AudioFlinger.cpp revision 28ed2f93324988767b5658eba7c1fa781a275183
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 == NULL) { 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, int channelCount) const 968{ 969 status_t ret = initCheck(); 970 if (ret != NO_ERROR) { 971 return 0; 972 } 973 974 AutoMutex lock(mHardwareLock); 975 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 976 struct audio_config config = { 977 sample_rate: sampleRate, 978 channel_mask: audio_channel_in_mask_from_count(channelCount), 979 format: format, 980 }; 981 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 982 mHardwareStatus = AUDIO_HW_IDLE; 983 return size; 984} 985 986unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 987{ 988 if (ioHandle == 0) { 989 return 0; 990 } 991 992 Mutex::Autolock _l(mLock); 993 994 RecordThread *recordThread = checkRecordThread_l(ioHandle); 995 if (recordThread != NULL) { 996 return recordThread->getInputFramesLost(); 997 } 998 return 0; 999} 1000 1001status_t AudioFlinger::setVoiceVolume(float value) 1002{ 1003 status_t ret = initCheck(); 1004 if (ret != NO_ERROR) { 1005 return ret; 1006 } 1007 1008 // check calling permissions 1009 if (!settingsAllowed()) { 1010 return PERMISSION_DENIED; 1011 } 1012 1013 AutoMutex lock(mHardwareLock); 1014 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1015 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1016 mHardwareStatus = AUDIO_HW_IDLE; 1017 1018 return ret; 1019} 1020 1021status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1022 audio_io_handle_t output) const 1023{ 1024 status_t status; 1025 1026 Mutex::Autolock _l(mLock); 1027 1028 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1029 if (playbackThread != NULL) { 1030 return playbackThread->getRenderPosition(halFrames, dspFrames); 1031 } 1032 1033 return BAD_VALUE; 1034} 1035 1036void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1037{ 1038 1039 Mutex::Autolock _l(mLock); 1040 1041 pid_t pid = IPCThreadState::self()->getCallingPid(); 1042 if (mNotificationClients.indexOfKey(pid) < 0) { 1043 sp<NotificationClient> notificationClient = new NotificationClient(this, 1044 client, 1045 pid); 1046 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1047 1048 mNotificationClients.add(pid, notificationClient); 1049 1050 sp<IBinder> binder = client->asBinder(); 1051 binder->linkToDeath(notificationClient); 1052 1053 // the config change is always sent from playback or record threads to avoid deadlock 1054 // with AudioSystem::gLock 1055 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1056 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1057 } 1058 1059 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1060 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1061 } 1062 } 1063} 1064 1065void AudioFlinger::removeNotificationClient(pid_t pid) 1066{ 1067 Mutex::Autolock _l(mLock); 1068 1069 mNotificationClients.removeItem(pid); 1070 1071 ALOGV("%d died, releasing its sessions", pid); 1072 size_t num = mAudioSessionRefs.size(); 1073 bool removed = false; 1074 for (size_t i = 0; i< num; ) { 1075 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1076 ALOGV(" pid %d @ %d", ref->mPid, i); 1077 if (ref->mPid == pid) { 1078 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1079 mAudioSessionRefs.removeAt(i); 1080 delete ref; 1081 removed = true; 1082 num--; 1083 } else { 1084 i++; 1085 } 1086 } 1087 if (removed) { 1088 purgeStaleEffects_l(); 1089 } 1090} 1091 1092// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1094{ 1095 size_t size = mNotificationClients.size(); 1096 for (size_t i = 0; i < size; i++) { 1097 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1098 param2); 1099 } 1100} 1101 1102// removeClient_l() must be called with AudioFlinger::mLock held 1103void AudioFlinger::removeClient_l(pid_t pid) 1104{ 1105 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1106 mClients.removeItem(pid); 1107} 1108 1109 1110// ---------------------------------------------------------------------------- 1111 1112AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1113 uint32_t device, type_t type) 1114 : Thread(false), 1115 mType(type), 1116 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1117 // mChannelMask 1118 mChannelCount(0), 1119 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1120 mParamStatus(NO_ERROR), 1121 mStandby(false), mId(id), 1122 mDevice(device), 1123 mDeathRecipient(new PMDeathRecipient(this)) 1124{ 1125} 1126 1127AudioFlinger::ThreadBase::~ThreadBase() 1128{ 1129 mParamCond.broadcast(); 1130 // do not lock the mutex in destructor 1131 releaseWakeLock_l(); 1132 if (mPowerManager != 0) { 1133 sp<IBinder> binder = mPowerManager->asBinder(); 1134 binder->unlinkToDeath(mDeathRecipient); 1135 } 1136} 1137 1138void AudioFlinger::ThreadBase::exit() 1139{ 1140 ALOGV("ThreadBase::exit"); 1141 { 1142 // This lock prevents the following race in thread (uniprocessor for illustration): 1143 // if (!exitPending()) { 1144 // // context switch from here to exit() 1145 // // exit() calls requestExit(), what exitPending() observes 1146 // // exit() calls signal(), which is dropped since no waiters 1147 // // context switch back from exit() to here 1148 // mWaitWorkCV.wait(...); 1149 // // now thread is hung 1150 // } 1151 AutoMutex lock(mLock); 1152 requestExit(); 1153 mWaitWorkCV.signal(); 1154 } 1155 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1156 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1157 requestExitAndWait(); 1158} 1159 1160status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1161{ 1162 status_t status; 1163 1164 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1165 Mutex::Autolock _l(mLock); 1166 1167 mNewParameters.add(keyValuePairs); 1168 mWaitWorkCV.signal(); 1169 // wait condition with timeout in case the thread loop has exited 1170 // before the request could be processed 1171 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1172 status = mParamStatus; 1173 mWaitWorkCV.signal(); 1174 } else { 1175 status = TIMED_OUT; 1176 } 1177 return status; 1178} 1179 1180void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1181{ 1182 Mutex::Autolock _l(mLock); 1183 sendConfigEvent_l(event, param); 1184} 1185 1186// sendConfigEvent_l() must be called with ThreadBase::mLock held 1187void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1188{ 1189 ConfigEvent configEvent; 1190 configEvent.mEvent = event; 1191 configEvent.mParam = param; 1192 mConfigEvents.add(configEvent); 1193 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1194 mWaitWorkCV.signal(); 1195} 1196 1197void AudioFlinger::ThreadBase::processConfigEvents() 1198{ 1199 mLock.lock(); 1200 while (!mConfigEvents.isEmpty()) { 1201 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1202 ConfigEvent configEvent = mConfigEvents[0]; 1203 mConfigEvents.removeAt(0); 1204 // release mLock before locking AudioFlinger mLock: lock order is always 1205 // AudioFlinger then ThreadBase to avoid cross deadlock 1206 mLock.unlock(); 1207 mAudioFlinger->mLock.lock(); 1208 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1209 mAudioFlinger->mLock.unlock(); 1210 mLock.lock(); 1211 } 1212 mLock.unlock(); 1213} 1214 1215status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1216{ 1217 const size_t SIZE = 256; 1218 char buffer[SIZE]; 1219 String8 result; 1220 1221 bool locked = tryLock(mLock); 1222 if (!locked) { 1223 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1224 write(fd, buffer, strlen(buffer)); 1225 } 1226 1227 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1246 result.append(buffer); 1247 1248 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1249 result.append(buffer); 1250 result.append(" Index Command"); 1251 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1252 snprintf(buffer, SIZE, "\n %02d ", i); 1253 result.append(buffer); 1254 result.append(mNewParameters[i]); 1255 } 1256 1257 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1258 result.append(buffer); 1259 snprintf(buffer, SIZE, " Index event param\n"); 1260 result.append(buffer); 1261 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1262 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1263 result.append(buffer); 1264 } 1265 result.append("\n"); 1266 1267 write(fd, result.string(), result.size()); 1268 1269 if (locked) { 1270 mLock.unlock(); 1271 } 1272 return NO_ERROR; 1273} 1274 1275status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1276{ 1277 const size_t SIZE = 256; 1278 char buffer[SIZE]; 1279 String8 result; 1280 1281 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1282 write(fd, buffer, strlen(buffer)); 1283 1284 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1285 sp<EffectChain> chain = mEffectChains[i]; 1286 if (chain != 0) { 1287 chain->dump(fd, args); 1288 } 1289 } 1290 return NO_ERROR; 1291} 1292 1293void AudioFlinger::ThreadBase::acquireWakeLock() 1294{ 1295 Mutex::Autolock _l(mLock); 1296 acquireWakeLock_l(); 1297} 1298 1299void AudioFlinger::ThreadBase::acquireWakeLock_l() 1300{ 1301 if (mPowerManager == 0) { 1302 // use checkService() to avoid blocking if power service is not up yet 1303 sp<IBinder> binder = 1304 defaultServiceManager()->checkService(String16("power")); 1305 if (binder == 0) { 1306 ALOGW("Thread %s cannot connect to the power manager service", mName); 1307 } else { 1308 mPowerManager = interface_cast<IPowerManager>(binder); 1309 binder->linkToDeath(mDeathRecipient); 1310 } 1311 } 1312 if (mPowerManager != 0) { 1313 sp<IBinder> binder = new BBinder(); 1314 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1315 binder, 1316 String16(mName)); 1317 if (status == NO_ERROR) { 1318 mWakeLockToken = binder; 1319 } 1320 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1321 } 1322} 1323 1324void AudioFlinger::ThreadBase::releaseWakeLock() 1325{ 1326 Mutex::Autolock _l(mLock); 1327 releaseWakeLock_l(); 1328} 1329 1330void AudioFlinger::ThreadBase::releaseWakeLock_l() 1331{ 1332 if (mWakeLockToken != 0) { 1333 ALOGV("releaseWakeLock_l() %s", mName); 1334 if (mPowerManager != 0) { 1335 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1336 } 1337 mWakeLockToken.clear(); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::clearPowerManager() 1342{ 1343 Mutex::Autolock _l(mLock); 1344 releaseWakeLock_l(); 1345 mPowerManager.clear(); 1346} 1347 1348void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1349{ 1350 sp<ThreadBase> thread = mThread.promote(); 1351 if (thread != 0) { 1352 thread->clearPowerManager(); 1353 } 1354 ALOGW("power manager service died !!!"); 1355} 1356 1357void AudioFlinger::ThreadBase::setEffectSuspended( 1358 const effect_uuid_t *type, bool suspend, int sessionId) 1359{ 1360 Mutex::Autolock _l(mLock); 1361 setEffectSuspended_l(type, suspend, sessionId); 1362} 1363 1364void AudioFlinger::ThreadBase::setEffectSuspended_l( 1365 const effect_uuid_t *type, bool suspend, int sessionId) 1366{ 1367 sp<EffectChain> chain = getEffectChain_l(sessionId); 1368 if (chain != 0) { 1369 if (type != NULL) { 1370 chain->setEffectSuspended_l(type, suspend); 1371 } else { 1372 chain->setEffectSuspendedAll_l(suspend); 1373 } 1374 } 1375 1376 updateSuspendedSessions_l(type, suspend, sessionId); 1377} 1378 1379void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1380{ 1381 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1382 if (index < 0) { 1383 return; 1384 } 1385 1386 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1387 mSuspendedSessions.editValueAt(index); 1388 1389 for (size_t i = 0; i < sessionEffects.size(); i++) { 1390 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1391 for (int j = 0; j < desc->mRefCount; j++) { 1392 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1393 chain->setEffectSuspendedAll_l(true); 1394 } else { 1395 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1396 desc->mType.timeLow); 1397 chain->setEffectSuspended_l(&desc->mType, true); 1398 } 1399 } 1400 } 1401} 1402 1403void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1404 bool suspend, 1405 int sessionId) 1406{ 1407 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1408 1409 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1410 1411 if (suspend) { 1412 if (index >= 0) { 1413 sessionEffects = mSuspendedSessions.editValueAt(index); 1414 } else { 1415 mSuspendedSessions.add(sessionId, sessionEffects); 1416 } 1417 } else { 1418 if (index < 0) { 1419 return; 1420 } 1421 sessionEffects = mSuspendedSessions.editValueAt(index); 1422 } 1423 1424 1425 int key = EffectChain::kKeyForSuspendAll; 1426 if (type != NULL) { 1427 key = type->timeLow; 1428 } 1429 index = sessionEffects.indexOfKey(key); 1430 1431 sp<SuspendedSessionDesc> desc; 1432 if (suspend) { 1433 if (index >= 0) { 1434 desc = sessionEffects.valueAt(index); 1435 } else { 1436 desc = new SuspendedSessionDesc(); 1437 if (type != NULL) { 1438 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1439 } 1440 sessionEffects.add(key, desc); 1441 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1442 } 1443 desc->mRefCount++; 1444 } else { 1445 if (index < 0) { 1446 return; 1447 } 1448 desc = sessionEffects.valueAt(index); 1449 if (--desc->mRefCount == 0) { 1450 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1451 sessionEffects.removeItemsAt(index); 1452 if (sessionEffects.isEmpty()) { 1453 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1454 sessionId); 1455 mSuspendedSessions.removeItem(sessionId); 1456 } 1457 } 1458 } 1459 if (!sessionEffects.isEmpty()) { 1460 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1461 } 1462} 1463 1464void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1465 bool enabled, 1466 int sessionId) 1467{ 1468 Mutex::Autolock _l(mLock); 1469 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1470} 1471 1472void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1473 bool enabled, 1474 int sessionId) 1475{ 1476 if (mType != RECORD) { 1477 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1478 // another session. This gives the priority to well behaved effect control panels 1479 // and applications not using global effects. 1480 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1481 // global effects 1482 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1483 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1484 } 1485 } 1486 1487 sp<EffectChain> chain = getEffectChain_l(sessionId); 1488 if (chain != 0) { 1489 chain->checkSuspendOnEffectEnabled(effect, enabled); 1490 } 1491} 1492 1493// ---------------------------------------------------------------------------- 1494 1495AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1496 AudioStreamOut* output, 1497 audio_io_handle_t id, 1498 uint32_t device, 1499 type_t type) 1500 : ThreadBase(audioFlinger, id, device, type), 1501 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1502 // Assumes constructor is called by AudioFlinger with it's mLock held, 1503 // but it would be safer to explicitly pass initial masterMute as parameter 1504 mMasterMute(audioFlinger->masterMute_l()), 1505 // mStreamTypes[] initialized in constructor body 1506 mOutput(output), 1507 // Assumes constructor is called by AudioFlinger with it's mLock held, 1508 // but it would be safer to explicitly pass initial masterVolume as parameter 1509 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1510 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1511 mMixerStatus(MIXER_IDLE), 1512 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1513 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1514 mScreenState(gScreenState), 1515 // index 0 is reserved for normal mixer's submix 1516 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1517{ 1518 snprintf(mName, kNameLength, "AudioOut_%X", id); 1519 1520 readOutputParameters(); 1521 1522 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1523 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1524 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1525 stream = (audio_stream_type_t) (stream + 1)) { 1526 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1527 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1528 } 1529 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1530 // because mAudioFlinger doesn't have one to copy from 1531} 1532 1533AudioFlinger::PlaybackThread::~PlaybackThread() 1534{ 1535 delete [] mMixBuffer; 1536} 1537 1538status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1539{ 1540 dumpInternals(fd, args); 1541 dumpTracks(fd, args); 1542 dumpEffectChains(fd, args); 1543 return NO_ERROR; 1544} 1545 1546status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1547{ 1548 const size_t SIZE = 256; 1549 char buffer[SIZE]; 1550 String8 result; 1551 1552 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1553 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1554 const stream_type_t *st = &mStreamTypes[i]; 1555 if (i > 0) { 1556 result.appendFormat(", "); 1557 } 1558 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1559 if (st->mute) { 1560 result.append("M"); 1561 } 1562 } 1563 result.append("\n"); 1564 write(fd, result.string(), result.length()); 1565 result.clear(); 1566 1567 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mTracks.size(); ++i) { 1571 sp<Track> track = mTracks[i]; 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 1578 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1579 result.append(buffer); 1580 Track::appendDumpHeader(result); 1581 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1582 sp<Track> track = mActiveTracks[i].promote(); 1583 if (track != 0) { 1584 track->dump(buffer, SIZE); 1585 result.append(buffer); 1586 } 1587 } 1588 write(fd, result.string(), result.size()); 1589 1590 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1591 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1592 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1593 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1594 1595 return NO_ERROR; 1596} 1597 1598status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1599{ 1600 const size_t SIZE = 256; 1601 char buffer[SIZE]; 1602 String8 result; 1603 1604 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1605 result.append(buffer); 1606 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1607 result.append(buffer); 