { 3420 mOutput->stream->common.standby(&mOutput->stream->common); 3421 mStandby = true; 3422 mBytesWritten = 0; 3423 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3424 keyValuePair.string()); 3425 } 3426 if (status == NO_ERROR && reconfig) { 3427 delete mAudioMixer; 3428 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3429 mAudioMixer = NULL; 3430 readOutputParameters(); 3431 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3432 for (size_t i = 0; i < mTracks.size() ; i++) { 3433 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3434 if (name < 0) break; 3435 mTracks[i]->mName = name; 3436 // limit track sample rate to 2 x new output sample rate 3437 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3438 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3439 } 3440 } 3441 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3442 } 3443 } 3444 3445 mNewParameters.removeAt(0); 3446 3447 mParamStatus = status; 3448 mParamCond.signal(); 3449 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3450 // already timed out waiting for the status and will never signal the condition. 3451 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3452 } 3453 3454 if (!(previousCommand & FastMixerState::IDLE)) { 3455 ALOG_ASSERT(mFastMixer != NULL); 3456 FastMixerStateQueue *sq = mFastMixer->sq(); 3457 FastMixerState *state = sq->begin(); 3458 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3459 state->mCommand = previousCommand; 3460 sq->end(); 3461 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3462 } 3463 3464 return reconfig; 3465} 3466 3467status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3468{ 3469 const size_t SIZE = 256; 3470 char buffer[SIZE]; 3471 String8 result; 3472 3473 PlaybackThread::dumpInternals(fd, args); 3474 3475 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3476 result.append(buffer); 3477 write(fd, result.string(), result.size()); 3478 3479 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3480 FastMixerDumpState copy = mFastMixerDumpState; 3481 copy.dump(fd); 3482 3483 // Write the tee output to a .wav file 3484 NBAIO_Source *teeSource = mTeeSource.get(); 3485 if (teeSource != NULL) { 3486 char teePath[64]; 3487 struct timeval tv; 3488 gettimeofday(&tv, NULL); 3489 struct tm tm; 3490 localtime_r(&tv.tv_sec, &tm); 3491 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3492 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3493 if (teeFd >= 0) { 3494 char wavHeader[44]; 3495 memcpy(wavHeader, 3496 "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", 3497 sizeof(wavHeader)); 3498 NBAIO_Format format = teeSource->format(); 3499 unsigned channelCount = Format_channelCount(format); 3500 ALOG_ASSERT(channelCount <= FCC_2); 3501 unsigned sampleRate = Format_sampleRate(format); 3502 wavHeader[22] = channelCount; // number of channels 3503 wavHeader[24] = sampleRate; // sample rate 3504 wavHeader[25] = sampleRate >> 8; 3505 wavHeader[32] = channelCount * 2; // block alignment 3506 write(teeFd, wavHeader, sizeof(wavHeader)); 3507 size_t total = 0; 3508 bool firstRead = true; 3509 for (;;) { 3510#define TEE_SINK_READ 1024 3511 short buffer[TEE_SINK_READ * FCC_2]; 3512 size_t count = TEE_SINK_READ; 3513 ssize_t actual = teeSource->read(buffer, count); 3514 bool wasFirstRead = firstRead; 3515 firstRead = false; 3516 if (actual <= 0) { 3517 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3518 continue; 3519 } 3520 break; 3521 } 3522 ALOG_ASSERT(actual <= count); 3523 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3524 total += actual; 3525 } 3526 lseek(teeFd, (off_t) 4, SEEK_SET); 3527 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3528 write(teeFd, &temp, sizeof(temp)); 3529 lseek(teeFd, (off_t) 40, SEEK_SET); 3530 temp = total * channelCount * sizeof(short); 3531 write(teeFd, &temp, sizeof(temp)); 3532 close(teeFd); 3533 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3534 } else { 3535 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3536 } 3537 } 3538 3539 return NO_ERROR; 3540} 3541 3542uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3543{ 3544 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3545} 3546 3547uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3548{ 3549 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3550} 3551 3552void AudioFlinger::MixerThread::cacheParameters_l() 3553{ 3554 PlaybackThread::cacheParameters_l(); 3555 3556 // FIXME: Relaxed timing because of a certain device that can't meet latency 3557 // Should be reduced to 2x after the vendor fixes the driver issue 3558 // increase threshold again due to low power audio mode. The way this warning 3559 // threshold is calculated and its usefulness should be reconsidered anyway. 3560 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3561} 3562 3563// ---------------------------------------------------------------------------- 3564AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3565 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3566 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3567 // mLeftVolFloat, mRightVolFloat 3568 // mLeftVolShort, mRightVolShort 3569{ 3570} 3571 3572AudioFlinger::DirectOutputThread::~DirectOutputThread() 3573{ 3574} 3575 3576AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3577 Vector< sp<Track> > *tracksToRemove 3578) 3579{ 3580 sp<Track> trackToRemove; 3581 3582 mixer_state mixerStatus = MIXER_IDLE; 3583 3584 // find out which tracks need to be processed 3585 if (mActiveTracks.size() != 0) { 3586 sp<Track> t = mActiveTracks[0].promote(); 3587 // The track died recently 3588 if (t == 0) return MIXER_IDLE; 3589 3590 Track* const track = t.get(); 3591 audio_track_cblk_t* cblk = track->cblk(); 3592 3593 // The first time a track is added we wait 3594 // for all its buffers to be filled before processing it 3595 if (cblk->framesReady() && track->isReady() && 3596 !track->isPaused() && !track->isTerminated()) 3597 { 3598 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3599 3600 if (track->mFillingUpStatus == Track::FS_FILLED) { 3601 track->mFillingUpStatus = Track::FS_ACTIVE; 3602 mLeftVolFloat = mRightVolFloat = 0; 3603 mLeftVolShort = mRightVolShort = 0; 3604 if (track->mState == TrackBase::RESUMING) { 3605 track->mState = TrackBase::ACTIVE; 3606 rampVolume = true; 3607 } 3608 } else if (cblk->server != 0) { 3609 // If the track is stopped before the first frame was mixed, 3610 // do not apply ramp 3611 rampVolume = true; 3612 } 3613 // compute volume for this track 3614 float left, right; 3615 if (track->isMuted() || mMasterMute || track->isPausing() || 3616 mStreamTypes[track->streamType()].mute) { 3617 left = right = 0; 3618 if (track->isPausing()) { 3619 track->setPaused(); 3620 } 3621 } else { 3622 float typeVolume = mStreamTypes[track->streamType()].volume; 3623 float v = mMasterVolume * typeVolume; 3624 uint32_t vlr = cblk->getVolumeLR(); 3625 float v_clamped = v * (vlr & 0xFFFF); 3626 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3627 left = v_clamped/MAX_GAIN; 3628 v_clamped = v * (vlr >> 16); 3629 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3630 right = v_clamped/MAX_GAIN; 3631 } 3632 3633 if (left != mLeftVolFloat || right != mRightVolFloat) { 3634 mLeftVolFloat = left; 3635 mRightVolFloat = right; 3636 3637 // If audio HAL implements volume control, 3638 // force software volume to nominal value 3639 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3640 left = 1.0f; 3641 right = 1.0f; 3642 } 3643 3644 // Convert volumes from float to 8.24 3645 uint32_t vl = (uint32_t)(left * (1 << 24)); 3646 uint32_t vr = (uint32_t)(right * (1 << 24)); 3647 3648 // Delegate volume control to effect in track effect chain if needed 3649 // only one effect chain can be present on DirectOutputThread, so if 3650 // there is one, the track is connected to it 3651 if (!mEffectChains.isEmpty()) { 3652 // Do not ramp volume if volume is controlled by effect 3653 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3654 rampVolume = false; 3655 } 3656 } 3657 3658 // Convert volumes from 8.24 to 4.12 format 3659 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3660 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3661 leftVol = (uint16_t)v_clamped; 3662 v_clamped = (vr + (1 << 11)) >> 12; 3663 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3664 rightVol = (uint16_t)v_clamped; 3665 } else { 3666 leftVol = mLeftVolShort; 3667 rightVol = mRightVolShort; 3668 rampVolume = false; 3669 } 3670 3671 // reset retry count 3672 track->mRetryCount = kMaxTrackRetriesDirect; 3673 mActiveTrack = t; 3674 mixerStatus = MIXER_TRACKS_READY; 3675 } else { 3676 // clear effect chain input buffer if an active track underruns to avoid sending 3677 // previous audio buffer again to effects 3678 if (!mEffectChains.isEmpty()) { 3679 mEffectChains[0]->clearInputBuffer(); 3680 } 3681 3682 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3683 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3684 // We have consumed all the buffers of this track. 3685 // Remove it from the list of active tracks. 3686 // TODO: implement behavior for compressed audio 3687 size_t audioHALFrames = 3688 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3689 size_t framesWritten = 3690 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3691 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3692 if (track->isStopped()) { 3693 track->reset(); 3694 } 3695 trackToRemove = track; 3696 } 3697 } else { 3698 // No buffers for this track. Give it a few chances to 3699 // fill a buffer, then remove it from active list. 3700 if (--(track->mRetryCount) <= 0) { 3701 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3702 trackToRemove = track; 3703 } else { 3704 mixerStatus = MIXER_TRACKS_ENABLED; 3705 } 3706 } 3707 } 3708 } 3709 3710 // FIXME merge this with similar code for removing multiple tracks 3711 // remove all the tracks that need to be... 3712 if (CC_UNLIKELY(trackToRemove != 0)) { 3713 tracksToRemove->add(trackToRemove); 3714 mActiveTracks.remove(trackToRemove); 3715 if (!mEffectChains.isEmpty()) { 3716 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3717 trackToRemove->sessionId()); 3718 mEffectChains[0]->decActiveTrackCnt(); 3719 } 3720 if (trackToRemove->isTerminated()) { 3721 removeTrack_l(trackToRemove); 3722 } 3723 } 3724 3725 return mixerStatus; 3726} 3727 3728void AudioFlinger::DirectOutputThread::threadLoop_mix() 3729{ 3730 AudioBufferProvider::Buffer buffer; 3731 size_t frameCount = mFrameCount; 3732 int8_t *curBuf = (int8_t *)mMixBuffer; 3733 // output audio to hardware 3734 while (frameCount) { 3735 buffer.frameCount = frameCount; 3736 mActiveTrack->getNextBuffer(&buffer); 3737 if (CC_UNLIKELY(buffer.raw == NULL)) { 3738 memset(curBuf, 0, frameCount * mFrameSize); 3739 break; 3740 } 3741 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3742 frameCount -= buffer.frameCount; 3743 curBuf += buffer.frameCount * mFrameSize; 3744 mActiveTrack->releaseBuffer(&buffer); 3745 } 3746 sleepTime = 0; 3747 standbyTime = systemTime() + standbyDelay; 3748 mActiveTrack.clear(); 3749 3750 // apply volume 3751 3752 // Do not apply volume on compressed audio 3753 if (!audio_is_linear_pcm(mFormat)) { 3754 return; 3755 } 3756 3757 // convert to signed 16 bit before volume calculation 3758 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3759 size_t count = mFrameCount * mChannelCount; 3760 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3761 int16_t *dst = mMixBuffer + count-1; 3762 while (count--) { 3763 *dst-- = (int16_t)(*src--^0x80) << 8; 3764 } 3765 } 3766 3767 frameCount = mFrameCount; 3768 int16_t *out = mMixBuffer; 3769 if (rampVolume) { 3770 if (mChannelCount == 1) { 3771 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3772 int32_t vlInc = d / (int32_t)frameCount; 3773 int32_t vl = ((int32_t)mLeftVolShort << 16); 3774 do { 3775 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3776 out++; 3777 vl += vlInc; 3778 } while (--frameCount); 3779 3780 } else { 3781 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3782 int32_t vlInc = d / (int32_t)frameCount; 3783 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3784 int32_t vrInc = d / (int32_t)frameCount; 3785 int32_t vl = ((int32_t)mLeftVolShort << 16); 3786 int32_t vr = ((int32_t)mRightVolShort << 16); 3787 do { 3788 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3789 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3790 out += 2; 3791 vl += vlInc; 3792 vr += vrInc; 3793 } while (--frameCount); 3794 } 3795 } else { 3796 if (mChannelCount == 1) { 3797 do { 3798 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3799 out++; 3800 } while (--frameCount); 3801 } else { 3802 do { 3803 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3804 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3805 out += 2; 3806 } while (--frameCount); 3807 } 3808 } 3809 3810 // convert back to unsigned 8 bit after volume calculation 3811 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3812 size_t count = mFrameCount * mChannelCount; 3813 int16_t *src = mMixBuffer; 3814 uint8_t *dst = (uint8_t *)mMixBuffer; 3815 while (count--) { 3816 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3817 } 3818 } 3819 3820 mLeftVolShort = leftVol; 3821 mRightVolShort = rightVol; 3822} 3823 3824void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3825{ 3826 if (sleepTime == 0) { 3827 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3828 sleepTime = activeSleepTime; 3829 } else { 3830 sleepTime = idleSleepTime; 3831 } 3832 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3833 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3834 sleepTime = 0; 3835 } 3836} 3837 3838// getTrackName_l() must be called with ThreadBase::mLock held 3839int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3840{ 3841 return 0; 3842} 3843 3844// deleteTrackName_l() must be called with ThreadBase::mLock held 3845void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3846{ 3847} 3848 3849// checkForNewParameters_l() must be called with ThreadBase::mLock held 3850bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3851{ 3852 bool reconfig = false; 3853 3854 while (!mNewParameters.isEmpty()) { 3855 status_t status = NO_ERROR; 3856 String8 keyValuePair = mNewParameters[0]; 3857 AudioParameter param = AudioParameter(keyValuePair); 3858 int value; 3859 3860 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3861 // do not accept frame count changes if tracks are open as the track buffer 3862 // size depends on frame count and correct behavior would not be garantied 3863 // if frame count is changed after track creation 3864 if (!mTracks.isEmpty()) { 3865 status = INVALID_OPERATION; 3866 } else { 3867 reconfig = true; 3868 } 3869 } 3870 if (status == NO_ERROR) { 3871 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3872 keyValuePair.string()); 3873 if (!mStandby && status == INVALID_OPERATION) { 3874 mOutput->stream->common.standby(&mOutput->stream->common); 3875 mStandby = true; 3876 mBytesWritten = 0; 3877 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3878 keyValuePair.string()); 3879 } 3880 if (status == NO_ERROR && reconfig) { 3881 readOutputParameters(); 3882 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3883 } 3884 } 3885 3886 mNewParameters.removeAt(0); 3887 3888 mParamStatus = status; 3889 mParamCond.signal(); 3890 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3891 // already timed out waiting for the status and will never signal the condition. 3892 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3893 } 3894 return reconfig; 3895} 3896 3897uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3898{ 3899 uint32_t time; 3900 if (audio_is_linear_pcm(mFormat)) { 3901 time = PlaybackThread::activeSleepTimeUs(); 3902 } else { 3903 time = 10000; 3904 } 3905 return time; 3906} 3907 3908uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3909{ 3910 uint32_t time; 3911 if (audio_is_linear_pcm(mFormat)) { 3912 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3913 } else { 3914 time = 10000; 3915 } 3916 return time; 3917} 3918 3919uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3920{ 3921 uint32_t time; 3922 if (audio_is_linear_pcm(mFormat)) { 3923 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3924 } else { 3925 time = 10000; 3926 } 3927 return time; 3928} 3929 3930void AudioFlinger::DirectOutputThread::cacheParameters_l() 3931{ 3932 PlaybackThread::cacheParameters_l(); 3933 3934 // use shorter standby delay as on normal output to release 3935 // hardware resources as soon as possible 3936 standbyDelay = microseconds(activeSleepTime*2); 3937} 3938 3939// ---------------------------------------------------------------------------- 3940 3941AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3942 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3943 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3944 mWaitTimeMs(UINT_MAX) 3945{ 3946 addOutputTrack(mainThread); 3947} 3948 3949AudioFlinger::DuplicatingThread::~DuplicatingThread() 3950{ 3951 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3952 mOutputTracks[i]->destroy(); 3953 } 3954} 3955 3956void AudioFlinger::DuplicatingThread::threadLoop_mix() 3957{ 3958 // mix buffers... 3959 if (outputsReady(outputTracks)) { 3960 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3961 } else { 3962 memset(mMixBuffer, 0, mixBufferSize); 3963 } 3964 sleepTime = 0; 3965 writeFrames = mNormalFrameCount; 3966} 3967 3968void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3969{ 3970 if (sleepTime == 0) { 3971 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3972 sleepTime = activeSleepTime; 3973 } else { 3974 sleepTime = idleSleepTime; 3975 } 3976 } else if (mBytesWritten != 0) { 3977 // flush remaining overflow buffers in output tracks 3978 for (size_t i = 0; i < outputTracks.size(); i++) { 3979 if (outputTracks[i]->isActive()) { 3980 sleepTime = 0; 3981 writeFrames = 0; 3982 memset(mMixBuffer, 0, mixBufferSize); 3983 break; 3984 } 3985 } 3986 } 3987} 3988 3989void AudioFlinger::DuplicatingThread::threadLoop_write() 3990{ 3991 standbyTime = systemTime() + standbyDelay; 3992 for (size_t i = 0; i < outputTracks.size(); i++) { 3993 outputTracks[i]->write(mMixBuffer, writeFrames); 3994 } 3995 mBytesWritten += mixBufferSize; 3996} 3997 3998void AudioFlinger::DuplicatingThread::threadLoop_standby() 3999{ 4000 // DuplicatingThread implements standby by stopping all tracks 4001 for (size_t i = 0; i < outputTracks.size(); i++) { 4002 outputTracks[i]->stop(); 4003 } 4004} 4005 4006void AudioFlinger::DuplicatingThread::saveOutputTracks() 4007{ 4008 outputTracks = mOutputTracks; 4009} 4010 4011void AudioFlinger::DuplicatingThread::clearOutputTracks() 4012{ 4013 outputTracks.clear(); 4014} 4015 4016void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4017{ 4018 Mutex::Autolock _l(mLock); 4019 // FIXME explain this formula 4020 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4021 OutputTrack *outputTrack = new OutputTrack(thread, 4022 this, 4023 mSampleRate, 4024 mFormat, 4025 mChannelMask, 4026 frameCount); 4027 if (outputTrack->cblk() != NULL) { 4028 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4029 mOutputTracks.add(outputTrack); 4030 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4031 updateWaitTime_l(); 4032 } 4033} 4034 4035void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4036{ 4037 Mutex::Autolock _l(mLock); 4038 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4039 if (mOutputTracks[i]->thread() == thread) { 4040 mOutputTracks[i]->destroy(); 4041 mOutputTracks.removeAt(i); 4042 updateWaitTime_l(); 4043 return; 4044 } 4045 } 4046 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4047} 4048 4049// caller must hold mLock 4050void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4051{ 4052 mWaitTimeMs = UINT_MAX; 4053 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4054 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4055 if (strong != 0) { 4056 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4057 if (waitTimeMs < mWaitTimeMs) { 4058 mWaitTimeMs = waitTimeMs; 4059 } 4060 } 4061 } 4062} 4063 4064 4065bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4066{ 4067 for (size_t i = 0; i < outputTracks.size(); i++) { 4068 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4069 if (thread == 0) { 4070 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4071 return false; 4072 } 4073 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4074 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4075 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4076 return false; 4077 } 4078 } 4079 return true; 4080} 4081 4082uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4083{ 4084 return (mWaitTimeMs * 1000) / 2; 4085} 4086 4087void AudioFlinger::DuplicatingThread::cacheParameters_l() 4088{ 4089 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4090 updateWaitTime_l(); 4091 4092 MixerThread::cacheParameters_l(); 4093} 4094 4095// ---------------------------------------------------------------------------- 4096 4097// TrackBase constructor must be called with AudioFlinger::mLock held 4098AudioFlinger::ThreadBase::TrackBase::TrackBase( 4099 ThreadBase *thread, 4100 const sp<Client>& client, 4101 uint32_t sampleRate, 4102 audio_format_t format, 4103 uint32_t channelMask, 4104 int frameCount, 4105 const sp<IMemory>& sharedBuffer, 4106 int sessionId) 4107 : RefBase(), 4108 mThread(thread), 4109 mClient(client), 4110 mCblk(NULL), 4111 // mBuffer 4112 // mBufferEnd 4113 mFrameCount(0), 4114 mState(IDLE), 4115 mSampleRate(sampleRate), 4116 mFormat(format), 4117 mStepServerFailed(false), 4118 mSessionId(sessionId) 4119 // mChannelCount 4120 // mChannelMask 4121{ 4122 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4123 4124 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4125 size_t size = sizeof(audio_track_cblk_t); 4126 uint8_t channelCount = popcount(channelMask); 4127 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4128 if (sharedBuffer == 0) { 4129 size += bufferSize; 4130 } 4131 4132 if (client != NULL) { 4133 mCblkMemory = client->heap()->allocate(size); 4134 if (mCblkMemory != 0) { 4135 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4136 if (mCblk != NULL) { // construct the shared structure in-place. 4137 new(mCblk) audio_track_cblk_t(); 4138 // clear all buffers 4139 mCblk->frameCount = frameCount; 4140 mCblk->sampleRate = sampleRate; 4141// uncomment the following lines to quickly test 32-bit wraparound 4142// mCblk->user = 0xffff0000; 4143// mCblk->server = 0xffff0000; 4144// mCblk->userBase = 0xffff0000; 4145// mCblk->serverBase = 0xffff0000; 4146 mChannelCount = channelCount; 4147 mChannelMask = channelMask; 4148 if (sharedBuffer == 0) { 4149 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4150 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4151 // Force underrun condition to avoid false underrun callback until first data is 4152 // written to buffer (other flags are cleared) 4153 mCblk->flags = CBLK_UNDERRUN_ON; 4154 } else { 4155 mBuffer = sharedBuffer->pointer(); 4156 } 4157 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4158 } 4159 } else { 4160 ALOGE("not enough memory for AudioTrack size=%u", size); 4161 client->heap()->dump("AudioTrack"); 4162 return; 4163 } 4164 } else { 4165 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4166 // construct the shared structure in-place. 