AudioFlinger.cpp revision dd8104cc5367262f0e5f13df4e79f131e8d560bb
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 168 // AudioFlinger::setParameters() updates, other threads read w/o lock 169 170// ---------------------------------------------------------------------------- 171 172#ifdef ADD_BATTERY_DATA 173// To collect the amplifier usage 174static void addBatteryData(uint32_t params) { 175 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 176 if (service == NULL) { 177 // it already logged 178 return; 179 } 180 181 service->addBatteryData(params); 182} 183#endif 184 185static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 186{ 187 const hw_module_t *mod; 188 int rc; 189 190 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 191 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 rc = audio_hw_device_open(mod, dev); 197 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 198 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 199 if (rc) { 200 goto out; 201 } 202 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 203 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 204 rc = BAD_VALUE; 205 goto out; 206 } 207 return 0; 208 209out: 210 *dev = NULL; 211 return rc; 212} 213 214// ---------------------------------------------------------------------------- 215 216AudioFlinger::AudioFlinger() 217 : BnAudioFlinger(), 218 mPrimaryHardwareDev(NULL), 219 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 220 mMasterVolume(1.0f), 221 mMasterVolumeSupportLvl(MVS_NONE), 222 mMasterMute(false), 223 mNextUniqueId(1), 224 mMode(AUDIO_MODE_INVALID), 225 mBtNrecIsOff(false) 226{ 227} 228 229void AudioFlinger::onFirstRef() 230{ 231 int rc = 0; 232 233 Mutex::Autolock _l(mLock); 234 235 /* TODO: move all this work into an Init() function */ 236 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 237 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 238 uint32_t int_val; 239 if (1 == sscanf(val_str, "%u", &int_val)) { 240 mStandbyTimeInNsecs = milliseconds(int_val); 241 ALOGI("Using %u mSec as standby time.", int_val); 242 } else { 243 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 244 ALOGI("Using default %u mSec as standby time.", 245 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 246 } 247 } 248 249 mMode = AUDIO_MODE_NORMAL; 250 mMasterVolumeSW = 1.0; 251 mMasterVolume = 1.0; 252 mHardwareStatus = AUDIO_HW_IDLE; 253} 254 255AudioFlinger::~AudioFlinger() 256{ 257 258 while (!mRecordThreads.isEmpty()) { 259 // closeInput() will remove first entry from mRecordThreads 260 closeInput(mRecordThreads.keyAt(0)); 261 } 262 while (!mPlaybackThreads.isEmpty()) { 263 // closeOutput() will remove first entry from mPlaybackThreads 264 closeOutput(mPlaybackThreads.keyAt(0)); 265 } 266 267 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 268 // no mHardwareLock needed, as there are no other references to this 269 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 270 delete mAudioHwDevs.valueAt(i); 271 } 272} 273 274static const char * const audio_interfaces[] = { 275 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 276 AUDIO_HARDWARE_MODULE_ID_A2DP, 277 AUDIO_HARDWARE_MODULE_ID_USB, 278}; 279#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 280 281audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 282{ 283 // if module is 0, the request comes from an old policy manager and we should load 284 // well known modules 285 if (module == 0) { 286 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 287 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 288 loadHwModule_l(audio_interfaces[i]); 289 } 290 } else { 291 // check a match for the requested module handle 292 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 293 if (audioHwdevice != NULL) { 294 return audioHwdevice->hwDevice(); 295 } 296 } 297 // then try to find a module supporting the requested device. 298 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 299 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 300 if ((dev->get_supported_devices(dev) & devices) == devices) 301 return dev; 302 } 303 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 IAudioFlinger::track_flags_t flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 pid_t tid, 449 int *sessionId, 450 status_t *status) 451{ 452 sp<PlaybackThread::Track> track; 453 sp<TrackHandle> trackHandle; 454 sp<Client> client; 455 status_t lStatus; 456 int lSessionId; 457 458 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 459 // but if someone uses binder directly they could bypass that and cause us to crash 460 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 461 ALOGE("createTrack() invalid stream type %d", streamType); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 466 { 467 Mutex::Autolock _l(mLock); 468 PlaybackThread *thread = checkPlaybackThread_l(output); 469 PlaybackThread *effectThread = NULL; 470 if (thread == NULL) { 471 ALOGE("unknown output thread"); 472 lStatus = BAD_VALUE; 473 goto Exit; 474 } 475 476 client = registerPid_l(pid); 477 478 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 479 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 480 // check if an effect chain with the same session ID is present on another 481 // output thread and move it here. 482 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 483 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 484 if (mPlaybackThreads.keyAt(i) != output) { 485 uint32_t sessions = t->hasAudioSession(*sessionId); 486 if (sessions & PlaybackThread::EFFECT_SESSION) { 487 effectThread = t.get(); 488 break; 489 } 490 } 491 } 492 lSessionId = *sessionId; 493 } else { 494 // if no audio session id is provided, create one here 495 lSessionId = nextUniqueId(); 496 if (sessionId != NULL) { 497 *sessionId = lSessionId; 498 } 499 } 500 ALOGV("createTrack() lSessionId: %d", lSessionId); 501 502 track = thread->createTrack_l(client, streamType, sampleRate, format, 503 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 504 505 // move effect chain to this output thread if an effect on same session was waiting 506 // for a track to be created 507 if (lStatus == NO_ERROR && effectThread != NULL) { 508 Mutex::Autolock _dl(thread->mLock); 509 Mutex::Autolock _sl(effectThread->mLock); 510 moveEffectChain_l(lSessionId, effectThread, thread, true); 511 } 512 513 // Look for sync events awaiting for a session to be used. 514 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 515 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 516 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 517 if (lStatus == NO_ERROR) { 518 track->setSyncEvent(mPendingSyncEvents[i]); 519 } else { 520 mPendingSyncEvents[i]->cancel(); 521 } 522 mPendingSyncEvents.removeAt(i); 523 i--; 524 } 525 } 526 } 527 } 528 if (lStatus == NO_ERROR) { 529 trackHandle = new TrackHandle(track); 530 } else { 531 // remove local strong reference to Client before deleting the Track so that the Client 532 // destructor is called by the TrackBase destructor with mLock held 533 client.clear(); 534 track.clear(); 535 } 536 537Exit: 538 if (status != NULL) { 539 *status = lStatus; 540 } 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 586 // should examine all callers and fix them to handle smaller counts 587 return thread->frameCount(); 588} 589 590uint32_t AudioFlinger::latency(audio_io_handle_t output) const 591{ 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGW("latency() unknown thread %d", output); 596 return 0; 597 } 598 return thread->latency(); 599} 600 601status_t AudioFlinger::setMasterVolume(float value) 602{ 603 status_t ret = initCheck(); 604 if (ret != NO_ERROR) { 605 return ret; 606 } 607 608 // check calling permissions 609 if (!settingsAllowed()) { 610 return PERMISSION_DENIED; 611 } 612 613 float swmv = value; 614 615 Mutex::Autolock _l(mLock); 616 617 // when hw supports master volume, don't scale in sw mixer 618 if (MVS_NONE != mMasterVolumeSupportLvl) { 619 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 620 AutoMutex lock(mHardwareLock); 621 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 622 623 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 624 if (NULL != dev->set_master_volume) { 625 dev->set_master_volume(dev, value); 626 } 627 mHardwareStatus = AUDIO_HW_IDLE; 628 } 629 630 swmv = 1.0; 631 } 632 633 mMasterVolume = value; 634 mMasterVolumeSW = swmv; 635 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 636 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 637 638 return NO_ERROR; 639} 640 641status_t AudioFlinger::setMode(audio_mode_t mode) 642{ 643 status_t ret = initCheck(); 644 if (ret != NO_ERROR) { 645 return ret; 646 } 647 648 // check calling permissions 649 if (!settingsAllowed()) { 650 return PERMISSION_DENIED; 651 } 652 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 653 ALOGW("Illegal value: setMode(%d)", mode); 654 return BAD_VALUE; 655 } 656 657 { // scope for the lock 658 AutoMutex lock(mHardwareLock); 659 mHardwareStatus = AUDIO_HW_SET_MODE; 660 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 661 mHardwareStatus = AUDIO_HW_IDLE; 662 } 663 664 if (NO_ERROR == ret) { 665 Mutex::Autolock _l(mLock); 666 mMode = mode; 667 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 668 mPlaybackThreads.valueAt(i)->setMode(mode); 669 } 670 671 return ret; 672} 673 674status_t AudioFlinger::setMicMute(bool state) 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return ret; 679 } 680 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 AutoMutex lock(mHardwareLock); 687 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 688 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 689 mHardwareStatus = AUDIO_HW_IDLE; 690 return ret; 691} 692 693bool AudioFlinger::getMicMute() const 694{ 695 status_t ret = initCheck(); 696 if (ret != NO_ERROR) { 697 return false; 698 } 699 700 bool state = AUDIO_MODE_INVALID; 701 AutoMutex lock(mHardwareLock); 702 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 703 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 704 mHardwareStatus = AUDIO_HW_IDLE; 705 return state; 706} 707 708status_t AudioFlinger::setMasterMute(bool muted) 709{ 710 // check calling permissions 711 if (!settingsAllowed()) { 712 return PERMISSION_DENIED; 713 } 714 715 Mutex::Autolock _l(mLock); 716 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 717 mMasterMute = muted; 718 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 719 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 720 721 return NO_ERROR; 722} 723 724float AudioFlinger::masterVolume() const 725{ 726 Mutex::Autolock _l(mLock); 727 return masterVolume_l(); 728} 729 730float AudioFlinger::masterVolumeSW() const 731{ 732 Mutex::Autolock _l(mLock); 733 return masterVolumeSW_l(); 734} 735 736bool AudioFlinger::masterMute() const 737{ 738 Mutex::Autolock _l(mLock); 739 return masterMute_l(); 740} 741 742float AudioFlinger::masterVolume_l() const 743{ 744 if (MVS_FULL == mMasterVolumeSupportLvl) { 745 float ret_val; 746 AutoMutex lock(mHardwareLock); 747 748 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 749 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 750 (NULL != mPrimaryHardwareDev->get_master_volume), 751 "can't get master volume"); 752 753 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 754 mHardwareStatus = AUDIO_HW_IDLE; 755 return ret_val; 756 } 757 758 return mMasterVolume; 759} 760 761status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 762 audio_io_handle_t output) 763{ 764 // check calling permissions 765 if (!settingsAllowed()) { 766 return PERMISSION_DENIED; 767 } 768 769 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 770 ALOGE("setStreamVolume() invalid stream %d", stream); 771 return BAD_VALUE; 772 } 773 774 AutoMutex lock(mLock); 775 PlaybackThread *thread = NULL; 776 if (output) { 777 thread = checkPlaybackThread_l(output); 778 if (thread == NULL) { 779 return BAD_VALUE; 780 } 781 } 782 783 mStreamTypes[stream].volume = value; 784 785 if (thread == NULL) { 786 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 787 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 788 } 789 } else { 790 thread->setStreamVolume(stream, value); 791 } 792 793 return NO_ERROR; 794} 795 796status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 797{ 798 // check calling permissions 799 if (!settingsAllowed()) { 800 return PERMISSION_DENIED; 801 } 802 803 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 804 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 805 ALOGE("setStreamMute() invalid stream %d", stream); 806 return BAD_VALUE; 807 } 808 809 AutoMutex lock(mLock); 810 mStreamTypes[stream].mute = muted; 811 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 812 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 813 814 return NO_ERROR; 815} 816 817float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 818{ 819 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 820 return 0.0f; 821 } 822 823 AutoMutex lock(mLock); 824 float volume; 825 if (output) { 826 PlaybackThread *thread = checkPlaybackThread_l(output); 827 if (thread == NULL) { 828 return 0.0f; 829 } 830 volume = thread->streamVolume(stream); 831 } else { 832 volume = streamVolume_l(stream); 833 } 834 835 return volume; 836} 837 838bool AudioFlinger::streamMute(audio_stream_type_t stream) const 839{ 840 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 841 return true; 842 } 843 844 AutoMutex lock(mLock); 845 return streamMute_l(stream); 846} 847 848status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 849{ 850 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 851 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 852 // check calling permissions 853 if (!settingsAllowed()) { 854 return PERMISSION_DENIED; 855 } 856 857 // ioHandle == 0 means the parameters are global to the audio hardware interface 858 if (ioHandle == 0) { 859 Mutex::Autolock _l(mLock); 860 status_t final_result = NO_ERROR; 861 { 862 AutoMutex lock(mHardwareLock); 863 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 864 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 865 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 866 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 867 final_result = result ?: final_result; 868 } 869 mHardwareStatus = AUDIO_HW_IDLE; 870 } 871 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 872 AudioParameter param = AudioParameter(keyValuePairs); 873 String8 value; 874 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 875 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 876 if (mBtNrecIsOff != btNrecIsOff) { 877 for (size_t i = 0; i < mRecordThreads.size(); i++) { 878 sp<RecordThread> thread = mRecordThreads.valueAt(i); 879 RecordThread::RecordTrack *track = thread->track(); 880 if (track != NULL) { 881 audio_devices_t device = (audio_devices_t)( 882 thread->device() & AUDIO_DEVICE_IN_ALL); 883 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 884 thread->setEffectSuspended(FX_IID_AEC, 885 suspend, 886 track->sessionId()); 887 thread->setEffectSuspended(FX_IID_NS, 888 suspend, 889 track->sessionId()); 890 } 891 } 892 mBtNrecIsOff = btNrecIsOff; 893 } 894 } 895 String8 screenState; 896 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 897 bool isOff = screenState == "off"; 898 if (isOff != (gScreenState & 1)) { 899 gScreenState = ((gScreenState & ~1) + 2) | isOff; 900 } 901 } 902 return final_result; 903 } 904 905 // hold a strong ref on thread in case closeOutput() or closeInput() is called 906 // and the thread is exited once the lock is released 907 sp<ThreadBase> thread; 908 { 909 Mutex::Autolock _l(mLock); 910 thread = checkPlaybackThread_l(ioHandle); 911 if (thread == 0) { 912 thread = checkRecordThread_l(ioHandle); 913 } else if (thread == primaryPlaybackThread_l()) { 914 // indicate output device change to all input threads for pre processing 915 AudioParameter param = AudioParameter(keyValuePairs); 916 int value; 917 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 918 (value != 0)) { 919 for (size_t i = 0; i < mRecordThreads.size(); i++) { 920 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 921 } 922 } 923 } 924 } 925 if (thread != 0) { 926 return thread->setParameters(keyValuePairs); 927 } 928 return BAD_VALUE; 929} 930 931String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 932{ 933// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 934// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 935 936 Mutex::Autolock _l(mLock); 937 938 if (ioHandle == 0) { 939 String8 out_s8; 940 941 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 942 char *s; 943 { 944 AutoMutex lock(mHardwareLock); 945 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 946 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 947 s = dev->get_parameters(dev, keys.string()); 948 mHardwareStatus = AUDIO_HW_IDLE; 949 } 950 out_s8 += String8(s ? s : ""); 951 free(s); 952 } 953 return out_s8; 954 } 955 956 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 957 if (playbackThread != NULL) { 958 return playbackThread->getParameters(keys); 959 } 960 RecordThread *recordThread = checkRecordThread_l(ioHandle); 961 if (recordThread != NULL) { 962 return recordThread->getParameters(keys); 963 } 964 return String8(""); 965} 966 967size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 968 audio_channel_mask_t channelMask) const 969{ 970 status_t ret = initCheck(); 971 if (ret != NO_ERROR) { 972 return 0; 973 } 974 975 AutoMutex lock(mHardwareLock); 976 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 977 struct audio_config config = { 978 sample_rate: sampleRate, 979 channel_mask: channelMask, 980 format: format, 981 }; 982 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 983 mHardwareStatus = AUDIO_HW_IDLE; 984 return size; 985} 986 987unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 988{ 989 if (ioHandle == 0) { 990 return 0; 991 } 992 993 Mutex::Autolock _l(mLock); 994 995 RecordThread *recordThread = checkRecordThread_l(ioHandle); 996 if (recordThread != NULL) { 997 return recordThread->getInputFramesLost(); 998 } 999 return 0; 1000} 1001 1002status_t AudioFlinger::setVoiceVolume(float value) 1003{ 1004 status_t ret = initCheck(); 1005 if (ret != NO_ERROR) { 1006 return ret; 1007 } 1008 1009 // check calling permissions 1010 if (!settingsAllowed()) { 1011 return PERMISSION_DENIED; 1012 } 1013 1014 AutoMutex lock(mHardwareLock); 1015 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1016 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1017 mHardwareStatus = AUDIO_HW_IDLE; 1018 1019 return ret; 1020} 1021 1022status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1023 audio_io_handle_t output) const 1024{ 1025 status_t status; 1026 1027 Mutex::Autolock _l(mLock); 1028 1029 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1030 if (playbackThread != NULL) { 1031 return playbackThread->getRenderPosition(halFrames, dspFrames); 1032 } 1033 1034 return BAD_VALUE; 1035} 1036 1037void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1038{ 1039 1040 Mutex::Autolock _l(mLock); 1041 1042 pid_t pid = IPCThreadState::self()->getCallingPid(); 1043 if (mNotificationClients.indexOfKey(pid) < 0) { 1044 sp<NotificationClient> notificationClient = new NotificationClient(this, 1045 client, 1046 pid); 1047 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1048 1049 mNotificationClients.add(pid, notificationClient); 1050 1051 sp<IBinder> binder = client->asBinder(); 1052 binder->linkToDeath(notificationClient); 1053 1054 // the config change is always sent from playback or record threads to avoid deadlock 1055 // with AudioSystem::gLock 1056 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1057 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1058 } 1059 1060 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1061 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1062 } 1063 } 1064} 1065 1066void AudioFlinger::removeNotificationClient(pid_t pid) 1067{ 1068 Mutex::Autolock _l(mLock); 1069 1070 mNotificationClients.removeItem(pid); 1071 1072 ALOGV("%d died, releasing its sessions", pid); 1073 size_t num = mAudioSessionRefs.size(); 1074 bool removed = false; 1075 for (size_t i = 0; i< num; ) { 1076 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1077 ALOGV(" pid %d @ %d", ref->mPid, i); 1078 if (ref->mPid == pid) { 1079 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1080 mAudioSessionRefs.removeAt(i); 1081 delete ref; 1082 removed = true; 1083 num--; 1084 } else { 1085 i++; 1086 } 1087 } 1088 if (removed) { 1089 purgeStaleEffects_l(); 1090 } 1091} 1092 1093// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1094void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1095{ 1096 size_t size = mNotificationClients.size(); 1097 for (size_t i = 0; i < size; i++) { 1098 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1099 param2); 1100 } 1101} 1102 1103// removeClient_l() must be called with AudioFlinger::mLock held 1104void AudioFlinger::removeClient_l(pid_t pid) 1105{ 1106 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1107 mClients.removeItem(pid); 1108} 1109 1110// getEffectThread_l() must be called with AudioFlinger::mLock held 1111sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1112{ 1113 sp<PlaybackThread> thread; 1114 1115 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1116 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1117 ALOG_ASSERT(thread == 0); 1118 thread = mPlaybackThreads.valueAt(i); 1119 } 1120 } 1121 1122 return thread; 1123} 1124 1125// ---------------------------------------------------------------------------- 1126 1127AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1128 uint32_t device, type_t type) 1129 : Thread(false), 1130 mType(type), 1131 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1132 // mChannelMask 1133 mChannelCount(0), 1134 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1135 mParamStatus(NO_ERROR), 1136 mStandby(false), mId(id), 1137 mDevice(device), 1138 mDeathRecipient(new PMDeathRecipient(this)) 1139{ 1140} 1141 1142AudioFlinger::ThreadBase::~ThreadBase() 1143{ 1144 mParamCond.broadcast(); 1145 // do not lock the mutex in destructor 1146 releaseWakeLock_l(); 1147 if (mPowerManager != 0) { 1148 sp<IBinder> binder = mPowerManager->asBinder(); 1149 binder->unlinkToDeath(mDeathRecipient); 1150 } 1151} 1152 1153void AudioFlinger::ThreadBase::exit() 1154{ 1155 ALOGV("ThreadBase::exit"); 1156 { 1157 // This lock prevents the following race in thread (uniprocessor for illustration): 1158 // if (!exitPending()) { 1159 // // context switch from here to exit() 1160 // // exit() calls requestExit(), what exitPending() observes 1161 // // exit() calls signal(), which is dropped since no waiters 1162 // // context switch back from exit() to here 1163 // mWaitWorkCV.wait(...); 1164 // // now thread is hung 1165 // } 1166 AutoMutex lock(mLock); 1167 requestExit(); 1168 mWaitWorkCV.signal(); 1169 } 1170 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1171 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1172 requestExitAndWait(); 1173} 1174 1175status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1176{ 1177 status_t status; 1178 1179 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1180 Mutex::Autolock _l(mLock); 1181 1182 mNewParameters.add(keyValuePairs); 1183 mWaitWorkCV.signal(); 1184 // wait condition with timeout in case the thread loop has exited 1185 // before the request could be processed 1186 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1187 status = mParamStatus; 1188 mWaitWorkCV.signal(); 1189 } else { 1190 status = TIMED_OUT; 1191 } 1192 return status; 1193} 1194 1195void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1196{ 1197 Mutex::Autolock _l(mLock); 1198 sendConfigEvent_l(event, param); 1199} 1200 1201// sendConfigEvent_l() must be called with ThreadBase::mLock held 1202void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1203{ 1204 ConfigEvent configEvent; 1205 configEvent.mEvent = event; 1206 configEvent.mParam = param; 1207 mConfigEvents.add(configEvent); 1208 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1209 mWaitWorkCV.signal(); 1210} 1211 1212void AudioFlinger::ThreadBase::processConfigEvents() 1213{ 1214 mLock.lock(); 1215 while (!mConfigEvents.isEmpty()) { 1216 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1217 ConfigEvent configEvent = mConfigEvents[0]; 1218 mConfigEvents.removeAt(0); 1219 // release mLock before locking AudioFlinger mLock: lock order is always 1220 // AudioFlinger then ThreadBase to avoid cross deadlock 1221 mLock.unlock(); 1222 mAudioFlinger->mLock.lock(); 1223 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1224 mAudioFlinger->mLock.unlock(); 1225 mLock.lock(); 1226 } 1227 mLock.unlock(); 1228} 1229 1230status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1231{ 1232 const size_t SIZE = 256; 1233 char buffer[SIZE]; 1234 String8 result; 1235 1236 bool locked = tryLock(mLock); 1237 if (!locked) { 1238 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1239 write(fd, buffer, strlen(buffer)); 1240 } 1241 1242 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1245 result.append(buffer); 1246 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1247 result.append(buffer); 1248 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1249 result.append(buffer); 1250 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1251 result.append(buffer); 1252 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1253 result.append(buffer); 1254 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1255 result.append(buffer); 1256 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1257 result.append(buffer); 1258 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1259 result.append(buffer); 1260 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1261 result.append(buffer); 1262 1263 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1264 result.append(buffer); 1265 result.append(" Index Command"); 1266 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1267 snprintf(buffer, SIZE, "\n %02d ", i); 1268 result.append(buffer); 1269 result.append(mNewParameters[i]); 1270 } 1271 1272 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1273 result.append(buffer); 1274 snprintf(buffer, SIZE, " Index event param\n"); 1275 result.append(buffer); 1276 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1277 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1278 result.append(buffer); 1279 } 1280 result.append("\n"); 1281 1282 write(fd, result.string(), result.size()); 1283 1284 if (locked) { 1285 mLock.unlock(); 1286 } 1287 return NO_ERROR; 1288} 1289 1290status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1291{ 1292 const size_t SIZE = 256; 1293 char buffer[SIZE]; 1294 String8 result; 1295 1296 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1297 write(fd, buffer, strlen(buffer)); 1298 1299 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1300 sp<EffectChain> chain = mEffectChains[i]; 1301 if (chain != 0) { 1302 chain->dump(fd, args); 1303 } 1304 } 1305 return NO_ERROR; 1306} 1307 1308void AudioFlinger::ThreadBase::acquireWakeLock() 1309{ 1310 Mutex::Autolock _l(mLock); 1311 acquireWakeLock_l(); 1312} 1313 1314void AudioFlinger::ThreadBase::acquireWakeLock_l() 1315{ 1316 if (mPowerManager == 0) { 1317 // use checkService() to avoid blocking if power service is not up yet 1318 sp<IBinder> binder = 1319 defaultServiceManager()->checkService(String16("power")); 1320 if (binder == 0) { 1321 ALOGW("Thread %s cannot connect to the power manager service", mName); 1322 } else { 1323 mPowerManager = interface_cast<IPowerManager>(binder); 1324 binder->linkToDeath(mDeathRecipient); 1325 } 1326 } 1327 if (mPowerManager != 0) { 1328 sp<IBinder> binder = new BBinder(); 1329 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1330 binder, 1331 String16(mName)); 1332 if (status == NO_ERROR) { 1333 mWakeLockToken = binder; 1334 } 1335 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1336 } 1337} 1338 1339void AudioFlinger::ThreadBase::releaseWakeLock() 1340{ 1341 Mutex::Autolock _l(mLock); 1342 releaseWakeLock_l(); 1343} 1344 1345void AudioFlinger::ThreadBase::releaseWakeLock_l() 1346{ 1347 if (mWakeLockToken != 0) { 1348 ALOGV("releaseWakeLock_l() %s", mName); 1349 if (mPowerManager != 0) { 1350 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1351 } 1352 mWakeLockToken.clear(); 1353 } 1354} 1355 1356void AudioFlinger::ThreadBase::clearPowerManager() 1357{ 1358 Mutex::Autolock _l(mLock); 1359 releaseWakeLock_l(); 1360 mPowerManager.clear(); 1361} 1362 1363void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1364{ 1365 sp<ThreadBase> thread = mThread.promote(); 1366 if (thread != 0) { 1367 thread->clearPowerManager(); 1368 } 1369 ALOGW("power manager service died !!!"); 1370} 1371 1372void AudioFlinger::ThreadBase::setEffectSuspended( 1373 const effect_uuid_t *type, bool suspend, int sessionId) 1374{ 1375 Mutex::Autolock _l(mLock); 1376 setEffectSuspended_l(type, suspend, sessionId); 1377} 1378 1379void AudioFlinger::ThreadBase::setEffectSuspended_l( 1380 const effect_uuid_t *type, bool suspend, int sessionId) 1381{ 1382 sp<EffectChain> chain = getEffectChain_l(sessionId); 1383 if (chain != 0) { 1384 if (type != NULL) { 1385 chain->setEffectSuspended_l(type, suspend); 1386 } else { 1387 chain->setEffectSuspendedAll_l(suspend); 1388 } 1389 } 1390 1391 updateSuspendedSessions_l(type, suspend, sessionId); 1392} 1393 1394void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1395{ 1396 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1397 if (index < 0) { 1398 return; 1399 } 1400 1401 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1402 mSuspendedSessions.editValueAt(index); 1403 1404 for (size_t i = 0; i < sessionEffects.size(); i++) { 1405 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1406 for (int j = 0; j < desc->mRefCount; j++) { 1407 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1408 chain->setEffectSuspendedAll_l(true); 1409 } else { 1410 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1411 desc->mType.timeLow); 1412 chain->setEffectSuspended_l(&desc->mType, true); 1413 } 1414 } 1415 } 1416} 1417 1418void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1419 bool suspend, 1420 int sessionId) 1421{ 1422 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1423 1424 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1425 1426 if (suspend) { 1427 if (index >= 0) { 1428 sessionEffects = mSuspendedSessions.editValueAt(index); 1429 } else { 1430 mSuspendedSessions.add(sessionId, sessionEffects); 1431 } 1432 } else { 1433 if (index < 0) { 1434 return; 1435 } 1436 sessionEffects = mSuspendedSessions.editValueAt(index); 1437 } 1438 1439 1440 int key = EffectChain::kKeyForSuspendAll; 1441 if (type != NULL) { 1442 key = type->timeLow; 1443 } 1444 index = sessionEffects.indexOfKey(key); 1445 1446 sp<SuspendedSessionDesc> desc; 1447 if (suspend) { 1448 if (index >= 0) { 1449 desc = sessionEffects.valueAt(index); 1450 } else { 1451 desc = new SuspendedSessionDesc(); 1452 if (type != NULL) { 1453 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1454 } 1455 sessionEffects.add(key, desc); 1456 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1457 } 1458 desc->mRefCount++; 1459 } else { 1460 if (index < 0) { 1461 return; 1462 } 1463 desc = sessionEffects.valueAt(index); 1464 if (--desc->mRefCount == 0) { 1465 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1466 sessionEffects.removeItemsAt(index); 1467 if (sessionEffects.isEmpty()) { 1468 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1469 sessionId); 1470 mSuspendedSessions.removeItem(sessionId); 1471 } 1472 } 1473 } 1474 if (!sessionEffects.isEmpty()) { 1475 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1476 } 1477} 1478 1479void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1480 bool enabled, 1481 int sessionId) 1482{ 1483 Mutex::Autolock _l(mLock); 1484 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1485} 1486 1487void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1488 bool enabled, 1489 int sessionId) 1490{ 1491 if (mType != RECORD) { 1492 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1493 // another session. This gives the priority to well behaved effect control panels 1494 // and applications not using global effects. 