AudioFlinger.cpp revision 35d7bfc359b3aa87ade92d1ab55c6992418cad48
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 1609 dumpBase(fd, args); 1610 1611 return NO_ERROR; 1612} 1613 1614// Thread virtuals 1615status_t AudioFlinger::PlaybackThread::readyToRun() 1616{ 1617 status_t status = initCheck(); 1618 if (status == NO_ERROR) { 1619 ALOGI("AudioFlinger's thread %p ready to run", this); 1620 } else { 1621 ALOGE("No working audio driver found."); 1622 } 1623 return status; 1624} 1625 1626void AudioFlinger::PlaybackThread::onFirstRef() 1627{ 1628 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1629} 1630 1631// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1632sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1633 const sp<AudioFlinger::Client>& client, 1634 audio_stream_type_t streamType, 1635 uint32_t sampleRate, 1636 audio_format_t format, 1637 uint32_t channelMask, 1638 int frameCount, 1639 const sp<IMemory>& sharedBuffer, 1640 int sessionId, 1641 IAudioFlinger::track_flags_t flags, 1642 pid_t tid, 1643 status_t *status) 1644{ 1645 sp<Track> track; 1646 status_t lStatus; 1647 1648 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1649 1650 // client expresses a preference for FAST, but we get the final say 1651 if (flags & IAudioFlinger::TRACK_FAST) { 1652 if ( 1653 // not timed 1654 (!isTimed) && 1655 // either of these use cases: 1656 ( 1657 // use case 1: shared buffer with any frame count 1658 ( 1659 (sharedBuffer != 0) 1660 ) || 1661 // use case 2: callback handler and frame count is default or at least as large as HAL 1662 ( 1663 (tid != -1) && 1664 ((frameCount == 0) || 1665 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1666 ) 1667 ) && 1668 // PCM data 1669 audio_is_linear_pcm(format) && 1670 // mono or stereo 1671 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1672 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1673#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1674 // hardware sample rate 1675 (sampleRate == mSampleRate) && 1676#endif 1677 // normal mixer has an associated fast mixer 1678 hasFastMixer() && 1679 // there are sufficient fast track slots available 1680 (mFastTrackAvailMask != 0) 1681 // FIXME test that MixerThread for this fast track has a capable output HAL 1682 // FIXME add a permission test also? 1683 ) { 1684 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1685 if (frameCount == 0) { 1686 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1687 } 1688 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1689 frameCount, mFrameCount); 1690 } else { 1691 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1692 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1693 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1694 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1695 audio_is_linear_pcm(format), 1696 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1697 flags &= ~IAudioFlinger::TRACK_FAST; 1698 // For compatibility with AudioTrack calculation, buffer depth is forced 1699 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1700 // This is probably too conservative, but legacy application code may depend on it. 1701 // If you change this calculation, also review the start threshold which is related. 1702 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1703 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1704 if (minBufCount < 2) { 1705 minBufCount = 2; 1706 } 1707 int minFrameCount = mNormalFrameCount * minBufCount; 1708 if (frameCount < minFrameCount) { 1709 frameCount = minFrameCount; 1710 } 1711 } 1712 } 1713 1714 if (mType == DIRECT) { 1715 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1716 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1717 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1718 "for output %p with format %d", 1719 sampleRate, format, channelMask, mOutput, mFormat); 1720 lStatus = BAD_VALUE; 1721 goto Exit; 1722 } 1723 } 1724 } else { 1725 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1726 if (sampleRate > mSampleRate*2) { 1727 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1728 lStatus = BAD_VALUE; 1729 goto Exit; 1730 } 1731 } 1732 1733 lStatus = initCheck(); 1734 if (lStatus != NO_ERROR) { 1735 ALOGE("Audio driver not initialized."); 1736 goto Exit; 1737 } 1738 1739 { // scope for mLock 1740 Mutex::Autolock _l(mLock); 1741 1742 // all tracks in same audio session must share the same routing strategy otherwise 1743 // conflicts will happen when tracks are moved from one output to another by audio policy 1744 // manager 1745 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1746 for (size_t i = 0; i < mTracks.size(); ++i) { 1747 sp<Track> t = mTracks[i]; 1748 if (t != 0 && !t->isOutputTrack()) { 1749 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1750 if (sessionId == t->sessionId() && strategy != actual) { 1751 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1752 strategy, actual); 1753 lStatus = BAD_VALUE; 1754 goto Exit; 1755 } 1756 } 1757 } 1758 1759 if (!isTimed) { 1760 track = new Track(this, client, streamType, sampleRate, format, 1761 channelMask, frameCount, sharedBuffer, sessionId, flags); 1762 } else { 1763 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1764 channelMask, frameCount, sharedBuffer, sessionId); 1765 } 1766 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1767 lStatus = NO_MEMORY; 1768 goto Exit; 1769 } 1770 mTracks.add(track); 1771 1772 sp<EffectChain> chain = getEffectChain_l(sessionId); 1773 if (chain != 0) { 1774 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1775 track->setMainBuffer(chain->inBuffer()); 1776 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1777 chain->incTrackCnt(); 1778 } 1779 } 1780 1781#ifdef HAVE_REQUEST_PRIORITY 1782 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1783 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1784 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1785 // so ask activity manager to do this on our behalf 1786 int err = requestPriority(callingPid, tid, 1); 1787 if (err != 0) { 1788 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1789 1, callingPid, tid, err); 1790 } 1791 } 1792#endif 1793 1794 lStatus = NO_ERROR; 1795 1796Exit: 1797 if (status) { 1798 *status = lStatus; 1799 } 1800 return track; 1801} 1802 1803uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1804{ 1805 if (mFastMixer != NULL) { 1806 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1807 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1808 } 1809 return latency; 1810} 1811 1812uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1813{ 1814 return latency; 1815} 1816 1817uint32_t AudioFlinger::PlaybackThread::latency() const 1818{ 1819 Mutex::Autolock _l(mLock); 1820 if (initCheck() == NO_ERROR) { 1821 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1822 } else { 1823 return 0; 1824 } 1825} 1826 1827void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1828{ 1829 Mutex::Autolock _l(mLock); 1830 mMasterVolume = value; 1831} 1832 1833void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1834{ 1835 Mutex::Autolock _l(mLock); 1836 setMasterMute_l(muted); 1837} 1838 1839void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 mStreamTypes[stream].volume = value; 1843} 1844 1845void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 mStreamTypes[stream].mute = muted; 1849} 1850 1851float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1852{ 1853 Mutex::Autolock _l(mLock); 1854 return mStreamTypes[stream].volume; 1855} 1856 1857// addTrack_l() must be called with ThreadBase::mLock held 1858status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1859{ 1860 status_t status = ALREADY_EXISTS; 1861 1862 // set retry count for buffer fill 1863 track->mRetryCount = kMaxTrackStartupRetries; 1864 if (mActiveTracks.indexOf(track) < 0) { 1865 // the track is newly added, make sure it fills up all its 1866 // buffers before playing. This is to ensure the client will 1867 // effectively get the latency it requested. 1868 track->mFillingUpStatus = Track::FS_FILLING; 1869 track->mResetDone = false; 1870 track->mPresentationCompleteFrames = 0; 1871 mActiveTracks.add(track); 1872 if (track->mainBuffer() != mMixBuffer) { 1873 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1874 if (chain != 0) { 1875 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1876 chain->incActiveTrackCnt(); 1877 } 1878 } 1879 1880 status = NO_ERROR; 1881 } 1882 1883 ALOGV("mWaitWorkCV.broadcast"); 1884 mWaitWorkCV.broadcast(); 1885 1886 return status; 1887} 1888 1889// destroyTrack_l() must be called with ThreadBase::mLock held 1890void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1891{ 1892 track->mState = TrackBase::TERMINATED; 1893 // active tracks are removed by threadLoop() 1894 if (mActiveTracks.indexOf(track) < 0) { 1895 removeTrack_l(track); 1896 } 1897} 1898 1899void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1900{ 1901 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1902 mTracks.remove(track); 1903 deleteTrackName_l(track->name()); 1904 // redundant as track is about to be destroyed, for dumpsys only 1905 track->mName = -1; 1906 if (track->isFastTrack()) { 1907 int index = track->mFastIndex; 1908 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1909 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1910 mFastTrackAvailMask |= 1 << index; 1911 // redundant as track is about to be destroyed, for dumpsys only 1912 track->mFastIndex = -1; 1913 } 1914 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1915 if (chain != 0) { 1916 chain->decTrackCnt(); 1917 } 1918} 1919 1920String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1921{ 1922 String8 out_s8 = String8(""); 1923 char *s; 1924 1925 Mutex::Autolock _l(mLock); 1926 if (initCheck() != NO_ERROR) { 1927 return out_s8; 1928 } 1929 1930 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1931 out_s8 = String8(s); 1932 free(s); 1933 return out_s8; 1934} 1935 1936// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1937void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1938 AudioSystem::OutputDescriptor desc; 1939 void *param2 = NULL; 1940 1941 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1942 1943 switch (event) { 1944 case AudioSystem::OUTPUT_OPENED: 1945 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1946 desc.channels = mChannelMask; 1947 desc.samplingRate = mSampleRate; 1948 desc.format = mFormat; 1949 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1950 desc.latency = latency(); 1951 param2 = &desc; 1952 break; 1953 1954 case AudioSystem::STREAM_CONFIG_CHANGED: 1955 param2 = ¶m; 1956 case AudioSystem::OUTPUT_CLOSED: 1957 default: 1958 break; 1959 } 1960 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1961} 1962 1963void AudioFlinger::PlaybackThread::readOutputParameters() 1964{ 1965 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1966 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1967 mChannelCount = (uint16_t)popcount(mChannelMask); 1968 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1969 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1970 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1971 if (mFrameCount & 15) { 1972 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1973 mFrameCount); 1974 } 1975 1976 // Calculate size of normal mix buffer relative to the HAL output buffer size 1977 double multiplier = 1.0; 1978 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1979 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1980 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1981 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1982 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1983 maxNormalFrameCount = maxNormalFrameCount & ~15; 1984 if (maxNormalFrameCount < minNormalFrameCount) { 1985 maxNormalFrameCount = minNormalFrameCount; 1986 } 1987 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1988 if (multiplier <= 1.0) { 1989 multiplier = 1.0; 1990 } else if (multiplier <= 2.0) { 1991 if (2 * mFrameCount <= maxNormalFrameCount) { 1992 multiplier = 2.0; 1993 } else { 1994 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1995 } 1996 } else { 1997 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1998 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1999 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2000 // FIXME this rounding up should not be done if no HAL SRC 2001 uint32_t truncMult = (uint32_t) multiplier; 2002 if ((truncMult & 1)) { 2003 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2004 ++truncMult; 2005 } 2006 } 2007 multiplier = (double) truncMult; 2008 } 2009 } 2010 mNormalFrameCount = multiplier * mFrameCount; 2011 // round up to nearest 16 frames to satisfy AudioMixer 2012 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2013 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2014 2015 // FIXME - Current mixer implementation only supports stereo output: Always 2016 // Allocate a stereo buffer even if HW output is mono. 2017 delete[] mMixBuffer; 2018 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2019 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2020 2021 // force reconfiguration of effect chains and engines to take new buffer size and audio 2022 // parameters into account 2023 // Note that mLock is not held when readOutputParameters() is called from the constructor 2024 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2025 // matter. 2026 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2027 Vector< sp<EffectChain> > effectChains = mEffectChains; 2028 for (size_t i = 0; i < effectChains.size(); i ++) { 2029 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2030 } 2031} 2032 2033 2034status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2035{ 2036 if (halFrames == NULL || dspFrames == NULL) { 2037 return BAD_VALUE; 2038 } 2039 Mutex::Autolock _l(mLock); 2040 if (initCheck() != NO_ERROR) { 2041 return INVALID_OPERATION; 2042 } 2043 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2044 2045 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2046} 2047 2048uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2049{ 2050 Mutex::Autolock _l(mLock); 2051 uint32_t result = 0; 2052 if (getEffectChain_l(sessionId) != 0) { 2053 result = EFFECT_SESSION; 2054 } 2055 2056 for (size_t i = 0; i < mTracks.size(); ++i) { 2057 sp<Track> track = mTracks[i]; 2058 if (sessionId == track->sessionId() && 2059 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2060 result |= TRACK_SESSION; 2061 break; 2062 } 2063 } 2064 2065 return result; 2066} 2067 2068uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2069{ 2070 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2071 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2072 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2073 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2074 } 2075 for (size_t i = 0; i < mTracks.size(); i++) { 2076 sp<Track> track = mTracks[i]; 2077 if (sessionId == track->sessionId() && 2078 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2079 return AudioSystem::getStrategyForStream(track->streamType()); 2080 } 2081 } 2082 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2083} 2084 2085 2086AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2087{ 2088 Mutex::Autolock _l(mLock); 2089 return mOutput; 2090} 2091 2092AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2093{ 2094 Mutex::Autolock _l(mLock); 2095 AudioStreamOut *output = mOutput; 2096 mOutput = NULL; 2097 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2098 // must push a NULL and wait for ack 2099 mOutputSink.clear(); 2100 mPipeSink.clear(); 2101 mNormalSink.clear(); 2102 return output; 2103} 2104 2105// this method must always be called either with ThreadBase mLock held or inside the thread loop 2106audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2107{ 2108 if (mOutput == NULL) { 2109 return NULL; 2110 } 2111 return &mOutput->stream->common; 2112} 2113 2114uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2115{ 2116 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2117 // decoding and transfer time. So sleeping for half of the latency would likely cause 2118 // underruns 2119 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2120 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2121 } else { 2122 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2123 } 2124} 2125 2126status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2127{ 2128 if (!isValidSyncEvent(event)) { 2129 return BAD_VALUE; 2130 } 2131 2132 Mutex::Autolock _l(mLock); 2133 2134 for (size_t i = 0; i < mTracks.size(); ++i) { 2135 sp<Track> track = mTracks[i]; 2136 if (event->triggerSession() == track->sessionId()) { 2137 track->setSyncEvent(event); 2138 return NO_ERROR; 2139 } 2140 } 2141 2142 return NAME_NOT_FOUND; 2143} 2144 2145bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2146{ 2147 switch (event->type()) { 2148 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2149 return true; 2150 default: 2151 break; 2152 } 2153 return false; 2154} 2155 2156void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2157{ 2158 size_t count = tracksToRemove.size(); 2159 if (CC_UNLIKELY(count)) { 2160 for (size_t i = 0 ; i < count ; i++) { 2161 const sp<Track>& track = tracksToRemove.itemAt(i); 2162 if ((track->sharedBuffer() != 0) && 2163 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2164 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2165 } 2166 } 2167 } 2168 2169} 2170 2171// ---------------------------------------------------------------------------- 2172 2173AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2174 audio_io_handle_t id, uint32_t device, type_t type) 2175 : PlaybackThread(audioFlinger, output, id, device, type), 2176 // mAudioMixer below 2177#ifdef SOAKER 2178 mSoaker(NULL), 2179#endif 2180 // mFastMixer below 2181 mFastMixerFutex(0) 2182 // mOutputSink below 2183 // mPipeSink below 2184 // mNormalSink below 2185{ 2186 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2187 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2188 "mFrameCount=%d, mNormalFrameCount=%d", 2189 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2190 mNormalFrameCount); 2191 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2192 2193 // FIXME - Current mixer implementation only supports stereo output 2194 if (mChannelCount == 1) { 2195 ALOGE("Invalid audio hardware channel count"); 2196 } 2197 2198 // create an NBAIO sink for the HAL output stream, and negotiate 2199 mOutputSink = new AudioStreamOutSink(output->stream); 2200 size_t numCounterOffers = 0; 2201 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2202 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2203 ALOG_ASSERT(index == 0); 2204 2205 // initialize fast mixer depending on configuration 2206 bool initFastMixer; 2207 switch (kUseFastMixer) { 2208 case FastMixer_Never: 2209 initFastMixer = false; 2210 break; 2211 case FastMixer_Always: 2212 initFastMixer = true; 2213 break; 2214 case FastMixer_Static: 2215 case FastMixer_Dynamic: 2216 initFastMixer = mFrameCount < mNormalFrameCount; 2217 break; 2218 } 2219 if (initFastMixer) { 2220 2221 // create a MonoPipe to connect our submix to FastMixer 2222 NBAIO_Format format = mOutputSink->format(); 2223 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2224 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2225 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2226 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2227 const NBAIO_Format offers[1] = {format}; 2228 size_t numCounterOffers = 0; 2229 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2230 ALOG_ASSERT(index == 0); 2231 mPipeSink = monoPipe; 2232 2233#ifdef TEE_SINK_FRAMES 2234 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2235 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2236 numCounterOffers = 0; 2237 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2238 ALOG_ASSERT(index == 0); 2239 mTeeSink = teeSink; 2240 PipeReader *teeSource = new PipeReader(*teeSink); 2241 numCounterOffers = 0; 2242 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2243 ALOG_ASSERT(index == 0); 2244 mTeeSource = teeSource; 2245#endif 2246 2247#ifdef SOAKER 2248 // create a soaker as workaround for governor issues 2249 mSoaker = new Soaker(); 2250 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2251 mSoaker->run("Soaker", PRIORITY_LOWEST); 2252#endif 2253 2254 // create fast mixer and configure it initially with just one fast track for our submix 2255 mFastMixer = new FastMixer(); 2256 FastMixerStateQueue *sq = mFastMixer->sq(); 2257 FastMixerState *state = sq->begin(); 2258 FastTrack *fastTrack = &state->mFastTracks[0]; 2259 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2260 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2261 fastTrack->mVolumeProvider = NULL; 2262 fastTrack->mGeneration++; 2263 state->mFastTracksGen++; 2264 state->mTrackMask = 1; 2265 // fast mixer will use the HAL output sink 2266 state->mOutputSink = mOutputSink.get(); 2267 state->mOutputSinkGen++; 2268 state->mFrameCount = mFrameCount; 2269 state->mCommand = FastMixerState::COLD_IDLE; 2270 // already done in constructor initialization list 2271 //mFastMixerFutex = 0; 2272 state->mColdFutexAddr = &mFastMixerFutex; 2273 state->mColdGen++; 2274 state->mDumpState = &mFastMixerDumpState; 2275 state->mTeeSink = mTeeSink.get(); 2276 sq->end(); 2277 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2278 2279 // start the fast mixer 2280 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2281#ifdef HAVE_REQUEST_PRIORITY 2282 pid_t tid = mFastMixer->getTid(); 2283 int err = requestPriority(getpid_cached, tid, 2); 2284 if (err != 0) { 2285 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2286 2, getpid_cached, tid, err); 2287 } 2288#endif 2289 2290 } else { 2291 mFastMixer = NULL; 2292 } 2293 2294 switch (kUseFastMixer) { 2295 case FastMixer_Never: 2296 case FastMixer_Dynamic: 2297 mNormalSink = mOutputSink; 2298 break; 2299 case FastMixer_Always: 2300 mNormalSink = mPipeSink; 2301 break; 2302 case FastMixer_Static: 2303 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2304 break; 2305 } 2306} 2307 2308AudioFlinger::MixerThread::~MixerThread() 2309{ 2310 if (mFastMixer != NULL) { 2311 FastMixerStateQueue *sq = mFastMixer->sq(); 2312 FastMixerState *state = sq->begin(); 2313 if (state->mCommand == FastMixerState::COLD_IDLE) { 2314 int32_t old = android_atomic_inc(&mFastMixerFutex); 2315 if (old == -1) { 2316 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2317 } 2318 } 2319 state->mCommand = FastMixerState::EXIT; 2320 sq->end(); 2321 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2322 mFastMixer->join(); 2323 // Though the fast mixer thread has exited, it's state queue is still valid. 2324 // We'll use that extract the final state which contains one remaining fast track 2325 // corresponding to our sub-mix. 2326 state = sq->begin(); 2327 ALOG_ASSERT(state->mTrackMask == 1); 2328 FastTrack *fastTrack = &state->mFastTracks[0]; 2329 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2330 delete fastTrack->mBufferProvider; 2331 sq->end(false /*didModify*/); 2332 delete mFastMixer; 2333#ifdef SOAKER 2334 if (mSoaker != NULL) { 2335 mSoaker->requestExitAndWait(); 2336 } 2337 delete mSoaker; 2338#endif 2339 } 2340 delete mAudioMixer; 2341} 2342 2343class CpuStats { 2344public: 2345 CpuStats(); 2346 void sample(const String8 &title); 2347#ifdef DEBUG_CPU_USAGE 2348private: 2349 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2350 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2351 2352 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2353 2354 int mCpuNum; // thread's current CPU number 2355 int mCpukHz; // frequency of thread's current CPU in kHz 2356#endif 2357}; 2358 2359CpuStats::CpuStats() 2360#ifdef DEBUG_CPU_USAGE 2361 : mCpuNum(-1), mCpukHz(-1) 2362#endif 2363{ 2364} 2365 2366void CpuStats::sample(const String8 &title) { 2367#ifdef DEBUG_CPU_USAGE 2368 // get current thread's delta CPU time in wall clock ns 2369 double wcNs; 2370 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2371 2372 // record sample for wall clock statistics 2373 if (valid) { 2374 mWcStats.sample(wcNs); 2375 } 2376 2377 // get the current CPU number 2378 int cpuNum = sched_getcpu(); 2379 2380 // get the current CPU frequency in kHz 2381 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2382 2383 // check if either CPU number or frequency changed 2384 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2385 mCpuNum = cpuNum; 2386 mCpukHz = cpukHz; 2387 // ignore sample for purposes of cycles 2388 valid = false; 2389 } 2390 2391 // if no change in CPU number or frequency, then record sample for cycle statistics 2392 if (valid && mCpukHz > 0) { 2393 double cycles = wcNs * cpukHz * 0.000001; 2394 mHzStats.sample(cycles); 2395 } 2396 2397 unsigned n = mWcStats.n(); 2398 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2399 if ((n & 127) == 1) { 2400 long long elapsed = mCpuUsage.elapsed(); 2401 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2402 double perLoop = elapsed / (double) n; 2403 double perLoop100 = perLoop * 0.01; 2404 double perLoop1k = perLoop * 0.001; 2405 double mean = mWcStats.mean(); 2406 double stddev = mWcStats.stddev(); 2407 double minimum = mWcStats.minimum(); 2408 double maximum = mWcStats.maximum(); 2409 double meanCycles = mHzStats.mean(); 2410 double stddevCycles = mHzStats.stddev(); 2411 double minCycles = mHzStats.minimum(); 2412 double maxCycles = mHzStats.maximum(); 2413 mCpuUsage.resetElapsed(); 2414 mWcStats.reset(); 2415 mHzStats.reset(); 2416 ALOGD("CPU usage for %s over past %.1f secs\n" 2417 " (%u mixer loops at %.1f mean ms per loop):\n" 2418 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2419 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2420 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2421 title.string(), 2422 elapsed * .000000001, n, perLoop * .000001, 2423 mean * .001, 2424 stddev * .001, 2425 minimum * .001, 2426 maximum * .001, 2427 mean / perLoop100, 2428 stddev / perLoop100, 2429 minimum / perLoop100, 2430 maximum / perLoop100, 2431 meanCycles / perLoop1k, 2432 stddevCycles / perLoop1k, 2433 minCycles / perLoop1k, 2434 maxCycles / perLoop1k); 2435 2436 } 2437 } 2438#endif 2439}; 2440 2441void AudioFlinger::PlaybackThread::checkSilentMode_l() 2442{ 2443 if (!mMasterMute) { 2444 char value[PROPERTY_VALUE_MAX]; 2445 if (property_get("ro.audio.silent", value, "0") > 0) { 2446 char *endptr; 2447 unsigned long ul = strtoul(value, &endptr, 0); 2448 if (*endptr == '\0' && ul != 0) { 2449 ALOGD("Silence is golden"); 2450 // The setprop command will not allow a property to be changed after 2451 // the first time it is set, so we don't have to worry about un-muting. 2452 setMasterMute_l(true); 2453 } 2454 } 2455 } 2456} 2457 2458bool AudioFlinger::PlaybackThread::threadLoop() 2459{ 2460 Vector< sp<Track> > tracksToRemove; 2461 2462 standbyTime = systemTime(); 2463 2464 // MIXER 2465 nsecs_t lastWarning = 0; 2466if (mType == MIXER) { 2467 longStandbyExit = false; 2468} 2469 2470 // DUPLICATING 2471 // FIXME could this be made local to while loop? 2472 writeFrames = 0; 2473 2474 cacheParameters_l(); 2475 sleepTime = idleSleepTime; 2476 2477if (mType == MIXER) { 2478 sleepTimeShift = 0; 2479} 2480 2481 CpuStats cpuStats; 2482 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2483 2484 acquireWakeLock(); 2485 2486 while (!exitPending()) 2487 { 2488 cpuStats.sample(myName); 2489 2490 Vector< sp<EffectChain> > effectChains; 2491 2492 processConfigEvents(); 2493 2494 { // scope for mLock 2495 2496 Mutex::Autolock _l(mLock); 2497 2498 if (checkForNewParameters_l()) { 2499 cacheParameters_l(); 2500 } 2501 2502 saveOutputTracks(); 2503 2504 // put audio hardware into standby after short delay 2505 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2506 mSuspended > 0)) { 2507 if (!mStandby) { 2508 2509 threadLoop_standby(); 2510 2511 mStandby = true; 2512 mBytesWritten = 0; 2513 } 2514 2515 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2516 // we're about to wait, flush the binder command buffer 2517 IPCThreadState::self()->flushCommands(); 2518 2519 clearOutputTracks(); 2520 2521 if (exitPending()) break; 2522 2523 releaseWakeLock_l(); 2524 // wait until we have something to do... 2525 ALOGV("%s going to sleep", myName.string()); 2526 mWaitWorkCV.wait(mLock); 2527 ALOGV("%s waking up", myName.string()); 2528 acquireWakeLock_l(); 2529 2530 mMixerStatus = MIXER_IDLE; 2531 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2532 2533 checkSilentMode_l(); 2534 2535 standbyTime = systemTime() + standbyDelay; 2536 sleepTime = idleSleepTime; 2537 if (mType == MIXER) { 2538 sleepTimeShift = 0; 2539 } 2540 2541 continue; 2542 } 2543 } 2544 2545 // mMixerStatusIgnoringFastTracks is also updated internally 2546 mMixerStatus = prepareTracks_l(&tracksToRemove); 2547 2548 // prevent any changes in effect chain list and in each effect chain 2549 // during mixing and effect process as the audio buffers could be deleted 2550 // or modified if an effect is created or deleted 2551 lockEffectChains_l(effectChains); 2552 } 2553 2554 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2555 threadLoop_mix(); 2556 } else { 2557 threadLoop_sleepTime(); 2558 } 2559 2560 if (mSuspended > 0) { 2561 sleepTime = suspendSleepTimeUs(); 2562 } 2563 2564 // only process effects if we're going to write 2565 if (sleepTime == 0) { 2566 for (size_t i = 0; i < effectChains.size(); i ++) { 2567 effectChains[i]->process_l(); 2568 } 2569 } 2570 2571 // enable changes in effect chain 2572 unlockEffectChains(effectChains); 2573 2574 // sleepTime == 0 means we must write to audio hardware 2575 if (sleepTime == 0) { 2576 2577 threadLoop_write(); 2578 2579if (mType == MIXER) { 2580 // write blocked detection 2581 nsecs_t now = systemTime(); 2582 nsecs_t delta = now - mLastWriteTime; 2583 if (!mStandby && delta > maxPeriod) { 2584 mNumDelayedWrites++; 2585 if ((now - lastWarning) > kWarningThrottleNs) { 2586#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2587 ScopedTrace st(ATRACE_TAG, "underrun"); 2588#endif 2589 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2590 ns2ms(delta), mNumDelayedWrites, this); 2591 lastWarning = now; 2592 } 2593 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2594 // a different threshold. Or completely removed for what it is worth anyway... 2595 if (mStandby) { 2596 longStandbyExit = true; 2597 } 2598 } 2599} 2600 2601 mStandby = false; 2602 } else { 2603 usleep(sleepTime); 2604 } 2605 2606 // Finally let go of removed track(s), without the lock held 2607 // since we can't guarantee the destructors won't acquire that 2608 // same lock. This will also mutate and push a new fast mixer state. 