4167 new(mCblk) audio_track_cblk_t(); 4168 // clear all buffers 4169 mCblk->frameCount = frameCount; 4170 mCblk->sampleRate = sampleRate; 4171// uncomment the following lines to quickly test 32-bit wraparound 4172// mCblk->user = 0xffff0000; 4173// mCblk->server = 0xffff0000; 4174// mCblk->userBase = 0xffff0000; 4175// mCblk->serverBase = 0xffff0000; 4176 mChannelCount = channelCount; 4177 mChannelMask = channelMask; 4178 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4179 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4180 // Force underrun condition to avoid false underrun callback until first data is 4181 // written to buffer (other flags are cleared) 4182 mCblk->flags = CBLK_UNDERRUN_ON; 4183 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4184 } 4185} 4186 4187AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4188{ 4189 if (mCblk != NULL) { 4190 if (mClient == 0) { 4191 delete mCblk; 4192 } else { 4193 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4194 } 4195 } 4196 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4197 if (mClient != 0) { 4198 // Client destructor must run with AudioFlinger mutex locked 4199 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4200 // If the client's reference count drops to zero, the associated destructor 4201 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4202 // relying on the automatic clear() at end of scope. 4203 mClient.clear(); 4204 } 4205} 4206 4207// AudioBufferProvider interface 4208// getNextBuffer() = 0; 4209// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4210void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4211{ 4212 buffer->raw = NULL; 4213 mFrameCount = buffer->frameCount; 4214 // FIXME See note at getNextBuffer() 4215 (void) step(); // ignore return value of step() 4216 buffer->frameCount = 0; 4217} 4218 4219bool AudioFlinger::ThreadBase::TrackBase::step() { 4220 bool result; 4221 audio_track_cblk_t* cblk = this->cblk(); 4222 4223 result = cblk->stepServer(mFrameCount); 4224 if (!result) { 4225 ALOGV("stepServer failed acquiring cblk mutex"); 4226 mStepServerFailed = true; 4227 } 4228 return result; 4229} 4230 4231void AudioFlinger::ThreadBase::TrackBase::reset() { 4232 audio_track_cblk_t* cblk = this->cblk(); 4233 4234 cblk->user = 0; 4235 cblk->server = 0; 4236 cblk->userBase = 0; 4237 cblk->serverBase = 0; 4238 mStepServerFailed = false; 4239 ALOGV("TrackBase::reset"); 4240} 4241 4242int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4243 return (int)mCblk->sampleRate; 4244} 4245 4246void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4247 audio_track_cblk_t* cblk = this->cblk(); 4248 size_t frameSize = cblk->frameSize; 4249 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4250 int8_t *bufferEnd = bufferStart + frames * frameSize; 4251 4252 // Check validity of returned pointer in case the track control block would have been corrupted. 4253 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4254 "TrackBase::getBuffer buffer out of range:\n" 4255 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4256 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4257 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4258 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4259 4260 return bufferStart; 4261} 4262 4263status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4264{ 4265 mSyncEvents.add(event); 4266 return NO_ERROR; 4267} 4268 4269// ---------------------------------------------------------------------------- 4270 4271// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4272AudioFlinger::PlaybackThread::Track::Track( 4273 PlaybackThread *thread, 4274 const sp<Client>& client, 4275 audio_stream_type_t streamType, 4276 uint32_t sampleRate, 4277 audio_format_t format, 4278 uint32_t channelMask, 4279 int frameCount, 4280 const sp<IMemory>& sharedBuffer, 4281 int sessionId, 4282 IAudioFlinger::track_flags_t flags) 4283 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4284 mMute(false), 4285 mFillingUpStatus(FS_INVALID), 4286 // mRetryCount initialized later when needed 4287 mSharedBuffer(sharedBuffer), 4288 mStreamType(streamType), 4289 mName(-1), // see note below 4290 mMainBuffer(thread->mixBuffer()), 4291 mAuxBuffer(NULL), 4292 mAuxEffectId(0), mHasVolumeController(false), 4293 mPresentationCompleteFrames(0), 4294 mFlags(flags), 4295 mFastIndex(-1), 4296 mUnderrunCount(0), 4297 mCachedVolume(1.0) 4298{ 4299 if (mCblk != NULL) { 4300 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4301 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4302 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4303 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4304 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4305 if (mName < 0) { 4306 ALOGE("no more track names available"); 4307 return; 4308 } 4309 // only allocate a fast track index if we were able to allocate a normal track name 4310 if (flags & IAudioFlinger::TRACK_FAST) { 4311 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4312 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4313 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4314 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4315 // FIXME This is too eager. We allocate a fast track index before the 4316 // fast track becomes active. Since fast tracks are a scarce resource, 4317 // this means we are potentially denying other more important fast tracks from 4318 // being created. It would be better to allocate the index dynamically. 4319 mFastIndex = i; 4320 // Read the initial underruns because this field is never cleared by the fast mixer 4321 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4322 thread->mFastTrackAvailMask &= ~(1 << i); 4323 } 4324 } 4325 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4326} 4327 4328AudioFlinger::PlaybackThread::Track::~Track() 4329{ 4330 ALOGV("PlaybackThread::Track destructor"); 4331 sp<ThreadBase> thread = mThread.promote(); 4332 if (thread != 0) { 4333 Mutex::Autolock _l(thread->mLock); 4334 mState = TERMINATED; 4335 } 4336} 4337 4338void AudioFlinger::PlaybackThread::Track::destroy() 4339{ 4340 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4341 // by removing it from mTracks vector, so there is a risk that this Tracks's 4342 // destructor is called. As the destructor needs to lock mLock, 4343 // we must acquire a strong reference on this Track before locking mLock 4344 // here so that the destructor is called only when exiting this function. 4345 // On the other hand, as long as Track::destroy() is only called by 4346 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4347 // this Track with its member mTrack. 4348 sp<Track> keep(this); 4349 { // scope for mLock 4350 sp<ThreadBase> thread = mThread.promote(); 4351 if (thread != 0) { 4352 if (!isOutputTrack()) { 4353 if (mState == ACTIVE || mState == RESUMING) { 4354 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4355 4356#ifdef ADD_BATTERY_DATA 4357 // to track the speaker usage 4358 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4359#endif 4360 } 4361 AudioSystem::releaseOutput(thread->id()); 4362 } 4363 Mutex::Autolock _l(thread->mLock); 4364 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4365 playbackThread->destroyTrack_l(this); 4366 } 4367 } 4368} 4369 4370/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4371{ 4372 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4373 " Server User Main buf Aux Buf Flags Underruns\n"); 4374} 4375 4376void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4377{ 4378 uint32_t vlr = mCblk->getVolumeLR(); 4379 if (isFastTrack()) { 4380 sprintf(buffer, " F %2d", mFastIndex); 4381 } else { 4382 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4383 } 4384 track_state state = mState; 4385 char stateChar; 4386 switch (state) { 4387 case IDLE: 4388 stateChar = 'I'; 4389 break; 4390 case TERMINATED: 4391 stateChar = 'T'; 4392 break; 4393 case STOPPING_1: 4394 stateChar = 's'; 4395 break; 4396 case STOPPING_2: 4397 stateChar = '5'; 4398 break; 4399 case STOPPED: 4400 stateChar = 'S'; 4401 break; 4402 case RESUMING: 4403 stateChar = 'R'; 4404 break; 4405 case ACTIVE: 4406 stateChar = 'A'; 4407 break; 4408 case PAUSING: 4409 stateChar = 'p'; 4410 break; 4411 case PAUSED: 4412 stateChar = 'P'; 4413 break; 4414 case FLUSHED: 4415 stateChar = 'F'; 4416 break; 4417 default: 4418 stateChar = '?'; 4419 break; 4420 } 4421 char nowInUnderrun; 4422 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4423 case UNDERRUN_FULL: 4424 nowInUnderrun = ' '; 4425 break; 4426 case UNDERRUN_PARTIAL: 4427 nowInUnderrun = '<'; 4428 break; 4429 case UNDERRUN_EMPTY: 4430 nowInUnderrun = '*'; 4431 break; 4432 default: 4433 nowInUnderrun = '?'; 4434 break; 4435 } 4436 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4437 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4438 (mClient == 0) ? getpid_cached : mClient->pid(), 4439 mStreamType, 4440 mFormat, 4441 mChannelMask, 4442 mSessionId, 4443 mFrameCount, 4444 mCblk->frameCount, 4445 stateChar, 4446 mMute, 4447 mFillingUpStatus, 4448 mCblk->sampleRate, 4449 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4450 20.0 * log10((vlr >> 16) / 4096.0), 4451 mCblk->server, 4452 mCblk->user, 4453 (int)mMainBuffer, 4454 (int)mAuxBuffer, 4455 mCblk->flags, 4456 mUnderrunCount, 4457 nowInUnderrun); 4458} 4459 4460// AudioBufferProvider interface 4461status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4462 AudioBufferProvider::Buffer* buffer, int64_t pts) 4463{ 4464 audio_track_cblk_t* cblk = this->cblk(); 4465 uint32_t framesReady; 4466 uint32_t framesReq = buffer->frameCount; 4467 4468 // Check if last stepServer failed, try to step now 4469 if (mStepServerFailed) { 4470 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4471 // Since the fast mixer is higher priority than client callback thread, 4472 // it does not result in priority inversion for client. 4473 // But a non-blocking solution would be preferable to avoid 4474 // fast mixer being unable to tryLock(), and 4475 // to avoid the extra context switches if the client wakes up, 4476 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4477 if (!step()) goto getNextBuffer_exit; 4478 ALOGV("stepServer recovered"); 4479 mStepServerFailed = false; 4480 } 4481 4482 // FIXME Same as above 4483 framesReady = cblk->framesReady(); 4484 4485 if (CC_LIKELY(framesReady)) { 4486 uint32_t s = cblk->server; 4487 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4488 4489 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4490 if (framesReq > framesReady) { 4491 framesReq = framesReady; 4492 } 4493 if (framesReq > bufferEnd - s) { 4494 framesReq = bufferEnd - s; 4495 } 4496 4497 buffer->raw = getBuffer(s, framesReq); 4498 if (buffer->raw == NULL) goto getNextBuffer_exit; 4499 4500 buffer->frameCount = framesReq; 4501 return NO_ERROR; 4502 } 4503 4504getNextBuffer_exit: 4505 buffer->raw = NULL; 4506 buffer->frameCount = 0; 4507 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4508 return NOT_ENOUGH_DATA; 4509} 4510 4511// Note that framesReady() takes a mutex on the control block using tryLock(). 4512// This could result in priority inversion if framesReady() is called by the normal mixer, 4513// as the normal mixer thread runs at lower 4514// priority than the client's callback thread: there is a short window within framesReady() 4515// during which the normal mixer could be preempted, and the client callback would block. 4516// Another problem can occur if framesReady() is called by the fast mixer: 4517// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4518// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4519size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4520 return mCblk->framesReady(); 4521} 4522 4523// Don't call for fast tracks; the framesReady() could result in priority inversion 4524bool AudioFlinger::PlaybackThread::Track::isReady() const { 4525 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4526 4527 if (framesReady() >= mCblk->frameCount || 4528 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4529 mFillingUpStatus = FS_FILLED; 4530 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4531 return true; 4532 } 4533 return false; 4534} 4535 4536status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4537 int triggerSession) 4538{ 4539 status_t status = NO_ERROR; 4540 ALOGV("start(%d), calling pid %d session %d", 4541 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4542 4543 sp<ThreadBase> thread = mThread.promote(); 4544 if (thread != 0) { 4545 Mutex::Autolock _l(thread->mLock); 4546 track_state state = mState; 4547 // here the track could be either new, or restarted 4548 // in both cases "unstop" the track 4549 if (mState == PAUSED) { 4550 mState = TrackBase::RESUMING; 4551 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4552 } else { 4553 mState = TrackBase::ACTIVE; 4554 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4555 } 4556 4557 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4558 thread->mLock.unlock(); 4559 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4560 thread->mLock.lock(); 4561 4562#ifdef ADD_BATTERY_DATA 4563 // to track the speaker usage 4564 if (status == NO_ERROR) { 4565 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4566 } 4567#endif 4568 } 4569 if (status == NO_ERROR) { 4570 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4571 playbackThread->addTrack_l(this); 4572 } else { 4573 mState = state; 4574 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4575 } 4576 } else { 4577 status = BAD_VALUE; 4578 } 4579 return status; 4580} 4581 4582void AudioFlinger::PlaybackThread::Track::stop() 4583{ 4584 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4585 sp<ThreadBase> thread = mThread.promote(); 4586 if (thread != 0) { 4587 Mutex::Autolock _l(thread->mLock); 4588 track_state state = mState; 4589 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4590 // If the track is not active (PAUSED and buffers full), flush buffers 4591 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4592 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4593 reset(); 4594 mState = STOPPED; 4595 } else if (!isFastTrack()) { 4596 mState = STOPPED; 4597 } else { 4598 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4599 // and then to STOPPED and reset() when presentation is complete 4600 mState = STOPPING_1; 4601 } 4602 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4603 } 4604 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4605 thread->mLock.unlock(); 4606 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4607 thread->mLock.lock(); 4608 4609#ifdef ADD_BATTERY_DATA 4610 // to track the speaker usage 4611 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4612#endif 4613 } 4614 } 4615} 4616 4617void AudioFlinger::PlaybackThread::Track::pause() 4618{ 4619 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4620 sp<ThreadBase> thread = mThread.promote(); 4621 if (thread != 0) { 4622 Mutex::Autolock _l(thread->mLock); 4623 if (mState == ACTIVE || mState == RESUMING) { 4624 mState = PAUSING; 4625 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4626 if (!isOutputTrack()) { 4627 thread->mLock.unlock(); 4628 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4629 thread->mLock.lock(); 4630 4631#ifdef ADD_BATTERY_DATA 4632 // to track the speaker usage 4633 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4634#endif 4635 } 4636 } 4637 } 4638} 4639 4640void AudioFlinger::PlaybackThread::Track::flush() 4641{ 4642 ALOGV("flush(%d)", mName); 4643 sp<ThreadBase> thread = mThread.promote(); 4644 if (thread != 0) { 4645 Mutex::Autolock _l(thread->mLock); 4646 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4647 mState != PAUSING) { 4648 return; 4649 } 4650 // No point remaining in PAUSED state after a flush => go to 4651 // FLUSHED state 4652 mState = FLUSHED; 4653 // do not reset the track if it is still in the process of being stopped or paused. 4654 // this will be done by prepareTracks_l() when the track is stopped. 4655 // prepareTracks_l() will see mState == FLUSHED, then 4656 // remove from active track list, reset(), and trigger presentation complete 4657 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4658 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4659 reset(); 4660 } 4661 } 4662} 4663 4664void AudioFlinger::PlaybackThread::Track::reset() 4665{ 4666 // Do not reset twice to avoid discarding data written just after a flush and before 4667 // the audioflinger thread detects the track is stopped. 4668 if (!mResetDone) { 4669 TrackBase::reset(); 4670 // Force underrun condition to avoid false underrun callback until first data is 4671 // written to buffer 4672 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4673 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4674 mFillingUpStatus = FS_FILLING; 4675 mResetDone = true; 4676 if (mState == FLUSHED) { 4677 mState = IDLE; 4678 } 4679 } 4680} 4681 4682void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4683{ 4684 mMute = muted; 4685} 4686 4687status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4688{ 4689 status_t status = DEAD_OBJECT; 4690 sp<ThreadBase> thread = mThread.promote(); 4691 if (thread != 0) { 4692 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4693 status = playbackThread->attachAuxEffect(this, EffectId); 4694 } 4695 return status; 4696} 4697 4698void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4699{ 4700 mAuxEffectId = EffectId; 4701 mAuxBuffer = buffer; 4702} 4703 4704bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4705 size_t audioHalFrames) 4706{ 4707 // a track is considered presented when the total number of frames written to audio HAL 4708 // corresponds to the number of frames written when presentationComplete() is called for the 4709 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4710 if (mPresentationCompleteFrames == 0) { 4711 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4712 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4713 mPresentationCompleteFrames, audioHalFrames); 4714 } 4715 if (framesWritten >= mPresentationCompleteFrames) { 4716 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4717 mSessionId, framesWritten); 4718 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4719 return true; 4720 } 4721 return false; 4722} 4723 4724void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4725{ 4726 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4727 if (mSyncEvents[i]->type() == type) { 4728 mSyncEvents[i]->trigger(); 4729 mSyncEvents.removeAt(i); 4730 i--; 4731 } 4732 } 4733} 4734 4735// implement VolumeBufferProvider interface 4736 4737uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4738{ 4739 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4740 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4741 uint32_t vlr = mCblk->getVolumeLR(); 4742 uint32_t vl = vlr & 0xFFFF; 4743 uint32_t vr = vlr >> 16; 4744 // track volumes come from shared memory, so can't be trusted and must be clamped 4745 if (vl > MAX_GAIN_INT) { 4746 vl = MAX_GAIN_INT; 4747 } 4748 if (vr > MAX_GAIN_INT) { 4749 vr = MAX_GAIN_INT; 4750 } 4751 // now apply the cached master volume and stream type volume; 4752 // this is trusted but lacks any synchronization or barrier so may be stale 4753 float v = mCachedVolume; 4754 vl *= v; 4755 vr *= v; 4756 // re-combine into U4.16 4757 vlr = (vr << 16) | (vl & 0xFFFF); 4758 // FIXME look at mute, pause, and stop flags 4759 return vlr; 4760} 4761 4762status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4763{ 4764 if (mState == TERMINATED || mState == PAUSED || 4765 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4766 (mState == STOPPED)))) { 4767 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4768 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4769 event->cancel(); 4770 return INVALID_OPERATION; 4771 } 4772 TrackBase::setSyncEvent(event); 4773 return NO_ERROR; 4774} 4775 4776// timed audio tracks 4777 4778sp<AudioFlinger::PlaybackThread::TimedTrack> 4779AudioFlinger::PlaybackThread::TimedTrack::create( 4780 PlaybackThread *thread, 4781 const sp<Client>& client, 4782 audio_stream_type_t streamType, 4783 uint32_t sampleRate, 4784 audio_format_t format, 4785 uint32_t channelMask, 4786 int frameCount, 4787 const sp<IMemory>& sharedBuffer, 4788 int sessionId) { 4789 if (!client->reserveTimedTrack()) 4790 return NULL; 4791 4792 return new TimedTrack( 4793 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4794 sharedBuffer, sessionId); 4795} 4796 4797AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4798 PlaybackThread *thread, 4799 const sp<Client>& client, 4800 audio_stream_type_t streamType, 4801 uint32_t sampleRate, 4802 audio_format_t format, 4803 uint32_t channelMask, 4804 int frameCount, 4805 const sp<IMemory>& sharedBuffer, 4806 int sessionId) 4807 : Track(thread, client, streamType, sampleRate, format, channelMask, 4808 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4809 mQueueHeadInFlight(false), 4810 mTrimQueueHeadOnRelease(false), 4811 mFramesPendingInQueue(0), 4812 mTimedSilenceBuffer(NULL), 4813 mTimedSilenceBufferSize(0), 4814 mTimedAudioOutputOnTime(false), 4815 mMediaTimeTransformValid(false) 4816{ 4817 LocalClock lc; 4818 mLocalTimeFreq = lc.getLocalFreq(); 4819 4820 mLocalTimeToSampleTransform.a_zero = 0; 4821 mLocalTimeToSampleTransform.b_zero = 0; 4822 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4823 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4824 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4825 &mLocalTimeToSampleTransform.a_to_b_denom); 4826 4827 mMediaTimeToSampleTransform.a_zero = 0; 4828 mMediaTimeToSampleTransform.b_zero = 0; 4829 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4830 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4831 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4832 &mMediaTimeToSampleTransform.a_to_b_denom); 4833} 4834 4835AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4836 mClient->releaseTimedTrack(); 4837 delete [] mTimedSilenceBuffer; 4838} 4839 4840status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4841 size_t size, sp<IMemory>* buffer) { 4842 4843 Mutex::Autolock _l(mTimedBufferQueueLock); 4844 4845 trimTimedBufferQueue_l(); 4846 4847 // lazily initialize the shared memory heap for timed buffers 4848 if (mTimedMemoryDealer == NULL) { 4849 const int kTimedBufferHeapSize = 512 << 10; 4850 4851 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4852 "AudioFlingerTimed"); 4853 if (mTimedMemoryDealer == NULL) 4854 return NO_MEMORY; 4855 } 4856 4857 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4858 if (newBuffer == NULL) { 4859 newBuffer = mTimedMemoryDealer->allocate(size); 4860 if (newBuffer == NULL) 4861 return NO_MEMORY; 4862 } 4863 4864 *buffer = newBuffer; 4865 return NO_ERROR; 4866} 4867 4868// caller must hold mTimedBufferQueueLock 4869void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4870 int64_t mediaTimeNow; 4871 { 4872 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4873 if (!mMediaTimeTransformValid) 4874 return; 4875 4876 int64_t targetTimeNow; 4877 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4878 ? mCCHelper.getCommonTime(&targetTimeNow) 4879 : mCCHelper.getLocalTime(&targetTimeNow); 4880 4881 if (OK != res) 4882 return; 4883 4884 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4885 &mediaTimeNow)) { 4886 return; 4887 } 4888 } 4889 4890 size_t trimEnd; 4891 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4892 int64_t bufEnd; 4893 4894 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4895 // We have a next buffer. Just use its PTS as the PTS of the frame 4896 // following the last frame in this buffer. If the stream is sparse 4897 // (ie, there are deliberate gaps left in the stream which should be 4898 // filled with silence by the TimedAudioTrack), then this can result 4899 // in one extra buffer being left un-trimmed when it could have 4900 // been. In general, this is not typical, and we would rather 4901 // optimized away the TS calculation below for the more common case 4902 // where PTSes are contiguous. 