1495 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1496 // global effects 1497 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1498 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1499 } 1500 } 1501 1502 sp<EffectChain> chain = getEffectChain_l(sessionId); 1503 if (chain != 0) { 1504 chain->checkSuspendOnEffectEnabled(effect, enabled); 1505 } 1506} 1507 1508// ---------------------------------------------------------------------------- 1509 1510AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1511 AudioStreamOut* output, 1512 audio_io_handle_t id, 1513 uint32_t device, 1514 type_t type) 1515 : ThreadBase(audioFlinger, id, device, type), 1516 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1517 // Assumes constructor is called by AudioFlinger with it's mLock held, 1518 // but it would be safer to explicitly pass initial masterMute as parameter 1519 mMasterMute(audioFlinger->masterMute_l()), 1520 // mStreamTypes[] initialized in constructor body 1521 mOutput(output), 1522 // Assumes constructor is called by AudioFlinger with it's mLock held, 1523 // but it would be safer to explicitly pass initial masterVolume as parameter 1524 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1525 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1526 mMixerStatus(MIXER_IDLE), 1527 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1528 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1529 mScreenState(gScreenState), 1530 // index 0 is reserved for normal mixer's submix 1531 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1532{ 1533 snprintf(mName, kNameLength, "AudioOut_%X", id); 1534 1535 readOutputParameters(); 1536 1537 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1538 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1539 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1540 stream = (audio_stream_type_t) (stream + 1)) { 1541 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1542 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1543 } 1544 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1545 // because mAudioFlinger doesn't have one to copy from 1546} 1547 1548AudioFlinger::PlaybackThread::~PlaybackThread() 1549{ 1550 delete [] mMixBuffer; 1551} 1552 1553status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1554{ 1555 dumpInternals(fd, args); 1556 dumpTracks(fd, args); 1557 dumpEffectChains(fd, args); 1558 return NO_ERROR; 1559} 1560 1561status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1562{ 1563 const size_t SIZE = 256; 1564 char buffer[SIZE]; 1565 String8 result; 1566 1567 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1568 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1569 const stream_type_t *st = &mStreamTypes[i]; 1570 if (i > 0) { 1571 result.appendFormat(", "); 1572 } 1573 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1574 if (st->mute) { 1575 result.append("M"); 1576 } 1577 } 1578 result.append("\n"); 1579 write(fd, result.string(), result.length()); 1580 result.clear(); 1581 1582 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1583 result.append(buffer); 1584 Track::appendDumpHeader(result); 1585 for (size_t i = 0; i < mTracks.size(); ++i) { 1586 sp<Track> track = mTracks[i]; 1587 if (track != 0) { 1588 track->dump(buffer, SIZE); 1589 result.append(buffer); 1590 } 1591 } 1592 1593 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1594 result.append(buffer); 1595 Track::appendDumpHeader(result); 1596 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1597 sp<Track> track = mActiveTracks[i].promote(); 1598 if (track != 0) { 1599 track->dump(buffer, SIZE); 1600 result.append(buffer); 1601 } 1602 } 1603 write(fd, result.string(), result.size()); 1604 1605 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1606 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1607 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1608 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1609 1610 return NO_ERROR; 1611} 1612 1613status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1614{ 1615 const size_t SIZE = 256; 1616 char buffer[SIZE]; 1617 String8 result; 1618 1619 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1620 result.append(buffer); 1621 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1622 result.append(buffer); 1623 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1624 result.append(buffer); 1625 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1626 result.append(buffer); 1627 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1628 result.append(buffer); 1629 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1630 result.append(buffer); 1631 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1632 result.append(buffer); 1633 write(fd, result.string(), result.size()); 1634 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1635 1636 dumpBase(fd, args); 1637 1638 return NO_ERROR; 1639} 1640 1641// Thread virtuals 1642status_t AudioFlinger::PlaybackThread::readyToRun() 1643{ 1644 status_t status = initCheck(); 1645 if (status == NO_ERROR) { 1646 ALOGI("AudioFlinger's thread %p ready to run", this); 1647 } else { 1648 ALOGE("No working audio driver found."); 1649 } 1650 return status; 1651} 1652 1653void AudioFlinger::PlaybackThread::onFirstRef() 1654{ 1655 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1656} 1657 1658// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1659sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1660 const sp<AudioFlinger::Client>& client, 1661 audio_stream_type_t streamType, 1662 uint32_t sampleRate, 1663 audio_format_t format, 1664 uint32_t channelMask, 1665 int frameCount, 1666 const sp<IMemory>& sharedBuffer, 1667 int sessionId, 1668 IAudioFlinger::track_flags_t flags, 1669 pid_t tid, 1670 status_t *status) 1671{ 1672 sp<Track> track; 1673 status_t lStatus; 1674 1675 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1676 1677 // client expresses a preference for FAST, but we get the final say 1678 if (flags & IAudioFlinger::TRACK_FAST) { 1679 if ( 1680 // not timed 1681 (!isTimed) && 1682 // either of these use cases: 1683 ( 1684 // use case 1: shared buffer with any frame count 1685 ( 1686 (sharedBuffer != 0) 1687 ) || 1688 // use case 2: callback handler and frame count is default or at least as large as HAL 1689 ( 1690 (tid != -1) && 1691 ((frameCount == 0) || 1692 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1693 ) 1694 ) && 1695 // PCM data 1696 audio_is_linear_pcm(format) && 1697 // mono or stereo 1698 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1699 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1700#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1701 // hardware sample rate 1702 (sampleRate == mSampleRate) && 1703#endif 1704 // normal mixer has an associated fast mixer 1705 hasFastMixer() && 1706 // there are sufficient fast track slots available 1707 (mFastTrackAvailMask != 0) 1708 // FIXME test that MixerThread for this fast track has a capable output HAL 1709 // FIXME add a permission test also? 1710 ) { 1711 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1712 if (frameCount == 0) { 1713 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1714 } 1715 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1716 frameCount, mFrameCount); 1717 } else { 1718 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1719 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1720 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1721 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1722 audio_is_linear_pcm(format), 1723 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1724 flags &= ~IAudioFlinger::TRACK_FAST; 1725 // For compatibility with AudioTrack calculation, buffer depth is forced 1726 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1727 // This is probably too conservative, but legacy application code may depend on it. 1728 // If you change this calculation, also review the start threshold which is related. 1729 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1730 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1731 if (minBufCount < 2) { 1732 minBufCount = 2; 1733 } 1734 int minFrameCount = mNormalFrameCount * minBufCount; 1735 if (frameCount < minFrameCount) { 1736 frameCount = minFrameCount; 1737 } 1738 } 1739 } 1740 1741 if (mType == DIRECT) { 1742 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1743 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1744 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1745 "for output %p with format %d", 1746 sampleRate, format, channelMask, mOutput, mFormat); 1747 lStatus = BAD_VALUE; 1748 goto Exit; 1749 } 1750 } 1751 } else { 1752 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1753 if (sampleRate > mSampleRate*2) { 1754 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1755 lStatus = BAD_VALUE; 1756 goto Exit; 1757 } 1758 } 1759 1760 lStatus = initCheck(); 1761 if (lStatus != NO_ERROR) { 1762 ALOGE("Audio driver not initialized."); 1763 goto Exit; 1764 } 1765 1766 { // scope for mLock 1767 Mutex::Autolock _l(mLock); 1768 1769 // all tracks in same audio session must share the same routing strategy otherwise 1770 // conflicts will happen when tracks are moved from one output to another by audio policy 1771 // manager 1772 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1773 for (size_t i = 0; i < mTracks.size(); ++i) { 1774 sp<Track> t = mTracks[i]; 1775 if (t != 0 && !t->isOutputTrack()) { 1776 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1777 if (sessionId == t->sessionId() && strategy != actual) { 1778 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1779 strategy, actual); 1780 lStatus = BAD_VALUE; 1781 goto Exit; 1782 } 1783 } 1784 } 1785 1786 if (!isTimed) { 1787 track = new Track(this, client, streamType, sampleRate, format, 1788 channelMask, frameCount, sharedBuffer, sessionId, flags); 1789 } else { 1790 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1791 channelMask, frameCount, sharedBuffer, sessionId); 1792 } 1793 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1794 lStatus = NO_MEMORY; 1795 goto Exit; 1796 } 1797 mTracks.add(track); 1798 1799 sp<EffectChain> chain = getEffectChain_l(sessionId); 1800 if (chain != 0) { 1801 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1802 track->setMainBuffer(chain->inBuffer()); 1803 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1804 chain->incTrackCnt(); 1805 } 1806 } 1807 1808#ifdef HAVE_REQUEST_PRIORITY 1809 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1810 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1811 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1812 // so ask activity manager to do this on our behalf 1813 int err = requestPriority(callingPid, tid, 1); 1814 if (err != 0) { 1815 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1816 1, callingPid, tid, err); 1817 } 1818 } 1819#endif 1820 1821 lStatus = NO_ERROR; 1822 1823Exit: 1824 if (status) { 1825 *status = lStatus; 1826 } 1827 return track; 1828} 1829 1830uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1831{ 1832 if (mFastMixer != NULL) { 1833 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1834 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1835 } 1836 return latency; 1837} 1838 1839uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1840{ 1841 return latency; 1842} 1843 1844uint32_t AudioFlinger::PlaybackThread::latency() const 1845{ 1846 Mutex::Autolock _l(mLock); 1847 return latency_l(); 1848} 1849uint32_t AudioFlinger::PlaybackThread::latency_l() const 1850{ 1851 if (initCheck() == NO_ERROR) { 1852 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1853 } else { 1854 return 0; 1855 } 1856} 1857 1858void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1859{ 1860 Mutex::Autolock _l(mLock); 1861 mMasterVolume = value; 1862} 1863 1864void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1865{ 1866 Mutex::Autolock _l(mLock); 1867 setMasterMute_l(muted); 1868} 1869 1870void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1871{ 1872 Mutex::Autolock _l(mLock); 1873 mStreamTypes[stream].volume = value; 1874} 1875 1876void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1877{ 1878 Mutex::Autolock _l(mLock); 1879 mStreamTypes[stream].mute = muted; 1880} 1881 1882float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1883{ 1884 Mutex::Autolock _l(mLock); 1885 return mStreamTypes[stream].volume; 1886} 1887 1888// addTrack_l() must be called with ThreadBase::mLock held 1889status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1890{ 1891 status_t status = ALREADY_EXISTS; 1892 1893 // set retry count for buffer fill 1894 track->mRetryCount = kMaxTrackStartupRetries; 1895 if (mActiveTracks.indexOf(track) < 0) { 1896 // the track is newly added, make sure it fills up all its 1897 // buffers before playing. This is to ensure the client will 1898 // effectively get the latency it requested. 1899 track->mFillingUpStatus = Track::FS_FILLING; 1900 track->mResetDone = false; 1901 track->mPresentationCompleteFrames = 0; 1902 mActiveTracks.add(track); 1903 if (track->mainBuffer() != mMixBuffer) { 1904 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1905 if (chain != 0) { 1906 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1907 chain->incActiveTrackCnt(); 1908 } 1909 } 1910 1911 status = NO_ERROR; 1912 } 1913 1914 ALOGV("mWaitWorkCV.broadcast"); 1915 mWaitWorkCV.broadcast(); 1916 1917 return status; 1918} 1919 1920// destroyTrack_l() must be called with ThreadBase::mLock held 1921void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1922{ 1923 track->mState = TrackBase::TERMINATED; 1924 // active tracks are removed by threadLoop() 1925 if (mActiveTracks.indexOf(track) < 0) { 1926 removeTrack_l(track); 1927 } 1928} 1929 1930void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1931{ 1932 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1933 mTracks.remove(track); 1934 deleteTrackName_l(track->name()); 1935 // redundant as track is about to be destroyed, for dumpsys only 1936 track->mName = -1; 1937 if (track->isFastTrack()) { 1938 int index = track->mFastIndex; 1939 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1940 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1941 mFastTrackAvailMask |= 1 << index; 1942 // redundant as track is about to be destroyed, for dumpsys only 1943 track->mFastIndex = -1; 1944 } 1945 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1946 if (chain != 0) { 1947 chain->decTrackCnt(); 1948 } 1949} 1950 1951String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1952{ 1953 String8 out_s8 = String8(""); 1954 char *s; 1955 1956 Mutex::Autolock _l(mLock); 1957 if (initCheck() != NO_ERROR) { 1958 return out_s8; 1959 } 1960 1961 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1962 out_s8 = String8(s); 1963 free(s); 1964 return out_s8; 1965} 1966 1967// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1968void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1969 AudioSystem::OutputDescriptor desc; 1970 void *param2 = NULL; 1971 1972 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1973 1974 switch (event) { 1975 case AudioSystem::OUTPUT_OPENED: 1976 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1977 desc.channels = mChannelMask; 1978 desc.samplingRate = mSampleRate; 1979 desc.format = mFormat; 1980 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1981 desc.latency = latency(); 1982 param2 = &desc; 1983 break; 1984 1985 case AudioSystem::STREAM_CONFIG_CHANGED: 1986 param2 = ¶m; 1987 case AudioSystem::OUTPUT_CLOSED: 1988 default: 1989 break; 1990 } 1991 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1992} 1993 1994void AudioFlinger::PlaybackThread::readOutputParameters() 1995{ 1996 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1997 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1998 mChannelCount = (uint16_t)popcount(mChannelMask); 1999 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2000 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2001 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2002 if (mFrameCount & 15) { 2003 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2004 mFrameCount); 2005 } 2006 2007 // Calculate size of normal mix buffer relative to the HAL output buffer size 2008 double multiplier = 1.0; 2009 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2010 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2011 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2012 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2013 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2014 maxNormalFrameCount = maxNormalFrameCount & ~15; 2015 if (maxNormalFrameCount < minNormalFrameCount) { 2016 maxNormalFrameCount = minNormalFrameCount; 2017 } 2018 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2019 if (multiplier <= 1.0) { 2020 multiplier = 1.0; 2021 } else if (multiplier <= 2.0) { 2022 if (2 * mFrameCount <= maxNormalFrameCount) { 2023 multiplier = 2.0; 2024 } else { 2025 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2026 } 2027 } else { 2028 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2029 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2030 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2031 // FIXME this rounding up should not be done if no HAL SRC 2032 uint32_t truncMult = (uint32_t) multiplier; 2033 if ((truncMult & 1)) { 2034 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2035 ++truncMult; 2036 } 2037 } 2038 multiplier = (double) truncMult; 2039 } 2040 } 2041 mNormalFrameCount = multiplier * mFrameCount; 2042 // round up to nearest 16 frames to satisfy AudioMixer 2043 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2044 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2045 2046 delete[] mMixBuffer; 2047 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2048 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2049 2050 // force reconfiguration of effect chains and engines to take new buffer size and audio 2051 // parameters into account 2052 // Note that mLock is not held when readOutputParameters() is called from the constructor 2053 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2054 // matter. 2055 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2056 Vector< sp<EffectChain> > effectChains = mEffectChains; 2057 for (size_t i = 0; i < effectChains.size(); i ++) { 2058 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2059 } 2060} 2061 2062 2063status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2064{ 2065 if (halFrames == NULL || dspFrames == NULL) { 2066 return BAD_VALUE; 2067 } 2068 Mutex::Autolock _l(mLock); 2069 if (initCheck() != NO_ERROR) { 2070 return INVALID_OPERATION; 2071 } 2072 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2073 2074 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2075} 2076 2077uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2078{ 2079 Mutex::Autolock _l(mLock); 2080 uint32_t result = 0; 2081 if (getEffectChain_l(sessionId) != 0) { 2082 result = EFFECT_SESSION; 2083 } 2084 2085 for (size_t i = 0; i < mTracks.size(); ++i) { 2086 sp<Track> track = mTracks[i]; 2087 if (sessionId == track->sessionId() && 2088 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2089 result |= TRACK_SESSION; 2090 break; 2091 } 2092 } 2093 2094 return result; 2095} 2096 2097uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2098{ 2099 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2100 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2101 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2102 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2103 } 2104 for (size_t i = 0; i < mTracks.size(); i++) { 2105 sp<Track> track = mTracks[i]; 2106 if (sessionId == track->sessionId() && 2107 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2108 return AudioSystem::getStrategyForStream(track->streamType()); 2109 } 2110 } 2111 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2112} 2113 2114 2115AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2116{ 2117 Mutex::Autolock _l(mLock); 2118 return mOutput; 2119} 2120 2121AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2122{ 2123 Mutex::Autolock _l(mLock); 2124 AudioStreamOut *output = mOutput; 2125 mOutput = NULL; 2126 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2127 // must push a NULL and wait for ack 2128 mOutputSink.clear(); 2129 mPipeSink.clear(); 2130 mNormalSink.clear(); 2131 return output; 2132} 2133 2134// this method must always be called either with ThreadBase mLock held or inside the thread loop 2135audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2136{ 2137 if (mOutput == NULL) { 2138 return NULL; 2139 } 2140 return &mOutput->stream->common; 2141} 2142 2143uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2144{ 2145 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2146} 2147 2148status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2149{ 2150 if (!isValidSyncEvent(event)) { 2151 return BAD_VALUE; 2152 } 2153 2154 Mutex::Autolock _l(mLock); 2155 2156 for (size_t i = 0; i < mTracks.size(); ++i) { 2157 sp<Track> track = mTracks[i]; 2158 if (event->triggerSession() == track->sessionId()) { 2159 track->setSyncEvent(event); 2160 return NO_ERROR; 2161 } 2162 } 2163 2164 return NAME_NOT_FOUND; 2165} 2166 2167bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2168{ 2169 switch (event->type()) { 2170 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2171 return true; 2172 default: 2173 break; 2174 } 2175 return false; 2176} 2177 2178void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2179{ 2180 size_t count = tracksToRemove.size(); 2181 if (CC_UNLIKELY(count)) { 2182 for (size_t i = 0 ; i < count ; i++) { 2183 const sp<Track>& track = tracksToRemove.itemAt(i); 2184 if ((track->sharedBuffer() != 0) && 2185 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2186 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2187 } 2188 } 2189 } 2190 2191} 2192 2193// ---------------------------------------------------------------------------- 2194 2195AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2196 audio_io_handle_t id, uint32_t device, type_t type) 2197 : PlaybackThread(audioFlinger, output, id, device, type), 2198 // mAudioMixer below 2199#ifdef SOAKER 2200 mSoaker(NULL), 2201#endif 2202 // mFastMixer below 2203 mFastMixerFutex(0) 2204 // mOutputSink below 2205 // mPipeSink below 2206 // mNormalSink below 2207{ 2208 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2209 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2210 "mFrameCount=%d, mNormalFrameCount=%d", 2211 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2212 mNormalFrameCount); 2213 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2214 2215 // FIXME - Current mixer implementation only supports stereo output 2216 if (mChannelCount == 1) { 2217 ALOGE("Invalid audio hardware channel count"); 2218 } 2219 2220 // create an NBAIO sink for the HAL output stream, and negotiate 2221 mOutputSink = new AudioStreamOutSink(output->stream); 2222 size_t numCounterOffers = 0; 2223 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2224 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2225 ALOG_ASSERT(index == 0); 2226 2227 // initialize fast mixer depending on configuration 2228 bool initFastMixer; 2229 switch (kUseFastMixer) { 2230 case FastMixer_Never: 2231 initFastMixer = false; 2232 break; 2233 case FastMixer_Always: 2234 initFastMixer = true; 2235 break; 2236 case FastMixer_Static: 2237 case FastMixer_Dynamic: 2238 initFastMixer = mFrameCount < mNormalFrameCount; 2239 break; 2240 } 2241 if (initFastMixer) { 2242 2243 // create a MonoPipe to connect our submix to FastMixer 2244 NBAIO_Format format = mOutputSink->format(); 2245 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2246 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2247 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2248 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2249 const NBAIO_Format offers[1] = {format}; 2250 size_t numCounterOffers = 0; 2251 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2252 ALOG_ASSERT(index == 0); 2253 monoPipe->setAvgFrames((mScreenState & 1) ? 2254 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2255 mPipeSink = monoPipe; 2256 2257#ifdef TEE_SINK_FRAMES 2258 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2259 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2260 numCounterOffers = 0; 2261 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2262 ALOG_ASSERT(index == 0); 2263 mTeeSink = teeSink; 2264 PipeReader *teeSource = new PipeReader(*teeSink); 2265 numCounterOffers = 0; 2266 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2267 ALOG_ASSERT(index == 0); 2268 mTeeSource = teeSource; 2269#endif 2270 2271#ifdef SOAKER 2272 // create a soaker as workaround for governor issues 2273 mSoaker = new Soaker(); 2274 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2275 mSoaker->run("Soaker", PRIORITY_LOWEST); 2276#endif 2277 2278 // create fast mixer and configure it initially with just one fast track for our submix 2279 mFastMixer = new FastMixer(); 2280 FastMixerStateQueue *sq = mFastMixer->sq(); 2281#ifdef STATE_QUEUE_DUMP 2282 sq->setObserverDump(&mStateQueueObserverDump); 2283 sq->setMutatorDump(&mStateQueueMutatorDump); 2284#endif 2285 FastMixerState *state = sq->begin(); 2286 FastTrack *fastTrack = &state->mFastTracks[0]; 2287 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2288 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2289 fastTrack->mVolumeProvider = NULL; 2290 fastTrack->mGeneration++; 2291 state->mFastTracksGen++; 2292 state->mTrackMask = 1; 2293 // fast mixer will use the HAL output sink 2294 state->mOutputSink = mOutputSink.get(); 2295 state->mOutputSinkGen++; 2296 state->mFrameCount = mFrameCount; 2297 state->mCommand = FastMixerState::COLD_IDLE; 2298 // already done in constructor initialization list 2299 //mFastMixerFutex = 0; 2300 state->mColdFutexAddr = &mFastMixerFutex; 2301 state->mColdGen++; 2302 state->mDumpState = &mFastMixerDumpState; 2303 state->mTeeSink = mTeeSink.get(); 2304 sq->end(); 2305 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2306 2307 // start the fast mixer 2308 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2309#ifdef HAVE_REQUEST_PRIORITY 2310 pid_t tid = mFastMixer->getTid(); 2311 int err = requestPriority(getpid_cached, tid, 2); 2312 if (err != 0) { 2313 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2314 2, getpid_cached, tid, err); 2315 } 2316#endif 2317 2318#ifdef AUDIO_WATCHDOG 2319 // create and start the watchdog 2320 mAudioWatchdog = new AudioWatchdog(); 2321 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2322 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2323 tid = mAudioWatchdog->getTid(); 2324 err = requestPriority(getpid_cached, tid, 1); 2325 if (err != 0) { 2326 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2327 1, getpid_cached, tid, err); 2328 } 2329#endif 2330 2331 } else { 2332 mFastMixer = NULL; 2333 } 2334 2335 switch (kUseFastMixer) { 2336 case FastMixer_Never: 2337 case FastMixer_Dynamic: 2338 mNormalSink = mOutputSink; 2339 break; 2340 case FastMixer_Always: 2341 mNormalSink = mPipeSink; 2342 break; 2343 case FastMixer_Static: 2344 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2345 break; 2346 } 2347} 2348 2349AudioFlinger::MixerThread::~MixerThread() 2350{ 2351 if (mFastMixer != NULL) { 2352 FastMixerStateQueue *sq = mFastMixer->sq(); 2353 FastMixerState *state = sq->begin(); 2354 if (state->mCommand == FastMixerState::COLD_IDLE) { 2355 int32_t old = android_atomic_inc(&mFastMixerFutex); 2356 if (old == -1) { 2357 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2358 } 2359 } 2360 state->mCommand = FastMixerState::EXIT; 2361 sq->end(); 2362 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2363 mFastMixer->join(); 2364 // Though the fast mixer thread has exited, it's state queue is still valid. 2365 // We'll use that extract the final state which contains one remaining fast track 2366 // corresponding to our sub-mix. 2367 state = sq->begin(); 2368 ALOG_ASSERT(state->mTrackMask == 1); 2369 FastTrack *fastTrack = &state->mFastTracks[0]; 2370 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2371 delete fastTrack->mBufferProvider; 2372 sq->end(false /*didModify*/); 2373 delete mFastMixer; 2374#ifdef SOAKER 2375 if (mSoaker != NULL) { 2376 mSoaker->requestExitAndWait(); 2377 } 2378 delete mSoaker; 2379#endif 2380 if (mAudioWatchdog != 0) { 2381 mAudioWatchdog->requestExit(); 2382 mAudioWatchdog->requestExitAndWait(); 2383 mAudioWatchdog.clear(); 2384 } 2385 } 2386 delete mAudioMixer; 2387} 2388 2389class CpuStats { 2390public: 2391 CpuStats(); 2392 void sample(const String8 &title); 2393#ifdef DEBUG_CPU_USAGE 2394private: 2395 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2396 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2397 2398 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2399 2400 int mCpuNum; // thread's current CPU number 2401 int mCpukHz; // frequency of thread's current CPU in kHz 2402#endif 2403}; 2404 2405CpuStats::CpuStats() 2406#ifdef DEBUG_CPU_USAGE 2407 : mCpuNum(-1), mCpukHz(-1) 2408#endif 2409{ 2410} 2411 2412void CpuStats::sample(const String8 &title) { 2413#ifdef DEBUG_CPU_USAGE 2414 // get current thread's delta CPU time in wall clock ns 2415 double wcNs; 2416 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2417 2418 // record sample for wall clock statistics 2419 if (valid) { 2420 mWcStats.sample(wcNs); 2421 } 2422 2423 // get the current CPU number 2424 int cpuNum = sched_getcpu(); 2425 2426 // get the current CPU frequency in kHz 2427 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2428 2429 // check if either CPU number or frequency changed 2430 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2431 mCpuNum = cpuNum; 2432 mCpukHz = cpukHz; 2433 // ignore sample for purposes of cycles 2434 valid = false; 2435 } 2436 2437 // if no change in CPU number or frequency, then record sample for cycle statistics 2438 if (valid && mCpukHz > 0) { 2439 double cycles = wcNs * cpukHz * 0.000001; 2440 mHzStats.sample(cycles); 2441 } 2442 2443 unsigned n = mWcStats.n(); 2444 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2445 if ((n & 127) == 1) { 2446 long long elapsed = mCpuUsage.elapsed(); 2447 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2448 double perLoop = elapsed / (double) n; 2449 double perLoop100 = perLoop * 0.01; 2450 double perLoop1k = perLoop * 0.001; 2451 double mean = mWcStats.mean(); 2452 double stddev = mWcStats.stddev(); 2453 double minimum = mWcStats.minimum(); 2454 double maximum = mWcStats.maximum(); 2455 double meanCycles = mHzStats.mean(); 2456 double stddevCycles = mHzStats.stddev(); 2457 double minCycles = mHzStats.minimum(); 2458 double maxCycles = mHzStats.maximum(); 2459 mCpuUsage.resetElapsed(); 2460 mWcStats.reset(); 2461 mHzStats.reset(); 2462 ALOGD("CPU usage for %s over past %.1f secs\n" 2463 " (%u mixer loops at %.1f mean ms per loop):\n" 2464 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2465 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2466 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2467 title.string(), 2468 elapsed * .000000001, n, perLoop * .000001, 2469 mean * .001, 2470 stddev * .001, 2471 minimum * .001, 2472 maximum * .001, 2473 mean / perLoop100, 2474 stddev / perLoop100, 2475 minimum / perLoop100, 2476 maximum / perLoop100, 2477 meanCycles / perLoop1k, 2478 stddevCycles / perLoop1k, 2479 minCycles / perLoop1k, 2480 maxCycles / perLoop1k); 2481 2482 } 2483 } 2484#endif 2485}; 2486 2487void AudioFlinger::PlaybackThread::checkSilentMode_l() 2488{ 2489 if (!mMasterMute) { 2490 char value[PROPERTY_VALUE_MAX]; 2491 if (property_get("ro.audio.silent", value, "0") > 0) { 2492 char *endptr; 2493 unsigned long ul = strtoul(value, &endptr, 0); 2494 if (*endptr == '\0' && ul != 0) { 2495 ALOGD("Silence is golden"); 2496 // The setprop command will not allow a property to be changed after 2497 // the first time it is set, so we don't have to worry about un-muting. 2498 setMasterMute_l(true); 2499 } 2500 } 2501 } 2502} 2503 2504bool AudioFlinger::PlaybackThread::threadLoop() 2505{ 2506 Vector< sp<Track> > tracksToRemove; 2507 2508 standbyTime = systemTime(); 2509 2510 // MIXER 2511 nsecs_t lastWarning = 0; 2512 2513 // DUPLICATING 2514 // FIXME could this be made local to while loop? 2515 writeFrames = 0; 2516 2517 cacheParameters_l(); 2518 sleepTime = idleSleepTime; 2519 2520if (mType == MIXER) { 2521 sleepTimeShift = 0; 2522} 2523 2524 CpuStats cpuStats; 2525 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2526 2527 acquireWakeLock(); 2528 2529 while (!exitPending()) 2530 { 2531 cpuStats.sample(myName); 2532 2533 Vector< sp<EffectChain> > effectChains; 2534 2535 processConfigEvents(); 2536 2537 { // scope for mLock 2538 2539 Mutex::Autolock _l(mLock); 2540 2541 if (checkForNewParameters_l()) { 2542 cacheParameters_l(); 2543 } 2544 2545 saveOutputTracks(); 2546 2547 // put audio hardware into standby after short delay 2548 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2549 mSuspended > 0)) { 2550 if (!mStandby) { 2551 2552 threadLoop_standby(); 2553 2554 mStandby = true; 2555 mBytesWritten = 0; 2556 } 2557 2558 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2559 // we're about to wait, flush the binder command buffer 2560 IPCThreadState::self()->flushCommands(); 2561 2562 clearOutputTracks(); 2563 2564 if (exitPending()) break; 2565 2566 releaseWakeLock_l(); 2567 // wait until we have something to do... 2568 ALOGV("%s going to sleep", myName.string()); 2569 mWaitWorkCV.wait(mLock); 2570 ALOGV("%s waking up", myName.string()); 2571 acquireWakeLock_l(); 2572 2573 mMixerStatus = MIXER_IDLE; 2574 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2575 2576 checkSilentMode_l(); 2577 2578 standbyTime = systemTime() + standbyDelay; 2579 sleepTime = idleSleepTime; 2580 if (mType == MIXER) { 2581 sleepTimeShift = 0; 2582 } 2583 2584 continue; 2585 } 2586 } 2587 2588 // mMixerStatusIgnoringFastTracks is also updated internally 2589 mMixerStatus = prepareTracks_l(&tracksToRemove); 2590 2591 // prevent any changes in effect chain list and in each effect chain 2592 // during mixing and effect process as the audio buffers could be deleted 2593 // or modified if an effect is created or deleted 2594 lockEffectChains_l(effectChains); 2595 } 2596 2597 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2598 threadLoop_mix(); 2599 } else { 2600 threadLoop_sleepTime(); 2601 } 2602 2603 if (mSuspended > 0) { 2604 sleepTime = suspendSleepTimeUs(); 2605 } 2606 2607 // only process effects if we're going to write 2608 if (sleepTime == 0) { 2609 for (size_t i = 0; i < effectChains.size(); i ++) { 2610 effectChains[i]->process_l(); 2611 } 2612 } 2613 2614 // enable changes in effect chain 2615 unlockEffectChains(effectChains); 2616 2617 // sleepTime == 0 means we must write to audio hardware 2618 if (sleepTime == 0) { 2619 2620 threadLoop_write(); 2621 2622if (mType == MIXER) { 2623 // write blocked detection 2624 nsecs_t now = systemTime(); 2625 nsecs_t delta = now - mLastWriteTime; 2626 if (!mStandby && delta > maxPeriod) { 2627 mNumDelayedWrites++; 2628 if ((now - lastWarning) > kWarningThrottleNs) { 2629#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2630 ScopedTrace st(ATRACE_TAG, "underrun"); 2631#endif 2632 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2633 ns2ms(delta), mNumDelayedWrites, this); 2634 lastWarning = now; 2635 } 2636 } 2637} 2638 2639 mStandby = false; 2640 } else { 2641 usleep(sleepTime); 2642 } 2643 2644 // Finally let go of removed track(s), without the lock held 2645 // since we can't guarantee the destructors won't acquire that 2646 // same lock. This will also mutate and push a new fast mixer state. 2647 threadLoop_removeTracks(tracksToRemove); 2648 tracksToRemove.clear(); 2649 2650 // FIXME I don't understand the need for this here; 2651 // it was in the original code but maybe the 2652 // assignment in saveOutputTracks() makes this unnecessary? 2653 clearOutputTracks(); 2654 2655 // Effect chains will be actually deleted here if they were removed from 2656 // mEffectChains list during mixing or effects processing 2657 effectChains.clear(); 2658 2659 // FIXME Note that the above .clear() is no longer necessary since effectChains 2660 // is now local to this block, but will keep it for now (at least until merge done). 2661 } 2662 2663if (mType == MIXER || mType == DIRECT) { 2664 // put output stream into standby mode 2665 if (!mStandby) { 2666 mOutput->stream->common.standby(&mOutput->stream->common); 2667 } 2668} 2669if (mType == DUPLICATING) { 2670 // for DuplicatingThread, standby mode is handled by the outputTracks 2671} 2672 2673 releaseWakeLock(); 2674 2675 ALOGV("Thread %p type %d exiting", this, mType); 2676 return false; 2677} 2678 2679void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2680{ 2681 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2682} 2683 2684void AudioFlinger::MixerThread::threadLoop_write() 2685{ 2686 // FIXME we should only do one push per cycle; confirm this is true 2687 // Start the fast mixer if it's not already running 2688 if (mFastMixer != NULL) { 2689 FastMixerStateQueue *sq = mFastMixer->sq(); 2690 FastMixerState *state = sq->begin(); 2691 if (state->mCommand != FastMixerState::MIX_WRITE && 2692 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2693 if (state->mCommand == FastMixerState::COLD_IDLE) { 2694 int32_t old = android_atomic_inc(&mFastMixerFutex); 2695 if (old == -1) { 2696 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2697 } 2698 if (mAudioWatchdog != 0) { 2699 mAudioWatchdog->resume(); 2700 } 2701 } 2702 state->mCommand = FastMixerState::MIX_WRITE; 2703 sq->end(); 2704 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2705 if (kUseFastMixer == FastMixer_Dynamic) { 2706 mNormalSink = mPipeSink; 2707 } 2708 } else { 2709 sq->end(false /*didModify*/); 2710 } 2711 } 2712 PlaybackThread::threadLoop_write(); 2713} 2714 2715// shared by MIXER and DIRECT, overridden by DUPLICATING 2716void AudioFlinger::PlaybackThread::threadLoop_write() 2717{ 2718 // FIXME rewrite to reduce number of system calls 2719 mLastWriteTime = systemTime(); 2720 mInWrite = true; 2721 int bytesWritten; 2722 2723 // If an NBAIO sink is present, use it to write the normal mixer's submix 2724 if (mNormalSink != 0) { 2725#define mBitShift 2 // FIXME 2726 size_t count = mixBufferSize >> mBitShift; 2727#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2728 Tracer::traceBegin(ATRACE_TAG, "write"); 2729#endif 2730 // update the setpoint when gScreenState changes 2731 uint32_t screenState = gScreenState; 2732 if (screenState != mScreenState) { 2733 mScreenState = screenState; 2734 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2735 if (pipe != NULL) { 2736 pipe->setAvgFrames((mScreenState & 1) ? 