2609 threadLoop_removeTracks(tracksToRemove); 2610 tracksToRemove.clear(); 2611 2612 // FIXME I don't understand the need for this here; 2613 // it was in the original code but maybe the 2614 // assignment in saveOutputTracks() makes this unnecessary? 2615 clearOutputTracks(); 2616 2617 // Effect chains will be actually deleted here if they were removed from 2618 // mEffectChains list during mixing or effects processing 2619 effectChains.clear(); 2620 2621 // FIXME Note that the above .clear() is no longer necessary since effectChains 2622 // is now local to this block, but will keep it for now (at least until merge done). 2623 } 2624 2625if (mType == MIXER || mType == DIRECT) { 2626 // put output stream into standby mode 2627 if (!mStandby) { 2628 mOutput->stream->common.standby(&mOutput->stream->common); 2629 } 2630} 2631if (mType == DUPLICATING) { 2632 // for DuplicatingThread, standby mode is handled by the outputTracks 2633} 2634 2635 releaseWakeLock(); 2636 2637 ALOGV("Thread %p type %d exiting", this, mType); 2638 return false; 2639} 2640 2641void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2642{ 2643 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_write() 2647{ 2648 // FIXME we should only do one push per cycle; confirm this is true 2649 // Start the fast mixer if it's not already running 2650 if (mFastMixer != NULL) { 2651 FastMixerStateQueue *sq = mFastMixer->sq(); 2652 FastMixerState *state = sq->begin(); 2653 if (state->mCommand != FastMixerState::MIX_WRITE && 2654 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2655 if (state->mCommand == FastMixerState::COLD_IDLE) { 2656 int32_t old = android_atomic_inc(&mFastMixerFutex); 2657 if (old == -1) { 2658 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2659 } 2660 } 2661 state->mCommand = FastMixerState::MIX_WRITE; 2662 sq->end(); 2663 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2664 if (kUseFastMixer == FastMixer_Dynamic) { 2665 mNormalSink = mPipeSink; 2666 } 2667 } else { 2668 sq->end(false /*didModify*/); 2669 } 2670 } 2671 PlaybackThread::threadLoop_write(); 2672} 2673 2674// shared by MIXER and DIRECT, overridden by DUPLICATING 2675void AudioFlinger::PlaybackThread::threadLoop_write() 2676{ 2677 // FIXME rewrite to reduce number of system calls 2678 mLastWriteTime = systemTime(); 2679 mInWrite = true; 2680 2681#define mBitShift 2 // FIXME 2682 size_t count = mixBufferSize >> mBitShift; 2683#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2684 Tracer::traceBegin(ATRACE_TAG, "write"); 2685#endif 2686 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2687#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2688 Tracer::traceEnd(ATRACE_TAG); 2689#endif 2690 if (framesWritten > 0) { 2691 size_t bytesWritten = framesWritten << mBitShift; 2692 mBytesWritten += bytesWritten; 2693 } 2694 2695 mNumWrites++; 2696 mInWrite = false; 2697} 2698 2699void AudioFlinger::MixerThread::threadLoop_standby() 2700{ 2701 // Idle the fast mixer if it's currently running 2702 if (mFastMixer != NULL) { 2703 FastMixerStateQueue *sq = mFastMixer->sq(); 2704 FastMixerState *state = sq->begin(); 2705 if (!(state->mCommand & FastMixerState::IDLE)) { 2706 state->mCommand = FastMixerState::COLD_IDLE; 2707 state->mColdFutexAddr = &mFastMixerFutex; 2708 state->mColdGen++; 2709 mFastMixerFutex = 0; 2710 sq->end(); 2711 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2712 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2713 if (kUseFastMixer == FastMixer_Dynamic) { 2714 mNormalSink = mOutputSink; 2715 } 2716 } else { 2717 sq->end(false /*didModify*/); 2718 } 2719 } 2720 PlaybackThread::threadLoop_standby(); 2721} 2722 2723// shared by MIXER and DIRECT, overridden by DUPLICATING 2724void AudioFlinger::PlaybackThread::threadLoop_standby() 2725{ 2726 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2727 mOutput->stream->common.standby(&mOutput->stream->common); 2728} 2729 2730void AudioFlinger::MixerThread::threadLoop_mix() 2731{ 2732 // obtain the presentation timestamp of the next output buffer 2733 int64_t pts; 2734 status_t status = INVALID_OPERATION; 2735 2736 if (NULL != mOutput->stream->get_next_write_timestamp) { 2737 status = mOutput->stream->get_next_write_timestamp( 2738 mOutput->stream, &pts); 2739 } 2740 2741 if (status != NO_ERROR) { 2742 pts = AudioBufferProvider::kInvalidPTS; 2743 } 2744 2745 // mix buffers... 2746 mAudioMixer->process(pts); 2747 // increase sleep time progressively when application underrun condition clears. 2748 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2749 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2750 // such that we would underrun the audio HAL. 2751 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2752 sleepTimeShift--; 2753 } 2754 sleepTime = 0; 2755 standbyTime = systemTime() + standbyDelay; 2756 //TODO: delay standby when effects have a tail 2757} 2758 2759void AudioFlinger::MixerThread::threadLoop_sleepTime() 2760{ 2761 // If no tracks are ready, sleep once for the duration of an output 2762 // buffer size, then write 0s to the output 2763 if (sleepTime == 0) { 2764 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2765 sleepTime = activeSleepTime >> sleepTimeShift; 2766 if (sleepTime < kMinThreadSleepTimeUs) { 2767 sleepTime = kMinThreadSleepTimeUs; 2768 } 2769 // reduce sleep time in case of consecutive application underruns to avoid 2770 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2771 // duration we would end up writing less data than needed by the audio HAL if 2772 // the condition persists. 2773 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2774 sleepTimeShift++; 2775 } 2776 } else { 2777 sleepTime = idleSleepTime; 2778 } 2779 } else if (mBytesWritten != 0 || 2780 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2781 memset (mMixBuffer, 0, mixBufferSize); 2782 sleepTime = 0; 2783 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2784 } 2785 // TODO add standby time extension fct of effect tail 2786} 2787 2788// prepareTracks_l() must be called with ThreadBase::mLock held 2789AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2790 Vector< sp<Track> > *tracksToRemove) 2791{ 2792 2793 mixer_state mixerStatus = MIXER_IDLE; 2794 // find out which tracks need to be processed 2795 size_t count = mActiveTracks.size(); 2796 size_t mixedTracks = 0; 2797 size_t tracksWithEffect = 0; 2798 // counts only _active_ fast tracks 2799 size_t fastTracks = 0; 2800 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2801 2802 float masterVolume = mMasterVolume; 2803 bool masterMute = mMasterMute; 2804 2805 if (masterMute) { 2806 masterVolume = 0; 2807 } 2808 // Delegate master volume control to effect in output mix effect chain if needed 2809 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2810 if (chain != 0) { 2811 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2812 chain->setVolume_l(&v, &v); 2813 masterVolume = (float)((v + (1 << 23)) >> 24); 2814 chain.clear(); 2815 } 2816 2817 // prepare a new state to push 2818 FastMixerStateQueue *sq = NULL; 2819 FastMixerState *state = NULL; 2820 bool didModify = false; 2821 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2822 if (mFastMixer != NULL) { 2823 sq = mFastMixer->sq(); 2824 state = sq->begin(); 2825 } 2826 2827 for (size_t i=0 ; i<count ; i++) { 2828 sp<Track> t = mActiveTracks[i].promote(); 2829 if (t == 0) continue; 2830 2831 // this const just means the local variable doesn't change 2832 Track* const track = t.get(); 2833 2834 // process fast tracks 2835 if (track->isFastTrack()) { 2836 2837 // It's theoretically possible (though unlikely) for a fast track to be created 2838 // and then removed within the same normal mix cycle. This is not a problem, as 2839 // the track never becomes active so it's fast mixer slot is never touched. 2840 // The converse, of removing an (active) track and then creating a new track 2841 // at the identical fast mixer slot within the same normal mix cycle, 2842 // is impossible because the slot isn't marked available until the end of each cycle. 2843 int j = track->mFastIndex; 2844 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2845 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2846 FastTrack *fastTrack = &state->mFastTracks[j]; 2847 2848 // Determine whether the track is currently in underrun condition, 2849 // and whether it had a recent underrun. 2850 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2851 uint32_t recentFull = (underruns.mBitFields.mFull - 2852 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2853 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2854 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2855 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2856 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2857 uint32_t recentUnderruns = recentPartial + recentEmpty; 2858 track->mObservedUnderruns = underruns; 2859 // don't count underruns that occur while stopping or pausing 2860 // or stopped which can occur when flush() is called while active 2861 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2862 track->mUnderrunCount += recentUnderruns; 2863 } 2864 2865 // This is similar to the state machine for normal tracks, 2866 // with a few modifications for fast tracks. 2867 bool isActive = true; 2868 switch (track->mState) { 2869 case TrackBase::STOPPING_1: 2870 // track stays active in STOPPING_1 state until first underrun 2871 if (recentUnderruns > 0) { 2872 track->mState = TrackBase::STOPPING_2; 2873 } 2874 break; 2875 case TrackBase::PAUSING: 2876 // ramp down is not yet implemented 2877 track->setPaused(); 2878 break; 2879 case TrackBase::RESUMING: 2880 // ramp up is not yet implemented 2881 track->mState = TrackBase::ACTIVE; 2882 break; 2883 case TrackBase::ACTIVE: 2884 if (recentFull > 0 || recentPartial > 0) { 2885 // track has provided at least some frames recently: reset retry count 2886 track->mRetryCount = kMaxTrackRetries; 2887 } 2888 if (recentUnderruns == 0) { 2889 // no recent underruns: stay active 2890 break; 2891 } 2892 // there has recently been an underrun of some kind 2893 if (track->sharedBuffer() == 0) { 2894 // were any of the recent underruns "empty" (no frames available)? 2895 if (recentEmpty == 0) { 2896 // no, then ignore the partial underruns as they are allowed indefinitely 2897 break; 2898 } 2899 // there has recently been an "empty" underrun: decrement the retry counter 2900 if (--(track->mRetryCount) > 0) { 2901 break; 2902 } 2903 // indicate to client process that the track was disabled because of underrun; 2904 // it will then automatically call start() when data is available 2905 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2906 // remove from active list, but state remains ACTIVE [confusing but true] 2907 isActive = false; 2908 break; 2909 } 2910 // fall through 2911 case TrackBase::STOPPING_2: 2912 case TrackBase::PAUSED: 2913 case TrackBase::TERMINATED: 2914 case TrackBase::STOPPED: 2915 case TrackBase::FLUSHED: // flush() while active 2916 // Check for presentation complete if track is inactive 2917 // We have consumed all the buffers of this track. 2918 // This would be incomplete if we auto-paused on underrun 2919 { 2920 size_t audioHALFrames = 2921 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2922 size_t framesWritten = 2923 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2924 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2925 // track stays in active list until presentation is complete 2926 break; 2927 } 2928 } 2929 if (track->isStopping_2()) { 2930 track->mState = TrackBase::STOPPED; 2931 } 2932 if (track->isStopped()) { 2933 // Can't reset directly, as fast mixer is still polling this track 2934 // track->reset(); 2935 // So instead mark this track as needing to be reset after push with ack 2936 resetMask |= 1 << i; 2937 } 2938 isActive = false; 2939 break; 2940 case TrackBase::IDLE: 2941 default: 2942 LOG_FATAL("unexpected track state %d", track->mState); 2943 } 2944 2945 if (isActive) { 2946 // was it previously inactive? 2947 if (!(state->mTrackMask & (1 << j))) { 2948 ExtendedAudioBufferProvider *eabp = track; 2949 VolumeProvider *vp = track; 2950 fastTrack->mBufferProvider = eabp; 2951 fastTrack->mVolumeProvider = vp; 2952 fastTrack->mSampleRate = track->mSampleRate; 2953 fastTrack->mChannelMask = track->mChannelMask; 2954 fastTrack->mGeneration++; 2955 state->mTrackMask |= 1 << j; 2956 didModify = true; 2957 // no acknowledgement required for newly active tracks 2958 } 2959 // cache the combined master volume and stream type volume for fast mixer; this 2960 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2961 track->mCachedVolume = track->isMuted() ? 2962 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2963 ++fastTracks; 2964 } else { 2965 // was it previously active? 2966 if (state->mTrackMask & (1 << j)) { 2967 fastTrack->mBufferProvider = NULL; 2968 fastTrack->mGeneration++; 2969 state->mTrackMask &= ~(1 << j); 2970 didModify = true; 2971 // If any fast tracks were removed, we must wait for acknowledgement 2972 // because we're about to decrement the last sp<> on those tracks. 2973 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2974 } else { 2975 LOG_FATAL("fast track %d should have been active", j); 2976 } 2977 tracksToRemove->add(track); 2978 // Avoids a misleading display in dumpsys 2979 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2980 } 2981 continue; 2982 } 2983 2984 { // local variable scope to avoid goto warning 2985 2986 audio_track_cblk_t* cblk = track->cblk(); 2987 2988 // The first time a track is added we wait 2989 // for all its buffers to be filled before processing it 2990 int name = track->name(); 2991 // make sure that we have enough frames to mix one full buffer. 2992 // enforce this condition only once to enable draining the buffer in case the client 2993 // app does not call stop() and relies on underrun to stop: 2994 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2995 // during last round 2996 uint32_t minFrames = 1; 2997 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2998 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2999 if (t->sampleRate() == (int)mSampleRate) { 3000 minFrames = mNormalFrameCount; 3001 } else { 3002 // +1 for rounding and +1 for additional sample needed for interpolation 3003 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3004 // add frames already consumed but not yet released by the resampler 3005 // because cblk->framesReady() will include these frames 3006 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3007 // the minimum track buffer size is normally twice the number of frames necessary 3008 // to fill one buffer and the resampler should not leave more than one buffer worth 3009 // of unreleased frames after each pass, but just in case... 3010 ALOG_ASSERT(minFrames <= cblk->frameCount); 3011 } 3012 } 3013 if ((track->framesReady() >= minFrames) && track->isReady() && 3014 !track->isPaused() && !track->isTerminated()) 3015 { 3016 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3017 3018 mixedTracks++; 3019 3020 // track->mainBuffer() != mMixBuffer means there is an effect chain 3021 // connected to the track 3022 chain.clear(); 3023 if (track->mainBuffer() != mMixBuffer) { 3024 chain = getEffectChain_l(track->sessionId()); 3025 // Delegate volume control to effect in track effect chain if needed 3026 if (chain != 0) { 3027 tracksWithEffect++; 3028 } else { 3029 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3030 name, track->sessionId()); 3031 } 3032 } 3033 3034 3035 int param = AudioMixer::VOLUME; 3036 if (track->mFillingUpStatus == Track::FS_FILLED) { 3037 // no ramp for the first volume setting 3038 track->mFillingUpStatus = Track::FS_ACTIVE; 3039 if (track->mState == TrackBase::RESUMING) { 3040 track->mState = TrackBase::ACTIVE; 3041 param = AudioMixer::RAMP_VOLUME; 3042 } 3043 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3044 } else if (cblk->server != 0) { 3045 // If the track is stopped before the first frame was mixed, 3046 // do not apply ramp 3047 param = AudioMixer::RAMP_VOLUME; 3048 } 3049 3050 // compute volume for this track 3051 uint32_t vl, vr, va; 3052 if (track->isMuted() || track->isPausing() || 3053 mStreamTypes[track->streamType()].mute) { 3054 vl = vr = va = 0; 3055 if (track->isPausing()) { 3056 track->setPaused(); 3057 } 3058 } else { 3059 3060 // read original volumes with volume control 3061 float typeVolume = mStreamTypes[track->streamType()].volume; 3062 float v = masterVolume * typeVolume; 3063 uint32_t vlr = cblk->getVolumeLR(); 3064 vl = vlr & 0xFFFF; 3065 vr = vlr >> 16; 3066 // track volumes come from shared memory, so can't be trusted and must be clamped 3067 if (vl > MAX_GAIN_INT) { 3068 ALOGV("Track left volume out of range: %04X", vl); 3069 vl = MAX_GAIN_INT; 3070 } 3071 if (vr > MAX_GAIN_INT) { 3072 ALOGV("Track right volume out of range: %04X", vr); 3073 vr = MAX_GAIN_INT; 3074 } 3075 // now apply the master volume and stream type volume 3076 vl = (uint32_t)(v * vl) << 12; 3077 vr = (uint32_t)(v * vr) << 12; 3078 // assuming master volume and stream type volume each go up to 1.0, 3079 // vl and vr are now in 8.24 format 3080 3081 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3082 // send level comes from shared memory and so may be corrupt 3083 if (sendLevel > MAX_GAIN_INT) { 3084 ALOGV("Track send level out of range: %04X", sendLevel); 3085 sendLevel = MAX_GAIN_INT; 3086 } 3087 va = (uint32_t)(v * sendLevel); 3088 } 3089 // Delegate volume control to effect in track effect chain if needed 3090 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3091 // Do not ramp volume if volume is controlled by effect 3092 param = AudioMixer::VOLUME; 3093 track->mHasVolumeController = true; 3094 } else { 3095 // force no volume ramp when volume controller was just disabled or removed 3096 // from effect chain to avoid volume spike 3097 if (track->mHasVolumeController) { 3098 param = AudioMixer::VOLUME; 3099 } 3100 track->mHasVolumeController = false; 3101 } 3102 3103 // Convert volumes from 8.24 to 4.12 format 3104 // This additional clamping is needed in case chain->setVolume_l() overshot 3105 vl = (vl + (1 << 11)) >> 12; 3106 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3107 vr = (vr + (1 << 11)) >> 12; 3108 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3109 3110 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3111 3112 // XXX: these things DON'T need to be done each time 3113 mAudioMixer->setBufferProvider(name, track); 3114 mAudioMixer->enable(name); 3115 3116 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3117 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3118 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3119 mAudioMixer->setParameter( 3120 name, 3121 AudioMixer::TRACK, 3122 AudioMixer::FORMAT, (void *)track->format()); 3123 mAudioMixer->setParameter( 3124 name, 3125 AudioMixer::TRACK, 3126 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3127 mAudioMixer->setParameter( 3128 name, 3129 AudioMixer::RESAMPLE, 3130 AudioMixer::SAMPLE_RATE, 3131 (void *)(cblk->sampleRate)); 3132 mAudioMixer->setParameter( 3133 name, 3134 AudioMixer::TRACK, 3135 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3136 mAudioMixer->setParameter( 3137 name, 3138 AudioMixer::TRACK, 3139 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3140 3141 // reset retry count 3142 track->mRetryCount = kMaxTrackRetries; 3143 3144 // If one track is ready, set the mixer ready if: 3145 // - the mixer was not ready during previous round OR 3146 // - no other track is not ready 3147 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3148 mixerStatus != MIXER_TRACKS_ENABLED) { 3149 mixerStatus = MIXER_TRACKS_READY; 3150 } 3151 } else { 3152 // clear effect chain input buffer if an active track underruns to avoid sending 3153 // previous audio buffer again to effects 3154 chain = getEffectChain_l(track->sessionId()); 3155 if (chain != 0) { 3156 chain->clearInputBuffer(); 3157 } 3158 3159 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3160 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3161 track->isStopped() || track->isPaused()) { 3162 // We have consumed all the buffers of this track. 3163 // Remove it from the list of active tracks. 3164 // TODO: use actual buffer filling status instead of latency when available from 3165 // audio HAL 3166 size_t audioHALFrames = 3167 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3168 size_t framesWritten = 3169 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3170 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3171 if (track->isStopped()) { 3172 track->reset(); 3173 } 3174 tracksToRemove->add(track); 3175 } 3176 } else { 3177 // No buffers for this track. Give it a few chances to 3178 // fill a buffer, then remove it from active list. 3179 if (--(track->mRetryCount) <= 0) { 3180 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3181 tracksToRemove->add(track); 3182 // indicate to client process that the track was disabled because of underrun; 3183 // it will then automatically call start() when data is available 3184 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3185 // If one track is not ready, mark the mixer also not ready if: 3186 // - the mixer was ready during previous round OR 3187 // - no other track is ready 3188 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3189 mixerStatus != MIXER_TRACKS_READY) { 3190 mixerStatus = MIXER_TRACKS_ENABLED; 3191 } 3192 } 3193 mAudioMixer->disable(name); 3194 } 3195 3196 } // local variable scope to avoid goto warning 3197track_is_ready: ; 3198 3199 } 3200 3201 // Push the new FastMixer state if necessary 3202 if (didModify) { 3203 state->mFastTracksGen++; 3204 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3205 if (kUseFastMixer == FastMixer_Dynamic && 3206 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3207 state->mCommand = FastMixerState::COLD_IDLE; 3208 state->mColdFutexAddr = &mFastMixerFutex; 3209 state->mColdGen++; 3210 mFastMixerFutex = 0; 3211 if (kUseFastMixer == FastMixer_Dynamic) { 3212 mNormalSink = mOutputSink; 3213 } 3214 // If we go into cold idle, need to wait for acknowledgement 3215 // so that fast mixer stops doing I/O. 3216 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3217 } 3218 sq->end(); 3219 } 3220 if (sq != NULL) { 3221 sq->end(didModify); 3222 sq->push(block); 3223 } 3224 3225 // Now perform the deferred reset on fast tracks that have stopped 3226 while (resetMask != 0) { 3227 size_t i = __builtin_ctz(resetMask); 3228 ALOG_ASSERT(i < count); 3229 resetMask &= ~(1 << i); 3230 sp<Track> t = mActiveTracks[i].promote(); 3231 if (t == 0) continue; 3232 Track* track = t.get(); 3233 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3234 track->reset(); 3235 } 3236 3237 // remove all the tracks that need to be... 3238 count = tracksToRemove->size(); 3239 if (CC_UNLIKELY(count)) { 3240 for (size_t i=0 ; i<count ; i++) { 3241 const sp<Track>& track = tracksToRemove->itemAt(i); 3242 mActiveTracks.remove(track); 3243 if (track->mainBuffer() != mMixBuffer) { 3244 chain = getEffectChain_l(track->sessionId()); 3245 if (chain != 0) { 3246 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3247 chain->decActiveTrackCnt(); 3248 } 3249 } 3250 if (track->isTerminated()) { 3251 removeTrack_l(track); 3252 } 3253 } 3254 } 3255 3256 // mix buffer must be cleared if all tracks are connected to an 3257 // effect chain as in this case the mixer will not write to 3258 // mix buffer and track effects will accumulate into it 3259 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3260 // FIXME as a performance optimization, should remember previous zero status 3261 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3262 } 3263 3264 // if any fast tracks, then status is ready 3265 mMixerStatusIgnoringFastTracks = mixerStatus; 3266 if (fastTracks > 0) { 3267 mixerStatus = MIXER_TRACKS_READY; 3268 } 3269 return mixerStatus; 3270} 3271 3272/* 3273The derived values that are cached: 3274 - mixBufferSize from frame count * frame size 3275 - activeSleepTime from activeSleepTimeUs() 3276 - idleSleepTime from idleSleepTimeUs() 3277 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3278 - maxPeriod from frame count and sample rate (MIXER only) 3279 3280The parameters that affect these derived values are: 3281 - frame count 3282 - frame size 3283 - sample rate 3284 - device type: A2DP or not 3285 - device latency 3286 - format: PCM or not 3287 - active sleep time 3288 - idle sleep time 3289*/ 3290 3291void AudioFlinger::PlaybackThread::cacheParameters_l() 3292{ 3293 mixBufferSize = mNormalFrameCount * mFrameSize; 3294 activeSleepTime = activeSleepTimeUs(); 3295 idleSleepTime = idleSleepTimeUs(); 3296} 3297 3298void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3299{ 3300 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3301 this, streamType, mTracks.size()); 3302 Mutex::Autolock _l(mLock); 3303 3304 size_t size = mTracks.size(); 3305 for (size_t i = 0; i < size; i++) { 3306 sp<Track> t = mTracks[i]; 3307 if (t->streamType() == streamType) { 3308 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3309 t->mCblk->cv.signal(); 3310 } 3311 } 3312} 3313 3314// getTrackName_l() must be called with ThreadBase::mLock held 3315int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3316{ 3317 return mAudioMixer->getTrackName(channelMask); 3318} 3319 3320// deleteTrackName_l() must be called with ThreadBase::mLock held 3321void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3322{ 3323 ALOGV("remove track (%d) and delete from mixer", name); 3324 mAudioMixer->deleteTrackName(name); 3325} 3326 3327// checkForNewParameters_l() must be called with ThreadBase::mLock held 3328bool AudioFlinger::MixerThread::checkForNewParameters_l() 3329{ 3330 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3331 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3332 bool reconfig = false; 3333 3334 while (!mNewParameters.isEmpty()) { 3335 3336 if (mFastMixer != NULL) { 3337 FastMixerStateQueue *sq = mFastMixer->sq(); 3338 FastMixerState *state = sq->begin(); 3339 if (!(state->mCommand & FastMixerState::IDLE)) { 3340 previousCommand = state->mCommand; 3341 state->mCommand = FastMixerState::HOT_IDLE; 3342 sq->end(); 3343 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3344 } else { 3345 sq->end(false /*didModify*/); 3346 } 3347 } 3348 3349 status_t status = NO_ERROR; 3350 String8 keyValuePair = mNewParameters[0]; 3351 AudioParameter param = AudioParameter(keyValuePair); 3352 int value; 3353 3354 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3355 reconfig = true; 3356 } 3357 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3358 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3359 status = BAD_VALUE; 3360 } else { 3361 reconfig = true; 3362 } 3363 } 3364 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3365 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3366 status = BAD_VALUE; 3367 } else { 3368 reconfig = true; 3369 } 3370 } 3371 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3372 // do not accept frame count changes if tracks are open as the track buffer 3373 // size depends on frame count and correct behavior would not be guaranteed 3374 // if frame count is changed after track creation 3375 if (!mTracks.isEmpty()) { 3376 status = INVALID_OPERATION; 3377 } else { 3378 reconfig = true; 3379 } 3380 } 3381 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3382#ifdef ADD_BATTERY_DATA 3383 // when changing the audio output device, call addBatteryData to notify 3384 // the change 3385 if ((int)mDevice != value) { 3386 uint32_t params = 0; 3387 // check whether speaker is on 3388 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3389 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3390 } 3391 3392 int deviceWithoutSpeaker 3393 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3394 // check if any other device (except speaker) is on 3395 if (value & deviceWithoutSpeaker ) { 3396 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3397 } 3398 3399 if (params != 0) { 3400 addBatteryData(params); 3401 } 3402 } 3403#endif 3404 3405 // forward device change to effects that have requested to be 3406 // aware of attached audio device. 3407 mDevice = (uint32_t)value; 3408 for (size_t i = 0; i < mEffectChains.size(); i++) { 3409 mEffectChains[i]->setDevice_l(mDevice); 3410 } 3411 } 3412 3413 if (status == NO_ERROR) { 3414 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3415 keyValuePair.string()); 3416 if (!mStandby && status == INVALID_OPERATION) { 3417 mOutput->stream->common.standby(&mOutput->stream->common); 3418 mStandby = true; 3419 mBytesWritten = 0; 3420 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3421 keyValuePair.string()); 3422 } 3423 if (status == NO_ERROR && reconfig) { 3424 delete mAudioMixer; 3425 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3426 mAudioMixer = NULL; 3427 readOutputParameters(); 3428 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3429 for (size_t i = 0; i < mTracks.size() ; i++) { 3430 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3431 if (name < 0) break; 3432 mTracks[i]->mName = name; 3433 // limit track sample rate to 2 x new output sample rate 3434 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3435 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3436 } 3437 } 3438 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3439 } 3440 } 3441 3442 mNewParameters.removeAt(0); 3443 3444 mParamStatus = status; 3445 mParamCond.signal(); 3446 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3447 // already timed out waiting for the status and will never signal the condition. 3448 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3449 } 3450 3451 if (!