4903 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4904 } else { 4905 // We have no next buffer. Compute the PTS of the frame following 4906 // the last frame in this buffer by computing the duration of of 4907 // this frame in media time units and adding it to the PTS of the 4908 // buffer. 4909 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4910 / mCblk->frameSize; 4911 4912 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4913 &bufEnd)) { 4914 ALOGE("Failed to convert frame count of %lld to media time" 4915 " duration" " (scale factor %d/%u) in %s", 4916 frameCount, 4917 mMediaTimeToSampleTransform.a_to_b_numer, 4918 mMediaTimeToSampleTransform.a_to_b_denom, 4919 __PRETTY_FUNCTION__); 4920 break; 4921 } 4922 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4923 } 4924 4925 if (bufEnd > mediaTimeNow) 4926 break; 4927 4928 // Is the buffer we want to use in the middle of a mix operation right 4929 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4930 // from the mixer which should be coming back shortly. 4931 if (!trimEnd && mQueueHeadInFlight) { 4932 mTrimQueueHeadOnRelease = true; 4933 } 4934 } 4935 4936 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4937 if (trimStart < trimEnd) { 4938 // Update the bookkeeping for framesReady() 4939 for (size_t i = trimStart; i < trimEnd; ++i) { 4940 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4941 } 4942 4943 // Now actually remove the buffers from the queue. 4944 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4945 } 4946} 4947 4948void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4949 const char* logTag) { 4950 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4951 "%s called (reason \"%s\"), but timed buffer queue has no" 4952 " elements to trim.", __FUNCTION__, logTag); 4953 4954 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4955 mTimedBufferQueue.removeAt(0); 4956} 4957 4958void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4959 const TimedBuffer& buf, 4960 const char* logTag) { 4961 uint32_t bufBytes = buf.buffer()->size(); 4962 uint32_t consumedAlready = buf.position(); 4963 4964 ALOG_ASSERT(consumedAlready <= bufBytes, 4965 "Bad bookkeeping while updating frames pending. Timed buffer is" 4966 " only %u bytes long, but claims to have consumed %u" 4967 " bytes. (update reason: \"%s\")", 4968 bufBytes, consumedAlready, logTag); 4969 4970 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4971 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4972 "Bad bookkeeping while updating frames pending. Should have at" 4973 " least %u queued frames, but we think we have only %u. (update" 4974 " reason: \"%s\")", 4975 bufFrames, mFramesPendingInQueue, logTag); 4976 4977 mFramesPendingInQueue -= bufFrames; 4978} 4979 4980status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4981 const sp<IMemory>& buffer, int64_t pts) { 4982 4983 { 4984 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4985 if (!mMediaTimeTransformValid) 4986 return INVALID_OPERATION; 4987 } 4988 4989 Mutex::Autolock _l(mTimedBufferQueueLock); 4990 4991 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4992 mFramesPendingInQueue += bufFrames; 4993 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4994 4995 return NO_ERROR; 4996} 4997 4998status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4999 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5000 5001 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5002 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5003 target); 5004 5005 if (!(target == TimedAudioTrack::LOCAL_TIME || 5006 target == TimedAudioTrack::COMMON_TIME)) { 5007 return BAD_VALUE; 5008 } 5009 5010 Mutex::Autolock lock(mMediaTimeTransformLock); 5011 mMediaTimeTransform = xform; 5012 mMediaTimeTransformTarget = target; 5013 mMediaTimeTransformValid = true; 5014 5015 return NO_ERROR; 5016} 5017 5018#define min(a, b) ((a) < (b) ? (a) : (b)) 5019 5020// implementation of getNextBuffer for tracks whose buffers have timestamps 5021status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5022 AudioBufferProvider::Buffer* buffer, int64_t pts) 5023{ 5024 if (pts == AudioBufferProvider::kInvalidPTS) { 5025 buffer->raw = 0; 5026 buffer->frameCount = 0; 5027 mTimedAudioOutputOnTime = false; 5028 return INVALID_OPERATION; 5029 } 5030 5031 Mutex::Autolock _l(mTimedBufferQueueLock); 5032 5033 ALOG_ASSERT(!mQueueHeadInFlight, 5034 "getNextBuffer called without releaseBuffer!"); 5035 5036 while (true) { 5037 5038 // if we have no timed buffers, then fail 5039 if (mTimedBufferQueue.isEmpty()) { 5040 buffer->raw = 0; 5041 buffer->frameCount = 0; 5042 return NOT_ENOUGH_DATA; 5043 } 5044 5045 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5046 5047 // calculate the PTS of the head of the timed buffer queue expressed in 5048 // local time 5049 int64_t headLocalPTS; 5050 { 5051 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5052 5053 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5054 5055 if (mMediaTimeTransform.a_to_b_denom == 0) { 5056 // the transform represents a pause, so yield silence 5057 timedYieldSilence_l(buffer->frameCount, buffer); 5058 return NO_ERROR; 5059 } 5060 5061 int64_t transformedPTS; 5062 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5063 &transformedPTS)) { 5064 // the transform failed. this shouldn't happen, but if it does 5065 // then just drop this buffer 5066 ALOGW("timedGetNextBuffer transform failed"); 5067 buffer->raw = 0; 5068 buffer->frameCount = 0; 5069 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5070 return NO_ERROR; 5071 } 5072 5073 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5074 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5075 &headLocalPTS)) { 5076 buffer->raw = 0; 5077 buffer->frameCount = 0; 5078 return INVALID_OPERATION; 5079 } 5080 } else { 5081 headLocalPTS = transformedPTS; 5082 } 5083 } 5084 5085 // adjust the head buffer's PTS to reflect the portion of the head buffer 5086 // that has already been consumed 5087 int64_t effectivePTS = headLocalPTS + 5088 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5089 5090 // Calculate the delta in samples between the head of the input buffer 5091 // queue and the start of the next output buffer that will be written. 5092 // If the transformation fails because of over or underflow, it means 5093 // that the sample's position in the output stream is so far out of 5094 // whack that it should just be dropped. 5095 int64_t sampleDelta; 5096 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5097 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5098 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5099 " mix"); 5100 continue; 5101 } 5102 if (!mLocalTimeToSampleTransform.doForwardTransform( 5103 (effectivePTS - pts) << 32, &sampleDelta)) { 5104 ALOGV("*** too late during sample rate transform: dropped buffer"); 5105 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5106 continue; 5107 } 5108 5109 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5110 " sampleDelta=[%d.%08x]", 5111 head.pts(), head.position(), pts, 5112 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5113 + (sampleDelta >> 32)), 5114 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5115 5116 // if the delta between the ideal placement for the next input sample and 5117 // the current output position is within this threshold, then we will 5118 // concatenate the next input samples to the previous output 5119 const int64_t kSampleContinuityThreshold = 5120 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5121 5122 // if this is the first buffer of audio that we're emitting from this track 5123 // then it should be almost exactly on time. 5124 const int64_t kSampleStartupThreshold = 1LL << 32; 5125 5126 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5127 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5128 // the next input is close enough to being on time, so concatenate it 5129 // with the last output 5130 timedYieldSamples_l(buffer); 5131 5132 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5133 head.position(), buffer->frameCount); 5134 return NO_ERROR; 5135 } 5136 5137 // Looks like our output is not on time. Reset our on timed status. 5138 // Next time we mix samples from our input queue, then should be within 5139 // the StartupThreshold. 5140 mTimedAudioOutputOnTime = false; 5141 if (sampleDelta > 0) { 5142 // the gap between the current output position and the proper start of 5143 // the next input sample is too big, so fill it with silence 5144 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5145 5146 timedYieldSilence_l(framesUntilNextInput, buffer); 5147 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5148 return NO_ERROR; 5149 } else { 5150 // the next input sample is late 5151 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5152 size_t onTimeSamplePosition = 5153 head.position() + lateFrames * mCblk->frameSize; 5154 5155 if (onTimeSamplePosition > head.buffer()->size()) { 5156 // all the remaining samples in the head are too late, so 5157 // drop it and move on 5158 ALOGV("*** too late: dropped buffer"); 5159 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5160 continue; 5161 } else { 5162 // skip over the late samples 5163 head.setPosition(onTimeSamplePosition); 5164 5165 // yield the available samples 5166 timedYieldSamples_l(buffer); 5167 5168 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5169 return NO_ERROR; 5170 } 5171 } 5172 } 5173} 5174 5175// Yield samples from the timed buffer queue head up to the given output 5176// buffer's capacity. 5177// 5178// Caller must hold mTimedBufferQueueLock 5179void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5180 AudioBufferProvider::Buffer* buffer) { 5181 5182 const TimedBuffer& head = mTimedBufferQueue[0]; 5183 5184 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5185 head.position()); 5186 5187 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5188 mCblk->frameSize); 5189 size_t framesRequested = buffer->frameCount; 5190 buffer->frameCount = min(framesLeftInHead, framesRequested); 5191 5192 mQueueHeadInFlight = true; 5193 mTimedAudioOutputOnTime = true; 5194} 5195 5196// Yield samples of silence up to the given output buffer's capacity 5197// 5198// Caller must hold mTimedBufferQueueLock 5199void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5200 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5201 5202 // lazily allocate a buffer filled with silence 5203 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5204 delete [] mTimedSilenceBuffer; 5205 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5206 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5207 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5208 } 5209 5210 buffer->raw = mTimedSilenceBuffer; 5211 size_t framesRequested = buffer->frameCount; 5212 buffer->frameCount = min(numFrames, framesRequested); 5213 5214 mTimedAudioOutputOnTime = false; 5215} 5216 5217// AudioBufferProvider interface 5218void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5219 AudioBufferProvider::Buffer* buffer) { 5220 5221 Mutex::Autolock _l(mTimedBufferQueueLock); 5222 5223 // If the buffer which was just released is part of the buffer at the head 5224 // of the queue, be sure to update the amt of the buffer which has been 5225 // consumed. If the buffer being returned is not part of the head of the 5226 // queue, its either because the buffer is part of the silence buffer, or 5227 // because the head of the timed queue was trimmed after the mixer called 5228 // getNextBuffer but before the mixer called releaseBuffer. 5229 if (buffer->raw == mTimedSilenceBuffer) { 5230 ALOG_ASSERT(!mQueueHeadInFlight, 5231 "Queue head in flight during release of silence buffer!"); 5232 goto done; 5233 } 5234 5235 ALOG_ASSERT(mQueueHeadInFlight, 5236 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5237 " head in flight."); 5238 5239 if (mTimedBufferQueue.size()) { 5240 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5241 5242 void* start = head.buffer()->pointer(); 5243 void* end = reinterpret_cast<void*>( 5244 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5245 + head.buffer()->size()); 5246 5247 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5248 "released buffer not within the head of the timed buffer" 5249 " queue; qHead = [%p, %p], released buffer = %p", 5250 start, end, buffer->raw); 5251 5252 head.setPosition(head.position() + 5253 (buffer->frameCount * mCblk->frameSize)); 5254 mQueueHeadInFlight = false; 5255 5256 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5257 "Bad bookkeeping during releaseBuffer! Should have at" 5258 " least %u queued frames, but we think we have only %u", 5259 buffer->frameCount, mFramesPendingInQueue); 5260 5261 mFramesPendingInQueue -= buffer->frameCount; 5262 5263 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5264 || mTrimQueueHeadOnRelease) { 5265 trimTimedBufferQueueHead_l("releaseBuffer"); 5266 mTrimQueueHeadOnRelease = false; 5267 } 5268 } else { 5269 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5270 " buffers in the timed buffer queue"); 5271 } 5272 5273done: 5274 buffer->raw = 0; 5275 buffer->frameCount = 0; 5276} 5277 5278size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5279 Mutex::Autolock _l(mTimedBufferQueueLock); 5280 return mFramesPendingInQueue; 5281} 5282 5283AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5284 : mPTS(0), mPosition(0) {} 5285 5286AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5287 const sp<IMemory>& buffer, int64_t pts) 5288 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5289 5290// ---------------------------------------------------------------------------- 5291 5292// RecordTrack constructor must be called with AudioFlinger::mLock held 5293AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5294 RecordThread *thread, 5295 const sp<Client>& client, 5296 uint32_t sampleRate, 5297 audio_format_t format, 5298 uint32_t channelMask, 5299 int frameCount, 5300 int sessionId) 5301 : TrackBase(thread, client, sampleRate, format, 5302 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5303 mOverflow(false) 5304{ 5305 if (mCblk != NULL) { 5306 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5307 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5308 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5309 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5310 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5311 } else { 5312 mCblk->frameSize = sizeof(int8_t); 5313 } 5314 } 5315} 5316 5317AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5318{ 5319 sp<ThreadBase> thread = mThread.promote(); 5320 if (thread != 0) { 5321 AudioSystem::releaseInput(thread->id()); 5322 } 5323} 5324 5325// AudioBufferProvider interface 5326status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5327{ 5328 audio_track_cblk_t* cblk = this->cblk(); 5329 uint32_t framesAvail; 5330 uint32_t framesReq = buffer->frameCount; 5331 5332 // Check if last stepServer failed, try to step now 5333 if (mStepServerFailed) { 5334 if (!step()) goto getNextBuffer_exit; 5335 ALOGV("stepServer recovered"); 5336 mStepServerFailed = false; 5337 } 5338 5339 framesAvail = cblk->framesAvailable_l(); 5340 5341 if (CC_LIKELY(framesAvail)) { 5342 uint32_t s = cblk->server; 5343 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5344 5345 if (framesReq > framesAvail) { 5346 framesReq = framesAvail; 5347 } 5348 if (framesReq > bufferEnd - s) { 5349 framesReq = bufferEnd - s; 5350 } 5351 5352 buffer->raw = getBuffer(s, framesReq); 5353 if (buffer->raw == NULL) goto getNextBuffer_exit; 5354 5355 buffer->frameCount = framesReq; 5356 return NO_ERROR; 5357 } 5358 5359getNextBuffer_exit: 5360 buffer->raw = NULL; 5361 buffer->frameCount = 0; 5362 return NOT_ENOUGH_DATA; 5363} 5364 5365status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5366 int triggerSession) 5367{ 5368 sp<ThreadBase> thread = mThread.promote(); 5369 if (thread != 0) { 5370 RecordThread *recordThread = (RecordThread *)thread.get(); 5371 return recordThread->start(this, event, triggerSession); 5372 } else { 5373 return BAD_VALUE; 5374 } 5375} 5376 5377void AudioFlinger::RecordThread::RecordTrack::stop() 5378{ 5379 sp<ThreadBase> thread = mThread.promote(); 5380 if (thread != 0) { 5381 RecordThread *recordThread = (RecordThread *)thread.get(); 5382 recordThread->stop(this); 5383 TrackBase::reset(); 5384 // Force overrun condition to avoid false overrun callback until first data is 5385 // read from buffer 5386 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5387 } 5388} 5389 5390void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5391{ 5392 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5393 (mClient == 0) ? getpid_cached : mClient->pid(), 5394 mFormat, 5395 mChannelMask, 5396 mSessionId, 5397 mFrameCount, 5398 mState, 5399 mCblk->sampleRate, 5400 mCblk->server, 5401 mCblk->user); 5402} 5403 5404 5405// ---------------------------------------------------------------------------- 5406 5407AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5408 PlaybackThread *playbackThread, 5409 DuplicatingThread *sourceThread, 5410 uint32_t sampleRate, 5411 audio_format_t format, 5412 uint32_t channelMask, 5413 int frameCount) 5414 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5415 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5416 mActive(false), mSourceThread(sourceThread) 5417{ 5418 5419 if (mCblk != NULL) { 5420 mCblk->flags |= CBLK_DIRECTION_OUT; 5421 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5422 mOutBuffer.frameCount = 0; 5423 playbackThread->mTracks.add(this); 5424 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5425 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5426 mCblk, mBuffer, mCblk->buffers, 5427 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5428 } else { 5429 ALOGW("Error creating output track on thread %p", playbackThread); 5430 } 5431} 5432 5433AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5434{ 5435 clearBufferQueue(); 5436} 5437 5438status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5439 int triggerSession) 5440{ 5441 status_t status = Track::start(event, triggerSession); 5442 if (status != NO_ERROR) { 5443 return status; 5444 } 5445 5446 mActive = true; 5447 mRetryCount = 127; 5448 return status; 5449} 5450 5451void AudioFlinger::PlaybackThread::OutputTrack::stop() 5452{ 5453 Track::stop(); 5454 clearBufferQueue(); 5455 mOutBuffer.frameCount = 0; 5456 mActive = false; 5457} 5458 5459bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5460{ 5461 Buffer *pInBuffer; 5462 Buffer inBuffer; 5463 uint32_t channelCount = mChannelCount; 5464 bool outputBufferFull = false; 5465 inBuffer.frameCount = frames; 5466 inBuffer.i16 = data; 5467 5468 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5469 5470 if (!mActive && frames != 0) { 5471 start(); 5472 sp<ThreadBase> thread = mThread.promote(); 5473 if (thread != 0) { 5474 MixerThread *mixerThread = (MixerThread *)thread.get(); 5475 if (mCblk->frameCount > frames){ 5476 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5477 uint32_t startFrames = (mCblk->frameCount - frames); 5478 pInBuffer = new Buffer; 5479 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5480 pInBuffer->frameCount = startFrames; 5481 pInBuffer->i16 = pInBuffer->mBuffer; 5482 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5483 mBufferQueue.add(pInBuffer); 5484 } else { 5485 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5486 } 5487 } 5488 } 5489 } 5490 5491 while (waitTimeLeftMs) { 5492 // First write pending buffers, then new data 5493 if (mBufferQueue.size()) { 5494 pInBuffer = mBufferQueue.itemAt(0); 5495 } else { 5496 pInBuffer = &inBuffer; 5497 } 5498 5499 if (pInBuffer->frameCount == 0) { 5500 break; 5501 } 5502 5503 if (mOutBuffer.frameCount == 0) { 5504 mOutBuffer.frameCount = pInBuffer->frameCount; 5505 nsecs_t startTime = systemTime(); 5506 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5507 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5508 outputBufferFull = true; 5509 break; 5510 } 5511 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5512 if (waitTimeLeftMs >= waitTimeMs) { 5513 waitTimeLeftMs -= waitTimeMs; 5514 } else { 5515 waitTimeLeftMs = 0; 5516 } 5517 } 5518 5519 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5520 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5521 mCblk->stepUser(outFrames); 5522 pInBuffer->frameCount -= outFrames; 5523 pInBuffer->i16 += outFrames * channelCount; 5524 mOutBuffer.frameCount -= outFrames; 5525 mOutBuffer.i16 += outFrames * channelCount; 5526 5527 if (pInBuffer->frameCount == 0) { 5528 if (mBufferQueue.size()) { 5529 mBufferQueue.removeAt(0); 5530 delete [] pInBuffer->mBuffer; 5531 delete pInBuffer; 5532 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5533 } else { 5534 break; 5535 } 5536 } 5537 } 5538 5539 // If we could not write all frames, allocate a buffer and queue it for next time. 5540 if (inBuffer.frameCount) { 5541 sp<ThreadBase> thread = mThread.promote(); 5542 if (thread != 0 && !thread->standby()) { 5543 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5544 pInBuffer = new Buffer; 5545 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5546 pInBuffer->frameCount = inBuffer.frameCount; 5547 pInBuffer->i16 = pInBuffer->mBuffer; 5548 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5549 mBufferQueue.add(pInBuffer); 5550 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5551 } else { 5552 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5553 } 5554 } 5555 } 5556 5557 // Calling write() with a 0 length buffer, means that no more data will be written: 5558 // If no more buffers are pending, fill output track buffer to make sure it is started 5559 // by output mixer. 