2737 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2738 } 2739 } 2740 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2741#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2742 Tracer::traceEnd(ATRACE_TAG); 2743#endif 2744 if (framesWritten > 0) { 2745 bytesWritten = framesWritten << mBitShift; 2746 } else { 2747 bytesWritten = framesWritten; 2748 } 2749 // otherwise use the HAL / AudioStreamOut directly 2750 } else { 2751 // Direct output thread. 2752 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2753 } 2754 2755 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2756 mNumWrites++; 2757 mInWrite = false; 2758} 2759 2760void AudioFlinger::MixerThread::threadLoop_standby() 2761{ 2762 // Idle the fast mixer if it's currently running 2763 if (mFastMixer != NULL) { 2764 FastMixerStateQueue *sq = mFastMixer->sq(); 2765 FastMixerState *state = sq->begin(); 2766 if (!(state->mCommand & FastMixerState::IDLE)) { 2767 state->mCommand = FastMixerState::COLD_IDLE; 2768 state->mColdFutexAddr = &mFastMixerFutex; 2769 state->mColdGen++; 2770 mFastMixerFutex = 0; 2771 sq->end(); 2772 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2773 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2774 if (kUseFastMixer == FastMixer_Dynamic) { 2775 mNormalSink = mOutputSink; 2776 } 2777 if (mAudioWatchdog != 0) { 2778 mAudioWatchdog->pause(); 2779 } 2780 } else { 2781 sq->end(false /*didModify*/); 2782 } 2783 } 2784 PlaybackThread::threadLoop_standby(); 2785} 2786 2787// shared by MIXER and DIRECT, overridden by DUPLICATING 2788void AudioFlinger::PlaybackThread::threadLoop_standby() 2789{ 2790 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2791 mOutput->stream->common.standby(&mOutput->stream->common); 2792} 2793 2794void AudioFlinger::MixerThread::threadLoop_mix() 2795{ 2796 // obtain the presentation timestamp of the next output buffer 2797 int64_t pts; 2798 status_t status = INVALID_OPERATION; 2799 2800 if (NULL != mOutput->stream->get_next_write_timestamp) { 2801 status = mOutput->stream->get_next_write_timestamp( 2802 mOutput->stream, &pts); 2803 } 2804 2805 if (status != NO_ERROR) { 2806 pts = AudioBufferProvider::kInvalidPTS; 2807 } 2808 2809 // mix buffers... 2810 mAudioMixer->process(pts); 2811 // increase sleep time progressively when application underrun condition clears. 2812 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2813 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2814 // such that we would underrun the audio HAL. 2815 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2816 sleepTimeShift--; 2817 } 2818 sleepTime = 0; 2819 standbyTime = systemTime() + standbyDelay; 2820 //TODO: delay standby when effects have a tail 2821} 2822 2823void AudioFlinger::MixerThread::threadLoop_sleepTime() 2824{ 2825 // If no tracks are ready, sleep once for the duration of an output 2826 // buffer size, then write 0s to the output 2827 if (sleepTime == 0) { 2828 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2829 sleepTime = activeSleepTime >> sleepTimeShift; 2830 if (sleepTime < kMinThreadSleepTimeUs) { 2831 sleepTime = kMinThreadSleepTimeUs; 2832 } 2833 // reduce sleep time in case of consecutive application underruns to avoid 2834 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2835 // duration we would end up writing less data than needed by the audio HAL if 2836 // the condition persists. 2837 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2838 sleepTimeShift++; 2839 } 2840 } else { 2841 sleepTime = idleSleepTime; 2842 } 2843 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2844 memset (mMixBuffer, 0, mixBufferSize); 2845 sleepTime = 0; 2846 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2847 } 2848 // TODO add standby time extension fct of effect tail 2849} 2850 2851// prepareTracks_l() must be called with ThreadBase::mLock held 2852AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2853 Vector< sp<Track> > *tracksToRemove) 2854{ 2855 2856 mixer_state mixerStatus = MIXER_IDLE; 2857 // find out which tracks need to be processed 2858 size_t count = mActiveTracks.size(); 2859 size_t mixedTracks = 0; 2860 size_t tracksWithEffect = 0; 2861 // counts only _active_ fast tracks 2862 size_t fastTracks = 0; 2863 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2864 2865 float masterVolume = mMasterVolume; 2866 bool masterMute = mMasterMute; 2867 2868 if (masterMute) { 2869 masterVolume = 0; 2870 } 2871 // Delegate master volume control to effect in output mix effect chain if needed 2872 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2873 if (chain != 0) { 2874 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2875 chain->setVolume_l(&v, &v); 2876 masterVolume = (float)((v + (1 << 23)) >> 24); 2877 chain.clear(); 2878 } 2879 2880 // prepare a new state to push 2881 FastMixerStateQueue *sq = NULL; 2882 FastMixerState *state = NULL; 2883 bool didModify = false; 2884 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2885 if (mFastMixer != NULL) { 2886 sq = mFastMixer->sq(); 2887 state = sq->begin(); 2888 } 2889 2890 for (size_t i=0 ; i<count ; i++) { 2891 sp<Track> t = mActiveTracks[i].promote(); 2892 if (t == 0) continue; 2893 2894 // this const just means the local variable doesn't change 2895 Track* const track = t.get(); 2896 2897 // process fast tracks 2898 if (track->isFastTrack()) { 2899 2900 // It's theoretically possible (though unlikely) for a fast track to be created 2901 // and then removed within the same normal mix cycle. This is not a problem, as 2902 // the track never becomes active so it's fast mixer slot is never touched. 2903 // The converse, of removing an (active) track and then creating a new track 2904 // at the identical fast mixer slot within the same normal mix cycle, 2905 // is impossible because the slot isn't marked available until the end of each cycle. 2906 int j = track->mFastIndex; 2907 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2908 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2909 FastTrack *fastTrack = &state->mFastTracks[j]; 2910 2911 // Determine whether the track is currently in underrun condition, 2912 // and whether it had a recent underrun. 2913 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2914 FastTrackUnderruns underruns = ftDump->mUnderruns; 2915 uint32_t recentFull = (underruns.mBitFields.mFull - 2916 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2917 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2918 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2919 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2920 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2921 uint32_t recentUnderruns = recentPartial + recentEmpty; 2922 track->mObservedUnderruns = underruns; 2923 // don't count underruns that occur while stopping or pausing 2924 // or stopped which can occur when flush() is called while active 2925 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2926 track->mUnderrunCount += recentUnderruns; 2927 } 2928 2929 // This is similar to the state machine for normal tracks, 2930 // with a few modifications for fast tracks. 2931 bool isActive = true; 2932 switch (track->mState) { 2933 case TrackBase::STOPPING_1: 2934 // track stays active in STOPPING_1 state until first underrun 2935 if (recentUnderruns > 0) { 2936 track->mState = TrackBase::STOPPING_2; 2937 } 2938 break; 2939 case TrackBase::PAUSING: 2940 // ramp down is not yet implemented 2941 track->setPaused(); 2942 break; 2943 case TrackBase::RESUMING: 2944 // ramp up is not yet implemented 2945 track->mState = TrackBase::ACTIVE; 2946 break; 2947 case TrackBase::ACTIVE: 2948 if (recentFull > 0 || recentPartial > 0) { 2949 // track has provided at least some frames recently: reset retry count 2950 track->mRetryCount = kMaxTrackRetries; 2951 } 2952 if (recentUnderruns == 0) { 2953 // no recent underruns: stay active 2954 break; 2955 } 2956 // there has recently been an underrun of some kind 2957 if (track->sharedBuffer() == 0) { 2958 // were any of the recent underruns "empty" (no frames available)? 2959 if (recentEmpty == 0) { 2960 // no, then ignore the partial underruns as they are allowed indefinitely 2961 break; 2962 } 2963 // there has recently been an "empty" underrun: decrement the retry counter 2964 if (--(track->mRetryCount) > 0) { 2965 break; 2966 } 2967 // indicate to client process that the track was disabled because of underrun; 2968 // it will then automatically call start() when data is available 2969 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2970 // remove from active list, but state remains ACTIVE [confusing but true] 2971 isActive = false; 2972 break; 2973 } 2974 // fall through 2975 case TrackBase::STOPPING_2: 2976 case TrackBase::PAUSED: 2977 case TrackBase::TERMINATED: 2978 case TrackBase::STOPPED: 2979 case TrackBase::FLUSHED: // flush() while active 2980 // Check for presentation complete if track is inactive 2981 // We have consumed all the buffers of this track. 2982 // This would be incomplete if we auto-paused on underrun 2983 { 2984 size_t audioHALFrames = 2985 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2986 size_t framesWritten = 2987 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2988 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2989 // track stays in active list until presentation is complete 2990 break; 2991 } 2992 } 2993 if (track->isStopping_2()) { 2994 track->mState = TrackBase::STOPPED; 2995 } 2996 if (track->isStopped()) { 2997 // Can't reset directly, as fast mixer is still polling this track 2998 // track->reset(); 2999 // So instead mark this track as needing to be reset after push with ack 3000 resetMask |= 1 << i; 3001 } 3002 isActive = false; 3003 break; 3004 case TrackBase::IDLE: 3005 default: 3006 LOG_FATAL("unexpected track state %d", track->mState); 3007 } 3008 3009 if (isActive) { 3010 // was it previously inactive? 3011 if (!(state->mTrackMask & (1 << j))) { 3012 ExtendedAudioBufferProvider *eabp = track; 3013 VolumeProvider *vp = track; 3014 fastTrack->mBufferProvider = eabp; 3015 fastTrack->mVolumeProvider = vp; 3016 fastTrack->mSampleRate = track->mSampleRate; 3017 fastTrack->mChannelMask = track->mChannelMask; 3018 fastTrack->mGeneration++; 3019 state->mTrackMask |= 1 << j; 3020 didModify = true; 3021 // no acknowledgement required for newly active tracks 3022 } 3023 // cache the combined master volume and stream type volume for fast mixer; this 3024 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3025 track->mCachedVolume = track->isMuted() ? 3026 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3027 ++fastTracks; 3028 } else { 3029 // was it previously active? 3030 if (state->mTrackMask & (1 << j)) { 3031 fastTrack->mBufferProvider = NULL; 3032 fastTrack->mGeneration++; 3033 state->mTrackMask &= ~(1 << j); 3034 didModify = true; 3035 // If any fast tracks were removed, we must wait for acknowledgement 3036 // because we're about to decrement the last sp<> on those tracks. 3037 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3038 } else { 3039 LOG_FATAL("fast track %d should have been active", j); 3040 } 3041 tracksToRemove->add(track); 3042 // Avoids a misleading display in dumpsys 3043 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3044 } 3045 continue; 3046 } 3047 3048 { // local variable scope to avoid goto warning 3049 3050 audio_track_cblk_t* cblk = track->cblk(); 3051 3052 // The first time a track is added we wait 3053 // for all its buffers to be filled before processing it 3054 int name = track->name(); 3055 // make sure that we have enough frames to mix one full buffer. 3056 // enforce this condition only once to enable draining the buffer in case the client 3057 // app does not call stop() and relies on underrun to stop: 3058 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3059 // during last round 3060 uint32_t minFrames = 1; 3061 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3062 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3063 if (t->sampleRate() == (int)mSampleRate) { 3064 minFrames = mNormalFrameCount; 3065 } else { 3066 // +1 for rounding and +1 for additional sample needed for interpolation 3067 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3068 // add frames already consumed but not yet released by the resampler 3069 // because cblk->framesReady() will include these frames 3070 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3071 // the minimum track buffer size is normally twice the number of frames necessary 3072 // to fill one buffer and the resampler should not leave more than one buffer worth 3073 // of unreleased frames after each pass, but just in case... 3074 ALOG_ASSERT(minFrames <= cblk->frameCount); 3075 } 3076 } 3077 if ((track->framesReady() >= minFrames) && track->isReady() && 3078 !track->isPaused() && !track->isTerminated()) 3079 { 3080 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3081 3082 mixedTracks++; 3083 3084 // track->mainBuffer() != mMixBuffer means there is an effect chain 3085 // connected to the track 3086 chain.clear(); 3087 if (track->mainBuffer() != mMixBuffer) { 3088 chain = getEffectChain_l(track->sessionId()); 3089 // Delegate volume control to effect in track effect chain if needed 3090 if (chain != 0) { 3091 tracksWithEffect++; 3092 } else { 3093 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3094 name, track->sessionId()); 3095 } 3096 } 3097 3098 3099 int param = AudioMixer::VOLUME; 3100 if (track->mFillingUpStatus == Track::FS_FILLED) { 3101 // no ramp for the first volume setting 3102 track->mFillingUpStatus = Track::FS_ACTIVE; 3103 if (track->mState == TrackBase::RESUMING) { 3104 track->mState = TrackBase::ACTIVE; 3105 param = AudioMixer::RAMP_VOLUME; 3106 } 3107 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3108 } else if (cblk->server != 0) { 3109 // If the track is stopped before the first frame was mixed, 3110 // do not apply ramp 3111 param = AudioMixer::RAMP_VOLUME; 3112 } 3113 3114 // compute volume for this track 3115 uint32_t vl, vr, va; 3116 if (track->isMuted() || track->isPausing() || 3117 mStreamTypes[track->streamType()].mute) { 3118 vl = vr = va = 0; 3119 if (track->isPausing()) { 3120 track->setPaused(); 3121 } 3122 } else { 3123 3124 // read original volumes with volume control 3125 float typeVolume = mStreamTypes[track->streamType()].volume; 3126 float v = masterVolume * typeVolume; 3127 uint32_t vlr = cblk->getVolumeLR(); 3128 vl = vlr & 0xFFFF; 3129 vr = vlr >> 16; 3130 // track volumes come from shared memory, so can't be trusted and must be clamped 3131 if (vl > MAX_GAIN_INT) { 3132 ALOGV("Track left volume out of range: %04X", vl); 3133 vl = MAX_GAIN_INT; 3134 } 3135 if (vr > MAX_GAIN_INT) { 3136 ALOGV("Track right volume out of range: %04X", vr); 3137 vr = MAX_GAIN_INT; 3138 } 3139 // now apply the master volume and stream type volume 3140 vl = (uint32_t)(v * vl) << 12; 3141 vr = (uint32_t)(v * vr) << 12; 3142 // assuming master volume and stream type volume each go up to 1.0, 3143 // vl and vr are now in 8.24 format 3144 3145 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3146 // send level comes from shared memory and so may be corrupt 3147 if (sendLevel > MAX_GAIN_INT) { 3148 ALOGV("Track send level out of range: %04X", sendLevel); 3149 sendLevel = MAX_GAIN_INT; 3150 } 3151 va = (uint32_t)(v * sendLevel); 3152 } 3153 // Delegate volume control to effect in track effect chain if needed 3154 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3155 // Do not ramp volume if volume is controlled by effect 3156 param = AudioMixer::VOLUME; 3157 track->mHasVolumeController = true; 3158 } else { 3159 // force no volume ramp when volume controller was just disabled or removed 3160 // from effect chain to avoid volume spike 3161 if (track->mHasVolumeController) { 3162 param = AudioMixer::VOLUME; 3163 } 3164 track->mHasVolumeController = false; 3165 } 3166 3167 // Convert volumes from 8.24 to 4.12 format 3168 // This additional clamping is needed in case chain->setVolume_l() overshot 3169 vl = (vl + (1 << 11)) >> 12; 3170 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3171 vr = (vr + (1 << 11)) >> 12; 3172 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3173 3174 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3175 3176 // XXX: these things DON'T need to be done each time 3177 mAudioMixer->setBufferProvider(name, track); 3178 mAudioMixer->enable(name); 3179 3180 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3181 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3182 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3183 mAudioMixer->setParameter( 3184 name, 3185 AudioMixer::TRACK, 3186 AudioMixer::FORMAT, (void *)track->format()); 3187 mAudioMixer->setParameter( 3188 name, 3189 AudioMixer::TRACK, 3190 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3191 mAudioMixer->setParameter( 3192 name, 3193 AudioMixer::RESAMPLE, 3194 AudioMixer::SAMPLE_RATE, 3195 (void *)(cblk->sampleRate)); 3196 mAudioMixer->setParameter( 3197 name, 3198 AudioMixer::TRACK, 3199 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3200 mAudioMixer->setParameter( 3201 name, 3202 AudioMixer::TRACK, 3203 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3204 3205 // reset retry count 3206 track->mRetryCount = kMaxTrackRetries; 3207 3208 // If one track is ready, set the mixer ready if: 3209 // - the mixer was not ready during previous round OR 3210 // - no other track is not ready 3211 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3212 mixerStatus != MIXER_TRACKS_ENABLED) { 3213 mixerStatus = MIXER_TRACKS_READY; 3214 } 3215 } else { 3216 // clear effect chain input buffer if an active track underruns to avoid sending 3217 // previous audio buffer again to effects 3218 chain = getEffectChain_l(track->sessionId()); 3219 if (chain != 0) { 3220 chain->clearInputBuffer(); 3221 } 3222 3223 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3224 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3225 track->isStopped() || track->isPaused()) { 3226 // We have consumed all the buffers of this track. 3227 // Remove it from the list of active tracks. 3228 // TODO: use actual buffer filling status instead of latency when available from 3229 // audio HAL 3230 size_t audioHALFrames = 3231 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3232 size_t framesWritten = 3233 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3234 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3235 if (track->isStopped()) { 3236 track->reset(); 3237 } 3238 tracksToRemove->add(track); 3239 } 3240 } else { 3241 track->mUnderrunCount++; 3242 // No buffers for this track. Give it a few chances to 3243 // fill a buffer, then remove it from active list. 3244 if (--(track->mRetryCount) <= 0) { 3245 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3246 tracksToRemove->add(track); 3247 // indicate to client process that the track was disabled because of underrun; 3248 // it will then automatically call start() when data is available 3249 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3250 // If one track is not ready, mark the mixer also not ready if: 3251 // - the mixer was ready during previous round OR 3252 // - no other track is ready 3253 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3254 mixerStatus != MIXER_TRACKS_READY) { 3255 mixerStatus = MIXER_TRACKS_ENABLED; 3256 } 3257 } 3258 mAudioMixer->disable(name); 3259 } 3260 3261 } // local variable scope to avoid goto warning 3262track_is_ready: ; 3263 3264 } 3265 3266 // Push the new FastMixer state if necessary 3267 bool pauseAudioWatchdog = false; 3268 if (didModify) { 3269 state->mFastTracksGen++; 3270 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3271 if (kUseFastMixer == FastMixer_Dynamic && 3272 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3273 state->mCommand = FastMixerState::COLD_IDLE; 3274 state->mColdFutexAddr = &mFastMixerFutex; 3275 state->mColdGen++; 3276 mFastMixerFutex = 0; 3277 if (kUseFastMixer == FastMixer_Dynamic) { 3278 mNormalSink = mOutputSink; 3279 } 3280 // If we go into cold idle, need to wait for acknowledgement 3281 // so that fast mixer stops doing I/O. 3282 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3283 pauseAudioWatchdog = true; 3284 } 3285 sq->end(); 3286 } 3287 if (sq != NULL) { 3288 sq->end(didModify); 3289 sq->push(block); 3290 } 3291 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3292 mAudioWatchdog->pause(); 3293 } 3294 3295 // Now perform the deferred reset on fast tracks that have stopped 3296 while (resetMask != 0) { 3297 size_t i = __builtin_ctz(resetMask); 3298 ALOG_ASSERT(i < count); 3299 resetMask &= ~(1 << i); 3300 sp<Track> t = mActiveTracks[i].promote(); 3301 if (t == 0) continue; 3302 Track* track = t.get(); 3303 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3304 track->reset(); 3305 } 3306 3307 // remove all the tracks that need to be... 3308 count = tracksToRemove->size(); 3309 if (CC_UNLIKELY(count)) { 3310 for (size_t i=0 ; i<count ; i++) { 3311 const sp<Track>& track = tracksToRemove->itemAt(i); 3312 mActiveTracks.remove(track); 3313 if (track->mainBuffer() != mMixBuffer) { 3314 chain = getEffectChain_l(track->sessionId()); 3315 if (chain != 0) { 3316 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3317 chain->decActiveTrackCnt(); 3318 } 3319 } 3320 if (track->isTerminated()) { 3321 removeTrack_l(track); 3322 } 3323 } 3324 } 3325 3326 // mix buffer must be cleared if all tracks are connected to an 3327 // effect chain as in this case the mixer will not write to 3328 // mix buffer and track effects will accumulate into it 3329 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3330 // FIXME as a performance optimization, should remember previous zero status 3331 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3332 } 3333 3334 // if any fast tracks, then status is ready 3335 mMixerStatusIgnoringFastTracks = mixerStatus; 3336 if (fastTracks > 0) { 3337 mixerStatus = MIXER_TRACKS_READY; 3338 } 3339 return mixerStatus; 3340} 3341 3342/* 3343The derived values that are cached: 3344 - mixBufferSize from frame count * frame size 3345 - activeSleepTime from activeSleepTimeUs() 3346 - idleSleepTime from idleSleepTimeUs() 3347 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3348 - maxPeriod from frame count and sample rate (MIXER only) 3349 3350The parameters that affect these derived values are: 3351 - frame count 3352 - frame size 3353 - sample rate 3354 - device type: A2DP or not 3355 - device latency 3356 - format: PCM or not 3357 - active sleep time 3358 - idle sleep time 3359*/ 3360 3361void AudioFlinger::PlaybackThread::cacheParameters_l() 3362{ 3363 mixBufferSize = mNormalFrameCount * mFrameSize; 3364 activeSleepTime = activeSleepTimeUs(); 3365 idleSleepTime = idleSleepTimeUs(); 3366} 3367 3368void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3369{ 3370 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3371 this, streamType, mTracks.size()); 3372 Mutex::Autolock _l(mLock); 3373 3374 size_t size = mTracks.size(); 3375 for (size_t i = 0; i < size; i++) { 3376 sp<Track> t = mTracks[i]; 3377 if (t->streamType() == streamType) { 3378 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3379 t->mCblk->cv.signal(); 3380 } 3381 } 3382} 3383 3384// getTrackName_l() must be called with ThreadBase::mLock held 3385int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3386{ 3387 return mAudioMixer->getTrackName(channelMask); 3388} 3389 3390// deleteTrackName_l() must be called with ThreadBase::mLock held 3391void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3392{ 3393 ALOGV("remove track (%d) and delete from mixer", name); 3394 mAudioMixer->deleteTrackName(name); 3395} 3396 3397// checkForNewParameters_l() must be called with ThreadBase::mLock held 3398bool AudioFlinger::MixerThread::checkForNewParameters_l() 3399{ 3400 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3401 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3402 bool reconfig = false; 3403 3404 while (!mNewParameters.isEmpty()) { 3405 3406 if (mFastMixer != NULL) { 3407 FastMixerStateQueue *sq = mFastMixer->sq(); 3408 FastMixerState *state = sq->begin(); 3409 if (!(state->mCommand & FastMixerState::IDLE)) { 3410 previousCommand = state->mCommand; 3411 state->mCommand = FastMixerState::HOT_IDLE; 3412 sq->end(); 3413 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3414 } else { 3415 sq->end(false /*didModify*/); 3416 } 3417 } 3418 3419 status_t status = NO_ERROR; 3420 String8 keyValuePair = mNewParameters[0]; 3421 AudioParameter param = AudioParameter(keyValuePair); 3422 int value; 3423 3424 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3425 reconfig = true; 3426 } 3427 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3428 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3429 status = BAD_VALUE; 3430 } else { 3431 reconfig = true; 3432 } 3433 } 3434 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3435 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3436 status = BAD_VALUE; 3437 } else { 3438 reconfig = true; 3439 } 3440 } 3441 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3442 // do not accept frame count changes if tracks are open as the track buffer 3443 // size depends on frame count and correct behavior would not be guaranteed 3444 // if frame count is changed after track creation 3445 if (!mTracks.isEmpty()) { 3446 status = INVALID_OPERATION; 3447 } else { 3448 reconfig = true; 3449 } 3450 } 3451 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3452#ifdef ADD_BATTERY_DATA 3453 // when changing the audio output device, call addBatteryData to notify 3454 // the change 3455 if ((int)mDevice != value) { 3456 uint32_t params = 0; 3457 // check whether speaker is on 3458 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3459 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3460 } 3461 3462 int deviceWithoutSpeaker 3463 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3464 // check if any other device (except speaker) is on 3465 if (value & deviceWithoutSpeaker ) { 3466 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3467 } 3468 3469 if (params != 0) { 3470 addBatteryData(params); 3471 } 3472 } 3473#endif 3474 3475 // forward device change to effects that have requested to be 3476 // aware of attached audio device. 3477 mDevice = (uint32_t)value; 3478 for (size_t i = 0; i < mEffectChains.size(); i++) { 3479 mEffectChains[i]->setDevice_l(mDevice); 3480 } 3481 } 3482 3483 if (status == NO_ERROR) { 3484 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3485 keyValuePair.string()); 3486 if (!mStandby && status == INVALID_OPERATION) { 3487 mOutput->stream->common.standby(&mOutput->stream->common); 3488 mStandby = true; 3489 mBytesWritten = 0; 3490 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3491 keyValuePair.string()); 3492 } 3493 if (status == NO_ERROR && reconfig) { 3494 delete mAudioMixer; 3495 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3496 mAudioMixer = NULL; 3497 readOutputParameters(); 3498 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3499 for (size_t i = 0; i < mTracks.size() ; i++) { 3500 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3501 if (name < 0) break; 3502 mTracks[i]->mName = name; 3503 // limit track sample rate to 2 x new output sample rate 3504 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3505 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3506 } 3507 } 3508 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3509 } 3510 } 3511 3512 mNewParameters.removeAt(0); 3513 3514 mParamStatus = status; 3515 mParamCond.signal(); 3516 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3517 // already timed out waiting for the status and will never signal the condition. 3518 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3519 } 3520 3521 if (!(previousCommand & FastMixerState::IDLE)) { 3522 ALOG_ASSERT(mFastMixer != NULL); 3523 FastMixerStateQueue *sq = mFastMixer->sq(); 3524 FastMixerState *state = sq->begin(); 3525 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3526 state->mCommand = previousCommand; 3527 sq->end(); 3528 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3529 } 3530 3531 return reconfig; 3532} 3533 3534status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3535{ 3536 const size_t SIZE = 256; 3537 char buffer[SIZE]; 3538 String8 result; 3539 3540 PlaybackThread::dumpInternals(fd, args); 3541 3542 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3543 result.append(buffer); 3544 write(fd, result.string(), result.size()); 3545 3546 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3547 FastMixerDumpState copy = mFastMixerDumpState; 3548 copy.dump(fd); 3549 3550#ifdef STATE_QUEUE_DUMP 3551 // Similar for state queue 3552 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3553 observerCopy.dump(fd); 3554 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3555 mutatorCopy.dump(fd); 3556#endif 3557 3558 // Write the tee output to a .wav file 3559 NBAIO_Source *teeSource = mTeeSource.get(); 3560 if (teeSource != NULL) { 3561 char teePath[64]; 3562 struct timeval tv; 3563 gettimeofday(&tv, NULL); 3564 struct tm tm; 3565 localtime_r(&tv.tv_sec, &tm); 3566 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3567 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3568 if (teeFd >= 0) { 3569 char wavHeader[44]; 3570 memcpy(wavHeader, 3571 "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", 3572 sizeof(wavHeader)); 3573 NBAIO_Format format = teeSource->format(); 3574 unsigned channelCount = Format_channelCount(format); 3575 ALOG_ASSERT(channelCount <= FCC_2); 3576 unsigned sampleRate = Format_sampleRate(format); 3577 wavHeader[22] = channelCount; // number of channels 3578 wavHeader[24] = sampleRate; // sample rate 3579 wavHeader[25] = sampleRate >> 8; 3580 wavHeader[32] = channelCount * 2; // block alignment 3581 write(teeFd, wavHeader, sizeof(wavHeader)); 3582 size_t total = 0; 3583 bool firstRead = true; 3584 for (;;) { 3585#define TEE_SINK_READ 1024 3586 short buffer[TEE_SINK_READ * FCC_2]; 3587 size_t count = TEE_SINK_READ; 3588 ssize_t actual = teeSource->read(buffer, count); 3589 bool wasFirstRead = firstRead; 3590 firstRead = false; 3591 if (actual <= 0) { 3592 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3593 continue; 3594 } 3595 break; 3596 } 3597 ALOG_ASSERT(actual <= count); 3598 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3599 total += actual; 3600 } 3601 lseek(teeFd, (off_t) 4, SEEK_SET); 3602 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3603 write(teeFd, &temp, sizeof(temp)); 3604 lseek(teeFd, (off_t) 40, SEEK_SET); 3605 temp = total * channelCount * sizeof(short); 3606 write(teeFd, &temp, sizeof(temp)); 3607 close(teeFd); 3608 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3609 } else { 3610 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3611 } 3612 } 3613 3614 if (mAudioWatchdog != 0) { 3615 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3616 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3617 wdCopy.dump(fd); 3618 } 3619 3620 return NO_ERROR; 3621} 3622 3623uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3624{ 3625 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3626} 3627 3628uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3629{ 3630 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3631} 3632 3633void AudioFlinger::MixerThread::cacheParameters_l() 3634{ 3635 PlaybackThread::cacheParameters_l(); 3636 3637 // FIXME: Relaxed timing because of a certain device that can't meet latency 3638 // Should be reduced to 2x after the vendor fixes the driver issue 3639 // increase threshold again due to low power audio mode. The way this warning 3640 // threshold is calculated and its usefulness should be reconsidered anyway. 3641 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3642} 3643 3644// ---------------------------------------------------------------------------- 3645AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3646 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3647 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3648 // mLeftVolFloat, mRightVolFloat 3649{ 3650} 3651 3652AudioFlinger::DirectOutputThread::~DirectOutputThread() 3653{ 3654} 3655 3656AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3657 Vector< sp<Track> > *tracksToRemove 3658) 3659{ 3660 sp<Track> trackToRemove; 3661 3662 mixer_state mixerStatus = MIXER_IDLE; 3663 3664 // find out which tracks need to be processed 3665 if (mActiveTracks.size() != 0) { 3666 sp<Track> t = mActiveTracks[0].promote(); 3667 // The track died recently 3668 if (t == 0) return MIXER_IDLE; 3669 3670 Track* const track = t.get(); 3671 audio_track_cblk_t* cblk = track->cblk(); 3672 3673 // The first time a track is added we wait 3674 // for all its buffers to be filled before processing it 3675 uint32_t minFrames; 3676 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3677 minFrames = mNormalFrameCount; 3678 } else { 3679 minFrames = 1; 3680 } 3681 if ((track->framesReady() >= minFrames) && track->isReady() && 3682 !track->isPaused() && !track->isTerminated()) 3683 { 3684 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3685 3686 if (track->mFillingUpStatus == Track::FS_FILLED) { 3687 track->mFillingUpStatus = Track::FS_ACTIVE; 3688 mLeftVolFloat = mRightVolFloat = 0; 3689 if (track->mState == TrackBase::RESUMING) { 3690 track->mState = TrackBase::ACTIVE; 3691 } 3692 } 3693 3694 // compute volume for this track 3695 float left, right; 3696 if (track->isMuted() || mMasterMute || track->isPausing() || 3697 mStreamTypes[track->streamType()].mute) { 3698 left = right = 0; 3699 if (track->isPausing()) { 3700 track->setPaused(); 3701 } 3702 } else { 3703 float typeVolume = mStreamTypes[track->streamType()].volume; 3704 float v = mMasterVolume * typeVolume; 3705 uint32_t vlr = cblk->getVolumeLR(); 3706 float v_clamped = v * (vlr & 0xFFFF); 3707 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3708 left = v_clamped/MAX_GAIN; 3709 v_clamped = v * (vlr >> 16); 3710 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3711 right = v_clamped/MAX_GAIN; 3712 } 3713 3714 if (left != mLeftVolFloat || right != mRightVolFloat) { 3715 mLeftVolFloat = left; 3716 mRightVolFloat = right; 3717 3718 // Convert volumes from float to 8.24 3719 uint32_t vl = (uint32_t)(left * (1 << 24)); 3720 uint32_t vr = (uint32_t)(right * (1 << 24)); 3721 3722 // Delegate volume control to effect in track effect chain if needed 3723 // only one effect chain can be present on DirectOutputThread, so if 3724 // there is one, the track is connected to it 3725 if (!mEffectChains.isEmpty()) { 3726 // Do not ramp volume if volume is controlled by effect 3727 mEffectChains[0]->setVolume_l(&vl, &vr); 3728 left = (float)vl / (1 << 24); 3729 right = (float)vr / (1 << 24); 3730 } 3731 mOutput->stream->set_volume(mOutput->stream, left, right); 3732 } 3733 3734 // reset retry count 3735 track->mRetryCount = kMaxTrackRetriesDirect; 3736 mActiveTrack = t; 3737 mixerStatus = MIXER_TRACKS_READY; 3738 } else { 3739 // clear effect chain input buffer if an active track underruns to avoid sending 3740 // previous audio buffer again to effects 3741 if (!mEffectChains.isEmpty()) { 3742 mEffectChains[0]->clearInputBuffer(); 3743 } 3744 3745 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3746 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3747 track->isStopped() || track->isPaused()) { 3748 // We have consumed all the buffers of this track. 