1608 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1609 result.append(buffer); 1610 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1611 result.append(buffer); 1612 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1613 result.append(buffer); 1614 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1615 result.append(buffer); 1616 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1617 result.append(buffer); 1618 write(fd, result.string(), result.size()); 1619 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1620 1621 dumpBase(fd, args); 1622 1623 return NO_ERROR; 1624} 1625 1626// Thread virtuals 1627status_t AudioFlinger::PlaybackThread::readyToRun() 1628{ 1629 status_t status = initCheck(); 1630 if (status == NO_ERROR) { 1631 ALOGI("AudioFlinger's thread %p ready to run", this); 1632 } else { 1633 ALOGE("No working audio driver found."); 1634 } 1635 return status; 1636} 1637 1638void AudioFlinger::PlaybackThread::onFirstRef() 1639{ 1640 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1641} 1642 1643// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1644sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1645 const sp<AudioFlinger::Client>& client, 1646 audio_stream_type_t streamType, 1647 uint32_t sampleRate, 1648 audio_format_t format, 1649 uint32_t channelMask, 1650 int frameCount, 1651 const sp<IMemory>& sharedBuffer, 1652 int sessionId, 1653 IAudioFlinger::track_flags_t flags, 1654 pid_t tid, 1655 status_t *status) 1656{ 1657 sp<Track> track; 1658 status_t lStatus; 1659 1660 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1661 1662 // client expresses a preference for FAST, but we get the final say 1663 if (flags & IAudioFlinger::TRACK_FAST) { 1664 if ( 1665 // not timed 1666 (!isTimed) && 1667 // either of these use cases: 1668 ( 1669 // use case 1: shared buffer with any frame count 1670 ( 1671 (sharedBuffer != 0) 1672 ) || 1673 // use case 2: callback handler and frame count is default or at least as large as HAL 1674 ( 1675 (tid != -1) && 1676 ((frameCount == 0) || 1677 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1678 ) 1679 ) && 1680 // PCM data 1681 audio_is_linear_pcm(format) && 1682 // mono or stereo 1683 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1684 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1685#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1686 // hardware sample rate 1687 (sampleRate == mSampleRate) && 1688#endif 1689 // normal mixer has an associated fast mixer 1690 hasFastMixer() && 1691 // there are sufficient fast track slots available 1692 (mFastTrackAvailMask != 0) 1693 // FIXME test that MixerThread for this fast track has a capable output HAL 1694 // FIXME add a permission test also? 1695 ) { 1696 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1697 if (frameCount == 0) { 1698 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1699 } 1700 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1701 frameCount, mFrameCount); 1702 } else { 1703 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1704 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1705 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1706 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1707 audio_is_linear_pcm(format), 1708 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1709 flags &= ~IAudioFlinger::TRACK_FAST; 1710 // For compatibility with AudioTrack calculation, buffer depth is forced 1711 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1712 // This is probably too conservative, but legacy application code may depend on it. 1713 // If you change this calculation, also review the start threshold which is related. 1714 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1715 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1716 if (minBufCount < 2) { 1717 minBufCount = 2; 1718 } 1719 int minFrameCount = mNormalFrameCount * minBufCount; 1720 if (frameCount < minFrameCount) { 1721 frameCount = minFrameCount; 1722 } 1723 } 1724 } 1725 1726 if (mType == DIRECT) { 1727 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1728 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1729 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1730 "for output %p with format %d", 1731 sampleRate, format, channelMask, mOutput, mFormat); 1732 lStatus = BAD_VALUE; 1733 goto Exit; 1734 } 1735 } 1736 } else { 1737 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1738 if (sampleRate > mSampleRate*2) { 1739 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1740 lStatus = BAD_VALUE; 1741 goto Exit; 1742 } 1743 } 1744 1745 lStatus = initCheck(); 1746 if (lStatus != NO_ERROR) { 1747 ALOGE("Audio driver not initialized."); 1748 goto Exit; 1749 } 1750 1751 { // scope for mLock 1752 Mutex::Autolock _l(mLock); 1753 1754 // all tracks in same audio session must share the same routing strategy otherwise 1755 // conflicts will happen when tracks are moved from one output to another by audio policy 1756 // manager 1757 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1758 for (size_t i = 0; i < mTracks.size(); ++i) { 1759 sp<Track> t = mTracks[i]; 1760 if (t != 0 && !t->isOutputTrack()) { 1761 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1762 if (sessionId == t->sessionId() && strategy != actual) { 1763 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1764 strategy, actual); 1765 lStatus = BAD_VALUE; 1766 goto Exit; 1767 } 1768 } 1769 } 1770 1771 if (!isTimed) { 1772 track = new Track(this, client, streamType, sampleRate, format, 1773 channelMask, frameCount, sharedBuffer, sessionId, flags); 1774 } else { 1775 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1776 channelMask, frameCount, sharedBuffer, sessionId); 1777 } 1778 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1779 lStatus = NO_MEMORY; 1780 goto Exit; 1781 } 1782 mTracks.add(track); 1783 1784 sp<EffectChain> chain = getEffectChain_l(sessionId); 1785 if (chain != 0) { 1786 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1787 track->setMainBuffer(chain->inBuffer()); 1788 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1789 chain->incTrackCnt(); 1790 } 1791 } 1792 1793#ifdef HAVE_REQUEST_PRIORITY 1794 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1795 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1796 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1797 // so ask activity manager to do this on our behalf 1798 int err = requestPriority(callingPid, tid, 1); 1799 if (err != 0) { 1800 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1801 1, callingPid, tid, err); 1802 } 1803 } 1804#endif 1805 1806 lStatus = NO_ERROR; 1807 1808Exit: 1809 if (status) { 1810 *status = lStatus; 1811 } 1812 return track; 1813} 1814 1815uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1816{ 1817 if (mFastMixer != NULL) { 1818 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1819 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1820 } 1821 return latency; 1822} 1823 1824uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1825{ 1826 return latency; 1827} 1828 1829uint32_t AudioFlinger::PlaybackThread::latency() const 1830{ 1831 Mutex::Autolock _l(mLock); 1832 return latency_l(); 1833} 1834uint32_t AudioFlinger::PlaybackThread::latency_l() const 1835{ 1836 if (initCheck() == NO_ERROR) { 1837 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1838 } else { 1839 return 0; 1840 } 1841} 1842 1843void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 mMasterVolume = value; 1847} 1848 1849void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 setMasterMute_l(muted); 1853} 1854 1855void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1856{ 1857 Mutex::Autolock _l(mLock); 1858 mStreamTypes[stream].volume = value; 1859} 1860 1861void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1862{ 1863 Mutex::Autolock _l(mLock); 1864 mStreamTypes[stream].mute = muted; 1865} 1866 1867float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1868{ 1869 Mutex::Autolock _l(mLock); 1870 return mStreamTypes[stream].volume; 1871} 1872 1873// addTrack_l() must be called with ThreadBase::mLock held 1874status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1875{ 1876 status_t status = ALREADY_EXISTS; 1877 1878 // set retry count for buffer fill 1879 track->mRetryCount = kMaxTrackStartupRetries; 1880 if (mActiveTracks.indexOf(track) < 0) { 1881 // the track is newly added, make sure it fills up all its 1882 // buffers before playing. This is to ensure the client will 1883 // effectively get the latency it requested. 1884 track->mFillingUpStatus = Track::FS_FILLING; 1885 track->mResetDone = false; 1886 track->mPresentationCompleteFrames = 0; 1887 mActiveTracks.add(track); 1888 if (track->mainBuffer() != mMixBuffer) { 1889 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1890 if (chain != 0) { 1891 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1892 chain->incActiveTrackCnt(); 1893 } 1894 } 1895 1896 status = NO_ERROR; 1897 } 1898 1899 ALOGV("mWaitWorkCV.broadcast"); 1900 mWaitWorkCV.broadcast(); 1901 1902 return status; 1903} 1904 1905// destroyTrack_l() must be called with ThreadBase::mLock held 1906void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1907{ 1908 track->mState = TrackBase::TERMINATED; 1909 // active tracks are removed by threadLoop() 1910 if (mActiveTracks.indexOf(track) < 0) { 1911 removeTrack_l(track); 1912 } 1913} 1914 1915void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1916{ 1917 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1918 mTracks.remove(track); 1919 deleteTrackName_l(track->name()); 1920 // redundant as track is about to be destroyed, for dumpsys only 1921 track->mName = -1; 1922 if (track->isFastTrack()) { 1923 int index = track->mFastIndex; 1924 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1925 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1926 mFastTrackAvailMask |= 1 << index; 1927 // redundant as track is about to be destroyed, for dumpsys only 1928 track->mFastIndex = -1; 1929 } 1930 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1931 if (chain != 0) { 1932 chain->decTrackCnt(); 1933 } 1934} 1935 1936String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1937{ 1938 String8 out_s8 = String8(""); 1939 char *s; 1940 1941 Mutex::Autolock _l(mLock); 1942 if (initCheck() != NO_ERROR) { 1943 return out_s8; 1944 } 1945 1946 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1947 out_s8 = String8(s); 1948 free(s); 1949 return out_s8; 1950} 1951 1952// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1953void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1954 AudioSystem::OutputDescriptor desc; 1955 void *param2 = NULL; 1956 1957 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1958 1959 switch (event) { 1960 case AudioSystem::OUTPUT_OPENED: 1961 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1962 desc.channels = mChannelMask; 1963 desc.samplingRate = mSampleRate; 1964 desc.format = mFormat; 1965 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1966 desc.latency = latency(); 1967 param2 = &desc; 1968 break; 1969 1970 case AudioSystem::STREAM_CONFIG_CHANGED: 1971 param2 = ¶m; 1972 case AudioSystem::OUTPUT_CLOSED: 1973 default: 1974 break; 1975 } 1976 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1977} 1978 1979void AudioFlinger::PlaybackThread::readOutputParameters() 1980{ 1981 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1982 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1983 mChannelCount = (uint16_t)popcount(mChannelMask); 1984 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1985 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1986 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1987 if (mFrameCount & 15) { 1988 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1989 mFrameCount); 1990 } 1991 1992 // Calculate size of normal mix buffer relative to the HAL output buffer size 1993 double multiplier = 1.0; 1994 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1995 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1996 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1997 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1998 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1999 maxNormalFrameCount = maxNormalFrameCount & ~15; 2000 if (maxNormalFrameCount < minNormalFrameCount) { 2001 maxNormalFrameCount = minNormalFrameCount; 2002 } 2003 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2004 if (multiplier <= 1.0) { 2005 multiplier = 1.0; 2006 } else if (multiplier <= 2.0) { 2007 if (2 * mFrameCount <= maxNormalFrameCount) { 2008 multiplier = 2.0; 2009 } else { 2010 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2011 } 2012 } else { 2013 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2014 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2015 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2016 // FIXME this rounding up should not be done if no HAL SRC 2017 uint32_t truncMult = (uint32_t) multiplier; 2018 if ((truncMult & 1)) { 2019 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2020 ++truncMult; 2021 } 2022 } 2023 multiplier = (double) truncMult; 2024 } 2025 } 2026 mNormalFrameCount = multiplier * mFrameCount; 2027 // round up to nearest 16 frames to satisfy AudioMixer 2028 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2029 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2030 2031 delete[] mMixBuffer; 2032 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2033 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2034 2035 // force reconfiguration of effect chains and engines to take new buffer size and audio 2036 // parameters into account 2037 // Note that mLock is not held when readOutputParameters() is called from the constructor 2038 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2039 // matter. 2040 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2041 Vector< sp<EffectChain> > effectChains = mEffectChains; 2042 for (size_t i = 0; i < effectChains.size(); i ++) { 2043 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2044 } 2045} 2046 2047 2048status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2049{ 2050 if (halFrames == NULL || dspFrames == NULL) { 2051 return BAD_VALUE; 2052 } 2053 Mutex::Autolock _l(mLock); 2054 if (initCheck() != NO_ERROR) { 2055 return INVALID_OPERATION; 2056 } 2057 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2058 2059 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2060} 2061 2062uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2063{ 2064 Mutex::Autolock _l(mLock); 2065 uint32_t result = 0; 2066 if (getEffectChain_l(sessionId) != 0) { 2067 result = EFFECT_SESSION; 2068 } 2069 2070 for (size_t i = 0; i < mTracks.size(); ++i) { 2071 sp<Track> track = mTracks[i]; 2072 if (sessionId == track->sessionId() && 2073 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2074 result |= TRACK_SESSION; 2075 break; 2076 } 2077 } 2078 2079 return result; 2080} 2081 2082uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2083{ 2084 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2085 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2086 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2087 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2088 } 2089 for (size_t i = 0; i < mTracks.size(); i++) { 2090 sp<Track> track = mTracks[i]; 2091 if (sessionId == track->sessionId() && 2092 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2093 return AudioSystem::getStrategyForStream(track->streamType()); 2094 } 2095 } 2096 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2097} 2098 2099 2100AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2101{ 2102 Mutex::Autolock _l(mLock); 2103 return mOutput; 2104} 2105 2106AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2107{ 2108 Mutex::Autolock _l(mLock); 2109 AudioStreamOut *output = mOutput; 2110 mOutput = NULL; 2111 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2112 // must push a NULL and wait for ack 2113 mOutputSink.clear(); 2114 mPipeSink.clear(); 2115 mNormalSink.clear(); 2116 return output; 2117} 2118 2119// this method must always be called either with ThreadBase mLock held or inside the thread loop 2120audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2121{ 2122 if (mOutput == NULL) { 2123 return NULL; 2124 } 2125 return &mOutput->stream->common; 2126} 2127 2128uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2129{ 2130 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2131} 2132 2133status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2134{ 2135 if (!isValidSyncEvent(event)) { 2136 return BAD_VALUE; 2137 } 2138 2139 Mutex::Autolock _l(mLock); 2140 2141 for (size_t i = 0; i < mTracks.size(); ++i) { 2142 sp<Track> track = mTracks[i]; 2143 if (event->triggerSession() == track->sessionId()) { 2144 track->setSyncEvent(event); 2145 return NO_ERROR; 2146 } 2147 } 2148 2149 return NAME_NOT_FOUND; 2150} 2151 2152bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2153{ 2154 switch (event->type()) { 2155 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2156 return true; 2157 default: 2158 break; 2159 } 2160 return false; 2161} 2162 2163void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2164{ 2165 size_t count = tracksToRemove.size(); 2166 if (CC_UNLIKELY(count)) { 2167 for (size_t i = 0 ; i < count ; i++) { 2168 const sp<Track>& track = tracksToRemove.itemAt(i); 2169 if ((track->sharedBuffer() != 0) && 2170 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2171 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2172 } 2173 } 2174 } 2175 2176} 2177 2178// ---------------------------------------------------------------------------- 2179 2180AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2181 audio_io_handle_t id, uint32_t device, type_t type) 2182 : PlaybackThread(audioFlinger, output, id, device, type), 2183 // mAudioMixer below 2184#ifdef SOAKER 2185 mSoaker(NULL), 2186#endif 2187 // mFastMixer below 2188 mFastMixerFutex(0) 2189 // mOutputSink below 2190 // mPipeSink below 2191 // mNormalSink below 2192{ 2193 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2194 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2195 "mFrameCount=%d, mNormalFrameCount=%d", 2196 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2197 mNormalFrameCount); 2198 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2199 2200 // FIXME - Current mixer implementation only supports stereo output 2201 if (mChannelCount == 1) { 2202 ALOGE("Invalid audio hardware channel count"); 2203 } 2204 2205 // create an NBAIO sink for the HAL output stream, and negotiate 2206 mOutputSink = new AudioStreamOutSink(output->stream); 2207 size_t numCounterOffers = 0; 2208 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2209 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2210 ALOG_ASSERT(index == 0); 2211 2212 // initialize fast mixer depending on configuration 2213 bool initFastMixer; 2214 switch (kUseFastMixer) { 2215 case FastMixer_Never: 2216 initFastMixer = false; 2217 break; 2218 case FastMixer_Always: 2219 initFastMixer = true; 2220 break; 2221 case FastMixer_Static: 2222 case FastMixer_Dynamic: 2223 initFastMixer = mFrameCount < mNormalFrameCount; 2224 break; 2225 } 2226 if (initFastMixer) { 2227 2228 // create a MonoPipe to connect our submix to FastMixer 2229 NBAIO_Format format = mOutputSink->format(); 2230 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2231 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2232 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2233 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2234 const NBAIO_Format offers[1] = {format}; 2235 size_t numCounterOffers = 0; 2236 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2237 ALOG_ASSERT(index == 0); 2238 monoPipe->setAvgFrames((mScreenState & 1) ? 