5560 if (frames == 0 && mBufferQueue.size() == 0) { 5561 if (mCblk->user < mCblk->frameCount) { 5562 frames = mCblk->frameCount - mCblk->user; 5563 pInBuffer = new Buffer; 5564 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5565 pInBuffer->frameCount = frames; 5566 pInBuffer->i16 = pInBuffer->mBuffer; 5567 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5568 mBufferQueue.add(pInBuffer); 5569 } else if (mActive) { 5570 stop(); 5571 } 5572 } 5573 5574 return outputBufferFull; 5575} 5576 5577status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5578{ 5579 int active; 5580 status_t result; 5581 audio_track_cblk_t* cblk = mCblk; 5582 uint32_t framesReq = buffer->frameCount; 5583 5584// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5585 buffer->frameCount = 0; 5586 5587 uint32_t framesAvail = cblk->framesAvailable(); 5588 5589 5590 if (framesAvail == 0) { 5591 Mutex::Autolock _l(cblk->lock); 5592 goto start_loop_here; 5593 while (framesAvail == 0) { 5594 active = mActive; 5595 if (CC_UNLIKELY(!active)) { 5596 ALOGV("Not active and NO_MORE_BUFFERS"); 5597 return NO_MORE_BUFFERS; 5598 } 5599 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5600 if (result != NO_ERROR) { 5601 return NO_MORE_BUFFERS; 5602 } 5603 // read the server count again 5604 start_loop_here: 5605 framesAvail = cblk->framesAvailable_l(); 5606 } 5607 } 5608 5609// if (framesAvail < framesReq) { 5610// return NO_MORE_BUFFERS; 5611// } 5612 5613 if (framesReq > framesAvail) { 5614 framesReq = framesAvail; 5615 } 5616 5617 uint32_t u = cblk->user; 5618 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5619 5620 if (framesReq > bufferEnd - u) { 5621 framesReq = bufferEnd - u; 5622 } 5623 5624 buffer->frameCount = framesReq; 5625 buffer->raw = (void *)cblk->buffer(u); 5626 return NO_ERROR; 5627} 5628 5629 5630void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5631{ 5632 size_t size = mBufferQueue.size(); 5633 5634 for (size_t i = 0; i < size; i++) { 5635 Buffer *pBuffer = mBufferQueue.itemAt(i); 5636 delete [] pBuffer->mBuffer; 5637 delete pBuffer; 5638 } 5639 mBufferQueue.clear(); 5640} 5641 5642// ---------------------------------------------------------------------------- 5643 5644AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5645 : RefBase(), 5646 mAudioFlinger(audioFlinger), 5647 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5648 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5649 mPid(pid), 5650 mTimedTrackCount(0) 5651{ 5652 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5653} 5654 5655// Client destructor must be called with AudioFlinger::mLock held 5656AudioFlinger::Client::~Client() 5657{ 5658 mAudioFlinger->removeClient_l(mPid); 5659} 5660 5661sp<MemoryDealer> AudioFlinger::Client::heap() const 5662{ 5663 return mMemoryDealer; 5664} 5665 5666// Reserve one of the limited slots for a timed audio track associated 5667// with this client 5668bool AudioFlinger::Client::reserveTimedTrack() 5669{ 5670 const int kMaxTimedTracksPerClient = 4; 5671 5672 Mutex::Autolock _l(mTimedTrackLock); 5673 5674 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5675 ALOGW("can not create timed track - pid %d has exceeded the limit", 5676 mPid); 5677 return false; 5678 } 5679 5680 mTimedTrackCount++; 5681 return true; 5682} 5683 5684// Release a slot for a timed audio track 5685void AudioFlinger::Client::releaseTimedTrack() 5686{ 5687 Mutex::Autolock _l(mTimedTrackLock); 5688 mTimedTrackCount--; 5689} 5690 5691// ---------------------------------------------------------------------------- 5692 5693AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5694 const sp<IAudioFlingerClient>& client, 5695 pid_t pid) 5696 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5697{ 5698} 5699 5700AudioFlinger::NotificationClient::~NotificationClient() 5701{ 5702} 5703 5704void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5705{ 5706 sp<NotificationClient> keep(this); 5707 mAudioFlinger->removeNotificationClient(mPid); 5708} 5709 5710// ---------------------------------------------------------------------------- 5711 5712AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5713 : BnAudioTrack(), 5714 mTrack(track) 5715{ 5716} 5717 5718AudioFlinger::TrackHandle::~TrackHandle() { 5719 // just stop the track on deletion, associated resources 5720 // will be freed from the main thread once all pending buffers have 5721 // been played. Unless it's not in the active track list, in which 5722 // case we free everything now... 5723 mTrack->destroy(); 5724} 5725 5726sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5727 return mTrack->getCblk(); 5728} 5729 5730status_t AudioFlinger::TrackHandle::start() { 5731 return mTrack->start(); 5732} 5733 5734void AudioFlinger::TrackHandle::stop() { 5735 mTrack->stop(); 5736} 5737 5738void AudioFlinger::TrackHandle::flush() { 5739 mTrack->flush(); 5740} 5741 5742void AudioFlinger::TrackHandle::mute(bool e) { 5743 mTrack->mute(e); 5744} 5745 5746void AudioFlinger::TrackHandle::pause() { 5747 mTrack->pause(); 5748} 5749 5750status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5751{ 5752 return mTrack->attachAuxEffect(EffectId); 5753} 5754 5755status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5756 sp<IMemory>* buffer) { 5757 if (!mTrack->isTimedTrack()) 5758 return INVALID_OPERATION; 5759 5760 PlaybackThread::TimedTrack* tt = 5761 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5762 return tt->allocateTimedBuffer(size, buffer); 5763} 5764 5765status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5766 int64_t pts) { 5767 if (!mTrack->isTimedTrack()) 5768 return INVALID_OPERATION; 5769 5770 PlaybackThread::TimedTrack* tt = 5771 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5772 return tt->queueTimedBuffer(buffer, pts); 5773} 5774 5775status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5776 const LinearTransform& xform, int target) { 5777 5778 if (!mTrack->isTimedTrack()) 5779 return INVALID_OPERATION; 5780 5781 PlaybackThread::TimedTrack* tt = 5782 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5783 return tt->setMediaTimeTransform( 5784 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5785} 5786 5787status_t AudioFlinger::TrackHandle::onTransact( 5788 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5789{ 5790 return BnAudioTrack::onTransact(code, data, reply, flags); 5791} 5792 5793// ---------------------------------------------------------------------------- 5794 5795sp<IAudioRecord> AudioFlinger::openRecord( 5796 pid_t pid, 5797 audio_io_handle_t input, 5798 uint32_t sampleRate, 5799 audio_format_t format, 5800 uint32_t channelMask, 5801 int frameCount, 5802 IAudioFlinger::track_flags_t flags, 5803 int *sessionId, 5804 status_t *status) 5805{ 5806 sp<RecordThread::RecordTrack> recordTrack; 5807 sp<RecordHandle> recordHandle; 5808 sp<Client> client; 5809 status_t lStatus; 5810 RecordThread *thread; 5811 size_t inFrameCount; 5812 int lSessionId; 5813 5814 // check calling permissions 5815 if (!recordingAllowed()) { 5816 lStatus = PERMISSION_DENIED; 5817 goto Exit; 5818 } 5819 5820 // add client to list 5821 { // scope for mLock 5822 Mutex::Autolock _l(mLock); 5823 thread = checkRecordThread_l(input); 5824 if (thread == NULL) { 5825 lStatus = BAD_VALUE; 5826 goto Exit; 5827 } 5828 5829 client = registerPid_l(pid); 5830 5831 // If no audio session id is provided, create one here 5832 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5833 lSessionId = *sessionId; 5834 } else { 5835 lSessionId = nextUniqueId(); 5836 if (sessionId != NULL) { 5837 *sessionId = lSessionId; 5838 } 5839 } 5840 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5841 recordTrack = thread->createRecordTrack_l(client, 5842 sampleRate, 5843 format, 5844 channelMask, 5845 frameCount, 5846 lSessionId, 5847 &lStatus); 5848 } 5849 if (lStatus != NO_ERROR) { 5850 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5851 // destructor is called by the TrackBase destructor with mLock held 5852 client.clear(); 5853 recordTrack.clear(); 5854 goto Exit; 5855 } 5856 5857 // return to handle to client 5858 recordHandle = new RecordHandle(recordTrack); 5859 lStatus = NO_ERROR; 5860 5861Exit: 5862 if (status) { 5863 *status = lStatus; 5864 } 5865 return recordHandle; 5866} 5867 5868// ---------------------------------------------------------------------------- 5869 5870AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5871 : BnAudioRecord(), 5872 mRecordTrack(recordTrack) 5873{ 5874} 5875 5876AudioFlinger::RecordHandle::~RecordHandle() { 5877 stop(); 5878} 5879 5880sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5881 return mRecordTrack->getCblk(); 5882} 5883 5884status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5885 ALOGV("RecordHandle::start()"); 5886 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5887} 5888 5889void AudioFlinger::RecordHandle::stop() { 5890 ALOGV("RecordHandle::stop()"); 5891 mRecordTrack->stop(); 5892} 5893 5894status_t AudioFlinger::RecordHandle::onTransact( 5895 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5896{ 5897 return BnAudioRecord::onTransact(code, data, reply, flags); 5898} 5899 5900// ---------------------------------------------------------------------------- 5901 5902AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5903 AudioStreamIn *input, 5904 uint32_t sampleRate, 5905 uint32_t channels, 5906 audio_io_handle_t id, 5907 uint32_t device) : 5908 ThreadBase(audioFlinger, id, device, RECORD), 5909 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5910 // mRsmpInIndex and mInputBytes set by readInputParameters() 5911 mReqChannelCount(popcount(channels)), 5912 mReqSampleRate(sampleRate) 5913 // mBytesRead is only meaningful while active, and so is cleared in start() 5914 // (but might be better to also clear here for dump?) 5915{ 5916 snprintf(mName, kNameLength, "AudioIn_%X", id); 5917 5918 readInputParameters(); 5919} 5920 5921 5922AudioFlinger::RecordThread::~RecordThread() 5923{ 5924 delete[] mRsmpInBuffer; 5925 delete mResampler; 5926 delete[] mRsmpOutBuffer; 5927} 5928 5929void AudioFlinger::RecordThread::onFirstRef() 5930{ 5931 run(mName, PRIORITY_URGENT_AUDIO); 5932} 5933 5934status_t AudioFlinger::RecordThread::readyToRun() 5935{ 5936 status_t status = initCheck(); 5937 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5938 return status; 5939} 5940 5941bool AudioFlinger::RecordThread::threadLoop() 5942{ 5943 AudioBufferProvider::Buffer buffer; 5944 sp<RecordTrack> activeTrack; 5945 Vector< sp<EffectChain> > effectChains; 5946 5947 nsecs_t lastWarning = 0; 5948 5949 acquireWakeLock(); 5950 5951 // start recording 5952 while (!exitPending()) { 5953 5954 processConfigEvents(); 5955 5956 { // scope for mLock 5957 Mutex::Autolock _l(mLock); 5958 checkForNewParameters_l(); 5959 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5960 if (!mStandby) { 5961 mInput->stream->common.standby(&mInput->stream->common); 5962 mStandby = true; 5963 } 5964 5965 if (exitPending()) break; 5966 5967 releaseWakeLock_l(); 5968 ALOGV("RecordThread: loop stopping"); 5969 // go to sleep 5970 mWaitWorkCV.wait(mLock); 5971 ALOGV("RecordThread: loop starting"); 5972 acquireWakeLock_l(); 5973 continue; 5974 } 5975 if (mActiveTrack != 0) { 5976 if (mActiveTrack->mState == TrackBase::PAUSING) { 5977 if (!mStandby) { 5978 mInput->stream->common.standby(&mInput->stream->common); 5979 mStandby = true; 5980 } 5981 mActiveTrack.clear(); 5982 mStartStopCond.broadcast(); 5983 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5984 if (mReqChannelCount != mActiveTrack->channelCount()) { 5985 mActiveTrack.clear(); 5986 mStartStopCond.broadcast(); 5987 } else if (mBytesRead != 0) { 5988 // record start succeeds only if first read from audio input 5989 // succeeds 5990 if (mBytesRead > 0) { 5991 mActiveTrack->mState = TrackBase::ACTIVE; 5992 } else { 5993 mActiveTrack.clear(); 5994 } 5995 mStartStopCond.broadcast(); 5996 } 5997 mStandby = false; 5998 } 5999 } 6000 lockEffectChains_l(effectChains); 6001 } 6002 6003 if (mActiveTrack != 0) { 6004 if (mActiveTrack->mState != TrackBase::ACTIVE && 6005 mActiveTrack->mState != TrackBase::RESUMING) { 6006 unlockEffectChains(effectChains); 6007 usleep(kRecordThreadSleepUs); 6008 continue; 6009 } 6010 for (size_t i = 0; i < effectChains.size(); i ++) { 6011 effectChains[i]->process_l(); 6012 } 6013 6014 buffer.frameCount = mFrameCount; 6015 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6016 size_t framesOut = buffer.frameCount; 6017 if (mResampler == NULL) { 6018 // no resampling 6019 while (framesOut) { 6020 size_t framesIn = mFrameCount - mRsmpInIndex; 6021 if (framesIn) { 6022 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6023 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6024 if (framesIn > framesOut) 6025 framesIn = framesOut; 6026 mRsmpInIndex += framesIn; 6027 framesOut -= framesIn; 6028 if ((int)mChannelCount == mReqChannelCount || 6029 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6030 memcpy(dst, src, framesIn * mFrameSize); 6031 } else { 6032 int16_t *src16 = (int16_t *)src; 6033 int16_t *dst16 = (int16_t *)dst; 6034 if (mChannelCount == 1) { 6035 while (framesIn--) { 6036 *dst16++ = *src16; 6037 *dst16++ = *src16++; 6038 } 6039 } else { 6040 while (framesIn--) { 6041 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6042 src16 += 2; 6043 } 6044 } 6045 } 6046 } 6047 if (framesOut && mFrameCount == mRsmpInIndex) { 6048 if (framesOut == mFrameCount && 6049 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6050 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6051 framesOut = 0; 6052 } else { 6053 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6054 mRsmpInIndex = 0; 6055 } 6056 if (mBytesRead < 0) { 6057 ALOGE("Error reading audio input"); 6058 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6059 // Force input into standby so that it tries to 6060 // recover at next read attempt 6061 mInput->stream->common.standby(&mInput->stream->common); 6062 usleep(kRecordThreadSleepUs); 6063 } 6064 mRsmpInIndex = mFrameCount; 6065 framesOut = 0; 6066 buffer.frameCount = 0; 6067 } 6068 } 6069 } 6070 } else { 6071 // resampling 6072 6073 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6074 // alter output frame count as if we were expecting stereo samples 6075 if (mChannelCount == 1 && mReqChannelCount == 1) { 6076 framesOut >>= 1; 6077 } 6078 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6079 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6080 // are 32 bit aligned which should be always true. 6081 if (mChannelCount == 2 && mReqChannelCount == 1) { 6082 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6083 // the resampler always outputs stereo samples: do post stereo to mono conversion 6084 int16_t *src = (int16_t *)mRsmpOutBuffer; 6085 int16_t *dst = buffer.i16; 6086 while (framesOut--) { 6087 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6088 src += 2; 6089 } 6090 } else { 6091 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6092 } 6093 6094 } 6095 if (mFramestoDrop == 0) { 6096 mActiveTrack->releaseBuffer(&buffer); 6097 } else { 6098 if (mFramestoDrop > 0) { 6099 mFramestoDrop -= buffer.frameCount; 6100 if (mFramestoDrop <= 0) { 6101 clearSyncStartEvent(); 6102 } 6103 } else { 6104 mFramestoDrop += buffer.frameCount; 6105 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6106 mSyncStartEvent->isCancelled()) { 6107 ALOGW("Synced record %s, session %d, trigger session %d", 6108 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6109 mActiveTrack->sessionId(), 6110 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6111 clearSyncStartEvent(); 6112 } 6113 } 6114 } 6115 mActiveTrack->overflow(); 6116 } 6117 // client isn't retrieving buffers fast enough 6118 else { 6119 if (!mActiveTrack->setOverflow()) { 6120 nsecs_t now = systemTime(); 6121 if ((now - lastWarning) > kWarningThrottleNs) { 6122 ALOGW("RecordThread: buffer overflow"); 6123 lastWarning = now; 6124 } 6125 } 6126 // Release the processor for a while before asking for a new buffer. 6127 // This will give the application more chance to read from the buffer and 6128 // clear the overflow. 6129 usleep(kRecordThreadSleepUs); 6130 } 6131 } 6132 // enable changes in effect chain 6133 unlockEffectChains(effectChains); 6134 effectChains.clear(); 6135 } 6136 6137 if (!mStandby) { 6138 mInput->stream->common.standby(&mInput->stream->common); 6139 } 6140 mActiveTrack.clear(); 6141 6142 mStartStopCond.broadcast(); 6143 6144 releaseWakeLock(); 6145 6146 ALOGV("RecordThread %p exiting", this); 6147 return false; 6148} 6149 6150 6151sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6152 const sp<AudioFlinger::Client>& client, 6153 uint32_t sampleRate, 6154 audio_format_t format, 6155 int channelMask, 6156 int frameCount, 6157 int sessionId, 6158 status_t *status) 6159{ 6160 sp<RecordTrack> track; 6161 status_t lStatus; 6162 6163 lStatus = initCheck(); 6164 if (lStatus != NO_ERROR) { 6165 ALOGE("Audio driver not initialized."); 6166 goto Exit; 6167 } 6168 6169 { // scope for mLock 6170 Mutex::Autolock _l(mLock); 6171 6172 track = new RecordTrack(this, client, sampleRate, 6173 format, channelMask, frameCount, sessionId); 6174 6175 if (track->getCblk() == 0) { 6176 lStatus = NO_MEMORY; 6177 goto Exit; 6178 } 6179 6180 mTrack = track.get(); 6181 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6182 bool suspend = audio_is_bluetooth_sco_device( 6183 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6184 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6185 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6186 } 6187 lStatus = NO_ERROR; 6188 6189Exit: 6190 if (status) { 6191 *status = lStatus; 6192 } 6193 return track; 6194} 6195 6196status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6197 AudioSystem::sync_event_t event, 6198 int triggerSession) 6199{ 6200 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6201 sp<ThreadBase> strongMe = this; 6202 status_t status = NO_ERROR; 6203 6204 if (event == AudioSystem::SYNC_EVENT_NONE) { 6205 clearSyncStartEvent(); 6206 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6207 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6208 triggerSession, 6209 recordTrack->sessionId(), 6210 syncStartEventCallback, 6211 this); 6212 // Sync event can be cancelled by the trigger session if the track is not in a 6213 // compatible state in which case we start record immediately 6214 if (mSyncStartEvent->isCancelled()) { 6215 clearSyncStartEvent(); 6216 } else { 6217 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6218 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6219 } 6220 } 6221 6222 { 6223 AutoMutex lock(mLock); 6224 if (mActiveTrack != 0) { 6225 if (recordTrack != mActiveTrack.get()) { 6226 status = -EBUSY; 6227 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6228 mActiveTrack->mState = TrackBase::ACTIVE; 6229 } 6230 return status; 6231 } 6232 6233 recordTrack->mState = TrackBase::IDLE; 6234 mActiveTrack = recordTrack; 6235 mLock.unlock(); 6236 status_t status = AudioSystem::startInput(mId); 6237 mLock.lock(); 6238 if (status != NO_ERROR) { 6239 mActiveTrack.clear(); 6240 clearSyncStartEvent(); 6241 return status; 6242 } 6243 mRsmpInIndex = mFrameCount; 6244 mBytesRead = 0; 6245 if (mResampler != NULL) { 6246 mResampler->reset(); 6247 } 6248 mActiveTrack->mState = TrackBase::RESUMING; 6249 // signal thread to start 6250 ALOGV("Signal record thread"); 6251 mWaitWorkCV.signal(); 6252 // do not wait for mStartStopCond if exiting 6253 if (exitPending()) { 6254 mActiveTrack.clear(); 6255 status = INVALID_OPERATION; 6256 goto startError; 6257 } 6258 mStartStopCond.wait(mLock); 6259 if (mActiveTrack == 0) { 6260 ALOGV("Record failed to start"); 6261 status = BAD_VALUE; 6262 goto startError; 6263 } 6264 ALOGV("Record started OK"); 6265 return status; 6266 } 6267startError: 6268 AudioSystem::stopInput(mId); 6269 clearSyncStartEvent(); 6270 return status; 6271} 6272 6273void AudioFlinger::RecordThread::clearSyncStartEvent() 6274{ 6275 if (mSyncStartEvent != 0) { 6276 mSyncStartEvent->cancel(); 6277 } 6278 mSyncStartEvent.clear(); 6279 mFramestoDrop = 0; 6280} 6281 6282void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6283{ 6284 sp<SyncEvent> strongEvent = event.promote(); 6285 6286 if (strongEvent != 0) { 6287 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6288 me->handleSyncStartEvent(strongEvent); 6289 } 6290} 6291 6292void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6293{ 6294 if (event == mSyncStartEvent) { 6295 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6296 // from audio HAL 6297 mFramestoDrop = mFrameCount * 2; 6298 } 6299} 6300 6301void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6302 ALOGV("RecordThread::stop"); 6303 sp<ThreadBase> strongMe = this; 6304 { 6305 AutoMutex lock(mLock); 6306 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6307 mActiveTrack->mState = TrackBase::PAUSING; 6308 // do not wait for mStartStopCond if exiting 6309 if (exitPending()) { 6310 return; 6311 } 6312 mStartStopCond.wait(mLock); 6313 // if we have been restarted, recordTrack == mActiveTrack.get() here 6314 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6315 mLock.unlock(); 6316 AudioSystem::stopInput(mId); 6317 mLock.lock(); 6318 ALOGV("Record stopped OK"); 6319 } 6320 } 6321 } 6322} 6323 6324bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6325{ 6326 return false; 6327} 6328 6329status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6330{ 6331 if (!isValidSyncEvent(event)) { 6332 return BAD_VALUE; 6333 } 6334 6335 Mutex::Autolock _l(mLock); 6336 6337 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6338 mTrack->setSyncEvent(event); 6339 return NO_ERROR; 6340 } 6341 return NAME_NOT_FOUND; 6342} 6343 6344status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6345{ 6346 const size_t SIZE = 256; 6347 char buffer[SIZE]; 6348 String8 result; 6349 6350 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6351 result.append(buffer); 6352 6353 if (mActiveTrack != 0) { 6354 result.append("Active Track:\n"); 6355 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6356 mActiveTrack->dump(buffer, SIZE); 6357 result.append(buffer); 6358 6359 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6360 result.append(buffer); 6361 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6362 result.append(buffer); 6363 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6364 result.append(buffer); 6365 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6366 result.append(buffer); 6367 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6368 result.append(buffer); 6369 6370 6371 } else { 6372 result.append("No record client\n"); 6373 } 6374 write(fd, result.string(), result.size()); 6375 6376 dumpBase(fd, args); 6377 dumpEffectChains(fd, args); 6378 6379 return NO_ERROR; 6380} 6381 6382// AudioBufferProvider interface 6383status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6384{ 6385 size_t framesReq = buffer->frameCount; 6386 size_t framesReady = mFrameCount - mRsmpInIndex; 6387 int channelCount; 6388 6389 if (framesReady == 0) { 6390 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6391 if (mBytesRead < 0) { 6392 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6393 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6394 // Force input into standby so that it tries to 6395 // recover at next read attempt 6396 mInput->stream->common.standby(&mInput->stream->common); 6397 usleep(kRecordThreadSleepUs); 6398 } 6399 buffer->raw = NULL; 6400 buffer->frameCount = 0; 6401 return NOT_ENOUGH_DATA; 6402 } 6403 mRsmpInIndex = 0; 6404 framesReady = mFrameCount; 6405 } 6406 6407 if (framesReq > framesReady) { 6408 framesReq = framesReady; 6409 } 6410 6411 if (mChannelCount == 1 && mReqChannelCount == 2) { 6412 channelCount = 1; 6413 } else { 6414 channelCount = 2; 6415 } 6416 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6417 buffer->frameCount = framesReq; 6418 return NO_ERROR; 6419} 6420 6421// AudioBufferProvider interface 6422void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6423{ 6424 mRsmpInIndex += buffer->frameCount; 6425 buffer->frameCount = 0; 6426} 6427 6428bool AudioFlinger::RecordThread::checkForNewParameters_l() 6429{ 6430 bool reconfig = false; 6431 6432 while (!mNewParameters.isEmpty()) { 6433 status_t status = NO_ERROR; 6434 String8 keyValuePair = mNewParameters[0]; 6435 AudioParameter param = AudioParameter(keyValuePair); 6436 int value; 6437 audio_format_t reqFormat = mFormat; 6438 int reqSamplingRate = mReqSampleRate; 6439 int reqChannelCount = mReqChannelCount; 6440 6441 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6442 reqSamplingRate = value; 6443 reconfig = true; 6444 } 6445 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6446 reqFormat = (audio_format_t) value; 6447 reconfig = true; 6448 } 6449 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6450 reqChannelCount = popcount(value); 6451 reconfig = true; 6452 } 6453 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6454 // do not accept frame count changes if tracks are open as the track buffer 6455 // size depends on frame count and correct behavior would not be guaranteed 6456 // if frame count is changed after track creation 6457 if (mActiveTrack != 0) { 6458 status = INVALID_OPERATION; 6459 } else { 6460 reconfig = true; 6461 } 6462 } 6463 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6464 // forward device change to effects that have requested to be 6465 // aware of attached audio device. 