3749 // Remove it from the list of active tracks. 3750 // TODO: implement behavior for compressed audio 3751 size_t audioHALFrames = 3752 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3753 size_t framesWritten = 3754 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3755 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3756 if (track->isStopped()) { 3757 track->reset(); 3758 } 3759 trackToRemove = track; 3760 } 3761 } else { 3762 // No buffers for this track. Give it a few chances to 3763 // fill a buffer, then remove it from active list. 3764 if (--(track->mRetryCount) <= 0) { 3765 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3766 trackToRemove = track; 3767 } else { 3768 mixerStatus = MIXER_TRACKS_ENABLED; 3769 } 3770 } 3771 } 3772 } 3773 3774 // FIXME merge this with similar code for removing multiple tracks 3775 // remove all the tracks that need to be... 3776 if (CC_UNLIKELY(trackToRemove != 0)) { 3777 tracksToRemove->add(trackToRemove); 3778 mActiveTracks.remove(trackToRemove); 3779 if (!mEffectChains.isEmpty()) { 3780 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3781 trackToRemove->sessionId()); 3782 mEffectChains[0]->decActiveTrackCnt(); 3783 } 3784 if (trackToRemove->isTerminated()) { 3785 removeTrack_l(trackToRemove); 3786 } 3787 } 3788 3789 return mixerStatus; 3790} 3791 3792void AudioFlinger::DirectOutputThread::threadLoop_mix() 3793{ 3794 AudioBufferProvider::Buffer buffer; 3795 size_t frameCount = mFrameCount; 3796 int8_t *curBuf = (int8_t *)mMixBuffer; 3797 // output audio to hardware 3798 while (frameCount) { 3799 buffer.frameCount = frameCount; 3800 mActiveTrack->getNextBuffer(&buffer); 3801 if (CC_UNLIKELY(buffer.raw == NULL)) { 3802 memset(curBuf, 0, frameCount * mFrameSize); 3803 break; 3804 } 3805 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3806 frameCount -= buffer.frameCount; 3807 curBuf += buffer.frameCount * mFrameSize; 3808 mActiveTrack->releaseBuffer(&buffer); 3809 } 3810 sleepTime = 0; 3811 standbyTime = systemTime() + standbyDelay; 3812 mActiveTrack.clear(); 3813 3814} 3815 3816void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3817{ 3818 if (sleepTime == 0) { 3819 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3820 sleepTime = activeSleepTime; 3821 } else { 3822 sleepTime = idleSleepTime; 3823 } 3824 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3825 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3826 sleepTime = 0; 3827 } 3828} 3829 3830// getTrackName_l() must be called with ThreadBase::mLock held 3831int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3832{ 3833 return 0; 3834} 3835 3836// deleteTrackName_l() must be called with ThreadBase::mLock held 3837void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3838{ 3839} 3840 3841// checkForNewParameters_l() must be called with ThreadBase::mLock held 3842bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3843{ 3844 bool reconfig = false; 3845 3846 while (!mNewParameters.isEmpty()) { 3847 status_t status = NO_ERROR; 3848 String8 keyValuePair = mNewParameters[0]; 3849 AudioParameter param = AudioParameter(keyValuePair); 3850 int value; 3851 3852 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3853 // do not accept frame count changes if tracks are open as the track buffer 3854 // size depends on frame count and correct behavior would not be garantied 3855 // if frame count is changed after track creation 3856 if (!mTracks.isEmpty()) { 3857 status = INVALID_OPERATION; 3858 } else { 3859 reconfig = true; 3860 } 3861 } 3862 if (status == NO_ERROR) { 3863 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3864 keyValuePair.string()); 3865 if (!mStandby && status == INVALID_OPERATION) { 3866 mOutput->stream->common.standby(&mOutput->stream->common); 3867 mStandby = true; 3868 mBytesWritten = 0; 3869 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3870 keyValuePair.string()); 3871 } 3872 if (status == NO_ERROR && reconfig) { 3873 readOutputParameters(); 3874 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3875 } 3876 } 3877 3878 mNewParameters.removeAt(0); 3879 3880 mParamStatus = status; 3881 mParamCond.signal(); 3882 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3883 // already timed out waiting for the status and will never signal the condition. 3884 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3885 } 3886 return reconfig; 3887} 3888 3889uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3890{ 3891 uint32_t time; 3892 if (audio_is_linear_pcm(mFormat)) { 3893 time = PlaybackThread::activeSleepTimeUs(); 3894 } else { 3895 time = 10000; 3896 } 3897 return time; 3898} 3899 3900uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3901{ 3902 uint32_t time; 3903 if (audio_is_linear_pcm(mFormat)) { 3904 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3905 } else { 3906 time = 10000; 3907 } 3908 return time; 3909} 3910 3911uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3912{ 3913 uint32_t time; 3914 if (audio_is_linear_pcm(mFormat)) { 3915 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3916 } else { 3917 time = 10000; 3918 } 3919 return time; 3920} 3921 3922void AudioFlinger::DirectOutputThread::cacheParameters_l() 3923{ 3924 PlaybackThread::cacheParameters_l(); 3925 3926 // use shorter standby delay as on normal output to release 3927 // hardware resources as soon as possible 3928 standbyDelay = microseconds(activeSleepTime*2); 3929} 3930 3931// ---------------------------------------------------------------------------- 3932 3933AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3934 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3935 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3936 mWaitTimeMs(UINT_MAX) 3937{ 3938 addOutputTrack(mainThread); 3939} 3940 3941AudioFlinger::DuplicatingThread::~DuplicatingThread() 3942{ 3943 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3944 mOutputTracks[i]->destroy(); 3945 } 3946} 3947 3948void AudioFlinger::DuplicatingThread::threadLoop_mix() 3949{ 3950 // mix buffers... 3951 if (outputsReady(outputTracks)) { 3952 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3953 } else { 3954 memset(mMixBuffer, 0, mixBufferSize); 3955 } 3956 sleepTime = 0; 3957 writeFrames = mNormalFrameCount; 3958 standbyTime = systemTime() + standbyDelay; 3959} 3960 3961void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3962{ 3963 if (sleepTime == 0) { 3964 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3965 sleepTime = activeSleepTime; 3966 } else { 3967 sleepTime = idleSleepTime; 3968 } 3969 } else if (mBytesWritten != 0) { 3970 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3971 writeFrames = mNormalFrameCount; 3972 memset(mMixBuffer, 0, mixBufferSize); 3973 } else { 3974 // flush remaining overflow buffers in output tracks 3975 writeFrames = 0; 3976 } 3977 sleepTime = 0; 3978 } 3979} 3980 3981void AudioFlinger::DuplicatingThread::threadLoop_write() 3982{ 3983 for (size_t i = 0; i < outputTracks.size(); i++) { 3984 outputTracks[i]->write(mMixBuffer, writeFrames); 3985 } 3986 mBytesWritten += mixBufferSize; 3987} 3988 3989void AudioFlinger::DuplicatingThread::threadLoop_standby() 3990{ 3991 // DuplicatingThread implements standby by stopping all tracks 3992 for (size_t i = 0; i < outputTracks.size(); i++) { 3993 outputTracks[i]->stop(); 3994 } 3995} 3996 3997void AudioFlinger::DuplicatingThread::saveOutputTracks() 3998{ 3999 outputTracks = mOutputTracks; 4000} 4001 4002void AudioFlinger::DuplicatingThread::clearOutputTracks() 4003{ 4004 outputTracks.clear(); 4005} 4006 4007void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4008{ 4009 Mutex::Autolock _l(mLock); 4010 // FIXME explain this formula 4011 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4012 OutputTrack *outputTrack = new OutputTrack(thread, 4013 this, 4014 mSampleRate, 4015 mFormat, 4016 mChannelMask, 4017 frameCount); 4018 if (outputTrack->cblk() != NULL) { 4019 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4020 mOutputTracks.add(outputTrack); 4021 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4022 updateWaitTime_l(); 4023 } 4024} 4025 4026void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4027{ 4028 Mutex::Autolock _l(mLock); 4029 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4030 if (mOutputTracks[i]->thread() == thread) { 4031 mOutputTracks[i]->destroy(); 4032 mOutputTracks.removeAt(i); 4033 updateWaitTime_l(); 4034 return; 4035 } 4036 } 4037 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4038} 4039 4040// caller must hold mLock 4041void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4042{ 4043 mWaitTimeMs = UINT_MAX; 4044 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4045 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4046 if (strong != 0) { 4047 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4048 if (waitTimeMs < mWaitTimeMs) { 4049 mWaitTimeMs = waitTimeMs; 4050 } 4051 } 4052 } 4053} 4054 4055 4056bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4057{ 4058 for (size_t i = 0; i < outputTracks.size(); i++) { 4059 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4060 if (thread == 0) { 4061 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4062 return false; 4063 } 4064 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4065 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4066 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4067 return false; 4068 } 4069 } 4070 return true; 4071} 4072 4073uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4074{ 4075 return (mWaitTimeMs * 1000) / 2; 4076} 4077 4078void AudioFlinger::DuplicatingThread::cacheParameters_l() 4079{ 4080 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4081 updateWaitTime_l(); 4082 4083 MixerThread::cacheParameters_l(); 4084} 4085 4086// ---------------------------------------------------------------------------- 4087 4088// TrackBase constructor must be called with AudioFlinger::mLock held 4089AudioFlinger::ThreadBase::TrackBase::TrackBase( 4090 ThreadBase *thread, 4091 const sp<Client>& client, 4092 uint32_t sampleRate, 4093 audio_format_t format, 4094 uint32_t channelMask, 4095 int frameCount, 4096 const sp<IMemory>& sharedBuffer, 4097 int sessionId) 4098 : RefBase(), 4099 mThread(thread), 4100 mClient(client), 4101 mCblk(NULL), 4102 // mBuffer 4103 // mBufferEnd 4104 mFrameCount(0), 4105 mState(IDLE), 4106 mSampleRate(sampleRate), 4107 mFormat(format), 4108 mStepServerFailed(false), 4109 mSessionId(sessionId) 4110 // mChannelCount 4111 // mChannelMask 4112{ 4113 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4114 4115 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4116 size_t size = sizeof(audio_track_cblk_t); 4117 uint8_t channelCount = popcount(channelMask); 4118 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4119 if (sharedBuffer == 0) { 4120 size += bufferSize; 4121 } 4122 4123 if (client != NULL) { 4124 mCblkMemory = client->heap()->allocate(size); 4125 if (mCblkMemory != 0) { 4126 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4127 if (mCblk != NULL) { // construct the shared structure in-place. 4128 new(mCblk) audio_track_cblk_t(); 4129 // clear all buffers 4130 mCblk->frameCount = frameCount; 4131 mCblk->sampleRate = sampleRate; 4132// uncomment the following lines to quickly test 32-bit wraparound 4133// mCblk->user = 0xffff0000; 4134// mCblk->server = 0xffff0000; 4135// mCblk->userBase = 0xffff0000; 4136// mCblk->serverBase = 0xffff0000; 4137 mChannelCount = channelCount; 4138 mChannelMask = channelMask; 4139 if (sharedBuffer == 0) { 4140 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4141 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4142 // Force underrun condition to avoid false underrun callback until first data is 4143 // written to buffer (other flags are cleared) 4144 mCblk->flags = CBLK_UNDERRUN_ON; 4145 } else { 4146 mBuffer = sharedBuffer->pointer(); 4147 } 4148 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4149 } 4150 } else { 4151 ALOGE("not enough memory for AudioTrack size=%u", size); 4152 client->heap()->dump("AudioTrack"); 4153 return; 4154 } 4155 } else { 4156 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4157 // construct the shared structure in-place. 4158 new(mCblk) audio_track_cblk_t(); 4159 // clear all buffers 4160 mCblk->frameCount = frameCount; 4161 mCblk->sampleRate = sampleRate; 4162// uncomment the following lines to quickly test 32-bit wraparound 4163// mCblk->user = 0xffff0000; 4164// mCblk->server = 0xffff0000; 4165// mCblk->userBase = 0xffff0000; 4166// mCblk->serverBase = 0xffff0000; 4167 mChannelCount = channelCount; 4168 mChannelMask = channelMask; 4169 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4170 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4171 // Force underrun condition to avoid false underrun callback until first data is 4172 // written to buffer (other flags are cleared) 4173 mCblk->flags = CBLK_UNDERRUN_ON; 4174 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4175 } 4176} 4177 4178AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4179{ 4180 if (mCblk != NULL) { 4181 if (mClient == 0) { 4182 delete mCblk; 4183 } else { 4184 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4185 } 4186 } 4187 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4188 if (mClient != 0) { 4189 // Client destructor must run with AudioFlinger mutex locked 4190 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4191 // If the client's reference count drops to zero, the associated destructor 4192 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4193 // relying on the automatic clear() at end of scope. 4194 mClient.clear(); 4195 } 4196} 4197 4198// AudioBufferProvider interface 4199// getNextBuffer() = 0; 4200// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4201void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4202{ 4203 buffer->raw = NULL; 4204 mFrameCount = buffer->frameCount; 4205 // FIXME See note at getNextBuffer() 4206 (void) step(); // ignore return value of step() 4207 buffer->frameCount = 0; 4208} 4209 4210bool AudioFlinger::ThreadBase::TrackBase::step() { 4211 bool result; 4212 audio_track_cblk_t* cblk = this->cblk(); 4213 4214 result = cblk->stepServer(mFrameCount); 4215 if (!result) { 4216 ALOGV("stepServer failed acquiring cblk mutex"); 4217 mStepServerFailed = true; 4218 } 4219 return result; 4220} 4221 4222void AudioFlinger::ThreadBase::TrackBase::reset() { 4223 audio_track_cblk_t* cblk = this->cblk(); 4224 4225 cblk->user = 0; 4226 cblk->server = 0; 4227 cblk->userBase = 0; 4228 cblk->serverBase = 0; 4229 mStepServerFailed = false; 4230 ALOGV("TrackBase::reset"); 4231} 4232 4233int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4234 return (int)mCblk->sampleRate; 4235} 4236 4237void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4238 audio_track_cblk_t* cblk = this->cblk(); 4239 size_t frameSize = cblk->frameSize; 4240 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4241 int8_t *bufferEnd = bufferStart + frames * frameSize; 4242 4243 // Check validity of returned pointer in case the track control block would have been corrupted. 4244 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4245 "TrackBase::getBuffer buffer out of range:\n" 4246 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4247 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4248 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4249 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4250 4251 return bufferStart; 4252} 4253 4254status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4255{ 4256 mSyncEvents.add(event); 4257 return NO_ERROR; 4258} 4259 4260// ---------------------------------------------------------------------------- 4261 4262// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4263AudioFlinger::PlaybackThread::Track::Track( 4264 PlaybackThread *thread, 4265 const sp<Client>& client, 4266 audio_stream_type_t streamType, 4267 uint32_t sampleRate, 4268 audio_format_t format, 4269 uint32_t channelMask, 4270 int frameCount, 4271 const sp<IMemory>& sharedBuffer, 4272 int sessionId, 4273 IAudioFlinger::track_flags_t flags) 4274 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4275 mMute(false), 4276 mFillingUpStatus(FS_INVALID), 4277 // mRetryCount initialized later when needed 4278 mSharedBuffer(sharedBuffer), 4279 mStreamType(streamType), 4280 mName(-1), // see note below 4281 mMainBuffer(thread->mixBuffer()), 4282 mAuxBuffer(NULL), 4283 mAuxEffectId(0), mHasVolumeController(false), 4284 mPresentationCompleteFrames(0), 4285 mFlags(flags), 4286 mFastIndex(-1), 4287 mUnderrunCount(0), 4288 mCachedVolume(1.0) 4289{ 4290 if (mCblk != NULL) { 4291 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4292 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4293 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4294 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4295 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4296 mCblk->mName = mName; 4297 if (mName < 0) { 4298 ALOGE("no more track names available"); 4299 return; 4300 } 4301 // only allocate a fast track index if we were able to allocate a normal track name 4302 if (flags & IAudioFlinger::TRACK_FAST) { 4303 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4304 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4305 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4306 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4307 // FIXME This is too eager. We allocate a fast track index before the 4308 // fast track becomes active. Since fast tracks are a scarce resource, 4309 // this means we are potentially denying other more important fast tracks from 4310 // being created. It would be better to allocate the index dynamically. 4311 mFastIndex = i; 4312 mCblk->mName = i; 4313 // Read the initial underruns because this field is never cleared by the fast mixer 4314 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4315 thread->mFastTrackAvailMask &= ~(1 << i); 4316 } 4317 } 4318 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4319} 4320 4321AudioFlinger::PlaybackThread::Track::~Track() 4322{ 4323 ALOGV("PlaybackThread::Track destructor"); 4324 sp<ThreadBase> thread = mThread.promote(); 4325 if (thread != 0) { 4326 Mutex::Autolock _l(thread->mLock); 4327 mState = TERMINATED; 4328 } 4329} 4330 4331void AudioFlinger::PlaybackThread::Track::destroy() 4332{ 4333 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4334 // by removing it from mTracks vector, so there is a risk that this Tracks's 4335 // destructor is called. As the destructor needs to lock mLock, 4336 // we must acquire a strong reference on this Track before locking mLock 4337 // here so that the destructor is called only when exiting this function. 4338 // On the other hand, as long as Track::destroy() is only called by 4339 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4340 // this Track with its member mTrack. 4341 sp<Track> keep(this); 4342 { // scope for mLock 4343 sp<ThreadBase> thread = mThread.promote(); 4344 if (thread != 0) { 4345 if (!isOutputTrack()) { 4346 if (mState == ACTIVE || mState == RESUMING) { 4347 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4348 4349#ifdef ADD_BATTERY_DATA 4350 // to track the speaker usage 4351 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4352#endif 4353 } 4354 AudioSystem::releaseOutput(thread->id()); 4355 } 4356 Mutex::Autolock _l(thread->mLock); 4357 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4358 playbackThread->destroyTrack_l(this); 4359 } 4360 } 4361} 4362 4363/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4364{ 4365 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4366 " Server User Main buf Aux Buf Flags Underruns\n"); 4367} 4368 4369void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4370{ 4371 uint32_t vlr = mCblk->getVolumeLR(); 4372 if (isFastTrack()) { 4373 sprintf(buffer, " F %2d", mFastIndex); 4374 } else { 4375 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4376 } 4377 track_state state = mState; 4378 char stateChar; 4379 switch (state) { 4380 case IDLE: 4381 stateChar = 'I'; 4382 break; 4383 case TERMINATED: 4384 stateChar = 'T'; 4385 break; 4386 case STOPPING_1: 4387 stateChar = 's'; 4388 break; 4389 case STOPPING_2: 4390 stateChar = '5'; 4391 break; 4392 case STOPPED: 4393 stateChar = 'S'; 4394 break; 4395 case RESUMING: 4396 stateChar = 'R'; 4397 break; 4398 case ACTIVE: 4399 stateChar = 'A'; 4400 break; 4401 case PAUSING: 4402 stateChar = 'p'; 4403 break; 4404 case PAUSED: 4405 stateChar = 'P'; 4406 break; 4407 case FLUSHED: 4408 stateChar = 'F'; 4409 break; 4410 default: 4411 stateChar = '?'; 4412 break; 4413 } 4414 char nowInUnderrun; 4415 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4416 case UNDERRUN_FULL: 4417 nowInUnderrun = ' '; 4418 break; 4419 case UNDERRUN_PARTIAL: 4420 nowInUnderrun = '<'; 4421 break; 4422 case UNDERRUN_EMPTY: 4423 nowInUnderrun = '*'; 4424 break; 4425 default: 4426 nowInUnderrun = '?'; 4427 break; 4428 } 4429 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4430 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4431 (mClient == 0) ? getpid_cached : mClient->pid(), 4432 mStreamType, 4433 mFormat, 4434 mChannelMask, 4435 mSessionId, 4436 mFrameCount, 4437 mCblk->frameCount, 4438 stateChar, 4439 mMute, 4440 mFillingUpStatus, 4441 mCblk->sampleRate, 4442 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4443 20.0 * log10((vlr >> 16) / 4096.0), 4444 mCblk->server, 4445 mCblk->user, 4446 (int)mMainBuffer, 4447 (int)mAuxBuffer, 4448 mCblk->flags, 4449 mUnderrunCount, 4450 nowInUnderrun); 4451} 4452 4453// AudioBufferProvider interface 4454status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4455 AudioBufferProvider::Buffer* buffer, int64_t pts) 4456{ 4457 audio_track_cblk_t* cblk = this->cblk(); 4458 uint32_t framesReady; 4459 uint32_t framesReq = buffer->frameCount; 4460 4461 // Check if last stepServer failed, try to step now 4462 if (mStepServerFailed) { 4463 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4464 // Since the fast mixer is higher priority than client callback thread, 4465 // it does not result in priority inversion for client. 4466 // But a non-blocking solution would be preferable to avoid 4467 // fast mixer being unable to tryLock(), and 4468 // to avoid the extra context switches if the client wakes up, 4469 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4470 if (!step()) goto getNextBuffer_exit; 4471 ALOGV("stepServer recovered"); 4472 mStepServerFailed = false; 4473 } 4474 4475 // FIXME Same as above 4476 framesReady = cblk->framesReady(); 4477 4478 if (CC_LIKELY(framesReady)) { 4479 uint32_t s = cblk->server; 4480 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4481 4482 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4483 if (framesReq > framesReady) { 4484 framesReq = framesReady; 4485 } 4486 if (framesReq > bufferEnd - s) { 4487 framesReq = bufferEnd - s; 4488 } 4489 4490 buffer->raw = getBuffer(s, framesReq); 4491 if (buffer->raw == NULL) goto getNextBuffer_exit; 4492 4493 buffer->frameCount = framesReq; 4494 return NO_ERROR; 4495 } 4496 4497getNextBuffer_exit: 4498 buffer->raw = NULL; 4499 buffer->frameCount = 0; 4500 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4501 return NOT_ENOUGH_DATA; 4502} 4503 4504// Note that framesReady() takes a mutex on the control block using tryLock(). 4505// This could result in priority inversion if framesReady() is called by the normal mixer, 4506// as the normal mixer thread runs at lower 4507// priority than the client's callback thread: there is a short window within framesReady() 4508// during which the normal mixer could be preempted, and the client callback would block. 4509// Another problem can occur if framesReady() is called by the fast mixer: 4510// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4511// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4512size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4513 return mCblk->framesReady(); 4514} 4515 4516// Don't call for fast tracks; the framesReady() could result in priority inversion 4517bool AudioFlinger::PlaybackThread::Track::isReady() const { 4518 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4519 4520 if (framesReady() >= mCblk->frameCount || 4521 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4522 mFillingUpStatus = FS_FILLED; 4523 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4524 return true; 4525 } 4526 return false; 4527} 4528 4529status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4530 int triggerSession) 4531{ 4532 status_t status = NO_ERROR; 4533 ALOGV("start(%d), calling pid %d session %d", 4534 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4535 4536 sp<ThreadBase> thread = mThread.promote(); 4537 if (thread != 0) { 4538 Mutex::Autolock _l(thread->mLock); 4539 track_state state = mState; 4540 // here the track could be either new, or restarted 4541 // in both cases "unstop" the track 4542 if (mState == PAUSED) { 4543 mState = TrackBase::RESUMING; 4544 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4545 } else { 4546 mState = TrackBase::ACTIVE; 4547 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4548 } 4549 4550 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4551 thread->mLock.unlock(); 4552 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4553 thread->mLock.lock(); 4554 4555#ifdef ADD_BATTERY_DATA 4556 // to track the speaker usage 4557 if (status == NO_ERROR) { 4558 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4559 } 4560#endif 4561 } 4562 if (status == NO_ERROR) { 4563 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4564 playbackThread->addTrack_l(this); 4565 } else { 4566 mState = state; 4567 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4568 } 4569 } else { 4570 status = BAD_VALUE; 4571 } 4572 return status; 4573} 4574 4575void AudioFlinger::PlaybackThread::Track::stop() 4576{ 4577 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4578 sp<ThreadBase> thread = mThread.promote(); 4579 if (thread != 0) { 4580 Mutex::Autolock _l(thread->mLock); 4581 track_state state = mState; 4582 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4583 // If the track is not active (PAUSED and buffers full), flush buffers 4584 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4585 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4586 reset(); 4587 mState = STOPPED; 4588 } else if (!isFastTrack()) { 4589 mState = STOPPED; 4590 } else { 4591 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4592 // and then to STOPPED and reset() when presentation is complete 4593 mState = STOPPING_1; 4594 } 4595 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4596 } 4597 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4598 thread->mLock.unlock(); 4599 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4600 thread->mLock.lock(); 4601 4602#ifdef ADD_BATTERY_DATA 4603 // to track the speaker usage 4604 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4605#endif 4606 } 4607 } 4608} 4609 4610void AudioFlinger::PlaybackThread::Track::pause() 4611{ 4612 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4613 sp<ThreadBase> thread = mThread.promote(); 4614 if (thread != 0) { 4615 Mutex::Autolock _l(thread->mLock); 4616 if (mState == ACTIVE || mState == RESUMING) { 4617 mState = PAUSING; 4618 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4619 if (!isOutputTrack()) { 4620 thread->mLock.unlock(); 4621 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4622 thread->mLock.lock(); 4623 4624#ifdef ADD_BATTERY_DATA 4625 // to track the speaker usage 4626 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4627#endif 4628 } 4629 } 4630 } 4631} 4632 4633void AudioFlinger::PlaybackThread::Track::flush() 4634{ 4635 ALOGV("flush(%d)", mName); 4636 sp<ThreadBase> thread = mThread.promote(); 4637 if (thread != 0) { 4638 Mutex::Autolock _l(thread->mLock); 4639 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4640 mState != PAUSING) { 4641 return; 4642 } 4643 // No point remaining in PAUSED state after a flush => go to 4644 // FLUSHED state 4645 mState = FLUSHED; 4646 // do not reset the track if it is still in the process of being stopped or paused. 4647 // this will be done by prepareTracks_l() when the track is stopped. 4648 // prepareTracks_l() will see mState == FLUSHED, then 4649 // remove from active track list, reset(), and trigger presentation complete 4650 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4651 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4652 reset(); 4653 } 4654 } 4655} 4656 4657void AudioFlinger::PlaybackThread::Track::reset() 4658{ 4659 // Do not reset twice to avoid discarding data written just after a flush and before 4660 // the audioflinger thread detects the track is stopped. 4661 if (!mResetDone) { 4662 TrackBase::reset(); 4663 // Force underrun condition to avoid false underrun callback until first data is 4664 // written to buffer 4665 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4666 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4667 mFillingUpStatus = FS_FILLING; 4668 mResetDone = true; 4669 if (mState == FLUSHED) { 4670 mState = IDLE; 4671 } 4672 } 4673} 4674 4675void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4676{ 4677 mMute = muted; 4678} 4679 4680status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4681{ 4682 status_t status = DEAD_OBJECT; 4683 sp<ThreadBase> thread = mThread.promote(); 4684 if (thread != 0) { 4685 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4686 sp<AudioFlinger> af = mClient->audioFlinger(); 4687 4688 Mutex::Autolock _l(af->mLock); 4689 4690 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4691 4692 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4693 Mutex::Autolock _dl(playbackThread->mLock); 4694 Mutex::Autolock _sl(srcThread->mLock); 4695 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4696 if (chain == 0) { 4697 return INVALID_OPERATION; 4698 } 4699 4700 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4701 if (effect == 0) { 4702 return INVALID_OPERATION; 4703 } 4704 srcThread->removeEffect_l(effect); 4705 playbackThread->addEffect_l(effect); 4706 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4707 if (effect->state() == EffectModule::ACTIVE || 4708 effect->state() == EffectModule::STOPPING) { 4709 effect->start(); 4710 } 4711 4712 sp<EffectChain> dstChain = effect->chain().promote(); 4713 if (dstChain == 0) { 4714 srcThread->addEffect_l(effect); 4715 return INVALID_OPERATION; 4716 } 4717 AudioSystem::unregisterEffect(effect->id()); 4718 AudioSystem::registerEffect(&effect->desc(), 4719 srcThread->id(), 4720 dstChain->strategy(), 4721 AUDIO_SESSION_OUTPUT_MIX, 4722 effect->id()); 4723 } 4724 status = playbackThread->attachAuxEffect(this, EffectId); 4725 } 4726 return status; 4727} 4728 4729void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4730{ 4731 mAuxEffectId = EffectId; 4732 mAuxBuffer = buffer; 4733} 4734 4735bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4736 size_t audioHalFrames) 4737{ 4738 // a track is considered presented when the total number of frames written to audio HAL 4739 // corresponds to the number of frames written when presentationComplete() is called for the 4740 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4741 if (mPresentationCompleteFrames == 0) { 4742 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4743 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4744 mPresentationCompleteFrames, audioHalFrames); 4745 } 4746 if (framesWritten >= mPresentationCompleteFrames) { 4747 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4748 mSessionId, framesWritten); 4749 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4750 return true; 4751 } 4752 return false; 4753} 4754 4755void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4756{ 4757 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4758 if (mSyncEvents[i]->type() == type) { 4759 mSyncEvents[i]->trigger(); 4760 mSyncEvents.removeAt(i); 4761 i--; 4762 } 4763 } 4764} 4765 4766// implement VolumeBufferProvider interface 4767 4768uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4769{ 4770 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4771 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4772 uint32_t vlr = mCblk->getVolumeLR(); 4773 uint32_t vl = vlr & 0xFFFF; 4774 uint32_t vr = vlr >> 16; 4775 // track volumes come from shared memory, so can't be trusted and must be clamped 4776 if (vl > MAX_GAIN_INT) { 4777 vl = MAX_GAIN_INT; 4778 } 4779 if (vr > MAX_GAIN_INT) { 4780 vr = MAX_GAIN_INT; 4781 } 4782 // now apply the cached master volume and stream type volume; 4783 // this is trusted but lacks any synchronization or barrier so may be stale 4784 float v = mCachedVolume; 4785 vl *= v; 4786 vr *= v; 4787 // re-combine into U4.16 4788 vlr = (vr << 16) | (vl & 0xFFFF); 4789 // FIXME look at mute, pause, and stop flags 4790 return vlr; 4791} 4792 4793status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4794{ 4795 if (mState == TERMINATED || mState == PAUSED || 4796 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4797 (mState == STOPPED)))) { 4798 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4799 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4800 event->cancel(); 4801 return INVALID_OPERATION; 4802 } 4803 TrackBase::setSyncEvent(event); 4804 return NO_ERROR; 4805} 4806 4807// timed audio tracks 4808 4809sp<AudioFlinger::PlaybackThread::TimedTrack> 4810AudioFlinger::PlaybackThread::TimedTrack::create( 4811 PlaybackThread *thread, 4812 const sp<Client>& client, 4813 audio_stream_type_t streamType, 4814 uint32_t sampleRate, 4815 audio_format_t format, 4816 uint32_t channelMask, 4817 int frameCount, 4818 const sp<IMemory>& sharedBuffer, 4819 int sessionId) { 4820 if (!client->reserveTimedTrack()) 4821 return 0; 4822 4823 return new TimedTrack( 4824 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4825 sharedBuffer, sessionId); 4826} 4827 4828AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4829 PlaybackThread *thread, 4830 const sp<Client>& client, 4831 audio_stream_type_t streamType, 4832 uint32_t sampleRate, 4833 audio_format_t format, 4834 uint32_t channelMask, 4835 int frameCount, 4836 const sp<IMemory>& sharedBuffer, 4837 int sessionId) 4838 : Track(thread, client, streamType, sampleRate, format, channelMask, 4839 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4840 mQueueHeadInFlight(false), 4841 mTrimQueueHeadOnRelease(false), 4842 mFramesPendingInQueue(0), 4843 mTimedSilenceBuffer(NULL), 4844 mTimedSilenceBufferSize(0), 4845 mTimedAudioOutputOnTime(false), 4846 mMediaTimeTransformValid(false) 4847{ 4848 LocalClock lc; 4849 mLocalTimeFreq = lc.getLocalFreq(); 4850 4851 mLocalTimeToSampleTransform.a_zero = 0; 4852 mLocalTimeToSampleTransform.b_zero = 0; 4853 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4854 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4855 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4856 &mLocalTimeToSampleTransform.a_to_b_denom); 4857 4858 mMediaTimeToSampleTransform.a_zero = 0; 4859 mMediaTimeToSampleTransform.b_zero = 0; 4860 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4861 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4862 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4863 &mMediaTimeToSampleTransform.a_to_b_denom); 4864} 4865 4866AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4867 mClient->releaseTimedTrack(); 4868 delete [] mTimedSilenceBuffer; 4869} 4870 4871status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4872 size_t size, sp<IMemory>* buffer) { 4873 4874 Mutex::Autolock _l(mTimedBufferQueueLock); 4875 4876 trimTimedBufferQueue_l(); 4877 4878 // lazily initialize the shared memory heap for timed buffers 4879 if (mTimedMemoryDealer == NULL) { 4880 const int kTimedBufferHeapSize = 512 << 10; 4881 4882 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4883 "AudioFlingerTimed"); 4884 if (mTimedMemoryDealer == NULL) 4885 return NO_MEMORY; 4886 } 4887 4888 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4889 if (newBuffer == NULL) { 4890 newBuffer = mTimedMemoryDealer->allocate(size); 4891 if (newBuffer == NULL) 4892 return NO_MEMORY; 4893 } 4894 4895 *buffer = newBuffer; 4896 return NO_ERROR; 4897} 4898 4899// caller must hold mTimedBufferQueueLock 4900void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4901 int64_t mediaTimeNow; 4902 { 4903 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4904 if (!mMediaTimeTransformValid) 4905 return; 4906 4907 int64_t targetTimeNow; 4908 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4909 ? mCCHelper.getCommonTime(&targetTimeNow) 4910 : mCCHelper.getLocalTime(&targetTimeNow); 4911 4912 if (OK != res) 4913 return; 4914 4915 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4916 &mediaTimeNow)) { 4917 return; 4918 } 4919 } 4920 4921 size_t trimEnd; 4922 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4923 int64_t bufEnd; 4924 4925 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4926 // We have a next buffer. Just use its PTS as the PTS of the frame 4927 // following the last frame in this buffer. If the stream is sparse 4928 // (ie, there are deliberate gaps left in the stream which should be 4929 // filled with silence by the TimedAudioTrack), then this can result 4930 // in one extra buffer being left un-trimmed when it could have 4931 // been. In general, this is not typical, and we would rather 4932 // optimized away the TS calculation below for the more common case 4933 // where PTSes are contiguous. 