(previousCommand & FastMixerState::IDLE)) { 3452 ALOG_ASSERT(mFastMixer != NULL); 3453 FastMixerStateQueue *sq = mFastMixer->sq(); 3454 FastMixerState *state = sq->begin(); 3455 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3456 state->mCommand = previousCommand; 3457 sq->end(); 3458 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3459 } 3460 3461 return reconfig; 3462} 3463 3464status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3465{ 3466 const size_t SIZE = 256; 3467 char buffer[SIZE]; 3468 String8 result; 3469 3470 PlaybackThread::dumpInternals(fd, args); 3471 3472 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3473 result.append(buffer); 3474 write(fd, result.string(), result.size()); 3475 3476 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3477 FastMixerDumpState copy = mFastMixerDumpState; 3478 copy.dump(fd); 3479 3480 // Write the tee output to a .wav file 3481 NBAIO_Source *teeSource = mTeeSource.get(); 3482 if (teeSource != NULL) { 3483 char teePath[64]; 3484 struct timeval tv; 3485 gettimeofday(&tv, NULL); 3486 struct tm tm; 3487 localtime_r(&tv.tv_sec, &tm); 3488 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3489 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3490 if (teeFd >= 0) { 3491 char wavHeader[44]; 3492 memcpy(wavHeader, 3493 "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", 3494 sizeof(wavHeader)); 3495 NBAIO_Format format = teeSource->format(); 3496 unsigned channelCount = Format_channelCount(format); 3497 ALOG_ASSERT(channelCount <= FCC_2); 3498 unsigned sampleRate = Format_sampleRate(format); 3499 wavHeader[22] = channelCount; // number of channels 3500 wavHeader[24] = sampleRate; // sample rate 3501 wavHeader[25] = sampleRate >> 8; 3502 wavHeader[32] = channelCount * 2; // block alignment 3503 write(teeFd, wavHeader, sizeof(wavHeader)); 3504 size_t total = 0; 3505 bool firstRead = true; 3506 for (;;) { 3507#define TEE_SINK_READ 1024 3508 short buffer[TEE_SINK_READ * FCC_2]; 3509 size_t count = TEE_SINK_READ; 3510 ssize_t actual = teeSource->read(buffer, count); 3511 bool wasFirstRead = firstRead; 3512 firstRead = false; 3513 if (actual <= 0) { 3514 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3515 continue; 3516 } 3517 break; 3518 } 3519 ALOG_ASSERT(actual <= count); 3520 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3521 total += actual; 3522 } 3523 lseek(teeFd, (off_t) 4, SEEK_SET); 3524 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3525 write(teeFd, &temp, sizeof(temp)); 3526 lseek(teeFd, (off_t) 40, SEEK_SET); 3527 temp = total * channelCount * sizeof(short); 3528 write(teeFd, &temp, sizeof(temp)); 3529 close(teeFd); 3530 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3531 } else { 3532 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3533 } 3534 } 3535 3536 return NO_ERROR; 3537} 3538 3539uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3540{ 3541 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3542} 3543 3544uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3545{ 3546 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3547} 3548 3549void AudioFlinger::MixerThread::cacheParameters_l() 3550{ 3551 PlaybackThread::cacheParameters_l(); 3552 3553 // FIXME: Relaxed timing because of a certain device that can't meet latency 3554 // Should be reduced to 2x after the vendor fixes the driver issue 3555 // increase threshold again due to low power audio mode. The way this warning 3556 // threshold is calculated and its usefulness should be reconsidered anyway. 3557 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3558} 3559 3560// ---------------------------------------------------------------------------- 3561AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3562 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3563 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3564 // mLeftVolFloat, mRightVolFloat 3565 // mLeftVolShort, mRightVolShort 3566{ 3567} 3568 3569AudioFlinger::DirectOutputThread::~DirectOutputThread() 3570{ 3571} 3572 3573AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3574 Vector< sp<Track> > *tracksToRemove 3575) 3576{ 3577 sp<Track> trackToRemove; 3578 3579 mixer_state mixerStatus = MIXER_IDLE; 3580 3581 // find out which tracks need to be processed 3582 if (mActiveTracks.size() != 0) { 3583 sp<Track> t = mActiveTracks[0].promote(); 3584 // The track died recently 3585 if (t == 0) return MIXER_IDLE; 3586 3587 Track* const track = t.get(); 3588 audio_track_cblk_t* cblk = track->cblk(); 3589 3590 // The first time a track is added we wait 3591 // for all its buffers to be filled before processing it 3592 if (cblk->framesReady() && track->isReady() && 3593 !track->isPaused() && !track->isTerminated()) 3594 { 3595 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3596 3597 if (track->mFillingUpStatus == Track::FS_FILLED) { 3598 track->mFillingUpStatus = Track::FS_ACTIVE; 3599 mLeftVolFloat = mRightVolFloat = 0; 3600 mLeftVolShort = mRightVolShort = 0; 3601 if (track->mState == TrackBase::RESUMING) { 3602 track->mState = TrackBase::ACTIVE; 3603 rampVolume = true; 3604 } 3605 } else if (cblk->server != 0) { 3606 // If the track is stopped before the first frame was mixed, 3607 // do not apply ramp 3608 rampVolume = true; 3609 } 3610 // compute volume for this track 3611 float left, right; 3612 if (track->isMuted() || mMasterMute || track->isPausing() || 3613 mStreamTypes[track->streamType()].mute) { 3614 left = right = 0; 3615 if (track->isPausing()) { 3616 track->setPaused(); 3617 } 3618 } else { 3619 float typeVolume = mStreamTypes[track->streamType()].volume; 3620 float v = mMasterVolume * typeVolume; 3621 uint32_t vlr = cblk->getVolumeLR(); 3622 float v_clamped = v * (vlr & 0xFFFF); 3623 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3624 left = v_clamped/MAX_GAIN; 3625 v_clamped = v * (vlr >> 16); 3626 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3627 right = v_clamped/MAX_GAIN; 3628 } 3629 3630 if (left != mLeftVolFloat || right != mRightVolFloat) { 3631 mLeftVolFloat = left; 3632 mRightVolFloat = right; 3633 3634 // If audio HAL implements volume control, 3635 // force software volume to nominal value 3636 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3637 left = 1.0f; 3638 right = 1.0f; 3639 } 3640 3641 // Convert volumes from float to 8.24 3642 uint32_t vl = (uint32_t)(left * (1 << 24)); 3643 uint32_t vr = (uint32_t)(right * (1 << 24)); 3644 3645 // Delegate volume control to effect in track effect chain if needed 3646 // only one effect chain can be present on DirectOutputThread, so if 3647 // there is one, the track is connected to it 3648 if (!mEffectChains.isEmpty()) { 3649 // Do not ramp volume if volume is controlled by effect 3650 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3651 rampVolume = false; 3652 } 3653 } 3654 3655 // Convert volumes from 8.24 to 4.12 format 3656 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3657 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3658 leftVol = (uint16_t)v_clamped; 3659 v_clamped = (vr + (1 << 11)) >> 12; 3660 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3661 rightVol = (uint16_t)v_clamped; 3662 } else { 3663 leftVol = mLeftVolShort; 3664 rightVol = mRightVolShort; 3665 rampVolume = false; 3666 } 3667 3668 // reset retry count 3669 track->mRetryCount = kMaxTrackRetriesDirect; 3670 mActiveTrack = t; 3671 mixerStatus = MIXER_TRACKS_READY; 3672 } else { 3673 // clear effect chain input buffer if an active track underruns to avoid sending 3674 // previous audio buffer again to effects 3675 if (!mEffectChains.isEmpty()) { 3676 mEffectChains[0]->clearInputBuffer(); 3677 } 3678 3679 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3680 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3681 // We have consumed all the buffers of this track. 3682 // Remove it from the list of active tracks. 3683 // TODO: implement behavior for compressed audio 3684 size_t audioHALFrames = 3685 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3686 size_t framesWritten = 3687 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3688 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3689 if (track->isStopped()) { 3690 track->reset(); 3691 } 3692 trackToRemove = track; 3693 } 3694 } else { 3695 // No buffers for this track. Give it a few chances to 3696 // fill a buffer, then remove it from active list. 3697 if (--(track->mRetryCount) <= 0) { 3698 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3699 trackToRemove = track; 3700 } else { 3701 mixerStatus = MIXER_TRACKS_ENABLED; 3702 } 3703 } 3704 } 3705 } 3706 3707 // FIXME merge this with similar code for removing multiple tracks 3708 // remove all the tracks that need to be... 3709 if (CC_UNLIKELY(trackToRemove != 0)) { 3710 tracksToRemove->add(trackToRemove); 3711 mActiveTracks.remove(trackToRemove); 3712 if (!mEffectChains.isEmpty()) { 3713 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3714 trackToRemove->sessionId()); 3715 mEffectChains[0]->decActiveTrackCnt(); 3716 } 3717 if (trackToRemove->isTerminated()) { 3718 removeTrack_l(trackToRemove); 3719 } 3720 } 3721 3722 return mixerStatus; 3723} 3724 3725void AudioFlinger::DirectOutputThread::threadLoop_mix() 3726{ 3727 AudioBufferProvider::Buffer buffer; 3728 size_t frameCount = mFrameCount; 3729 int8_t *curBuf = (int8_t *)mMixBuffer; 3730 // output audio to hardware 3731 while (frameCount) { 3732 buffer.frameCount = frameCount; 3733 mActiveTrack->getNextBuffer(&buffer); 3734 if (CC_UNLIKELY(buffer.raw == NULL)) { 3735 memset(curBuf, 0, frameCount * mFrameSize); 3736 break; 3737 } 3738 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3739 frameCount -= buffer.frameCount; 3740 curBuf += buffer.frameCount * mFrameSize; 3741 mActiveTrack->releaseBuffer(&buffer); 3742 } 3743 sleepTime = 0; 3744 standbyTime = systemTime() + standbyDelay; 3745 mActiveTrack.clear(); 3746 3747 // apply volume 3748 3749 // Do not apply volume on compressed audio 3750 if (!audio_is_linear_pcm(mFormat)) { 3751 return; 3752 } 3753 3754 // convert to signed 16 bit before volume calculation 3755 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3756 size_t count = mFrameCount * mChannelCount; 3757 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3758 int16_t *dst = mMixBuffer + count-1; 3759 while (count--) { 3760 *dst-- = (int16_t)(*src--^0x80) << 8; 3761 } 3762 } 3763 3764 frameCount = mFrameCount; 3765 int16_t *out = mMixBuffer; 3766 if (rampVolume) { 3767 if (mChannelCount == 1) { 3768 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3769 int32_t vlInc = d / (int32_t)frameCount; 3770 int32_t vl = ((int32_t)mLeftVolShort << 16); 3771 do { 3772 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3773 out++; 3774 vl += vlInc; 3775 } while (--frameCount); 3776 3777 } else { 3778 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3779 int32_t vlInc = d / (int32_t)frameCount; 3780 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3781 int32_t vrInc = d / (int32_t)frameCount; 3782 int32_t vl = ((int32_t)mLeftVolShort << 16); 3783 int32_t vr = ((int32_t)mRightVolShort << 16); 3784 do { 3785 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3786 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3787 out += 2; 3788 vl += vlInc; 3789 vr += vrInc; 3790 } while (--frameCount); 3791 } 3792 } else { 3793 if (mChannelCount == 1) { 3794 do { 3795 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3796 out++; 3797 } while (--frameCount); 3798 } else { 3799 do { 3800 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3801 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3802 out += 2; 3803 } while (--frameCount); 3804 } 3805 } 3806 3807 // convert back to unsigned 8 bit after volume calculation 3808 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3809 size_t count = mFrameCount * mChannelCount; 3810 int16_t *src = mMixBuffer; 3811 uint8_t *dst = (uint8_t *)mMixBuffer; 3812 while (count--) { 3813 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3814 } 3815 } 3816 3817 mLeftVolShort = leftVol; 3818 mRightVolShort = rightVol; 3819} 3820 3821void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3822{ 3823 if (sleepTime == 0) { 3824 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3825 sleepTime = activeSleepTime; 3826 } else { 3827 sleepTime = idleSleepTime; 3828 } 3829 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3830 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3831 sleepTime = 0; 3832 } 3833} 3834 3835// getTrackName_l() must be called with ThreadBase::mLock held 3836int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3837{ 3838 return 0; 3839} 3840 3841// deleteTrackName_l() must be called with ThreadBase::mLock held 3842void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3843{ 3844} 3845 3846// checkForNewParameters_l() must be called with ThreadBase::mLock held 3847bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3848{ 3849 bool reconfig = false; 3850 3851 while (!mNewParameters.isEmpty()) { 3852 status_t status = NO_ERROR; 3853 String8 keyValuePair = mNewParameters[0]; 3854 AudioParameter param = AudioParameter(keyValuePair); 3855 int value; 3856 3857 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3858 // do not accept frame count changes if tracks are open as the track buffer 3859 // size depends on frame count and correct behavior would not be garantied 3860 // if frame count is changed after track creation 3861 if (!mTracks.isEmpty()) { 3862 status = INVALID_OPERATION; 3863 } else { 3864 reconfig = true; 3865 } 3866 } 3867 if (status == NO_ERROR) { 3868 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3869 keyValuePair.string()); 3870 if (!mStandby && status == INVALID_OPERATION) { 3871 mOutput->stream->common.standby(&mOutput->stream->common); 3872 mStandby = true; 3873 mBytesWritten = 0; 3874 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3875 keyValuePair.string()); 3876 } 3877 if (status == NO_ERROR && reconfig) { 3878 readOutputParameters(); 3879 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3880 } 3881 } 3882 3883 mNewParameters.removeAt(0); 3884 3885 mParamStatus = status; 3886 mParamCond.signal(); 3887 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3888 // already timed out waiting for the status and will never signal the condition. 3889 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3890 } 3891 return reconfig; 3892} 3893 3894uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3895{ 3896 uint32_t time; 3897 if (audio_is_linear_pcm(mFormat)) { 3898 time = PlaybackThread::activeSleepTimeUs(); 3899 } else { 3900 time = 10000; 3901 } 3902 return time; 3903} 3904 3905uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3906{ 3907 uint32_t time; 3908 if (audio_is_linear_pcm(mFormat)) { 3909 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3910 } else { 3911 time = 10000; 3912 } 3913 return time; 3914} 3915 3916uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3917{ 3918 uint32_t time; 3919 if (audio_is_linear_pcm(mFormat)) { 3920 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3921 } else { 3922 time = 10000; 3923 } 3924 return time; 3925} 3926 3927void AudioFlinger::DirectOutputThread::cacheParameters_l() 3928{ 3929 PlaybackThread::cacheParameters_l(); 3930 3931 // use shorter standby delay as on normal output to release 3932 // hardware resources as soon as possible 3933 standbyDelay = microseconds(activeSleepTime*2); 3934} 3935 3936// ---------------------------------------------------------------------------- 3937 3938AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3939 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3940 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3941 mWaitTimeMs(UINT_MAX) 3942{ 3943 addOutputTrack(mainThread); 3944} 3945 3946AudioFlinger::DuplicatingThread::~DuplicatingThread() 3947{ 3948 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3949 mOutputTracks[i]->destroy(); 3950 } 3951} 3952 3953void AudioFlinger::DuplicatingThread::threadLoop_mix() 3954{ 3955 // mix buffers... 3956 if (outputsReady(outputTracks)) { 3957 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3958 } else { 3959 memset(mMixBuffer, 0, mixBufferSize); 3960 } 3961 sleepTime = 0; 3962 writeFrames = mNormalFrameCount; 3963} 3964 3965void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3966{ 3967 if (sleepTime == 0) { 3968 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3969 sleepTime = activeSleepTime; 3970 } else { 3971 sleepTime = idleSleepTime; 3972 } 3973 } else if (mBytesWritten != 0) { 3974 // flush remaining overflow buffers in output tracks 3975 for (size_t i = 0; i < outputTracks.size(); i++) { 3976 if (outputTracks[i]->isActive()) { 3977 sleepTime = 0; 3978 writeFrames = 0; 3979 memset(mMixBuffer, 0, mixBufferSize); 3980 break; 3981 } 3982 } 3983 } 3984} 3985 3986void AudioFlinger::DuplicatingThread::threadLoop_write() 3987{ 3988 standbyTime = systemTime() + standbyDelay; 3989 for (size_t i = 0; i < outputTracks.size(); i++) { 3990 outputTracks[i]->write(mMixBuffer, writeFrames); 3991 } 3992 mBytesWritten += mixBufferSize; 3993} 3994 3995void AudioFlinger::DuplicatingThread::threadLoop_standby() 3996{ 3997 // DuplicatingThread implements standby by stopping all tracks 3998 for (size_t i = 0; i < outputTracks.size(); i++) { 3999 outputTracks[i]->stop(); 4000 } 4001} 4002 4003void AudioFlinger::DuplicatingThread::saveOutputTracks() 4004{ 4005 outputTracks = mOutputTracks; 4006} 4007 4008void AudioFlinger::DuplicatingThread::clearOutputTracks() 4009{ 4010 outputTracks.clear(); 4011} 4012 4013void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4014{ 4015 Mutex::Autolock _l(mLock); 4016 // FIXME explain this formula 4017 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4018 OutputTrack *outputTrack = new OutputTrack(thread, 4019 this, 4020 mSampleRate, 4021 mFormat, 4022 mChannelMask, 4023 frameCount); 4024 if (outputTrack->cblk() != NULL) { 4025 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4026 mOutputTracks.add(outputTrack); 4027 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4028 updateWaitTime_l(); 4029 } 4030} 4031 4032void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4033{ 4034 Mutex::Autolock _l(mLock); 4035 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4036 if (mOutputTracks[i]->thread() == thread) { 4037 mOutputTracks[i]->destroy(); 4038 mOutputTracks.removeAt(i); 4039 updateWaitTime_l(); 4040 return; 4041 } 4042 } 4043 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4044} 4045 4046// caller must hold mLock 4047void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4048{ 4049 mWaitTimeMs = UINT_MAX; 4050 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4051 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4052 if (strong != 0) { 4053 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4054 if (waitTimeMs < mWaitTimeMs) { 4055 mWaitTimeMs = waitTimeMs; 4056 } 4057 } 4058 } 4059} 4060 4061 4062bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4063{ 4064 for (size_t i = 0; i < outputTracks.size(); i++) { 4065 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4066 if (thread == 0) { 4067 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4068 return false; 4069 } 4070 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4071 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4072 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4073 return false; 4074 } 4075 } 4076 return true; 4077} 4078 4079uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4080{ 4081 return (mWaitTimeMs * 1000) / 2; 4082} 4083 4084void AudioFlinger::DuplicatingThread::cacheParameters_l() 4085{ 4086 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4087 updateWaitTime_l(); 4088 4089 MixerThread::cacheParameters_l(); 4090} 4091 4092// ---------------------------------------------------------------------------- 4093 4094// TrackBase constructor must be called with AudioFlinger::mLock held 4095AudioFlinger::ThreadBase::TrackBase::TrackBase( 4096 ThreadBase *thread, 4097 const sp<Client>& client, 4098 uint32_t sampleRate, 4099 audio_format_t format, 4100 uint32_t channelMask, 4101 int frameCount, 4102 const sp<IMemory>& sharedBuffer, 4103 int sessionId) 4104 : RefBase(), 4105 mThread(thread), 4106 mClient(client), 4107 mCblk(NULL), 4108 // mBuffer 4109 // mBufferEnd 4110 mFrameCount(0), 4111 mState(IDLE), 4112 mSampleRate(sampleRate), 4113 mFormat(format), 4114 mStepServerFailed(false), 4115 mSessionId(sessionId) 4116 // mChannelCount 4117 // mChannelMask 4118{ 4119 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4120 4121 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4122 size_t size = sizeof(audio_track_cblk_t); 4123 uint8_t channelCount = popcount(channelMask); 4124 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4125 if (sharedBuffer == 0) { 4126 size += bufferSize; 4127 } 4128 4129 if (client != NULL) { 4130 mCblkMemory = client->heap()->allocate(size); 4131 if (mCblkMemory != 0) { 4132 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4133 if (mCblk != NULL) { // construct the shared structure in-place. 4134 new(mCblk) audio_track_cblk_t(); 4135 // clear all buffers 4136 mCblk->frameCount = frameCount; 4137 mCblk->sampleRate = sampleRate; 4138// uncomment the following lines to quickly test 32-bit wraparound 4139// mCblk->user = 0xffff0000; 4140// mCblk->server = 0xffff0000; 4141// mCblk->userBase = 0xffff0000; 4142// mCblk->serverBase = 0xffff0000; 4143 mChannelCount = channelCount; 4144 mChannelMask = channelMask; 4145 if (sharedBuffer == 0) { 4146 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4147 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4148 // Force underrun condition to avoid false underrun callback until first data is 4149 // written to buffer (other flags are cleared) 4150 mCblk->flags = CBLK_UNDERRUN_ON; 4151 } else { 4152 mBuffer = sharedBuffer->pointer(); 4153 } 4154 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4155 } 4156 } else { 4157 ALOGE("not enough memory for AudioTrack size=%u", size); 4158 client->heap()->dump("AudioTrack"); 4159 return; 4160 } 4161 } else { 4162 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4163 // construct the shared structure in-place. 4164 new(mCblk) audio_track_cblk_t(); 4165 // clear all buffers 4166 mCblk->frameCount = frameCount; 4167 mCblk->sampleRate = sampleRate; 4168// uncomment the following lines to quickly test 32-bit wraparound 4169// mCblk->user = 0xffff0000; 4170// mCblk->server = 0xffff0000; 4171// mCblk->userBase = 0xffff0000; 4172// mCblk->serverBase = 0xffff0000; 4173 mChannelCount = channelCount; 4174 mChannelMask = channelMask; 4175 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4176 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4177 // Force underrun condition to avoid false underrun callback until first data is 4178 // written to buffer (other flags are cleared) 4179 mCblk->flags = CBLK_UNDERRUN_ON; 4180 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4181 } 4182} 4183 4184AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4185{ 4186 if (mCblk != NULL) { 4187 if (mClient == 0) { 4188 delete mCblk; 4189 } else { 4190 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4191 } 4192 } 4193 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4194 if (mClient != 0) { 4195 // Client destructor must run with AudioFlinger mutex locked 4196 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4197 // If the client's reference count drops to zero, the associated destructor 4198 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4199 // relying on the automatic clear() at end of scope. 4200 mClient.clear(); 4201 } 4202} 4203 4204// AudioBufferProvider interface 4205// getNextBuffer() = 0; 4206// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4207void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4208{ 4209 buffer->raw = NULL; 4210 mFrameCount = buffer->frameCount; 4211 // FIXME See note at getNextBuffer() 4212 (void) step(); // ignore return value of step() 4213 buffer->frameCount = 0; 4214} 4215 4216bool AudioFlinger::ThreadBase::TrackBase::step() { 4217 bool result; 4218 audio_track_cblk_t* cblk = this->cblk(); 4219 4220 result = cblk->stepServer(mFrameCount); 4221 if (!result) { 4222 ALOGV("stepServer failed acquiring cblk mutex"); 4223 mStepServerFailed = true; 4224 } 4225 return result; 4226} 4227 4228void AudioFlinger::ThreadBase::TrackBase::reset() { 4229 audio_track_cblk_t* cblk = this->cblk(); 4230 4231 cblk->user = 0; 4232 cblk->server = 0; 4233 cblk->userBase = 0; 4234 cblk->serverBase = 0; 4235 mStepServerFailed = false; 4236 ALOGV("TrackBase::reset"); 4237} 4238 4239int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4240 return (int)mCblk->sampleRate; 4241} 4242 4243void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4244 audio_track_cblk_t* cblk = this->cblk(); 4245 size_t frameSize = cblk->frameSize; 4246 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4247 int8_t *bufferEnd = bufferStart + frames * frameSize; 4248 4249 // Check validity of returned pointer in case the track control block would have been corrupted. 4250 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4251 "TrackBase::getBuffer buffer out of range:\n" 4252 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4253 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4254 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4255 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4256 4257 return bufferStart; 4258} 4259 4260status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4261{ 4262 mSyncEvents.add(event); 4263 return NO_ERROR; 4264} 4265 4266// ---------------------------------------------------------------------------- 4267 4268// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4269AudioFlinger::PlaybackThread::Track::Track( 4270 PlaybackThread *thread, 4271 const sp<Client>& client, 4272 audio_stream_type_t streamType, 4273 uint32_t sampleRate, 4274 audio_format_t format, 4275 uint32_t channelMask, 4276 int frameCount, 4277 const sp<IMemory>& sharedBuffer, 4278 int sessionId, 4279 IAudioFlinger::track_flags_t flags) 4280 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4281 mMute(false), 4282 mFillingUpStatus(FS_INVALID), 4283 // mRetryCount initialized later when needed 4284 mSharedBuffer(sharedBuffer), 4285 mStreamType(streamType), 4286 mName(-1), // see note below 4287 mMainBuffer(thread->mixBuffer()), 4288 mAuxBuffer(NULL), 4289 mAuxEffectId(0), mHasVolumeController(false), 4290 mPresentationCompleteFrames(0), 4291 mFlags(flags), 4292 mFastIndex(-1), 4293 mUnderrunCount(0), 4294 mCachedVolume(1.0) 4295{ 4296 if (mCblk != NULL) { 4297 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4298 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4299 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4300 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4301 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4302 if (mName < 0) { 4303 ALOGE("no more track names available"); 4304 return; 4305 } 4306 // only allocate a fast track index if we were able to allocate a normal track name 4307 if (flags & IAudioFlinger::TRACK_FAST) { 4308 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4309 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4310 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4311 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4312 // FIXME This is too eager. We allocate a fast track index before the 4313 // fast track becomes active. Since fast tracks are a scarce resource, 4314 // this means we are potentially denying other more important fast tracks from 4315 // being created. It would be better to allocate the index dynamically. 4316 mFastIndex = i; 4317 // Read the initial underruns because this field is never cleared by the fast mixer 4318 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4319 thread->mFastTrackAvailMask &= ~(1 << i); 4320 } 4321 } 4322 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4323} 4324 4325AudioFlinger::PlaybackThread::Track::~Track() 4326{ 4327 ALOGV("PlaybackThread::Track destructor"); 4328 sp<ThreadBase> thread = mThread.promote(); 4329 if (thread != 0) { 4330 Mutex::Autolock _l(thread->mLock); 4331 mState = TERMINATED; 4332 } 4333} 4334 4335void AudioFlinger::PlaybackThread::Track::destroy() 4336{ 4337 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4338 // by removing it from mTracks vector, so there is a risk that this Tracks's 4339 // destructor is called. As the destructor needs to lock mLock, 4340 // we must acquire a strong reference on this Track before locking mLock 4341 // here so that the destructor is called only when exiting this function. 4342 // On the other hand, as long as Track::destroy() is only called by 4343 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4344 // this Track with its member mTrack. 4345 sp<Track> keep(this); 4346 { // scope for mLock 4347 sp<ThreadBase> thread = mThread.promote(); 4348 if (thread != 0) { 4349 if (!isOutputTrack()) { 4350 if (mState == ACTIVE || mState == RESUMING) { 4351 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4352 4353#ifdef ADD_BATTERY_DATA 4354 // to track the speaker usage 4355 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4356#endif 4357 } 4358 AudioSystem::releaseOutput(thread->id()); 4359 } 4360 Mutex::Autolock _l(thread->mLock); 4361 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4362 playbackThread->destroyTrack_l(this); 4363 } 4364 } 4365} 4366 4367/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4368{ 4369 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4370 " Server User Main buf Aux Buf Flags FastUnder\n"); 4371} 4372 4373void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4374{ 4375 uint32_t vlr = mCblk->getVolumeLR(); 4376 if (isFastTrack()) { 4377 sprintf(buffer, " F %2d", mFastIndex); 4378 } else { 4379 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4380 } 4381 track_state state = mState; 4382 char stateChar; 4383 switch (state) { 4384 case IDLE: 4385 stateChar = 'I'; 4386 break; 4387 case TERMINATED: 4388 stateChar = 'T'; 4389 break; 4390 case STOPPING_1: 4391 stateChar = 's'; 4392 break; 4393 case STOPPING_2: 4394 stateChar = '5'; 4395 break; 4396 case STOPPED: 4397 stateChar = 'S'; 4398 break; 4399 case RESUMING: 4400 stateChar = 'R'; 4401 break; 4402 case ACTIVE: 4403 stateChar = 'A'; 4404 break; 4405 case PAUSING: 4406 stateChar = 'p'; 4407 break; 4408 case PAUSED: 4409 stateChar = 'P'; 4410 break; 4411 case FLUSHED: 4412 stateChar = 'F'; 4413 break; 4414 default: 4415 stateChar = '?'; 4416 break; 4417 } 4418 char nowInUnderrun; 4419 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4420 case UNDERRUN_FULL: 4421 nowInUnderrun = ' '; 4422 break; 4423 case UNDERRUN_PARTIAL: 4424 nowInUnderrun = '<'; 4425 break; 4426 case UNDERRUN_EMPTY: 4427 nowInUnderrun = '*'; 4428 break; 4429 default: 4430 nowInUnderrun = '?'; 4431 break; 4432 } 4433 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4434 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4435 (mClient == 0) ? getpid_cached : mClient->pid(), 4436 mStreamType, 4437 mFormat, 4438 mChannelMask, 4439 mSessionId, 4440 mFrameCount, 4441 mCblk->frameCount, 4442 stateChar, 4443 mMute, 4444 mFillingUpStatus, 4445 mCblk->sampleRate, 4446 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4447 20.0 * log10((vlr >> 16) / 4096.0), 4448 mCblk->server, 4449 mCblk->user, 4450 (int)mMainBuffer, 4451 (int)mAuxBuffer, 4452 mCblk->flags, 4453 mUnderrunCount, 4454 nowInUnderrun); 4455} 4456 4457// AudioBufferProvider interface 4458status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4459 AudioBufferProvider::Buffer* buffer, int64_t pts) 4460{ 4461 audio_track_cblk_t* cblk = this->cblk(); 4462 uint32_t framesReady; 4463 uint32_t framesReq = buffer->frameCount; 4464 4465 // Check if last stepServer failed, try to step now 4466 if (mStepServerFailed) { 4467 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4468 // Since the fast mixer is higher priority than client callback thread, 4469 // it does not result in priority inversion for client. 