6466 for (size_t i = 0; i < mEffectChains.size(); i++) { 6467 mEffectChains[i]->setDevice_l(value); 6468 } 6469 // store input device and output device but do not forward output device to audio HAL. 6470 // Note that status is ignored by the caller for output device 6471 // (see AudioFlinger::setParameters() 6472 if (value & AUDIO_DEVICE_OUT_ALL) { 6473 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6474 status = BAD_VALUE; 6475 } else { 6476 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6477 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6478 if (mTrack != NULL) { 6479 bool suspend = audio_is_bluetooth_sco_device( 6480 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6481 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6482 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6483 } 6484 } 6485 mDevice |= (uint32_t)value; 6486 } 6487 if (status == NO_ERROR) { 6488 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6489 if (status == INVALID_OPERATION) { 6490 mInput->stream->common.standby(&mInput->stream->common); 6491 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6492 keyValuePair.string()); 6493 } 6494 if (reconfig) { 6495 if (status == BAD_VALUE && 6496 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6497 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6498 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6499 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6500 (reqChannelCount <= FCC_2)) { 6501 status = NO_ERROR; 6502 } 6503 if (status == NO_ERROR) { 6504 readInputParameters(); 6505 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6506 } 6507 } 6508 } 6509 6510 mNewParameters.removeAt(0); 6511 6512 mParamStatus = status; 6513 mParamCond.signal(); 6514 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6515 // already timed out waiting for the status and will never signal the condition. 6516 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6517 } 6518 return reconfig; 6519} 6520 6521String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6522{ 6523 char *s; 6524 String8 out_s8 = String8(); 6525 6526 Mutex::Autolock _l(mLock); 6527 if (initCheck() != NO_ERROR) { 6528 return out_s8; 6529 } 6530 6531 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6532 out_s8 = String8(s); 6533 free(s); 6534 return out_s8; 6535} 6536 6537void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6538 AudioSystem::OutputDescriptor desc; 6539 void *param2 = NULL; 6540 6541 switch (event) { 6542 case AudioSystem::INPUT_OPENED: 6543 case AudioSystem::INPUT_CONFIG_CHANGED: 6544 desc.channels = mChannelMask; 6545 desc.samplingRate = mSampleRate; 6546 desc.format = mFormat; 6547 desc.frameCount = mFrameCount; 6548 desc.latency = 0; 6549 param2 = &desc; 6550 break; 6551 6552 case AudioSystem::INPUT_CLOSED: 6553 default: 6554 break; 6555 } 6556 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6557} 6558 6559void AudioFlinger::RecordThread::readInputParameters() 6560{ 6561 delete mRsmpInBuffer; 6562 // mRsmpInBuffer is always assigned a new[] below 6563 delete mRsmpOutBuffer; 6564 mRsmpOutBuffer = NULL; 6565 delete mResampler; 6566 mResampler = NULL; 6567 6568 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6569 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6570 mChannelCount = (uint16_t)popcount(mChannelMask); 6571 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6572 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6573 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6574 mFrameCount = mInputBytes / mFrameSize; 6575 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6576 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6577 6578 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6579 { 6580 int channelCount; 6581 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6582 // stereo to mono post process as the resampler always outputs stereo. 6583 if (mChannelCount == 1 && mReqChannelCount == 2) { 6584 channelCount = 1; 6585 } else { 6586 channelCount = 2; 6587 } 6588 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6589 mResampler->setSampleRate(mSampleRate); 6590 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6591 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6592 6593 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6594 if (mChannelCount == 1 && mReqChannelCount == 1) { 6595 mFrameCount >>= 1; 6596 } 6597 6598 } 6599 mRsmpInIndex = mFrameCount; 6600} 6601 6602unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6603{ 6604 Mutex::Autolock _l(mLock); 6605 if (initCheck() != NO_ERROR) { 6606 return 0; 6607 } 6608 6609 return mInput->stream->get_input_frames_lost(mInput->stream); 6610} 6611 6612uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6613{ 6614 Mutex::Autolock _l(mLock); 6615 uint32_t result = 0; 6616 if (getEffectChain_l(sessionId) != 0) { 6617 result = EFFECT_SESSION; 6618 } 6619 6620 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6621 result |= TRACK_SESSION; 6622 } 6623 6624 return result; 6625} 6626 6627AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6628{ 6629 Mutex::Autolock _l(mLock); 6630 return mTrack; 6631} 6632 6633AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6634{ 6635 Mutex::Autolock _l(mLock); 6636 return mInput; 6637} 6638 6639AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6640{ 6641 Mutex::Autolock _l(mLock); 6642 AudioStreamIn *input = mInput; 6643 mInput = NULL; 6644 return input; 6645} 6646 6647// this method must always be called either with ThreadBase mLock held or inside the thread loop 6648audio_stream_t* AudioFlinger::RecordThread::stream() const 6649{ 6650 if (mInput == NULL) { 6651 return NULL; 6652 } 6653 return &mInput->stream->common; 6654} 6655 6656 6657// ---------------------------------------------------------------------------- 6658 6659audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6660{ 6661 if (!settingsAllowed()) { 6662 return 0; 6663 } 6664 Mutex::Autolock _l(mLock); 6665 return loadHwModule_l(name); 6666} 6667 6668// loadHwModule_l() must be called with AudioFlinger::mLock held 6669audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6670{ 6671 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6672 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6673 ALOGW("loadHwModule() module %s already loaded", name); 6674 return mAudioHwDevs.keyAt(i); 6675 } 6676 } 6677 6678 audio_hw_device_t *dev; 6679 6680 int rc = load_audio_interface(name, &dev); 6681 if (rc) { 6682 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6683 return 0; 6684 } 6685 6686 mHardwareStatus = AUDIO_HW_INIT; 6687 rc = dev->init_check(dev); 6688 mHardwareStatus = AUDIO_HW_IDLE; 6689 if (rc) { 6690 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6691 return 0; 6692 } 6693 6694 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6695 (NULL != dev->set_master_volume)) { 6696 AutoMutex lock(mHardwareLock); 6697 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6698 dev->set_master_volume(dev, mMasterVolume); 6699 mHardwareStatus = AUDIO_HW_IDLE; 6700 } 6701 6702 audio_module_handle_t handle = nextUniqueId(); 6703 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6704 6705 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6706 name, dev->common.module->name, dev->common.module->id, handle); 6707 6708 return handle; 6709 6710} 6711 6712audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6713 audio_devices_t *pDevices, 6714 uint32_t *pSamplingRate, 6715 audio_format_t *pFormat, 6716 audio_channel_mask_t *pChannelMask, 6717 uint32_t *pLatencyMs, 6718 audio_output_flags_t flags) 6719{ 6720 status_t status; 6721 PlaybackThread *thread = NULL; 6722 struct audio_config config = { 6723 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6724 channel_mask: pChannelMask ? *pChannelMask : 0, 6725 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6726 }; 6727 audio_stream_out_t *outStream = NULL; 6728 audio_hw_device_t *outHwDev; 6729 6730 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6731 module, 6732 (pDevices != NULL) ? (int)*pDevices : 0, 6733 config.sample_rate, 6734 config.format, 6735 config.channel_mask, 6736 flags); 6737 6738 if (pDevices == NULL || *pDevices == 0) { 6739 return 0; 6740 } 6741 6742 Mutex::Autolock _l(mLock); 6743 6744 outHwDev = findSuitableHwDev_l(module, *pDevices); 6745 if (outHwDev == NULL) 6746 return 0; 6747 6748 audio_io_handle_t id = nextUniqueId(); 6749 6750 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6751 6752 status = outHwDev->open_output_stream(outHwDev, 6753 id, 6754 *pDevices, 6755 (audio_output_flags_t)flags, 6756 &config, 6757 &outStream); 6758 6759 mHardwareStatus = AUDIO_HW_IDLE; 6760 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6761 outStream, 6762 config.sample_rate, 6763 config.format, 6764 config.channel_mask, 6765 status); 6766 6767 if (status == NO_ERROR && outStream != NULL) { 6768 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6769 6770 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6771 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6772 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6773 thread = new DirectOutputThread(this, output, id, *pDevices); 6774 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6775 } else { 6776 thread = new MixerThread(this, output, id, *pDevices); 6777 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6778 } 6779 mPlaybackThreads.add(id, thread); 6780 6781 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6782 if (pFormat != NULL) *pFormat = config.format; 6783 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6784 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6785 6786 // notify client processes of the new output creation 6787 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6788 6789 // the first primary output opened designates the primary hw device 6790 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6791 ALOGI("Using module %d has the primary audio interface", module); 6792 mPrimaryHardwareDev = outHwDev; 6793 6794 AutoMutex lock(mHardwareLock); 6795 mHardwareStatus = AUDIO_HW_SET_MODE; 6796 outHwDev->set_mode(outHwDev, mMode); 6797 6798 // Determine the level of master volume support the primary audio HAL has, 6799 // and set the initial master volume at the same time. 6800 float initialVolume = 1.0; 6801 mMasterVolumeSupportLvl = MVS_NONE; 6802 6803 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6804 if ((NULL != outHwDev->get_master_volume) && 6805 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6806 mMasterVolumeSupportLvl = MVS_FULL; 6807 } else { 6808 mMasterVolumeSupportLvl = MVS_SETONLY; 6809 initialVolume = 1.0; 6810 } 6811 6812 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6813 if ((NULL == outHwDev->set_master_volume) || 6814 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6815 mMasterVolumeSupportLvl = MVS_NONE; 6816 } 6817 // now that we have a primary device, initialize master volume on other devices 6818 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6819 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6820 6821 if ((dev != mPrimaryHardwareDev) && 6822 (NULL != dev->set_master_volume)) { 6823 dev->set_master_volume(dev, initialVolume); 6824 } 6825 } 6826 mHardwareStatus = AUDIO_HW_IDLE; 6827 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6828 ? initialVolume 6829 : 1.0; 6830 mMasterVolume = initialVolume; 6831 } 6832 return id; 6833 } 6834 6835 return 0; 6836} 6837 6838audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6839 audio_io_handle_t output2) 6840{ 6841 Mutex::Autolock _l(mLock); 6842 MixerThread *thread1 = checkMixerThread_l(output1); 6843 MixerThread *thread2 = checkMixerThread_l(output2); 6844 6845 if (thread1 == NULL || thread2 == NULL) { 6846 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6847 return 0; 6848 } 6849 6850 audio_io_handle_t id = nextUniqueId(); 6851 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6852 thread->addOutputTrack(thread2); 6853 mPlaybackThreads.add(id, thread); 6854 // notify client processes of the new output creation 6855 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6856 return id; 6857} 6858 6859status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6860{ 6861 // keep strong reference on the playback thread so that 6862 // it is not destroyed while exit() is executed 6863 sp<PlaybackThread> thread; 6864 { 6865 Mutex::Autolock _l(mLock); 6866 thread = checkPlaybackThread_l(output); 6867 if (thread == NULL) { 6868 return BAD_VALUE; 6869 } 6870 6871 ALOGV("closeOutput() %d", output); 6872 6873 if (thread->type() == ThreadBase::MIXER) { 6874 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6875 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6876 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6877 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6878 } 6879 } 6880 } 6881 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6882 mPlaybackThreads.removeItem(output); 6883 } 6884 thread->exit(); 6885 // The thread entity (active unit of execution) is no longer running here, 6886 // but the ThreadBase container still exists. 6887 6888 if (thread->type() != ThreadBase::DUPLICATING) { 6889 AudioStreamOut *out = thread->clearOutput(); 6890 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6891 // from now on thread->mOutput is NULL 6892 out->hwDev->close_output_stream(out->hwDev, out->stream); 6893 delete out; 6894 } 6895 return NO_ERROR; 6896} 6897 6898status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6899{ 6900 Mutex::Autolock _l(mLock); 6901 PlaybackThread *thread = checkPlaybackThread_l(output); 6902 6903 if (thread == NULL) { 6904 return BAD_VALUE; 6905 } 6906 6907 ALOGV("suspendOutput() %d", output); 6908 thread->suspend(); 6909 6910 return NO_ERROR; 6911} 6912 6913status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6914{ 6915 Mutex::Autolock _l(mLock); 6916 PlaybackThread *thread = checkPlaybackThread_l(output); 6917 6918 if (thread == NULL) { 6919 return BAD_VALUE; 6920 } 6921 6922 ALOGV("restoreOutput() %d", output); 6923 6924 thread->restore(); 6925 6926 return NO_ERROR; 6927} 6928 6929audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6930 audio_devices_t *pDevices, 6931 uint32_t *pSamplingRate, 6932 audio_format_t *pFormat, 6933 uint32_t *pChannelMask) 6934{ 6935 status_t status; 6936 RecordThread *thread = NULL; 6937 struct audio_config config = { 6938 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6939 channel_mask: pChannelMask ? *pChannelMask : 0, 6940 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6941 }; 6942 uint32_t reqSamplingRate = config.sample_rate; 6943 audio_format_t reqFormat = config.format; 6944 audio_channel_mask_t reqChannels = config.channel_mask; 6945 audio_stream_in_t *inStream = NULL; 6946 audio_hw_device_t *inHwDev; 6947 6948 if (pDevices == NULL || *pDevices == 0) { 6949 return 0; 6950 } 6951 6952 Mutex::Autolock _l(mLock); 6953 6954 inHwDev = findSuitableHwDev_l(module, *pDevices); 6955 if (inHwDev == NULL) 6956 return 0; 6957 6958 audio_io_handle_t id = nextUniqueId(); 6959 6960 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6961 &inStream); 6962 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6963 inStream, 6964 config.sample_rate, 6965 config.format, 6966 config.channel_mask, 6967 status); 6968 6969 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6970 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6971 // or stereo to mono conversions on 16 bit PCM inputs. 6972 if (status == BAD_VALUE && 6973 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6974 (config.sample_rate <= 2 * reqSamplingRate) && 6975 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6976 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6977 inStream = NULL; 6978 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6979 } 6980 6981 if (status == NO_ERROR && inStream != NULL) { 6982 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6983 6984 // Start record thread 6985 // RecorThread require both input and output device indication to forward to audio 6986 // pre processing modules 6987 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6988 thread = new RecordThread(this, 6989 input, 6990 reqSamplingRate, 6991 reqChannels, 6992 id, 6993 device); 6994 mRecordThreads.add(id, thread); 6995 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6996 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6997 if (pFormat != NULL) *pFormat = config.format; 6998 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6999 7000 input->stream->common.standby(&input->stream->common); 7001 7002 // notify client processes of the new input creation 7003 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7004 return id; 7005 } 7006 7007 return 0; 7008} 7009 7010status_t AudioFlinger::closeInput(audio_io_handle_t input) 7011{ 7012 // keep strong reference on the record thread so that 7013 // it is not destroyed while exit() is executed 7014 sp<RecordThread> thread; 7015 { 7016 Mutex::Autolock _l(mLock); 7017 thread = checkRecordThread_l(input); 7018 if (thread == NULL) { 7019 return BAD_VALUE; 7020 } 7021 7022 ALOGV("closeInput() %d", input); 7023 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7024 mRecordThreads.removeItem(input); 7025 } 7026 thread->exit(); 7027 // The thread entity (active unit of execution) is no longer running here, 7028 // but the ThreadBase container still exists. 7029 7030 AudioStreamIn *in = thread->clearInput(); 7031 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7032 // from now on thread->mInput is NULL 7033 in->hwDev->close_input_stream(in->hwDev, in->stream); 7034 delete in; 7035 7036 return NO_ERROR; 7037} 7038 7039status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7040{ 7041 Mutex::Autolock _l(mLock); 7042 MixerThread *dstThread = checkMixerThread_l(output); 7043 if (dstThread == NULL) { 7044 ALOGW("setStreamOutput() bad output id %d", output); 7045 return BAD_VALUE; 7046 } 7047 7048 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7049 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7050 7051 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7052 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7053 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7054 MixerThread *srcThread = (MixerThread *)thread; 7055 srcThread->invalidateTracks(stream); 7056 } 7057 } 7058 7059 return NO_ERROR; 7060} 7061 7062 7063int AudioFlinger::newAudioSessionId() 7064{ 7065 return nextUniqueId(); 7066} 7067 7068void AudioFlinger::acquireAudioSessionId(int audioSession) 7069{ 7070 Mutex::Autolock _l(mLock); 7071 pid_t caller = IPCThreadState::self()->getCallingPid(); 7072 ALOGV("acquiring %d from %d", audioSession, caller); 7073 size_t num = mAudioSessionRefs.size(); 7074 for (size_t i = 0; i< num; i++) { 7075 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7076 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7077 ref->mCnt++; 7078 ALOGV(" incremented refcount to %d", ref->mCnt); 7079 return; 7080 } 7081 } 7082 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7083 ALOGV(" added new entry for %d", audioSession); 7084} 7085 7086void AudioFlinger::releaseAudioSessionId(int audioSession) 7087{ 7088 Mutex::Autolock _l(mLock); 7089 pid_t caller = IPCThreadState::self()->getCallingPid(); 7090 ALOGV("releasing %d from %d", audioSession, caller); 7091 size_t num = mAudioSessionRefs.size(); 7092 for (size_t i = 0; i< num; i++) { 7093 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7094 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7095 ref->mCnt--; 7096 ALOGV(" decremented refcount to %d", ref->mCnt); 7097 if (ref->mCnt == 0) { 7098 mAudioSessionRefs.removeAt(i); 7099 delete ref; 7100 purgeStaleEffects_l(); 7101 } 7102 return; 7103 } 7104 } 7105 ALOGW("session id %d not found for pid %d", audioSession, caller); 7106} 7107 7108void AudioFlinger::purgeStaleEffects_l() { 7109 7110 ALOGV("purging stale effects"); 7111 7112 Vector< sp<EffectChain> > chains; 7113 7114 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7115 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7116 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7117 sp<EffectChain> ec = t->mEffectChains[j]; 7118 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7119 chains.push(ec); 7120 } 7121 } 7122 } 7123 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7124 sp<RecordThread> t = mRecordThreads.valueAt(i); 7125 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7126 sp<EffectChain> ec = t->mEffectChains[j]; 7127 chains.push(ec); 7128 } 7129 } 7130 7131 for (size_t i = 0; i < chains.size(); i++) { 7132 sp<EffectChain> ec = chains[i]; 7133 int sessionid = ec->sessionId(); 7134 sp<ThreadBase> t = ec->mThread.promote(); 7135 if (t == 0) { 7136 continue; 7137 } 7138 size_t numsessionrefs = mAudioSessionRefs.size(); 7139 bool found = false; 7140 for (size_t k = 0; k < numsessionrefs; k++) { 7141 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7142 if (ref->mSessionid == sessionid) { 7143 ALOGV(" session %d still exists for %d with %d refs", 7144 sessionid, ref->mPid, ref->mCnt); 7145 found = true; 7146 break; 7147 } 7148 } 7149 if (!found) { 7150 // remove all effects from the chain 7151 while (ec->mEffects.size()) { 7152 sp<EffectModule> effect = ec->mEffects[0]; 7153 effect->unPin(); 7154 Mutex::Autolock _l (t->mLock); 7155 t->removeEffect_l(effect); 7156 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7157 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7158 if (handle != 0) { 7159 handle->mEffect.clear(); 7160 if (handle->mHasControl && handle->mEnabled) { 7161 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7162 } 7163 } 7164 } 7165 AudioSystem::unregisterEffect(effect->id()); 7166 } 7167 } 7168 } 7169 return; 7170} 7171 7172// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7173AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7174{ 