4934 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4935 } else { 4936 // We have no next buffer. Compute the PTS of the frame following 4937 // the last frame in this buffer by computing the duration of of 4938 // this frame in media time units and adding it to the PTS of the 4939 // buffer. 4940 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4941 / mCblk->frameSize; 4942 4943 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4944 &bufEnd)) { 4945 ALOGE("Failed to convert frame count of %lld to media time" 4946 " duration" " (scale factor %d/%u) in %s", 4947 frameCount, 4948 mMediaTimeToSampleTransform.a_to_b_numer, 4949 mMediaTimeToSampleTransform.a_to_b_denom, 4950 __PRETTY_FUNCTION__); 4951 break; 4952 } 4953 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4954 } 4955 4956 if (bufEnd > mediaTimeNow) 4957 break; 4958 4959 // Is the buffer we want to use in the middle of a mix operation right 4960 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4961 // from the mixer which should be coming back shortly. 4962 if (!trimEnd && mQueueHeadInFlight) { 4963 mTrimQueueHeadOnRelease = true; 4964 } 4965 } 4966 4967 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4968 if (trimStart < trimEnd) { 4969 // Update the bookkeeping for framesReady() 4970 for (size_t i = trimStart; i < trimEnd; ++i) { 4971 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4972 } 4973 4974 // Now actually remove the buffers from the queue. 4975 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4976 } 4977} 4978 4979void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4980 const char* logTag) { 4981 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4982 "%s called (reason \"%s\"), but timed buffer queue has no" 4983 " elements to trim.", __FUNCTION__, logTag); 4984 4985 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4986 mTimedBufferQueue.removeAt(0); 4987} 4988 4989void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4990 const TimedBuffer& buf, 4991 const char* logTag) { 4992 uint32_t bufBytes = buf.buffer()->size(); 4993 uint32_t consumedAlready = buf.position(); 4994 4995 ALOG_ASSERT(consumedAlready <= bufBytes, 4996 "Bad bookkeeping while updating frames pending. Timed buffer is" 4997 " only %u bytes long, but claims to have consumed %u" 4998 " bytes. (update reason: \"%s\")", 4999 bufBytes, consumedAlready, logTag); 5000 5001 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 5002 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5003 "Bad bookkeeping while updating frames pending. Should have at" 5004 " least %u queued frames, but we think we have only %u. (update" 5005 " reason: \"%s\")", 5006 bufFrames, mFramesPendingInQueue, logTag); 5007 5008 mFramesPendingInQueue -= bufFrames; 5009} 5010 5011status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5012 const sp<IMemory>& buffer, int64_t pts) { 5013 5014 { 5015 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5016 if (!mMediaTimeTransformValid) 5017 return INVALID_OPERATION; 5018 } 5019 5020 Mutex::Autolock _l(mTimedBufferQueueLock); 5021 5022 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5023 mFramesPendingInQueue += bufFrames; 5024 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5025 5026 return NO_ERROR; 5027} 5028 5029status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5030 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5031 5032 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5033 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5034 target); 5035 5036 if (!(target == TimedAudioTrack::LOCAL_TIME || 5037 target == TimedAudioTrack::COMMON_TIME)) { 5038 return BAD_VALUE; 5039 } 5040 5041 Mutex::Autolock lock(mMediaTimeTransformLock); 5042 mMediaTimeTransform = xform; 5043 mMediaTimeTransformTarget = target; 5044 mMediaTimeTransformValid = true; 5045 5046 return NO_ERROR; 5047} 5048 5049#define min(a, b) ((a) < (b) ? (a) : (b)) 5050 5051// implementation of getNextBuffer for tracks whose buffers have timestamps 5052status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5053 AudioBufferProvider::Buffer* buffer, int64_t pts) 5054{ 5055 if (pts == AudioBufferProvider::kInvalidPTS) { 5056 buffer->raw = NULL; 5057 buffer->frameCount = 0; 5058 mTimedAudioOutputOnTime = false; 5059 return INVALID_OPERATION; 5060 } 5061 5062 Mutex::Autolock _l(mTimedBufferQueueLock); 5063 5064 ALOG_ASSERT(!mQueueHeadInFlight, 5065 "getNextBuffer called without releaseBuffer!"); 5066 5067 while (true) { 5068 5069 // if we have no timed buffers, then fail 5070 if (mTimedBufferQueue.isEmpty()) { 5071 buffer->raw = NULL; 5072 buffer->frameCount = 0; 5073 return NOT_ENOUGH_DATA; 5074 } 5075 5076 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5077 5078 // calculate the PTS of the head of the timed buffer queue expressed in 5079 // local time 5080 int64_t headLocalPTS; 5081 { 5082 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5083 5084 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5085 5086 if (mMediaTimeTransform.a_to_b_denom == 0) { 5087 // the transform represents a pause, so yield silence 5088 timedYieldSilence_l(buffer->frameCount, buffer); 5089 return NO_ERROR; 5090 } 5091 5092 int64_t transformedPTS; 5093 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5094 &transformedPTS)) { 5095 // the transform failed. this shouldn't happen, but if it does 5096 // then just drop this buffer 5097 ALOGW("timedGetNextBuffer transform failed"); 5098 buffer->raw = NULL; 5099 buffer->frameCount = 0; 5100 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5101 return NO_ERROR; 5102 } 5103 5104 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5105 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5106 &headLocalPTS)) { 5107 buffer->raw = NULL; 5108 buffer->frameCount = 0; 5109 return INVALID_OPERATION; 5110 } 5111 } else { 5112 headLocalPTS = transformedPTS; 5113 } 5114 } 5115 5116 // adjust the head buffer's PTS to reflect the portion of the head buffer 5117 // that has already been consumed 5118 int64_t effectivePTS = headLocalPTS + 5119 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5120 5121 // Calculate the delta in samples between the head of the input buffer 5122 // queue and the start of the next output buffer that will be written. 5123 // If the transformation fails because of over or underflow, it means 5124 // that the sample's position in the output stream is so far out of 5125 // whack that it should just be dropped. 5126 int64_t sampleDelta; 5127 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5128 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5129 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5130 " mix"); 5131 continue; 5132 } 5133 if (!mLocalTimeToSampleTransform.doForwardTransform( 5134 (effectivePTS - pts) << 32, &sampleDelta)) { 5135 ALOGV("*** too late during sample rate transform: dropped buffer"); 5136 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5137 continue; 5138 } 5139 5140 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5141 " sampleDelta=[%d.%08x]", 5142 head.pts(), head.position(), pts, 5143 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5144 + (sampleDelta >> 32)), 5145 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5146 5147 // if the delta between the ideal placement for the next input sample and 5148 // the current output position is within this threshold, then we will 5149 // concatenate the next input samples to the previous output 5150 const int64_t kSampleContinuityThreshold = 5151 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5152 5153 // if this is the first buffer of audio that we're emitting from this track 5154 // then it should be almost exactly on time. 5155 const int64_t kSampleStartupThreshold = 1LL << 32; 5156 5157 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5158 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5159 // the next input is close enough to being on time, so concatenate it 5160 // with the last output 5161 timedYieldSamples_l(buffer); 5162 5163 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5164 head.position(), buffer->frameCount); 5165 return NO_ERROR; 5166 } 5167 5168 // Looks like our output is not on time. Reset our on timed status. 5169 // Next time we mix samples from our input queue, then should be within 5170 // the StartupThreshold. 5171 mTimedAudioOutputOnTime = false; 5172 if (sampleDelta > 0) { 5173 // the gap between the current output position and the proper start of 5174 // the next input sample is too big, so fill it with silence 5175 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5176 5177 timedYieldSilence_l(framesUntilNextInput, buffer); 5178 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5179 return NO_ERROR; 5180 } else { 5181 // the next input sample is late 5182 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5183 size_t onTimeSamplePosition = 5184 head.position() + lateFrames * mCblk->frameSize; 5185 5186 if (onTimeSamplePosition > head.buffer()->size()) { 5187 // all the remaining samples in the head are too late, so 5188 // drop it and move on 5189 ALOGV("*** too late: dropped buffer"); 5190 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5191 continue; 5192 } else { 5193 // skip over the late samples 5194 head.setPosition(onTimeSamplePosition); 5195 5196 // yield the available samples 5197 timedYieldSamples_l(buffer); 5198 5199 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5200 return NO_ERROR; 5201 } 5202 } 5203 } 5204} 5205 5206// Yield samples from the timed buffer queue head up to the given output 5207// buffer's capacity. 5208// 5209// Caller must hold mTimedBufferQueueLock 5210void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5211 AudioBufferProvider::Buffer* buffer) { 5212 5213 const TimedBuffer& head = mTimedBufferQueue[0]; 5214 5215 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5216 head.position()); 5217 5218 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5219 mCblk->frameSize); 5220 size_t framesRequested = buffer->frameCount; 5221 buffer->frameCount = min(framesLeftInHead, framesRequested); 5222 5223 mQueueHeadInFlight = true; 5224 mTimedAudioOutputOnTime = true; 5225} 5226 5227// Yield samples of silence up to the given output buffer's capacity 5228// 5229// Caller must hold mTimedBufferQueueLock 5230void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5231 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5232 5233 // lazily allocate a buffer filled with silence 5234 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5235 delete [] mTimedSilenceBuffer; 5236 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5237 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5238 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5239 } 5240 5241 buffer->raw = mTimedSilenceBuffer; 5242 size_t framesRequested = buffer->frameCount; 5243 buffer->frameCount = min(numFrames, framesRequested); 5244 5245 mTimedAudioOutputOnTime = false; 5246} 5247 5248// AudioBufferProvider interface 5249void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5250 AudioBufferProvider::Buffer* buffer) { 5251 5252 Mutex::Autolock _l(mTimedBufferQueueLock); 5253 5254 // If the buffer which was just released is part of the buffer at the head 5255 // of the queue, be sure to update the amt of the buffer which has been 5256 // consumed. If the buffer being returned is not part of the head of the 5257 // queue, its either because the buffer is part of the silence buffer, or 5258 // because the head of the timed queue was trimmed after the mixer called 5259 // getNextBuffer but before the mixer called releaseBuffer. 5260 if (buffer->raw == mTimedSilenceBuffer) { 5261 ALOG_ASSERT(!mQueueHeadInFlight, 5262 "Queue head in flight during release of silence buffer!"); 5263 goto done; 5264 } 5265 5266 ALOG_ASSERT(mQueueHeadInFlight, 5267 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5268 " head in flight."); 5269 5270 if (mTimedBufferQueue.size()) { 5271 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5272 5273 void* start = head.buffer()->pointer(); 5274 void* end = reinterpret_cast<void*>( 5275 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5276 + head.buffer()->size()); 5277 5278 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5279 "released buffer not within the head of the timed buffer" 5280 " queue; qHead = [%p, %p], released buffer = %p", 5281 start, end, buffer->raw); 5282 5283 head.setPosition(head.position() + 5284 (buffer->frameCount * mCblk->frameSize)); 5285 mQueueHeadInFlight = false; 5286 5287 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5288 "Bad bookkeeping during releaseBuffer! Should have at" 5289 " least %u queued frames, but we think we have only %u", 5290 buffer->frameCount, mFramesPendingInQueue); 5291 5292 mFramesPendingInQueue -= buffer->frameCount; 5293 5294 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5295 || mTrimQueueHeadOnRelease) { 5296 trimTimedBufferQueueHead_l("releaseBuffer"); 5297 mTrimQueueHeadOnRelease = false; 5298 } 5299 } else { 5300 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5301 " buffers in the timed buffer queue"); 5302 } 5303 5304done: 5305 buffer->raw = 0; 5306 buffer->frameCount = 0; 5307} 5308 5309size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5310 Mutex::Autolock _l(mTimedBufferQueueLock); 5311 return mFramesPendingInQueue; 5312} 5313 5314AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5315 : mPTS(0), mPosition(0) {} 5316 5317AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5318 const sp<IMemory>& buffer, int64_t pts) 5319 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5320 5321// ---------------------------------------------------------------------------- 5322 5323// RecordTrack constructor must be called with AudioFlinger::mLock held 5324AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5325 RecordThread *thread, 5326 const sp<Client>& client, 5327 uint32_t sampleRate, 5328 audio_format_t format, 5329 uint32_t channelMask, 5330 int frameCount, 5331 int sessionId) 5332 : TrackBase(thread, client, sampleRate, format, 5333 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5334 mOverflow(false) 5335{ 5336 if (mCblk != NULL) { 5337 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5338 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5339 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5340 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5341 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5342 } else { 5343 mCblk->frameSize = sizeof(int8_t); 5344 } 5345 } 5346} 5347 5348AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5349{ 5350 sp<ThreadBase> thread = mThread.promote(); 5351 if (thread != 0) { 5352 AudioSystem::releaseInput(thread->id()); 5353 } 5354} 5355 5356// AudioBufferProvider interface 5357status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5358{ 5359 audio_track_cblk_t* cblk = this->cblk(); 5360 uint32_t framesAvail; 5361 uint32_t framesReq = buffer->frameCount; 5362 5363 // Check if last stepServer failed, try to step now 5364 if (mStepServerFailed) { 5365 if (!step()) goto getNextBuffer_exit; 5366 ALOGV("stepServer recovered"); 5367 mStepServerFailed = false; 5368 } 5369 5370 framesAvail = cblk->framesAvailable_l(); 5371 5372 if (CC_LIKELY(framesAvail)) { 5373 uint32_t s = cblk->server; 5374 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5375 5376 if (framesReq > framesAvail) { 5377 framesReq = framesAvail; 5378 } 5379 if (framesReq > bufferEnd - s) { 5380 framesReq = bufferEnd - s; 5381 } 5382 5383 buffer->raw = getBuffer(s, framesReq); 5384 if (buffer->raw == NULL) goto getNextBuffer_exit; 5385 5386 buffer->frameCount = framesReq; 5387 return NO_ERROR; 5388 } 5389 5390getNextBuffer_exit: 5391 buffer->raw = NULL; 5392 buffer->frameCount = 0; 5393 return NOT_ENOUGH_DATA; 5394} 5395 5396status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5397 int triggerSession) 5398{ 5399 sp<ThreadBase> thread = mThread.promote(); 5400 if (thread != 0) { 5401 RecordThread *recordThread = (RecordThread *)thread.get(); 5402 return recordThread->start(this, event, triggerSession); 5403 } else { 5404 return BAD_VALUE; 5405 } 5406} 5407 5408void AudioFlinger::RecordThread::RecordTrack::stop() 5409{ 5410 sp<ThreadBase> thread = mThread.promote(); 5411 if (thread != 0) { 5412 RecordThread *recordThread = (RecordThread *)thread.get(); 5413 recordThread->stop(this); 5414 TrackBase::reset(); 5415 // Force overrun condition to avoid false overrun callback until first data is 5416 // read from buffer 5417 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5418 } 5419} 5420 5421void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5422{ 5423 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5424 (mClient == 0) ? getpid_cached : mClient->pid(), 5425 mFormat, 5426 mChannelMask, 5427 mSessionId, 5428 mFrameCount, 5429 mState, 5430 mCblk->sampleRate, 5431 mCblk->server, 5432 mCblk->user); 5433} 5434 5435 5436// ---------------------------------------------------------------------------- 5437 5438AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5439 PlaybackThread *playbackThread, 5440 DuplicatingThread *sourceThread, 5441 uint32_t sampleRate, 5442 audio_format_t format, 5443 uint32_t channelMask, 5444 int frameCount) 5445 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5446 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5447 mActive(false), mSourceThread(sourceThread) 5448{ 5449 5450 if (mCblk != NULL) { 5451 mCblk->flags |= CBLK_DIRECTION_OUT; 5452 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5453 mOutBuffer.frameCount = 0; 5454 playbackThread->mTracks.add(this); 5455 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5456 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5457 mCblk, mBuffer, mCblk->buffers, 5458 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5459 } else { 5460 ALOGW("Error creating output track on thread %p", playbackThread); 5461 } 5462} 5463 5464AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5465{ 5466 clearBufferQueue(); 5467} 5468 5469status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5470 int triggerSession) 5471{ 5472 status_t status = Track::start(event, triggerSession); 5473 if (status != NO_ERROR) { 5474 return status; 5475 } 5476 5477 mActive = true; 5478 mRetryCount = 127; 5479 return status; 5480} 5481 5482void AudioFlinger::PlaybackThread::OutputTrack::stop() 5483{ 5484 Track::stop(); 5485 clearBufferQueue(); 5486 mOutBuffer.frameCount = 0; 5487 mActive = false; 5488} 5489 5490bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5491{ 5492 Buffer *pInBuffer; 5493 Buffer inBuffer; 5494 uint32_t channelCount = mChannelCount; 5495 bool outputBufferFull = false; 5496 inBuffer.frameCount = frames; 5497 inBuffer.i16 = data; 5498 5499 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5500 5501 if (!mActive && frames != 0) { 5502 start(); 5503 sp<ThreadBase> thread = mThread.promote(); 5504 if (thread != 0) { 5505 MixerThread *mixerThread = (MixerThread *)thread.get(); 5506 if (mCblk->frameCount > frames){ 5507 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5508 uint32_t startFrames = (mCblk->frameCount - frames); 5509 pInBuffer = new Buffer; 5510 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5511 pInBuffer->frameCount = startFrames; 5512 pInBuffer->i16 = pInBuffer->mBuffer; 5513 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5514 mBufferQueue.add(pInBuffer); 5515 } else { 5516 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5517 } 5518 } 5519 } 5520 } 5521 5522 while (waitTimeLeftMs) { 5523 // First write pending buffers, then new data 5524 if (mBufferQueue.size()) { 5525 pInBuffer = mBufferQueue.itemAt(0); 5526 } else { 5527 pInBuffer = &inBuffer; 5528 } 5529 5530 if (pInBuffer->frameCount == 0) { 5531 break; 5532 } 5533 5534 if (mOutBuffer.frameCount == 0) { 5535 mOutBuffer.frameCount = pInBuffer->frameCount; 5536 nsecs_t startTime = systemTime(); 5537 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5538 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5539 outputBufferFull = true; 5540 break; 5541 } 5542 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5543 if (waitTimeLeftMs >= waitTimeMs) { 5544 waitTimeLeftMs -= waitTimeMs; 5545 } else { 5546 waitTimeLeftMs = 0; 5547 } 5548 } 5549 5550 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5551 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5552 mCblk->stepUser(outFrames); 5553 pInBuffer->frameCount -= outFrames; 5554 pInBuffer->i16 += outFrames * channelCount; 5555 mOutBuffer.frameCount -= outFrames; 5556 mOutBuffer.i16 += outFrames * channelCount; 5557 5558 if (pInBuffer->frameCount == 0) { 5559 if (mBufferQueue.size()) { 5560 mBufferQueue.removeAt(0); 5561 delete [] pInBuffer->mBuffer; 5562 delete pInBuffer; 5563 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5564 } else { 5565 break; 5566 } 5567 } 5568 } 5569 5570 // If we could not write all frames, allocate a buffer and queue it for next time. 5571 if (inBuffer.frameCount) { 5572 sp<ThreadBase> thread = mThread.promote(); 5573 if (thread != 0 && !thread->standby()) { 5574 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5575 pInBuffer = new Buffer; 5576 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5577 pInBuffer->frameCount = inBuffer.frameCount; 5578 pInBuffer->i16 = pInBuffer->mBuffer; 5579 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5580 mBufferQueue.add(pInBuffer); 5581 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5582 } else { 5583 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5584 } 5585 } 5586 } 5587 5588 // Calling write() with a 0 length buffer, means that no more data will be written: 5589 // If no more buffers are pending, fill output track buffer to make sure it is started 5590 // by output mixer. 5591 if (frames == 0 && mBufferQueue.size() == 0) { 5592 if (mCblk->user < mCblk->frameCount) { 5593 frames = mCblk->frameCount - mCblk->user; 5594 pInBuffer = new Buffer; 5595 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5596 pInBuffer->frameCount = frames; 5597 pInBuffer->i16 = pInBuffer->mBuffer; 5598 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5599 mBufferQueue.add(pInBuffer); 5600 } else if (mActive) { 5601 stop(); 5602 } 5603 } 5604 5605 return outputBufferFull; 5606} 5607 5608status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5609{ 5610 int active; 5611 status_t result; 5612 audio_track_cblk_t* cblk = mCblk; 5613 uint32_t framesReq = buffer->frameCount; 5614 5615// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5616 buffer->frameCount = 0; 5617 5618 uint32_t framesAvail = cblk->framesAvailable(); 5619 5620 5621 if (framesAvail == 0) { 5622 Mutex::Autolock _l(cblk->lock); 5623 goto start_loop_here; 5624 while (framesAvail == 0) { 5625 active = mActive; 5626 if (CC_UNLIKELY(!active)) { 5627 ALOGV("Not active and NO_MORE_BUFFERS"); 5628 return NO_MORE_BUFFERS; 5629 } 5630 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5631 if (result != NO_ERROR) { 5632 return NO_MORE_BUFFERS; 5633 } 5634 // read the server count again 5635 start_loop_here: 5636 framesAvail = cblk->framesAvailable_l(); 5637 } 5638 } 5639 5640// if (framesAvail < framesReq) { 5641// return NO_MORE_BUFFERS; 5642// } 5643 5644 if (framesReq > framesAvail) { 5645 framesReq = framesAvail; 5646 } 5647 5648 uint32_t u = cblk->user; 5649 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5650 5651 if (framesReq > bufferEnd - u) { 5652 framesReq = bufferEnd - u; 5653 } 5654 5655 buffer->frameCount = framesReq; 5656 buffer->raw = (void *)cblk->buffer(u); 5657 return NO_ERROR; 5658} 5659 5660 5661void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5662{ 5663 size_t size = mBufferQueue.size(); 5664 5665 for (size_t i = 0; i < size; i++) { 5666 Buffer *pBuffer = mBufferQueue.itemAt(i); 5667 delete [] pBuffer->mBuffer; 5668 delete pBuffer; 5669 } 5670 mBufferQueue.clear(); 5671} 5672 5673// ---------------------------------------------------------------------------- 5674 5675AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5676 : RefBase(), 5677 mAudioFlinger(audioFlinger), 5678 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5679 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5680 mPid(pid), 5681 mTimedTrackCount(0) 5682{ 5683 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5684} 5685 5686// Client destructor must be called with AudioFlinger::mLock held 5687AudioFlinger::Client::~Client() 5688{ 5689 mAudioFlinger->removeClient_l(mPid); 5690} 5691 5692sp<MemoryDealer> AudioFlinger::Client::heap() const 5693{ 5694 return mMemoryDealer; 5695} 5696 5697// Reserve one of the limited slots for a timed audio track associated 5698// with this client 5699bool AudioFlinger::Client::reserveTimedTrack() 5700{ 5701 const int kMaxTimedTracksPerClient = 4; 5702 5703 Mutex::Autolock _l(mTimedTrackLock); 5704 5705 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5706 ALOGW("can not create timed track - pid %d has exceeded the limit", 5707 mPid); 5708 return false; 5709 } 5710 5711 mTimedTrackCount++; 5712 return true; 5713} 5714 5715// Release a slot for a timed audio track 5716void AudioFlinger::Client::releaseTimedTrack() 5717{ 5718 Mutex::Autolock _l(mTimedTrackLock); 5719 mTimedTrackCount--; 5720} 5721 5722// ---------------------------------------------------------------------------- 5723 5724AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5725 const sp<IAudioFlingerClient>& client, 5726 pid_t pid) 5727 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5728{ 5729} 5730 5731AudioFlinger::NotificationClient::~NotificationClient() 5732{ 5733} 5734 5735void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5736{ 5737 sp<NotificationClient> keep(this); 5738 mAudioFlinger->removeNotificationClient(mPid); 5739} 5740 5741// ---------------------------------------------------------------------------- 5742 5743AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5744 : BnAudioTrack(), 5745 mTrack(track) 5746{ 5747} 5748 5749AudioFlinger::TrackHandle::~TrackHandle() { 5750 // just stop the track on deletion, associated resources 5751 // will be freed from the main thread once all pending buffers have 5752 // been played. Unless it's not in the active track list, in which 5753 // case we free everything now... 5754 mTrack->destroy(); 5755} 5756 5757sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5758 return mTrack->getCblk(); 5759} 5760 5761status_t AudioFlinger::TrackHandle::start() { 5762 return mTrack->start(); 5763} 5764 5765void AudioFlinger::TrackHandle::stop() { 5766 mTrack->stop(); 5767} 5768 5769void AudioFlinger::TrackHandle::flush() { 5770 mTrack->flush(); 5771} 5772 5773void AudioFlinger::TrackHandle::mute(bool e) { 5774 mTrack->mute(e); 5775} 5776 5777void AudioFlinger::TrackHandle::pause() { 5778 mTrack->pause(); 5779} 5780 5781status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5782{ 5783 return mTrack->attachAuxEffect(EffectId); 5784} 5785 5786status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5787 sp<IMemory>* buffer) { 5788 if (!mTrack->isTimedTrack()) 5789 return INVALID_OPERATION; 5790 5791 PlaybackThread::TimedTrack* tt = 5792 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5793 return tt->allocateTimedBuffer(size, buffer); 5794} 5795 5796status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5797 int64_t pts) { 5798 if (!mTrack->isTimedTrack()) 5799 return INVALID_OPERATION; 5800 5801 PlaybackThread::TimedTrack* tt = 5802 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5803 return tt->queueTimedBuffer(buffer, pts); 5804} 5805 5806status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5807 const LinearTransform& xform, int target) { 5808 5809 if (!mTrack->isTimedTrack()) 5810 return INVALID_OPERATION; 5811 5812 PlaybackThread::TimedTrack* tt = 5813 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5814 return tt->setMediaTimeTransform( 5815 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5816} 5817 5818status_t AudioFlinger::TrackHandle::onTransact( 5819 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5820{ 5821 return BnAudioTrack::onTransact(code, data, reply, flags); 5822} 5823 5824// ---------------------------------------------------------------------------- 5825 5826sp<IAudioRecord> AudioFlinger::openRecord( 5827 pid_t pid, 5828 audio_io_handle_t input, 5829 uint32_t sampleRate, 5830 audio_format_t format, 5831 uint32_t channelMask, 5832 int frameCount, 5833 IAudioFlinger::track_flags_t flags, 5834 int *sessionId, 5835 status_t *status) 5836{ 5837 sp<RecordThread::RecordTrack> recordTrack; 5838 sp<RecordHandle> recordHandle; 5839 sp<Client> client; 5840 status_t lStatus; 5841 RecordThread *thread; 5842 size_t inFrameCount; 5843 int lSessionId; 5844 5845 // check calling permissions 5846 if (!recordingAllowed()) { 5847 lStatus = PERMISSION_DENIED; 5848 goto Exit; 5849 } 5850 5851 // add client to list 5852 { // scope for mLock 5853 Mutex::Autolock _l(mLock); 5854 thread = checkRecordThread_l(input); 5855 if (thread == NULL) { 5856 lStatus = BAD_VALUE; 5857 goto Exit; 5858 } 5859 5860 client = registerPid_l(pid); 5861 5862 // If no audio session id is provided, create one here 5863 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5864 lSessionId = *sessionId; 5865 } else { 5866 lSessionId = nextUniqueId(); 5867 if (sessionId != NULL) { 5868 *sessionId = lSessionId; 5869 } 5870 } 5871 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5872 recordTrack = thread->createRecordTrack_l(client, 5873 sampleRate, 5874 format, 5875 channelMask, 5876 frameCount, 5877 lSessionId, 5878 &lStatus); 5879 } 5880 if (lStatus != NO_ERROR) { 5881 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5882 // destructor is called by the TrackBase destructor with mLock held 5883 client.clear(); 5884 recordTrack.clear(); 5885 goto Exit; 5886 } 5887 5888 // return to handle to client 5889 recordHandle = new RecordHandle(recordTrack); 5890 lStatus = NO_ERROR; 5891 5892Exit: 5893 if (status) { 5894 *status = lStatus; 5895 } 5896 return recordHandle; 5897} 5898 5899// ---------------------------------------------------------------------------- 5900 5901AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5902 : BnAudioRecord(), 5903 mRecordTrack(recordTrack) 5904{ 5905} 5906 5907AudioFlinger::RecordHandle::~RecordHandle() { 5908 stop(); 5909} 5910 5911sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5912 return mRecordTrack->getCblk(); 5913} 5914 5915status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5916 ALOGV("RecordHandle::start()"); 5917 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5918} 5919 5920void AudioFlinger::RecordHandle::stop() { 5921 ALOGV("RecordHandle::stop()"); 5922 mRecordTrack->stop(); 5923} 5924 5925status_t AudioFlinger::RecordHandle::onTransact( 5926 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5927{ 5928 return BnAudioRecord::onTransact(code, data, reply, flags); 5929} 5930 5931// ---------------------------------------------------------------------------- 5932 5933AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5934 AudioStreamIn *input, 5935 uint32_t sampleRate, 5936 uint32_t channels, 5937 audio_io_handle_t id, 5938 uint32_t device) : 5939 ThreadBase(audioFlinger, id, device, RECORD), 5940 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5941 // mRsmpInIndex and mInputBytes set by readInputParameters() 5942 mReqChannelCount(popcount(channels)), 5943 mReqSampleRate(sampleRate) 5944 // mBytesRead is only meaningful while active, and so is cleared in start() 5945 // (but might be better to also clear here for dump?) 5946{ 5947 snprintf(mName, kNameLength, "AudioIn_%X", id); 5948 5949 readInputParameters(); 5950} 5951 5952 5953AudioFlinger::RecordThread::~RecordThread() 5954{ 5955 delete[] mRsmpInBuffer; 5956 delete mResampler; 5957 delete[] mRsmpOutBuffer; 5958} 5959 5960void AudioFlinger::RecordThread::onFirstRef() 5961{ 5962 run(mName, PRIORITY_URGENT_AUDIO); 5963} 5964 5965status_t AudioFlinger::RecordThread::readyToRun() 5966{ 5967 status_t status = initCheck(); 5968 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5969 return status; 5970} 5971 5972bool AudioFlinger::RecordThread::threadLoop() 5973{ 5974 AudioBufferProvider::Buffer buffer; 5975 sp<RecordTrack> activeTrack; 5976 Vector< sp<EffectChain> > effectChains; 5977 5978 nsecs_t lastWarning = 0; 5979 5980 acquireWakeLock(); 5981 5982 // start recording 5983 while (!exitPending()) { 5984 5985 processConfigEvents(); 5986 5987 { // scope for mLock 5988 Mutex::Autolock _l(mLock); 5989 checkForNewParameters_l(); 5990 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5991 if (!mStandby) { 5992 mInput->stream->common.standby(&mInput->stream->common); 5993 mStandby = true; 5994 } 5995 5996 if (exitPending()) break; 5997 5998 releaseWakeLock_l(); 5999 ALOGV("RecordThread: loop stopping"); 6000 // go to sleep 6001 mWaitWorkCV.wait(mLock); 6002 ALOGV("RecordThread: loop starting"); 6003 acquireWakeLock_l(); 6004 continue; 6005 } 6006 if (mActiveTrack != 0) { 6007 if (mActiveTrack->mState == TrackBase::PAUSING) { 6008 if (!mStandby) { 6009 mInput->stream->common.standby(&mInput->stream->common); 6010 mStandby = true; 6011 } 6012 mActiveTrack.clear(); 6013 mStartStopCond.broadcast(); 6014 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6015 if (mReqChannelCount != mActiveTrack->channelCount()) { 6016 mActiveTrack.clear(); 6017 mStartStopCond.broadcast(); 6018 } else if (mBytesRead != 0) { 6019 // record start succeeds only if first read from audio input 6020 // succeeds 6021 if (mBytesRead > 0) { 6022 mActiveTrack->mState = TrackBase::ACTIVE; 6023 } else { 6024 mActiveTrack.clear(); 6025 } 6026 mStartStopCond.broadcast(); 6027 } 6028 mStandby = false; 6029 } 6030 } 6031 lockEffectChains_l(effectChains); 6032 } 6033 6034 if (mActiveTrack != 0) { 6035 if (mActiveTrack->mState != TrackBase::ACTIVE && 6036 mActiveTrack->mState != TrackBase::RESUMING) { 6037 unlockEffectChains(effectChains); 6038 usleep(kRecordThreadSleepUs); 6039 continue; 6040 } 6041 for (size_t i = 0; i < effectChains.size(); i ++) { 6042 effectChains[i]->process_l(); 6043 } 6044 6045 buffer.frameCount = mFrameCount; 6046 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6047 size_t framesOut = buffer.frameCount; 6048 if (mResampler == NULL) { 6049 // no resampling 6050 while (framesOut) { 6051 size_t framesIn = mFrameCount - mRsmpInIndex; 6052 if (framesIn) { 6053 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6054 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6055 if (framesIn > framesOut) 6056 framesIn = framesOut; 6057 mRsmpInIndex += framesIn; 6058 framesOut -= framesIn; 6059 if ((int)mChannelCount == mReqChannelCount || 6060 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6061 memcpy(dst, src, framesIn * mFrameSize); 6062 } else { 6063 int16_t *src16 = (int16_t *)src; 6064 int16_t *dst16 = (int16_t *)dst; 6065 if (mChannelCount == 1) { 6066 while (framesIn--) { 6067 *dst16++ = *src16; 6068 *dst16++ = *src16++; 6069 } 6070 } else { 6071 while (framesIn--) { 6072 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6073 src16 += 2; 6074 } 6075 } 6076 } 6077 } 6078 if (framesOut && mFrameCount == mRsmpInIndex) { 6079 if (framesOut == mFrameCount && 6080 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6081 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6082 framesOut = 0; 6083 } else { 6084 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6085 mRsmpInIndex = 0; 6086 } 6087 if (mBytesRead < 0) { 6088 ALOGE("Error reading audio input"); 6089 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6090 // Force input into standby so that it tries to 6091 // recover at next read attempt 6092 mInput->stream->common.standby(&mInput->stream->common); 6093 usleep(kRecordThreadSleepUs); 6094 } 6095 mRsmpInIndex = mFrameCount; 6096 framesOut = 0; 6097 buffer.frameCount = 0; 6098 } 6099 } 6100 } 6101 } else { 6102 // resampling 6103 6104 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6105 // alter output frame count as if we were expecting stereo samples 6106 if (mChannelCount == 1 && mReqChannelCount == 1) { 6107 framesOut >>= 1; 6108 } 6109 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6110 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6111 // are 32 bit aligned which should be always true. 6112 if (mChannelCount == 2 && mReqChannelCount == 1) { 6113 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6114 // the resampler always outputs stereo samples: do post stereo to mono conversion 6115 int16_t *src = (int16_t *)mRsmpOutBuffer; 6116 int16_t *dst = buffer.i16; 6117 while (framesOut--) { 6118 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6119 src += 2; 6120 } 6121 } else { 6122 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6123 } 6124 6125 } 6126 if (mFramestoDrop == 0) { 6127 mActiveTrack->releaseBuffer(&buffer); 6128 } else { 6129 if (mFramestoDrop > 0) { 6130 mFramestoDrop -= buffer.frameCount; 6131 if (mFramestoDrop <= 0) { 6132 clearSyncStartEvent(); 6133 } 6134 } else { 6135 mFramestoDrop += buffer.frameCount; 6136 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6137 mSyncStartEvent->isCancelled()) { 6138 ALOGW("Synced record %s, session %d, trigger session %d", 6139 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6140 mActiveTrack->sessionId(), 6141 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6142 clearSyncStartEvent(); 6143 } 6144 } 6145 } 6146 mActiveTrack->overflow(); 6147 } 6148 // client isn't retrieving buffers fast enough 6149 else { 6150 if (!mActiveTrack->setOverflow()) { 6151 nsecs_t now = systemTime(); 6152 if ((now - lastWarning) > kWarningThrottleNs) { 6153 ALOGW("RecordThread: buffer overflow"); 6154 lastWarning = now; 6155 } 6156 } 6157 // Release the processor for a while before asking for a new buffer. 