2239 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2240 mPipeSink = monoPipe; 2241 2242#ifdef TEE_SINK_FRAMES 2243 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2244 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2245 numCounterOffers = 0; 2246 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2247 ALOG_ASSERT(index == 0); 2248 mTeeSink = teeSink; 2249 PipeReader *teeSource = new PipeReader(*teeSink); 2250 numCounterOffers = 0; 2251 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2252 ALOG_ASSERT(index == 0); 2253 mTeeSource = teeSource; 2254#endif 2255 2256#ifdef SOAKER 2257 // create a soaker as workaround for governor issues 2258 mSoaker = new Soaker(); 2259 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2260 mSoaker->run("Soaker", PRIORITY_LOWEST); 2261#endif 2262 2263 // create fast mixer and configure it initially with just one fast track for our submix 2264 mFastMixer = new FastMixer(); 2265 FastMixerStateQueue *sq = mFastMixer->sq(); 2266#ifdef STATE_QUEUE_DUMP 2267 sq->setObserverDump(&mStateQueueObserverDump); 2268 sq->setMutatorDump(&mStateQueueMutatorDump); 2269#endif 2270 FastMixerState *state = sq->begin(); 2271 FastTrack *fastTrack = &state->mFastTracks[0]; 2272 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2273 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2274 fastTrack->mVolumeProvider = NULL; 2275 fastTrack->mGeneration++; 2276 state->mFastTracksGen++; 2277 state->mTrackMask = 1; 2278 // fast mixer will use the HAL output sink 2279 state->mOutputSink = mOutputSink.get(); 2280 state->mOutputSinkGen++; 2281 state->mFrameCount = mFrameCount; 2282 state->mCommand = FastMixerState::COLD_IDLE; 2283 // already done in constructor initialization list 2284 //mFastMixerFutex = 0; 2285 state->mColdFutexAddr = &mFastMixerFutex; 2286 state->mColdGen++; 2287 state->mDumpState = &mFastMixerDumpState; 2288 state->mTeeSink = mTeeSink.get(); 2289 sq->end(); 2290 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2291 2292 // start the fast mixer 2293 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2294#ifdef HAVE_REQUEST_PRIORITY 2295 pid_t tid = mFastMixer->getTid(); 2296 int err = requestPriority(getpid_cached, tid, 2); 2297 if (err != 0) { 2298 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2299 2, getpid_cached, tid, err); 2300 } 2301#endif 2302 2303 } else { 2304 mFastMixer = NULL; 2305 } 2306 2307 switch (kUseFastMixer) { 2308 case FastMixer_Never: 2309 case FastMixer_Dynamic: 2310 mNormalSink = mOutputSink; 2311 break; 2312 case FastMixer_Always: 2313 mNormalSink = mPipeSink; 2314 break; 2315 case FastMixer_Static: 2316 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2317 break; 2318 } 2319} 2320 2321AudioFlinger::MixerThread::~MixerThread() 2322{ 2323 if (mFastMixer != NULL) { 2324 FastMixerStateQueue *sq = mFastMixer->sq(); 2325 FastMixerState *state = sq->begin(); 2326 if (state->mCommand == FastMixerState::COLD_IDLE) { 2327 int32_t old = android_atomic_inc(&mFastMixerFutex); 2328 if (old == -1) { 2329 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2330 } 2331 } 2332 state->mCommand = FastMixerState::EXIT; 2333 sq->end(); 2334 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2335 mFastMixer->join(); 2336 // Though the fast mixer thread has exited, it's state queue is still valid. 2337 // We'll use that extract the final state which contains one remaining fast track 2338 // corresponding to our sub-mix. 2339 state = sq->begin(); 2340 ALOG_ASSERT(state->mTrackMask == 1); 2341 FastTrack *fastTrack = &state->mFastTracks[0]; 2342 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2343 delete fastTrack->mBufferProvider; 2344 sq->end(false /*didModify*/); 2345 delete mFastMixer; 2346#ifdef SOAKER 2347 if (mSoaker != NULL) { 2348 mSoaker->requestExitAndWait(); 2349 } 2350 delete mSoaker; 2351#endif 2352 } 2353 delete mAudioMixer; 2354} 2355 2356class CpuStats { 2357public: 2358 CpuStats(); 2359 void sample(const String8 &title); 2360#ifdef DEBUG_CPU_USAGE 2361private: 2362 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2363 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2364 2365 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2366 2367 int mCpuNum; // thread's current CPU number 2368 int mCpukHz; // frequency of thread's current CPU in kHz 2369#endif 2370}; 2371 2372CpuStats::CpuStats() 2373#ifdef DEBUG_CPU_USAGE 2374 : mCpuNum(-1), mCpukHz(-1) 2375#endif 2376{ 2377} 2378 2379void CpuStats::sample(const String8 &title) { 2380#ifdef DEBUG_CPU_USAGE 2381 // get current thread's delta CPU time in wall clock ns 2382 double wcNs; 2383 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2384 2385 // record sample for wall clock statistics 2386 if (valid) { 2387 mWcStats.sample(wcNs); 2388 } 2389 2390 // get the current CPU number 2391 int cpuNum = sched_getcpu(); 2392 2393 // get the current CPU frequency in kHz 2394 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2395 2396 // check if either CPU number or frequency changed 2397 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2398 mCpuNum = cpuNum; 2399 mCpukHz = cpukHz; 2400 // ignore sample for purposes of cycles 2401 valid = false; 2402 } 2403 2404 // if no change in CPU number or frequency, then record sample for cycle statistics 2405 if (valid && mCpukHz > 0) { 2406 double cycles = wcNs * cpukHz * 0.000001; 2407 mHzStats.sample(cycles); 2408 } 2409 2410 unsigned n = mWcStats.n(); 2411 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2412 if ((n & 127) == 1) { 2413 long long elapsed = mCpuUsage.elapsed(); 2414 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2415 double perLoop = elapsed / (double) n; 2416 double perLoop100 = perLoop * 0.01; 2417 double perLoop1k = perLoop * 0.001; 2418 double mean = mWcStats.mean(); 2419 double stddev = mWcStats.stddev(); 2420 double minimum = mWcStats.minimum(); 2421 double maximum = mWcStats.maximum(); 2422 double meanCycles = mHzStats.mean(); 2423 double stddevCycles = mHzStats.stddev(); 2424 double minCycles = mHzStats.minimum(); 2425 double maxCycles = mHzStats.maximum(); 2426 mCpuUsage.resetElapsed(); 2427 mWcStats.reset(); 2428 mHzStats.reset(); 2429 ALOGD("CPU usage for %s over past %.1f secs\n" 2430 " (%u mixer loops at %.1f mean ms per loop):\n" 2431 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2432 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2433 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2434 title.string(), 2435 elapsed * .000000001, n, perLoop * .000001, 2436 mean * .001, 2437 stddev * .001, 2438 minimum * .001, 2439 maximum * .001, 2440 mean / perLoop100, 2441 stddev / perLoop100, 2442 minimum / perLoop100, 2443 maximum / perLoop100, 2444 meanCycles / perLoop1k, 2445 stddevCycles / perLoop1k, 2446 minCycles / perLoop1k, 2447 maxCycles / perLoop1k); 2448 2449 } 2450 } 2451#endif 2452}; 2453 2454void AudioFlinger::PlaybackThread::checkSilentMode_l() 2455{ 2456 if (!mMasterMute) { 2457 char value[PROPERTY_VALUE_MAX]; 2458 if (property_get("ro.audio.silent", value, "0") > 0) { 2459 char *endptr; 2460 unsigned long ul = strtoul(value, &endptr, 0); 2461 if (*endptr == '\0' && ul != 0) { 2462 ALOGD("Silence is golden"); 2463 // The setprop command will not allow a property to be changed after 2464 // the first time it is set, so we don't have to worry about un-muting. 2465 setMasterMute_l(true); 2466 } 2467 } 2468 } 2469} 2470 2471bool AudioFlinger::PlaybackThread::threadLoop() 2472{ 2473 Vector< sp<Track> > tracksToRemove; 2474 2475 standbyTime = systemTime(); 2476 2477 // MIXER 2478 nsecs_t lastWarning = 0; 2479if (mType == MIXER) { 2480 longStandbyExit = false; 2481} 2482 2483 // DUPLICATING 2484 // FIXME could this be made local to while loop? 2485 writeFrames = 0; 2486 2487 cacheParameters_l(); 2488 sleepTime = idleSleepTime; 2489 2490if (mType == MIXER) { 2491 sleepTimeShift = 0; 2492} 2493 2494 CpuStats cpuStats; 2495 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2496 2497 acquireWakeLock(); 2498 2499 while (!exitPending()) 2500 { 2501 cpuStats.sample(myName); 2502 2503 Vector< sp<EffectChain> > effectChains; 2504 2505 processConfigEvents(); 2506 2507 { // scope for mLock 2508 2509 Mutex::Autolock _l(mLock); 2510 2511 if (checkForNewParameters_l()) { 2512 cacheParameters_l(); 2513 } 2514 2515 saveOutputTracks(); 2516 2517 // put audio hardware into standby after short delay 2518 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2519 mSuspended > 0)) { 2520 if (!mStandby) { 2521 2522 threadLoop_standby(); 2523 2524 mStandby = true; 2525 mBytesWritten = 0; 2526 } 2527 2528 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2529 // we're about to wait, flush the binder command buffer 2530 IPCThreadState::self()->flushCommands(); 2531 2532 clearOutputTracks(); 2533 2534 if (exitPending()) break; 2535 2536 releaseWakeLock_l(); 2537 // wait until we have something to do... 2538 ALOGV("%s going to sleep", myName.string()); 2539 mWaitWorkCV.wait(mLock); 2540 ALOGV("%s waking up", myName.string()); 2541 acquireWakeLock_l(); 2542 2543 mMixerStatus = MIXER_IDLE; 2544 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2545 2546 checkSilentMode_l(); 2547 2548 standbyTime = systemTime() + standbyDelay; 2549 sleepTime = idleSleepTime; 2550 if (mType == MIXER) { 2551 sleepTimeShift = 0; 2552 } 2553 2554 continue; 2555 } 2556 } 2557 2558 // mMixerStatusIgnoringFastTracks is also updated internally 2559 mMixerStatus = prepareTracks_l(&tracksToRemove); 2560 2561 // prevent any changes in effect chain list and in each effect chain 2562 // during mixing and effect process as the audio buffers could be deleted 2563 // or modified if an effect is created or deleted 2564 lockEffectChains_l(effectChains); 2565 } 2566 2567 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2568 threadLoop_mix(); 2569 } else { 2570 threadLoop_sleepTime(); 2571 } 2572 2573 if (mSuspended > 0) { 2574 sleepTime = suspendSleepTimeUs(); 2575 } 2576 2577 // only process effects if we're going to write 2578 if (sleepTime == 0) { 2579 for (size_t i = 0; i < effectChains.size(); i ++) { 2580 effectChains[i]->process_l(); 2581 } 2582 } 2583 2584 // enable changes in effect chain 2585 unlockEffectChains(effectChains); 2586 2587 // sleepTime == 0 means we must write to audio hardware 2588 if (sleepTime == 0) { 2589 2590 threadLoop_write(); 2591 2592if (mType == MIXER) { 2593 // write blocked detection 2594 nsecs_t now = systemTime(); 2595 nsecs_t delta = now - mLastWriteTime; 2596 if (!mStandby && delta > maxPeriod) { 2597 mNumDelayedWrites++; 2598 if ((now - lastWarning) > kWarningThrottleNs) { 2599#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2600 ScopedTrace st(ATRACE_TAG, "underrun"); 2601#endif 2602 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2603 ns2ms(delta), mNumDelayedWrites, this); 2604 lastWarning = now; 2605 } 2606 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2607 // a different threshold. Or completely removed for what it is worth anyway... 2608 if (mStandby) { 2609 longStandbyExit = true; 2610 } 2611 } 2612} 2613 2614 mStandby = false; 2615 } else { 2616 usleep(sleepTime); 2617 } 2618 2619 // Finally let go of removed track(s), without the lock held 2620 // since we can't guarantee the destructors won't acquire that 2621 // same lock. This will also mutate and push a new fast mixer state. 2622 threadLoop_removeTracks(tracksToRemove); 2623 tracksToRemove.clear(); 2624 2625 // FIXME I don't understand the need for this here; 2626 // it was in the original code but maybe the 2627 // assignment in saveOutputTracks() makes this unnecessary? 2628 clearOutputTracks(); 2629 2630 // Effect chains will be actually deleted here if they were removed from 2631 // mEffectChains list during mixing or effects processing 2632 effectChains.clear(); 2633 2634 // FIXME Note that the above .clear() is no longer necessary since effectChains 2635 // is now local to this block, but will keep it for now (at least until merge done). 2636 } 2637 2638if (mType == MIXER || mType == DIRECT) { 2639 // put output stream into standby mode 2640 if (!mStandby) { 2641 mOutput->stream->common.standby(&mOutput->stream->common); 2642 } 2643} 2644if (mType == DUPLICATING) { 2645 // for DuplicatingThread, standby mode is handled by the outputTracks 2646} 2647 2648 releaseWakeLock(); 2649 2650 ALOGV("Thread %p type %d exiting", this, mType); 2651 return false; 2652} 2653 2654void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2655{ 2656 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2657} 2658 2659void AudioFlinger::MixerThread::threadLoop_write() 2660{ 2661 // FIXME we should only do one push per cycle; confirm this is true 2662 // Start the fast mixer if it's not already running 2663 if (mFastMixer != NULL) { 2664 FastMixerStateQueue *sq = mFastMixer->sq(); 2665 FastMixerState *state = sq->begin(); 2666 if (state->mCommand != FastMixerState::MIX_WRITE && 2667 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2668 if (state->mCommand == FastMixerState::COLD_IDLE) { 2669 int32_t old = android_atomic_inc(&mFastMixerFutex); 2670 if (old == -1) { 2671 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2672 } 2673 } 2674 state->mCommand = FastMixerState::MIX_WRITE; 2675 sq->end(); 2676 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2677 if (kUseFastMixer == FastMixer_Dynamic) { 2678 mNormalSink = mPipeSink; 2679 } 2680 } else { 2681 sq->end(false /*didModify*/); 2682 } 2683 } 2684 PlaybackThread::threadLoop_write(); 2685} 2686 2687// shared by MIXER and DIRECT, overridden by DUPLICATING 2688void AudioFlinger::PlaybackThread::threadLoop_write() 2689{ 2690 // FIXME rewrite to reduce number of system calls 2691 mLastWriteTime = systemTime(); 2692 mInWrite = true; 2693 int bytesWritten; 2694 2695 // If an NBAIO sink is present, use it to write the normal mixer's submix 2696 if (mNormalSink != 0) { 2697#define mBitShift 2 // FIXME 2698 size_t count = mixBufferSize >> mBitShift; 2699#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2700 Tracer::traceBegin(ATRACE_TAG, "write"); 2701#endif 2702 // update the setpoint when gScreenState changes 2703 uint32_t screenState = gScreenState; 2704 if (screenState != mScreenState) { 2705 mScreenState = screenState; 2706 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2707 if (pipe != NULL) { 2708 pipe->setAvgFrames((mScreenState & 1) ? 2709 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2710 } 2711 } 2712 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2713#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2714 Tracer::traceEnd(ATRACE_TAG); 2715#endif 2716 if (framesWritten > 0) { 2717 bytesWritten = framesWritten << mBitShift; 2718 } else { 2719 bytesWritten = framesWritten; 2720 } 2721 // otherwise use the HAL / AudioStreamOut directly 2722 } else { 2723 // Direct output thread. 2724 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2725 } 2726 2727 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2728 mNumWrites++; 2729 mInWrite = false; 2730} 2731 2732void AudioFlinger::MixerThread::threadLoop_standby() 2733{ 2734 // Idle the fast mixer if it's currently running 2735 if (mFastMixer != NULL) { 2736 FastMixerStateQueue *sq = mFastMixer->sq(); 2737 FastMixerState *state = sq->begin(); 2738 if (!(state->mCommand & FastMixerState::IDLE)) { 2739 state->mCommand = FastMixerState::COLD_IDLE; 2740 state->mColdFutexAddr = &mFastMixerFutex; 2741 state->mColdGen++; 2742 mFastMixerFutex = 0; 2743 sq->end(); 2744 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2745 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2746 if (kUseFastMixer == FastMixer_Dynamic) { 2747 mNormalSink = mOutputSink; 2748 } 2749 } else { 2750 sq->end(false /*didModify*/); 2751 } 2752 } 2753 PlaybackThread::threadLoop_standby(); 2754} 2755 2756// shared by MIXER and DIRECT, overridden by DUPLICATING 2757void AudioFlinger::PlaybackThread::threadLoop_standby() 2758{ 2759 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2760 mOutput->stream->common.standby(&mOutput->stream->common); 2761} 2762 2763void AudioFlinger::MixerThread::threadLoop_mix() 2764{ 2765 // obtain the presentation timestamp of the next output buffer 2766 int64_t pts; 2767 status_t status = INVALID_OPERATION; 2768 2769 if (NULL != mOutput->stream->get_next_write_timestamp) { 2770 status = mOutput->stream->get_next_write_timestamp( 2771 mOutput->stream, &pts); 2772 } 2773 2774 if (status != NO_ERROR) { 2775 pts = AudioBufferProvider::kInvalidPTS; 2776 } 2777 2778 // mix buffers... 2779 mAudioMixer->process(pts); 2780 // increase sleep time progressively when application underrun condition clears. 