4470 // But a non-blocking solution would be preferable to avoid 4471 // fast mixer being unable to tryLock(), and 4472 // to avoid the extra context switches if the client wakes up, 4473 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4474 if (!step()) goto getNextBuffer_exit; 4475 ALOGV("stepServer recovered"); 4476 mStepServerFailed = false; 4477 } 4478 4479 // FIXME Same as above 4480 framesReady = cblk->framesReady(); 4481 4482 if (CC_LIKELY(framesReady)) { 4483 uint32_t s = cblk->server; 4484 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4485 4486 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4487 if (framesReq > framesReady) { 4488 framesReq = framesReady; 4489 } 4490 if (framesReq > bufferEnd - s) { 4491 framesReq = bufferEnd - s; 4492 } 4493 4494 buffer->raw = getBuffer(s, framesReq); 4495 if (buffer->raw == NULL) goto getNextBuffer_exit; 4496 4497 buffer->frameCount = framesReq; 4498 return NO_ERROR; 4499 } 4500 4501getNextBuffer_exit: 4502 buffer->raw = NULL; 4503 buffer->frameCount = 0; 4504 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4505 return NOT_ENOUGH_DATA; 4506} 4507 4508// Note that framesReady() takes a mutex on the control block using tryLock(). 4509// This could result in priority inversion if framesReady() is called by the normal mixer, 4510// as the normal mixer thread runs at lower 4511// priority than the client's callback thread: there is a short window within framesReady() 4512// during which the normal mixer could be preempted, and the client callback would block. 4513// Another problem can occur if framesReady() is called by the fast mixer: 4514// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4515// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4516size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4517 return mCblk->framesReady(); 4518} 4519 4520// Don't call for fast tracks; the framesReady() could result in priority inversion 4521bool AudioFlinger::PlaybackThread::Track::isReady() const { 4522 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4523 4524 if (framesReady() >= mCblk->frameCount || 4525 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4526 mFillingUpStatus = FS_FILLED; 4527 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4528 return true; 4529 } 4530 return false; 4531} 4532 4533status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4534 int triggerSession) 4535{ 4536 status_t status = NO_ERROR; 4537 ALOGV("start(%d), calling pid %d session %d", 4538 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4539 4540 sp<ThreadBase> thread = mThread.promote(); 4541 if (thread != 0) { 4542 Mutex::Autolock _l(thread->mLock); 4543 track_state state = mState; 4544 // here the track could be either new, or restarted 4545 // in both cases "unstop" the track 4546 if (mState == PAUSED) { 4547 mState = TrackBase::RESUMING; 4548 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4549 } else { 4550 mState = TrackBase::ACTIVE; 4551 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4552 } 4553 4554 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4555 thread->mLock.unlock(); 4556 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4557 thread->mLock.lock(); 4558 4559#ifdef ADD_BATTERY_DATA 4560 // to track the speaker usage 4561 if (status == NO_ERROR) { 4562 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4563 } 4564#endif 4565 } 4566 if (status == NO_ERROR) { 4567 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4568 playbackThread->addTrack_l(this); 4569 } else { 4570 mState = state; 4571 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4572 } 4573 } else { 4574 status = BAD_VALUE; 4575 } 4576 return status; 4577} 4578 4579void AudioFlinger::PlaybackThread::Track::stop() 4580{ 4581 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4582 sp<ThreadBase> thread = mThread.promote(); 4583 if (thread != 0) { 4584 Mutex::Autolock _l(thread->mLock); 4585 track_state state = mState; 4586 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4587 // If the track is not active (PAUSED and buffers full), flush buffers 4588 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4589 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4590 reset(); 4591 mState = STOPPED; 4592 } else if (!isFastTrack()) { 4593 mState = STOPPED; 4594 } else { 4595 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4596 // and then to STOPPED and reset() when presentation is complete 4597 mState = STOPPING_1; 4598 } 4599 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4600 } 4601 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4602 thread->mLock.unlock(); 4603 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4604 thread->mLock.lock(); 4605 4606#ifdef ADD_BATTERY_DATA 4607 // to track the speaker usage 4608 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4609#endif 4610 } 4611 } 4612} 4613 4614void AudioFlinger::PlaybackThread::Track::pause() 4615{ 4616 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4617 sp<ThreadBase> thread = mThread.promote(); 4618 if (thread != 0) { 4619 Mutex::Autolock _l(thread->mLock); 4620 if (mState == ACTIVE || mState == RESUMING) { 4621 mState = PAUSING; 4622 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4623 if (!isOutputTrack()) { 4624 thread->mLock.unlock(); 4625 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4626 thread->mLock.lock(); 4627 4628#ifdef ADD_BATTERY_DATA 4629 // to track the speaker usage 4630 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4631#endif 4632 } 4633 } 4634 } 4635} 4636 4637void AudioFlinger::PlaybackThread::Track::flush() 4638{ 4639 ALOGV("flush(%d)", mName); 4640 sp<ThreadBase> thread = mThread.promote(); 4641 if (thread != 0) { 4642 Mutex::Autolock _l(thread->mLock); 4643 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4644 mState != PAUSING) { 4645 return; 4646 } 4647 // No point remaining in PAUSED state after a flush => go to 4648 // FLUSHED state 4649 mState = FLUSHED; 4650 // do not reset the track if it is still in the process of being stopped or paused. 4651 // this will be done by prepareTracks_l() when the track is stopped. 4652 // prepareTracks_l() will see mState == FLUSHED, then 4653 // remove from active track list, reset(), and trigger presentation complete 4654 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4655 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4656 reset(); 4657 } 4658 } 4659} 4660 4661void AudioFlinger::PlaybackThread::Track::reset() 4662{ 4663 // Do not reset twice to avoid discarding data written just after a flush and before 4664 // the audioflinger thread detects the track is stopped. 4665 if (!mResetDone) { 4666 TrackBase::reset(); 4667 // Force underrun condition to avoid false underrun callback until first data is 4668 // written to buffer 4669 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4670 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4671 mFillingUpStatus = FS_FILLING; 4672 mResetDone = true; 4673 if (mState == FLUSHED) { 4674 mState = IDLE; 4675 } 4676 } 4677} 4678 4679void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4680{ 4681 mMute = muted; 4682} 4683 4684status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4685{ 4686 status_t status = DEAD_OBJECT; 4687 sp<ThreadBase> thread = mThread.promote(); 4688 if (thread != 0) { 4689 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4690 status = playbackThread->attachAuxEffect(this, EffectId); 4691 } 4692 return status; 4693} 4694 4695void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4696{ 4697 mAuxEffectId = EffectId; 4698 mAuxBuffer = buffer; 4699} 4700 4701bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4702 size_t audioHalFrames) 4703{ 4704 // a track is considered presented when the total number of frames written to audio HAL 4705 // corresponds to the number of frames written when presentationComplete() is called for the 4706 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4707 if (mPresentationCompleteFrames == 0) { 4708 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4709 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4710 mPresentationCompleteFrames, audioHalFrames); 4711 } 4712 if (framesWritten >= mPresentationCompleteFrames) { 4713 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4714 mSessionId, framesWritten); 4715 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4716 return true; 4717 } 4718 return false; 4719} 4720 4721void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4722{ 4723 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4724 if (mSyncEvents[i]->type() == type) { 4725 mSyncEvents[i]->trigger(); 4726 mSyncEvents.removeAt(i); 4727 i--; 4728 } 4729 } 4730} 4731 4732// implement VolumeBufferProvider interface 4733 4734uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4735{ 4736 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4737 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4738 uint32_t vlr = mCblk->getVolumeLR(); 4739 uint32_t vl = vlr & 0xFFFF; 4740 uint32_t vr = vlr >> 16; 4741 // track volumes come from shared memory, so can't be trusted and must be clamped 4742 if (vl > MAX_GAIN_INT) { 4743 vl = MAX_GAIN_INT; 4744 } 4745 if (vr > MAX_GAIN_INT) { 4746 vr = MAX_GAIN_INT; 4747 } 4748 // now apply the cached master volume and stream type volume; 4749 // this is trusted but lacks any synchronization or barrier so may be stale 4750 float v = mCachedVolume; 4751 vl *= v; 4752 vr *= v; 4753 // re-combine into U4.16 4754 vlr = (vr << 16) | (vl & 0xFFFF); 4755 // FIXME look at mute, pause, and stop flags 4756 return vlr; 4757} 4758 4759status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4760{ 4761 if (mState == TERMINATED || mState == PAUSED || 4762 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4763 (mState == STOPPED)))) { 4764 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4765 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4766 event->cancel(); 4767 return INVALID_OPERATION; 4768 } 4769 TrackBase::setSyncEvent(event); 4770 return NO_ERROR; 4771} 4772 4773// timed audio tracks 4774 4775sp<AudioFlinger::PlaybackThread::TimedTrack> 4776AudioFlinger::PlaybackThread::TimedTrack::create( 4777 PlaybackThread *thread, 4778 const sp<Client>& client, 4779 audio_stream_type_t streamType, 4780 uint32_t sampleRate, 4781 audio_format_t format, 4782 uint32_t channelMask, 4783 int frameCount, 4784 const sp<IMemory>& sharedBuffer, 4785 int sessionId) { 4786 if (!client->reserveTimedTrack()) 4787 return NULL; 4788 4789 return new TimedTrack( 4790 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4791 sharedBuffer, sessionId); 4792} 4793 4794AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4795 PlaybackThread *thread, 4796 const sp<Client>& client, 4797 audio_stream_type_t streamType, 4798 uint32_t sampleRate, 4799 audio_format_t format, 4800 uint32_t channelMask, 4801 int frameCount, 4802 const sp<IMemory>& sharedBuffer, 4803 int sessionId) 4804 : Track(thread, client, streamType, sampleRate, format, channelMask, 4805 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4806 mQueueHeadInFlight(false), 4807 mTrimQueueHeadOnRelease(false), 4808 mFramesPendingInQueue(0), 4809 mTimedSilenceBuffer(NULL), 4810 mTimedSilenceBufferSize(0), 4811 mTimedAudioOutputOnTime(false), 4812 mMediaTimeTransformValid(false) 4813{ 4814 LocalClock lc; 4815 mLocalTimeFreq = lc.getLocalFreq(); 4816 4817 mLocalTimeToSampleTransform.a_zero = 0; 4818 mLocalTimeToSampleTransform.b_zero = 0; 4819 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4820 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4821 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4822 &mLocalTimeToSampleTransform.a_to_b_denom); 4823 4824 mMediaTimeToSampleTransform.a_zero = 0; 4825 mMediaTimeToSampleTransform.b_zero = 0; 4826 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4827 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4828 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4829 &mMediaTimeToSampleTransform.a_to_b_denom); 4830} 4831 4832AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4833 mClient->releaseTimedTrack(); 4834 delete [] mTimedSilenceBuffer; 4835} 4836 4837status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4838 size_t size, sp<IMemory>* buffer) { 4839 4840 Mutex::Autolock _l(mTimedBufferQueueLock); 4841 4842 trimTimedBufferQueue_l(); 4843 4844 // lazily initialize the shared memory heap for timed buffers 4845 if (mTimedMemoryDealer == NULL) { 4846 const int kTimedBufferHeapSize = 512 << 10; 4847 4848 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4849 "AudioFlingerTimed"); 4850 if (mTimedMemoryDealer == NULL) 4851 return NO_MEMORY; 4852 } 4853 4854 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4855 if (newBuffer == NULL) { 4856 newBuffer = mTimedMemoryDealer->allocate(size); 4857 if (newBuffer == NULL) 4858 return NO_MEMORY; 4859 } 4860 4861 *buffer = newBuffer; 4862 return NO_ERROR; 4863} 4864 4865// caller must hold mTimedBufferQueueLock 4866void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4867 int64_t mediaTimeNow; 4868 { 4869 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4870 if (!mMediaTimeTransformValid) 4871 return; 4872 4873 int64_t targetTimeNow; 4874 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4875 ? mCCHelper.getCommonTime(&targetTimeNow) 4876 : mCCHelper.getLocalTime(&targetTimeNow); 4877 4878 if (OK != res) 4879 return; 4880 4881 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4882 &mediaTimeNow)) { 4883 return; 4884 } 4885 } 4886 4887 size_t trimEnd; 4888 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4889 int64_t bufEnd; 4890 4891 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4892 // We have a next buffer. Just use its PTS as the PTS of the frame 4893 // following the last frame in this buffer. If the stream is sparse 4894 // (ie, there are deliberate gaps left in the stream which should be 4895 // filled with silence by the TimedAudioTrack), then this can result 4896 // in one extra buffer being left un-trimmed when it could have 4897 // been. In general, this is not typical, and we would rather 4898 // optimized away the TS calculation below for the more common case 4899 // where PTSes are contiguous. 4900 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4901 } else { 4902 // We have no next buffer. Compute the PTS of the frame following 4903 // the last frame in this buffer by computing the duration of of 4904 // this frame in media time units and adding it to the PTS of the 4905 // buffer. 4906 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4907 / mCblk->frameSize; 4908 4909 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4910 &bufEnd)) { 4911 ALOGE("Failed to convert frame count of %lld to media time" 4912 " duration" " (scale factor %d/%u) in %s", 4913 frameCount, 4914 mMediaTimeToSampleTransform.a_to_b_numer, 4915 mMediaTimeToSampleTransform.a_to_b_denom, 4916 __PRETTY_FUNCTION__); 4917 break; 4918 } 4919 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4920 } 4921 4922 if (bufEnd > mediaTimeNow) 4923 break; 4924 4925 // Is the buffer we want to use in the middle of a mix operation right 4926 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4927 // from the mixer which should be coming back shortly. 4928 if (!trimEnd && mQueueHeadInFlight) { 4929 mTrimQueueHeadOnRelease = true; 4930 } 4931 } 4932 4933 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4934 if (trimStart < trimEnd) { 4935 // Update the bookkeeping for framesReady() 4936 for (size_t i = trimStart; i < trimEnd; ++i) { 4937 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4938 } 4939 4940 // Now actually remove the buffers from the queue. 4941 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4942 } 4943} 4944 4945void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4946 const char* logTag) { 4947 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4948 "%s called (reason \"%s\"), but timed buffer queue has no" 4949 " elements to trim.", __FUNCTION__, logTag); 4950 4951 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4952 mTimedBufferQueue.removeAt(0); 4953} 4954 4955void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4956 const TimedBuffer& buf, 4957 const char* logTag) { 4958 uint32_t bufBytes = buf.buffer()->size(); 4959 uint32_t consumedAlready = buf.position(); 4960 4961 ALOG_ASSERT(consumedAlready <= bufBytes, 4962 "Bad bookkeeping while updating frames pending. Timed buffer is" 4963 " only %u bytes long, but claims to have consumed %u" 4964 " bytes. (update reason: \"%s\")", 4965 bufBytes, consumedAlready, logTag); 4966 4967 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4968 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4969 "Bad bookkeeping while updating frames pending. Should have at" 4970 " least %u queued frames, but we think we have only %u. (update" 4971 " reason: \"%s\")", 4972 bufFrames, mFramesPendingInQueue, logTag); 4973 4974 mFramesPendingInQueue -= bufFrames; 4975} 4976 4977status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4978 const sp<IMemory>& buffer, int64_t pts) { 4979 4980 { 4981 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4982 if (!mMediaTimeTransformValid) 4983 return INVALID_OPERATION; 4984 } 4985 4986 Mutex::Autolock _l(mTimedBufferQueueLock); 4987 4988 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4989 mFramesPendingInQueue += bufFrames; 4990 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4991 4992 return NO_ERROR; 4993} 4994 4995status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4996 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4997 4998 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4999 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5000 target); 5001 5002 if (!(target == TimedAudioTrack::LOCAL_TIME || 5003 target == TimedAudioTrack::COMMON_TIME)) { 5004 return BAD_VALUE; 5005 } 5006 5007 Mutex::Autolock lock(mMediaTimeTransformLock); 5008 mMediaTimeTransform = xform; 5009 mMediaTimeTransformTarget = target; 5010 mMediaTimeTransformValid = true; 5011 5012 return NO_ERROR; 5013} 5014 5015#define min(a, b) ((a) < (b) ? (a) : (b)) 5016 5017// implementation of getNextBuffer for tracks whose buffers have timestamps 5018status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5019 AudioBufferProvider::Buffer* buffer, int64_t pts) 5020{ 5021 if (pts == AudioBufferProvider::kInvalidPTS) { 5022 buffer->raw = 0; 5023 buffer->frameCount = 0; 5024 mTimedAudioOutputOnTime = false; 5025 return INVALID_OPERATION; 5026 } 5027 5028 Mutex::Autolock _l(mTimedBufferQueueLock); 5029 5030 ALOG_ASSERT(!mQueueHeadInFlight, 5031 "getNextBuffer called without releaseBuffer!"); 5032 5033 while (true) { 5034 5035 // if we have no timed buffers, then fail 5036 if (mTimedBufferQueue.isEmpty()) { 5037 buffer->raw = 0; 5038 buffer->frameCount = 0; 5039 return NOT_ENOUGH_DATA; 5040 } 5041 5042 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5043 5044 // calculate the PTS of the head of the timed buffer queue expressed in 5045 // local time 5046 int64_t headLocalPTS; 5047 { 5048 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5049 5050 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5051 5052 if (mMediaTimeTransform.a_to_b_denom == 0) { 5053 // the transform represents a pause, so yield silence 5054 timedYieldSilence_l(buffer->frameCount, buffer); 5055 return NO_ERROR; 5056 } 5057 5058 int64_t transformedPTS; 5059 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5060 &transformedPTS)) { 5061 // the transform failed. this shouldn't happen, but if it does 5062 // then just drop this buffer 5063 ALOGW("timedGetNextBuffer transform failed"); 5064 buffer->raw = 0; 5065 buffer->frameCount = 0; 5066 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5067 return NO_ERROR; 5068 } 5069 5070 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5071 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5072 &headLocalPTS)) { 5073 buffer->raw = 0; 5074 buffer->frameCount = 0; 5075 return INVALID_OPERATION; 5076 } 5077 } else { 5078 headLocalPTS = transformedPTS; 5079 } 5080 } 5081 5082 // adjust the head buffer's PTS to reflect the portion of the head buffer 5083 // that has already been consumed 5084 int64_t effectivePTS = headLocalPTS + 5085 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5086 5087 // Calculate the delta in samples between the head of the input buffer 5088 // queue and the start of the next output buffer that will be written. 5089 // If the transformation fails because of over or underflow, it means 5090 // that the sample's position in the output stream is so far out of 5091 // whack that it should just be dropped. 5092 int64_t sampleDelta; 5093 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5094 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5095 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5096 " mix"); 5097 continue; 5098 } 5099 if (!mLocalTimeToSampleTransform.doForwardTransform( 5100 (effectivePTS - pts) << 32, &sampleDelta)) { 5101 ALOGV("*** too late during sample rate transform: dropped buffer"); 5102 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5103 continue; 5104 } 5105 5106 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5107 " sampleDelta=[%d.%08x]", 5108 head.pts(), head.position(), pts, 5109 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5110 + (sampleDelta >> 32)), 5111 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5112 5113 // if the delta between the ideal placement for the next input sample and 5114 // the current output position is within this threshold, then we will 5115 // concatenate the next input samples to the previous output 5116 const int64_t kSampleContinuityThreshold = 5117 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5118 5119 // if this is the first buffer of audio that we're emitting from this track 5120 // then it should be almost exactly on time. 5121 const int64_t kSampleStartupThreshold = 1LL << 32; 5122 5123 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5124 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5125 // the next input is close enough to being on time, so concatenate it 5126 // with the last output 5127 timedYieldSamples_l(buffer); 5128 5129 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5130 head.position(), buffer->frameCount); 5131 return NO_ERROR; 5132 } 5133 5134 // Looks like our output is not on time. Reset our on timed status. 5135 // Next time we mix samples from our input queue, then should be within 5136 // the StartupThreshold. 5137 mTimedAudioOutputOnTime = false; 5138 if (sampleDelta > 0) { 5139 // the gap between the current output position and the proper start of 5140 // the next input sample is too big, so fill it with silence 5141 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5142 5143 timedYieldSilence_l(framesUntilNextInput, buffer); 5144 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5145 return NO_ERROR; 5146 } else { 5147 // the next input sample is late 5148 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5149 size_t onTimeSamplePosition = 5150 head.position() + lateFrames * mCblk->frameSize; 5151 5152 if (onTimeSamplePosition > head.buffer()->size()) { 5153 // all the remaining samples in the head are too late, so 5154 // drop it and move on 5155 ALOGV("*** too late: dropped buffer"); 5156 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5157 continue; 5158 } else { 5159 // skip over the late samples 5160 head.setPosition(onTimeSamplePosition); 5161 5162 // yield the available samples 5163 timedYieldSamples_l(buffer); 5164 5165 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5166 return NO_ERROR; 5167 } 5168 } 5169 } 5170} 5171 5172// Yield samples from the timed buffer queue head up to the given output 5173// buffer's capacity. 5174// 5175// Caller must hold mTimedBufferQueueLock 5176void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5177 AudioBufferProvider::Buffer* buffer) { 5178 5179 const TimedBuffer& head = mTimedBufferQueue[0]; 5180 5181 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5182 head.position()); 5183 5184 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5185 mCblk->frameSize); 5186 size_t framesRequested = buffer->frameCount; 5187 buffer->frameCount = min(framesLeftInHead, framesRequested); 5188 5189 mQueueHeadInFlight = true; 5190 mTimedAudioOutputOnTime = true; 5191} 5192 5193// Yield samples of silence up to the given output buffer's capacity 5194// 5195// Caller must hold mTimedBufferQueueLock 5196void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5197 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5198 5199 // lazily allocate a buffer filled with silence 5200 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5201 delete [] mTimedSilenceBuffer; 5202 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5203 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5204 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5205 } 5206 5207 buffer->raw = mTimedSilenceBuffer; 5208 size_t framesRequested = buffer->frameCount; 5209 buffer->frameCount = min(numFrames, framesRequested); 5210 5211 mTimedAudioOutputOnTime = false; 5212} 5213 5214// AudioBufferProvider interface 5215void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5216 AudioBufferProvider::Buffer* buffer) { 5217 5218 Mutex::Autolock _l(mTimedBufferQueueLock); 5219 5220 // If the buffer which was just released is part of the buffer at the head 5221 // of the queue, be sure to update the amt of the buffer which has been 5222 // consumed. If the buffer being returned is not part of the head of the 5223 // queue, its either because the buffer is part of the silence buffer, or 5224 // because the head of the timed queue was trimmed after the mixer called 5225 // getNextBuffer but before the mixer called releaseBuffer. 