7175 return mPlaybackThreads.valueFor(output).get(); 7176} 7177 7178// checkMixerThread_l() must be called with AudioFlinger::mLock held 7179AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7180{ 7181 PlaybackThread *thread = checkPlaybackThread_l(output); 7182 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7183} 7184 7185// checkRecordThread_l() must be called with AudioFlinger::mLock held 7186AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7187{ 7188 return mRecordThreads.valueFor(input).get(); 7189} 7190 7191uint32_t AudioFlinger::nextUniqueId() 7192{ 7193 return android_atomic_inc(&mNextUniqueId); 7194} 7195 7196AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7197{ 7198 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7199 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7200 AudioStreamOut *output = thread->getOutput(); 7201 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7202 return thread; 7203 } 7204 } 7205 return NULL; 7206} 7207 7208uint32_t AudioFlinger::primaryOutputDevice_l() const 7209{ 7210 PlaybackThread *thread = primaryPlaybackThread_l(); 7211 7212 if (thread == NULL) { 7213 return 0; 7214 } 7215 7216 return thread->device(); 7217} 7218 7219sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7220 int triggerSession, 7221 int listenerSession, 7222 sync_event_callback_t callBack, 7223 void *cookie) 7224{ 7225 Mutex::Autolock _l(mLock); 7226 7227 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7228 status_t playStatus = NAME_NOT_FOUND; 7229 status_t recStatus = NAME_NOT_FOUND; 7230 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7231 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7232 if (playStatus == NO_ERROR) { 7233 return event; 7234 } 7235 } 7236 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7237 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7238 if (recStatus == NO_ERROR) { 7239 return event; 7240 } 7241 } 7242 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7243 mPendingSyncEvents.add(event); 7244 } else { 7245 ALOGV("createSyncEvent() invalid event %d", event->type()); 7246 event.clear(); 7247 } 7248 return event; 7249} 7250 7251// ---------------------------------------------------------------------------- 7252// Effect management 7253// ---------------------------------------------------------------------------- 7254 7255 7256status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7257{ 7258 Mutex::Autolock _l(mLock); 7259 return EffectQueryNumberEffects(numEffects); 7260} 7261 7262status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7263{ 7264 Mutex::Autolock _l(mLock); 7265 return EffectQueryEffect(index, descriptor); 7266} 7267 7268status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7269 effect_descriptor_t *descriptor) const 7270{ 7271 Mutex::Autolock _l(mLock); 7272 return EffectGetDescriptor(pUuid, descriptor); 7273} 7274 7275 7276sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7277 effect_descriptor_t *pDesc, 7278 const sp<IEffectClient>& effectClient, 7279 int32_t priority, 7280 audio_io_handle_t io, 7281 int sessionId, 7282 status_t *status, 7283 int *id, 7284 int *enabled) 7285{ 7286 status_t lStatus = NO_ERROR; 7287 sp<EffectHandle> handle; 7288 effect_descriptor_t desc; 7289 7290 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7291 pid, effectClient.get(), priority, sessionId, io); 7292 7293 if (pDesc == NULL) { 7294 lStatus = BAD_VALUE; 7295 goto Exit; 7296 } 7297 7298 // check audio settings permission for global effects 7299 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7300 lStatus = PERMISSION_DENIED; 7301 goto Exit; 7302 } 7303 7304 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7305 // that can only be created by audio policy manager (running in same process) 7306 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7307 lStatus = PERMISSION_DENIED; 7308 goto Exit; 7309 } 7310 7311 if (io == 0) { 7312 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7313 // output must be specified by AudioPolicyManager when using session 7314 // AUDIO_SESSION_OUTPUT_STAGE 7315 lStatus = BAD_VALUE; 7316 goto Exit; 7317 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7318 // if the output returned by getOutputForEffect() is removed before we lock the 7319 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7320 // and we will exit safely 7321 io = AudioSystem::getOutputForEffect(&desc); 7322 } 7323 } 7324 7325 { 7326 Mutex::Autolock _l(mLock); 7327 7328 7329 if (!EffectIsNullUuid(&pDesc->uuid)) { 7330 // if uuid is specified, request effect descriptor 7331 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7332 if (lStatus < 0) { 7333 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7334 goto Exit; 7335 } 7336 } else { 7337 // if uuid is not specified, look for an available implementation 7338 // of the required type in effect factory 7339 if (EffectIsNullUuid(&pDesc->type)) { 7340 ALOGW("createEffect() no effect type"); 7341 lStatus = BAD_VALUE; 7342 goto Exit; 7343 } 7344 uint32_t numEffects = 0; 7345 effect_descriptor_t d; 7346 d.flags = 0; // prevent compiler warning 7347 bool found = false; 7348 7349 lStatus = EffectQueryNumberEffects(&numEffects); 7350 if (lStatus < 0) { 7351 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7352 goto Exit; 7353 } 7354 for (uint32_t i = 0; i < numEffects; i++) { 7355 lStatus = EffectQueryEffect(i, &desc); 7356 if (lStatus < 0) { 7357 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7358 continue; 7359 } 7360 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7361 // If matching type found save effect descriptor. If the session is 7362 // 0 and the effect is not auxiliary, continue enumeration in case 7363 // an auxiliary version of this effect type is available 7364 found = true; 7365 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7366 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7367 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7368 break; 7369 } 7370 } 7371 } 7372 if (!found) { 7373 lStatus = BAD_VALUE; 7374 ALOGW("createEffect() effect not found"); 7375 goto Exit; 7376 } 7377 // For same effect type, chose auxiliary version over insert version if 7378 // connect to output mix (Compliance to OpenSL ES) 7379 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7380 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7381 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7382 } 7383 } 7384 7385 // Do not allow auxiliary effects on a session different from 0 (output mix) 7386 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7387 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7388 lStatus = INVALID_OPERATION; 7389 goto Exit; 7390 } 7391 7392 // check recording permission for visualizer 7393 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7394 !recordingAllowed()) { 7395 lStatus = PERMISSION_DENIED; 7396 goto Exit; 7397 } 7398 7399 // return effect descriptor 7400 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7401 7402 // If output is not specified try to find a matching audio session ID in one of the 7403 // output threads. 7404 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7405 // because of code checking output when entering the function. 7406 // Note: io is never 0 when creating an effect on an input 7407 if (io == 0) { 7408 // look for the thread where the specified audio session is present 7409 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7410 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7411 io = mPlaybackThreads.keyAt(i); 7412 break; 7413 } 7414 } 7415 if (io == 0) { 7416 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7417 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7418 io = mRecordThreads.keyAt(i); 7419 break; 7420 } 7421 } 7422 } 7423 // If no output thread contains the requested session ID, default to 7424 // first output. The effect chain will be moved to the correct output 7425 // thread when a track with the same session ID is created 7426 if (io == 0 && mPlaybackThreads.size()) { 7427 io = mPlaybackThreads.keyAt(0); 7428 } 7429 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7430 } 7431 ThreadBase *thread = checkRecordThread_l(io); 7432 if (thread == NULL) { 7433 thread = checkPlaybackThread_l(io); 7434 if (thread == NULL) { 7435 ALOGE("createEffect() unknown output thread"); 7436 lStatus = BAD_VALUE; 7437 goto Exit; 7438 } 7439 } 7440 7441 sp<Client> client = registerPid_l(pid); 7442 7443 // create effect on selected output thread 7444 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7445 &desc, enabled, &lStatus); 7446 if (handle != 0 && id != NULL) { 7447 *id = handle->id(); 7448 } 7449 } 7450 7451Exit: 7452 if (status != NULL) { 7453 *status = lStatus; 7454 } 7455 return handle; 7456} 7457 7458status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7459 audio_io_handle_t dstOutput) 7460{ 7461 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7462 sessionId, srcOutput, dstOutput); 7463 Mutex::Autolock _l(mLock); 7464 if (srcOutput == dstOutput) { 7465 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7466 return NO_ERROR; 7467 } 7468 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7469 if (srcThread == NULL) { 7470 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7471 return BAD_VALUE; 7472 } 7473 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7474 if (dstThread == NULL) { 7475 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7476 return BAD_VALUE; 7477 } 7478 7479 Mutex::Autolock _dl(dstThread->mLock); 7480 Mutex::Autolock _sl(srcThread->mLock); 7481 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7482 7483 return NO_ERROR; 7484} 7485 7486// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7487status_t AudioFlinger::moveEffectChain_l(int sessionId, 7488 AudioFlinger::PlaybackThread *srcThread, 7489 AudioFlinger::PlaybackThread *dstThread, 7490 bool reRegister) 7491{ 7492 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7493 sessionId, srcThread, dstThread); 7494 7495 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7496 if (chain == 0) { 7497 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7498 sessionId, srcThread); 7499 return INVALID_OPERATION; 7500 } 7501 7502 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7503 // so that a new chain is created with correct parameters when first effect is added. This is 7504 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7505 // removed. 7506 srcThread->removeEffectChain_l(chain); 7507 7508 // transfer all effects one by one so that new effect chain is created on new thread with 7509 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7510 audio_io_handle_t dstOutput = dstThread->id(); 7511 sp<EffectChain> dstChain; 7512 uint32_t strategy = 0; // prevent compiler warning 7513 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7514 while (effect != 0) { 7515 srcThread->removeEffect_l(effect); 7516 dstThread->addEffect_l(effect); 7517 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7518 if (effect->state() == EffectModule::ACTIVE || 7519 effect->state() == EffectModule::STOPPING) { 7520 effect->start(); 7521 } 7522 // if the move request is not received from audio policy manager, the effect must be 7523 // re-registered with the new strategy and output 7524 if (dstChain == 0) { 7525 dstChain = effect->chain().promote(); 7526 if (dstChain == 0) { 7527 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7528 srcThread->addEffect_l(effect); 7529 return NO_INIT; 7530 } 7531 strategy = dstChain->strategy(); 7532 } 7533 if (reRegister) { 7534 AudioSystem::unregisterEffect(effect->id()); 7535 AudioSystem::registerEffect(&effect->desc(), 7536 dstOutput, 7537 strategy, 7538 sessionId, 7539 effect->id()); 7540 } 7541 effect = chain->getEffectFromId_l(0); 7542 } 7543 7544 return NO_ERROR; 7545} 7546 7547 7548// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7549sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7550 const sp<AudioFlinger::Client>& client, 7551 const sp<IEffectClient>& effectClient, 7552 int32_t priority, 7553 int sessionId, 7554 effect_descriptor_t *desc, 7555 int *enabled, 7556 status_t *status 7557 ) 7558{ 7559 sp<EffectModule> effect; 7560 sp<EffectHandle> handle; 7561 status_t lStatus; 7562 sp<EffectChain> chain; 7563 bool chainCreated = false; 7564 bool effectCreated = false; 7565 bool effectRegistered = false; 7566 7567 lStatus = initCheck(); 7568 if (lStatus != NO_ERROR) { 7569 ALOGW("createEffect_l() Audio driver not initialized."); 7570 goto Exit; 7571 } 7572 7573 // Do not allow effects with session ID 0 on direct output or duplicating threads 7574 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7575 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7576 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7577 desc->name, sessionId); 7578 lStatus = BAD_VALUE; 7579 goto Exit; 7580 } 7581 // Only Pre processor effects are allowed on input threads and only on input threads 7582 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7583 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7584 desc->name, desc->flags, mType); 7585 lStatus = BAD_VALUE; 7586 goto Exit; 7587 } 7588 7589 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7590 7591 { // scope for mLock 7592 Mutex::Autolock _l(mLock); 7593 7594 // check for existing effect chain with the requested audio session 7595 chain = getEffectChain_l(sessionId); 7596 if (chain == 0) { 7597 // create a new chain for this session 7598 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7599 chain = new EffectChain(this, sessionId); 7600 addEffectChain_l(chain); 7601 chain->setStrategy(getStrategyForSession_l(sessionId)); 7602 chainCreated = true; 7603 } else { 7604 effect = chain->getEffectFromDesc_l(desc); 7605 } 7606 7607 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7608 7609 if (effect == 0) { 7610 int id = mAudioFlinger->nextUniqueId(); 7611 // Check CPU and memory usage 7612 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7613 if (lStatus != NO_ERROR) { 7614 goto Exit; 7615 } 7616 effectRegistered = true; 7617 // create a new effect module if none present in the chain 7618 effect = new EffectModule(this, chain, desc, id, sessionId); 7619 lStatus = effect->status(); 7620 if (lStatus != NO_ERROR) { 7621 goto Exit; 7622 } 7623 lStatus = chain->addEffect_l(effect); 7624 if (lStatus != NO_ERROR) { 7625 goto Exit; 7626 } 7627 effectCreated = true; 7628 7629 effect->setDevice(mDevice); 7630 effect->setMode(mAudioFlinger->getMode()); 7631 } 7632 // create effect handle and connect it to effect module 7633 handle = new EffectHandle(effect, client, effectClient, priority); 7634 lStatus = effect->addHandle(handle); 7635 if (enabled != NULL) { 7636 *enabled = (int)effect->isEnabled(); 7637 } 7638 } 7639 7640Exit: 7641 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7642 Mutex::Autolock _l(mLock); 7643 if (effectCreated) { 7644 chain->removeEffect_l(effect); 7645 } 7646 if (effectRegistered) { 7647 AudioSystem::unregisterEffect(effect->id()); 7648 } 7649 if (chainCreated) { 7650 removeEffectChain_l(chain); 7651 } 7652 handle.clear(); 7653 } 7654 7655 if (status != NULL) { 7656 *status = lStatus; 7657 } 7658 return handle; 7659} 7660 7661sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7662{ 7663 sp<EffectChain> chain = getEffectChain_l(sessionId); 7664 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7665} 7666 7667// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7668// PlaybackThread::mLock held 7669status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7670{ 7671 // check for existing effect chain with the requested audio session 7672 int sessionId = effect->sessionId(); 7673 sp<EffectChain> chain = getEffectChain_l(sessionId); 7674 bool chainCreated = false; 7675 7676 if (chain == 0) { 7677 // create a new chain for this session 7678 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7679 chain = new EffectChain(this, sessionId); 7680 addEffectChain_l(chain); 7681 chain->setStrategy(getStrategyForSession_l(sessionId)); 7682 chainCreated = true; 7683 } 7684 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7685 7686 if (chain->getEffectFromId_l(effect->id()) != 0) { 7687 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7688 this, effect->desc().name, chain.get()); 7689 return BAD_VALUE; 7690 } 7691 7692 status_t status = chain->addEffect_l(effect); 7693 if (status != NO_ERROR) { 7694 if (chainCreated) { 7695 removeEffectChain_l(chain); 7696 } 7697 return status; 7698 } 7699 7700 effect->setDevice(mDevice); 7701 effect->setMode(mAudioFlinger->getMode()); 7702 return NO_ERROR; 7703} 7704 7705void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7706 7707 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7708 effect_descriptor_t desc = effect->desc(); 7709 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7710 detachAuxEffect_l(effect->id()); 7711 } 7712 7713 sp<EffectChain> chain = effect->chain().promote(); 7714 if (chain != 0) { 7715 // remove effect chain if removing last effect 7716 if (chain->removeEffect_l(effect) == 0) { 7717 removeEffectChain_l(chain); 7718 } 7719 } else { 7720 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7721 } 7722} 7723 7724void AudioFlinger::ThreadBase::lockEffectChains_l( 7725 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7726{ 7727 effectChains = mEffectChains; 7728 for (size_t i = 0; i < mEffectChains.size(); i++) { 7729 mEffectChains[i]->lock(); 7730 } 7731} 7732 7733void AudioFlinger::ThreadBase::unlockEffectChains( 7734 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7735{ 7736 for (size_t i = 0; i < effectChains.size(); i++) { 7737 effectChains[i]->unlock(); 7738 } 7739} 7740 7741sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7742{ 7743 Mutex::Autolock _l(mLock); 7744 return getEffectChain_l(sessionId); 7745} 7746 7747sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7748{ 7749 size_t size = mEffectChains.size(); 7750 for (size_t i = 0; i < size; i++) { 7751 if (mEffectChains[i]->sessionId() == sessionId) { 7752 return mEffectChains[i]; 7753 } 7754 } 7755 return 0; 7756} 7757 7758void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7759{ 7760 Mutex::Autolock _l(mLock); 7761 size_t size = mEffectChains.size(); 7762 for (size_t i = 0; i < size; i++) { 7763 mEffectChains[i]->setMode_l(mode); 7764 } 7765} 7766 7767void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7768 const wp<EffectHandle>& handle, 7769 bool unpinIfLast) { 7770 7771 Mutex::Autolock _l(mLock); 7772 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7773 // delete the effect module if removing last handle on it 7774 if (effect->removeHandle(handle) == 0) { 7775 if (!effect->isPinned() || unpinIfLast) { 7776 removeEffect_l(effect); 7777 AudioSystem::unregisterEffect(effect->id()); 7778 } 7779 } 7780} 7781 7782status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7783{ 7784 int session = chain->sessionId(); 7785 int16_t *buffer = mMixBuffer; 7786 bool ownsBuffer = false; 7787 7788 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7789 if (session > 0) { 7790 // Only one effect chain can be present in direct output thread and it uses 7791 // the mix buffer as input 7792 if (mType != DIRECT) { 7793 size_t numSamples = mNormalFrameCount * mChannelCount; 7794 buffer = new int16_t[numSamples]; 7795 memset(buffer, 0, numSamples * sizeof(int16_t)); 7796 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7797 ownsBuffer = true; 7798 } 7799 7800 // Attach all tracks with same session ID to this chain. 7801 for (size_t i = 0; i < mTracks.size(); ++i) { 7802 sp<Track> track = mTracks[i]; 7803 if (session == track->sessionId()) { 7804 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7805 track->setMainBuffer(buffer); 7806 chain->incTrackCnt(); 7807 } 7808 } 7809 7810 // indicate all active tracks in the chain 7811 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7812 sp<Track> track = mActiveTracks[i].promote(); 7813 if (track == 0) continue; 7814 if (session == track->sessionId()) { 7815 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7816 chain->incActiveTrackCnt(); 7817 } 7818 } 7819 } 7820 7821 chain->setInBuffer(buffer, ownsBuffer); 7822 chain->setOutBuffer(mMixBuffer); 7823 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7824 // chains list in order to be processed last as it contains output stage effects 7825 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7826 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7827 // after track specific effects and before output stage 7828 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7829 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7830 // Effect chain for other sessions are inserted at beginning of effect 7831 // chains list to be processed before output mix effects. Relative order between other 7832 // sessions is not important 7833 size_t size = mEffectChains.size(); 7834 size_t i = 0; 7835 for (i = 0; i < size; i++) { 7836 if (mEffectChains[i]->sessionId() < session) break; 7837 } 7838 mEffectChains.insertAt(chain, i); 7839 checkSuspendOnAddEffectChain_l(chain); 7840 7841 return NO_ERROR; 7842} 7843 7844size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7845{ 7846 int session = chain->sessionId(); 7847 7848 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7849 7850 for (size_t i = 0; i < mEffectChains.size(); i++) { 7851 if (chain == mEffectChains[i]) { 7852 mEffectChains.removeAt(i); 7853 // detach all active tracks from the chain 7854 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7855 sp<Track> track = mActiveTracks[i].promote(); 7856 if (track == 0) continue; 7857 if (session == track->sessionId()) { 7858 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7859 chain.get(), session); 7860 chain->decActiveTrackCnt(); 7861 } 7862 } 7863 7864 // detach all tracks with same session ID from this chain 7865 for (size_t i = 0; i < mTracks.size(); ++i) { 7866 sp<Track> track = mTracks[i]; 7867 if (session == track->sessionId()) { 7868 track->setMainBuffer(mMixBuffer); 7869 chain->decTrackCnt(); 7870 } 7871 } 7872 break; 7873 } 7874 } 7875 return mEffectChains.size(); 7876} 7877 7878status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7879 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7880{ 7881 Mutex::Autolock _l(mLock); 7882 return attachAuxEffect_l(track, EffectId); 7883} 7884 7885status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7886 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7887{ 7888 status_t status = NO_ERROR; 7889 7890 if (EffectId == 0) { 7891 track->setAuxBuffer(0, NULL); 7892 } else { 7893 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7894 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7895 if (effect != 0) { 7896 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7897 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7898 } else { 7899 status = INVALID_OPERATION; 7900 } 7901 } else { 7902 status = BAD_VALUE; 7903 } 7904 } 7905 return status; 7906} 7907 7908void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7909{ 7910 for (size_t i = 0; i < mTracks.size(); ++i) { 7911 sp<Track> track = mTracks[i]; 7912 if (track->auxEffectId() == effectId) { 7913 attachAuxEffect_l(track, 0); 7914 } 7915 } 7916} 7917 7918status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7919{ 