2781 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2782 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2783 // such that we would underrun the audio HAL. 2784 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2785 sleepTimeShift--; 2786 } 2787 sleepTime = 0; 2788 standbyTime = systemTime() + standbyDelay; 2789 //TODO: delay standby when effects have a tail 2790} 2791 2792void AudioFlinger::MixerThread::threadLoop_sleepTime() 2793{ 2794 // If no tracks are ready, sleep once for the duration of an output 2795 // buffer size, then write 0s to the output 2796 if (sleepTime == 0) { 2797 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2798 sleepTime = activeSleepTime >> sleepTimeShift; 2799 if (sleepTime < kMinThreadSleepTimeUs) { 2800 sleepTime = kMinThreadSleepTimeUs; 2801 } 2802 // reduce sleep time in case of consecutive application underruns to avoid 2803 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2804 // duration we would end up writing less data than needed by the audio HAL if 2805 // the condition persists. 2806 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2807 sleepTimeShift++; 2808 } 2809 } else { 2810 sleepTime = idleSleepTime; 2811 } 2812 } else if (mBytesWritten != 0 || 2813 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2814 memset (mMixBuffer, 0, mixBufferSize); 2815 sleepTime = 0; 2816 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2817 } 2818 // TODO add standby time extension fct of effect tail 2819} 2820 2821// prepareTracks_l() must be called with ThreadBase::mLock held 2822AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2823 Vector< sp<Track> > *tracksToRemove) 2824{ 2825 2826 mixer_state mixerStatus = MIXER_IDLE; 2827 // find out which tracks need to be processed 2828 size_t count = mActiveTracks.size(); 2829 size_t mixedTracks = 0; 2830 size_t tracksWithEffect = 0; 2831 // counts only _active_ fast tracks 2832 size_t fastTracks = 0; 2833 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2834 2835 float masterVolume = mMasterVolume; 2836 bool masterMute = mMasterMute; 2837 2838 if (masterMute) { 2839 masterVolume = 0; 2840 } 2841 // Delegate master volume control to effect in output mix effect chain if needed 2842 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2843 if (chain != 0) { 2844 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2845 chain->setVolume_l(&v, &v); 2846 masterVolume = (float)((v + (1 << 23)) >> 24); 2847 chain.clear(); 2848 } 2849 2850 // prepare a new state to push 2851 FastMixerStateQueue *sq = NULL; 2852 FastMixerState *state = NULL; 2853 bool didModify = false; 2854 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2855 if (mFastMixer != NULL) { 2856 sq = mFastMixer->sq(); 2857 state = sq->begin(); 2858 } 2859 2860 for (size_t i=0 ; i<count ; i++) { 2861 sp<Track> t = mActiveTracks[i].promote(); 2862 if (t == 0) continue; 2863 2864 // this const just means the local variable doesn't change 2865 Track* const track = t.get(); 2866 2867 // process fast tracks 2868 if (track->isFastTrack()) { 2869 2870 // It's theoretically possible (though unlikely) for a fast track to be created 2871 // and then removed within the same normal mix cycle. This is not a problem, as 2872 // the track never becomes active so it's fast mixer slot is never touched. 2873 // The converse, of removing an (active) track and then creating a new track 2874 // at the identical fast mixer slot within the same normal mix cycle, 2875 // is impossible because the slot isn't marked available until the end of each cycle. 2876 int j = track->mFastIndex; 2877 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2878 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2879 FastTrack *fastTrack = &state->mFastTracks[j]; 2880 2881 // Determine whether the track is currently in underrun condition, 2882 // and whether it had a recent underrun. 2883 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2884 FastTrackUnderruns underruns = ftDump->mUnderruns; 2885 uint32_t recentFull = (underruns.mBitFields.mFull - 2886 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2887 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2888 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2889 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2890 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2891 uint32_t recentUnderruns = recentPartial + recentEmpty; 2892 track->mObservedUnderruns = underruns; 2893 // don't count underruns that occur while stopping or pausing 2894 // or stopped which can occur when flush() is called while active 2895 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2896 track->mUnderrunCount += recentUnderruns; 2897 } 2898 2899 // This is similar to the state machine for normal tracks, 2900 // with a few modifications for fast tracks. 2901 bool isActive = true; 2902 switch (track->mState) { 2903 case TrackBase::STOPPING_1: 2904 // track stays active in STOPPING_1 state until first underrun 2905 if (recentUnderruns > 0) { 2906 track->mState = TrackBase::STOPPING_2; 2907 } 2908 break; 2909 case TrackBase::PAUSING: 2910 // ramp down is not yet implemented 2911 track->setPaused(); 2912 break; 2913 case TrackBase::RESUMING: 2914 // ramp up is not yet implemented 2915 track->mState = TrackBase::ACTIVE; 2916 break; 2917 case TrackBase::ACTIVE: 2918 if (recentFull > 0 || recentPartial > 0) { 2919 // track has provided at least some frames recently: reset retry count 2920 track->mRetryCount = kMaxTrackRetries; 2921 } 2922 if (recentUnderruns == 0) { 2923 // no recent underruns: stay active 2924 break; 2925 } 2926 // there has recently been an underrun of some kind 2927 if (track->sharedBuffer() == 0) { 2928 // were any of the recent underruns "empty" (no frames available)? 2929 if (recentEmpty == 0) { 2930 // no, then ignore the partial underruns as they are allowed indefinitely 2931 break; 2932 } 2933 // there has recently been an "empty" underrun: decrement the retry counter 2934 if (--(track->mRetryCount) > 0) { 2935 break; 2936 } 2937 // indicate to client process that the track was disabled because of underrun; 2938 // it will then automatically call start() when data is available 2939 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2940 // remove from active list, but state remains ACTIVE [confusing but true] 2941 isActive = false; 2942 break; 2943 } 2944 // fall through 2945 case TrackBase::STOPPING_2: 2946 case TrackBase::PAUSED: 2947 case TrackBase::TERMINATED: 2948 case TrackBase::STOPPED: 2949 case TrackBase::FLUSHED: // flush() while active 2950 // Check for presentation complete if track is inactive 2951 // We have consumed all the buffers of this track. 2952 // This would be incomplete if we auto-paused on underrun 2953 { 2954 size_t audioHALFrames = 2955 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2956 size_t framesWritten = 2957 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2958 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2959 // track stays in active list until presentation is complete 2960 break; 2961 } 2962 } 2963 if (track->isStopping_2()) { 2964 track->mState = TrackBase::STOPPED; 2965 } 2966 if (track->isStopped()) { 2967 // Can't reset directly, as fast mixer is still polling this track 2968 // track->reset(); 2969 // So instead mark this track as needing to be reset after push with ack 2970 resetMask |= 1 << i; 2971 } 2972 isActive = false; 2973 break; 2974 case TrackBase::IDLE: 2975 default: 2976 LOG_FATAL("unexpected track state %d", track->mState); 2977 } 2978 2979 if (isActive) { 2980 // was it previously inactive? 2981 if (!(state->mTrackMask & (1 << j))) { 2982 ExtendedAudioBufferProvider *eabp = track; 2983 VolumeProvider *vp = track; 2984 fastTrack->mBufferProvider = eabp; 2985 fastTrack->mVolumeProvider = vp; 2986 fastTrack->mSampleRate = track->mSampleRate; 2987 fastTrack->mChannelMask = track->mChannelMask; 2988 fastTrack->mGeneration++; 2989 state->mTrackMask |= 1 << j; 2990 didModify = true; 2991 // no acknowledgement required for newly active tracks 2992 } 2993 // cache the combined master volume and stream type volume for fast mixer; this 2994 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2995 track->mCachedVolume = track->isMuted() ? 2996 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2997 ++fastTracks; 2998 } else { 2999 // was it previously active? 3000 if (state->mTrackMask & (1 << j)) { 3001 fastTrack->mBufferProvider = NULL; 3002 fastTrack->mGeneration++; 3003 state->mTrackMask &= ~(1 << j); 3004 didModify = true; 3005 // If any fast tracks were removed, we must wait for acknowledgement 3006 // because we're about to decrement the last sp<> on those tracks. 3007 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3008 } else { 3009 LOG_FATAL("fast track %d should have been active", j); 3010 } 3011 tracksToRemove->add(track); 3012 // Avoids a misleading display in dumpsys 3013 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3014 } 3015 continue; 3016 } 3017 3018 { // local variable scope to avoid goto warning 3019 3020 audio_track_cblk_t* cblk = track->cblk(); 3021 3022 // The first time a track is added we wait 3023 // for all its buffers to be filled before processing it 3024 int name = track->name(); 3025 // make sure that we have enough frames to mix one full buffer. 3026 // enforce this condition only once to enable draining the buffer in case the client 3027 // app does not call stop() and relies on underrun to stop: 3028 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3029 // during last round 3030 uint32_t minFrames = 1; 3031 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3032 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3033 if (t->sampleRate() == (int)mSampleRate) { 3034 minFrames = mNormalFrameCount; 3035 } else { 3036 // +1 for rounding and +1 for additional sample needed for interpolation 3037 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3038 // add frames already consumed but not yet released by the resampler 3039 // because cblk->framesReady() will include these frames 3040 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3041 // the minimum track buffer size is normally twice the number of frames necessary 3042 // to fill one buffer and the resampler should not leave more than one buffer worth 3043 // of unreleased frames after each pass, but just in case... 3044 ALOG_ASSERT(minFrames <= cblk->frameCount); 3045 } 3046 } 3047 if ((track->framesReady() >= minFrames) && track->isReady() && 3048 !track->isPaused() && !track->isTerminated()) 3049 { 3050 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3051 3052 mixedTracks++; 3053 3054 // track->mainBuffer() != mMixBuffer means there is an effect chain 3055 // connected to the track 3056 chain.clear(); 3057 if (track->mainBuffer() != mMixBuffer) { 3058 chain = getEffectChain_l(track->sessionId()); 3059 // Delegate volume control to effect in track effect chain if needed 3060 if (chain != 0) { 3061 tracksWithEffect++; 3062 } else { 3063 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3064 name, track->sessionId()); 3065 } 3066 } 3067 3068 3069 int param = AudioMixer::VOLUME; 3070 if (track->mFillingUpStatus == Track::FS_FILLED) { 3071 // no ramp for the first volume setting 3072 track->mFillingUpStatus = Track::FS_ACTIVE; 3073 if (track->mState == TrackBase::RESUMING) { 3074 track->mState = TrackBase::ACTIVE; 3075 param = AudioMixer::RAMP_VOLUME; 3076 } 3077 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3078 } else if (cblk->server != 0) { 3079 // If the track is stopped before the first frame was mixed, 3080 // do not apply ramp 3081 param = AudioMixer::RAMP_VOLUME; 3082 } 3083 3084 // compute volume for this track 3085 uint32_t vl, vr, va; 3086 if (track->isMuted() || track->isPausing() || 3087 mStreamTypes[track->streamType()].mute) { 3088 vl = vr = va = 0; 3089 if (track->isPausing()) { 3090 track->setPaused(); 3091 } 3092 } else { 3093 3094 // read original volumes with volume control 3095 float typeVolume = mStreamTypes[track->streamType()].volume; 3096 float v = masterVolume * typeVolume; 3097 uint32_t vlr = cblk->getVolumeLR(); 3098 vl = vlr & 0xFFFF; 3099 vr = vlr >> 16; 3100 // track volumes come from shared memory, so can't be trusted and must be clamped 3101 if (vl > MAX_GAIN_INT) { 3102 ALOGV("Track left volume out of range: %04X", vl); 3103 vl = MAX_GAIN_INT; 3104 } 3105 if (vr > MAX_GAIN_INT) { 3106 ALOGV("Track right volume out of range: %04X", vr); 3107 vr = MAX_GAIN_INT; 3108 } 3109 // now apply the master volume and stream type volume 3110 vl = (uint32_t)(v * vl) << 12; 3111 vr = (uint32_t)(v * vr) << 12; 3112 // assuming master volume and stream type volume each go up to 1.0, 3113 // vl and vr are now in 8.24 format 3114 3115 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3116 // send level comes from shared memory and so may be corrupt 3117 if (sendLevel > MAX_GAIN_INT) { 3118 ALOGV("Track send level out of range: %04X", sendLevel); 3119 sendLevel = MAX_GAIN_INT; 3120 } 3121 va = (uint32_t)(v * sendLevel); 3122 } 3123 // Delegate volume control to effect in track effect chain if needed 3124 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3125 // Do not ramp volume if volume is controlled by effect 3126 param = AudioMixer::VOLUME; 3127 track->mHasVolumeController = true; 3128 } else { 3129 // force no volume ramp when volume controller was just disabled or removed 3130 // from effect chain to avoid volume spike 3131 if (track->mHasVolumeController) { 3132 param = AudioMixer::VOLUME; 3133 } 3134 track->mHasVolumeController = false; 3135 } 3136 3137 // Convert volumes from 8.24 to 4.12 format 3138 // This additional clamping is needed in case chain->setVolume_l() overshot 3139 vl = (vl + (1 << 11)) >> 12; 3140 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3141 vr = (vr + (1 << 11)) >> 12; 3142 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3143 3144 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3145 3146 // XXX: these things DON'T need to be done each time 3147 mAudioMixer->setBufferProvider(name, track); 3148 mAudioMixer->enable(name); 3149 3150 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3151 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3152 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3153 mAudioMixer->setParameter( 3154 name, 3155 AudioMixer::TRACK, 3156 AudioMixer::FORMAT, (void *)track->format()); 3157 mAudioMixer->setParameter( 3158 name, 3159 AudioMixer::TRACK, 3160 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3161 mAudioMixer->setParameter( 3162 name, 3163 AudioMixer::RESAMPLE, 3164 AudioMixer::SAMPLE_RATE, 3165 (void *)(cblk->sampleRate)); 3166 mAudioMixer->setParameter( 3167 name, 3168 AudioMixer::TRACK, 3169 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3170 mAudioMixer->setParameter( 3171 name, 3172 AudioMixer::TRACK, 3173 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3174 3175 // reset retry count 3176 track->mRetryCount = kMaxTrackRetries; 3177 3178 // If one track is ready, set the mixer ready if: 3179 // - the mixer was not ready during previous round OR 3180 // - no other track is not ready 3181 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3182 mixerStatus != MIXER_TRACKS_ENABLED) { 3183 mixerStatus = MIXER_TRACKS_READY; 3184 } 3185 } else { 3186 // clear effect chain input buffer if an active track underruns to avoid sending 3187 // previous audio buffer again to effects 3188 chain = getEffectChain_l(track->sessionId()); 3189 if (chain != 0) { 3190 chain->clearInputBuffer(); 3191 } 3192 3193 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3194 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3195 track->isStopped() || track->isPaused()) { 3196 // We have consumed all the buffers of this track. 3197 // Remove it from the list of active tracks. 