6158 // This will give the application more chance to read from the buffer and 6159 // clear the overflow. 6160 usleep(kRecordThreadSleepUs); 6161 } 6162 } 6163 // enable changes in effect chain 6164 unlockEffectChains(effectChains); 6165 effectChains.clear(); 6166 } 6167 6168 if (!mStandby) { 6169 mInput->stream->common.standby(&mInput->stream->common); 6170 } 6171 mActiveTrack.clear(); 6172 6173 mStartStopCond.broadcast(); 6174 6175 releaseWakeLock(); 6176 6177 ALOGV("RecordThread %p exiting", this); 6178 return false; 6179} 6180 6181 6182sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6183 const sp<AudioFlinger::Client>& client, 6184 uint32_t sampleRate, 6185 audio_format_t format, 6186 int channelMask, 6187 int frameCount, 6188 int sessionId, 6189 status_t *status) 6190{ 6191 sp<RecordTrack> track; 6192 status_t lStatus; 6193 6194 lStatus = initCheck(); 6195 if (lStatus != NO_ERROR) { 6196 ALOGE("Audio driver not initialized."); 6197 goto Exit; 6198 } 6199 6200 { // scope for mLock 6201 Mutex::Autolock _l(mLock); 6202 6203 track = new RecordTrack(this, client, sampleRate, 6204 format, channelMask, frameCount, sessionId); 6205 6206 if (track->getCblk() == 0) { 6207 lStatus = NO_MEMORY; 6208 goto Exit; 6209 } 6210 6211 mTrack = track.get(); 6212 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6213 bool suspend = audio_is_bluetooth_sco_device( 6214 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6215 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6216 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6217 } 6218 lStatus = NO_ERROR; 6219 6220Exit: 6221 if (status) { 6222 *status = lStatus; 6223 } 6224 return track; 6225} 6226 6227status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6228 AudioSystem::sync_event_t event, 6229 int triggerSession) 6230{ 6231 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6232 sp<ThreadBase> strongMe = this; 6233 status_t status = NO_ERROR; 6234 6235 if (event == AudioSystem::SYNC_EVENT_NONE) { 6236 clearSyncStartEvent(); 6237 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6238 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6239 triggerSession, 6240 recordTrack->sessionId(), 6241 syncStartEventCallback, 6242 this); 6243 // Sync event can be cancelled by the trigger session if the track is not in a 6244 // compatible state in which case we start record immediately 6245 if (mSyncStartEvent->isCancelled()) { 6246 clearSyncStartEvent(); 6247 } else { 6248 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6249 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6250 } 6251 } 6252 6253 { 6254 AutoMutex lock(mLock); 6255 if (mActiveTrack != 0) { 6256 if (recordTrack != mActiveTrack.get()) { 6257 status = -EBUSY; 6258 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6259 mActiveTrack->mState = TrackBase::ACTIVE; 6260 } 6261 return status; 6262 } 6263 6264 recordTrack->mState = TrackBase::IDLE; 6265 mActiveTrack = recordTrack; 6266 mLock.unlock(); 6267 status_t status = AudioSystem::startInput(mId); 6268 mLock.lock(); 6269 if (status != NO_ERROR) { 6270 mActiveTrack.clear(); 6271 clearSyncStartEvent(); 6272 return status; 6273 } 6274 mRsmpInIndex = mFrameCount; 6275 mBytesRead = 0; 6276 if (mResampler != NULL) { 6277 mResampler->reset(); 6278 } 6279 mActiveTrack->mState = TrackBase::RESUMING; 6280 // signal thread to start 6281 ALOGV("Signal record thread"); 6282 mWaitWorkCV.signal(); 6283 // do not wait for mStartStopCond if exiting 6284 if (exitPending()) { 6285 mActiveTrack.clear(); 6286 status = INVALID_OPERATION; 6287 goto startError; 6288 } 6289 mStartStopCond.wait(mLock); 6290 if (mActiveTrack == 0) { 6291 ALOGV("Record failed to start"); 6292 status = BAD_VALUE; 6293 goto startError; 6294 } 6295 ALOGV("Record started OK"); 6296 return status; 6297 } 6298startError: 6299 AudioSystem::stopInput(mId); 6300 clearSyncStartEvent(); 6301 return status; 6302} 6303 6304void AudioFlinger::RecordThread::clearSyncStartEvent() 6305{ 6306 if (mSyncStartEvent != 0) { 6307 mSyncStartEvent->cancel(); 6308 } 6309 mSyncStartEvent.clear(); 6310 mFramestoDrop = 0; 6311} 6312 6313void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6314{ 6315 sp<SyncEvent> strongEvent = event.promote(); 6316 6317 if (strongEvent != 0) { 6318 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6319 me->handleSyncStartEvent(strongEvent); 6320 } 6321} 6322 6323void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6324{ 6325 if (event == mSyncStartEvent) { 6326 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6327 // from audio HAL 6328 mFramestoDrop = mFrameCount * 2; 6329 } 6330} 6331 6332void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6333 ALOGV("RecordThread::stop"); 6334 sp<ThreadBase> strongMe = this; 6335 { 6336 AutoMutex lock(mLock); 6337 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6338 mActiveTrack->mState = TrackBase::PAUSING; 6339 // do not wait for mStartStopCond if exiting 6340 if (exitPending()) { 6341 return; 6342 } 6343 mStartStopCond.wait(mLock); 6344 // if we have been restarted, recordTrack == mActiveTrack.get() here 6345 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6346 mLock.unlock(); 6347 AudioSystem::stopInput(mId); 6348 mLock.lock(); 6349 ALOGV("Record stopped OK"); 6350 } 6351 } 6352 } 6353} 6354 6355bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6356{ 6357 return false; 6358} 6359 6360status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6361{ 6362 if (!isValidSyncEvent(event)) { 6363 return BAD_VALUE; 6364 } 6365 6366 Mutex::Autolock _l(mLock); 6367 6368 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6369 mTrack->setSyncEvent(event); 6370 return NO_ERROR; 6371 } 6372 return NAME_NOT_FOUND; 6373} 6374 6375status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6376{ 6377 const size_t SIZE = 256; 6378 char buffer[SIZE]; 6379 String8 result; 6380 6381 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6382 result.append(buffer); 6383 6384 if (mActiveTrack != 0) { 6385 result.append("Active Track:\n"); 6386 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6387 mActiveTrack->dump(buffer, SIZE); 6388 result.append(buffer); 6389 6390 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6391 result.append(buffer); 6392 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6393 result.append(buffer); 6394 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6395 result.append(buffer); 6396 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6397 result.append(buffer); 6398 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6399 result.append(buffer); 6400 6401 6402 } else { 6403 result.append("No record client\n"); 6404 } 6405 write(fd, result.string(), result.size()); 6406 6407 dumpBase(fd, args); 6408 dumpEffectChains(fd, args); 6409 6410 return NO_ERROR; 6411} 6412 6413// AudioBufferProvider interface 6414status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6415{ 6416 size_t framesReq = buffer->frameCount; 6417 size_t framesReady = mFrameCount - mRsmpInIndex; 6418 int channelCount; 6419 6420 if (framesReady == 0) { 6421 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6422 if (mBytesRead < 0) { 6423 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6424 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6425 // Force input into standby so that it tries to 6426 // recover at next read attempt 6427 mInput->stream->common.standby(&mInput->stream->common); 6428 usleep(kRecordThreadSleepUs); 6429 } 6430 buffer->raw = NULL; 6431 buffer->frameCount = 0; 6432 return NOT_ENOUGH_DATA; 6433 } 6434 mRsmpInIndex = 0; 6435 framesReady = mFrameCount; 6436 } 6437 6438 if (framesReq > framesReady) { 6439 framesReq = framesReady; 6440 } 6441 6442 if (mChannelCount == 1 && mReqChannelCount == 2) { 6443 channelCount = 1; 6444 } else { 6445 channelCount = 2; 6446 } 6447 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6448 buffer->frameCount = framesReq; 6449 return NO_ERROR; 6450} 6451 6452// AudioBufferProvider interface 6453void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6454{ 6455 mRsmpInIndex += buffer->frameCount; 6456 buffer->frameCount = 0; 6457} 6458 6459bool AudioFlinger::RecordThread::checkForNewParameters_l() 6460{ 6461 bool reconfig = false; 6462 6463 while (!mNewParameters.isEmpty()) { 6464 status_t status = NO_ERROR; 6465 String8 keyValuePair = mNewParameters[0]; 6466 AudioParameter param = AudioParameter(keyValuePair); 6467 int value; 6468 audio_format_t reqFormat = mFormat; 6469 int reqSamplingRate = mReqSampleRate; 6470 int reqChannelCount = mReqChannelCount; 6471 6472 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6473 reqSamplingRate = value; 6474 reconfig = true; 6475 } 6476 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6477 reqFormat = (audio_format_t) value; 6478 reconfig = true; 6479 } 6480 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6481 reqChannelCount = popcount(value); 6482 reconfig = true; 6483 } 6484 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6485 // do not accept frame count changes if tracks are open as the track buffer 6486 // size depends on frame count and correct behavior would not be guaranteed 6487 // if frame count is changed after track creation 6488 if (mActiveTrack != 0) { 6489 status = INVALID_OPERATION; 6490 } else { 6491 reconfig = true; 6492 } 6493 } 6494 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6495 // forward device change to effects that have requested to be 6496 // aware of attached audio device. 6497 for (size_t i = 0; i < mEffectChains.size(); i++) { 6498 mEffectChains[i]->setDevice_l(value); 6499 } 6500 // store input device and output device but do not forward output device to audio HAL. 6501 // Note that status is ignored by the caller for output device 6502 // (see AudioFlinger::setParameters() 6503 if (value & AUDIO_DEVICE_OUT_ALL) { 6504 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6505 status = BAD_VALUE; 6506 } else { 6507 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6508 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6509 if (mTrack != NULL) { 6510 bool suspend = audio_is_bluetooth_sco_device( 6511 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6512 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6513 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6514 } 6515 } 6516 mDevice |= (uint32_t)value; 6517 } 6518 if (status == NO_ERROR) { 6519 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6520 if (status == INVALID_OPERATION) { 6521 mInput->stream->common.standby(&mInput->stream->common); 6522 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6523 keyValuePair.string()); 6524 } 6525 if (reconfig) { 6526 if (status == BAD_VALUE && 6527 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6528 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6529 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6530 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6531 (reqChannelCount <= FCC_2)) { 6532 status = NO_ERROR; 6533 } 6534 if (status == NO_ERROR) { 6535 readInputParameters(); 6536 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6537 } 6538 } 6539 } 6540 6541 mNewParameters.removeAt(0); 6542 6543 mParamStatus = status; 6544 mParamCond.signal(); 6545 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6546 // already timed out waiting for the status and will never signal the condition. 6547 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6548 } 6549 return reconfig; 6550} 6551 6552String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6553{ 6554 char *s; 6555 String8 out_s8 = String8(); 6556 6557 Mutex::Autolock _l(mLock); 6558 if (initCheck() != NO_ERROR) { 6559 return out_s8; 6560 } 6561 6562 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6563 out_s8 = String8(s); 6564 free(s); 6565 return out_s8; 6566} 6567 6568void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6569 AudioSystem::OutputDescriptor desc; 6570 void *param2 = NULL; 6571 6572 switch (event) { 6573 case AudioSystem::INPUT_OPENED: 6574 case AudioSystem::INPUT_CONFIG_CHANGED: 6575 desc.channels = mChannelMask; 6576 desc.samplingRate = mSampleRate; 6577 desc.format = mFormat; 6578 desc.frameCount = mFrameCount; 6579 desc.latency = 0; 6580 param2 = &desc; 6581 break; 6582 6583 case AudioSystem::INPUT_CLOSED: 6584 default: 6585 break; 6586 } 6587 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6588} 6589 6590void AudioFlinger::RecordThread::readInputParameters() 6591{ 6592 delete mRsmpInBuffer; 6593 // mRsmpInBuffer is always assigned a new[] below 6594 delete mRsmpOutBuffer; 6595 mRsmpOutBuffer = NULL; 6596 delete mResampler; 6597 mResampler = NULL; 6598 6599 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6600 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6601 mChannelCount = (uint16_t)popcount(mChannelMask); 6602 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6603 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6604 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6605 mFrameCount = mInputBytes / mFrameSize; 6606 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6607 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6608 6609 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6610 { 6611 int channelCount; 6612 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6613 // stereo to mono post process as the resampler always outputs stereo. 6614 if (mChannelCount == 1 && mReqChannelCount == 2) { 6615 channelCount = 1; 6616 } else { 6617 channelCount = 2; 6618 } 6619 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6620 mResampler->setSampleRate(mSampleRate); 6621 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6622 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6623 6624 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6625 if (mChannelCount == 1 && mReqChannelCount == 1) { 6626 mFrameCount >>= 1; 6627 } 6628 6629 } 6630 mRsmpInIndex = mFrameCount; 6631} 6632 6633unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6634{ 6635 Mutex::Autolock _l(mLock); 6636 if (initCheck() != NO_ERROR) { 6637 return 0; 6638 } 6639 6640 return mInput->stream->get_input_frames_lost(mInput->stream); 6641} 6642 6643uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6644{ 6645 Mutex::Autolock _l(mLock); 6646 uint32_t result = 0; 6647 if (getEffectChain_l(sessionId) != 0) { 6648 result = EFFECT_SESSION; 6649 } 6650 6651 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6652 result |= TRACK_SESSION; 6653 } 6654 6655 return result; 6656} 6657 6658AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6659{ 6660 Mutex::Autolock _l(mLock); 6661 return mTrack; 6662} 6663 6664AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6665{ 6666 Mutex::Autolock _l(mLock); 6667 return mInput; 6668} 6669 6670AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6671{ 6672 Mutex::Autolock _l(mLock); 6673 AudioStreamIn *input = mInput; 6674 mInput = NULL; 6675 return input; 6676} 6677 6678// this method must always be called either with ThreadBase mLock held or inside the thread loop 6679audio_stream_t* AudioFlinger::RecordThread::stream() const 6680{ 6681 if (mInput == NULL) { 6682 return NULL; 6683 } 6684 return &mInput->stream->common; 6685} 6686 6687 6688// ---------------------------------------------------------------------------- 6689 6690audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6691{ 6692 if (!settingsAllowed()) { 6693 return 0; 6694 } 6695 Mutex::Autolock _l(mLock); 6696 return loadHwModule_l(name); 6697} 6698 6699// loadHwModule_l() must be called with AudioFlinger::mLock held 6700audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6701{ 6702 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6703 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6704 ALOGW("loadHwModule() module %s already loaded", name); 6705 return mAudioHwDevs.keyAt(i); 6706 } 6707 } 6708 6709 audio_hw_device_t *dev; 6710 6711 int rc = load_audio_interface(name, &dev); 6712 if (rc) { 6713 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6714 return 0; 6715 } 6716 6717 mHardwareStatus = AUDIO_HW_INIT; 6718 rc = dev->init_check(dev); 6719 mHardwareStatus = AUDIO_HW_IDLE; 6720 if (rc) { 6721 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6722 return 0; 6723 } 6724 6725 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6726 (NULL != dev->set_master_volume)) { 6727 AutoMutex lock(mHardwareLock); 6728 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6729 dev->set_master_volume(dev, mMasterVolume); 6730 mHardwareStatus = AUDIO_HW_IDLE; 6731 } 6732 6733 audio_module_handle_t handle = nextUniqueId(); 6734 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6735 6736 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6737 name, dev->common.module->name, dev->common.module->id, handle); 6738 6739 return handle; 6740 6741} 6742 6743audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6744 audio_devices_t *pDevices, 6745 uint32_t *pSamplingRate, 6746 audio_format_t *pFormat, 6747 audio_channel_mask_t *pChannelMask, 6748 uint32_t *pLatencyMs, 6749 audio_output_flags_t flags) 6750{ 6751 status_t status; 6752 PlaybackThread *thread = NULL; 6753 struct audio_config config = { 6754 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6755 channel_mask: pChannelMask ? *pChannelMask : 0, 6756 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6757 }; 6758 audio_stream_out_t *outStream = NULL; 6759 audio_hw_device_t *outHwDev; 6760 6761 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6762 module, 6763 (pDevices != NULL) ? (int)*pDevices : 0, 6764 config.sample_rate, 6765 config.format, 6766 config.channel_mask, 6767 flags); 6768 6769 if (pDevices == NULL || *pDevices == 0) { 6770 return 0; 6771 } 6772 6773 Mutex::Autolock _l(mLock); 6774 6775 outHwDev = findSuitableHwDev_l(module, *pDevices); 6776 if (outHwDev == NULL) 6777 return 0; 6778 6779 audio_io_handle_t id = nextUniqueId(); 6780 6781 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6782 6783 status = outHwDev->open_output_stream(outHwDev, 6784 id, 6785 *pDevices, 6786 (audio_output_flags_t)flags, 6787 &config, 6788 &outStream); 6789 6790 mHardwareStatus = AUDIO_HW_IDLE; 6791 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6792 outStream, 6793 config.sample_rate, 6794 config.format, 6795 config.channel_mask, 6796 status); 6797 6798 if (status == NO_ERROR && outStream != NULL) { 6799 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6800 6801 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6802 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6803 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6804 thread = new DirectOutputThread(this, output, id, *pDevices); 6805 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6806 } else { 6807 thread = new MixerThread(this, output, id, *pDevices); 6808 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6809 } 6810 mPlaybackThreads.add(id, thread); 6811 6812 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6813 if (pFormat != NULL) *pFormat = config.format; 6814 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6815 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6816 6817 // notify client processes of the new output creation 6818 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6819 6820 // the first primary output opened designates the primary hw device 6821 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6822 ALOGI("Using module %d has the primary audio interface", module); 6823 mPrimaryHardwareDev = outHwDev; 6824 6825 AutoMutex lock(mHardwareLock); 6826 mHardwareStatus = AUDIO_HW_SET_MODE; 6827 outHwDev->set_mode(outHwDev, mMode); 6828 6829 // Determine the level of master volume support the primary audio HAL has, 6830 // and set the initial master volume at the same time. 6831 float initialVolume = 1.0; 6832 mMasterVolumeSupportLvl = MVS_NONE; 6833 6834 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6835 if ((NULL != outHwDev->get_master_volume) && 6836 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6837 mMasterVolumeSupportLvl = MVS_FULL; 6838 } else { 6839 mMasterVolumeSupportLvl = MVS_SETONLY; 6840 initialVolume = 1.0; 6841 } 6842 6843 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6844 if ((NULL == outHwDev->set_master_volume) || 6845 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6846 mMasterVolumeSupportLvl = MVS_NONE; 6847 } 6848 // now that we have a primary device, initialize master volume on other devices 6849 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6850 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6851 6852 if ((dev != mPrimaryHardwareDev) && 6853 (NULL != dev->set_master_volume)) { 6854 dev->set_master_volume(dev, initialVolume); 6855 } 6856 } 6857 mHardwareStatus = AUDIO_HW_IDLE; 6858 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6859 ? initialVolume 6860 : 1.0; 6861 mMasterVolume = initialVolume; 6862 } 6863 return id; 6864 } 6865 6866 return 0; 6867} 6868 6869audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6870 audio_io_handle_t output2) 6871{ 6872 Mutex::Autolock _l(mLock); 6873 MixerThread *thread1 = checkMixerThread_l(output1); 6874 MixerThread *thread2 = checkMixerThread_l(output2); 6875 6876 if (thread1 == NULL || thread2 == NULL) { 6877 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6878 return 0; 6879 } 6880 6881 audio_io_handle_t id = nextUniqueId(); 6882 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6883 thread->addOutputTrack(thread2); 6884 mPlaybackThreads.add(id, thread); 6885 // notify client processes of the new output creation 6886 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6887 return id; 6888} 6889 6890status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6891{ 6892 // keep strong reference on the playback thread so that 6893 // it is not destroyed while exit() is executed 6894 sp<PlaybackThread> thread; 6895 { 6896 Mutex::Autolock _l(mLock); 6897 thread = checkPlaybackThread_l(output); 6898 if (thread == NULL) { 6899 return BAD_VALUE; 6900 } 6901 6902 ALOGV("closeOutput() %d", output); 6903 6904 if (thread->type() == ThreadBase::MIXER) { 6905 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6906 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6907 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6908 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6909 } 6910 } 6911 } 6912 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6913 mPlaybackThreads.removeItem(output); 6914 } 6915 thread->exit(); 6916 // The thread entity (active unit of execution) is no longer running here, 6917 // but the ThreadBase container still exists. 6918 6919 if (thread->type() != ThreadBase::DUPLICATING) { 6920 AudioStreamOut *out = thread->clearOutput(); 6921 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6922 // from now on thread->mOutput is NULL 6923 out->hwDev->close_output_stream(out->hwDev, out->stream); 6924 delete out; 6925 } 6926 return NO_ERROR; 6927} 6928 6929status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6930{ 6931 Mutex::Autolock _l(mLock); 6932 PlaybackThread *thread = checkPlaybackThread_l(output); 6933 6934 if (thread == NULL) { 6935 return BAD_VALUE; 6936 } 6937 6938 ALOGV("suspendOutput() %d", output); 6939 thread->suspend(); 6940 6941 return NO_ERROR; 6942} 6943 6944status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6945{ 6946 Mutex::Autolock _l(mLock); 6947 PlaybackThread *thread = checkPlaybackThread_l(output); 6948 6949 if (thread == NULL) { 6950 return BAD_VALUE; 6951 } 6952 6953 ALOGV("restoreOutput() %d", output); 6954 6955 thread->restore(); 6956 6957 return NO_ERROR; 6958} 6959 6960audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6961 audio_devices_t *pDevices, 6962 uint32_t *pSamplingRate, 6963 audio_format_t *pFormat, 6964 uint32_t *pChannelMask) 6965{ 6966 status_t status; 6967 RecordThread *thread = NULL; 6968 struct audio_config config = { 6969 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6970 channel_mask: pChannelMask ? *pChannelMask : 0, 6971 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6972 }; 6973 uint32_t reqSamplingRate = config.sample_rate; 6974 audio_format_t reqFormat = config.format; 6975 audio_channel_mask_t reqChannels = config.channel_mask; 6976 audio_stream_in_t *inStream = NULL; 6977 audio_hw_device_t *inHwDev; 6978 6979 if (pDevices == NULL || *pDevices == 0) { 6980 return 0; 6981 } 6982 6983 Mutex::Autolock _l(mLock); 6984 6985 inHwDev = findSuitableHwDev_l(module, *pDevices); 6986 if (inHwDev == NULL) 6987 return 0; 6988 6989 audio_io_handle_t id = nextUniqueId(); 6990 6991 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6992 &inStream); 6993 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6994 inStream, 6995 config.sample_rate, 6996 config.format, 6997 config.channel_mask, 6998 status); 6999 7000 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7001 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7002 // or stereo to mono conversions on 16 bit PCM inputs. 7003 if (status == BAD_VALUE && 7004 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7005 (config.sample_rate <= 2 * reqSamplingRate) && 7006 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7007 ALOGV("openInput() reopening with proposed sampling rate and channels"); 7008 inStream = NULL; 7009 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 7010 } 7011 7012 if (status == NO_ERROR && inStream != NULL) { 7013 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7014 7015 // Start record thread 7016 // RecorThread require both input and output device indication to forward to audio 7017 // pre processing modules 7018 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 7019 thread = new RecordThread(this, 7020 input, 7021 reqSamplingRate, 7022 reqChannels, 7023 id, 7024 device); 7025 mRecordThreads.add(id, thread); 7026 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7027 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7028 if (pFormat != NULL) *pFormat = config.format; 7029 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7030 7031 input->stream->common.standby(&input->stream->common); 7032 7033 // notify client processes of the new input creation 7034 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7035 return id; 7036 } 7037 7038 return 0; 7039} 7040 7041status_t AudioFlinger::closeInput(audio_io_handle_t input) 7042{ 7043 // keep strong reference on the record thread so that 7044 // it is not destroyed while exit() is executed 7045 sp<RecordThread> thread; 7046 { 7047 Mutex::Autolock _l(mLock); 7048 thread = checkRecordThread_l(input); 7049 if (thread == 0) { 7050 return BAD_VALUE; 7051 } 7052 7053 ALOGV("closeInput() %d", input); 7054 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7055 mRecordThreads.removeItem(input); 7056 } 7057 thread->exit(); 7058 // The thread entity (active unit of execution) is no longer running here, 7059 // but the ThreadBase container still exists. 