7920 // only one chain per input thread 7921 if (mEffectChains.size() != 0) { 7922 return INVALID_OPERATION; 7923 } 7924 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7925 7926 chain->setInBuffer(NULL); 7927 chain->setOutBuffer(NULL); 7928 7929 checkSuspendOnAddEffectChain_l(chain); 7930 7931 mEffectChains.add(chain); 7932 7933 return NO_ERROR; 7934} 7935 7936size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7937{ 7938 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7939 ALOGW_IF(mEffectChains.size() != 1, 7940 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7941 chain.get(), mEffectChains.size(), this); 7942 if (mEffectChains.size() == 1) { 7943 mEffectChains.removeAt(0); 7944 } 7945 return 0; 7946} 7947 7948// ---------------------------------------------------------------------------- 7949// EffectModule implementation 7950// ---------------------------------------------------------------------------- 7951 7952#undef LOG_TAG 7953#define LOG_TAG "AudioFlinger::EffectModule" 7954 7955AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7956 const wp<AudioFlinger::EffectChain>& chain, 7957 effect_descriptor_t *desc, 7958 int id, 7959 int sessionId) 7960 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7961 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7962{ 7963 ALOGV("Constructor %p", this); 7964 int lStatus; 7965 if (thread == NULL) { 7966 return; 7967 } 7968 7969 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7970 7971 // create effect engine from effect factory 7972 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7973 7974 if (mStatus != NO_ERROR) { 7975 return; 7976 } 7977 lStatus = init(); 7978 if (lStatus < 0) { 7979 mStatus = lStatus; 7980 goto Error; 7981 } 7982 7983 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7984 mPinned = true; 7985 } 7986 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7987 return; 7988Error: 7989 EffectRelease(mEffectInterface); 7990 mEffectInterface = NULL; 7991 ALOGV("Constructor Error %d", mStatus); 7992} 7993 7994AudioFlinger::EffectModule::~EffectModule() 7995{ 7996 ALOGV("Destructor %p", this); 7997 if (mEffectInterface != NULL) { 7998 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7999 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8000 sp<ThreadBase> thread = mThread.promote(); 8001 if (thread != 0) { 8002 audio_stream_t *stream = thread->stream(); 8003 if (stream != NULL) { 8004 stream->remove_audio_effect(stream, mEffectInterface); 8005 } 8006 } 8007 } 8008 // release effect engine 8009 EffectRelease(mEffectInterface); 8010 } 8011} 8012 8013status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8014{ 8015 status_t status; 8016 8017 Mutex::Autolock _l(mLock); 8018 int priority = handle->priority(); 8019 size_t size = mHandles.size(); 8020 sp<EffectHandle> h; 8021 size_t i; 8022 for (i = 0; i < size; i++) { 8023 h = mHandles[i].promote(); 8024 if (h == 0) continue; 8025 if (h->priority() <= priority) break; 8026 } 8027 // if inserted in first place, move effect control from previous owner to this handle 8028 if (i == 0) { 8029 bool enabled = false; 8030 if (h != 0) { 8031 enabled = h->enabled(); 8032 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8033 } 8034 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8035 status = NO_ERROR; 8036 } else { 8037 status = ALREADY_EXISTS; 8038 } 8039 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8040 mHandles.insertAt(handle, i); 8041 return status; 8042} 8043 8044size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8045{ 8046 Mutex::Autolock _l(mLock); 8047 size_t size = mHandles.size(); 8048 size_t i; 8049 for (i = 0; i < size; i++) { 8050 if (mHandles[i] == handle) break; 8051 } 8052 if (i == size) { 8053 return size; 8054 } 8055 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8056 8057 bool enabled = false; 8058 EffectHandle *hdl = handle.unsafe_get(); 8059 if (hdl != NULL) { 8060 ALOGV("removeHandle() unsafe_get OK"); 8061 enabled = hdl->enabled(); 8062 } 8063 mHandles.removeAt(i); 8064 size = mHandles.size(); 8065 // if removed from first place, move effect control from this handle to next in line 8066 if (i == 0 && size != 0) { 8067 sp<EffectHandle> h = mHandles[0].promote(); 8068 if (h != 0) { 8069 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8070 } 8071 } 8072 8073 // Prevent calls to process() and other functions on effect interface from now on. 8074 // The effect engine will be released by the destructor when the last strong reference on 8075 // this object is released which can happen after next process is called. 8076 if (size == 0 && !mPinned) { 8077 mState = DESTROYED; 8078 } 8079 8080 return size; 8081} 8082 8083sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8084{ 8085 Mutex::Autolock _l(mLock); 8086 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8087} 8088 8089void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8090{ 8091 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8092 // keep a strong reference on this EffectModule to avoid calling the 8093 // destructor before we exit 8094 sp<EffectModule> keep(this); 8095 { 8096 sp<ThreadBase> thread = mThread.promote(); 8097 if (thread != 0) { 8098 thread->disconnectEffect(keep, handle, unpinIfLast); 8099 } 8100 } 8101} 8102 8103void AudioFlinger::EffectModule::updateState() { 8104 Mutex::Autolock _l(mLock); 8105 8106 switch (mState) { 8107 case RESTART: 8108 reset_l(); 8109 // FALL THROUGH 8110 8111 case STARTING: 8112 // clear auxiliary effect input buffer for next accumulation 8113 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8114 memset(mConfig.inputCfg.buffer.raw, 8115 0, 8116 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8117 } 8118 start_l(); 8119 mState = ACTIVE; 8120 break; 8121 case STOPPING: 8122 stop_l(); 8123 mDisableWaitCnt = mMaxDisableWaitCnt; 8124 mState = STOPPED; 8125 break; 8126 case STOPPED: 8127 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8128 // turn off sequence. 8129 if (--mDisableWaitCnt == 0) { 8130 reset_l(); 8131 mState = IDLE; 8132 } 8133 break; 8134 default: //IDLE , ACTIVE, DESTROYED 8135 break; 8136 } 8137} 8138 8139void AudioFlinger::EffectModule::process() 8140{ 8141 Mutex::Autolock _l(mLock); 8142 8143 if (mState == DESTROYED || mEffectInterface == NULL || 8144 mConfig.inputCfg.buffer.raw == NULL || 8145 mConfig.outputCfg.buffer.raw == NULL) { 8146 return; 8147 } 8148 8149 if (isProcessEnabled()) { 8150 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8151 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8152 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8153 mConfig.inputCfg.buffer.s32, 8154 mConfig.inputCfg.buffer.frameCount/2); 8155 } 8156 8157 // do the actual processing in the effect engine 8158 int ret = (*mEffectInterface)->process(mEffectInterface, 8159 &mConfig.inputCfg.buffer, 8160 &mConfig.outputCfg.buffer); 8161 8162 // force transition to IDLE state when engine is ready 8163 if (mState == STOPPED && ret == -ENODATA) { 8164 mDisableWaitCnt = 1; 8165 } 8166 8167 // clear auxiliary effect input buffer for next accumulation 8168 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8169 memset(mConfig.inputCfg.buffer.raw, 0, 8170 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8171 } 8172 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8173 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8174 // If an insert effect is idle and input buffer is different from output buffer, 8175 // accumulate input onto output 8176 sp<EffectChain> chain = mChain.promote(); 8177 if (chain != 0 && chain->activeTrackCnt() != 0) { 8178 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8179 int16_t *in = mConfig.inputCfg.buffer.s16; 8180 int16_t *out = mConfig.outputCfg.buffer.s16; 8181 for (size_t i = 0; i < frameCnt; i++) { 8182 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8183 } 8184 } 8185 } 8186} 8187 8188void AudioFlinger::EffectModule::reset_l() 8189{ 8190 if (mEffectInterface == NULL) { 8191 return; 8192 } 8193 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8194} 8195 8196status_t AudioFlinger::EffectModule::configure() 8197{ 8198 uint32_t channels; 8199 if (mEffectInterface == NULL) { 8200 return NO_INIT; 8201 } 8202 8203 sp<ThreadBase> thread = mThread.promote(); 8204 if (thread == 0) { 8205 return DEAD_OBJECT; 8206 } 8207 8208 // TODO: handle configuration of effects replacing track process 8209 if (thread->channelCount() == 1) { 8210 channels = AUDIO_CHANNEL_OUT_MONO; 8211 } else { 8212 channels = AUDIO_CHANNEL_OUT_STEREO; 8213 } 8214 8215 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8216 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8217 } else { 8218 mConfig.inputCfg.channels = channels; 8219 } 8220 mConfig.outputCfg.channels = channels; 8221 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8222 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8223 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8224 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8225 mConfig.inputCfg.bufferProvider.cookie = NULL; 8226 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8227 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8228 mConfig.outputCfg.bufferProvider.cookie = NULL; 8229 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8230 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8231 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8232 // Insert effect: 8233 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8234 // always overwrites output buffer: input buffer == output buffer 8235 // - in other sessions: 8236 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8237 // other effect: overwrites output buffer: input buffer == output buffer 8238 // Auxiliary effect: 8239 // accumulates in output buffer: input buffer != output buffer 8240 // Therefore: accumulate <=> input buffer != output buffer 8241 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8242 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8243 } else { 8244 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8245 } 8246 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8247 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8248 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8249 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8250 8251 ALOGV("configure() %p thread %p buffer %p framecount %d", 8252 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8253 8254 status_t cmdStatus; 8255 uint32_t size = sizeof(int); 8256 status_t status = (*mEffectInterface)->command(mEffectInterface, 8257 EFFECT_CMD_SET_CONFIG, 8258 sizeof(effect_config_t), 8259 &mConfig, 8260 &size, 8261 &cmdStatus); 8262 if (status == 0) { 8263 status = cmdStatus; 8264 } 8265 8266 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8267 (1000 * mConfig.outputCfg.buffer.frameCount); 8268 8269 return status; 8270} 8271 8272status_t AudioFlinger::EffectModule::init() 8273{ 8274 Mutex::Autolock _l(mLock); 8275 if (mEffectInterface == NULL) { 8276 return NO_INIT; 8277 } 8278 status_t cmdStatus; 8279 uint32_t size = sizeof(status_t); 8280 status_t status = (*mEffectInterface)->command(mEffectInterface, 8281 EFFECT_CMD_INIT, 8282 0, 8283 NULL, 8284 &size, 8285 &cmdStatus); 8286 if (status == 0) { 8287 status = cmdStatus; 8288 } 8289 return status; 8290} 8291 8292status_t AudioFlinger::EffectModule::start() 8293{ 8294 Mutex::Autolock _l(mLock); 8295 return start_l(); 8296} 8297 8298status_t AudioFlinger::EffectModule::start_l() 8299{ 8300 if (mEffectInterface == NULL) { 8301 return NO_INIT; 8302 } 8303 status_t cmdStatus; 8304 uint32_t size = sizeof(status_t); 8305 status_t status = (*mEffectInterface)->command(mEffectInterface, 8306 EFFECT_CMD_ENABLE, 8307 0, 8308 NULL, 8309 &size, 8310 &cmdStatus); 8311 if (status == 0) { 8312 status = cmdStatus; 8313 } 8314 if (status == 0 && 8315 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8316 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8317 sp<ThreadBase> thread = mThread.promote(); 8318 if (thread != 0) { 8319 audio_stream_t *stream = thread->stream(); 8320 if (stream != NULL) { 8321 stream->add_audio_effect(stream, mEffectInterface); 8322 } 8323 } 8324 } 8325 return status; 8326} 8327 8328status_t AudioFlinger::EffectModule::stop() 8329{ 8330 Mutex::Autolock _l(mLock); 8331 return stop_l(); 8332} 8333 8334status_t AudioFlinger::EffectModule::stop_l() 8335{ 8336 if (mEffectInterface == NULL) { 8337 return NO_INIT; 8338 } 8339 status_t cmdStatus; 8340 uint32_t size = sizeof(status_t); 8341 status_t status = (*mEffectInterface)->command(mEffectInterface, 8342 EFFECT_CMD_DISABLE, 8343 0, 8344 NULL, 8345 &size, 8346 &cmdStatus); 8347 if (status == 0) { 8348 status = cmdStatus; 8349 } 8350 if (status == 0 && 8351 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8352 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8353 sp<ThreadBase> thread = mThread.promote(); 8354 if (thread != 0) { 8355 audio_stream_t *stream = thread->stream(); 8356 if (stream != NULL) { 8357 stream->remove_audio_effect(stream, mEffectInterface); 8358 } 8359 } 8360 } 8361 return status; 8362} 8363 8364status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8365 uint32_t cmdSize, 8366 void *pCmdData, 8367 uint32_t *replySize, 8368 void *pReplyData) 8369{ 8370 Mutex::Autolock _l(mLock); 8371// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8372 8373 if (mState == DESTROYED || mEffectInterface == NULL) { 8374 return NO_INIT; 8375 } 8376 status_t status = (*mEffectInterface)->command(mEffectInterface, 8377 cmdCode, 8378 cmdSize, 8379 pCmdData, 8380 replySize, 8381 pReplyData); 8382 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8383 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8384 for (size_t i = 1; i < mHandles.size(); i++) { 8385 sp<EffectHandle> h = mHandles[i].promote(); 8386 if (h != 0) { 8387 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8388 } 8389 } 8390 } 8391 return status; 8392} 8393 8394status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8395{ 8396 8397 Mutex::Autolock _l(mLock); 8398 ALOGV("setEnabled %p enabled %d", this, enabled); 8399 8400 if (enabled != isEnabled()) { 8401 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8402 if (enabled && status != NO_ERROR) { 8403 return status; 8404 } 8405 8406 switch (mState) { 8407 // going from disabled to enabled 8408 case IDLE: 8409 mState = STARTING; 8410 break; 8411 case STOPPED: 8412 mState = RESTART; 8413 break; 8414 case STOPPING: 8415 mState = ACTIVE; 8416 break; 8417 8418 // going from enabled to disabled 8419 case RESTART: 8420 mState = STOPPED; 8421 break; 8422 case STARTING: 8423 mState = IDLE; 8424 break; 8425 case ACTIVE: 8426 mState = STOPPING; 8427 break; 8428 case DESTROYED: 8429 return NO_ERROR; // simply ignore as we are being destroyed 8430 } 8431 for (size_t i = 1; i < mHandles.size(); i++) { 8432 sp<EffectHandle> h = mHandles[i].promote(); 8433 if (h != 0) { 8434 h->setEnabled(enabled); 8435 } 8436 } 8437 } 8438 return NO_ERROR; 8439} 8440 8441bool AudioFlinger::EffectModule::isEnabled() const 8442{ 8443 switch (mState) { 8444 case RESTART: 8445 case STARTING: 8446 case ACTIVE: 8447 return true; 8448 case IDLE: 8449 case STOPPING: 8450 case STOPPED: 8451 case DESTROYED: 8452 default: 8453 return false; 8454 } 8455} 8456 8457bool AudioFlinger::EffectModule::isProcessEnabled() const 8458{ 8459 switch (mState) { 8460 case RESTART: 8461 case ACTIVE: 8462 case STOPPING: 8463 case STOPPED: 8464 return true; 8465 case IDLE: 8466 case STARTING: 8467 case DESTROYED: 8468 default: 8469 return false; 8470 } 8471} 8472 8473status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8474{ 8475 Mutex::Autolock _l(mLock); 8476 status_t status = NO_ERROR; 8477 8478 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8479 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8480 if (isProcessEnabled() && 8481 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8482 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8483 status_t cmdStatus; 8484 uint32_t volume[2]; 8485 uint32_t *pVolume = NULL; 8486 uint32_t size = sizeof(volume); 8487 volume[0] = *left; 8488 volume[1] = *right; 8489 if (controller) { 8490 pVolume = volume; 8491 } 8492 status = (*mEffectInterface)->command(mEffectInterface, 8493 EFFECT_CMD_SET_VOLUME, 8494 size, 8495 volume, 8496 &size, 8497 pVolume); 8498 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8499 *left = volume[0]; 8500 *right = volume[1]; 8501 } 8502 } 8503 return status; 8504} 8505 8506status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8507{ 8508 Mutex::Autolock _l(mLock); 8509 status_t status = NO_ERROR; 8510 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8511 // audio pre processing modules on RecordThread can receive both output and 8512 // input device indication in the same call 8513 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8514 if (dev) { 8515 status_t cmdStatus; 8516 uint32_t size = sizeof(status_t); 8517 8518 status = (*mEffectInterface)->command(mEffectInterface, 8519 EFFECT_CMD_SET_DEVICE, 8520 sizeof(uint32_t), 8521 &dev, 8522 &size, 8523 &cmdStatus); 8524 if (status == NO_ERROR) { 8525 status = cmdStatus; 8526 } 8527 } 8528 dev = device & AUDIO_DEVICE_IN_ALL; 8529 if (dev) { 8530 status_t cmdStatus; 8531 uint32_t size = sizeof(status_t); 8532 8533 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8534 EFFECT_CMD_SET_INPUT_DEVICE, 8535 sizeof(uint32_t), 8536 &dev, 8537 &size, 8538 &cmdStatus); 8539 if (status2 == NO_ERROR) { 8540 status2 = cmdStatus; 8541 } 8542 if (status == NO_ERROR) { 8543 status = status2; 8544 } 8545 } 8546 } 8547 return status; 8548} 8549 8550status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8551{ 8552 Mutex::Autolock _l(mLock); 8553 status_t status = NO_ERROR; 8554 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8555 status_t cmdStatus; 8556 uint32_t size = sizeof(status_t); 8557 status = (*mEffectInterface)->command(mEffectInterface, 8558 EFFECT_CMD_SET_AUDIO_MODE, 8559 sizeof(audio_mode_t), 8560 &mode, 8561 &size, 8562 &cmdStatus); 8563 if (status == NO_ERROR) { 8564 status = cmdStatus; 8565 } 8566 } 8567 return status; 8568} 8569 8570void AudioFlinger::EffectModule::setSuspended(bool suspended) 8571{ 8572 Mutex::Autolock _l(mLock); 8573 mSuspended = suspended; 8574} 8575 8576bool AudioFlinger::EffectModule::suspended() const 8577{ 8578 Mutex::Autolock _l(mLock); 8579 return mSuspended; 8580} 8581 8582status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8583{ 8584 const size_t SIZE = 256; 8585 char buffer[SIZE]; 8586 String8 result; 8587 8588 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8589 result.append(buffer); 8590 8591 bool locked = tryLock(mLock); 8592 // failed to lock - AudioFlinger is probably deadlocked 8593 if (!locked) { 8594 result.append("\t\tCould not lock Fx mutex:\n"); 8595 } 8596 8597 result.append("\t\tSession Status State Engine:\n"); 8598 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8599 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8600 result.append(buffer); 8601 8602 result.append("\t\tDescriptor:\n"); 8603 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8604 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8605 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8606 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8607 result.append(buffer); 8608 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8609 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8610 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8611 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8612 result.append(buffer); 8613 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8614 mDescriptor.apiVersion, 8615 mDescriptor.flags); 8616 result.append(buffer); 8617 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8618 mDescriptor.name); 8619 result.append(buffer); 8620 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8621 mDescriptor.implementor); 8622 result.append(buffer); 8623 8624 result.append("\t\t- Input configuration:\n"); 8625 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8626 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8627 (uint32_t)mConfig.inputCfg.buffer.raw, 8628 mConfig.inputCfg.buffer.frameCount, 8629 mConfig.inputCfg.samplingRate, 8630 mConfig.inputCfg.channels, 8631 mConfig.inputCfg.format); 8632 result.append(buffer); 8633 8634 result.append("\t\t- Output configuration:\n"); 8635 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8636 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8637 (uint32_t)mConfig.outputCfg.buffer.raw, 8638 mConfig.outputCfg.buffer.frameCount, 8639 mConfig.outputCfg.samplingRate, 8640 mConfig.outputCfg.channels, 8641 mConfig.outputCfg.format); 8642 result.append(buffer); 8643 8644 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8645 result.append(buffer); 8646 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8647 for (size_t i = 0; i < mHandles.size(); ++i) { 8648 sp<EffectHandle> handle = mHandles[i].promote(); 8649 if (handle != 0) { 8650 handle->dump(buffer, SIZE); 8651 result.append(buffer); 8652 } 8653 } 8654 8655 result.append("\n"); 8656 8657 write(fd, result.string(), result.length()); 8658 8659 if (locked) { 8660 mLock.unlock(); 8661 } 8662 8663 return NO_ERROR; 8664} 8665 8666// ---------------------------------------------------------------------------- 8667// EffectHandle implementation 8668// ---------------------------------------------------------------------------- 8669 8670#undef LOG_TAG 8671#define LOG_TAG "AudioFlinger::EffectHandle" 8672 8673AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8674 const sp<AudioFlinger::Client>& client, 8675 const sp<IEffectClient>& effectClient, 8676 int32_t priority) 8677 : BnEffect(), 8678 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8679 mPriority(priority), mHasControl(false), mEnabled(false) 8680{ 8681 ALOGV("constructor %p", this); 8682 8683 if (client == 0) { 8684 return; 8685 } 8686 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8687 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8688 if (mCblkMemory != 0) { 8689 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8690 8691 if (mCblk != NULL) { 8692 new(mCblk) effect_param_cblk_t(); 8693 mBuffer = (uint8_t *)mCblk + bufOffset; 8694 } 8695 } else { 8696 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8697 return; 8698 } 8699} 8700 8701AudioFlinger::EffectHandle::~EffectHandle() 8702{ 8703 ALOGV("Destructor %p", this); 8704 disconnect(false); 8705 ALOGV("Destructor DONE %p", this); 8706} 8707 8708status_t AudioFlinger::EffectHandle::enable() 8709{ 8710 ALOGV("enable %p", this); 8711 if (!mHasControl) return INVALID_OPERATION; 8712 if (mEffect == 0) return DEAD_OBJECT; 8713 8714 if (mEnabled) { 8715 return NO_ERROR; 8716 } 8717 8718 mEnabled = true; 8719 8720 sp<ThreadBase> thread = mEffect->thread().promote(); 8721 if (thread != 0) { 8722 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8723 } 8724 8725 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8726 if (mEffect->suspended()) { 8727 return NO_ERROR; 8728 } 8729 8730 status_t status = mEffect->setEnabled(true); 8731 if (status != NO_ERROR) { 8732 if (thread != 0) { 8733 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8734 } 8735 mEnabled = false; 8736 } 8737 return status; 8738} 8739 8740status_t