3198 // TODO: use actual buffer filling status instead of latency when available from 3199 // audio HAL 3200 size_t audioHALFrames = 3201 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3202 size_t framesWritten = 3203 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3204 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3205 if (track->isStopped()) { 3206 track->reset(); 3207 } 3208 tracksToRemove->add(track); 3209 } 3210 } else { 3211 track->mUnderrunCount++; 3212 // No buffers for this track. Give it a few chances to 3213 // fill a buffer, then remove it from active list. 3214 if (--(track->mRetryCount) <= 0) { 3215 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3216 tracksToRemove->add(track); 3217 // indicate to client process that the track was disabled because of underrun; 3218 // it will then automatically call start() when data is available 3219 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3220 // If one track is not ready, mark the mixer also not ready if: 3221 // - the mixer was ready during previous round OR 3222 // - no other track is ready 3223 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3224 mixerStatus != MIXER_TRACKS_READY) { 3225 mixerStatus = MIXER_TRACKS_ENABLED; 3226 } 3227 } 3228 mAudioMixer->disable(name); 3229 } 3230 3231 } // local variable scope to avoid goto warning 3232track_is_ready: ; 3233 3234 } 3235 3236 // Push the new FastMixer state if necessary 3237 if (didModify) { 3238 state->mFastTracksGen++; 3239 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3240 if (kUseFastMixer == FastMixer_Dynamic && 3241 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3242 state->mCommand = FastMixerState::COLD_IDLE; 3243 state->mColdFutexAddr = &mFastMixerFutex; 3244 state->mColdGen++; 3245 mFastMixerFutex = 0; 3246 if (kUseFastMixer == FastMixer_Dynamic) { 3247 mNormalSink = mOutputSink; 3248 } 3249 // If we go into cold idle, need to wait for acknowledgement 3250 // so that fast mixer stops doing I/O. 3251 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3252 } 3253 sq->end(); 3254 } 3255 if (sq != NULL) { 3256 sq->end(didModify); 3257 sq->push(block); 3258 } 3259 3260 // Now perform the deferred reset on fast tracks that have stopped 3261 while (resetMask != 0) { 3262 size_t i = __builtin_ctz(resetMask); 3263 ALOG_ASSERT(i < count); 3264 resetMask &= ~(1 << i); 3265 sp<Track> t = mActiveTracks[i].promote(); 3266 if (t == 0) continue; 3267 Track* track = t.get(); 3268 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3269 track->reset(); 3270 } 3271 3272 // remove all the tracks that need to be... 3273 count = tracksToRemove->size(); 3274 if (CC_UNLIKELY(count)) { 3275 for (size_t i=0 ; i<count ; i++) { 3276 const sp<Track>& track = tracksToRemove->itemAt(i); 3277 mActiveTracks.remove(track); 3278 if (track->mainBuffer() != mMixBuffer) { 3279 chain = getEffectChain_l(track->sessionId()); 3280 if (chain != 0) { 3281 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3282 chain->decActiveTrackCnt(); 3283 } 3284 } 3285 if (track->isTerminated()) { 3286 removeTrack_l(track); 3287 } 3288 } 3289 } 3290 3291 // mix buffer must be cleared if all tracks are connected to an 3292 // effect chain as in this case the mixer will not write to 3293 // mix buffer and track effects will accumulate into it 3294 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3295 // FIXME as a performance optimization, should remember previous zero status 3296 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3297 } 3298 3299 // if any fast tracks, then status is ready 3300 mMixerStatusIgnoringFastTracks = mixerStatus; 3301 if (fastTracks > 0) { 3302 mixerStatus = MIXER_TRACKS_READY; 3303 } 3304 return mixerStatus; 3305} 3306 3307/* 3308The derived values that are cached: 3309 - mixBufferSize from frame count * frame size 3310 - activeSleepTime from activeSleepTimeUs() 3311 - idleSleepTime from idleSleepTimeUs() 3312 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3313 - maxPeriod from frame count and sample rate (MIXER only) 3314 3315The parameters that affect these derived values are: 3316 - frame count 3317 - frame size 3318 - sample rate 3319 - device type: A2DP or not 3320 - device latency 3321 - format: PCM or not 3322 - active sleep time 3323 - idle sleep time 3324*/ 3325 3326void AudioFlinger::PlaybackThread::cacheParameters_l() 3327{ 3328 mixBufferSize = mNormalFrameCount * mFrameSize; 3329 activeSleepTime = activeSleepTimeUs(); 3330 idleSleepTime = idleSleepTimeUs(); 3331} 3332 3333void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3334{ 3335 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3336 this, streamType, mTracks.size()); 3337 Mutex::Autolock _l(mLock); 3338 3339 size_t size = mTracks.size(); 3340 for (size_t i = 0; i < size; i++) { 3341 sp<Track> t = mTracks[i]; 3342 if (t->streamType() == streamType) { 3343 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3344 t->mCblk->cv.signal(); 3345 } 3346 } 3347} 3348 3349// getTrackName_l() must be called with ThreadBase::mLock held 3350int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3351{ 3352 return mAudioMixer->getTrackName(channelMask); 3353} 3354 3355// deleteTrackName_l() must be called with ThreadBase::mLock held 3356void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3357{ 3358 ALOGV("remove track (%d) and delete from mixer", name); 3359 mAudioMixer->deleteTrackName(name); 3360} 3361 3362// checkForNewParameters_l() must be called with ThreadBase::mLock held 3363bool AudioFlinger::MixerThread::checkForNewParameters_l() 3364{ 3365 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3366 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3367 bool reconfig = false; 3368 3369 while (!mNewParameters.isEmpty()) { 3370 3371 if (mFastMixer != NULL) { 3372 FastMixerStateQueue *sq = mFastMixer->sq(); 3373 FastMixerState *state = sq->begin(); 3374 if (!(state->mCommand & FastMixerState::IDLE)) { 3375 previousCommand = state->mCommand; 3376 state->mCommand = FastMixerState::HOT_IDLE; 3377 sq->end(); 3378 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3379 } else { 3380 sq->end(false /*didModify*/); 3381 } 3382 } 3383 3384 status_t status = NO_ERROR; 3385 String8 keyValuePair = mNewParameters[0]; 3386 AudioParameter param = AudioParameter(keyValuePair); 3387 int value; 3388 3389 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3390 reconfig = true; 3391 } 3392 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3393 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3394 status = BAD_VALUE; 3395 } else { 3396 reconfig = true; 3397 } 3398 } 3399 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3400 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3401 status = BAD_VALUE; 3402 } else { 3403 reconfig = true; 3404 } 3405 } 3406 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3407 // do not accept frame count changes if tracks are open as the track buffer 3408 // size depends on frame count and correct behavior would not be guaranteed 3409 // if frame count is changed after track creation 3410 if (!mTracks.isEmpty()) { 3411 status = INVALID_OPERATION; 3412 } else { 3413 reconfig = true; 3414 } 3415 } 3416 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3417#ifdef ADD_BATTERY_DATA 3418 // when changing the audio output device, call addBatteryData to notify 3419 // the change 3420 if ((int)mDevice != value) { 3421 uint32_t params = 0; 3422 // check whether speaker is on 3423 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3424 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3425 } 3426 3427 int deviceWithoutSpeaker 3428 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3429 // check if any other device (except speaker) is on 3430 if (value & deviceWithoutSpeaker ) { 3431 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3432 } 3433 3434 if (params != 0) { 3435 addBatteryData(params); 3436 } 3437 } 3438#endif 3439 3440 // forward device change to effects that have requested to be 3441 // aware of attached audio device. 3442 mDevice = (uint32_t)value; 3443 for (size_t i = 0; i < mEffectChains.size(); i++) { 3444 mEffectChains[i]->setDevice_l(mDevice); 3445 } 3446 } 3447 3448 if (status == NO_ERROR) { 3449 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3450 keyValuePair.string()); 3451 if (!mStandby && status == INVALID_OPERATION) { 3452 mOutput->stream->common.standby(&mOutput->stream->common); 3453 mStandby = true; 3454 mBytesWritten = 0; 3455 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3456 keyValuePair.string()); 3457 } 3458 if (status == NO_ERROR && reconfig) { 3459 delete mAudioMixer; 3460 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3461 mAudioMixer = NULL; 3462 readOutputParameters(); 3463 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3464 for (size_t i = 0; i < mTracks.size() ; i++) { 3465 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3466 if (name < 0) break; 3467 mTracks[i]->mName = name; 3468 // limit track sample rate to 2 x new output sample rate 3469 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3470 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3471 } 3472 } 3473 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3474 } 3475 } 3476 3477 mNewParameters.removeAt(0); 3478 3479 mParamStatus = status; 3480 mParamCond.signal(); 3481 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3482 // already timed out waiting for the status and will never signal the condition. 3483 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3484 } 3485 3486 if (!(previousCommand & FastMixerState::IDLE)) { 3487 ALOG_ASSERT(mFastMixer != NULL); 3488 FastMixerStateQueue *sq = mFastMixer->sq(); 3489 FastMixerState *state = sq->begin(); 3490 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3491 state->mCommand = previousCommand; 3492 sq->end(); 3493 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3494 } 3495 3496 return reconfig; 3497} 3498 3499status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3500{ 3501 const size_t SIZE = 256; 3502 char buffer[SIZE]; 3503 String8 result; 3504 3505 PlaybackThread::dumpInternals(fd, args); 3506 3507 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3508 result.append(buffer); 3509 write(fd, result.string(), result.size()); 3510 3511 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3512 FastMixerDumpState copy = mFastMixerDumpState; 3513 copy.dump(fd); 3514 3515#ifdef STATE_QUEUE_DUMP 3516 // Similar for state queue 3517 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3518 observerCopy.dump(fd); 3519 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3520 mutatorCopy.dump(fd); 3521#endif 3522 3523 // Write the tee output to a .wav file 3524 NBAIO_Source *teeSource = mTeeSource.get(); 3525 if (teeSource != NULL) { 3526 char teePath[64]; 3527 struct timeval tv; 3528 gettimeofday(&tv, NULL); 3529 struct tm tm; 3530 localtime_r(&tv.tv_sec, &tm); 3531 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3532 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3533 if (teeFd >= 0) { 3534 char wavHeader[44]; 3535 memcpy(wavHeader, 3536 "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", 3537 sizeof(wavHeader)); 3538 NBAIO_Format format = teeSource->format(); 3539 unsigned channelCount = Format_channelCount(format); 3540 ALOG_ASSERT(channelCount <= FCC_2); 3541 unsigned sampleRate = Format_sampleRate(format); 3542 wavHeader[22] = channelCount; // number of channels 3543 wavHeader[24] = sampleRate; // sample rate 3544 wavHeader[25] = sampleRate >> 8; 3545 wavHeader[32] = channelCount * 2; // block alignment 3546 write(teeFd, wavHeader, sizeof(wavHeader)); 3547 size_t total = 0; 3548 bool firstRead = true; 3549 for (;;) { 3550#define TEE_SINK_READ 1024 3551 short buffer[TEE_SINK_READ * FCC_2]; 3552 size_t count = TEE_SINK_READ; 3553 ssize_t actual = teeSource->read(buffer, count); 3554 bool wasFirstRead = firstRead; 3555 firstRead = false; 3556 if (actual <= 0) { 3557 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3558 continue; 3559 } 3560 break; 3561 } 3562 ALOG_ASSERT(actual <= count); 3563 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3564 total += actual; 3565 } 3566 lseek(teeFd, (off_t) 4, SEEK_SET); 3567 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3568 write(teeFd, &temp, sizeof(temp)); 3569 lseek(teeFd, (off_t) 40, SEEK_SET); 3570 temp = total * channelCount * sizeof(short); 3571 write(teeFd, &temp, sizeof(temp)); 3572 close(teeFd); 3573 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3574 } else { 3575 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3576 } 3577 } 3578 3579 return NO_ERROR; 3580} 3581 3582uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3583{ 3584 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3585} 3586 3587uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3588{ 3589 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3590} 3591 3592void AudioFlinger::MixerThread::cacheParameters_l() 3593{ 3594 PlaybackThread::cacheParameters_l(); 3595 3596 // FIXME: Relaxed timing because of a certain device that can't meet latency 3597 // Should be reduced to 2x after the vendor fixes the driver issue 3598 // increase threshold again due to low power audio mode. The way this warning 3599 // threshold is calculated and its usefulness should be reconsidered anyway. 3600 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3601} 3602 3603// ---------------------------------------------------------------------------- 3604AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3605 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3606 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3607 // mLeftVolFloat, mRightVolFloat 3608{ 3609} 3610 3611AudioFlinger::DirectOutputThread::~DirectOutputThread() 3612{ 3613} 3614 3615AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3616 Vector< sp<Track> > *tracksToRemove 3617) 3618{ 3619 sp<Track> trackToRemove; 3620 3621 mixer_state mixerStatus = MIXER_IDLE; 3622 3623 // find out which tracks need to be processed 3624 if (mActiveTracks.size() != 0) { 3625 sp<Track> t = mActiveTracks[0].promote(); 3626 // The track died recently 3627 if (t == 0) return MIXER_IDLE; 3628 3629 Track* const track = t.get(); 3630 audio_track_cblk_t* cblk = track->cblk(); 3631 3632 // The first time a track is added we wait 3633 // for all its buffers to be filled before processing it 3634 uint32_t minFrames; 3635 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3636 minFrames = mNormalFrameCount; 3637 } else { 3638 minFrames = 1; 3639 } 3640 if ((track->framesReady() >= minFrames) && track->isReady() && 3641 !track->isPaused() && !track->isTerminated()) 3642 { 3643 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3644 3645 if (track->mFillingUpStatus == Track::FS_FILLED) { 3646 track->mFillingUpStatus = Track::FS_ACTIVE; 3647 mLeftVolFloat = mRightVolFloat = 0; 3648 if (track->mState == TrackBase::RESUMING) { 3649 track->mState = TrackBase::ACTIVE; 3650 } 3651 } 3652 3653 // compute volume for this track 3654 float left, right; 3655 if (track->isMuted() || mMasterMute || track->isPausing() || 3656 mStreamTypes[track->streamType()].mute) { 3657 left = right = 0; 3658 if (track->isPausing()) { 3659 track->setPaused(); 3660 } 3661 } else { 3662 float typeVolume = mStreamTypes[track->streamType()].volume; 3663 float v = mMasterVolume * typeVolume; 3664 uint32_t vlr = cblk->getVolumeLR(); 3665 float v_clamped = v * (vlr & 0xFFFF); 3666 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3667 left = v_clamped/MAX_GAIN; 3668 v_clamped = v * (vlr >> 16); 3669 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3670 right = v_clamped/MAX_GAIN; 3671 } 3672 3673 if (left != mLeftVolFloat || right != mRightVolFloat) { 3674 mLeftVolFloat = left; 3675 mRightVolFloat = right; 3676 3677 // Convert volumes from float to 8.24 3678 uint32_t vl = (uint32_t)(left * (1 << 24)); 3679 uint32_t vr = (uint32_t)(right * (1 << 24)); 3680 3681 // Delegate volume control to effect in track effect chain if needed 3682 // only one effect chain can be present on DirectOutputThread, so if 3683 // there is one, the track is connected to it 3684 if (!mEffectChains.isEmpty()) { 3685 // Do not ramp volume if volume is controlled by effect 3686 mEffectChains[0]->setVolume_l(&vl, &vr); 3687 left = (float)vl / (1 << 24); 3688 right = (float)vr / (1 << 24); 3689 } 3690 mOutput->stream->set_volume(mOutput->stream, left, right); 3691 } 3692 3693 // reset retry count 3694 track->mRetryCount = kMaxTrackRetriesDirect; 3695 mActiveTrack = t; 3696 mixerStatus = MIXER_TRACKS_READY; 3697 } else { 3698 // clear effect chain input buffer if an active track underruns to avoid sending 3699 // previous audio buffer again to effects 3700 if (!mEffectChains.isEmpty()) { 3701 mEffectChains[0]->clearInputBuffer(); 3702 } 3703 3704 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3705 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3706 track->isStopped() || track->isPaused()) { 3707 // We have consumed all the buffers of this track. 3708 // Remove it from the list of active tracks. 3709 // TODO: implement behavior for compressed audio 3710 size_t audioHALFrames = 3711 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3712 size_t framesWritten = 3713 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3714 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3715 if (track->isStopped()) { 3716 track->reset(); 3717 } 3718 trackToRemove = track; 3719 } 3720 } else { 3721 // No buffers for this track. Give it a few chances to 3722 // fill a buffer, then remove it from active list. 3723 if (--(track->mRetryCount) <= 0) { 3724 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3725 trackToRemove = track; 3726 } else { 3727 mixerStatus = MIXER_TRACKS_ENABLED; 3728 } 3729 } 3730 } 3731 } 3732 3733 // FIXME merge this with similar code for removing multiple tracks 3734 // remove all the tracks that need to be... 3735 if (CC_UNLIKELY(trackToRemove != 0)) { 3736 tracksToRemove->add(trackToRemove); 3737 mActiveTracks.remove(trackToRemove); 3738 if (!mEffectChains.isEmpty()) { 3739 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3740 trackToRemove->sessionId()); 3741 mEffectChains[0]->decActiveTrackCnt(); 3742 } 3743 if (trackToRemove->isTerminated()) { 3744 removeTrack_l(trackToRemove); 3745 } 3746 } 3747 3748 return mixerStatus; 3749} 3750 3751void AudioFlinger::DirectOutputThread::threadLoop_mix() 3752{ 3753 AudioBufferProvider::Buffer buffer; 3754 size_t frameCount = mFrameCount; 3755 int8_t *curBuf = (int8_t *)mMixBuffer; 3756 // output audio to hardware 3757 while (frameCount) { 3758 buffer.frameCount = frameCount; 3759 mActiveTrack->getNextBuffer(&buffer); 3760 if (CC_UNLIKELY(buffer.raw == NULL)) { 3761 memset(curBuf, 0, frameCount * mFrameSize); 3762 break; 3763 } 3764 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3765 frameCount -= buffer.frameCount; 3766 curBuf += buffer.frameCount * mFrameSize; 3767 mActiveTrack->releaseBuffer(&buffer); 3768 } 3769 sleepTime = 0; 3770 standbyTime = systemTime() + standbyDelay; 3771 mActiveTrack.clear(); 3772 3773} 3774 3775void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3776{ 3777 if (sleepTime == 0) { 3778 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3779 sleepTime = activeSleepTime; 3780 } else { 3781 sleepTime = idleSleepTime; 3782 } 3783 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3784 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3785 sleepTime = 0; 3786 } 3787} 3788 3789// getTrackName_l() must be called with ThreadBase::mLock held 3790int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3791{ 3792 return 0; 3793} 3794 3795// deleteTrackName_l() must be called with ThreadBase::mLock held 3796void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3797{ 3798} 3799 3800// checkForNewParameters_l() must be called with ThreadBase::mLock held 3801bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3802{ 3803 bool reconfig = false; 3804 3805 while (!mNewParameters.isEmpty()) { 3806 status_t status = NO_ERROR; 3807 String8 keyValuePair = mNewParameters[0]; 3808 AudioParameter param = AudioParameter(keyValuePair); 3809 int value; 3810 3811 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3812 // do not accept frame count changes if tracks are open as the track buffer 3813 // size depends on frame count and correct behavior would not be garantied 3814 // if frame count is changed after track creation 3815 if (!mTracks.isEmpty()) { 3816 status = INVALID_OPERATION; 3817 } else { 3818 reconfig = true; 3819 } 3820 } 3821 if (status == NO_ERROR) { 3822 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3823 keyValuePair.string()); 3824 if (!mStandby && status == INVALID_OPERATION) { 3825 mOutput->stream->common.standby(&mOutput->stream->common); 3826 mStandby = true; 3827 mBytesWritten = 0; 3828 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3829 keyValuePair.string()); 3830 } 3831 if (status == NO_ERROR && reconfig) { 3832 readOutputParameters(); 3833 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3834 } 3835 } 3836 3837 mNewParameters.removeAt(0); 3838 3839 mParamStatus = status; 3840 mParamCond.signal(); 3841 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3842 // already timed out waiting for the status and will never signal the condition. 