5226 if (buffer->raw == mTimedSilenceBuffer) { 5227 ALOG_ASSERT(!mQueueHeadInFlight, 5228 "Queue head in flight during release of silence buffer!"); 5229 goto done; 5230 } 5231 5232 ALOG_ASSERT(mQueueHeadInFlight, 5233 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5234 " head in flight."); 5235 5236 if (mTimedBufferQueue.size()) { 5237 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5238 5239 void* start = head.buffer()->pointer(); 5240 void* end = reinterpret_cast<void*>( 5241 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5242 + head.buffer()->size()); 5243 5244 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5245 "released buffer not within the head of the timed buffer" 5246 " queue; qHead = [%p, %p], released buffer = %p", 5247 start, end, buffer->raw); 5248 5249 head.setPosition(head.position() + 5250 (buffer->frameCount * mCblk->frameSize)); 5251 mQueueHeadInFlight = false; 5252 5253 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5254 "Bad bookkeeping during releaseBuffer! Should have at" 5255 " least %u queued frames, but we think we have only %u", 5256 buffer->frameCount, mFramesPendingInQueue); 5257 5258 mFramesPendingInQueue -= buffer->frameCount; 5259 5260 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5261 || mTrimQueueHeadOnRelease) { 5262 trimTimedBufferQueueHead_l("releaseBuffer"); 5263 mTrimQueueHeadOnRelease = false; 5264 } 5265 } else { 5266 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5267 " buffers in the timed buffer queue"); 5268 } 5269 5270done: 5271 buffer->raw = 0; 5272 buffer->frameCount = 0; 5273} 5274 5275size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5276 Mutex::Autolock _l(mTimedBufferQueueLock); 5277 return mFramesPendingInQueue; 5278} 5279 5280AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5281 : mPTS(0), mPosition(0) {} 5282 5283AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5284 const sp<IMemory>& buffer, int64_t pts) 5285 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5286 5287// ---------------------------------------------------------------------------- 5288 5289// RecordTrack constructor must be called with AudioFlinger::mLock held 5290AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5291 RecordThread *thread, 5292 const sp<Client>& client, 5293 uint32_t sampleRate, 5294 audio_format_t format, 5295 uint32_t channelMask, 5296 int frameCount, 5297 int sessionId) 5298 : TrackBase(thread, client, sampleRate, format, 5299 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5300 mOverflow(false) 5301{ 5302 if (mCblk != NULL) { 5303 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5304 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5305 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5306 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5307 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5308 } else { 5309 mCblk->frameSize = sizeof(int8_t); 5310 } 5311 } 5312} 5313 5314AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5315{ 5316 sp<ThreadBase> thread = mThread.promote(); 5317 if (thread != 0) { 5318 AudioSystem::releaseInput(thread->id()); 5319 } 5320} 5321 5322// AudioBufferProvider interface 5323status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5324{ 5325 audio_track_cblk_t* cblk = this->cblk(); 5326 uint32_t framesAvail; 5327 uint32_t framesReq = buffer->frameCount; 5328 5329 // Check if last stepServer failed, try to step now 5330 if (mStepServerFailed) { 5331 if (!step()) goto getNextBuffer_exit; 5332 ALOGV("stepServer recovered"); 5333 mStepServerFailed = false; 5334 } 5335 5336 framesAvail = cblk->framesAvailable_l(); 5337 5338 if (CC_LIKELY(framesAvail)) { 5339 uint32_t s = cblk->server; 5340 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5341 5342 if (framesReq > framesAvail) { 5343 framesReq = framesAvail; 5344 } 5345 if (framesReq > bufferEnd - s) { 5346 framesReq = bufferEnd - s; 5347 } 5348 5349 buffer->raw = getBuffer(s, framesReq); 5350 if (buffer->raw == NULL) goto getNextBuffer_exit; 5351 5352 buffer->frameCount = framesReq; 5353 return NO_ERROR; 5354 } 5355 5356getNextBuffer_exit: 5357 buffer->raw = NULL; 5358 buffer->frameCount = 0; 5359 return NOT_ENOUGH_DATA; 5360} 5361 5362status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5363 int triggerSession) 5364{ 5365 sp<ThreadBase> thread = mThread.promote(); 5366 if (thread != 0) { 5367 RecordThread *recordThread = (RecordThread *)thread.get(); 5368 return recordThread->start(this, event, triggerSession); 5369 } else { 5370 return BAD_VALUE; 5371 } 5372} 5373 5374void AudioFlinger::RecordThread::RecordTrack::stop() 5375{ 5376 sp<ThreadBase> thread = mThread.promote(); 5377 if (thread != 0) { 5378 RecordThread *recordThread = (RecordThread *)thread.get(); 5379 recordThread->stop(this); 5380 TrackBase::reset(); 5381 // Force overrun condition to avoid false overrun callback until first data is 5382 // read from buffer 5383 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5384 } 5385} 5386 5387void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5388{ 5389 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5390 (mClient == 0) ? getpid_cached : mClient->pid(), 5391 mFormat, 5392 mChannelMask, 5393 mSessionId, 5394 mFrameCount, 5395 mState, 5396 mCblk->sampleRate, 5397 mCblk->server, 5398 mCblk->user); 5399} 5400 5401 5402// ---------------------------------------------------------------------------- 5403 5404AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5405 PlaybackThread *playbackThread, 5406 DuplicatingThread *sourceThread, 5407 uint32_t sampleRate, 5408 audio_format_t format, 5409 uint32_t channelMask, 5410 int frameCount) 5411 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5412 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5413 mActive(false), mSourceThread(sourceThread) 5414{ 5415 5416 if (mCblk != NULL) { 5417 mCblk->flags |= CBLK_DIRECTION_OUT; 5418 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5419 mOutBuffer.frameCount = 0; 5420 playbackThread->mTracks.add(this); 5421 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5422 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5423 mCblk, mBuffer, mCblk->buffers, 5424 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5425 } else { 5426 ALOGW("Error creating output track on thread %p", playbackThread); 5427 } 5428} 5429 5430AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5431{ 5432 clearBufferQueue(); 5433} 5434 5435status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5436 int triggerSession) 5437{ 5438 status_t status = Track::start(event, triggerSession); 5439 if (status != NO_ERROR) { 5440 return status; 5441 } 5442 5443 mActive = true; 5444 mRetryCount = 127; 5445 return status; 5446} 5447 5448void AudioFlinger::PlaybackThread::OutputTrack::stop() 5449{ 5450 Track::stop(); 5451 clearBufferQueue(); 5452 mOutBuffer.frameCount = 0; 5453 mActive = false; 5454} 5455 5456bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5457{ 5458 Buffer *pInBuffer; 5459 Buffer inBuffer; 5460 uint32_t channelCount = mChannelCount; 5461 bool outputBufferFull = false; 5462 inBuffer.frameCount = frames; 5463 inBuffer.i16 = data; 5464 5465 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5466 5467 if (!mActive && frames != 0) { 5468 start(); 5469 sp<ThreadBase> thread = mThread.promote(); 5470 if (thread != 0) { 5471 MixerThread *mixerThread = (MixerThread *)thread.get(); 5472 if (mCblk->frameCount > frames){ 5473 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5474 uint32_t startFrames = (mCblk->frameCount - frames); 5475 pInBuffer = new Buffer; 5476 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5477 pInBuffer->frameCount = startFrames; 5478 pInBuffer->i16 = pInBuffer->mBuffer; 5479 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5480 mBufferQueue.add(pInBuffer); 5481 } else { 5482 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5483 } 5484 } 5485 } 5486 } 5487 5488 while (waitTimeLeftMs) { 5489 // First write pending buffers, then new data 5490 if (mBufferQueue.size()) { 5491 pInBuffer = mBufferQueue.itemAt(0); 5492 } else { 5493 pInBuffer = &inBuffer; 5494 } 5495 5496 if (pInBuffer->frameCount == 0) { 5497 break; 5498 } 5499 5500 if (mOutBuffer.frameCount == 0) { 5501 mOutBuffer.frameCount = pInBuffer->frameCount; 5502 nsecs_t startTime = systemTime(); 5503 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5504 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5505 outputBufferFull = true; 5506 break; 5507 } 5508 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5509 if (waitTimeLeftMs >= waitTimeMs) { 5510 waitTimeLeftMs -= waitTimeMs; 5511 } else { 5512 waitTimeLeftMs = 0; 5513 } 5514 } 5515 5516 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5517 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5518 mCblk->stepUser(outFrames); 5519 pInBuffer->frameCount -= outFrames; 5520 pInBuffer->i16 += outFrames * channelCount; 5521 mOutBuffer.frameCount -= outFrames; 5522 mOutBuffer.i16 += outFrames * channelCount; 5523 5524 if (pInBuffer->frameCount == 0) { 5525 if (mBufferQueue.size()) { 5526 mBufferQueue.removeAt(0); 5527 delete [] pInBuffer->mBuffer; 5528 delete pInBuffer; 5529 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5530 } else { 5531 break; 5532 } 5533 } 5534 } 5535 5536 // If we could not write all frames, allocate a buffer and queue it for next time. 5537 if (inBuffer.frameCount) { 5538 sp<ThreadBase> thread = mThread.promote(); 5539 if (thread != 0 && !thread->standby()) { 5540 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5541 pInBuffer = new Buffer; 5542 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5543 pInBuffer->frameCount = inBuffer.frameCount; 5544 pInBuffer->i16 = pInBuffer->mBuffer; 5545 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5546 mBufferQueue.add(pInBuffer); 5547 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5548 } else { 5549 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5550 } 5551 } 5552 } 5553 5554 // Calling write() with a 0 length buffer, means that no more data will be written: 5555 // If no more buffers are pending, fill output track buffer to make sure it is started 5556 // by output mixer. 5557 if (frames == 0 && mBufferQueue.size() == 0) { 5558 if (mCblk->user < mCblk->frameCount) { 5559 frames = mCblk->frameCount - mCblk->user; 5560 pInBuffer = new Buffer; 5561 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5562 pInBuffer->frameCount = frames; 5563 pInBuffer->i16 = pInBuffer->mBuffer; 5564 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5565 mBufferQueue.add(pInBuffer); 5566 } else if (mActive) { 5567 stop(); 5568 } 5569 } 5570 5571 return outputBufferFull; 5572} 5573 5574status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5575{ 5576 int active; 5577 status_t result; 5578 audio_track_cblk_t* cblk = mCblk; 5579 uint32_t framesReq = buffer->frameCount; 5580 5581// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5582 buffer->frameCount = 0; 5583 5584 uint32_t framesAvail = cblk->framesAvailable(); 5585 5586 5587 if (framesAvail == 0) { 5588 Mutex::Autolock _l(cblk->lock); 5589 goto start_loop_here; 5590 while (framesAvail == 0) { 5591 active = mActive; 5592 if (CC_UNLIKELY(!active)) { 5593 ALOGV("Not active and NO_MORE_BUFFERS"); 5594 return NO_MORE_BUFFERS; 5595 } 5596 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5597 if (result != NO_ERROR) { 5598 return NO_MORE_BUFFERS; 5599 } 5600 // read the server count again 5601 start_loop_here: 5602 framesAvail = cblk->framesAvailable_l(); 5603 } 5604 } 5605 5606// if (framesAvail < framesReq) { 5607// return NO_MORE_BUFFERS; 5608// } 5609 5610 if (framesReq > framesAvail) { 5611 framesReq = framesAvail; 5612 } 5613 5614 uint32_t u = cblk->user; 5615 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5616 5617 if (framesReq > bufferEnd - u) { 5618 framesReq = bufferEnd - u; 5619 } 5620 5621 buffer->frameCount = framesReq; 5622 buffer->raw = (void *)cblk->buffer(u); 5623 return NO_ERROR; 5624} 5625 5626 5627void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5628{ 5629 size_t size = mBufferQueue.size(); 5630 5631 for (size_t i = 0; i < size; i++) { 5632 Buffer *pBuffer = mBufferQueue.itemAt(i); 5633 delete [] pBuffer->mBuffer; 5634 delete pBuffer; 5635 } 5636 mBufferQueue.clear(); 5637} 5638 5639// ---------------------------------------------------------------------------- 5640 5641AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5642 : RefBase(), 5643 mAudioFlinger(audioFlinger), 5644 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5645 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5646 mPid(pid), 5647 mTimedTrackCount(0) 5648{ 5649 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5650} 5651 5652// Client destructor must be called with AudioFlinger::mLock held 5653AudioFlinger::Client::~Client() 5654{ 5655 mAudioFlinger->removeClient_l(mPid); 5656} 5657 5658sp<MemoryDealer> AudioFlinger::Client::heap() const 5659{ 5660 return mMemoryDealer; 5661} 5662 5663// Reserve one of the limited slots for a timed audio track associated 5664// with this client 5665bool AudioFlinger::Client::reserveTimedTrack() 5666{ 5667 const int kMaxTimedTracksPerClient = 4; 5668 5669 Mutex::Autolock _l(mTimedTrackLock); 5670 5671 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5672 ALOGW("can not create timed track - pid %d has exceeded the limit", 5673 mPid); 5674 return false; 5675 } 5676 5677 mTimedTrackCount++; 5678 return true; 5679} 5680 5681// Release a slot for a timed audio track 5682void AudioFlinger::Client::releaseTimedTrack() 5683{ 5684 Mutex::Autolock _l(mTimedTrackLock); 5685 mTimedTrackCount--; 5686} 5687 5688// ---------------------------------------------------------------------------- 5689 5690AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5691 const sp<IAudioFlingerClient>& client, 5692 pid_t pid) 5693 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5694{ 5695} 5696 5697AudioFlinger::NotificationClient::~NotificationClient() 5698{ 5699} 5700 5701void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5702{ 5703 sp<NotificationClient> keep(this); 5704 mAudioFlinger->removeNotificationClient(mPid); 5705} 5706 5707// ---------------------------------------------------------------------------- 5708 5709AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5710 : BnAudioTrack(), 5711 mTrack(track) 5712{ 5713} 5714 5715AudioFlinger::TrackHandle::~TrackHandle() { 5716 // just stop the track on deletion, associated resources 5717 // will be freed from the main thread once all pending buffers have 5718 // been played. Unless it's not in the active track list, in which 5719 // case we free everything now... 5720 mTrack->destroy(); 5721} 5722 5723sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5724 return mTrack->getCblk(); 5725} 5726 5727status_t AudioFlinger::TrackHandle::start() { 5728 return mTrack->start(); 5729} 5730 5731void AudioFlinger::TrackHandle::stop() { 5732 mTrack->stop(); 5733} 5734 5735void AudioFlinger::TrackHandle::flush() { 5736 mTrack->flush(); 5737} 5738 5739void AudioFlinger::TrackHandle::mute(bool e) { 5740 mTrack->mute(e); 5741} 5742 5743void AudioFlinger::TrackHandle::pause() { 5744 mTrack->pause(); 5745} 5746 5747status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5748{ 5749 return mTrack->attachAuxEffect(EffectId); 5750} 5751 5752status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5753 sp<IMemory>* buffer) { 5754 if (!mTrack->isTimedTrack()) 5755 return INVALID_OPERATION; 5756 5757 PlaybackThread::TimedTrack* tt = 5758 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5759 return tt->allocateTimedBuffer(size, buffer); 5760} 5761 5762status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5763 int64_t pts) { 5764 if (!mTrack->isTimedTrack()) 5765 return INVALID_OPERATION; 5766 5767 PlaybackThread::TimedTrack* tt = 5768 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5769 return tt->queueTimedBuffer(buffer, pts); 5770} 5771 5772status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5773 const LinearTransform& xform, int target) { 5774 5775 if (!mTrack->isTimedTrack()) 5776 return INVALID_OPERATION; 5777 5778 PlaybackThread::TimedTrack* tt = 5779 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5780 return tt->setMediaTimeTransform( 5781 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5782} 5783 5784status_t AudioFlinger::TrackHandle::onTransact( 5785 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5786{ 5787 return BnAudioTrack::onTransact(code, data, reply, flags); 5788} 5789 5790// ---------------------------------------------------------------------------- 5791 5792sp<IAudioRecord> AudioFlinger::openRecord( 5793 pid_t pid, 5794 audio_io_handle_t input, 5795 uint32_t sampleRate, 5796 audio_format_t format, 5797 uint32_t channelMask, 5798 int frameCount, 5799 IAudioFlinger::track_flags_t flags, 5800 int *sessionId, 5801 status_t *status) 5802{ 5803 sp<RecordThread::RecordTrack> recordTrack; 5804 sp<RecordHandle> recordHandle; 5805 sp<Client> client; 5806 status_t lStatus; 5807 RecordThread *thread; 5808 size_t inFrameCount; 5809 int lSessionId; 5810 5811 // check calling permissions 5812 if (!recordingAllowed()) { 5813 lStatus = PERMISSION_DENIED; 5814 goto Exit; 5815 } 5816 5817 // add client to list 5818 { // scope for mLock 5819 Mutex::Autolock _l(mLock); 5820 thread = checkRecordThread_l(input); 5821 if (thread == NULL) { 5822 lStatus = BAD_VALUE; 5823 goto Exit; 5824 } 5825 5826 client = registerPid_l(pid); 5827 5828 // If no audio session id is provided, create one here 5829 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5830 lSessionId = *sessionId; 5831 } else { 5832 lSessionId = nextUniqueId(); 5833 if (sessionId != NULL) { 5834 *sessionId = lSessionId; 5835 } 5836 } 5837 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5838 recordTrack = thread->createRecordTrack_l(client, 5839 sampleRate, 5840 format, 5841 channelMask, 5842 frameCount, 5843 lSessionId, 5844 &lStatus); 5845 } 5846 if (lStatus != NO_ERROR) { 5847 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5848 // destructor is called by the TrackBase destructor with mLock held 5849 client.clear(); 5850 recordTrack.clear(); 5851 goto Exit; 5852 } 5853 5854 // return to handle to client 5855 recordHandle = new RecordHandle(recordTrack); 5856 lStatus = NO_ERROR; 5857 5858Exit: 5859 if (status) { 5860 *status = lStatus; 5861 } 5862 return recordHandle; 5863} 5864 5865// ---------------------------------------------------------------------------- 5866 5867AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5868 : BnAudioRecord(), 5869 mRecordTrack(recordTrack) 5870{ 5871} 5872 5873AudioFlinger::RecordHandle::~RecordHandle() { 5874 stop(); 5875} 5876 5877sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5878 return mRecordTrack->getCblk(); 5879} 5880 5881status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5882 ALOGV("RecordHandle::start()"); 5883 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5884} 5885 5886void AudioFlinger::RecordHandle::stop() { 5887 ALOGV("RecordHandle::stop()"); 5888 mRecordTrack->stop(); 5889} 5890 5891status_t AudioFlinger::RecordHandle::onTransact( 5892 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5893{ 5894 return BnAudioRecord::onTransact(code, data, reply, flags); 5895} 5896 5897// ---------------------------------------------------------------------------- 5898 5899AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5900 AudioStreamIn *input, 5901 uint32_t sampleRate, 5902 uint32_t channels, 5903 audio_io_handle_t id, 5904 uint32_t device) : 5905 ThreadBase(audioFlinger, id, device, RECORD), 5906 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5907 // mRsmpInIndex and mInputBytes set by readInputParameters() 5908 mReqChannelCount(popcount(channels)), 5909 mReqSampleRate(sampleRate) 5910 // mBytesRead is only meaningful while active, and so is cleared in start() 5911 // (but might be better to also clear here for dump?) 5912{ 5913 snprintf(mName, kNameLength, "AudioIn_%X", id); 5914 5915 readInputParameters(); 5916} 5917 5918 5919AudioFlinger::RecordThread::~RecordThread() 5920{ 5921 delete[] mRsmpInBuffer; 5922 delete mResampler; 5923 delete[] mRsmpOutBuffer; 5924} 5925 5926void AudioFlinger::RecordThread::onFirstRef() 5927{ 5928 run(mName, PRIORITY_URGENT_AUDIO); 5929} 5930 5931status_t AudioFlinger::RecordThread::readyToRun() 5932{ 5933 status_t status = initCheck(); 5934 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5935 return status; 5936} 5937 5938bool AudioFlinger::RecordThread::threadLoop() 5939{ 5940 AudioBufferProvider::Buffer buffer; 5941 sp<RecordTrack> activeTrack; 5942 Vector< sp<EffectChain> > effectChains; 5943 5944 nsecs_t lastWarning = 0; 5945 5946 acquireWakeLock(); 5947 5948 // start recording 5949 while (!exitPending()) { 5950 5951 processConfigEvents(); 5952 5953 { // scope for mLock 5954 Mutex::Autolock _l(mLock); 5955 checkForNewParameters_l(); 5956 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5957 if (!mStandby) { 5958 mInput->stream->common.standby(&mInput->stream->common); 5959 mStandby = true; 5960 } 5961 5962 if (exitPending()) break; 5963 5964 releaseWakeLock_l(); 5965 ALOGV("RecordThread: loop stopping"); 5966 // go to sleep 5967 mWaitWorkCV.wait(mLock); 5968 ALOGV("RecordThread: loop starting"); 5969 acquireWakeLock_l(); 5970 continue; 5971 } 5972 if (mActiveTrack != 0) { 5973 if (mActiveTrack->mState == TrackBase::PAUSING) { 5974 if (!mStandby) { 5975 mInput->stream->common.standby(&mInput->stream->common); 5976 mStandby = true; 5977 } 5978 mActiveTrack.clear(); 5979 mStartStopCond.broadcast(); 5980 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5981 if (mReqChannelCount != mActiveTrack->channelCount()) { 5982 mActiveTrack.clear(); 5983 mStartStopCond.broadcast(); 5984 } else if (mBytesRead != 0) { 5985 // record start succeeds only if first read from audio input 5986 // succeeds 5987 if (mBytesRead > 0) { 5988 mActiveTrack->mState = TrackBase::ACTIVE; 5989 } else { 5990 mActiveTrack.clear(); 5991 } 5992 mStartStopCond.broadcast(); 5993 } 5994 mStandby = false; 5995 } 5996 } 5997 lockEffectChains_l(effectChains); 5998 } 5999 6000 if (mActiveTrack != 0) { 6001 if (mActiveTrack->mState != TrackBase::ACTIVE && 6002 mActiveTrack->mState != TrackBase::RESUMING) { 6003 unlockEffectChains(effectChains); 6004 usleep(kRecordThreadSleepUs); 6005 continue; 6006 } 6007 for (size_t i = 0; i < effectChains.size(); i ++) { 6008 effectChains[i]->process_l(); 6009 } 6010 6011 buffer.frameCount = mFrameCount; 6012 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6013 size_t framesOut = buffer.frameCount; 6014 if (mResampler == NULL) { 6015 // no resampling 6016 while (framesOut) { 6017 size_t framesIn = mFrameCount - mRsmpInIndex; 6018 if (framesIn) { 6019 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6020 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6021 if (framesIn > framesOut) 6022 framesIn = framesOut; 6023 mRsmpInIndex += framesIn; 6024 framesOut -= framesIn; 6025 if ((int)mChannelCount == mReqChannelCount || 6026 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6027 memcpy(dst, src, framesIn * mFrameSize); 6028 } else { 6029 int16_t *src16 = (int16_t *)src; 6030 int16_t *dst16 = (int16_t *)dst; 6031 if (mChannelCount == 1) { 6032 while (framesIn--) { 6033 *dst16++ = *src16; 6034 *dst16++ = *src16++; 6035 } 6036 } else { 6037 while (framesIn--) { 6038 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6039 src16 += 2; 6040 } 6041 } 6042 } 6043 } 6044 if (framesOut && mFrameCount == mRsmpInIndex) { 6045 if (framesOut == mFrameCount && 6046 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6047 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6048 framesOut = 0; 6049 } else { 6050 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6051 mRsmpInIndex = 0; 6052 } 6053 if (mBytesRead < 0) { 6054 ALOGE("Error reading audio input"); 6055 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6056 // Force input into standby so that it tries to 6057 // recover at next read attempt 6058 mInput->stream->common.standby(&mInput->stream->common); 6059 usleep(kRecordThreadSleepUs); 6060 } 6061 mRsmpInIndex = mFrameCount; 6062 framesOut = 0; 6063 buffer.frameCount = 0; 6064 } 6065 } 6066 } 6067 } else { 6068 // resampling 6069 6070 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6071 // alter output frame count as if we were expecting stereo samples 6072 if (mChannelCount == 1 && mReqChannelCount == 1) { 6073 framesOut >>= 1; 6074 } 6075 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6076 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6077 // are 32 bit aligned which should be always true. 6078 if (mChannelCount == 2 && mReqChannelCount == 1) { 6079 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6080 // the resampler always outputs stereo samples: do post stereo to mono conversion 6081 int16_t *src = (int16_t *)mRsmpOutBuffer; 6082 int16_t *dst = buffer.i16; 6083 while (framesOut--) { 6084 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6085 src += 2; 6086 } 6087 } else { 6088 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6089 } 6090 6091 } 6092 if (mFramestoDrop == 0) { 6093 mActiveTrack->releaseBuffer(&buffer); 6094 } else { 6095 if (mFramestoDrop > 0) { 6096 mFramestoDrop -= buffer.frameCount; 6097 if (mFramestoDrop <= 0) { 6098 clearSyncStartEvent(); 6099 } 6100 } else { 6101 mFramestoDrop += buffer.frameCount; 6102 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6103 mSyncStartEvent->isCancelled()) { 6104 ALOGW("Synced record %s, session %d, trigger session %d", 6105 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6106 mActiveTrack->sessionId(), 6107 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6108 clearSyncStartEvent(); 6109 } 6110 } 6111 } 6112 mActiveTrack->overflow(); 6113 } 6114 // client isn't retrieving buffers fast enough 6115 else { 6116 if (!mActiveTrack->setOverflow()) { 6117 nsecs_t now = systemTime(); 6118 if ((now - lastWarning) > kWarningThrottleNs) { 6119 ALOGW("RecordThread: buffer overflow"); 6120 lastWarning = now; 6121 } 6122 } 6123 // Release the processor for a while before asking for a new buffer. 6124 // This will give the application more chance to read from the buffer and 6125 // clear the overflow. 6126 usleep(kRecordThreadSleepUs); 6127 } 6128 } 6129 // enable changes in effect chain 6130 unlockEffectChains(effectChains); 6131 effectChains.clear(); 6132 } 6133 6134 if (!mStandby) { 6135 mInput->stream->common.standby(&mInput->stream->common); 6136 } 6137 mActiveTrack.clear(); 6138 6139 mStartStopCond.broadcast(); 6140 6141 releaseWakeLock(); 6142 6143 ALOGV("RecordThread %p exiting", this); 6144 return false; 6145} 6146 6147 6148sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6149 const sp<AudioFlinger::Client>& client, 6150 uint32_t sampleRate, 6151 audio_format_t format, 6152 int channelMask, 6153 int frameCount, 6154 int sessionId, 6155 status_t *status) 6156{ 6157 sp<RecordTrack> track; 6158 status_t lStatus; 6159 6160 lStatus = initCheck(); 6161 if (lStatus != NO_ERROR) { 6162 ALOGE("Audio driver not initialized."); 6163 goto Exit; 6164 } 6165 6166 { // scope for mLock 6167 Mutex::Autolock _l(mLock); 6168 6169 track = new RecordTrack(this, client, sampleRate, 6170 format, channelMask, frameCount, sessionId); 6171 6172 if (track->getCblk() == 0) { 6173 lStatus = NO_MEMORY; 6174 goto Exit; 6175 } 6176 6177 mTrack = track.get(); 6178 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6179 bool suspend = audio_is_bluetooth_sco_device( 6180 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6181 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6182 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6183 } 6184 lStatus = NO_ERROR; 6185 6186Exit: 6187 if (status) { 6188 *status = lStatus; 6189 } 6190 return track; 6191} 6192 6193status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6194 AudioSystem::sync_event_t event, 6195 int triggerSession) 6196{ 6197 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6198 sp<ThreadBase> strongMe = this; 6199 status_t status = NO_ERROR; 6200 6201 if (event == AudioSystem::SYNC_EVENT_NONE) { 6202 clearSyncStartEvent(); 6203 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6204 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6205 triggerSession, 6206 recordTrack->sessionId(), 6207 syncStartEventCallback, 6208 this); 6209 // Sync event can be cancelled by the trigger session if the track is not in a 6210 // compatible state in which case we start record immediately 6211 if (mSyncStartEvent->isCancelled()) { 6212 clearSyncStartEvent(); 6213 } else { 6214 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6215 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6216 } 6217 } 6218 6219 { 6220 AutoMutex lock(mLock); 6221 if (mActiveTrack != 0) { 6222 if (recordTrack != mActiveTrack.get()) { 6223 status = -EBUSY; 6224 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6225 mActiveTrack->mState = TrackBase::ACTIVE; 6226 } 6227 return status; 6228 } 6229 6230 recordTrack->mState = TrackBase::IDLE; 6231 mActiveTrack = recordTrack; 6232 mLock.unlock(); 6233 status_t status = AudioSystem::startInput(mId); 6234 mLock.lock(); 6235 if (status != NO_ERROR) { 6236 mActiveTrack.clear(); 6237 clearSyncStartEvent(); 6238 return status; 6239 } 6240 mRsmpInIndex = mFrameCount; 6241 mBytesRead = 0; 6242 if (mResampler != NULL) { 6243 mResampler->reset(); 6244 } 6245 mActiveTrack->mState = TrackBase::RESUMING; 6246 // signal thread to start 6247 ALOGV("Signal record thread"); 6248 mWaitWorkCV.signal(); 6249 // do not wait for mStartStopCond if exiting 6250 if (exitPending()) { 6251 mActiveTrack.clear(); 6252 status = INVALID_OPERATION; 6253 goto startError; 6254 } 6255 mStartStopCond.wait(mLock); 6256 if (mActiveTrack == 0) { 6257 ALOGV("Record failed to start"); 6258 status = BAD_VALUE; 6259 goto startError; 6260 } 6261 ALOGV("Record started OK"); 6262 return status; 6263 } 6264startError: 6265 AudioSystem::stopInput(mId); 6266 clearSyncStartEvent(); 6267 return status; 6268} 6269 6270void AudioFlinger::RecordThread::clearSyncStartEvent() 6271{ 6272 if (mSyncStartEvent != 0) { 6273 mSyncStartEvent->cancel(); 6274 } 6275 mSyncStartEvent.clear(); 6276 mFramestoDrop = 0; 6277} 6278 6279void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6280{ 6281 sp<SyncEvent> strongEvent = event.promote(); 6282 6283 if (strongEvent != 0) { 6284 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6285 me->handleSyncStartEvent(strongEvent); 6286 } 6287} 6288 6289void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6290{ 6291 if (event == mSyncStartEvent) { 6292 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6293 // from audio HAL 6294 mFramestoDrop = mFrameCount * 2; 6295 } 6296} 6297 6298void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6299 ALOGV("RecordThread::stop"); 6300 sp<ThreadBase> strongMe = this; 6301 { 6302 AutoMutex lock(mLock); 6303 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6304 mActiveTrack->mState = TrackBase::PAUSING; 6305 // do not wait for mStartStopCond if exiting 6306 if (exitPending()) { 6307 return; 6308 } 6309 mStartStopCond.wait(mLock); 6310 // if we have been restarted, recordTrack == mActiveTrack.get() here 6311 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6312 mLock.unlock(); 6313 AudioSystem::stopInput(mId); 6314 mLock.lock(); 6315 ALOGV("Record stopped OK"); 6316 } 6317 } 6318 } 6319} 6320 6321bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6322{ 6323 return false; 6324} 6325 6326status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6327{ 6328 if (!isValidSyncEvent(event)) { 6329 return BAD_VALUE; 6330 } 6331 6332 Mutex::Autolock _l(mLock); 6333 6334 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6335 mTrack->setSyncEvent(event); 6336 return NO_ERROR; 6337 } 6338 return NAME_NOT_FOUND; 6339} 6340 6341status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6342{ 6343 const size_t SIZE = 256; 6344 char buffer[SIZE]; 6345 String8 result; 6346 6347 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6348 result.append(buffer); 6349 6350 if (mActiveTrack != 0) { 6351 result.append("Active Track:\n"); 6352 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6353 mActiveTrack->dump(buffer, SIZE); 6354 result.append(buffer); 6355 6356 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6357 result.append(buffer); 6358 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6359 result.append(buffer); 6360 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6361 result.append(buffer); 6362 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6363 result.append(buffer); 6364 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6365 result.append(buffer); 6366 6367 6368 } else { 6369 result.append("No record client\n"); 6370 } 6371 write(fd, result.string(), result.size()); 6372 6373 dumpBase(fd, args); 6374 dumpEffectChains(fd, args); 6375 6376 return NO_ERROR; 6377} 6378 6379// AudioBufferProvider interface 6380status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6381{ 6382 size_t framesReq = buffer->frameCount; 6383 size_t framesReady = mFrameCount - mRsmpInIndex; 6384 int channelCount; 6385 6386 if (framesReady == 0) { 6387 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6388 if (mBytesRead < 0) { 6389 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6390 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6391 // Force input into standby so that it tries to 6392 // recover at next read attempt 6393 mInput->stream->common.standby(&mInput->stream->common); 6394 usleep(kRecordThreadSleepUs); 6395 } 6396 buffer->raw = NULL; 6397 buffer->frameCount = 0; 6398 return NOT_ENOUGH_DATA; 6399 } 6400 mRsmpInIndex = 0; 6401 framesReady = mFrameCount; 6402 } 6403 6404 if (framesReq > framesReady) { 6405 framesReq = framesReady; 6406 } 6407 6408 if (mChannelCount == 1 && mReqChannelCount == 2) { 6409 channelCount = 1; 6410 } else { 6411 channelCount = 2; 6412 } 6413 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6414 buffer->frameCount = framesReq; 6415 return NO_ERROR; 6416} 6417 6418// AudioBufferProvider interface 6419void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6420{ 6421 mRsmpInIndex += buffer->frameCount; 6422 buffer->frameCount = 0; 6423} 6424 6425bool AudioFlinger::RecordThread::checkForNewParameters_l() 6426{ 6427 bool reconfig = false; 6428 6429 while (!mNewParameters.isEmpty()) { 6430 status_t status = NO_ERROR; 6431 String8 keyValuePair = mNewParameters[0]; 6432 AudioParameter param = AudioParameter(keyValuePair); 6433 int value; 6434 audio_format_t reqFormat = mFormat; 6435 int reqSamplingRate = mReqSampleRate; 6436 int reqChannelCount = mReqChannelCount; 6437 6438 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6439 reqSamplingRate = value; 6440 reconfig = true; 6441 } 6442 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6443 reqFormat = (audio_format_t) value; 6444 reconfig = true; 6445 } 6446 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6447 reqChannelCount = popcount(value); 6448 reconfig = true; 6449 } 6450 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6451 // do not accept frame count changes if tracks are open as the track buffer 6452 // size depends on frame count and correct behavior would not be guaranteed 6453 // if frame count is changed after track creation 6454 if (mActiveTrack != 0) { 6455 status = INVALID_OPERATION; 6456 } else { 6457 reconfig = true; 6458 } 6459 } 6460 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6461 // forward device change to effects that have requested to be 6462 // aware of attached audio device. 