7060 7061 AudioStreamIn *in = thread->clearInput(); 7062 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7063 // from now on thread->mInput is NULL 7064 in->hwDev->close_input_stream(in->hwDev, in->stream); 7065 delete in; 7066 7067 return NO_ERROR; 7068} 7069 7070status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7071{ 7072 Mutex::Autolock _l(mLock); 7073 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7074 7075 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7076 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7077 thread->invalidateTracks(stream); 7078 } 7079 7080 return NO_ERROR; 7081} 7082 7083 7084int AudioFlinger::newAudioSessionId() 7085{ 7086 return nextUniqueId(); 7087} 7088 7089void AudioFlinger::acquireAudioSessionId(int audioSession) 7090{ 7091 Mutex::Autolock _l(mLock); 7092 pid_t caller = IPCThreadState::self()->getCallingPid(); 7093 ALOGV("acquiring %d from %d", audioSession, caller); 7094 size_t num = mAudioSessionRefs.size(); 7095 for (size_t i = 0; i< num; i++) { 7096 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7097 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7098 ref->mCnt++; 7099 ALOGV(" incremented refcount to %d", ref->mCnt); 7100 return; 7101 } 7102 } 7103 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7104 ALOGV(" added new entry for %d", audioSession); 7105} 7106 7107void AudioFlinger::releaseAudioSessionId(int audioSession) 7108{ 7109 Mutex::Autolock _l(mLock); 7110 pid_t caller = IPCThreadState::self()->getCallingPid(); 7111 ALOGV("releasing %d from %d", audioSession, caller); 7112 size_t num = mAudioSessionRefs.size(); 7113 for (size_t i = 0; i< num; i++) { 7114 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7115 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7116 ref->mCnt--; 7117 ALOGV(" decremented refcount to %d", ref->mCnt); 7118 if (ref->mCnt == 0) { 7119 mAudioSessionRefs.removeAt(i); 7120 delete ref; 7121 purgeStaleEffects_l(); 7122 } 7123 return; 7124 } 7125 } 7126 ALOGW("session id %d not found for pid %d", audioSession, caller); 7127} 7128 7129void AudioFlinger::purgeStaleEffects_l() { 7130 7131 ALOGV("purging stale effects"); 7132 7133 Vector< sp<EffectChain> > chains; 7134 7135 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7136 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7137 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7138 sp<EffectChain> ec = t->mEffectChains[j]; 7139 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7140 chains.push(ec); 7141 } 7142 } 7143 } 7144 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7145 sp<RecordThread> t = mRecordThreads.valueAt(i); 7146 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7147 sp<EffectChain> ec = t->mEffectChains[j]; 7148 chains.push(ec); 7149 } 7150 } 7151 7152 for (size_t i = 0; i < chains.size(); i++) { 7153 sp<EffectChain> ec = chains[i]; 7154 int sessionid = ec->sessionId(); 7155 sp<ThreadBase> t = ec->mThread.promote(); 7156 if (t == 0) { 7157 continue; 7158 } 7159 size_t numsessionrefs = mAudioSessionRefs.size(); 7160 bool found = false; 7161 for (size_t k = 0; k < numsessionrefs; k++) { 7162 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7163 if (ref->mSessionid == sessionid) { 7164 ALOGV(" session %d still exists for %d with %d refs", 7165 sessionid, ref->mPid, ref->mCnt); 7166 found = true; 7167 break; 7168 } 7169 } 7170 if (!found) { 7171 // remove all effects from the chain 7172 while (ec->mEffects.size()) { 7173 sp<EffectModule> effect = ec->mEffects[0]; 7174 effect->unPin(); 7175 Mutex::Autolock _l (t->mLock); 7176 t->removeEffect_l(effect); 7177 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7178 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7179 if (handle != 0) { 7180 handle->mEffect.clear(); 7181 if (handle->mHasControl && handle->mEnabled) { 7182 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7183 } 7184 } 7185 } 7186 AudioSystem::unregisterEffect(effect->id()); 7187 } 7188 } 7189 } 7190 return; 7191} 7192 7193// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7194AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7195{ 7196 return mPlaybackThreads.valueFor(output).get(); 7197} 7198 7199// checkMixerThread_l() must be called with AudioFlinger::mLock held 7200AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7201{ 7202 PlaybackThread *thread = checkPlaybackThread_l(output); 7203 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7204} 7205 7206// checkRecordThread_l() must be called with AudioFlinger::mLock held 7207AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7208{ 7209 return mRecordThreads.valueFor(input).get(); 7210} 7211 7212uint32_t AudioFlinger::nextUniqueId() 7213{ 7214 return android_atomic_inc(&mNextUniqueId); 7215} 7216 7217AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7218{ 7219 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7220 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7221 AudioStreamOut *output = thread->getOutput(); 7222 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7223 return thread; 7224 } 7225 } 7226 return NULL; 7227} 7228 7229uint32_t AudioFlinger::primaryOutputDevice_l() const 7230{ 7231 PlaybackThread *thread = primaryPlaybackThread_l(); 7232 7233 if (thread == NULL) { 7234 return 0; 7235 } 7236 7237 return thread->device(); 7238} 7239 7240sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7241 int triggerSession, 7242 int listenerSession, 7243 sync_event_callback_t callBack, 7244 void *cookie) 7245{ 7246 Mutex::Autolock _l(mLock); 7247 7248 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7249 status_t playStatus = NAME_NOT_FOUND; 7250 status_t recStatus = NAME_NOT_FOUND; 7251 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7252 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7253 if (playStatus == NO_ERROR) { 7254 return event; 7255 } 7256 } 7257 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7258 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7259 if (recStatus == NO_ERROR) { 7260 return event; 7261 } 7262 } 7263 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7264 mPendingSyncEvents.add(event); 7265 } else { 7266 ALOGV("createSyncEvent() invalid event %d", event->type()); 7267 event.clear(); 7268 } 7269 return event; 7270} 7271 7272// ---------------------------------------------------------------------------- 7273// Effect management 7274// ---------------------------------------------------------------------------- 7275 7276 7277status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7278{ 7279 Mutex::Autolock _l(mLock); 7280 return EffectQueryNumberEffects(numEffects); 7281} 7282 7283status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7284{ 7285 Mutex::Autolock _l(mLock); 7286 return EffectQueryEffect(index, descriptor); 7287} 7288 7289status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7290 effect_descriptor_t *descriptor) const 7291{ 7292 Mutex::Autolock _l(mLock); 7293 return EffectGetDescriptor(pUuid, descriptor); 7294} 7295 7296 7297sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7298 effect_descriptor_t *pDesc, 7299 const sp<IEffectClient>& effectClient, 7300 int32_t priority, 7301 audio_io_handle_t io, 7302 int sessionId, 7303 status_t *status, 7304 int *id, 7305 int *enabled) 7306{ 7307 status_t lStatus = NO_ERROR; 7308 sp<EffectHandle> handle; 7309 effect_descriptor_t desc; 7310 7311 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7312 pid, effectClient.get(), priority, sessionId, io); 7313 7314 if (pDesc == NULL) { 7315 lStatus = BAD_VALUE; 7316 goto Exit; 7317 } 7318 7319 // check audio settings permission for global effects 7320 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7321 lStatus = PERMISSION_DENIED; 7322 goto Exit; 7323 } 7324 7325 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7326 // that can only be created by audio policy manager (running in same process) 7327 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7328 lStatus = PERMISSION_DENIED; 7329 goto Exit; 7330 } 7331 7332 if (io == 0) { 7333 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7334 // output must be specified by AudioPolicyManager when using session 7335 // AUDIO_SESSION_OUTPUT_STAGE 7336 lStatus = BAD_VALUE; 7337 goto Exit; 7338 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7339 // if the output returned by getOutputForEffect() is removed before we lock the 7340 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7341 // and we will exit safely 7342 io = AudioSystem::getOutputForEffect(&desc); 7343 } 7344 } 7345 7346 { 7347 Mutex::Autolock _l(mLock); 7348 7349 7350 if (!EffectIsNullUuid(&pDesc->uuid)) { 7351 // if uuid is specified, request effect descriptor 7352 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7353 if (lStatus < 0) { 7354 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7355 goto Exit; 7356 } 7357 } else { 7358 // if uuid is not specified, look for an available implementation 7359 // of the required type in effect factory 7360 if (EffectIsNullUuid(&pDesc->type)) { 7361 ALOGW("createEffect() no effect type"); 7362 lStatus = BAD_VALUE; 7363 goto Exit; 7364 } 7365 uint32_t numEffects = 0; 7366 effect_descriptor_t d; 7367 d.flags = 0; // prevent compiler warning 7368 bool found = false; 7369 7370 lStatus = EffectQueryNumberEffects(&numEffects); 7371 if (lStatus < 0) { 7372 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7373 goto Exit; 7374 } 7375 for (uint32_t i = 0; i < numEffects; i++) { 7376 lStatus = EffectQueryEffect(i, &desc); 7377 if (lStatus < 0) { 7378 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7379 continue; 7380 } 7381 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7382 // If matching type found save effect descriptor. If the session is 7383 // 0 and the effect is not auxiliary, continue enumeration in case 7384 // an auxiliary version of this effect type is available 7385 found = true; 7386 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7387 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7388 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7389 break; 7390 } 7391 } 7392 } 7393 if (!found) { 7394 lStatus = BAD_VALUE; 7395 ALOGW("createEffect() effect not found"); 7396 goto Exit; 7397 } 7398 // For same effect type, chose auxiliary version over insert version if 7399 // connect to output mix (Compliance to OpenSL ES) 7400 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7401 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7402 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7403 } 7404 } 7405 7406 // Do not allow auxiliary effects on a session different from 0 (output mix) 7407 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7408 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7409 lStatus = INVALID_OPERATION; 7410 goto Exit; 7411 } 7412 7413 // check recording permission for visualizer 7414 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7415 !recordingAllowed()) { 7416 lStatus = PERMISSION_DENIED; 7417 goto Exit; 7418 } 7419 7420 // return effect descriptor 7421 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7422 7423 // If output is not specified try to find a matching audio session ID in one of the 7424 // output threads. 7425 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7426 // because of code checking output when entering the function. 7427 // Note: io is never 0 when creating an effect on an input 7428 if (io == 0) { 7429 // look for the thread where the specified audio session is present 7430 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7431 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7432 io = mPlaybackThreads.keyAt(i); 7433 break; 7434 } 7435 } 7436 if (io == 0) { 7437 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7438 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7439 io = mRecordThreads.keyAt(i); 7440 break; 7441 } 7442 } 7443 } 7444 // If no output thread contains the requested session ID, default to 7445 // first output. The effect chain will be moved to the correct output 7446 // thread when a track with the same session ID is created 7447 if (io == 0 && mPlaybackThreads.size()) { 7448 io = mPlaybackThreads.keyAt(0); 7449 } 7450 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7451 } 7452 ThreadBase *thread = checkRecordThread_l(io); 7453 if (thread == NULL) { 7454 thread = checkPlaybackThread_l(io); 7455 if (thread == NULL) { 7456 ALOGE("createEffect() unknown output thread"); 7457 lStatus = BAD_VALUE; 7458 goto Exit; 7459 } 7460 } 7461 7462 sp<Client> client = registerPid_l(pid); 7463 7464 // create effect on selected output thread 7465 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7466 &desc, enabled, &lStatus); 7467 if (handle != 0 && id != NULL) { 7468 *id = handle->id(); 7469 } 7470 } 7471 7472Exit: 7473 if (status != NULL) { 7474 *status = lStatus; 7475 } 7476 return handle; 7477} 7478 7479status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7480 audio_io_handle_t dstOutput) 7481{ 7482 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7483 sessionId, srcOutput, dstOutput); 7484 Mutex::Autolock _l(mLock); 7485 if (srcOutput == dstOutput) { 7486 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7487 return NO_ERROR; 7488 } 7489 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7490 if (srcThread == NULL) { 7491 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7492 return BAD_VALUE; 7493 } 7494 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7495 if (dstThread == NULL) { 7496 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7497 return BAD_VALUE; 7498 } 7499 7500 Mutex::Autolock _dl(dstThread->mLock); 7501 Mutex::Autolock _sl(srcThread->mLock); 7502 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7503 7504 return NO_ERROR; 7505} 7506 7507// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7508status_t AudioFlinger::moveEffectChain_l(int sessionId, 7509 AudioFlinger::PlaybackThread *srcThread, 7510 AudioFlinger::PlaybackThread *dstThread, 7511 bool reRegister) 7512{ 7513 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7514 sessionId, srcThread, dstThread); 7515 7516 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7517 if (chain == 0) { 7518 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7519 sessionId, srcThread); 7520 return INVALID_OPERATION; 7521 } 7522 7523 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7524 // so that a new chain is created with correct parameters when first effect is added. This is 7525 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7526 // removed. 7527 srcThread->removeEffectChain_l(chain); 7528 7529 // transfer all effects one by one so that new effect chain is created on new thread with 7530 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7531 audio_io_handle_t dstOutput = dstThread->id(); 7532 sp<EffectChain> dstChain; 7533 uint32_t strategy = 0; // prevent compiler warning 7534 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7535 while (effect != 0) { 7536 srcThread->removeEffect_l(effect); 7537 dstThread->addEffect_l(effect); 7538 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7539 if (effect->state() == EffectModule::ACTIVE || 7540 effect->state() == EffectModule::STOPPING) { 7541 effect->start(); 7542 } 7543 // if the move request is not received from audio policy manager, the effect must be 7544 // re-registered with the new strategy and output 7545 if (dstChain == 0) { 7546 dstChain = effect->chain().promote(); 7547 if (dstChain == 0) { 7548 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7549 srcThread->addEffect_l(effect); 7550 return NO_INIT; 7551 } 7552 strategy = dstChain->strategy(); 7553 } 7554 if (reRegister) { 7555 AudioSystem::unregisterEffect(effect->id()); 7556 AudioSystem::registerEffect(&effect->desc(), 7557 dstOutput, 7558 strategy, 7559 sessionId, 7560 effect->id()); 7561 } 7562 effect = chain->getEffectFromId_l(0); 7563 } 7564 7565 return NO_ERROR; 7566} 7567 7568 7569// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7570sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7571 const sp<AudioFlinger::Client>& client, 7572 const sp<IEffectClient>& effectClient, 7573 int32_t priority, 7574 int sessionId, 7575 effect_descriptor_t *desc, 7576 int *enabled, 7577 status_t *status 7578 ) 7579{ 7580 sp<EffectModule> effect; 7581 sp<EffectHandle> handle; 7582 status_t lStatus; 7583 sp<EffectChain> chain; 7584 bool chainCreated = false; 7585 bool effectCreated = false; 7586 bool effectRegistered = false; 7587 7588 lStatus = initCheck(); 7589 if (lStatus != NO_ERROR) { 7590 ALOGW("createEffect_l() Audio driver not initialized."); 7591 goto Exit; 7592 } 7593 7594 // Do not allow effects with session ID 0 on direct output or duplicating threads 7595 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7596 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7597 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7598 desc->name, sessionId); 7599 lStatus = BAD_VALUE; 7600 goto Exit; 7601 } 7602 // Only Pre processor effects are allowed on input threads and only on input threads 7603 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7604 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7605 desc->name, desc->flags, mType); 7606 lStatus = BAD_VALUE; 7607 goto Exit; 7608 } 7609 7610 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7611 7612 { // scope for mLock 7613 Mutex::Autolock _l(mLock); 7614 7615 // check for existing effect chain with the requested audio session 7616 chain = getEffectChain_l(sessionId); 7617 if (chain == 0) { 7618 // create a new chain for this session 7619 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7620 chain = new EffectChain(this, sessionId); 7621 addEffectChain_l(chain); 7622 chain->setStrategy(getStrategyForSession_l(sessionId)); 7623 chainCreated = true; 7624 } else { 7625 effect = chain->getEffectFromDesc_l(desc); 7626 } 7627 7628 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7629 7630 if (effect == 0) { 7631 int id = mAudioFlinger->nextUniqueId(); 7632 // Check CPU and memory usage 7633 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7634 if (lStatus != NO_ERROR) { 7635 goto Exit; 7636 } 7637 effectRegistered = true; 7638 // create a new effect module if none present in the chain 7639 effect = new EffectModule(this, chain, desc, id, sessionId); 7640 lStatus = effect->status(); 7641 if (lStatus != NO_ERROR) { 7642 goto Exit; 7643 } 7644 lStatus = chain->addEffect_l(effect); 7645 if (lStatus != NO_ERROR) { 7646 goto Exit; 7647 } 7648 effectCreated = true; 7649 7650 effect->setDevice(mDevice); 7651 effect->setMode(mAudioFlinger->getMode()); 7652 } 7653 // create effect handle and connect it to effect module 7654 handle = new EffectHandle(effect, client, effectClient, priority); 7655 lStatus = effect->addHandle(handle); 7656 if (enabled != NULL) { 7657 *enabled = (int)effect->isEnabled(); 7658 } 7659 } 7660 7661Exit: 7662 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7663 Mutex::Autolock _l(mLock); 7664 if (effectCreated) { 7665 chain->removeEffect_l(effect); 7666 } 7667 if (effectRegistered) { 7668 AudioSystem::unregisterEffect(effect->id()); 7669 } 7670 if (chainCreated) { 7671 removeEffectChain_l(chain); 7672 } 7673 handle.clear(); 7674 } 7675 7676 if (status != NULL) { 7677 *status = lStatus; 7678 } 7679 return handle; 7680} 7681 7682sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7683{ 7684 Mutex::Autolock _l(mLock); 7685 return getEffect_l(sessionId, effectId); 7686} 7687 7688sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7689{ 7690 sp<EffectChain> chain = getEffectChain_l(sessionId); 7691 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7692} 7693 7694// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7695// PlaybackThread::mLock held 7696status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7697{ 7698 // check for existing effect chain with the requested audio session 7699 int sessionId = effect->sessionId(); 7700 sp<EffectChain> chain = getEffectChain_l(sessionId); 7701 bool chainCreated = false; 7702 7703 if (chain == 0) { 7704 // create a new chain for this session 7705 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7706 chain = new EffectChain(this, sessionId); 7707 addEffectChain_l(chain); 7708 chain->setStrategy(getStrategyForSession_l(sessionId)); 7709 chainCreated = true; 7710 } 7711 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7712 7713 if (chain->getEffectFromId_l(effect->id()) != 0) { 7714 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7715 this, effect->desc().name, chain.get()); 7716 return BAD_VALUE; 7717 } 7718 7719 status_t status = chain->addEffect_l(effect); 7720 if (status != NO_ERROR) { 7721 if (chainCreated) { 7722 removeEffectChain_l(chain); 7723 } 7724 return status; 7725 } 7726 7727 effect->setDevice(mDevice); 7728 effect->setMode(mAudioFlinger->getMode()); 7729 return NO_ERROR; 7730} 7731 7732void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7733 7734 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7735 effect_descriptor_t desc = effect->desc(); 7736 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7737 detachAuxEffect_l(effect->id()); 7738 } 7739 7740 sp<EffectChain> chain = effect->chain().promote(); 7741 if (chain != 0) { 7742 // remove effect chain if removing last effect 7743 if (chain->removeEffect_l(effect) == 0) { 7744 removeEffectChain_l(chain); 7745 } 7746 } else { 7747 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7748 } 7749} 7750 7751void AudioFlinger::ThreadBase::lockEffectChains_l( 7752 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7753{ 7754 effectChains = mEffectChains; 7755 for (size_t i = 0; i < mEffectChains.size(); i++) { 7756 mEffectChains[i]->lock(); 7757 } 7758} 7759 7760void AudioFlinger::ThreadBase::unlockEffectChains( 7761 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7762{ 7763 for (size_t i = 0; i < effectChains.size(); i++) { 7764 effectChains[i]->unlock(); 7765 } 7766} 7767 7768sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7769{ 7770 Mutex::Autolock _l(mLock); 7771 return getEffectChain_l(sessionId); 7772} 7773 7774sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7775{ 7776 size_t size = mEffectChains.size(); 7777 for (size_t i = 0; i < size; i++) { 7778 if (mEffectChains[i]->sessionId() == sessionId) { 7779 return mEffectChains[i]; 7780 } 7781 } 7782 return 0; 7783} 7784 7785void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7786{ 7787 Mutex::Autolock _l(mLock); 7788 size_t size = mEffectChains.size(); 7789 for (size_t i = 0; i < size; i++) { 7790 mEffectChains[i]->setMode_l(mode); 7791 } 7792} 7793 7794void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7795 const wp<EffectHandle>& handle, 7796 bool unpinIfLast) { 7797 7798 Mutex::Autolock _l(mLock); 7799 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7800 // delete the effect module if removing last handle on it 7801 if (effect->removeHandle(handle) == 0) { 7802 if (!effect->isPinned() || unpinIfLast) { 7803 removeEffect_l(effect); 7804 AudioSystem::unregisterEffect(effect->id()); 7805 } 7806 } 7807} 7808 7809status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7810{ 7811 int session = chain->sessionId(); 7812 int16_t *buffer = mMixBuffer; 7813 bool ownsBuffer = false; 7814 7815 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7816 if (session > 0) { 7817 // Only one effect chain can be present in direct output thread and it uses 7818 // the mix buffer as input 7819 if (mType != DIRECT) { 7820 size_t numSamples = mNormalFrameCount * mChannelCount; 7821 buffer = new int16_t[numSamples]; 7822 memset(buffer, 0, numSamples * sizeof(int16_t)); 7823 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7824 ownsBuffer = true; 7825 } 7826 7827 // Attach all tracks with same session ID to this chain. 7828 for (size_t i = 0; i < mTracks.size(); ++i) { 7829 sp<Track> track = mTracks[i]; 7830 if (session == track->sessionId()) { 7831 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7832 track->setMainBuffer(buffer); 7833 chain->incTrackCnt(); 7834 } 7835 } 7836 7837 // indicate all active tracks in the chain 7838 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7839 sp<Track> track = mActiveTracks[i].promote(); 7840 if (track == 0) continue; 7841 if (session == track->sessionId()) { 7842 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7843 chain->incActiveTrackCnt(); 7844 } 7845 } 7846 } 7847 7848 chain->setInBuffer(buffer, ownsBuffer); 7849 chain->setOutBuffer(mMixBuffer); 7850 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7851 // chains list in order to be processed last as it contains output stage effects 7852 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7853 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7854 // after track specific effects and before output stage 7855 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7856 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7857 // Effect chain for other sessions are inserted at beginning of effect 7858 // chains list to be processed before output mix effects. Relative order between other 7859 // sessions is not important 7860 size_t size = mEffectChains.size(); 7861 size_t i = 0; 7862 for (i = 0; i < size; i++) { 7863 if (mEffectChains[i]->sessionId() < session) break; 7864 } 7865 mEffectChains.insertAt(chain, i); 7866 checkSuspendOnAddEffectChain_l(chain); 7867 7868 return NO_ERROR; 7869} 7870 7871size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7872{ 7873 int session = chain->sessionId(); 7874 7875 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7876 7877 for (size_t i = 0; i < mEffectChains.size(); i++) { 7878 if (chain == mEffectChains[i]) { 7879 mEffectChains.removeAt(i); 7880 // detach all active tracks from the chain 7881 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7882 sp<Track> track = mActiveTracks[i].promote(); 7883 if (track == 0) continue; 7884 if (session == track->sessionId()) { 7885 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7886 chain.get(), session); 7887 chain->decActiveTrackCnt(); 7888 } 7889 } 7890 7891 // detach all tracks with same session ID from this chain 7892 for (size_t i = 0; i < mTracks.size(); ++i) { 7893 sp<Track> track = mTracks[i]; 7894 if (session == track->sessionId()) { 7895 track->setMainBuffer(mMixBuffer); 7896 chain->decTrackCnt(); 7897 } 7898 } 7899 break; 7900 } 7901 } 7902 return mEffectChains.size(); 7903} 7904 7905status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7906 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7907{ 7908 Mutex::Autolock _l(mLock); 7909 return attachAuxEffect_l(track, EffectId); 7910} 7911 7912status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7913 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7914{ 7915 status_t status = NO_ERROR; 7916 7917 if (EffectId == 0) { 7918 track->setAuxBuffer(0, NULL); 7919 } else { 7920 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7921 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7922 if (effect != 0) { 7923 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7924 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7925 } else { 7926 status = INVALID_OPERATION; 7927 } 7928 } else { 7929 status = BAD_VALUE; 7930 } 7931 } 7932 return status; 7933} 7934 7935void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7936{ 7937 for (size_t i = 0; i < mTracks.size(); ++i) { 7938 sp<Track> track = mTracks[i]; 7939 if (track->auxEffectId() == effectId) { 7940 attachAuxEffect_l(track, 0); 7941 } 7942 } 7943} 7944 7945status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7946{ 7947 // only one chain per input thread 7948 if (mEffectChains.size() != 0) { 7949 return INVALID_OPERATION; 7950 } 7951 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7952 7953 chain->setInBuffer(NULL); 7954 chain->setOutBuffer(NULL); 7955 7956 checkSuspendOnAddEffectChain_l(chain); 7957 7958 mEffectChains.add(chain); 7959 7960 return NO_ERROR; 7961} 7962 7963size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7964{ 7965 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7966 ALOGW_IF(mEffectChains.size() != 1, 7967 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7968 chain.get(), mEffectChains.size(), this); 7969 if (mEffectChains.size() == 1) { 7970 mEffectChains.removeAt(0); 7971 } 7972 return 0; 7973} 7974 7975// ---------------------------------------------------------------------------- 7976// EffectModule implementation 7977// ---------------------------------------------------------------------------- 7978 7979#undef LOG_TAG 7980#define LOG_TAG "AudioFlinger::EffectModule" 7981 7982AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7983 const wp<AudioFlinger::EffectChain>& chain, 7984 effect_descriptor_t *desc, 7985 int id, 7986 int sessionId) 7987 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7988 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7989 // mDescriptor is set below 7990 // mConfig is set by configure() and not used before then 7991 mEffectInterface(NULL), 7992 mStatus(NO_INIT), mState(IDLE), 7993 // mMaxDisableWaitCnt is set by configure() and not used before then 7994 // mDisableWaitCnt is set by process() and updateState() and not used before then 7995 mSuspended(false) 7996{ 7997 ALOGV("Constructor %p", this); 7998 int lStatus; 7999 if (thread == NULL) { 8000 return; 8001 } 8002 8003 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 8004 8005 // create effect engine from effect factory 8006 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8007 8008 if (mStatus != NO_ERROR) { 8009 return; 8010 } 8011 lStatus = init(); 8012 if (lStatus < 0) { 8013 mStatus = lStatus; 8014 goto Error; 8015 } 8016 8017 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8018 return; 8019Error: 8020 EffectRelease(mEffectInterface); 8021 mEffectInterface = NULL; 8022 ALOGV("Constructor Error %d", mStatus); 8023} 8024 8025AudioFlinger::EffectModule::~EffectModule() 8026{ 8027 ALOGV("Destructor %p", this); 8028 if (mEffectInterface != NULL) { 8029 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8030 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8031 sp<ThreadBase> thread = mThread.promote(); 8032 if (thread != 0) { 8033 audio_stream_t *stream = thread->stream(); 8034 if (stream != NULL) { 8035 stream->remove_audio_effect(stream, mEffectInterface); 8036 } 8037 } 8038 } 8039 // release effect engine 8040 EffectRelease(mEffectInterface); 8041 } 8042} 8043 8044status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8045{ 8046 status_t status; 8047 8048 Mutex::Autolock _l(mLock); 8049 int priority = handle->priority(); 8050 size_t size = mHandles.size(); 8051 sp<EffectHandle> h; 8052 size_t i; 8053 for (i = 0; i < size; i++) { 8054 h = mHandles[i].promote(); 8055 if (h == 0) continue; 8056 if (h->priority() <= priority) break; 8057 } 8058 // if inserted in first place, move effect control from previous owner to this handle 8059 if (i == 0) { 8060 bool enabled = false; 8061 if (h != 0) { 8062 enabled = h->enabled(); 8063 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8064 } 8065 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8066 status = NO_ERROR; 8067 } else { 8068 status = ALREADY_EXISTS; 8069 } 8070 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8071 mHandles.insertAt(handle, i); 8072 return status; 8073} 8074 8075size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8076{ 8077 Mutex::Autolock _l(mLock); 8078 size_t size = mHandles.size(); 8079 size_t i; 8080 for (i = 0; i < size; i++) { 8081 if (mHandles[i] == handle) break; 8082 } 8083 if (i == size) { 8084 return size; 8085 } 8086 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8087 8088 bool enabled = false; 8089 EffectHandle *hdl = handle.unsafe_get(); 8090 if (hdl != NULL) { 8091 ALOGV("removeHandle() unsafe_get OK"); 8092 enabled = hdl->enabled(); 8093 } 8094 mHandles.removeAt(i); 8095 size = mHandles.size(); 8096 // if removed from first place, move effect control from this handle to next in line 8097 if (i == 0 && size != 0) { 8098 sp<EffectHandle> h = mHandles[0].promote(); 8099 if (h != 0) { 8100 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8101 } 8102 } 8103 8104 // Prevent calls to process() and other functions on effect interface from now on. 8105 // The effect engine will be released by the destructor when the last strong reference on 8106 // this object is released which can happen after next process is called. 8107 if (size == 0 && !mPinned) { 8108 mState = DESTROYED; 8109 } 8110 8111 return size; 8112} 8113 8114sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8115{ 8116 Mutex::Autolock _l(mLock); 8117 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8118} 8119 8120void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8121{ 8122 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8123 // keep a strong reference on this EffectModule to avoid calling the 8124 // destructor before we exit 8125 sp<EffectModule> keep(this); 8126 { 8127 sp<ThreadBase> thread = mThread.promote(); 8128 if (thread != 0) { 8129 thread->disconnectEffect(keep, handle, unpinIfLast); 8130 } 8131 } 8132} 8133 8134void AudioFlinger::EffectModule::updateState() { 8135 Mutex::Autolock _l(mLock); 8136 8137 switch (mState) { 8138 case RESTART: 8139 reset_l(); 8140 // FALL THROUGH 8141 8142 case STARTING: 8143 // clear auxiliary effect input buffer for next accumulation 8144 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8145 memset(mConfig.inputCfg.buffer.raw, 8146 0, 8147 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8148 } 8149 start_l(); 8150 mState = ACTIVE; 8151 break; 8152 case STOPPING: 8153 stop_l(); 8154 mDisableWaitCnt = mMaxDisableWaitCnt; 8155 mState = STOPPED; 8156 break; 8157 case STOPPED: 8158 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8159 // turn off sequence. 8160 if (--mDisableWaitCnt == 0) { 8161 reset_l(); 8162 mState = IDLE; 8163 } 8164 break; 8165 default: //IDLE , ACTIVE, DESTROYED 8166 break; 8167 } 8168} 8169 8170void AudioFlinger::EffectModule::process() 8171{ 8172 Mutex::Autolock _l(mLock); 8173 8174 if (mState == DESTROYED || mEffectInterface == NULL || 8175 mConfig.inputCfg.buffer.raw == NULL || 8176 mConfig.outputCfg.buffer.raw == NULL) { 8177 return; 8178 } 8179 8180 if (isProcessEnabled()) { 8181 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8182 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8183 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8184 mConfig.inputCfg.buffer.s32, 8185 mConfig.inputCfg.buffer.frameCount/2); 8186 } 8187 8188 // do the actual processing in the effect engine 8189 int ret = (*mEffectInterface)->process(mEffectInterface, 8190 &mConfig.inputCfg.buffer, 8191 &mConfig.outputCfg.buffer); 8192 8193 // force transition to IDLE state when engine is ready 8194 if (mState == STOPPED && ret == -ENODATA) { 8195 mDisableWaitCnt = 1; 8196 } 8197 8198 // clear auxiliary effect input buffer for next accumulation 8199 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8200 memset(mConfig.inputCfg.buffer.raw, 0, 8201 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8202 } 8203 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8204 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8205 // If an insert effect is idle and input buffer is different from output buffer, 8206 // accumulate input onto output 8207 sp<EffectChain> chain = mChain.promote(); 8208 if (chain != 0 && chain->activeTrackCnt() != 0) { 8209 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8210 int16_t *in = mConfig.inputCfg.buffer.s16; 8211 int16_t *out = mConfig.outputCfg.buffer.s16; 8212 for (size_t i = 0; i < frameCnt; i++) { 8213 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8214 } 8215 } 8216 } 8217} 8218 8219void AudioFlinger::EffectModule::reset_l() 8220{ 8221 if (mEffectInterface == NULL) { 8222 return; 8223 } 8224 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8225} 8226 8227status_t AudioFlinger::EffectModule::configure() 8228{ 8229 uint32_t channels; 8230 if (mEffectInterface == NULL) { 8231 return NO_INIT; 8232 } 8233 8234 sp<ThreadBase> thread = mThread.promote(); 8235 if (thread == 0) { 8236 return DEAD_OBJECT; 8237 } 8238 8239 // TODO: handle configuration of effects replacing track process 8240 if (thread->channelCount() == 1) { 8241 channels = AUDIO_CHANNEL_OUT_MONO; 8242 } else { 8243 channels = AUDIO_CHANNEL_OUT_STEREO; 8244 } 8245 8246 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8247 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8248 } else { 8249 mConfig.inputCfg.channels = channels; 8250 } 8251 mConfig.outputCfg.channels = channels; 8252 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8253 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8254 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8255 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8256 mConfig.inputCfg.bufferProvider.cookie = NULL; 8257 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8258 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8259 mConfig.outputCfg.bufferProvider.cookie = NULL; 8260 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8261 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8262 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8263 // Insert effect: 8264 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8265 // always overwrites output buffer: input buffer == output buffer 8266 // - in other sessions: 8267 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8268 // other effect: overwrites output buffer: input buffer == output buffer 8269 // Auxiliary effect: 8270 // accumulates in output buffer: input buffer != output buffer 8271 // Therefore: accumulate <=> input buffer != output buffer 8272 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8273 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8274 } else { 8275 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8276 } 8277 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8278 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8279 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8280 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8281 8282 ALOGV("configure() %p thread %p buffer %p framecount %d", 8283 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8284 8285 status_t cmdStatus; 8286 uint32_t size = sizeof(int); 8287 status_t status = (*mEffectInterface)->command(mEffectInterface, 8288 EFFECT_CMD_SET_CONFIG, 8289 sizeof(effect_config_t), 8290 &mConfig, 8291 &size, 8292 &cmdStatus); 8293 if (status == 0) { 8294 status = cmdStatus; 8295 } 8296 8297 if (status == 0 && 8298 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8299 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8300 effect_param_t *p = (effect_param_t *)buf32; 8301 8302 p->psize = sizeof(uint32_t); 8303 p->vsize = sizeof(uint32_t); 8304 size = sizeof(int); 8305 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8306 8307 uint32_t latency = 0; 8308 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8309 if (pbt != NULL) { 8310 latency = pbt->latency_l(); 8311 } 8312 8313 *((int32_t *)p->data + 1)= latency; 8314 (*mEffectInterface)->command(mEffectInterface, 8315 EFFECT_CMD_SET_PARAM, 8316 sizeof(effect_param_t) + 8, 8317 &buf32, 8318 &size, 8319 &cmdStatus); 8320 } 8321 8322 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8323 (1000 * mConfig.outputCfg.buffer.frameCount); 8324 8325 return status; 8326} 8327 8328status_t AudioFlinger::EffectModule::init() 8329{ 8330 Mutex::Autolock _l(mLock); 8331 if (mEffectInterface == NULL) { 8332 return NO_INIT; 8333 } 8334 status_t cmdStatus; 8335 uint32_t size = sizeof(status_t); 8336 status_t status = (*mEffectInterface)->command(mEffectInterface, 