3843 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3844 } 3845 return reconfig; 3846} 3847 3848uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3849{ 3850 uint32_t time; 3851 if (audio_is_linear_pcm(mFormat)) { 3852 time = PlaybackThread::activeSleepTimeUs(); 3853 } else { 3854 time = 10000; 3855 } 3856 return time; 3857} 3858 3859uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3860{ 3861 uint32_t time; 3862 if (audio_is_linear_pcm(mFormat)) { 3863 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3864 } else { 3865 time = 10000; 3866 } 3867 return time; 3868} 3869 3870uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3871{ 3872 uint32_t time; 3873 if (audio_is_linear_pcm(mFormat)) { 3874 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3875 } else { 3876 time = 10000; 3877 } 3878 return time; 3879} 3880 3881void AudioFlinger::DirectOutputThread::cacheParameters_l() 3882{ 3883 PlaybackThread::cacheParameters_l(); 3884 3885 // use shorter standby delay as on normal output to release 3886 // hardware resources as soon as possible 3887 standbyDelay = microseconds(activeSleepTime*2); 3888} 3889 3890// ---------------------------------------------------------------------------- 3891 3892AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3893 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3894 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3895 mWaitTimeMs(UINT_MAX) 3896{ 3897 addOutputTrack(mainThread); 3898} 3899 3900AudioFlinger::DuplicatingThread::~DuplicatingThread() 3901{ 3902 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3903 mOutputTracks[i]->destroy(); 3904 } 3905} 3906 3907void AudioFlinger::DuplicatingThread::threadLoop_mix() 3908{ 3909 // mix buffers... 3910 if (outputsReady(outputTracks)) { 3911 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3912 } else { 3913 memset(mMixBuffer, 0, mixBufferSize); 3914 } 3915 sleepTime = 0; 3916 writeFrames = mNormalFrameCount; 3917 standbyTime = systemTime() + standbyDelay; 3918} 3919 3920void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3921{ 3922 if (sleepTime == 0) { 3923 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3924 sleepTime = activeSleepTime; 3925 } else { 3926 sleepTime = idleSleepTime; 3927 } 3928 } else if (mBytesWritten != 0) { 3929 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3930 writeFrames = mNormalFrameCount; 3931 memset(mMixBuffer, 0, mixBufferSize); 3932 } else { 3933 // flush remaining overflow buffers in output tracks 3934 writeFrames = 0; 3935 } 3936 sleepTime = 0; 3937 } 3938} 3939 3940void AudioFlinger::DuplicatingThread::threadLoop_write() 3941{ 3942 for (size_t i = 0; i < outputTracks.size(); i++) { 3943 outputTracks[i]->write(mMixBuffer, writeFrames); 3944 } 3945 mBytesWritten += mixBufferSize; 3946} 3947 3948void AudioFlinger::DuplicatingThread::threadLoop_standby() 3949{ 3950 // DuplicatingThread implements standby by stopping all tracks 3951 for (size_t i = 0; i < outputTracks.size(); i++) { 3952 outputTracks[i]->stop(); 3953 } 3954} 3955 3956void AudioFlinger::DuplicatingThread::saveOutputTracks() 3957{ 3958 outputTracks = mOutputTracks; 3959} 3960 3961void AudioFlinger::DuplicatingThread::clearOutputTracks() 3962{ 3963 outputTracks.clear(); 3964} 3965 3966void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3967{ 3968 Mutex::Autolock _l(mLock); 3969 // FIXME explain this formula 3970 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3971 OutputTrack *outputTrack = new OutputTrack(thread, 3972 this, 3973 mSampleRate, 3974 mFormat, 3975 mChannelMask, 3976 frameCount); 3977 if (outputTrack->cblk() != NULL) { 3978 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3979 mOutputTracks.add(outputTrack); 3980 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3981 updateWaitTime_l(); 3982 } 3983} 3984 3985void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3986{ 3987 Mutex::Autolock _l(mLock); 3988 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3989 if (mOutputTracks[i]->thread() == thread) { 3990 mOutputTracks[i]->destroy(); 3991 mOutputTracks.removeAt(i); 3992 updateWaitTime_l(); 3993 return; 3994 } 3995 } 3996 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3997} 3998 3999// caller must hold mLock 4000void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4001{ 4002 mWaitTimeMs = UINT_MAX; 4003 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4004 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4005 if (strong != 0) { 4006 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4007 if (waitTimeMs < mWaitTimeMs) { 4008 mWaitTimeMs = waitTimeMs; 4009 } 4010 } 4011 } 4012} 4013 4014 4015bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4016{ 4017 for (size_t i = 0; i < outputTracks.size(); i++) { 4018 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4019 if (thread == 0) { 4020 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4021 return false; 4022 } 4023 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4024 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4025 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4026 return false; 4027 } 4028 } 4029 return true; 4030} 4031 4032uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4033{ 4034 return (mWaitTimeMs * 1000) / 2; 4035} 4036 4037void AudioFlinger::DuplicatingThread::cacheParameters_l() 4038{ 4039 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4040 updateWaitTime_l(); 4041 4042 MixerThread::cacheParameters_l(); 4043} 4044 4045// ---------------------------------------------------------------------------- 4046 4047// TrackBase constructor must be called with AudioFlinger::mLock held 4048AudioFlinger::ThreadBase::TrackBase::TrackBase( 4049 ThreadBase *thread, 4050 const sp<Client>& client, 4051 uint32_t sampleRate, 4052 audio_format_t format, 4053 uint32_t channelMask, 4054 int frameCount, 4055 const sp<IMemory>& sharedBuffer, 4056 int sessionId) 4057 : RefBase(), 4058 mThread(thread), 4059 mClient(client), 4060 mCblk(NULL), 4061 // mBuffer 4062 // mBufferEnd 4063 mFrameCount(0), 4064 mState(IDLE), 4065 mSampleRate(sampleRate), 4066 mFormat(format), 4067 mStepServerFailed(false), 4068 mSessionId(sessionId) 4069 // mChannelCount 4070 // mChannelMask 4071{ 4072 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4073 4074 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4075 size_t size = sizeof(audio_track_cblk_t); 4076 uint8_t channelCount = popcount(channelMask); 4077 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4078 if (sharedBuffer == 0) { 4079 size += bufferSize; 4080 } 4081 4082 if (client != NULL) { 4083 mCblkMemory = client->heap()->allocate(size); 4084 if (mCblkMemory != 0) { 4085 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4086 if (mCblk != NULL) { // construct the shared structure in-place. 4087 new(mCblk) audio_track_cblk_t(); 4088 // clear all buffers 4089 mCblk->frameCount = frameCount; 4090 mCblk->sampleRate = sampleRate; 4091// uncomment the following lines to quickly test 32-bit wraparound 4092// mCblk->user = 0xffff0000; 4093// mCblk->server = 0xffff0000; 4094// mCblk->userBase = 0xffff0000; 4095// mCblk->serverBase = 0xffff0000; 4096 mChannelCount = channelCount; 4097 mChannelMask = channelMask; 4098 if (sharedBuffer == 0) { 4099 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4100 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4101 // Force underrun condition to avoid false underrun callback until first data is 4102 // written to buffer (other flags are cleared) 4103 mCblk->flags = CBLK_UNDERRUN_ON; 4104 } else { 4105 mBuffer = sharedBuffer->pointer(); 4106 } 4107 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4108 } 4109 } else { 4110 ALOGE("not enough memory for AudioTrack size=%u", size); 4111 client->heap()->dump("AudioTrack"); 4112 return; 4113 } 4114 } else { 4115 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4116 // construct the shared structure in-place. 4117 new(mCblk) audio_track_cblk_t(); 4118 // clear all buffers 4119 mCblk->frameCount = frameCount; 4120 mCblk->sampleRate = sampleRate; 4121// uncomment the following lines to quickly test 32-bit wraparound 4122// mCblk->user = 0xffff0000; 4123// mCblk->server = 0xffff0000; 4124// mCblk->userBase = 0xffff0000; 4125// mCblk->serverBase = 0xffff0000; 4126 mChannelCount = channelCount; 4127 mChannelMask = channelMask; 4128 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4129 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4130 // Force underrun condition to avoid false underrun callback until first data is 4131 // written to buffer (other flags are cleared) 4132 mCblk->flags = CBLK_UNDERRUN_ON; 4133 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4134 } 4135} 4136 4137AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4138{ 4139 if (mCblk != NULL) { 4140 if (mClient == 0) { 4141 delete mCblk; 4142 } else { 4143 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4144 } 4145 } 4146 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4147 if (mClient != 0) { 4148 // Client destructor must run with AudioFlinger mutex locked 4149 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4150 // If the client's reference count drops to zero, the associated destructor 4151 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4152 // relying on the automatic clear() at end of scope. 4153 mClient.clear(); 4154 } 4155} 4156 4157// AudioBufferProvider interface 4158// getNextBuffer() = 0; 4159// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4160void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4161{ 4162 buffer->raw = NULL; 4163 mFrameCount = buffer->frameCount; 4164 // FIXME See note at getNextBuffer() 4165 (void) step(); // ignore return value of step() 4166 buffer->frameCount = 0; 4167} 4168 4169bool AudioFlinger::ThreadBase::TrackBase::step() { 4170 bool result; 4171 audio_track_cblk_t* cblk = this->cblk(); 4172 4173 result = cblk->stepServer(mFrameCount); 4174 if (!result) { 4175 ALOGV("stepServer failed acquiring cblk mutex"); 4176 mStepServerFailed = true; 4177 } 4178 return result; 4179} 4180 4181void AudioFlinger::ThreadBase::TrackBase::reset() { 4182 audio_track_cblk_t* cblk = this->cblk(); 4183 4184 cblk->user = 0; 4185 cblk->server = 0; 4186 cblk->userBase = 0; 4187 cblk->serverBase = 0; 4188 mStepServerFailed = false; 4189 ALOGV("TrackBase::reset"); 4190} 4191 4192int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4193 return (int)mCblk->sampleRate; 4194} 4195 4196void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4197 audio_track_cblk_t* cblk = this->cblk(); 4198 size_t frameSize = cblk->frameSize; 4199 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4200 int8_t *bufferEnd = bufferStart + frames * frameSize; 4201 4202 // Check validity of returned pointer in case the track control block would have been corrupted. 4203 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4204 "TrackBase::getBuffer buffer out of range:\n" 4205 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4206 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4207 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4208 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4209 4210 return bufferStart; 4211} 4212 4213status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4214{ 4215 mSyncEvents.add(event); 4216 return NO_ERROR; 4217} 4218 4219// ---------------------------------------------------------------------------- 4220 4221// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4222AudioFlinger::PlaybackThread::Track::Track( 4223 PlaybackThread *thread, 4224 const sp<Client>& client, 4225 audio_stream_type_t streamType, 4226 uint32_t sampleRate, 4227 audio_format_t format, 4228 uint32_t channelMask, 4229 int frameCount, 4230 const sp<IMemory>& sharedBuffer, 4231 int sessionId, 4232 IAudioFlinger::track_flags_t flags) 4233 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4234 mMute(false), 4235 mFillingUpStatus(FS_INVALID), 4236 // mRetryCount initialized later when needed 4237 mSharedBuffer(sharedBuffer), 4238 mStreamType(streamType), 4239 mName(-1), // see note below 4240 mMainBuffer(thread->mixBuffer()), 4241 mAuxBuffer(NULL), 4242 mAuxEffectId(0), mHasVolumeController(false), 4243 mPresentationCompleteFrames(0), 4244 mFlags(flags), 4245 mFastIndex(-1), 4246 mUnderrunCount(0), 4247 mCachedVolume(1.0) 4248{ 4249 if (mCblk != NULL) { 4250 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4251 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4252 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4253 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4254 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4255 if (mName < 0) { 4256 ALOGE("no more track names available"); 4257 return; 4258 } 4259 // only allocate a fast track index if we were able to allocate a normal track name 4260 if (flags & IAudioFlinger::TRACK_FAST) { 4261 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4262 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4263 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4264 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4265 // FIXME This is too eager. We allocate a fast track index before the 4266 // fast track becomes active. Since fast tracks are a scarce resource, 4267 // this means we are potentially denying other more important fast tracks from 4268 // being created. It would be better to allocate the index dynamically. 4269 mFastIndex = i; 4270 // Read the initial underruns because this field is never cleared by the fast mixer 4271 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4272 thread->mFastTrackAvailMask &= ~(1 << i); 4273 } 4274 } 4275 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4276} 4277 4278AudioFlinger::PlaybackThread::Track::~Track() 4279{ 4280 ALOGV("PlaybackThread::Track destructor"); 4281 sp<ThreadBase> thread = mThread.promote(); 4282 if (thread != 0) { 4283 Mutex::Autolock _l(thread->mLock); 4284 mState = TERMINATED; 4285 } 4286} 4287 4288void AudioFlinger::PlaybackThread::Track::destroy() 4289{ 4290 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4291 // by removing it from mTracks vector, so there is a risk that this Tracks's 4292 // destructor is called. As the destructor needs to lock mLock, 4293 // we must acquire a strong reference on this Track before locking mLock 4294 // here so that the destructor is called only when exiting this function. 4295 // On the other hand, as long as Track::destroy() is only called by 4296 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4297 // this Track with its member mTrack. 4298 sp<Track> keep(this); 4299 { // scope for mLock 4300 sp<ThreadBase> thread = mThread.promote(); 4301 if (thread != 0) { 4302 if (!isOutputTrack()) { 4303 if (mState == ACTIVE || mState == RESUMING) { 4304 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4305 4306#ifdef ADD_BATTERY_DATA 4307 // to track the speaker usage 4308 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4309#endif 4310 } 4311 AudioSystem::releaseOutput(thread->id()); 4312 } 4313 Mutex::Autolock _l(thread->mLock); 4314 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4315 playbackThread->destroyTrack_l(this); 4316 } 4317 } 4318} 4319 4320/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4321{ 4322 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4323 " Server User Main buf Aux Buf Flags Underruns\n"); 4324} 4325 4326void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4327{ 4328 uint32_t vlr = mCblk->getVolumeLR(); 4329 if (isFastTrack()) { 4330 sprintf(buffer, " F %2d", mFastIndex); 4331 } else { 4332 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4333 } 4334 track_state state = mState; 4335 char stateChar; 4336 switch (state) { 4337 case IDLE: 4338 stateChar = 'I'; 4339 break; 4340 case TERMINATED: 4341 stateChar = 'T'; 4342 break; 4343 case STOPPING_1: 4344 stateChar = 's'; 4345 break; 4346 case STOPPING_2: 4347 stateChar = '5'; 4348 break; 4349 case STOPPED: 4350 stateChar = 'S'; 4351 break; 4352 case RESUMING: 4353 stateChar = 'R'; 4354 break; 4355 case ACTIVE: 4356 stateChar = 'A'; 4357 break; 4358 case PAUSING: 4359 stateChar = 'p'; 4360 break; 4361 case PAUSED: 4362 stateChar = 'P'; 4363 break; 4364 case FLUSHED: 4365 stateChar = 'F'; 4366 break; 4367 default: 4368 stateChar = '?'; 4369 break; 4370 } 4371 char nowInUnderrun; 4372 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4373 case UNDERRUN_FULL: 4374 nowInUnderrun = ' '; 4375 break; 4376 case UNDERRUN_PARTIAL: 4377 nowInUnderrun = '<'; 4378 break; 4379 case UNDERRUN_EMPTY: 4380 nowInUnderrun = '*'; 4381 break; 4382 default: 4383 nowInUnderrun = '?'; 4384 break; 4385 } 4386 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4387 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4388 (mClient == 0) ? getpid_cached : mClient->pid(), 4389 mStreamType, 4390 mFormat, 4391 mChannelMask, 4392 mSessionId, 4393 mFrameCount, 4394 mCblk->frameCount, 4395 stateChar, 4396 mMute, 4397 mFillingUpStatus, 4398 mCblk->sampleRate, 4399 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4400 20.0 * log10((vlr >> 16) / 4096.0), 4401 mCblk->server, 4402 mCblk->user, 4403 (int)mMainBuffer, 4404 (int)mAuxBuffer, 4405 mCblk->flags, 4406 mUnderrunCount, 4407 nowInUnderrun); 4408} 4409 4410// AudioBufferProvider interface 4411status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4412 AudioBufferProvider::Buffer* buffer, int64_t pts) 4413{ 4414 audio_track_cblk_t* cblk = this->cblk(); 4415 uint32_t framesReady; 4416 uint32_t framesReq = buffer->frameCount; 4417 4418 // Check if last stepServer failed, try to step now 4419 if (mStepServerFailed) { 4420 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4421 // Since the fast mixer is higher priority than client callback thread, 4422 // it does not result in priority inversion for client. 