6463 for (size_t i = 0; i < mEffectChains.size(); i++) { 6464 mEffectChains[i]->setDevice_l(value); 6465 } 6466 // store input device and output device but do not forward output device to audio HAL. 6467 // Note that status is ignored by the caller for output device 6468 // (see AudioFlinger::setParameters() 6469 if (value & AUDIO_DEVICE_OUT_ALL) { 6470 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6471 status = BAD_VALUE; 6472 } else { 6473 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6474 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6475 if (mTrack != NULL) { 6476 bool suspend = audio_is_bluetooth_sco_device( 6477 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6478 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6479 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6480 } 6481 } 6482 mDevice |= (uint32_t)value; 6483 } 6484 if (status == NO_ERROR) { 6485 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6486 if (status == INVALID_OPERATION) { 6487 mInput->stream->common.standby(&mInput->stream->common); 6488 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6489 keyValuePair.string()); 6490 } 6491 if (reconfig) { 6492 if (status == BAD_VALUE && 6493 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6494 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6495 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6496 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6497 (reqChannelCount <= FCC_2)) { 6498 status = NO_ERROR; 6499 } 6500 if (status == NO_ERROR) { 6501 readInputParameters(); 6502 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6503 } 6504 } 6505 } 6506 6507 mNewParameters.removeAt(0); 6508 6509 mParamStatus = status; 6510 mParamCond.signal(); 6511 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6512 // already timed out waiting for the status and will never signal the condition. 6513 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6514 } 6515 return reconfig; 6516} 6517 6518String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6519{ 6520 char *s; 6521 String8 out_s8 = String8(); 6522 6523 Mutex::Autolock _l(mLock); 6524 if (initCheck() != NO_ERROR) { 6525 return out_s8; 6526 } 6527 6528 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6529 out_s8 = String8(s); 6530 free(s); 6531 return out_s8; 6532} 6533 6534void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6535 AudioSystem::OutputDescriptor desc; 6536 void *param2 = NULL; 6537 6538 switch (event) { 6539 case AudioSystem::INPUT_OPENED: 6540 case AudioSystem::INPUT_CONFIG_CHANGED: 6541 desc.channels = mChannelMask; 6542 desc.samplingRate = mSampleRate; 6543 desc.format = mFormat; 6544 desc.frameCount = mFrameCount; 6545 desc.latency = 0; 6546 param2 = &desc; 6547 break; 6548 6549 case AudioSystem::INPUT_CLOSED: 6550 default: 6551 break; 6552 } 6553 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6554} 6555 6556void AudioFlinger::RecordThread::readInputParameters() 6557{ 6558 delete mRsmpInBuffer; 6559 // mRsmpInBuffer is always assigned a new[] below 6560 delete mRsmpOutBuffer; 6561 mRsmpOutBuffer = NULL; 6562 delete mResampler; 6563 mResampler = NULL; 6564 6565 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6566 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6567 mChannelCount = (uint16_t)popcount(mChannelMask); 6568 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6569 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6570 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6571 mFrameCount = mInputBytes / mFrameSize; 6572 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6573 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6574 6575 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6576 { 6577 int channelCount; 6578 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6579 // stereo to mono post process as the resampler always outputs stereo. 6580 if (mChannelCount == 1 && mReqChannelCount == 2) { 6581 channelCount = 1; 6582 } else { 6583 channelCount = 2; 6584 } 6585 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6586 mResampler->setSampleRate(mSampleRate); 6587 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6588 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6589 6590 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6591 if (mChannelCount == 1 && mReqChannelCount == 1) { 6592 mFrameCount >>= 1; 6593 } 6594 6595 } 6596 mRsmpInIndex = mFrameCount; 6597} 6598 6599unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6600{ 6601 Mutex::Autolock _l(mLock); 6602 if (initCheck() != NO_ERROR) { 6603 return 0; 6604 } 6605 6606 return mInput->stream->get_input_frames_lost(mInput->stream); 6607} 6608 6609uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6610{ 6611 Mutex::Autolock _l(mLock); 6612 uint32_t result = 0; 6613 if (getEffectChain_l(sessionId) != 0) { 6614 result = EFFECT_SESSION; 6615 } 6616 6617 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6618 result |= TRACK_SESSION; 6619 } 6620 6621 return result; 6622} 6623 6624AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6625{ 6626 Mutex::Autolock _l(mLock); 6627 return mTrack; 6628} 6629 6630AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6631{ 6632 Mutex::Autolock _l(mLock); 6633 return mInput; 6634} 6635 6636AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6637{ 6638 Mutex::Autolock _l(mLock); 6639 AudioStreamIn *input = mInput; 6640 mInput = NULL; 6641 return input; 6642} 6643 6644// this method must always be called either with ThreadBase mLock held or inside the thread loop 6645audio_stream_t* AudioFlinger::RecordThread::stream() const 6646{ 6647 if (mInput == NULL) { 6648 return NULL; 6649 } 6650 return &mInput->stream->common; 6651} 6652 6653 6654// ---------------------------------------------------------------------------- 6655 6656audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6657{ 6658 if (!settingsAllowed()) { 6659 return 0; 6660 } 6661 Mutex::Autolock _l(mLock); 6662 return loadHwModule_l(name); 6663} 6664 6665// loadHwModule_l() must be called with AudioFlinger::mLock held 6666audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6667{ 6668 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6669 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6670 ALOGW("loadHwModule() module %s already loaded", name); 6671 return mAudioHwDevs.keyAt(i); 6672 } 6673 } 6674 6675 audio_hw_device_t *dev; 6676 6677 int rc = load_audio_interface(name, &dev); 6678 if (rc) { 6679 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6680 return 0; 6681 } 6682 6683 mHardwareStatus = AUDIO_HW_INIT; 6684 rc = dev->init_check(dev); 6685 mHardwareStatus = AUDIO_HW_IDLE; 6686 if (rc) { 6687 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6688 return 0; 6689 } 6690 6691 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6692 (NULL != dev->set_master_volume)) { 6693 AutoMutex lock(mHardwareLock); 6694 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6695 dev->set_master_volume(dev, mMasterVolume); 6696 mHardwareStatus = AUDIO_HW_IDLE; 6697 } 6698 6699 audio_module_handle_t handle = nextUniqueId(); 6700 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6701 6702 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6703 name, dev->common.module->name, dev->common.module->id, handle); 6704 6705 return handle; 6706 6707} 6708 6709audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6710 audio_devices_t *pDevices, 6711 uint32_t *pSamplingRate, 6712 audio_format_t *pFormat, 6713 audio_channel_mask_t *pChannelMask, 6714 uint32_t *pLatencyMs, 6715 audio_output_flags_t flags) 6716{ 6717 status_t status; 6718 PlaybackThread *thread = NULL; 6719 struct audio_config config = { 6720 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6721 channel_mask: pChannelMask ? *pChannelMask : 0, 6722 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6723 }; 6724 audio_stream_out_t *outStream = NULL; 6725 audio_hw_device_t *outHwDev; 6726 6727 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6728 module, 6729 (pDevices != NULL) ? (int)*pDevices : 0, 6730 config.sample_rate, 6731 config.format, 6732 config.channel_mask, 6733 flags); 6734 6735 if (pDevices == NULL || *pDevices == 0) { 6736 return 0; 6737 } 6738 6739 Mutex::Autolock _l(mLock); 6740 6741 outHwDev = findSuitableHwDev_l(module, *pDevices); 6742 if (outHwDev == NULL) 6743 return 0; 6744 6745 audio_io_handle_t id = nextUniqueId(); 6746 6747 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6748 6749 status = outHwDev->open_output_stream(outHwDev, 6750 id, 6751 *pDevices, 6752 (audio_output_flags_t)flags, 6753 &config, 6754 &outStream); 6755 6756 mHardwareStatus = AUDIO_HW_IDLE; 6757 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6758 outStream, 6759 config.sample_rate, 6760 config.format, 6761 config.channel_mask, 6762 status); 6763 6764 if (status == NO_ERROR && outStream != NULL) { 6765 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6766 6767 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6768 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6769 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6770 thread = new DirectOutputThread(this, output, id, *pDevices); 6771 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6772 } else { 6773 thread = new MixerThread(this, output, id, *pDevices); 6774 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6775 } 6776 mPlaybackThreads.add(id, thread); 6777 6778 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6779 if (pFormat != NULL) *pFormat = config.format; 6780 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6781 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6782 6783 // notify client processes of the new output creation 6784 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6785 6786 // the first primary output opened designates the primary hw device 6787 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6788 ALOGI("Using module %d has the primary audio interface", module); 6789 mPrimaryHardwareDev = outHwDev; 6790 6791 AutoMutex lock(mHardwareLock); 6792 mHardwareStatus = AUDIO_HW_SET_MODE; 6793 outHwDev->set_mode(outHwDev, mMode); 6794 6795 // Determine the level of master volume support the primary audio HAL has, 6796 // and set the initial master volume at the same time. 6797 float initialVolume = 1.0; 6798 mMasterVolumeSupportLvl = MVS_NONE; 6799 6800 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6801 if ((NULL != outHwDev->get_master_volume) && 6802 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6803 mMasterVolumeSupportLvl = MVS_FULL; 6804 } else { 6805 mMasterVolumeSupportLvl = MVS_SETONLY; 6806 initialVolume = 1.0; 6807 } 6808 6809 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6810 if ((NULL == outHwDev->set_master_volume) || 6811 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6812 mMasterVolumeSupportLvl = MVS_NONE; 6813 } 6814 // now that we have a primary device, initialize master volume on other devices 6815 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6816 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6817 6818 if ((dev != mPrimaryHardwareDev) && 6819 (NULL != dev->set_master_volume)) { 6820 dev->set_master_volume(dev, initialVolume); 6821 } 6822 } 6823 mHardwareStatus = AUDIO_HW_IDLE; 6824 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6825 ? initialVolume 6826 : 1.0; 6827 mMasterVolume = initialVolume; 6828 } 6829 return id; 6830 } 6831 6832 return 0; 6833} 6834 6835audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6836 audio_io_handle_t output2) 6837{ 6838 Mutex::Autolock _l(mLock); 6839 MixerThread *thread1 = checkMixerThread_l(output1); 6840 MixerThread *thread2 = checkMixerThread_l(output2); 6841 6842 if (thread1 == NULL || thread2 == NULL) { 6843 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6844 return 0; 6845 } 6846 6847 audio_io_handle_t id = nextUniqueId(); 6848 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6849 thread->addOutputTrack(thread2); 6850 mPlaybackThreads.add(id, thread); 6851 // notify client processes of the new output creation 6852 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6853 return id; 6854} 6855 6856status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6857{ 6858 // keep strong reference on the playback thread so that 6859 // it is not destroyed while exit() is executed 6860 sp<PlaybackThread> thread; 6861 { 6862 Mutex::Autolock _l(mLock); 6863 thread = checkPlaybackThread_l(output); 6864 if (thread == NULL) { 6865 return BAD_VALUE; 6866 } 6867 6868 ALOGV("closeOutput() %d", output); 6869 6870 if (thread->type() == ThreadBase::MIXER) { 6871 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6872 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6873 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6874 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6875 } 6876 } 6877 } 6878 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6879 mPlaybackThreads.removeItem(output); 6880 } 6881 thread->exit(); 6882 // The thread entity (active unit of execution) is no longer running here, 6883 // but the ThreadBase container still exists. 6884 6885 if (thread->type() != ThreadBase::DUPLICATING) { 6886 AudioStreamOut *out = thread->clearOutput(); 6887 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6888 // from now on thread->mOutput is NULL 6889 out->hwDev->close_output_stream(out->hwDev, out->stream); 6890 delete out; 6891 } 6892 return NO_ERROR; 6893} 6894 6895status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6896{ 6897 Mutex::Autolock _l(mLock); 6898 PlaybackThread *thread = checkPlaybackThread_l(output); 6899 6900 if (thread == NULL) { 6901 return BAD_VALUE; 6902 } 6903 6904 ALOGV("suspendOutput() %d", output); 6905 thread->suspend(); 6906 6907 return NO_ERROR; 6908} 6909 6910status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6911{ 6912 Mutex::Autolock _l(mLock); 6913 PlaybackThread *thread = checkPlaybackThread_l(output); 6914 6915 if (thread == NULL) { 6916 return BAD_VALUE; 6917 } 6918 6919 ALOGV("restoreOutput() %d", output); 6920 6921 thread->restore(); 6922 6923 return NO_ERROR; 6924} 6925 6926audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6927 audio_devices_t *pDevices, 6928 uint32_t *pSamplingRate, 6929 audio_format_t *pFormat, 6930 uint32_t *pChannelMask) 6931{ 6932 status_t status; 6933 RecordThread *thread = NULL; 6934 struct audio_config config = { 6935 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6936 channel_mask: pChannelMask ? *pChannelMask : 0, 6937 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6938 }; 6939 uint32_t reqSamplingRate = config.sample_rate; 6940 audio_format_t reqFormat = config.format; 6941 audio_channel_mask_t reqChannels = config.channel_mask; 6942 audio_stream_in_t *inStream = NULL; 6943 audio_hw_device_t *inHwDev; 6944 6945 if (pDevices == NULL || *pDevices == 0) { 6946 return 0; 6947 } 6948 6949 Mutex::Autolock _l(mLock); 6950 6951 inHwDev = findSuitableHwDev_l(module, *pDevices); 6952 if (inHwDev == NULL) 6953 return 0; 6954 6955 audio_io_handle_t id = nextUniqueId(); 6956 6957 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6958 &inStream); 6959 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6960 inStream, 6961 config.sample_rate, 6962 config.format, 6963 config.channel_mask, 6964 status); 6965 6966 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6967 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6968 // or stereo to mono conversions on 16 bit PCM inputs. 6969 if (status == BAD_VALUE && 6970 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6971 (config.sample_rate <= 2 * reqSamplingRate) && 6972 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6973 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6974 inStream = NULL; 6975 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6976 } 6977 6978 if (status == NO_ERROR && inStream != NULL) { 6979 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6980 6981 // Start record thread 6982 // RecorThread require both input and output device indication to forward to audio 6983 // pre processing modules 6984 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6985 thread = new RecordThread(this, 6986 input, 6987 reqSamplingRate, 6988 reqChannels, 6989 id, 6990 device); 6991 mRecordThreads.add(id, thread); 6992 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6993 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6994 if (pFormat != NULL) *pFormat = config.format; 6995 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6996 6997 input->stream->common.standby(&input->stream->common); 6998 6999 // notify client processes of the new input creation 7000 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7001 return id; 7002 } 7003 7004 return 0; 7005} 7006 7007status_t AudioFlinger::closeInput(audio_io_handle_t input) 7008{ 7009 // keep strong reference on the record thread so that 7010 // it is not destroyed while exit() is executed 7011 sp<RecordThread> thread; 7012 { 7013 Mutex::Autolock _l(mLock); 7014 thread = checkRecordThread_l(input); 7015 if (thread == NULL) { 7016 return BAD_VALUE; 7017 } 7018 7019 ALOGV("closeInput() %d", input); 7020 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7021 mRecordThreads.removeItem(input); 7022 } 7023 thread->exit(); 7024 // The thread entity (active unit of execution) is no longer running here, 7025 // but the ThreadBase container still exists. 7026 7027 AudioStreamIn *in = thread->clearInput(); 7028 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7029 // from now on thread->mInput is NULL 7030 in->hwDev->close_input_stream(in->hwDev, in->stream); 7031 delete in; 7032 7033 return NO_ERROR; 7034} 7035 7036status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7037{ 7038 Mutex::Autolock _l(mLock); 7039 MixerThread *dstThread = checkMixerThread_l(output); 7040 if (dstThread == NULL) { 7041 ALOGW("setStreamOutput() bad output id %d", output); 7042 return BAD_VALUE; 7043 } 7044 7045 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7046 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7047 7048 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7049 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7050 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7051 MixerThread *srcThread = (MixerThread *)thread; 7052 srcThread->invalidateTracks(stream); 7053 } 7054 } 7055 7056 return NO_ERROR; 7057} 7058 7059 7060int AudioFlinger::newAudioSessionId() 7061{ 7062 return nextUniqueId(); 7063} 7064 7065void AudioFlinger::acquireAudioSessionId(int audioSession) 7066{ 7067 Mutex::Autolock _l(mLock); 7068 pid_t caller = IPCThreadState::self()->getCallingPid(); 7069 ALOGV("acquiring %d from %d", audioSession, caller); 7070 size_t num = mAudioSessionRefs.size(); 7071 for (size_t i = 0; i< num; i++) { 7072 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7073 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7074 ref->mCnt++; 7075 ALOGV(" incremented refcount to %d", ref->mCnt); 7076 return; 7077 } 7078 } 7079 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7080 ALOGV(" added new entry for %d", audioSession); 7081} 7082 7083void AudioFlinger::releaseAudioSessionId(int audioSession) 7084{ 7085 Mutex::Autolock _l(mLock); 7086 pid_t caller = IPCThreadState::self()->getCallingPid(); 7087 ALOGV("releasing %d from %d", audioSession, caller); 7088 size_t num = mAudioSessionRefs.size(); 7089 for (size_t i = 0; i< num; i++) { 7090 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7091 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7092 ref->mCnt--; 7093 ALOGV(" decremented refcount to %d", ref->mCnt); 7094 if (ref->mCnt == 0) { 7095 mAudioSessionRefs.removeAt(i); 7096 delete ref; 7097 purgeStaleEffects_l(); 7098 } 7099 return; 7100 } 7101 } 7102 ALOGW("session id %d not found for pid %d", audioSession, caller); 7103} 7104 7105void AudioFlinger::purgeStaleEffects_l() { 7106 7107 ALOGV("purging stale effects"); 7108 7109 Vector< sp<EffectChain> > chains; 7110 7111 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7112 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7113 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7114 sp<EffectChain> ec = t->mEffectChains[j]; 7115 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7116 chains.push(ec); 7117 } 7118 } 7119 } 7120 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7121 sp<RecordThread> t = mRecordThreads.valueAt(i); 7122 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7123 sp<EffectChain> ec = t->mEffectChains[j]; 7124 chains.push(ec); 7125 } 7126 } 7127 7128 for (size_t i = 0; i < chains.size(); i++) { 7129 sp<EffectChain> ec = chains[i]; 7130 int sessionid = ec->sessionId(); 7131 sp<ThreadBase> t = ec->mThread.promote(); 7132 if (t == 0) { 7133 continue; 7134 } 7135 size_t numsessionrefs = mAudioSessionRefs.size(); 7136 bool found = false; 7137 for (size_t k = 0; k < numsessionrefs; k++) { 7138 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7139 if (ref->mSessionid == sessionid) { 7140 ALOGV(" session %d still exists for %d with %d refs", 7141 sessionid, ref->mPid, ref->mCnt); 7142 found = true; 7143 break; 7144 } 7145 } 7146 if (!found) { 7147 // remove all effects from the chain 7148 while (ec->mEffects.size()) { 7149 sp<EffectModule> effect = ec->mEffects[0]; 7150 effect->unPin(); 7151 Mutex::Autolock _l (t->mLock); 7152 t->removeEffect_l(effect); 7153 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7154 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7155 if (handle != 0) { 7156 handle->mEffect.clear(); 7157 if (handle->mHasControl && handle->mEnabled) { 7158 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7159 } 7160 } 7161 } 7162 AudioSystem::unregisterEffect(effect->id()); 7163 } 7164 } 7165 } 7166 return; 7167} 7168 7169// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7170AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7171{ 7172 return mPlaybackThreads.valueFor(output).get(); 7173} 7174 7175// checkMixerThread_l() must be called with AudioFlinger::mLock held 7176AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7177{ 7178 PlaybackThread *thread = checkPlaybackThread_l(output); 7179 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7180} 7181 7182// checkRecordThread_l() must be called with AudioFlinger::mLock held 7183AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7184{ 7185 return mRecordThreads.valueFor(input).get(); 7186} 7187 7188uint32_t AudioFlinger::nextUniqueId() 7189{ 7190 return android_atomic_inc(&mNextUniqueId); 7191} 7192 7193AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7194{ 7195 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7196 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7197 AudioStreamOut *output = thread->getOutput(); 7198 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7199 return thread; 7200 } 7201 } 7202 return NULL; 7203} 7204 7205uint32_t AudioFlinger::primaryOutputDevice_l() const 7206{ 7207 PlaybackThread *thread = primaryPlaybackThread_l(); 7208 7209 if (thread == NULL) { 7210 return 0; 7211 } 7212 7213 return thread->device(); 7214} 7215 7216sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7217 int triggerSession, 7218 int listenerSession, 7219 sync_event_callback_t callBack, 7220 void *cookie) 7221{ 7222 Mutex::Autolock _l(mLock); 7223 7224 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7225 status_t playStatus = NAME_NOT_FOUND; 7226 status_t recStatus = NAME_NOT_FOUND; 7227 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7228 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7229 if (playStatus == NO_ERROR) { 7230 return event; 7231 } 7232 } 7233 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7234 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7235 if (recStatus == NO_ERROR) { 7236 return event; 7237 } 7238 } 7239 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7240 mPendingSyncEvents.add(event); 7241 } else { 7242 ALOGV("createSyncEvent() invalid event %d", event->type()); 7243 event.clear(); 7244 } 7245 return event; 7246} 7247 7248// ---------------------------------------------------------------------------- 7249// Effect management 7250// ---------------------------------------------------------------------------- 7251 7252 7253status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7254{ 7255 Mutex::Autolock _l(mLock); 7256 return EffectQueryNumberEffects(numEffects); 7257} 7258 7259status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7260{ 7261 Mutex::Autolock _l(mLock); 7262 return EffectQueryEffect(index, descriptor); 7263} 7264 7265status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7266 effect_descriptor_t *descriptor) const 7267{ 7268 Mutex::Autolock _l(mLock); 7269 return EffectGetDescriptor(pUuid, descriptor); 7270} 7271 7272 7273sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7274 effect_descriptor_t *pDesc, 7275 const sp<IEffectClient>& effectClient, 7276 int32_t priority, 7277 audio_io_handle_t io, 7278 int sessionId, 7279 status_t *status, 7280 int *id, 7281 int *enabled) 7282{ 7283 status_t lStatus = NO_ERROR; 7284 sp<EffectHandle> handle; 7285 effect_descriptor_t desc; 7286 7287 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7288 pid, effectClient.get(), priority, sessionId, io); 7289 7290 if (pDesc == NULL) { 7291 lStatus = BAD_VALUE; 7292 goto Exit; 7293 } 7294 7295 // check audio settings permission for global effects 7296 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7297 lStatus = PERMISSION_DENIED; 7298 goto Exit; 7299 } 7300 7301 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7302 // that can only be created by audio policy manager (running in same process) 7303 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7304 lStatus = PERMISSION_DENIED; 7305 goto Exit; 7306 } 7307 7308 if (io == 0) { 7309 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7310 // output must be specified by AudioPolicyManager when using session 7311 // AUDIO_SESSION_OUTPUT_STAGE 7312 lStatus = BAD_VALUE; 7313 goto Exit; 7314 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7315 // if the output returned by getOutputForEffect() is removed before we lock the 7316 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7317 // and we will exit safely 7318 io = AudioSystem::getOutputForEffect(&desc); 7319 } 7320 } 7321 7322 { 7323 Mutex::Autolock _l(mLock); 7324 7325 7326 if (!EffectIsNullUuid(&pDesc->uuid)) { 7327 // if uuid is specified, request effect descriptor 7328 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7329 if (lStatus < 0) { 7330 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7331 goto Exit; 7332 } 7333 } else { 7334 // if uuid is not specified, look for an available implementation 7335 // of the required type in effect factory 7336 if (EffectIsNullUuid(&pDesc->type)) { 7337 ALOGW("createEffect() no effect type"); 7338 lStatus = BAD_VALUE; 7339 goto Exit; 7340 } 7341 uint32_t numEffects = 0; 7342 effect_descriptor_t d; 7343 d.flags = 0; // prevent compiler warning 7344 bool found = false; 7345 7346 lStatus = EffectQueryNumberEffects(&numEffects); 7347 if (lStatus < 0) { 7348 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7349 goto Exit; 7350 } 7351 for (uint32_t i = 0; i < numEffects; i++) { 7352 lStatus = EffectQueryEffect(i, &desc); 7353 if (lStatus < 0) { 7354 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7355 continue; 7356 } 7357 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7358 // If matching type found save effect descriptor. If the session is 7359 // 0 and the effect is not auxiliary, continue enumeration in case 7360 // an auxiliary version of this effect type is available 7361 found = true; 7362 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7363 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7364 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7365 break; 7366 } 7367 } 7368 } 7369 if (!found) { 7370 lStatus = BAD_VALUE; 7371 ALOGW("createEffect() effect not found"); 7372 goto Exit; 7373 } 7374 // For same effect type, chose auxiliary version over insert version if 7375 // connect to output mix (Compliance to OpenSL ES) 7376 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7377 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7378 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7379 } 7380 } 7381 7382 // Do not allow auxiliary effects on a session different from 0 (output mix) 7383 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7384 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7385 lStatus = INVALID_OPERATION; 7386 goto Exit; 7387 } 7388 7389 // check recording permission for visualizer 7390 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7391 !recordingAllowed()) { 7392 lStatus = PERMISSION_DENIED; 7393 goto Exit; 7394 } 7395 7396 // return effect descriptor 7397 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7398 7399 // If output is not specified try to find a matching audio session ID in one of the 7400 // output threads. 7401 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7402 // because of code checking output when entering the function. 