8337 EFFECT_CMD_INIT, 8338 0, 8339 NULL, 8340 &size, 8341 &cmdStatus); 8342 if (status == 0) { 8343 status = cmdStatus; 8344 } 8345 return status; 8346} 8347 8348status_t AudioFlinger::EffectModule::start() 8349{ 8350 Mutex::Autolock _l(mLock); 8351 return start_l(); 8352} 8353 8354status_t AudioFlinger::EffectModule::start_l() 8355{ 8356 if (mEffectInterface == NULL) { 8357 return NO_INIT; 8358 } 8359 status_t cmdStatus; 8360 uint32_t size = sizeof(status_t); 8361 status_t status = (*mEffectInterface)->command(mEffectInterface, 8362 EFFECT_CMD_ENABLE, 8363 0, 8364 NULL, 8365 &size, 8366 &cmdStatus); 8367 if (status == 0) { 8368 status = cmdStatus; 8369 } 8370 if (status == 0 && 8371 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8372 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8373 sp<ThreadBase> thread = mThread.promote(); 8374 if (thread != 0) { 8375 audio_stream_t *stream = thread->stream(); 8376 if (stream != NULL) { 8377 stream->add_audio_effect(stream, mEffectInterface); 8378 } 8379 } 8380 } 8381 return status; 8382} 8383 8384status_t AudioFlinger::EffectModule::stop() 8385{ 8386 Mutex::Autolock _l(mLock); 8387 return stop_l(); 8388} 8389 8390status_t AudioFlinger::EffectModule::stop_l() 8391{ 8392 if (mEffectInterface == NULL) { 8393 return NO_INIT; 8394 } 8395 status_t cmdStatus; 8396 uint32_t size = sizeof(status_t); 8397 status_t status = (*mEffectInterface)->command(mEffectInterface, 8398 EFFECT_CMD_DISABLE, 8399 0, 8400 NULL, 8401 &size, 8402 &cmdStatus); 8403 if (status == 0) { 8404 status = cmdStatus; 8405 } 8406 if (status == 0 && 8407 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8408 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8409 sp<ThreadBase> thread = mThread.promote(); 8410 if (thread != 0) { 8411 audio_stream_t *stream = thread->stream(); 8412 if (stream != NULL) { 8413 stream->remove_audio_effect(stream, mEffectInterface); 8414 } 8415 } 8416 } 8417 return status; 8418} 8419 8420status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8421 uint32_t cmdSize, 8422 void *pCmdData, 8423 uint32_t *replySize, 8424 void *pReplyData) 8425{ 8426 Mutex::Autolock _l(mLock); 8427// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8428 8429 if (mState == DESTROYED || mEffectInterface == NULL) { 8430 return NO_INIT; 8431 } 8432 status_t status = (*mEffectInterface)->command(mEffectInterface, 8433 cmdCode, 8434 cmdSize, 8435 pCmdData, 8436 replySize, 8437 pReplyData); 8438 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8439 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8440 for (size_t i = 1; i < mHandles.size(); i++) { 8441 sp<EffectHandle> h = mHandles[i].promote(); 8442 if (h != 0) { 8443 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8444 } 8445 } 8446 } 8447 return status; 8448} 8449 8450status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8451{ 8452 8453 Mutex::Autolock _l(mLock); 8454 ALOGV("setEnabled %p enabled %d", this, enabled); 8455 8456 if (enabled != isEnabled()) { 8457 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8458 if (enabled && status != NO_ERROR) { 8459 return status; 8460 } 8461 8462 switch (mState) { 8463 // going from disabled to enabled 8464 case IDLE: 8465 mState = STARTING; 8466 break; 8467 case STOPPED: 8468 mState = RESTART; 8469 break; 8470 case STOPPING: 8471 mState = ACTIVE; 8472 break; 8473 8474 // going from enabled to disabled 8475 case RESTART: 8476 mState = STOPPED; 8477 break; 8478 case STARTING: 8479 mState = IDLE; 8480 break; 8481 case ACTIVE: 8482 mState = STOPPING; 8483 break; 8484 case DESTROYED: 8485 return NO_ERROR; // simply ignore as we are being destroyed 8486 } 8487 for (size_t i = 1; i < mHandles.size(); i++) { 8488 sp<EffectHandle> h = mHandles[i].promote(); 8489 if (h != 0) { 8490 h->setEnabled(enabled); 8491 } 8492 } 8493 } 8494 return NO_ERROR; 8495} 8496 8497bool AudioFlinger::EffectModule::isEnabled() const 8498{ 8499 switch (mState) { 8500 case RESTART: 8501 case STARTING: 8502 case ACTIVE: 8503 return true; 8504 case IDLE: 8505 case STOPPING: 8506 case STOPPED: 8507 case DESTROYED: 8508 default: 8509 return false; 8510 } 8511} 8512 8513bool AudioFlinger::EffectModule::isProcessEnabled() const 8514{ 8515 switch (mState) { 8516 case RESTART: 8517 case ACTIVE: 8518 case STOPPING: 8519 case STOPPED: 8520 return true; 8521 case IDLE: 8522 case STARTING: 8523 case DESTROYED: 8524 default: 8525 return false; 8526 } 8527} 8528 8529status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8530{ 8531 Mutex::Autolock _l(mLock); 8532 status_t status = NO_ERROR; 8533 8534 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8535 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8536 if (isProcessEnabled() && 8537 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8538 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8539 status_t cmdStatus; 8540 uint32_t volume[2]; 8541 uint32_t *pVolume = NULL; 8542 uint32_t size = sizeof(volume); 8543 volume[0] = *left; 8544 volume[1] = *right; 8545 if (controller) { 8546 pVolume = volume; 8547 } 8548 status = (*mEffectInterface)->command(mEffectInterface, 8549 EFFECT_CMD_SET_VOLUME, 8550 size, 8551 volume, 8552 &size, 8553 pVolume); 8554 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8555 *left = volume[0]; 8556 *right = volume[1]; 8557 } 8558 } 8559 return status; 8560} 8561 8562status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8563{ 8564 Mutex::Autolock _l(mLock); 8565 status_t status = NO_ERROR; 8566 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8567 // audio pre processing modules on RecordThread can receive both output and 8568 // input device indication in the same call 8569 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8570 if (dev) { 8571 status_t cmdStatus; 8572 uint32_t size = sizeof(status_t); 8573 8574 status = (*mEffectInterface)->command(mEffectInterface, 8575 EFFECT_CMD_SET_DEVICE, 8576 sizeof(uint32_t), 8577 &dev, 8578 &size, 8579 &cmdStatus); 8580 if (status == NO_ERROR) { 8581 status = cmdStatus; 8582 } 8583 } 8584 dev = device & AUDIO_DEVICE_IN_ALL; 8585 if (dev) { 8586 status_t cmdStatus; 8587 uint32_t size = sizeof(status_t); 8588 8589 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8590 EFFECT_CMD_SET_INPUT_DEVICE, 8591 sizeof(uint32_t), 8592 &dev, 8593 &size, 8594 &cmdStatus); 8595 if (status2 == NO_ERROR) { 8596 status2 = cmdStatus; 8597 } 8598 if (status == NO_ERROR) { 8599 status = status2; 8600 } 8601 } 8602 } 8603 return status; 8604} 8605 8606status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8607{ 8608 Mutex::Autolock _l(mLock); 8609 status_t status = NO_ERROR; 8610 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8611 status_t cmdStatus; 8612 uint32_t size = sizeof(status_t); 8613 status = (*mEffectInterface)->command(mEffectInterface, 8614 EFFECT_CMD_SET_AUDIO_MODE, 8615 sizeof(audio_mode_t), 8616 &mode, 8617 &size, 8618 &cmdStatus); 8619 if (status == NO_ERROR) { 8620 status = cmdStatus; 8621 } 8622 } 8623 return status; 8624} 8625 8626void AudioFlinger::EffectModule::setSuspended(bool suspended) 8627{ 8628 Mutex::Autolock _l(mLock); 8629 mSuspended = suspended; 8630} 8631 8632bool AudioFlinger::EffectModule::suspended() const 8633{ 8634 Mutex::Autolock _l(mLock); 8635 return mSuspended; 8636} 8637 8638status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8639{ 8640 const size_t SIZE = 256; 8641 char buffer[SIZE]; 8642 String8 result; 8643 8644 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8645 result.append(buffer); 8646 8647 bool locked = tryLock(mLock); 8648 // failed to lock - AudioFlinger is probably deadlocked 8649 if (!locked) { 8650 result.append("\t\tCould not lock Fx mutex:\n"); 8651 } 8652 8653 result.append("\t\tSession Status State Engine:\n"); 8654 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8655 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8656 result.append(buffer); 8657 8658 result.append("\t\tDescriptor:\n"); 8659 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8660 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8661 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8662 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8663 result.append(buffer); 8664 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8665 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8666 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8667 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8668 result.append(buffer); 8669 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8670 mDescriptor.apiVersion, 8671 mDescriptor.flags); 8672 result.append(buffer); 8673 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8674 mDescriptor.name); 8675 result.append(buffer); 8676 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8677 mDescriptor.implementor); 8678 result.append(buffer); 8679 8680 result.append("\t\t- Input configuration:\n"); 8681 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8682 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8683 (uint32_t)mConfig.inputCfg.buffer.raw, 8684 mConfig.inputCfg.buffer.frameCount, 8685 mConfig.inputCfg.samplingRate, 8686 mConfig.inputCfg.channels, 8687 mConfig.inputCfg.format); 8688 result.append(buffer); 8689 8690 result.append("\t\t- Output configuration:\n"); 8691 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8692 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8693 (uint32_t)mConfig.outputCfg.buffer.raw, 8694 mConfig.outputCfg.buffer.frameCount, 8695 mConfig.outputCfg.samplingRate, 8696 mConfig.outputCfg.channels, 8697 mConfig.outputCfg.format); 8698 result.append(buffer); 8699 8700 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8701 result.append(buffer); 8702 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8703 for (size_t i = 0; i < mHandles.size(); ++i) { 8704 sp<EffectHandle> handle = mHandles[i].promote(); 8705 if (handle != 0) { 8706 handle->dump(buffer, SIZE); 8707 result.append(buffer); 8708 } 8709 } 8710 8711 result.append("\n"); 8712 8713 write(fd, result.string(), result.length()); 8714 8715 if (locked) { 8716 mLock.unlock(); 8717 } 8718 8719 return NO_ERROR; 8720} 8721 8722// ---------------------------------------------------------------------------- 8723// EffectHandle implementation 8724// ---------------------------------------------------------------------------- 8725 8726#undef LOG_TAG 8727#define LOG_TAG "AudioFlinger::EffectHandle" 8728 8729AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8730 const sp<AudioFlinger::Client>& client, 8731 const sp<IEffectClient>& effectClient, 8732 int32_t priority) 8733 : BnEffect(), 8734 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8735 mPriority(priority), mHasControl(false), mEnabled(false) 8736{ 8737 ALOGV("constructor %p", this); 8738 8739 if (client == 0) { 8740 return; 8741 } 8742 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8743 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8744 if (mCblkMemory != 0) { 8745 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8746 8747 if (mCblk != NULL) { 8748 new(mCblk) effect_param_cblk_t(); 8749 mBuffer = (uint8_t *)mCblk + bufOffset; 8750 } 8751 } else { 8752 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8753 return; 8754 } 8755} 8756 8757AudioFlinger::EffectHandle::~EffectHandle() 8758{ 8759 ALOGV("Destructor %p", this); 8760 disconnect(false); 8761 ALOGV("Destructor DONE %p", this); 8762} 8763 8764status_t AudioFlinger::EffectHandle::enable() 8765{ 8766 ALOGV("enable %p", this); 8767 if (!mHasControl) return INVALID_OPERATION; 8768 if (mEffect == 0) return DEAD_OBJECT; 8769 8770 if (mEnabled) { 8771 return NO_ERROR; 8772 } 8773 8774 mEnabled = true; 8775 8776 sp<ThreadBase> thread = mEffect->thread().promote(); 8777 if (thread != 0) { 8778 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8779 } 8780 8781 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8782 if (mEffect->suspended()) { 8783 return NO_ERROR; 8784 } 8785 8786 status_t status = mEffect->setEnabled(true); 8787 if (status != NO_ERROR) { 8788 if (thread != 0) { 8789 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8790 } 8791 mEnabled = false; 8792 } 8793 return status; 8794} 8795 8796status_t AudioFlinger::EffectHandle::disable() 8797{ 8798 ALOGV("disable %p", this); 8799 if (!mHasControl) return INVALID_OPERATION; 8800 if (mEffect == 0) return DEAD_OBJECT; 8801 8802 if (!mEnabled) { 8803 return NO_ERROR; 8804 } 8805 mEnabled = false; 8806 8807 if (mEffect->suspended()) { 8808 return NO_ERROR; 8809 } 8810 8811 status_t status = mEffect->setEnabled(false); 8812 8813 sp<ThreadBase> thread = mEffect->thread().promote(); 8814 if (thread != 0) { 8815 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8816 } 8817 8818 return status; 8819} 8820 8821void AudioFlinger::EffectHandle::disconnect() 8822{ 8823 disconnect(true); 8824} 8825 8826void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8827{ 8828 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8829 if (mEffect == 0) { 8830 return; 8831 } 8832 mEffect->disconnect(this, unpinIfLast); 8833 8834 if (mHasControl && mEnabled) { 8835 sp<ThreadBase> thread = mEffect->thread().promote(); 8836 if (thread != 0) { 8837 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8838 } 8839 } 8840 8841 // release sp on module => module destructor can be called now 8842 mEffect.clear(); 8843 if (mClient != 0) { 8844 if (mCblk != NULL) { 8845 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8846 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8847 } 8848 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8849 // Client destructor must run with AudioFlinger mutex locked 8850 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8851 mClient.clear(); 8852 } 8853} 8854 8855status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8856 uint32_t cmdSize, 8857 void *pCmdData, 8858 uint32_t *replySize, 8859 void *pReplyData) 8860{ 8861// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8862// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8863 8864 // only get parameter command is permitted for applications not controlling the effect 8865 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8866 return INVALID_OPERATION; 8867 } 8868 if (mEffect == 0) return DEAD_OBJECT; 8869 if (mClient == 0) return INVALID_OPERATION; 8870 8871 // handle commands that are not forwarded transparently to effect engine 8872 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8873 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8874 // no risk to block the whole media server process or mixer threads is we are stuck here 8875 Mutex::Autolock _l(mCblk->lock); 8876 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8877 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8878 mCblk->serverIndex = 0; 8879 mCblk->clientIndex = 0; 8880 return BAD_VALUE; 8881 } 8882 status_t status = NO_ERROR; 8883 while (mCblk->serverIndex < mCblk->clientIndex) { 8884 int reply; 8885 uint32_t rsize = sizeof(int); 8886 int *p = (int *)(mBuffer + mCblk->serverIndex); 8887 int size = *p++; 8888 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8889 ALOGW("command(): invalid parameter block size"); 8890 break; 8891 } 8892 effect_param_t *param = (effect_param_t *)p; 8893 if (param->psize == 0 || param->vsize == 0) { 8894 ALOGW("command(): null parameter or value size"); 8895 mCblk->serverIndex += size; 8896 continue; 8897 } 8898 uint32_t psize = sizeof(effect_param_t) + 8899 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8900 param->vsize; 8901 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8902 psize, 8903 p, 8904 &rsize, 8905 &reply); 8906 // stop at first error encountered 8907 if (ret != NO_ERROR) { 8908 status = ret; 8909 *(int *)pReplyData = reply; 8910 break; 8911 } else if (reply != NO_ERROR) { 8912 *(int *)pReplyData = reply; 8913 break; 8914 } 8915 mCblk->serverIndex += size; 8916 } 8917 mCblk->serverIndex = 0; 8918 mCblk->clientIndex = 0; 8919 return status; 8920 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8921 *(int *)pReplyData = NO_ERROR; 8922 return enable(); 8923 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8924 *(int *)pReplyData = NO_ERROR; 8925 return disable(); 8926 } 8927 8928 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8929} 8930 8931void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8932{ 8933 ALOGV("setControl %p control %d", this, hasControl); 8934 8935 mHasControl = hasControl; 8936 mEnabled = enabled; 8937 8938 if (signal && mEffectClient != 0) { 8939 mEffectClient->controlStatusChanged(hasControl); 8940 } 8941} 8942 8943void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8944 uint32_t cmdSize, 8945 void *pCmdData, 8946 uint32_t replySize, 8947 void *pReplyData) 8948{ 8949 if (mEffectClient != 0) { 8950 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8951 } 8952} 8953 8954 8955 8956void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8957{ 8958 if (mEffectClient != 0) { 8959 mEffectClient->enableStatusChanged(enabled); 8960 } 8961} 8962 8963status_t AudioFlinger::EffectHandle::onTransact( 8964 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8965{ 8966 return BnEffect::onTransact(code, data, reply, flags); 8967} 8968 8969 8970void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8971{ 8972 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8973 8974 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8975 (mClient == 0) ? getpid_cached : mClient->pid(), 8976 mPriority, 8977 mHasControl, 8978 !locked, 8979 mCblk ? mCblk->clientIndex : 0, 8980 mCblk ? mCblk->serverIndex : 0 8981 ); 8982 8983 if (locked) { 8984 mCblk->lock.unlock(); 8985 } 8986} 8987 8988#undef LOG_TAG 8989#define LOG_TAG "AudioFlinger::EffectChain" 8990 8991AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8992 int sessionId) 8993 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8994 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8995 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8996{ 8997 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8998 if (thread == NULL) { 8999 return; 9000 } 9001 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9002 thread->frameCount(); 9003} 9004 9005AudioFlinger::EffectChain::~EffectChain() 9006{ 9007 if (mOwnInBuffer) { 9008 delete mInBuffer; 9009 } 9010 9011} 9012 9013// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9014sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9015{ 9016 size_t size = mEffects.size(); 9017 9018 for (size_t i = 0; i < size; i++) { 9019 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9020 return mEffects[i]; 9021 } 9022 } 9023 return 0; 9024} 9025 9026// getEffectFromId_l() must be called with ThreadBase::mLock held 9027sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9028{ 9029 size_t size = mEffects.size(); 9030 9031 for (size_t i = 0; i < size; i++) { 9032 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9033 if (id == 0 || mEffects[i]->id() == id) { 9034 return mEffects[i]; 9035 } 9036 } 9037 return 0; 9038} 9039 9040// getEffectFromType_l() must be called with ThreadBase::mLock held 9041sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9042 const effect_uuid_t *type) 9043{ 9044 size_t size = mEffects.size(); 9045 9046 for (size_t i = 0; i < size; i++) { 9047 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9048 return mEffects[i]; 9049 } 9050 } 9051 return 0; 9052} 9053 9054void AudioFlinger::EffectChain::clearInputBuffer() 9055{ 9056 Mutex::Autolock _l(mLock); 9057 sp<ThreadBase> thread = mThread.promote(); 9058 if (thread == 0) { 9059 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9060 return; 9061 } 9062 clearInputBuffer_l(thread); 9063} 9064 9065// Must be called with EffectChain::mLock locked 9066void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9067{ 9068 size_t numSamples = thread->frameCount() * thread->channelCount(); 9069 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9070 9071} 9072 9073// Must be called with EffectChain::mLock locked 9074void AudioFlinger::EffectChain::process_l() 9075{ 9076 sp<ThreadBase> thread = mThread.promote(); 9077 if (thread == 0) { 9078 ALOGW("process_l(): cannot promote mixer thread"); 9079 return; 9080 } 9081 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9082 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9083 // always process effects unless no more tracks are on the session and the effect tail 9084 // has been rendered 9085 bool doProcess = true; 9086 if (!isGlobalSession) { 9087 bool tracksOnSession = (trackCnt() != 0); 9088 9089 if (!tracksOnSession && mTailBufferCount == 0) { 9090 doProcess = false; 9091 } 9092 9093 if (activeTrackCnt() == 0) { 9094 // if no track is active and the effect tail has not been rendered, 9095 // the input buffer must be cleared here as the mixer process will not do it 9096 if (tracksOnSession || mTailBufferCount > 0) { 9097 clearInputBuffer_l(thread); 9098 if (mTailBufferCount > 0) { 9099 mTailBufferCount--; 9100 } 9101 } 9102 } 9103 } 9104 9105 size_t size = mEffects.size(); 9106 if (doProcess) { 9107 for (size_t i = 0; i < size; i++) { 9108 mEffects[i]->process(); 9109 } 9110 } 9111 for (size_t i = 0; i < size; i++) { 9112 mEffects[i]->updateState(); 9113 } 9114} 9115 9116// addEffect_l() must be called with PlaybackThread::mLock held 9117status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9118{ 9119 effect_descriptor_t desc = effect->desc(); 9120 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9121 9122 Mutex::Autolock _l(mLock); 9123 effect->setChain(this); 9124 sp<ThreadBase> thread = mThread.promote(); 9125 if (thread == 0) { 9126 return NO_INIT; 9127 } 9128 effect->setThread(thread); 9129 9130 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9131 // Auxiliary effects are inserted at the beginning of mEffects vector as 9132 // they are processed first and accumulated in chain input buffer 9133 mEffects.insertAt(effect, 0); 9134 9135 // the input buffer for auxiliary effect contains mono samples in 9136 // 32 bit format. This is to avoid saturation in AudoMixer 9137 // accumulation stage. Saturation is done in EffectModule::process() before 9138 // calling the process in effect engine 9139 size_t numSamples = thread->frameCount(); 9140 int32_t *buffer = new int32_t[numSamples]; 9141 memset(buffer, 0, numSamples * sizeof(int32_t)); 9142 effect->setInBuffer((int16_t *)buffer); 9143 // auxiliary effects output samples to chain input buffer for further processing 9144 // by insert effects 9145 effect->setOutBuffer(mInBuffer); 9146 } else { 9147 // Insert effects are inserted at the end of mEffects vector as they are processed 9148 // after track and auxiliary effects. 9149 // Insert effect order as a function of indicated preference: 9150 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9151 // another effect is present 9152 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9153 // last effect claiming first position 9154 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9155 // first effect claiming last position 9156 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9157 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9158 // already present 9159 9160 size_t size = mEffects.size(); 9161 size_t idx_insert = size; 9162 ssize_t idx_insert_first = -1; 9163 ssize_t idx_insert_last = -1; 9164 9165 for (size_t i = 0; i < size; i++) { 9166 effect_descriptor_t d = mEffects[i]->desc(); 9167 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9168 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9169 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9170 // check invalid effect chaining combinations 9171 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9172 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9173 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9174 return INVALID_OPERATION; 9175 } 9176 // remember position of first insert effect and by default 9177 // select this as insert position for new effect 9178 if (idx_insert == size) { 9179 idx_insert = i; 9180 } 9181 // remember position of last insert effect claiming 9182 // first position 9183 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9184 idx_insert_first = i; 9185 } 9186 // remember position of first insert effect claiming 9187 // last position 9188 if (iPref == EFFECT_FLAG_INSERT_LAST && 9189 idx_insert_last == -1) { 9190 idx_insert_last = i; 9191 } 9192 } 9193 } 9194 9195 // modify idx_insert from first position if needed 9196 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9197 if (idx_insert_last != -1) { 9198 idx_insert = idx_insert_last; 9199 } else { 9200 idx_insert = size; 9201 } 9202 } else { 9203 if (idx_insert_first != -1) { 9204 idx_insert = idx_insert_first + 1; 9205 } 9206 } 9207 9208 // always read samples from chain input buffer 9209 effect->setInBuffer(mInBuffer); 9210 9211 // if last effect in the chain, output samples to chain 9212 // output buffer, otherwise to chain input buffer 9213 if (idx_insert == size) { 9214 if (idx_insert != 0) { 9215 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9216 mEffects[idx_insert-1]->configure(); 9217 } 9218 effect->setOutBuffer(mOutBuffer); 9219 } else { 9220 effect->setOutBuffer(mInBuffer); 9221 } 9222 mEffects.insertAt(effect, idx_insert); 9223 9224 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9225 } 9226 effect->configure(); 9227 return NO_ERROR; 9228} 9229 9230// removeEffect_l() must be called with PlaybackThread::mLock held 9231size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9232{ 9233 Mutex::Autolock _l(mLock); 9234 size_t size = mEffects.size(); 9235 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9236 9237 for (size_t i = 0; i < size; i++) { 9238 if (effect == mEffects[i]) { 9239 // calling stop here will remove pre-processing effect from the audio HAL. 9240 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9241 // the middle of a read from audio HAL 9242 if (mEffects[i]->state() == EffectModule::ACTIVE || 9243 mEffects[i]->state() == EffectModule::STOPPING) { 9244 mEffects[i]->stop(); 9245 } 9246 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9247 delete[] effect->inBuffer(); 9248 } else { 9249 if (i == size - 1 && i != 0) { 9250 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9251 mEffects[i - 1]->configure(); 9252 } 9253 } 9254 mEffects.removeAt(i); 9255 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9256 break; 9257 } 9258 } 9259 9260 return mEffects.size(); 9261} 9262 9263// setDevice_l() must be called with PlaybackThread::mLock held 9264void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9265{ 9266 size_t size = mEffects.size(); 9267 for (size_t i = 0; i < size; i++) { 9268 mEffects[i]->setDevice(device); 9269 } 9270} 9271 9272// setMode_l() must be called with PlaybackThread::mLock held 9273void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9274{ 9275 size_t size = mEffects.size(); 9276 for (size_t i = 0; i < size; i++) { 9277 mEffects[i]->setMode(mode); 9278 } 9279} 9280 9281// setVolume_l() must be called with PlaybackThread::mLock held 9282bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9283{ 9284 uint32_t newLeft = *left; 9285 uint32_t newRight = *right; 9286 bool hasControl = false; 9287 int ctrlIdx = -1; 9288 size_t size = mEffects.size(); 9289 9290 // first update volume controller 9291 for (size_t i = size; i > 0; i--) { 9292 if (mEffects[i - 1]->isProcessEnabled() && 9293 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9294 ctrlIdx = i - 1; 9295 hasControl = true; 9296 break; 9297 } 9298 } 9299 9300 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9301 if (hasControl) { 9302 *left = mNewLeftVolume; 9303 *right = mNewRightVolume; 9304 } 9305 return hasControl; 9306 } 9307 9308 mVolumeCtrlIdx = ctrlIdx; 9309 mLeftVolume = newLeft; 9310 mRightVolume = newRight; 9311 9312 // second get volume update from volume controller 9313 if (ctrlIdx >= 0) { 9314 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9315 mNewLeftVolume = newLeft; 9316 mNewRightVolume = newRight; 9317 } 9318 // then indicate volume to all other effects in chain. 9319 // Pass altered volume to effects before volume controller 9320 // and requested volume to effects after controller 9321 uint32_t lVol = newLeft; 9322 uint32_t rVol = newRight; 9323 9324 for (size_t i = 0; i < size; i++) { 9325 if ((int)i == ctrlIdx) continue; 9326 // this also works for ctrlIdx == -1 when there is no volume controller 9327 if ((int)i > ctrlIdx) { 9328 lVol = *left; 9329 rVol = *right; 9330 } 9331 mEffects[i]->setVolume(&lVol, &rVol, false); 9332 } 9333 *left = newLeft; 9334 *right = newRight; 9335 9336 return hasControl; 9337} 9338 9339status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9340{ 9341 const size_t SIZE = 256; 9342 char buffer[SIZE]; 9343 String8 result; 9344 9345 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9346 result.append(buffer); 9347 9348 bool locked = tryLock(mLock); 9349 // failed to lock - AudioFlinger is probably deadlocked 9350 if (!locked) { 9351 result.append("\tCould not lock mutex:\n"); 9352 } 9353 9354 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9355 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9356 mEffects.size(), 9357 (uint32_t)mInBuffer, 9358 (uint32_t)mOutBuffer, 9359 mActiveTrackCnt); 9360 result.append(buffer); 9361 write(fd, result.string(), result.size()); 9362 9363 for (size_t i = 0; i < mEffects.size(); ++i) { 9364 sp<EffectModule> effect = mEffects[i]; 9365 if (effect != 0) { 9366 effect->dump(fd, args); 9367 } 9368 } 9369 9370 if (locked) { 9371 mLock.unlock(); 9372 } 9373 9374 return NO_ERROR; 9375} 9376 9377// must be called with ThreadBase::mLock held 9378void AudioFlinger::EffectChain::setEffectSuspended_l( 9379 const effect_uuid_t *type, bool suspend) 9380{ 9381 sp<SuspendedEffectDesc> desc; 9382 // use effect type UUID timelow as key as there is no real risk of identical 9383 // timeLow fields among effect type UUIDs. 9384 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9385 if (suspend) { 9386 if (index >= 0) { 9387 desc = mSuspendedEffects.valueAt(index); 9388 } else { 9389 desc = new SuspendedEffectDesc(); 9390 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9391 mSuspendedEffects.add(type->timeLow, desc); 9392 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9393 } 9394 if (desc->mRefCount++ == 0) { 9395 sp<EffectModule> effect = getEffectIfEnabled(type); 9396 if (effect != 0) { 9397 desc->mEffect = effect; 9398 effect->setSuspended(true); 9399 effect->setEnabled(false); 9400 } 9401 } 9402 } else { 9403 if (index < 0) { 9404 return; 9405 } 9406 desc = mSuspendedEffects.valueAt(index); 9407 if (desc->mRefCount <= 0) { 9408 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9409 desc->mRefCount = 1; 9410 } 9411 if (--desc->mRefCount == 0) { 9412 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9413 if (desc->mEffect != 0) { 9414 sp<EffectModule> effect = desc->mEffect.promote(); 9415 if (effect != 0) { 9416 effect->setSuspended(false); 9417 sp<EffectHandle> handle = effect->controlHandle(); 9418 if (handle != 0) { 9419 effect->setEnabled(handle->enabled()); 9420 } 9421 } 9422 desc->mEffect.clear(); 9423 } 9424 mSuspendedEffects.removeItemsAt(index); 9425 } 9426 } 9427} 9428 9429// must be called with ThreadBase::mLock held 9430void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9431{ 9432 sp<SuspendedEffectDesc> desc; 9433 9434 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9435 if (suspend) { 9436 if (index >= 0) { 9437 desc = mSuspendedEffects.valueAt(index); 9438 } else { 9439 desc = new SuspendedEffectDesc(); 9440 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9441 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9442 } 9443 if (desc->mRefCount++ == 0) { 9444 Vector< sp<EffectModule> > effects; 9445 getSuspendEligibleEffects(effects); 9446 for (size_t i = 0; i < effects.size(); i++) { 9447 setEffectSuspended_l(&effects[i]->desc().type, true); 9448 } 9449 } 9450 } else { 9451 if (index < 0) { 9452 return; 9453 } 9454 desc = mSuspendedEffects.valueAt(index); 9455 if (desc->mRefCount <= 0) { 9456 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9457 desc->mRefCount = 1; 9458 } 9459 if (--desc->mRefCount == 0) { 9460 Vector<const effect_uuid_t *> types; 9461 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9462 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9463 continue; 9464 } 9465 types.add(&mSuspendedEffects.valueAt(i)->mType); 9466 } 9467 for (size_t i = 0; i < types.size(); i++) { 9468 setEffectSuspended_l(types[i], false); 9469 } 9470 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9471 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9472 } 9473 } 9474} 9475 9476 9477// The volume effect is used for automated tests only 9478#ifndef OPENSL_ES_H_ 9479static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9480 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9481const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9482#endif //OPENSL_ES_H_ 9483 9484bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9485{ 9486 // auxiliary effects and visualizer are never suspended on output mix 9487 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9488 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9489 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9490 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9491 return false; 9492 } 9493 return true; 9494} 9495 9496void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9497{ 9498 effects.clear(); 9499 for (size_t i = 0; i < mEffects.size(); i++) { 9500 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9501 effects.add(mEffects[i]); 9502 } 9503 } 9504} 9505 9506sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9507 const effect_uuid_t *type) 9508{ 9509 sp<EffectModule> effect = getEffectFromType_l(type); 9510 return effect != 0 && effect->isEnabled() ? effect : 0; 9511} 9512 9513void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9514 bool enabled) 9515{ 9516 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9517 if (enabled) { 9518 if (index < 0) { 9519 // if the effect is not suspend check if all effects are suspended 9520 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9521 if (index < 0) { 9522 return; 9523 } 9524 if (!isEffectEligibleForSuspend(effect->desc())) { 9525 return; 9526 } 9527 setEffectSuspended_l(&effect->desc().type, enabled); 9528 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9529 if (index < 0) { 9530 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9531 return; 9532 } 9533 } 9534 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9535 effect->desc().type.timeLow); 9536 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9537 // if effect is requested to suspended but was not yet enabled, supend it now. 9538 if (desc->mEffect == 0) { 9539 desc->mEffect = effect; 9540 effect->setEnabled(false); 9541 effect->setSuspended(true); 9542 } 9543 } else { 9544 if (index < 0) { 9545 return; 9546 } 9547 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9548 effect->desc().type.timeLow); 9549 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9550 desc->mEffect.clear(); 9551 effect->setSuspended(false); 9552 } 9553} 9554 9555#undef LOG_TAG 9556#define LOG_TAG "AudioFlinger" 9557 9558// ---------------------------------------------------------------------------- 9559 9560status_t AudioFlinger::onTransact( 9561 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9562{ 9563 return BnAudioFlinger::onTransact(code, data, reply, flags); 9564} 9565 9566}; // namespace android 9567