4423 // But a non-blocking solution would be preferable to avoid 4424 // fast mixer being unable to tryLock(), and 4425 // to avoid the extra context switches if the client wakes up, 4426 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4427 if (!step()) goto getNextBuffer_exit; 4428 ALOGV("stepServer recovered"); 4429 mStepServerFailed = false; 4430 } 4431 4432 // FIXME Same as above 4433 framesReady = cblk->framesReady(); 4434 4435 if (CC_LIKELY(framesReady)) { 4436 uint32_t s = cblk->server; 4437 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4438 4439 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4440 if (framesReq > framesReady) { 4441 framesReq = framesReady; 4442 } 4443 if (framesReq > bufferEnd - s) { 4444 framesReq = bufferEnd - s; 4445 } 4446 4447 buffer->raw = getBuffer(s, framesReq); 4448 if (buffer->raw == NULL) goto getNextBuffer_exit; 4449 4450 buffer->frameCount = framesReq; 4451 return NO_ERROR; 4452 } 4453 4454getNextBuffer_exit: 4455 buffer->raw = NULL; 4456 buffer->frameCount = 0; 4457 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4458 return NOT_ENOUGH_DATA; 4459} 4460 4461// Note that framesReady() takes a mutex on the control block using tryLock(). 4462// This could result in priority inversion if framesReady() is called by the normal mixer, 4463// as the normal mixer thread runs at lower 4464// priority than the client's callback thread: there is a short window within framesReady() 4465// during which the normal mixer could be preempted, and the client callback would block. 4466// Another problem can occur if framesReady() is called by the fast mixer: 4467// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4468// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4469size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4470 return mCblk->framesReady(); 4471} 4472 4473// Don't call for fast tracks; the framesReady() could result in priority inversion 4474bool AudioFlinger::PlaybackThread::Track::isReady() const { 4475 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4476 4477 if (framesReady() >= mCblk->frameCount || 4478 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4479 mFillingUpStatus = FS_FILLED; 4480 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4481 return true; 4482 } 4483 return false; 4484} 4485 4486status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4487 int triggerSession) 4488{ 4489 status_t status = NO_ERROR; 4490 ALOGV("start(%d), calling pid %d session %d", 4491 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4492 4493 sp<ThreadBase> thread = mThread.promote(); 4494 if (thread != 0) { 4495 Mutex::Autolock _l(thread->mLock); 4496 track_state state = mState; 4497 // here the track could be either new, or restarted 4498 // in both cases "unstop" the track 4499 if (mState == PAUSED) { 4500 mState = TrackBase::RESUMING; 4501 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4502 } else { 4503 mState = TrackBase::ACTIVE; 4504 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4505 } 4506 4507 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4508 thread->mLock.unlock(); 4509 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4510 thread->mLock.lock(); 4511 4512#ifdef ADD_BATTERY_DATA 4513 // to track the speaker usage 4514 if (status == NO_ERROR) { 4515 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4516 } 4517#endif 4518 } 4519 if (status == NO_ERROR) { 4520 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4521 playbackThread->addTrack_l(this); 4522 } else { 4523 mState = state; 4524 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4525 } 4526 } else { 4527 status = BAD_VALUE; 4528 } 4529 return status; 4530} 4531 4532void AudioFlinger::PlaybackThread::Track::stop() 4533{ 4534 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4535 sp<ThreadBase> thread = mThread.promote(); 4536 if (thread != 0) { 4537 Mutex::Autolock _l(thread->mLock); 4538 track_state state = mState; 4539 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4540 // If the track is not active (PAUSED and buffers full), flush buffers 4541 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4542 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4543 reset(); 4544 mState = STOPPED; 4545 } else if (!isFastTrack()) { 4546 mState = STOPPED; 4547 } else { 4548 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4549 // and then to STOPPED and reset() when presentation is complete 4550 mState = STOPPING_1; 4551 } 4552 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4553 } 4554 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4555 thread->mLock.unlock(); 4556 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4557 thread->mLock.lock(); 4558 4559#ifdef ADD_BATTERY_DATA 4560 // to track the speaker usage 4561 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4562#endif 4563 } 4564 } 4565} 4566 4567void AudioFlinger::PlaybackThread::Track::pause() 4568{ 4569 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4570 sp<ThreadBase> thread = mThread.promote(); 4571 if (thread != 0) { 4572 Mutex::Autolock _l(thread->mLock); 4573 if (mState == ACTIVE || mState == RESUMING) { 4574 mState = PAUSING; 4575 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4576 if (!isOutputTrack()) { 4577 thread->mLock.unlock(); 4578 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4579 thread->mLock.lock(); 4580 4581#ifdef ADD_BATTERY_DATA 4582 // to track the speaker usage 4583 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4584#endif 4585 } 4586 } 4587 } 4588} 4589 4590void AudioFlinger::PlaybackThread::Track::flush() 4591{ 4592 ALOGV("flush(%d)", mName); 4593 sp<ThreadBase> thread = mThread.promote(); 4594 if (thread != 0) { 4595 Mutex::Autolock _l(thread->mLock); 4596 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4597 mState != PAUSING) { 4598 return; 4599 } 4600 // No point remaining in PAUSED state after a flush => go to 4601 // FLUSHED state 4602 mState = FLUSHED; 4603 // do not reset the track if it is still in the process of being stopped or paused. 4604 // this will be done by prepareTracks_l() when the track is stopped. 4605 // prepareTracks_l() will see mState == FLUSHED, then 4606 // remove from active track list, reset(), and trigger presentation complete 4607 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4608 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4609 reset(); 4610 } 4611 } 4612} 4613 4614void AudioFlinger::PlaybackThread::Track::reset() 4615{ 4616 // Do not reset twice to avoid discarding data written just after a flush and before 4617 // the audioflinger thread detects the track is stopped. 4618 if (!mResetDone) { 4619 TrackBase::reset(); 4620 // Force underrun condition to avoid false underrun callback until first data is 4621 // written to buffer 4622 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4623 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4624 mFillingUpStatus = FS_FILLING; 4625 mResetDone = true; 4626 if (mState == FLUSHED) { 4627 mState = IDLE; 4628 } 4629 } 4630} 4631 4632void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4633{ 4634 mMute = muted; 4635} 4636 4637status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4638{ 4639 status_t status = DEAD_OBJECT; 4640 sp<ThreadBase> thread = mThread.promote(); 4641 if (thread != 0) { 4642 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4643 status = playbackThread->attachAuxEffect(this, EffectId); 4644 } 4645 return status; 4646} 4647 4648void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4649{ 4650 mAuxEffectId = EffectId; 4651 mAuxBuffer = buffer; 4652} 4653 4654bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4655 size_t audioHalFrames) 4656{ 4657 // a track is considered presented when the total number of frames written to audio HAL 4658 // corresponds to the number of frames written when presentationComplete() is called for the 4659 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4660 if (mPresentationCompleteFrames == 0) { 4661 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4662 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4663 mPresentationCompleteFrames, audioHalFrames); 4664 } 4665 if (framesWritten >= mPresentationCompleteFrames) { 4666 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4667 mSessionId, framesWritten); 4668 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4669 return true; 4670 } 4671 return false; 4672} 4673 4674void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4675{ 4676 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4677 if (mSyncEvents[i]->type() == type) { 4678 mSyncEvents[i]->trigger(); 4679 mSyncEvents.removeAt(i); 4680 i--; 4681 } 4682 } 4683} 4684 4685// implement VolumeBufferProvider interface 4686 4687uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4688{ 4689 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4690 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4691 uint32_t vlr = mCblk->getVolumeLR(); 4692 uint32_t vl = vlr & 0xFFFF; 4693 uint32_t vr = vlr >> 16; 4694 // track volumes come from shared memory, so can't be trusted and must be clamped 4695 if (vl > MAX_GAIN_INT) { 4696 vl = MAX_GAIN_INT; 4697 } 4698 if (vr > MAX_GAIN_INT) { 4699 vr = MAX_GAIN_INT; 4700 } 4701 // now apply the cached master volume and stream type volume; 4702 // this is trusted but lacks any synchronization or barrier so may be stale 4703 float v = mCachedVolume; 4704 vl *= v; 4705 vr *= v; 4706 // re-combine into U4.16 4707 vlr = (vr << 16) | (vl & 0xFFFF); 4708 // FIXME look at mute, pause, and stop flags 4709 return vlr; 4710} 4711 4712status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4713{ 4714 if (mState == TERMINATED || mState == PAUSED || 4715 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4716 (mState == STOPPED)))) { 4717 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4718 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4719 event->cancel(); 4720 return INVALID_OPERATION; 4721 } 4722 TrackBase::setSyncEvent(event); 4723 return NO_ERROR; 4724} 4725 4726// timed audio tracks 4727 4728sp<AudioFlinger::PlaybackThread::TimedTrack> 4729AudioFlinger::PlaybackThread::TimedTrack::create( 4730 PlaybackThread *thread, 4731 const sp<Client>& client, 4732 audio_stream_type_t streamType, 4733 uint32_t sampleRate, 4734 audio_format_t format, 4735 uint32_t channelMask, 4736 int frameCount, 4737 const sp<IMemory>& sharedBuffer, 4738 int sessionId) { 4739 if (!client->reserveTimedTrack()) 4740 return NULL; 4741 4742 return new TimedTrack( 4743 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4744 sharedBuffer, sessionId); 4745} 4746 4747AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4748 PlaybackThread *thread, 4749 const sp<Client>& client, 4750 audio_stream_type_t streamType, 4751 uint32_t sampleRate, 4752 audio_format_t format, 4753 uint32_t channelMask, 4754 int frameCount, 4755 const sp<IMemory>& sharedBuffer, 4756 int sessionId) 4757 : Track(thread, client, streamType, sampleRate, format, channelMask, 4758 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4759 mQueueHeadInFlight(false), 4760 mTrimQueueHeadOnRelease(false), 4761 mFramesPendingInQueue(0), 4762 mTimedSilenceBuffer(NULL), 4763 mTimedSilenceBufferSize(0), 4764 mTimedAudioOutputOnTime(false), 4765 mMediaTimeTransformValid(false) 4766{ 4767 LocalClock lc; 4768 mLocalTimeFreq = lc.getLocalFreq(); 4769 4770 mLocalTimeToSampleTransform.a_zero = 0; 4771 mLocalTimeToSampleTransform.b_zero = 0; 4772 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4773 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4774 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4775 &mLocalTimeToSampleTransform.a_to_b_denom); 4776 4777 mMediaTimeToSampleTransform.a_zero = 0; 4778 mMediaTimeToSampleTransform.b_zero = 0; 4779 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4780 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4781 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4782 &mMediaTimeToSampleTransform.a_to_b_denom); 4783} 4784 4785AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4786 mClient->releaseTimedTrack(); 4787 delete [] mTimedSilenceBuffer; 4788} 4789 4790status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4791 size_t size, sp<IMemory>* buffer) { 4792 4793 Mutex::Autolock _l(mTimedBufferQueueLock); 4794 4795 trimTimedBufferQueue_l(); 4796 4797 // lazily initialize the shared memory heap for timed buffers 4798 if (mTimedMemoryDealer == NULL) { 4799 const int kTimedBufferHeapSize = 512 << 10; 4800 4801 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4802 "AudioFlingerTimed"); 4803 if (mTimedMemoryDealer == NULL) 4804 return NO_MEMORY; 4805 } 4806 4807 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4808 if (newBuffer == NULL) { 4809 newBuffer = mTimedMemoryDealer->allocate(size); 4810 if (newBuffer == NULL) 4811 return NO_MEMORY; 4812 } 4813 4814 *buffer = newBuffer; 4815 return NO_ERROR; 4816} 4817 4818// caller must hold mTimedBufferQueueLock 4819void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4820 int64_t mediaTimeNow; 4821 { 4822 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4823 if (!mMediaTimeTransformValid) 4824 return; 4825 4826 int64_t targetTimeNow; 4827 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4828 ? mCCHelper.getCommonTime(&targetTimeNow) 4829 : mCCHelper.getLocalTime(&targetTimeNow); 4830 4831 if (OK != res) 4832 return; 4833 4834 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4835 &mediaTimeNow)) { 4836 return; 4837 } 4838 } 4839 4840 size_t trimEnd; 4841 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4842 int64_t bufEnd; 4843 4844 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4845 // We have a next buffer. Just use its PTS as the PTS of the frame 4846 // following the last frame in this buffer. If the stream is sparse 4847 // (ie, there are deliberate gaps left in the stream which should be 4848 // filled with silence by the TimedAudioTrack), then this can result 4849 // in one extra buffer being left un-trimmed when it could have 4850 // been. In general, this is not typical, and we would rather 4851 // optimized away the TS calculation below for the more common case 4852 // where PTSes are contiguous. 4853 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4854 } else { 4855 // We have no next buffer. Compute the PTS of the frame following 4856 // the last frame in this buffer by computing the duration of of 4857 // this frame in media time units and adding it to the PTS of the 4858 // buffer. 4859 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4860 / mCblk->frameSize; 4861 4862 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4863 &bufEnd)) { 4864 ALOGE("Failed to convert frame count of %lld to media time" 4865 " duration" " (scale factor %d/%u) in %s", 4866 frameCount, 4867 mMediaTimeToSampleTransform.a_to_b_numer, 4868 mMediaTimeToSampleTransform.a_to_b_denom, 4869 __PRETTY_FUNCTION__); 4870 break; 4871 } 4872 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4873 } 4874 4875 if (bufEnd > mediaTimeNow) 4876 break; 4877 4878 // Is the buffer we want to use in the middle of a mix operation right 4879 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4880 // from the mixer which should be coming back shortly. 4881 if (!trimEnd && mQueueHeadInFlight) { 4882 mTrimQueueHeadOnRelease = true; 4883 } 4884 } 4885 4886 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4887 if (trimStart < trimEnd) { 4888 // Update the bookkeeping for framesReady() 4889 for (size_t i = trimStart; i < trimEnd; ++i) { 4890 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4891 } 4892 4893 // Now actually remove the buffers from the queue. 4894 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4895 } 4896} 4897 4898void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4899 const char* logTag) { 4900 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4901 "%s called (reason \"%s\"), but timed buffer queue has no" 4902 " elements to trim.", __FUNCTION__, logTag); 4903 4904 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4905 mTimedBufferQueue.removeAt(0); 4906} 4907 4908void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4909 const TimedBuffer& buf, 4910 const char* logTag) { 4911 uint32_t bufBytes = buf.buffer()->size(); 4912 uint32_t consumedAlready = buf.position(); 4913 4914 ALOG_ASSERT(consumedAlready <= bufBytes, 4915 "Bad bookkeeping while updating frames pending. Timed buffer is" 4916 " only %u bytes long, but claims to have consumed %u" 4917 " bytes. (update reason: \"%s\")", 4918 bufBytes, consumedAlready, logTag); 4919 4920 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4921 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4922 "Bad bookkeeping while updating frames pending. Should have at" 4923 " least %u queued frames, but we think we have only %u. (update" 4924 " reason: \"%s\")", 4925 bufFrames, mFramesPendingInQueue, logTag); 4926 4927 mFramesPendingInQueue -= bufFrames; 4928} 4929 4930status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4931 const sp<IMemory>& buffer, int64_t pts) { 4932 4933 { 4934 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4935 if (!mMediaTimeTransformValid) 4936 return INVALID_OPERATION; 4937 } 4938 4939 Mutex::Autolock _l(mTimedBufferQueueLock); 4940 4941 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4942 mFramesPendingInQueue += bufFrames; 4943 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4944 4945 return NO_ERROR; 4946} 4947 4948status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4949 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4950 4951 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4952 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4953 target); 4954 4955 if (!