7403 // Note: io is never 0 when creating an effect on an input 7404 if (io == 0) { 7405 // look for the thread where the specified audio session is present 7406 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7407 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7408 io = mPlaybackThreads.keyAt(i); 7409 break; 7410 } 7411 } 7412 if (io == 0) { 7413 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7414 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7415 io = mRecordThreads.keyAt(i); 7416 break; 7417 } 7418 } 7419 } 7420 // If no output thread contains the requested session ID, default to 7421 // first output. The effect chain will be moved to the correct output 7422 // thread when a track with the same session ID is created 7423 if (io == 0 && mPlaybackThreads.size()) { 7424 io = mPlaybackThreads.keyAt(0); 7425 } 7426 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7427 } 7428 ThreadBase *thread = checkRecordThread_l(io); 7429 if (thread == NULL) { 7430 thread = checkPlaybackThread_l(io); 7431 if (thread == NULL) { 7432 ALOGE("createEffect() unknown output thread"); 7433 lStatus = BAD_VALUE; 7434 goto Exit; 7435 } 7436 } 7437 7438 sp<Client> client = registerPid_l(pid); 7439 7440 // create effect on selected output thread 7441 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7442 &desc, enabled, &lStatus); 7443 if (handle != 0 && id != NULL) { 7444 *id = handle->id(); 7445 } 7446 } 7447 7448Exit: 7449 if (status != NULL) { 7450 *status = lStatus; 7451 } 7452 return handle; 7453} 7454 7455status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7456 audio_io_handle_t dstOutput) 7457{ 7458 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7459 sessionId, srcOutput, dstOutput); 7460 Mutex::Autolock _l(mLock); 7461 if (srcOutput == dstOutput) { 7462 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7463 return NO_ERROR; 7464 } 7465 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7466 if (srcThread == NULL) { 7467 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7468 return BAD_VALUE; 7469 } 7470 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7471 if (dstThread == NULL) { 7472 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7473 return BAD_VALUE; 7474 } 7475 7476 Mutex::Autolock _dl(dstThread->mLock); 7477 Mutex::Autolock _sl(srcThread->mLock); 7478 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7479 7480 return NO_ERROR; 7481} 7482 7483// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7484status_t AudioFlinger::moveEffectChain_l(int sessionId, 7485 AudioFlinger::PlaybackThread *srcThread, 7486 AudioFlinger::PlaybackThread *dstThread, 7487 bool reRegister) 7488{ 7489 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7490 sessionId, srcThread, dstThread); 7491 7492 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7493 if (chain == 0) { 7494 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7495 sessionId, srcThread); 7496 return INVALID_OPERATION; 7497 } 7498 7499 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7500 // so that a new chain is created with correct parameters when first effect is added. This is 7501 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7502 // removed. 7503 srcThread->removeEffectChain_l(chain); 7504 7505 // transfer all effects one by one so that new effect chain is created on new thread with 7506 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7507 audio_io_handle_t dstOutput = dstThread->id(); 7508 sp<EffectChain> dstChain; 7509 uint32_t strategy = 0; // prevent compiler warning 7510 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7511 while (effect != 0) { 7512 srcThread->removeEffect_l(effect); 7513 dstThread->addEffect_l(effect); 7514 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7515 if (effect->state() == EffectModule::ACTIVE || 7516 effect->state() == EffectModule::STOPPING) { 7517 effect->start(); 7518 } 7519 // if the move request is not received from audio policy manager, the effect must be 7520 // re-registered with the new strategy and output 7521 if (dstChain == 0) { 7522 dstChain = effect->chain().promote(); 7523 if (dstChain == 0) { 7524 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7525 srcThread->addEffect_l(effect); 7526 return NO_INIT; 7527 } 7528 strategy = dstChain->strategy(); 7529 } 7530 if (reRegister) { 7531 AudioSystem::unregisterEffect(effect->id()); 7532 AudioSystem::registerEffect(&effect->desc(), 7533 dstOutput, 7534 strategy, 7535 sessionId, 7536 effect->id()); 7537 } 7538 effect = chain->getEffectFromId_l(0); 7539 } 7540 7541 return NO_ERROR; 7542} 7543 7544 7545// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7546sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7547 const sp<AudioFlinger::Client>& client, 7548 const sp<IEffectClient>& effectClient, 7549 int32_t priority, 7550 int sessionId, 7551 effect_descriptor_t *desc, 7552 int *enabled, 7553 status_t *status 7554 ) 7555{ 7556 sp<EffectModule> effect; 7557 sp<EffectHandle> handle; 7558 status_t lStatus; 7559 sp<EffectChain> chain; 7560 bool chainCreated = false; 7561 bool effectCreated = false; 7562 bool effectRegistered = false; 7563 7564 lStatus = initCheck(); 7565 if (lStatus != NO_ERROR) { 7566 ALOGW("createEffect_l() Audio driver not initialized."); 7567 goto Exit; 7568 } 7569 7570 // Do not allow effects with session ID 0 on direct output or duplicating threads 7571 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7572 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7573 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7574 desc->name, sessionId); 7575 lStatus = BAD_VALUE; 7576 goto Exit; 7577 } 7578 // Only Pre processor effects are allowed on input threads and only on input threads 7579 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7580 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7581 desc->name, desc->flags, mType); 7582 lStatus = BAD_VALUE; 7583 goto Exit; 7584 } 7585 7586 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7587 7588 { // scope for mLock 7589 Mutex::Autolock _l(mLock); 7590 7591 // check for existing effect chain with the requested audio session 7592 chain = getEffectChain_l(sessionId); 7593 if (chain == 0) { 7594 // create a new chain for this session 7595 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7596 chain = new EffectChain(this, sessionId); 7597 addEffectChain_l(chain); 7598 chain->setStrategy(getStrategyForSession_l(sessionId)); 7599 chainCreated = true; 7600 } else { 7601 effect = chain->getEffectFromDesc_l(desc); 7602 } 7603 7604 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7605 7606 if (effect == 0) { 7607 int id = mAudioFlinger->nextUniqueId(); 7608 // Check CPU and memory usage 7609 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7610 if (lStatus != NO_ERROR) { 7611 goto Exit; 7612 } 7613 effectRegistered = true; 7614 // create a new effect module if none present in the chain 7615 effect = new EffectModule(this, chain, desc, id, sessionId); 7616 lStatus = effect->status(); 7617 if (lStatus != NO_ERROR) { 7618 goto Exit; 7619 } 7620 lStatus = chain->addEffect_l(effect); 7621 if (lStatus != NO_ERROR) { 7622 goto Exit; 7623 } 7624 effectCreated = true; 7625 7626 effect->setDevice(mDevice); 7627 effect->setMode(mAudioFlinger->getMode()); 7628 } 7629 // create effect handle and connect it to effect module 7630 handle = new EffectHandle(effect, client, effectClient, priority); 7631 lStatus = effect->addHandle(handle); 7632 if (enabled != NULL) { 7633 *enabled = (int)effect->isEnabled(); 7634 } 7635 } 7636 7637Exit: 7638 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7639 Mutex::Autolock _l(mLock); 7640 if (effectCreated) { 7641 chain->removeEffect_l(effect); 7642 } 7643 if (effectRegistered) { 7644 AudioSystem::unregisterEffect(effect->id()); 7645 } 7646 if (chainCreated) { 7647 removeEffectChain_l(chain); 7648 } 7649 handle.clear(); 7650 } 7651 7652 if (status != NULL) { 7653 *status = lStatus; 7654 } 7655 return handle; 7656} 7657 7658sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7659{ 7660 sp<EffectChain> chain = getEffectChain_l(sessionId); 7661 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7662} 7663 7664// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7665// PlaybackThread::mLock held 7666status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7667{ 7668 // check for existing effect chain with the requested audio session 7669 int sessionId = effect->sessionId(); 7670 sp<EffectChain> chain = getEffectChain_l(sessionId); 7671 bool chainCreated = false; 7672 7673 if (chain == 0) { 7674 // create a new chain for this session 7675 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7676 chain = new EffectChain(this, sessionId); 7677 addEffectChain_l(chain); 7678 chain->setStrategy(getStrategyForSession_l(sessionId)); 7679 chainCreated = true; 7680 } 7681 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7682 7683 if (chain->getEffectFromId_l(effect->id()) != 0) { 7684 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7685 this, effect->desc().name, chain.get()); 7686 return BAD_VALUE; 7687 } 7688 7689 status_t status = chain->addEffect_l(effect); 7690 if (status != NO_ERROR) { 7691 if (chainCreated) { 7692 removeEffectChain_l(chain); 7693 } 7694 return status; 7695 } 7696 7697 effect->setDevice(mDevice); 7698 effect->setMode(mAudioFlinger->getMode()); 7699 return NO_ERROR; 7700} 7701 7702void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7703 7704 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7705 effect_descriptor_t desc = effect->desc(); 7706 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7707 detachAuxEffect_l(effect->id()); 7708 } 7709 7710 sp<EffectChain> chain = effect->chain().promote(); 7711 if (chain != 0) { 7712 // remove effect chain if removing last effect 7713 if (chain->removeEffect_l(effect) == 0) { 7714 removeEffectChain_l(chain); 7715 } 7716 } else { 7717 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7718 } 7719} 7720 7721void AudioFlinger::ThreadBase::lockEffectChains_l( 7722 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7723{ 7724 effectChains = mEffectChains; 7725 for (size_t i = 0; i < mEffectChains.size(); i++) { 7726 mEffectChains[i]->lock(); 7727 } 7728} 7729 7730void AudioFlinger::ThreadBase::unlockEffectChains( 7731 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7732{ 7733 for (size_t i = 0; i < effectChains.size(); i++) { 7734 effectChains[i]->unlock(); 7735 } 7736} 7737 7738sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7739{ 7740 Mutex::Autolock _l(mLock); 7741 return getEffectChain_l(sessionId); 7742} 7743 7744sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7745{ 7746 size_t size = mEffectChains.size(); 7747 for (size_t i = 0; i < size; i++) { 7748 if (mEffectChains[i]->sessionId() == sessionId) { 7749 return mEffectChains[i]; 7750 } 7751 } 7752 return 0; 7753} 7754 7755void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7756{ 7757 Mutex::Autolock _l(mLock); 7758 size_t size = mEffectChains.size(); 7759 for (size_t i = 0; i < size; i++) { 7760 mEffectChains[i]->setMode_l(mode); 7761 } 7762} 7763 7764void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7765 const wp<EffectHandle>& handle, 7766 bool unpinIfLast) { 7767 7768 Mutex::Autolock _l(mLock); 7769 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7770 // delete the effect module if removing last handle on it 7771 if (effect->removeHandle(handle) == 0) { 7772 if (!effect->isPinned() || unpinIfLast) { 7773 removeEffect_l(effect); 7774 AudioSystem::unregisterEffect(effect->id()); 7775 } 7776 } 7777} 7778 7779status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7780{ 7781 int session = chain->sessionId(); 7782 int16_t *buffer = mMixBuffer; 7783 bool ownsBuffer = false; 7784 7785 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7786 if (session > 0) { 7787 // Only one effect chain can be present in direct output thread and it uses 7788 // the mix buffer as input 7789 if (mType != DIRECT) { 7790 size_t numSamples = mNormalFrameCount * mChannelCount; 7791 buffer = new int16_t[numSamples]; 7792 memset(buffer, 0, numSamples * sizeof(int16_t)); 7793 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7794 ownsBuffer = true; 7795 } 7796 7797 // Attach all tracks with same session ID to this chain. 7798 for (size_t i = 0; i < mTracks.size(); ++i) { 7799 sp<Track> track = mTracks[i]; 7800 if (session == track->sessionId()) { 7801 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7802 track->setMainBuffer(buffer); 7803 chain->incTrackCnt(); 7804 } 7805 } 7806 7807 // indicate all active tracks in the chain 7808 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7809 sp<Track> track = mActiveTracks[i].promote(); 7810 if (track == 0) continue; 7811 if (session == track->sessionId()) { 7812 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7813 chain->incActiveTrackCnt(); 7814 } 7815 } 7816 } 7817 7818 chain->setInBuffer(buffer, ownsBuffer); 7819 chain->setOutBuffer(mMixBuffer); 7820 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7821 // chains list in order to be processed last as it contains output stage effects 7822 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7823 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7824 // after track specific effects and before output stage 7825 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7826 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7827 // Effect chain for other sessions are inserted at beginning of effect 7828 // chains list to be processed before output mix effects. Relative order between other 7829 // sessions is not important 7830 size_t size = mEffectChains.size(); 7831 size_t i = 0; 7832 for (i = 0; i < size; i++) { 7833 if (mEffectChains[i]->sessionId() < session) break; 7834 } 7835 mEffectChains.insertAt(chain, i); 7836 checkSuspendOnAddEffectChain_l(chain); 7837 7838 return NO_ERROR; 7839} 7840 7841size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7842{ 7843 int session = chain->sessionId(); 7844 7845 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7846 7847 for (size_t i = 0; i < mEffectChains.size(); i++) { 7848 if (chain == mEffectChains[i]) { 7849 mEffectChains.removeAt(i); 7850 // detach all active tracks from the chain 7851 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7852 sp<Track> track = mActiveTracks[i].promote(); 7853 if (track == 0) continue; 7854 if (session == track->sessionId()) { 7855 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7856 chain.get(), session); 7857 chain->decActiveTrackCnt(); 7858 } 7859 } 7860 7861 // detach all tracks with same session ID from this chain 7862 for (size_t i = 0; i < mTracks.size(); ++i) { 7863 sp<Track> track = mTracks[i]; 7864 if (session == track->sessionId()) { 7865 track->setMainBuffer(mMixBuffer); 7866 chain->decTrackCnt(); 7867 } 7868 } 7869 break; 7870 } 7871 } 7872 return mEffectChains.size(); 7873} 7874 7875status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7876 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7877{ 7878 Mutex::Autolock _l(mLock); 7879 return attachAuxEffect_l(track, EffectId); 7880} 7881 7882status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7883 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7884{ 7885 status_t status = NO_ERROR; 7886 7887 if (EffectId == 0) { 7888 track->setAuxBuffer(0, NULL); 7889 } else { 7890 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7891 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7892 if (effect != 0) { 7893 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7894 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7895 } else { 7896 status = INVALID_OPERATION; 7897 } 7898 } else { 7899 status = BAD_VALUE; 7900 } 7901 } 7902 return status; 7903} 7904 7905void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7906{ 7907 for (size_t i = 0; i < mTracks.size(); ++i) { 7908 sp<Track> track = mTracks[i]; 7909 if (track->auxEffectId() == effectId) { 7910 attachAuxEffect_l(track, 0); 7911 } 7912 } 7913} 7914 7915status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7916{ 7917 // only one chain per input thread 7918 if (mEffectChains.size() != 0) { 7919 return INVALID_OPERATION; 7920 } 7921 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7922 7923 chain->setInBuffer(NULL); 7924 chain->setOutBuffer(NULL); 7925 7926 checkSuspendOnAddEffectChain_l(chain); 7927 7928 mEffectChains.add(chain); 7929 7930 return NO_ERROR; 7931} 7932 7933size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7934{ 7935 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7936 ALOGW_IF(mEffectChains.size() != 1, 7937 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7938 chain.get(), mEffectChains.size(), this); 7939 if (mEffectChains.size() == 1) { 7940 mEffectChains.removeAt(0); 7941 } 7942 return 0; 7943} 7944 7945// ---------------------------------------------------------------------------- 7946// EffectModule implementation 7947// ---------------------------------------------------------------------------- 7948 7949#undef LOG_TAG 7950#define LOG_TAG "AudioFlinger::EffectModule" 7951 7952AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7953 const wp<AudioFlinger::EffectChain>& chain, 7954 effect_descriptor_t *desc, 7955 int id, 7956 int sessionId) 7957 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7958 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7959{ 7960 ALOGV("Constructor %p", this); 7961 int lStatus; 7962 if (thread == NULL) { 7963 return; 7964 } 7965 7966 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7967 7968 // create effect engine from effect factory 7969 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7970 7971 if (mStatus != NO_ERROR) { 7972 return; 7973 } 7974 lStatus = init(); 7975 if (lStatus < 0) { 7976 mStatus = lStatus; 7977 goto Error; 7978 } 7979 7980 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7981 mPinned = true; 7982 } 7983 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7984 return; 7985Error: 7986 EffectRelease(mEffectInterface); 7987 mEffectInterface = NULL; 7988 ALOGV("Constructor Error %d", mStatus); 7989} 7990 7991AudioFlinger::EffectModule::~EffectModule() 7992{ 7993 ALOGV("Destructor %p", this); 7994 if (mEffectInterface != NULL) { 7995 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7996 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7997 sp<ThreadBase> thread = mThread.promote(); 7998 if (thread != 0) { 7999 audio_stream_t *stream = thread->stream(); 8000 if (stream != NULL) { 8001 stream->remove_audio_effect(stream, mEffectInterface); 8002 } 8003 } 8004 } 8005 // release effect engine 8006 EffectRelease(mEffectInterface); 8007 } 8008} 8009 8010status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8011{ 8012 status_t status; 8013 8014 Mutex::Autolock _l(mLock); 8015 int priority = handle->priority(); 8016 size_t size = mHandles.size(); 8017 sp<EffectHandle> h; 8018 size_t i; 8019 for (i = 0; i < size; i++) { 8020 h = mHandles[i].promote(); 8021 if (h == 0) continue; 8022 if (h->priority() <= priority) break; 8023 } 8024 // if inserted in first place, move effect control from previous owner to this handle 8025 if (i == 0) { 8026 bool enabled = false; 8027 if (h != 0) { 8028 enabled = h->enabled(); 8029 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8030 } 8031 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8032 status = NO_ERROR; 8033 } else { 8034 status = ALREADY_EXISTS; 8035 } 8036 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8037 mHandles.insertAt(handle, i); 8038 return status; 8039} 8040 8041size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8042{ 8043 Mutex::Autolock _l(mLock); 8044 size_t size = mHandles.size(); 8045 size_t i; 8046 for (i = 0; i < size; i++) { 8047 if (mHandles[i] == handle) break; 8048 } 8049 if (i == size) { 8050 return size; 8051 } 8052 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8053 8054 bool enabled = false; 8055 EffectHandle *hdl = handle.unsafe_get(); 8056 if (hdl != NULL) { 8057 ALOGV("removeHandle() unsafe_get OK"); 8058 enabled = hdl->enabled(); 8059 } 8060 mHandles.removeAt(i); 8061 size = mHandles.size(); 8062 // if removed from first place, move effect control from this handle to next in line 8063 if (i == 0 && size != 0) { 8064 sp<EffectHandle> h = mHandles[0].promote(); 8065 if (h != 0) { 8066 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8067 } 8068 } 8069 8070 // Prevent calls to process() and other functions on effect interface from now on. 8071 // The effect engine will be released by the destructor when the last strong reference on 8072 // this object is released which can happen after next process is called. 8073 if (size == 0 && !mPinned) { 8074 mState = DESTROYED; 8075 } 8076 8077 return size; 8078} 8079 8080sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8081{ 8082 Mutex::Autolock _l(mLock); 8083 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8084} 8085 8086void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8087{ 8088 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8089 // keep a strong reference on this EffectModule to avoid calling the 8090 // destructor before we exit 8091 sp<EffectModule> keep(this); 8092 { 8093 sp<ThreadBase> thread = mThread.promote(); 8094 if (thread != 0) { 8095 thread->disconnectEffect(keep, handle, unpinIfLast); 8096 } 8097 } 8098} 8099 8100void AudioFlinger::EffectModule::updateState() { 8101 Mutex::Autolock _l(mLock); 8102 8103 switch (mState) { 8104 case RESTART: 8105 reset_l(); 8106 // FALL THROUGH 8107 8108 case STARTING: 8109 // clear auxiliary effect input buffer for next accumulation 8110 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8111 memset(mConfig.inputCfg.buffer.raw, 8112 0, 8113 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8114 } 8115 start_l(); 8116 mState = ACTIVE; 8117 break; 8118 case STOPPING: 8119 stop_l(); 8120 mDisableWaitCnt = mMaxDisableWaitCnt; 8121 mState = STOPPED; 8122 break; 8123 case STOPPED: 8124 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8125 // turn off sequence. 8126 if (--mDisableWaitCnt == 0) { 8127 reset_l(); 8128 mState = IDLE; 8129 } 8130 break; 8131 default: //IDLE , ACTIVE, DESTROYED 8132 break; 8133 } 8134} 8135 8136void AudioFlinger::EffectModule::process() 8137{ 8138 Mutex::Autolock _l(mLock); 8139 8140 if (mState == DESTROYED || mEffectInterface == NULL || 8141 mConfig.inputCfg.buffer.raw == NULL || 8142 mConfig.outputCfg.buffer.raw == NULL) { 8143 return; 8144 } 8145 8146 if (isProcessEnabled()) { 8147 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8148 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8149 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8150 mConfig.inputCfg.buffer.s32, 8151 mConfig.inputCfg.buffer.frameCount/2); 8152 } 8153 8154 // do the actual processing in the effect engine 8155 int ret = (*mEffectInterface)->process(mEffectInterface, 8156 &mConfig.inputCfg.buffer, 8157 &mConfig.outputCfg.buffer); 8158 8159 // force transition to IDLE state when engine is ready 8160 if (mState == STOPPED && ret == -ENODATA) { 8161 mDisableWaitCnt = 1; 8162 } 8163 8164 // clear auxiliary effect input buffer for next accumulation 8165 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8166 memset(mConfig.inputCfg.buffer.raw, 0, 8167 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8168 } 8169 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8170 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8171 // If an insert effect is idle and input buffer is different from output buffer, 8172 // accumulate input onto output 8173 sp<EffectChain> chain = mChain.promote(); 8174 if (chain != 0 && chain->activeTrackCnt() != 0) { 8175 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8176 int16_t *in = mConfig.inputCfg.buffer.s16; 8177 int16_t *out = mConfig.outputCfg.buffer.s16; 8178 for (size_t i = 0; i < frameCnt; i++) { 8179 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8180 } 8181 } 8182 } 8183} 8184 8185void AudioFlinger::EffectModule::reset_l() 8186{ 8187 if (mEffectInterface == NULL) { 8188 return; 8189 } 8190 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8191} 8192 8193status_t AudioFlinger::EffectModule::configure() 8194{ 8195 uint32_t channels; 8196 if (mEffectInterface == NULL) { 8197 return NO_INIT; 8198 } 8199 8200 sp<ThreadBase> thread = mThread.promote(); 8201 if (thread == 0) { 8202 return DEAD_OBJECT; 8203 } 8204 8205 // TODO: handle configuration of effects replacing track process 8206 if (thread->channelCount() == 1) { 8207 channels = AUDIO_CHANNEL_OUT_MONO; 8208 } else { 8209 channels = AUDIO_CHANNEL_OUT_STEREO; 8210 } 8211 8212 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8213 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8214 } else { 8215 mConfig.inputCfg.channels = channels; 8216 } 8217 mConfig.outputCfg.channels = channels; 8218 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8219 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8220 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8221 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8222 mConfig.inputCfg.bufferProvider.cookie = NULL; 8223 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8224 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8225 mConfig.outputCfg.bufferProvider.cookie = NULL; 8226 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8227 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8228 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8229 // Insert effect: 8230 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8231 // always overwrites output buffer: input buffer == output buffer 8232 // - in other sessions: 8233 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8234 // other effect: overwrites output buffer: input buffer == output buffer 8235 // Auxiliary effect: 8236 // accumulates in output buffer: input buffer != output buffer 8237 // Therefore: accumulate <=> input buffer != output buffer 8238 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8239 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8240 } else { 8241 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8242 } 8243 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8244 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8245 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8246 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8247 8248 ALOGV("configure() %p thread %p buffer %p framecount %d", 8249 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8250 8251 status_t cmdStatus; 8252 uint32_t size = sizeof(int); 8253 status_t status = (*mEffectInterface)->command(mEffectInterface, 8254 EFFECT_CMD_SET_CONFIG, 8255 sizeof(effect_config_t), 8256 &mConfig, 8257 &size, 8258 &cmdStatus); 8259 if (status == 0) { 8260 status = cmdStatus; 8261 } 8262 8263 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8264 (1000 * mConfig.outputCfg.buffer.frameCount); 8265 8266 return status; 8267} 8268 8269status_t AudioFlinger::EffectModule::init() 8270{ 8271 Mutex::Autolock _l(mLock); 8272 if (mEffectInterface == NULL) { 8273 return NO_INIT; 8274 } 8275 status_t cmdStatus; 8276 uint32_t size = sizeof(status_t); 8277 status_t status = (*mEffectInterface)->command(mEffectInterface, 8278 EFFECT_CMD_INIT, 8279 0, 8280 NULL, 8281 &size, 8282 &cmdStatus); 8283 if (status == 0) { 8284 status = cmdStatus; 8285 } 8286 return status; 8287} 8288 8289status_t AudioFlinger::EffectModule::start() 8290{ 8291 Mutex::Autolock _l(mLock); 8292 return start_l(); 8293} 8294 8295status_t AudioFlinger::EffectModule::start_l() 8296{ 8297 if (mEffectInterface == NULL) { 8298 return NO_INIT; 8299 } 8300 status_t cmdStatus; 8301 uint32_t size = sizeof(status_t); 8302 status_t status = (*mEffectInterface)->command(mEffectInterface, 8303 EFFECT_CMD_ENABLE, 8304 0, 8305 NULL, 8306 &size, 8307 &cmdStatus); 8308 if (status == 0) { 8309 status = cmdStatus; 8310 } 8311 if (status == 0 && 8312 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8313 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8314 sp<ThreadBase> thread = mThread.promote(); 8315 if (thread != 0) { 8316 audio_stream_t *stream = thread->stream(); 8317 if (stream != NULL) { 8318 stream->add_audio_effect(stream, mEffectInterface); 8319 } 8320 } 8321 } 8322 return status; 8323} 8324 8325status_t AudioFlinger::EffectModule::stop() 8326{ 8327 Mutex::Autolock _l(mLock); 8328 return stop_l(); 8329} 8330 8331status_t AudioFlinger::EffectModule::stop_l() 8332{ 8333 if (mEffectInterface == NULL) { 8334 return NO_INIT; 8335 } 8336 status_t cmdStatus; 8337 uint32_t size = sizeof(status_t); 8338 status_t status = (*mEffectInterface)->command(mEffectInterface, 8339 EFFECT_CMD_DISABLE, 8340 0, 8341 NULL, 8342 &size, 8343 &cmdStatus); 8344 if (status == 0) { 8345 status = cmdStatus; 8346 } 8347 if (status == 0 && 8348 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8349 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8350 sp<ThreadBase> thread = mThread.promote(); 8351 if (thread != 0) { 8352 audio_stream_t *stream = thread->stream(); 8353 if (stream != NULL) { 8354 stream->remove_audio_effect(stream, mEffectInterface); 8355 } 8356 } 8357 } 8358 return status; 8359} 8360 8361status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8362 uint32_t cmdSize, 8363 void *pCmdData, 8364 uint32_t *replySize, 8365 void *pReplyData) 8366{ 8367 Mutex::Autolock _l(mLock); 8368// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8369 8370 if (mState == DESTROYED || mEffectInterface == NULL) { 8371 return NO_INIT; 8372 } 8373 status_t status = (*mEffectInterface)->command(mEffectInterface, 8374 cmdCode, 8375 cmdSize, 8376 pCmdData, 8377 replySize, 8378 pReplyData); 8379 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8380 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8381 for (size_t i = 1; i < mHandles.size(); i++) { 8382 sp<EffectHandle> h = mHandles[i].promote(); 8383 if (h != 0) { 8384 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8385 } 8386 } 8387 } 8388 return status; 8389} 8390 8391status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8392{ 8393 8394 Mutex::Autolock _l(mLock); 8395 ALOGV("setEnabled %p enabled %d", this, enabled); 8396 8397 if (enabled != isEnabled()) { 8398 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8399 if (enabled && status != NO_ERROR) { 8400 return status; 8401 } 8402 8403 switch (mState) { 8404 // going from disabled to enabled 8405 case IDLE: 8406 mState = STARTING; 8407 break; 8408 case STOPPED: 8409 mState = RESTART; 8410 break; 8411 case STOPPING: 8412 mState = ACTIVE; 8413 break; 8414 8415 // going from enabled to disabled 8416 case RESTART: 8417 mState = STOPPED; 8418 break; 8419 case STARTING: 8420 mState = IDLE; 8421 break; 8422 case ACTIVE: 8423 mState = STOPPING; 8424 break; 8425 case DESTROYED: 8426 return NO_ERROR; // simply ignore as we are being destroyed 8427 } 8428 for (size_t i = 1; i < mHandles.size(); i++) { 8429 sp<EffectHandle> h = mHandles[i].promote(); 8430 if (h != 0) { 8431 h->setEnabled(enabled); 8432 } 8433 } 8434 } 8435 return NO_ERROR; 8436} 8437 8438bool AudioFlinger::EffectModule::isEnabled() const 8439{ 8440 switch (mState) { 8441 case RESTART: 8442 case STARTING: 8443 case ACTIVE: 8444 return true; 8445 case IDLE: 8446 case STOPPING: 8447 case STOPPED: 8448 case DESTROYED: 8449 default: 8450 return false; 8451 } 8452} 8453 8454bool AudioFlinger::EffectModule::isProcessEnabled() const 8455{ 8456 switch (mState) { 8457 case RESTART: 8458 case ACTIVE: 8459 case STOPPING: 8460 case STOPPED: 8461 return true; 8462 case IDLE: 8463 case STARTING: 8464 case DESTROYED: 8465 default: 8466 return false; 8467 } 8468} 8469 8470status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8471{ 8472 Mutex::Autolock _l(mLock); 8473 status_t status = NO_ERROR; 8474 8475 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8476 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8477 if (isProcessEnabled() && 8478 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8479 