AudioFlinger::EffectHandle::disable() 8741{ 8742 ALOGV("disable %p", this); 8743 if (!mHasControl) return INVALID_OPERATION; 8744 if (mEffect == 0) return DEAD_OBJECT; 8745 8746 if (!mEnabled) { 8747 return NO_ERROR; 8748 } 8749 mEnabled = false; 8750 8751 if (mEffect->suspended()) { 8752 return NO_ERROR; 8753 } 8754 8755 status_t status = mEffect->setEnabled(false); 8756 8757 sp<ThreadBase> thread = mEffect->thread().promote(); 8758 if (thread != 0) { 8759 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8760 } 8761 8762 return status; 8763} 8764 8765void AudioFlinger::EffectHandle::disconnect() 8766{ 8767 disconnect(true); 8768} 8769 8770void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8771{ 8772 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8773 if (mEffect == 0) { 8774 return; 8775 } 8776 mEffect->disconnect(this, unpinIfLast); 8777 8778 if (mHasControl && mEnabled) { 8779 sp<ThreadBase> thread = mEffect->thread().promote(); 8780 if (thread != 0) { 8781 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8782 } 8783 } 8784 8785 // release sp on module => module destructor can be called now 8786 mEffect.clear(); 8787 if (mClient != 0) { 8788 if (mCblk != NULL) { 8789 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8790 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8791 } 8792 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8793 // Client destructor must run with AudioFlinger mutex locked 8794 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8795 mClient.clear(); 8796 } 8797} 8798 8799status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8800 uint32_t cmdSize, 8801 void *pCmdData, 8802 uint32_t *replySize, 8803 void *pReplyData) 8804{ 8805// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8806// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8807 8808 // only get parameter command is permitted for applications not controlling the effect 8809 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8810 return INVALID_OPERATION; 8811 } 8812 if (mEffect == 0) return DEAD_OBJECT; 8813 if (mClient == 0) return INVALID_OPERATION; 8814 8815 // handle commands that are not forwarded transparently to effect engine 8816 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8817 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8818 // no risk to block the whole media server process or mixer threads is we are stuck here 8819 Mutex::Autolock _l(mCblk->lock); 8820 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8821 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8822 mCblk->serverIndex = 0; 8823 mCblk->clientIndex = 0; 8824 return BAD_VALUE; 8825 } 8826 status_t status = NO_ERROR; 8827 while (mCblk->serverIndex < mCblk->clientIndex) { 8828 int reply; 8829 uint32_t rsize = sizeof(int); 8830 int *p = (int *)(mBuffer + mCblk->serverIndex); 8831 int size = *p++; 8832 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8833 ALOGW("command(): invalid parameter block size"); 8834 break; 8835 } 8836 effect_param_t *param = (effect_param_t *)p; 8837 if (param->psize == 0 || param->vsize == 0) { 8838 ALOGW("command(): null parameter or value size"); 8839 mCblk->serverIndex += size; 8840 continue; 8841 } 8842 uint32_t psize = sizeof(effect_param_t) + 8843 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8844 param->vsize; 8845 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8846 psize, 8847 p, 8848 &rsize, 8849 &reply); 8850 // stop at first error encountered 8851 if (ret != NO_ERROR) { 8852 status = ret; 8853 *(int *)pReplyData = reply; 8854 break; 8855 } else if (reply != NO_ERROR) { 8856 *(int *)pReplyData = reply; 8857 break; 8858 } 8859 mCblk->serverIndex += size; 8860 } 8861 mCblk->serverIndex = 0; 8862 mCblk->clientIndex = 0; 8863 return status; 8864 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8865 *(int *)pReplyData = NO_ERROR; 8866 return enable(); 8867 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8868 *(int *)pReplyData = NO_ERROR; 8869 return disable(); 8870 } 8871 8872 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8873} 8874 8875void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8876{ 8877 ALOGV("setControl %p control %d", this, hasControl); 8878 8879 mHasControl = hasControl; 8880 mEnabled = enabled; 8881 8882 if (signal && mEffectClient != 0) { 8883 mEffectClient->controlStatusChanged(hasControl); 8884 } 8885} 8886 8887void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8888 uint32_t cmdSize, 8889 void *pCmdData, 8890 uint32_t replySize, 8891 void *pReplyData) 8892{ 8893 if (mEffectClient != 0) { 8894 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8895 } 8896} 8897 8898 8899 8900void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8901{ 8902 if (mEffectClient != 0) { 8903 mEffectClient->enableStatusChanged(enabled); 8904 } 8905} 8906 8907status_t AudioFlinger::EffectHandle::onTransact( 8908 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8909{ 8910 return BnEffect::onTransact(code, data, reply, flags); 8911} 8912 8913 8914void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8915{ 8916 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8917 8918 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8919 (mClient == 0) ? getpid_cached : mClient->pid(), 8920 mPriority, 8921 mHasControl, 8922 !locked, 8923 mCblk ? mCblk->clientIndex : 0, 8924 mCblk ? mCblk->serverIndex : 0 8925 ); 8926 8927 if (locked) { 8928 mCblk->lock.unlock(); 8929 } 8930} 8931 8932#undef LOG_TAG 8933#define LOG_TAG "AudioFlinger::EffectChain" 8934 8935AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8936 int sessionId) 8937 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8938 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8939 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8940{ 8941 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8942 if (thread == NULL) { 8943 return; 8944 } 8945 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8946 thread->frameCount(); 8947} 8948 8949AudioFlinger::EffectChain::~EffectChain() 8950{ 8951 if (mOwnInBuffer) { 8952 delete mInBuffer; 8953 } 8954 8955} 8956 8957// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8958sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8959{ 8960 size_t size = mEffects.size(); 8961 8962 for (size_t i = 0; i < size; i++) { 8963 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8964 return mEffects[i]; 8965 } 8966 } 8967 return 0; 8968} 8969 8970// getEffectFromId_l() must be called with ThreadBase::mLock held 8971sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8972{ 8973 size_t size = mEffects.size(); 8974 8975 for (size_t i = 0; i < size; i++) { 8976 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8977 if (id == 0 || mEffects[i]->id() == id) { 8978 return mEffects[i]; 8979 } 8980 } 8981 return 0; 8982} 8983 8984// getEffectFromType_l() must be called with ThreadBase::mLock held 8985sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8986 const effect_uuid_t *type) 8987{ 8988 size_t size = mEffects.size(); 8989 8990 for (size_t i = 0; i < size; i++) { 8991 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8992 return mEffects[i]; 8993 } 8994 } 8995 return 0; 8996} 8997 8998void AudioFlinger::EffectChain::clearInputBuffer() 8999{ 9000 Mutex::Autolock _l(mLock); 9001 sp<ThreadBase> thread = mThread.promote(); 9002 if (thread == 0) { 9003 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9004 return; 9005 } 9006 clearInputBuffer_l(thread); 9007} 9008 9009// Must be called with EffectChain::mLock locked 9010void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9011{ 9012 size_t numSamples = thread->frameCount() * thread->channelCount(); 9013 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9014 9015} 9016 9017// Must be called with EffectChain::mLock locked 9018void AudioFlinger::EffectChain::process_l() 9019{ 9020 sp<ThreadBase> thread = mThread.promote(); 9021 if (thread == 0) { 9022 ALOGW("process_l(): cannot promote mixer thread"); 9023 return; 9024 } 9025 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9026 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9027 // always process effects unless no more tracks are on the session and the effect tail 9028 // has been rendered 9029 bool doProcess = true; 9030 if (!isGlobalSession) { 9031 bool tracksOnSession = (trackCnt() != 0); 9032 9033 if (!tracksOnSession && mTailBufferCount == 0) { 9034 doProcess = false; 9035 } 9036 9037 if (activeTrackCnt() == 0) { 9038 // if no track is active and the effect tail has not been rendered, 9039 // the input buffer must be cleared here as the mixer process will not do it 9040 if (tracksOnSession || mTailBufferCount > 0) { 9041 clearInputBuffer_l(thread); 9042 if (mTailBufferCount > 0) { 9043 mTailBufferCount--; 9044 } 9045 } 9046 } 9047 } 9048 9049 size_t size = mEffects.size(); 9050 if (doProcess) { 9051 for (size_t i = 0; i < size; i++) { 9052 mEffects[i]->process(); 9053 } 9054 } 9055 for (size_t i = 0; i < size; i++) { 9056 mEffects[i]->updateState(); 9057 } 9058} 9059 9060// addEffect_l() must be called with PlaybackThread::mLock held 9061status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9062{ 9063 effect_descriptor_t desc = effect->desc(); 9064 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9065 9066 Mutex::Autolock _l(mLock); 9067 effect->setChain(this); 9068 sp<ThreadBase> thread = mThread.promote(); 9069 if (thread == 0) { 9070 return NO_INIT; 9071 } 9072 effect->setThread(thread); 9073 9074 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9075 // Auxiliary effects are inserted at the beginning of mEffects vector as 9076 // they are processed first and accumulated in chain input buffer 9077 mEffects.insertAt(effect, 0); 9078 9079 // the input buffer for auxiliary effect contains mono samples in 9080 // 32 bit format. This is to avoid saturation in AudoMixer 9081 // accumulation stage. Saturation is done in EffectModule::process() before 9082 // calling the process in effect engine 9083 size_t numSamples = thread->frameCount(); 9084 int32_t *buffer = new int32_t[numSamples]; 9085 memset(buffer, 0, numSamples * sizeof(int32_t)); 9086 effect->setInBuffer((int16_t *)buffer); 9087 // auxiliary effects output samples to chain input buffer for further processing 9088 // by insert effects 9089 effect->setOutBuffer(mInBuffer); 9090 } else { 9091 // Insert effects are inserted at the end of mEffects vector as they are processed 9092 // after track and auxiliary effects. 9093 // Insert effect order as a function of indicated preference: 9094 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9095 // another effect is present 9096 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9097 // last effect claiming first position 9098 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9099 // first effect claiming last position 9100 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9101 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9102 // already present 9103 9104 size_t size = mEffects.size(); 9105 size_t idx_insert = size; 9106 ssize_t idx_insert_first = -1; 9107 ssize_t idx_insert_last = -1; 9108 9109 for (size_t i = 0; i < size; i++) { 9110 effect_descriptor_t d = mEffects[i]->desc(); 9111 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9112 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9113 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9114 // check invalid effect chaining combinations 9115 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9116 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9117 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9118 return INVALID_OPERATION; 9119 } 9120 // remember position of first insert effect and by default 9121 // select this as insert position for new effect 9122 if (idx_insert == size) { 9123 idx_insert = i; 9124 } 9125 // remember position of last insert effect claiming 9126 // first position 9127 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9128 idx_insert_first = i; 9129 } 9130 // remember position of first insert effect claiming 9131 // last position 9132 if (iPref == EFFECT_FLAG_INSERT_LAST && 9133 idx_insert_last == -1) { 9134 idx_insert_last = i; 9135 } 9136 } 9137 } 9138 9139 // modify idx_insert from first position if needed 9140 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9141 if (idx_insert_last != -1) { 9142 idx_insert = idx_insert_last; 9143 } else { 9144 idx_insert = size; 9145 } 9146 } else { 9147 if (idx_insert_first != -1) { 9148 idx_insert = idx_insert_first + 1; 9149 } 9150 } 9151 9152 // always read samples from chain input buffer 9153 effect->setInBuffer(mInBuffer); 9154 9155 // if last effect in the chain, output samples to chain 9156 // output buffer, otherwise to chain input buffer 9157 if (idx_insert == size) { 9158 if (idx_insert != 0) { 9159 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9160 mEffects[idx_insert-1]->configure(); 9161 } 9162 effect->setOutBuffer(mOutBuffer); 9163 } else { 9164 effect->setOutBuffer(mInBuffer); 9165 } 9166 mEffects.insertAt(effect, idx_insert); 9167 9168 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9169 } 9170 effect->configure(); 9171 return NO_ERROR; 9172} 9173 9174// removeEffect_l() must be called with PlaybackThread::mLock held 9175size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9176{ 9177 Mutex::Autolock _l(mLock); 9178 size_t size = mEffects.size(); 9179 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9180 9181 for (size_t i = 0; i < size; i++) { 9182 if (effect == mEffects[i]) { 9183 // calling stop here will remove pre-processing effect from the audio HAL. 9184 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9185 // the middle of a read from audio HAL 9186 if (mEffects[i]->state() == EffectModule::ACTIVE || 9187 mEffects[i]->state() == EffectModule::STOPPING) { 9188 mEffects[i]->stop(); 9189 } 9190 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9191 delete[] effect->inBuffer(); 9192 } else { 9193 if (i == size - 1 && i != 0) { 9194 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9195 mEffects[i - 1]->configure(); 9196 } 9197 } 9198 mEffects.removeAt(i); 9199 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9200 break; 9201 } 9202 } 9203 9204 return mEffects.size(); 9205} 9206 9207// setDevice_l() must be called with PlaybackThread::mLock held 9208void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9209{ 9210 size_t size = mEffects.size(); 9211 for (size_t i = 0; i < size; i++) { 9212 mEffects[i]->setDevice(device); 9213 } 9214} 9215 9216// setMode_l() must be called with PlaybackThread::mLock held 9217void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9218{ 9219 size_t size = mEffects.size(); 9220 for (size_t i = 0; i < size; i++) { 9221 mEffects[i]->setMode(mode); 9222 } 9223} 9224 9225// setVolume_l() must be called with PlaybackThread::mLock held 9226bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9227{ 9228 uint32_t newLeft = *left; 9229 uint32_t newRight = *right; 9230 bool hasControl = false; 9231 int ctrlIdx = -1; 9232 size_t size = mEffects.size(); 9233 9234 // first update volume controller 9235 for (size_t i = size; i > 0; i--) { 9236 if (mEffects[i - 1]->isProcessEnabled() && 9237 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9238 ctrlIdx = i - 1; 9239 hasControl = true; 9240 break; 9241 } 9242 } 9243 9244 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9245 if (hasControl) { 9246 *left = mNewLeftVolume; 9247 *right = mNewRightVolume; 9248 } 9249 return hasControl; 9250 } 9251 9252 mVolumeCtrlIdx = ctrlIdx; 9253 mLeftVolume = newLeft; 9254 mRightVolume = newRight; 9255 9256 // second get volume update from volume controller 9257 if (ctrlIdx >= 0) { 9258 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9259 mNewLeftVolume = newLeft; 9260 mNewRightVolume = newRight; 9261 } 9262 // then indicate volume to all other effects in chain. 9263 // Pass altered volume to effects before volume controller 9264 // and requested volume to effects after controller 9265 uint32_t lVol = newLeft; 9266 uint32_t rVol = newRight; 9267 9268 for (size_t i = 0; i < size; i++) { 9269 if ((int)i == ctrlIdx) continue; 9270 // this also works for ctrlIdx == -1 when there is no volume controller 9271 if ((int)i > ctrlIdx) { 9272 lVol = *left; 9273 rVol = *right; 9274 } 9275 mEffects[i]->setVolume(&lVol, &rVol, false); 9276 } 9277 *left = newLeft; 9278 *right = newRight; 9279 9280 return hasControl; 9281} 9282 9283status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9284{ 9285 const size_t SIZE = 256; 9286 char buffer[SIZE]; 9287 String8 result; 9288 9289 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9290 result.append(buffer); 9291 9292 bool locked = tryLock(mLock); 9293 // failed to lock - AudioFlinger is probably deadlocked 9294 if (!locked) { 9295 result.append("\tCould not lock mutex:\n"); 9296 } 9297 9298 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9299 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9300 mEffects.size(), 9301 (uint32_t)mInBuffer, 9302 (uint32_t)mOutBuffer, 9303 mActiveTrackCnt); 9304 result.append(buffer); 9305 write(fd, result.string(), result.size()); 9306 9307 for (size_t i = 0; i < mEffects.size(); ++i) { 9308 sp<EffectModule> effect = mEffects[i]; 9309 if (effect != 0) { 9310 effect->dump(fd, args); 9311 } 9312 } 9313 9314 if (locked) { 9315 mLock.unlock(); 9316 } 9317 9318 return NO_ERROR; 9319} 9320 9321// must be called with ThreadBase::mLock held 9322void AudioFlinger::EffectChain::setEffectSuspended_l( 9323 const effect_uuid_t *type, bool suspend) 9324{ 9325 sp<SuspendedEffectDesc> desc; 9326 // use effect type UUID timelow as key as there is no real risk of identical 9327 // timeLow fields among effect type UUIDs. 9328 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9329 if (suspend) { 9330 if (index >= 0) { 9331 desc = mSuspendedEffects.valueAt(index); 9332 } else { 9333 desc = new SuspendedEffectDesc(); 9334 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9335 mSuspendedEffects.add(type->timeLow, desc); 9336 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9337 } 9338 if (desc->mRefCount++ == 0) { 9339 sp<EffectModule> effect = getEffectIfEnabled(type); 9340 if (effect != 0) { 9341 desc->mEffect = effect; 9342 effect->setSuspended(true); 9343 effect->setEnabled(false); 9344 } 9345 } 9346 } else { 9347 if (index < 0) { 9348 return; 9349 } 9350 desc = mSuspendedEffects.valueAt(index); 9351 if (desc->mRefCount <= 0) { 9352 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9353 desc->mRefCount = 1; 9354 } 9355 if (--desc->mRefCount == 0) { 9356 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9357 if (desc->mEffect != 0) { 9358 sp<EffectModule> effect = desc->mEffect.promote(); 9359 if (effect != 0) { 9360 effect->setSuspended(false); 9361 sp<EffectHandle> handle = effect->controlHandle(); 9362 if (handle != 0) { 9363 effect->setEnabled(handle->enabled()); 9364 } 9365 } 9366 desc->mEffect.clear(); 9367 } 9368 mSuspendedEffects.removeItemsAt(index); 9369 } 9370 } 9371} 9372 9373// must be called with ThreadBase::mLock held 9374void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9375{ 9376 sp<SuspendedEffectDesc> desc; 9377 9378 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9379 if (suspend) { 9380 if (index >= 0) { 9381 desc = mSuspendedEffects.valueAt(index); 9382 } else { 9383 desc = new SuspendedEffectDesc(); 9384 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9385 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9386 } 9387 if (desc->mRefCount++ == 0) { 9388 Vector< sp<EffectModule> > effects; 9389 getSuspendEligibleEffects(effects); 9390 for (size_t i = 0; i < effects.size(); i++) { 9391 setEffectSuspended_l(&effects[i]->desc().type, true); 9392 } 9393 } 9394 } else { 9395 if (index < 0) { 9396 return; 9397 } 9398 desc = mSuspendedEffects.valueAt(index); 9399 if (desc->mRefCount <= 0) { 9400 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9401 desc->mRefCount = 1; 9402 } 9403 if (--desc->mRefCount == 0) { 9404 Vector<const effect_uuid_t *> types; 9405 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9406 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9407 continue; 9408 } 9409 types.add(&mSuspendedEffects.valueAt(i)->mType); 9410 } 9411 for (size_t i = 0; i < types.size(); i++) { 9412 setEffectSuspended_l(types[i], false); 9413 } 9414 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9415 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9416 } 9417 } 9418} 9419 9420 9421// The volume effect is used for automated tests only 9422#ifndef OPENSL_ES_H_ 9423static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9424 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9425const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9426#endif //OPENSL_ES_H_ 9427 9428bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9429{ 9430 // auxiliary effects and visualizer are never suspended on output mix 9431 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9432 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9433 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9434 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9435 return false; 9436 } 9437 return true; 9438} 9439 9440void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9441{ 9442 effects.clear(); 9443 for (size_t i = 0; i < mEffects.size(); i++) { 9444 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9445 effects.add(mEffects[i]); 9446 } 9447 } 9448} 9449 9450sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9451 const effect_uuid_t *type) 9452{ 9453 sp<EffectModule> effect = getEffectFromType_l(type); 9454 return effect != 0 && effect->isEnabled() ? effect : 0; 9455} 9456 9457void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9458 bool enabled) 9459{ 9460 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9461 if (enabled) { 9462 if (index < 0) { 9463 // if the effect is not suspend check if all effects are suspended 9464 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9465 if (index < 0) { 9466 return; 9467 } 9468 if (!isEffectEligibleForSuspend(effect->desc())) { 9469 return; 9470 } 9471 setEffectSuspended_l(&effect->desc().type, enabled); 9472 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9473 if (index < 0) { 9474 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9475 return; 9476 } 9477 } 9478 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9479 effect->desc().type.timeLow); 9480 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9481 // if effect is requested to suspended but was not yet enabled, supend it now. 9482 if (desc->mEffect == 0) { 9483 desc->mEffect = effect; 9484 effect->setEnabled(false); 9485 effect->setSuspended(true); 9486 } 9487 } else { 9488 if (index < 0) { 9489 return; 9490 } 9491 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9492 effect->desc().type.timeLow); 9493 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9494 desc->mEffect.clear(); 9495 effect->setSuspended(false); 9496 } 9497} 9498 9499#undef LOG_TAG 9500#define LOG_TAG "AudioFlinger" 9501 9502// ---------------------------------------------------------------------------- 9503 9504status_t AudioFlinger::onTransact( 9505 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9506{ 9507 return BnAudioFlinger::onTransact(code, data, reply, flags); 9508} 9509 9510}; // namespace android 9511