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8480 status_t cmdStatus; 8481 uint32_t volume[2]; 8482 uint32_t *pVolume = NULL; 8483 uint32_t size = sizeof(volume); 8484 volume[0] = *left; 8485 volume[1] = *right; 8486 if (controller) { 8487 pVolume = volume; 8488 } 8489 status = (*mEffectInterface)->command(mEffectInterface, 8490 EFFECT_CMD_SET_VOLUME, 8491 size, 8492 volume, 8493 &size, 8494 pVolume); 8495 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8496 *left = volume[0]; 8497 *right = volume[1]; 8498 } 8499 } 8500 return status; 8501} 8502 8503status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8504{ 8505 Mutex::Autolock _l(mLock); 8506 status_t status = NO_ERROR; 8507 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8508 // audio pre processing modules on RecordThread can receive both output and 8509 // input device indication in the same call 8510 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8511 if (dev) { 8512 status_t cmdStatus; 8513 uint32_t size = sizeof(status_t); 8514 8515 status = (*mEffectInterface)->command(mEffectInterface, 8516 EFFECT_CMD_SET_DEVICE, 8517 sizeof(uint32_t), 8518 &dev, 8519 &size, 8520 &cmdStatus); 8521 if (status == NO_ERROR) { 8522 status = cmdStatus; 8523 } 8524 } 8525 dev = device & AUDIO_DEVICE_IN_ALL; 8526 if (dev) { 8527 status_t cmdStatus; 8528 uint32_t size = sizeof(status_t); 8529 8530 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8531 EFFECT_CMD_SET_INPUT_DEVICE, 8532 sizeof(uint32_t), 8533 &dev, 8534 &size, 8535 &cmdStatus); 8536 if (status2 == NO_ERROR) { 8537 status2 = cmdStatus; 8538 } 8539 if (status == NO_ERROR) { 8540 status = status2; 8541 } 8542 } 8543 } 8544 return status; 8545} 8546 8547status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8548{ 8549 Mutex::Autolock _l(mLock); 8550 status_t status = NO_ERROR; 8551 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8552 status_t cmdStatus; 8553 uint32_t size = sizeof(status_t); 8554 status = (*mEffectInterface)->command(mEffectInterface, 8555 EFFECT_CMD_SET_AUDIO_MODE, 8556 sizeof(audio_mode_t), 8557 &mode, 8558 &size, 8559 &cmdStatus); 8560 if (status == NO_ERROR) { 8561 status = cmdStatus; 8562 } 8563 } 8564 return status; 8565} 8566 8567void AudioFlinger::EffectModule::setSuspended(bool suspended) 8568{ 8569 Mutex::Autolock _l(mLock); 8570 mSuspended = suspended; 8571} 8572 8573bool AudioFlinger::EffectModule::suspended() const 8574{ 8575 Mutex::Autolock _l(mLock); 8576 return mSuspended; 8577} 8578 8579status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8580{ 8581 const size_t SIZE = 256; 8582 char buffer[SIZE]; 8583 String8 result; 8584 8585 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8586 result.append(buffer); 8587 8588 bool locked = tryLock(mLock); 8589 // failed to lock - AudioFlinger is probably deadlocked 8590 if (!locked) { 8591 result.append("\t\tCould not lock Fx mutex:\n"); 8592 } 8593 8594 result.append("\t\tSession Status State Engine:\n"); 8595 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8596 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8597 result.append(buffer); 8598 8599 result.append("\t\tDescriptor:\n"); 8600 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8601 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8602 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8603 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8604 result.append(buffer); 8605 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8606 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8607 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8608 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8609 result.append(buffer); 8610 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8611 mDescriptor.apiVersion, 8612 mDescriptor.flags); 8613 result.append(buffer); 8614 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8615 mDescriptor.name); 8616 result.append(buffer); 8617 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8618 mDescriptor.implementor); 8619 result.append(buffer); 8620 8621 result.append("\t\t- Input configuration:\n"); 8622 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8623 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8624 (uint32_t)mConfig.inputCfg.buffer.raw, 8625 mConfig.inputCfg.buffer.frameCount, 8626 mConfig.inputCfg.samplingRate, 8627 mConfig.inputCfg.channels, 8628 mConfig.inputCfg.format); 8629 result.append(buffer); 8630 8631 result.append("\t\t- Output configuration:\n"); 8632 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8633 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8634 (uint32_t)mConfig.outputCfg.buffer.raw, 8635 mConfig.outputCfg.buffer.frameCount, 8636 mConfig.outputCfg.samplingRate, 8637 mConfig.outputCfg.channels, 8638 mConfig.outputCfg.format); 8639 result.append(buffer); 8640 8641 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8642 result.append(buffer); 8643 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8644 for (size_t i = 0; i < mHandles.size(); ++i) { 8645 sp<EffectHandle> handle = mHandles[i].promote(); 8646 if (handle != 0) { 8647 handle->dump(buffer, SIZE); 8648 result.append(buffer); 8649 } 8650 } 8651 8652 result.append("\n"); 8653 8654 write(fd, result.string(), result.length()); 8655 8656 if (locked) { 8657 mLock.unlock(); 8658 } 8659 8660 return NO_ERROR; 8661} 8662 8663// ---------------------------------------------------------------------------- 8664// EffectHandle implementation 8665// ---------------------------------------------------------------------------- 8666 8667#undef LOG_TAG 8668#define LOG_TAG "AudioFlinger::EffectHandle" 8669 8670AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8671 const sp<AudioFlinger::Client>& client, 8672 const sp<IEffectClient>& effectClient, 8673 int32_t priority) 8674 : BnEffect(), 8675 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8676 mPriority(priority), mHasControl(false), mEnabled(false) 8677{ 8678 ALOGV("constructor %p", this); 8679 8680 if (client == 0) { 8681 return; 8682 } 8683 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8684 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8685 if (mCblkMemory != 0) { 8686 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8687 8688 if (mCblk != NULL) { 8689 new(mCblk) effect_param_cblk_t(); 8690 mBuffer = (uint8_t *)mCblk + bufOffset; 8691 } 8692 } else { 8693 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8694 return; 8695 } 8696} 8697 8698AudioFlinger::EffectHandle::~EffectHandle() 8699{ 8700 ALOGV("Destructor %p", this); 8701 disconnect(false); 8702 ALOGV("Destructor DONE %p", this); 8703} 8704 8705status_t AudioFlinger::EffectHandle::enable() 8706{ 8707 ALOGV("enable %p", this); 8708 if (!mHasControl) return INVALID_OPERATION; 8709 if (mEffect == 0) return DEAD_OBJECT; 8710 8711 if (mEnabled) { 8712 return NO_ERROR; 8713 } 8714 8715 mEnabled = true; 8716 8717 sp<ThreadBase> thread = mEffect->thread().promote(); 8718 if (thread != 0) { 8719 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8720 } 8721 8722 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8723 if (mEffect->suspended()) { 8724 return NO_ERROR; 8725 } 8726 8727 status_t status = mEffect->setEnabled(true); 8728 if (status != NO_ERROR) { 8729 if (thread != 0) { 8730 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8731 } 8732 mEnabled = false; 8733 } 8734 return status; 8735} 8736 8737status_t AudioFlinger::EffectHandle::disable() 8738{ 8739 ALOGV("disable %p", this); 8740 if (!mHasControl) return INVALID_OPERATION; 8741 if (mEffect == 0) return DEAD_OBJECT; 8742 8743 if (!mEnabled) { 8744 return NO_ERROR; 8745 } 8746 mEnabled = false; 8747 8748 if (mEffect->suspended()) { 8749 return NO_ERROR; 8750 } 8751 8752 status_t status = mEffect->setEnabled(false); 8753 8754 sp<ThreadBase> thread = mEffect->thread().promote(); 8755 if (thread != 0) { 8756 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8757 } 8758 8759 return status; 8760} 8761 8762void AudioFlinger::EffectHandle::disconnect() 8763{ 8764 disconnect(true); 8765} 8766 8767void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8768{ 8769 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8770 if (mEffect == 0) { 8771 return; 8772 } 8773 mEffect->disconnect(this, unpinIfLast); 8774 8775 if (mHasControl && mEnabled) { 8776 sp<ThreadBase> thread = mEffect->thread().promote(); 8777 if (thread != 0) { 8778 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8779 } 8780 } 8781 8782 // release sp on module => module destructor can be called now 8783 mEffect.clear(); 8784 if (mClient != 0) { 8785 if (mCblk != NULL) { 8786 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8787 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8788 } 8789 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8790 // Client destructor must run with AudioFlinger mutex locked 8791 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8792 mClient.clear(); 8793 } 8794} 8795 8796status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8797 uint32_t cmdSize, 8798 void *pCmdData, 8799 uint32_t *replySize, 8800 void *pReplyData) 8801{ 8802// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8803// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8804 8805 // only get parameter command is permitted for applications not controlling the effect 8806 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8807 return INVALID_OPERATION; 8808 } 8809 if (mEffect == 0) return DEAD_OBJECT; 8810 if (mClient == 0) return INVALID_OPERATION; 8811 8812 // handle commands that are not forwarded transparently to effect engine 8813 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8814 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8815 // no risk to block the whole media server process or mixer threads is we are stuck here 8816 Mutex::Autolock _l(mCblk->lock); 8817 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8818 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8819 mCblk->serverIndex = 0; 8820 mCblk->clientIndex = 0; 8821 return BAD_VALUE; 8822 } 8823 status_t status = NO_ERROR; 8824 while (mCblk->serverIndex < mCblk->clientIndex) { 8825 int reply; 8826 uint32_t rsize = sizeof(int); 8827 int *p = (int *)(mBuffer + mCblk->serverIndex); 8828 int size = *p++; 8829 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8830 ALOGW("command(): invalid parameter block size"); 8831 break; 8832 } 8833 effect_param_t *param = (effect_param_t *)p; 8834 if (param->psize == 0 || param->vsize == 0) { 8835 ALOGW("command(): null parameter or value size"); 8836 mCblk->serverIndex += size; 8837 continue; 8838 } 8839 uint32_t psize = sizeof(effect_param_t) + 8840 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8841 param->vsize; 8842 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8843 psize, 8844 p, 8845 &rsize, 8846 &reply); 8847 // stop at first error encountered 8848 if (ret != NO_ERROR) { 8849 status = ret; 8850 *(int *)pReplyData = reply; 8851 break; 8852 } else if (reply != NO_ERROR) { 8853 *(int *)pReplyData = reply; 8854 break; 8855 } 8856 mCblk->serverIndex += size; 8857 } 8858 mCblk->serverIndex = 0; 8859 mCblk->clientIndex = 0; 8860 return status; 8861 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8862 *(int *)pReplyData = NO_ERROR; 8863 return enable(); 8864 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8865 *(int *)pReplyData = NO_ERROR; 8866 return disable(); 8867 } 8868 8869 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8870} 8871 8872void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8873{ 8874 ALOGV("setControl %p control %d", this, hasControl); 8875 8876 mHasControl = hasControl; 8877 mEnabled = enabled; 8878 8879 if (signal && mEffectClient != 0) { 8880 mEffectClient->controlStatusChanged(hasControl); 8881 } 8882} 8883 8884void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8885 uint32_t cmdSize, 8886 void *pCmdData, 8887 uint32_t replySize, 8888 void *pReplyData) 8889{ 8890 if (mEffectClient != 0) { 8891 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8892 } 8893} 8894 8895 8896 8897void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8898{ 8899 if (mEffectClient != 0) { 8900 mEffectClient->enableStatusChanged(enabled); 8901 } 8902} 8903 8904status_t AudioFlinger::EffectHandle::onTransact( 8905 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8906{ 8907 return BnEffect::onTransact(code, data, reply, flags); 8908} 8909 8910 8911void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8912{ 8913 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8914 8915 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8916 (mClient == 0) ? getpid_cached : mClient->pid(), 8917 mPriority, 8918 mHasControl, 8919 !locked, 8920 mCblk ? mCblk->clientIndex : 0, 8921 mCblk ? mCblk->serverIndex : 0 8922 ); 8923 8924 if (locked) { 8925 mCblk->lock.unlock(); 8926 } 8927} 8928 8929#undef LOG_TAG 8930#define LOG_TAG "AudioFlinger::EffectChain" 8931 8932AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8933 int sessionId) 8934 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8935 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8936 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8937{ 8938 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8939 if (thread == NULL) { 8940 return; 8941 } 8942 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8943 thread->frameCount(); 8944} 8945 8946AudioFlinger::EffectChain::~EffectChain() 8947{ 8948 if (mOwnInBuffer) { 8949 delete mInBuffer; 8950 } 8951 8952} 8953 8954// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8955sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8956{ 8957 size_t size = mEffects.size(); 8958 8959 for (size_t i = 0; i < size; i++) { 8960 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8961 return mEffects[i]; 8962 } 8963 } 8964 return 0; 8965} 8966 8967// getEffectFromId_l() must be called with ThreadBase::mLock held 8968sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8969{ 8970 size_t size = mEffects.size(); 8971 8972 for (size_t i = 0; i < size; i++) { 8973 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8974 if (id == 0 || mEffects[i]->id() == id) { 8975 return mEffects[i]; 8976 } 8977 } 8978 return 0; 8979} 8980 8981// getEffectFromType_l() must be called with ThreadBase::mLock held 8982sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8983 const effect_uuid_t *type) 8984{ 8985 size_t size = mEffects.size(); 8986 8987 for (size_t i = 0; i < size; i++) { 8988 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8989 return mEffects[i]; 8990 } 8991 } 8992 return 0; 8993} 8994 8995void AudioFlinger::EffectChain::clearInputBuffer() 8996{ 8997 Mutex::Autolock _l(mLock); 8998 sp<ThreadBase> thread = mThread.promote(); 8999 if (thread == 0) { 9000 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9001 return; 9002 } 9003 clearInputBuffer_l(thread); 9004} 9005 9006// Must be called with EffectChain::mLock locked 9007void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9008{ 9009 size_t numSamples = thread->frameCount() * thread->channelCount(); 9010 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9011 9012} 9013 9014// Must be called with EffectChain::mLock locked 9015void AudioFlinger::EffectChain::process_l() 9016{ 9017 sp<ThreadBase> thread = mThread.promote(); 9018 if (thread == 0) { 9019 ALOGW("process_l(): cannot promote mixer thread"); 9020 return; 9021 } 9022 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9023 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9024 // always process effects unless no more tracks are on the session and the effect tail 9025 // has been rendered 9026 bool doProcess = true; 9027 if (!isGlobalSession) { 9028 bool tracksOnSession = (trackCnt() != 0); 9029 9030 if (!tracksOnSession && mTailBufferCount == 0) { 9031 doProcess = false; 9032 } 9033 9034 if (activeTrackCnt() == 0) { 9035 // if no track is active and the effect tail has not been rendered, 9036 // the input buffer must be cleared here as the mixer process will not do it 9037 if (tracksOnSession || mTailBufferCount > 0) { 9038 clearInputBuffer_l(thread); 9039 if (mTailBufferCount > 0) { 9040 mTailBufferCount--; 9041 } 9042 } 9043 } 9044 } 9045 9046 size_t size = mEffects.size(); 9047 if (doProcess) { 9048 for (size_t i = 0; i < size; i++) { 9049 mEffects[i]->process(); 9050 } 9051 } 9052 for (size_t i = 0; i < size; i++) { 9053 mEffects[i]->updateState(); 9054 } 9055} 9056 9057// addEffect_l() must be called with PlaybackThread::mLock held 9058status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9059{ 9060 effect_descriptor_t desc = effect->desc(); 9061 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9062 9063 Mutex::Autolock _l(mLock); 9064 effect->setChain(this); 9065 sp<ThreadBase> thread = mThread.promote(); 9066 if (thread == 0) { 9067 return NO_INIT; 9068 } 9069 effect->setThread(thread); 9070 9071 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9072 // Auxiliary effects are inserted at the beginning of mEffects vector as 9073 // they are processed first and accumulated in chain input buffer 9074 mEffects.insertAt(effect, 0); 9075 9076 // the input buffer for auxiliary effect contains mono samples in 9077 // 32 bit format. This is to avoid saturation in AudoMixer 9078 // accumulation stage. Saturation is done in EffectModule::process() before 9079 // calling the process in effect engine 9080 size_t numSamples = thread->frameCount(); 9081 int32_t *buffer = new int32_t[numSamples]; 9082 memset(buffer, 0, numSamples * sizeof(int32_t)); 9083 effect->setInBuffer((int16_t *)buffer); 9084 // auxiliary effects output samples to chain input buffer for further processing 9085 // by insert effects 9086 effect->setOutBuffer(mInBuffer); 9087 } else { 9088 // Insert effects are inserted at the end of mEffects vector as they are processed 9089 // after track and auxiliary effects. 9090 // Insert effect order as a function of indicated preference: 9091 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9092 // another effect is present 9093 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9094 // last effect claiming first position 9095 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9096 // first effect claiming last position 9097 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9098 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9099 // already present 9100 9101 size_t size = mEffects.size(); 9102 size_t idx_insert = size; 9103 ssize_t idx_insert_first = -1; 9104 ssize_t idx_insert_last = -1; 9105 9106 for (size_t i = 0; i < size; i++) { 9107 effect_descriptor_t d = mEffects[i]->desc(); 9108 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9109 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9110 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9111 // check invalid effect chaining combinations 9112 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9113 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9114 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9115 return INVALID_OPERATION; 9116 } 9117 // remember position of first insert effect and by default 9118 // select this as insert position for new effect 9119 if (idx_insert == size) { 9120 idx_insert = i; 9121 } 9122 // remember position of last insert effect claiming 9123 // first position 9124 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9125 idx_insert_first = i; 9126 } 9127 // remember position of first insert effect claiming 9128 // last position 9129 if (iPref == EFFECT_FLAG_INSERT_LAST && 9130 idx_insert_last == -1) { 9131 idx_insert_last = i; 9132 } 9133 } 9134 } 9135 9136 // modify idx_insert from first position if needed 9137 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9138 if (idx_insert_last != -1) { 9139 idx_insert = idx_insert_last; 9140 } else { 9141 idx_insert = size; 9142 } 9143 } else { 9144 if (idx_insert_first != -1) { 9145 idx_insert = idx_insert_first + 1; 9146 } 9147 } 9148 9149 // always read samples from chain input buffer 9150 effect->setInBuffer(mInBuffer); 9151 9152 // if last effect in the chain, output samples to chain 9153 // output buffer, otherwise to chain input buffer 9154 if (idx_insert == size) { 9155 if (idx_insert != 0) { 9156 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9157 mEffects[idx_insert-1]->configure(); 9158 } 9159 effect->setOutBuffer(mOutBuffer); 9160 } else { 9161 effect->setOutBuffer(mInBuffer); 9162 } 9163 mEffects.insertAt(effect, idx_insert); 9164 9165 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9166 } 9167 effect->configure(); 9168 return NO_ERROR; 9169} 9170 9171// removeEffect_l() must be called with PlaybackThread::mLock held 9172size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9173{ 9174 Mutex::Autolock _l(mLock); 9175 size_t size = mEffects.size(); 9176 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9177 9178 for (size_t i = 0; i < size; i++) { 9179 if (effect == mEffects[i]) { 9180 // calling stop here will remove pre-processing effect from the audio HAL. 9181 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9182 // the middle of a read from audio HAL 9183 if (mEffects[i]->state() == EffectModule::ACTIVE || 9184 mEffects[i]->state() == EffectModule::STOPPING) { 9185 mEffects[i]->stop(); 9186 } 9187 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9188 delete[] effect->inBuffer(); 9189 } else { 9190 if (i == size - 1 && i != 0) { 9191 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9192 mEffects[i - 1]->configure(); 9193 } 9194 } 9195 mEffects.removeAt(i); 9196 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9197 break; 9198 } 9199 } 9200 9201 return mEffects.size(); 9202} 9203 9204// setDevice_l() must be called with PlaybackThread::mLock held 9205void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9206{ 9207 size_t size = mEffects.size(); 9208 for (size_t i = 0; i < size; i++) { 9209 mEffects[i]->setDevice(device); 9210 } 9211} 9212 9213// setMode_l() must be called with PlaybackThread::mLock held 9214void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9215{ 9216 size_t size = mEffects.size(); 9217 for (size_t i = 0; i < size; i++) { 9218 mEffects[i]->setMode(mode); 9219 } 9220} 9221 9222// setVolume_l() must be called with PlaybackThread::mLock held 9223bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9224{ 9225 uint32_t newLeft = *left; 9226 uint32_t newRight = *right; 9227 bool hasControl = false; 9228 int ctrlIdx = -1; 9229 size_t size = mEffects.size(); 9230 9231 // first update volume controller 9232 for (size_t i = size; i > 0; i--) { 9233 if (mEffects[i - 1]->isProcessEnabled() && 9234 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9235 ctrlIdx = i - 1; 9236 hasControl = true; 9237 break; 9238 } 9239 } 9240 9241 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9242 if (hasControl) { 9243 *left = mNewLeftVolume; 9244 *right = mNewRightVolume; 9245 } 9246 return hasControl; 9247 } 9248 9249 mVolumeCtrlIdx = ctrlIdx; 9250 mLeftVolume = newLeft; 9251 mRightVolume = newRight; 9252 9253 // second get volume update from volume controller 9254 if (ctrlIdx >= 0) { 9255 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9256 mNewLeftVolume = newLeft; 9257 mNewRightVolume = newRight; 9258 } 9259 // then indicate volume to all other effects in chain. 9260 // Pass altered volume to effects before volume controller 9261 // and requested volume to effects after controller 9262 uint32_t lVol = newLeft; 9263 uint32_t rVol = newRight; 9264 9265 for (size_t i = 0; i < size; i++) { 9266 if ((int)i == ctrlIdx) continue; 9267 // this also works for ctrlIdx == -1 when there is no volume controller 9268 if ((int)i > ctrlIdx) { 9269 lVol = *left; 9270 rVol = *right; 9271 } 9272 mEffects[i]->setVolume(&lVol, &rVol, false); 9273 } 9274 *left = newLeft; 9275 *right = newRight; 9276 9277 return hasControl; 9278} 9279 9280status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9281{ 9282 const size_t SIZE = 256; 9283 char buffer[SIZE]; 9284 String8 result; 9285 9286 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9287 result.append(buffer); 9288 9289 bool locked = tryLock(mLock); 9290 // failed to lock - AudioFlinger is probably deadlocked 9291 if (!locked) { 9292 result.append("\tCould not lock mutex:\n"); 9293 } 9294 9295 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9296 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9297 mEffects.size(), 9298 (uint32_t)mInBuffer, 9299 (uint32_t)mOutBuffer, 9300 mActiveTrackCnt); 9301 result.append(buffer); 9302 write(fd, result.string(), result.size()); 9303 9304 for (size_t i = 0; i < mEffects.size(); ++i) { 9305 sp<EffectModule> effect = mEffects[i]; 9306 if (effect != 0) { 9307 effect->dump(fd, args); 9308 } 9309 } 9310 9311 if (locked) { 9312 mLock.unlock(); 9313 } 9314 9315 return NO_ERROR; 9316} 9317 9318// must be called with ThreadBase::mLock held 9319void AudioFlinger::EffectChain::setEffectSuspended_l( 9320 const effect_uuid_t *type, bool suspend) 9321{ 9322 sp<SuspendedEffectDesc> desc; 9323 // use effect type UUID timelow as key as there is no real risk of identical 9324 // timeLow fields among effect type UUIDs. 9325 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9326 if (suspend) { 9327 if (index >= 0) { 9328 desc = mSuspendedEffects.valueAt(index); 9329 } else { 9330 desc = new SuspendedEffectDesc(); 9331 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9332 mSuspendedEffects.add(type->timeLow, desc); 9333 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9334 } 9335 if (desc->mRefCount++ == 0) { 9336 sp<EffectModule> effect = getEffectIfEnabled(type); 9337 if (effect != 0) { 9338 desc->mEffect = effect; 9339 effect->setSuspended(true); 9340 effect->setEnabled(false); 9341 } 9342 } 9343 } else { 9344 if (index < 0) { 9345 return; 9346 } 9347 desc = mSuspendedEffects.valueAt(index); 9348 if (desc->mRefCount <= 0) { 9349 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9350 desc->mRefCount = 1; 9351 } 9352 if (--desc->mRefCount == 0) { 9353 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9354 if (desc->mEffect != 0) { 9355 sp<EffectModule> effect = desc->mEffect.promote(); 9356 if (effect != 0) { 9357 effect->setSuspended(false); 9358 sp<EffectHandle> handle = effect->controlHandle(); 9359 if (handle != 0) { 9360 effect->setEnabled(handle->enabled()); 9361 } 9362 } 9363 desc->mEffect.clear(); 9364 } 9365 mSuspendedEffects.removeItemsAt(index); 9366 } 9367 } 9368} 9369 9370// must be called with ThreadBase::mLock held 9371void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9372{ 9373 sp<SuspendedEffectDesc> desc; 9374 9375 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9376 if (suspend) { 9377 if (index >= 0) { 9378 desc = mSuspendedEffects.valueAt(index); 9379 } else { 9380 desc = new SuspendedEffectDesc(); 9381 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9382 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9383 } 9384 if (desc->mRefCount++ == 0) { 9385 Vector< sp<EffectModule> > effects; 9386 getSuspendEligibleEffects(effects); 9387 for (size_t i = 0; i < effects.size(); i++) { 9388 setEffectSuspended_l(&effects[i]->desc().type, true); 9389 } 9390 } 9391 } else { 9392 if (index < 0) { 9393 return; 9394 } 9395 desc = mSuspendedEffects.valueAt(index); 9396 if (desc->mRefCount <= 0) { 9397 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9398 desc->mRefCount = 1; 9399 } 9400 if (--desc->mRefCount == 0) { 9401 Vector<const effect_uuid_t *> types; 9402 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9403 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9404 continue; 9405 } 9406 types.add(&mSuspendedEffects.valueAt(i)->mType); 9407 } 9408 for (size_t i = 0; i < types.size(); i++) { 9409 setEffectSuspended_l(types[i], false); 9410 } 9411 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9412 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9413 } 9414 } 9415} 9416 9417 9418// The volume effect is used for automated tests only 9419#ifndef OPENSL_ES_H_ 9420static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9421 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9422const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9423#endif //OPENSL_ES_H_ 9424 9425bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9426{ 9427 // auxiliary effects and visualizer are never suspended on output mix 9428 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9429 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9430 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9431 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9432 return false; 9433 } 9434 return true; 9435} 9436 9437void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9438{ 9439 effects.clear(); 9440 for (size_t i = 0; i < mEffects.size(); i++) { 9441 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9442 effects.add(mEffects[i]); 9443 } 9444 } 9445} 9446 9447sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9448 const effect_uuid_t *type) 9449{ 9450 sp<EffectModule> effect = getEffectFromType_l(type); 9451 return effect != 0 && effect->isEnabled() ? effect : 0; 9452} 9453 9454void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9455 bool enabled) 9456{ 9457 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9458 if (enabled) { 9459 if (index < 0) { 9460 // if the effect is not suspend check if all effects are suspended 9461 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9462 if (index < 0) { 9463 return; 9464 } 9465 if (!isEffectEligibleForSuspend(effect->desc())) { 9466 return; 9467 } 9468 setEffectSuspended_l(&effect->desc().type, enabled); 9469 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9470 if (index < 0) { 9471 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9472 return; 9473 } 9474 } 9475 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9476 effect->desc().type.timeLow); 9477 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9478 // if effect is requested to suspended but was not yet enabled, supend it now. 9479 if (desc->mEffect == 0) { 9480 desc->mEffect = effect; 9481 effect->setEnabled(false); 9482 effect->setSuspended(true); 9483 } 9484 } else { 9485 if (index < 0) { 9486 return; 9487 } 9488 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9489 effect->desc().type.timeLow); 9490 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9491 desc->mEffect.clear(); 9492 effect->setSuspended(false); 9493 } 9494} 9495 9496#undef LOG_TAG 9497#define LOG_TAG "AudioFlinger" 9498 9499// ---------------------------------------------------------------------------- 9500 9501status_t AudioFlinger::onTransact( 9502 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9503{ 9504 return BnAudioFlinger::onTransact(code, data, reply, flags); 9505} 9506 9507}; // namespace android 9508