(target == TimedAudioTrack::LOCAL_TIME || 4956 target == TimedAudioTrack::COMMON_TIME)) { 4957 return BAD_VALUE; 4958 } 4959 4960 Mutex::Autolock lock(mMediaTimeTransformLock); 4961 mMediaTimeTransform = xform; 4962 mMediaTimeTransformTarget = target; 4963 mMediaTimeTransformValid = true; 4964 4965 return NO_ERROR; 4966} 4967 4968#define min(a, b) ((a) < (b) ? (a) : (b)) 4969 4970// implementation of getNextBuffer for tracks whose buffers have timestamps 4971status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4972 AudioBufferProvider::Buffer* buffer, int64_t pts) 4973{ 4974 if (pts == AudioBufferProvider::kInvalidPTS) { 4975 buffer->raw = 0; 4976 buffer->frameCount = 0; 4977 mTimedAudioOutputOnTime = false; 4978 return INVALID_OPERATION; 4979 } 4980 4981 Mutex::Autolock _l(mTimedBufferQueueLock); 4982 4983 ALOG_ASSERT(!mQueueHeadInFlight, 4984 "getNextBuffer called without releaseBuffer!"); 4985 4986 while (true) { 4987 4988 // if we have no timed buffers, then fail 4989 if (mTimedBufferQueue.isEmpty()) { 4990 buffer->raw = 0; 4991 buffer->frameCount = 0; 4992 return NOT_ENOUGH_DATA; 4993 } 4994 4995 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4996 4997 // calculate the PTS of the head of the timed buffer queue expressed in 4998 // local time 4999 int64_t headLocalPTS; 5000 { 5001 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5002 5003 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5004 5005 if (mMediaTimeTransform.a_to_b_denom == 0) { 5006 // the transform represents a pause, so yield silence 5007 timedYieldSilence_l(buffer->frameCount, buffer); 5008 return NO_ERROR; 5009 } 5010 5011 int64_t transformedPTS; 5012 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5013 &transformedPTS)) { 5014 // the transform failed. this shouldn't happen, but if it does 5015 // then just drop this buffer 5016 ALOGW("timedGetNextBuffer transform failed"); 5017 buffer->raw = 0; 5018 buffer->frameCount = 0; 5019 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5020 return NO_ERROR; 5021 } 5022 5023 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5024 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5025 &headLocalPTS)) { 5026 buffer->raw = 0; 5027 buffer->frameCount = 0; 5028 return INVALID_OPERATION; 5029 } 5030 } else { 5031 headLocalPTS = transformedPTS; 5032 } 5033 } 5034 5035 // adjust the head buffer's PTS to reflect the portion of the head buffer 5036 // that has already been consumed 5037 int64_t effectivePTS = headLocalPTS + 5038 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5039 5040 // Calculate the delta in samples between the head of the input buffer 5041 // queue and the start of the next output buffer that will be written. 5042 // If the transformation fails because of over or underflow, it means 5043 // that the sample's position in the output stream is so far out of 5044 // whack that it should just be dropped. 5045 int64_t sampleDelta; 5046 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5047 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5048 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5049 " mix"); 5050 continue; 5051 } 5052 if (!mLocalTimeToSampleTransform.doForwardTransform( 5053 (effectivePTS - pts) << 32, &sampleDelta)) { 5054 ALOGV("*** too late during sample rate transform: dropped buffer"); 5055 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5056 continue; 5057 } 5058 5059 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5060 " sampleDelta=[%d.%08x]", 5061 head.pts(), head.position(), pts, 5062 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5063 + (sampleDelta >> 32)), 5064 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5065 5066 // if the delta between the ideal placement for the next input sample and 5067 // the current output position is within this threshold, then we will 5068 // concatenate the next input samples to the previous output 5069 const int64_t kSampleContinuityThreshold = 5070 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5071 5072 // if this is the first buffer of audio that we're emitting from this track 5073 // then it should be almost exactly on time. 5074 const int64_t kSampleStartupThreshold = 1LL << 32; 5075 5076 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5077 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5078 // the next input is close enough to being on time, so concatenate it 5079 // with the last output 5080 timedYieldSamples_l(buffer); 5081 5082 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5083 head.position(), buffer->frameCount); 5084 return NO_ERROR; 5085 } 5086 5087 // Looks like our output is not on time. Reset our on timed status. 5088 // Next time we mix samples from our input queue, then should be within 5089 // the StartupThreshold. 5090 mTimedAudioOutputOnTime = false; 5091 if (sampleDelta > 0) { 5092 // the gap between the current output position and the proper start of 5093 // the next input sample is too big, so fill it with silence 5094 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5095 5096 timedYieldSilence_l(framesUntilNextInput, buffer); 5097 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5098 return NO_ERROR; 5099 } else { 5100 // the next input sample is late 5101 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5102 size_t onTimeSamplePosition = 5103 head.position() + lateFrames * mCblk->frameSize; 5104 5105 if (onTimeSamplePosition > head.buffer()->size()) { 5106 // all the remaining samples in the head are too late, so 5107 // drop it and move on 5108 ALOGV("*** too late: dropped buffer"); 5109 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5110 continue; 5111 } else { 5112 // skip over the late samples 5113 head.setPosition(onTimeSamplePosition); 5114 5115 // yield the available samples 5116 timedYieldSamples_l(buffer); 5117 5118 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5119 return NO_ERROR; 5120 } 5121 } 5122 } 5123} 5124 5125// Yield samples from the timed buffer queue head up to the given output 5126// buffer's capacity. 5127// 5128// Caller must hold mTimedBufferQueueLock 5129void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5130 AudioBufferProvider::Buffer* buffer) { 5131 5132 const TimedBuffer& head = mTimedBufferQueue[0]; 5133 5134 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5135 head.position()); 5136 5137 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5138 mCblk->frameSize); 5139 size_t framesRequested = buffer->frameCount; 5140 buffer->frameCount = min(framesLeftInHead, framesRequested); 5141 5142 mQueueHeadInFlight = true; 5143 mTimedAudioOutputOnTime = true; 5144} 5145 5146// Yield samples of silence up to the given output buffer's capacity 5147// 5148// Caller must hold mTimedBufferQueueLock 5149void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5150 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5151 5152 // lazily allocate a buffer filled with silence 5153 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5154 delete [] mTimedSilenceBuffer; 5155 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5156 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5157 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5158 } 5159 5160 buffer->raw = mTimedSilenceBuffer; 5161 size_t framesRequested = buffer->frameCount; 5162 buffer->frameCount = min(numFrames, framesRequested); 5163 5164 mTimedAudioOutputOnTime = false; 5165} 5166 5167// AudioBufferProvider interface 5168void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5169 AudioBufferProvider::Buffer* buffer) { 5170 5171 Mutex::Autolock _l(mTimedBufferQueueLock); 5172 5173 // If the buffer which was just released is part of the buffer at the head 5174 // of the queue, be sure to update the amt of the buffer which has been 5175 // consumed. If the buffer being returned is not part of the head of the 5176 // queue, its either because the buffer is part of the silence buffer, or 5177 // because the head of the timed queue was trimmed after the mixer called 5178 // getNextBuffer but before the mixer called releaseBuffer. 5179 if (buffer->raw == mTimedSilenceBuffer) { 5180 ALOG_ASSERT(!mQueueHeadInFlight, 5181 "Queue head in flight during release of silence buffer!"); 5182 goto done; 5183 } 5184 5185 ALOG_ASSERT(mQueueHeadInFlight, 5186 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5187 " head in flight."); 5188 5189 if (mTimedBufferQueue.size()) { 5190 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5191 5192 void* start = head.buffer()->pointer(); 5193 void* end = reinterpret_cast<void*>( 5194 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5195 + head.buffer()->size()); 5196 5197 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5198 "released buffer not within the head of the timed buffer" 5199 " queue; qHead = [%p, %p], released buffer = %p", 5200 start, end, buffer->raw); 5201 5202 head.setPosition(head.position() + 5203 (buffer->frameCount * mCblk->frameSize)); 5204 mQueueHeadInFlight = false; 5205 5206 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5207 "Bad bookkeeping during releaseBuffer! Should have at" 5208 " least %u queued frames, but we think we have only %u", 5209 buffer->frameCount, mFramesPendingInQueue); 5210 5211 mFramesPendingInQueue -= buffer->frameCount; 5212 5213 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5214 || mTrimQueueHeadOnRelease) { 5215 trimTimedBufferQueueHead_l("releaseBuffer"); 5216 mTrimQueueHeadOnRelease = false; 5217 } 5218 } else { 5219 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5220 " buffers in the timed buffer queue"); 5221 } 5222 5223done: 5224 buffer->raw = 0; 5225 buffer->frameCount = 0; 5226} 5227 5228size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5229 Mutex::Autolock _l(mTimedBufferQueueLock); 5230 return mFramesPendingInQueue; 5231} 5232 5233AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5234 : mPTS(0), mPosition(0) {} 5235 5236AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5237 const sp<IMemory>& buffer, int64_t pts) 5238 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5239 5240// ---------------------------------------------------------------------------- 5241 5242// RecordTrack constructor must be called with AudioFlinger::mLock held 5243AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5244 RecordThread *thread, 5245 const sp<Client>& client, 5246 uint32_t sampleRate, 5247 audio_format_t format, 5248 uint32_t channelMask, 5249 int frameCount, 5250 int sessionId) 5251 : TrackBase(thread, client, sampleRate, format, 5252 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5253 mOverflow(false) 5254{ 5255 if (mCblk != NULL) { 5256 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5257 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5258 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5259 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5260 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5261 } else { 5262 mCblk->frameSize = sizeof(int8_t); 5263 } 5264 } 5265} 5266 5267AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5268{ 5269 sp<ThreadBase> thread = mThread.promote(); 5270 if (thread != 0) { 5271 AudioSystem::releaseInput(thread->id()); 5272 } 5273} 5274 5275// AudioBufferProvider interface 5276status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5277{ 5278 audio_track_cblk_t* cblk = this->cblk(); 5279 uint32_t framesAvail; 5280 uint32_t framesReq = buffer->frameCount; 5281 5282 // Check if last stepServer failed, try to step now 5283 if (mStepServerFailed) { 5284 if (!step()) goto getNextBuffer_exit; 5285 ALOGV("stepServer recovered"); 5286 mStepServerFailed = false; 5287 } 5288 5289 framesAvail = cblk->framesAvailable_l(); 5290 5291 if (CC_LIKELY(framesAvail)) { 5292 uint32_t s = cblk->server; 5293 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5294 5295 if (framesReq > framesAvail) { 5296 framesReq = framesAvail; 5297 } 5298 if (framesReq > bufferEnd - s) { 5299 framesReq = bufferEnd - s; 5300 } 5301 5302 buffer->raw = getBuffer(s, framesReq); 5303 if (buffer->raw == NULL) goto getNextBuffer_exit; 5304 5305 buffer->frameCount = framesReq; 5306 return NO_ERROR; 5307 } 5308 5309getNextBuffer_exit: 5310 buffer->raw = NULL; 5311 buffer->frameCount = 0; 5312 return NOT_ENOUGH_DATA; 5313} 5314 5315status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5316 int triggerSession) 5317{ 5318 sp<ThreadBase> thread = mThread.promote(); 5319 if (thread != 0) { 5320 RecordThread *recordThread = (RecordThread *)thread.get(); 5321 return recordThread->start(this, event, triggerSession); 5322 } else { 5323 return BAD_VALUE; 5324 } 5325} 5326 5327void AudioFlinger::RecordThread::RecordTrack::stop() 5328{ 5329 sp<ThreadBase> thread = mThread.promote(); 5330 if (thread != 0) { 5331 RecordThread *recordThread = (RecordThread *)thread.get(); 5332 recordThread->stop(this); 5333 TrackBase::reset(); 5334 // Force overrun condition to avoid false overrun callback until first data is 5335 // read from buffer 5336 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5337 } 5338} 5339 5340void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5341{ 5342 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5343 (mClient == 0) ? getpid_cached : mClient->pid(), 5344 mFormat, 5345 mChannelMask, 5346 mSessionId, 5347 mFrameCount, 5348 mState, 5349 mCblk->sampleRate, 5350 mCblk->server, 5351 mCblk->user); 5352} 5353 5354 5355// ---------------------------------------------------------------------------- 5356 5357AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5358 PlaybackThread *playbackThread, 5359 DuplicatingThread *sourceThread, 5360 uint32_t sampleRate, 5361 audio_format_t format, 5362 uint32_t channelMask, 5363 int frameCount) 5364 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5365 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5366 mActive(false), mSourceThread(sourceThread) 5367{ 5368 5369 if (mCblk != NULL) { 5370 mCblk->flags |= CBLK_DIRECTION_OUT; 5371 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5372 mOutBuffer.frameCount = 0; 5373 playbackThread->mTracks.add(this); 5374 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5375 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5376 mCblk, mBuffer, mCblk->buffers, 5377 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5378 } else { 5379 ALOGW("Error creating output track on thread %p", playbackThread); 5380 } 5381} 5382 5383AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5384{ 5385 clearBufferQueue(); 5386} 5387 5388status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5389 int triggerSession) 5390{ 5391 status_t status = Track::start(event, triggerSession); 5392 if (status != NO_ERROR) { 5393 return status; 5394 } 5395 5396 mActive = true; 5397 mRetryCount = 127; 5398 return status; 5399} 5400 5401void AudioFlinger::PlaybackThread::OutputTrack::stop() 5402{ 5403 Track::stop(); 5404 clearBufferQueue(); 5405 mOutBuffer.frameCount = 0; 5406 mActive = false; 5407} 5408 5409bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5410{ 5411 Buffer *pInBuffer; 5412 Buffer inBuffer; 5413 uint32_t channelCount = mChannelCount; 5414 bool outputBufferFull = false; 5415 inBuffer.frameCount = frames; 5416 inBuffer.i16 = data; 5417 5418 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5419 5420 if (!mActive && frames != 0) { 5421 start(); 5422 sp<ThreadBase> thread = mThread.promote(); 5423 if (thread != 0) { 5424 MixerThread *mixerThread = (MixerThread *)thread.get(); 5425 if (mCblk->frameCount > frames){ 5426 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5427 uint32_t startFrames = (mCblk->frameCount - frames); 5428 pInBuffer = new Buffer; 5429 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5430 pInBuffer->frameCount = startFrames; 5431 pInBuffer->i16 = pInBuffer->mBuffer; 5432 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5433 mBufferQueue.add(pInBuffer); 5434 } else { 5435 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5436 } 5437 } 5438 } 5439 } 5440 5441 while (waitTimeLeftMs) { 5442 // First write pending buffers, then new data 5443 if (mBufferQueue.size()) { 5444 pInBuffer = mBufferQueue.itemAt(0); 5445 } else { 5446 pInBuffer = &inBuffer; 5447 } 5448 5449 if (pInBuffer->frameCount == 0) { 5450 break; 5451 } 5452 5453 if (mOutBuffer.frameCount == 0) { 5454 mOutBuffer.frameCount = pInBuffer->frameCount; 5455 nsecs_t startTime = systemTime(); 5456 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5457 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5458 outputBufferFull = true; 5459 break; 5460 } 5461 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5462 if (waitTimeLeftMs >= waitTimeMs) { 5463 waitTimeLeftMs -= waitTimeMs; 5464 } else { 5465 waitTimeLeftMs = 0; 5466 } 5467 } 5468 5469 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5470 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5471 mCblk->stepUser(outFrames); 5472 pInBuffer->frameCount -= outFrames; 5473 pInBuffer->i16 += outFrames * channelCount; 5474 mOutBuffer.frameCount -= outFrames; 5475 mOutBuffer.i16 += outFrames * channelCount; 5476 5477 if (pInBuffer->frameCount == 0) { 5478 if (mBufferQueue.size()) { 5479 mBufferQueue.removeAt(0); 5480 delete [] pInBuffer->mBuffer; 5481 delete pInBuffer; 5482 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5483 } else { 5484 break; 5485 } 5486 } 5487 } 5488 5489 // If we could not write all frames, allocate a buffer and queue it for next time. 5490 if (inBuffer.frameCount) { 5491 sp<ThreadBase> thread = mThread.promote(); 5492 if (thread != 0 && !thread->standby()) { 5493 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5494 pInBuffer = new Buffer; 5495 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5496 pInBuffer->frameCount = inBuffer.frameCount; 5497 pInBuffer->i16 = pInBuffer->mBuffer; 5498 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5499 mBufferQueue.add(pInBuffer); 5500 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5501 } else { 5502 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5503 } 5504 } 5505 } 5506 5507 // Calling write() with a 0 length buffer, means that no more data will be written: 5508 // If no more buffers are pending, fill output track buffer to make sure it is started 5509 // by output mixer. 5510 if (frames == 0 && mBufferQueue.size() == 0) { 5511 if (mCblk->user < mCblk->frameCount) { 5512 frames = mCblk->frameCount - mCblk->user; 5513 pInBuffer = new Buffer; 5514 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5515 pInBuffer->frameCount = frames; 5516 pInBuffer->i16 = pInBuffer->mBuffer; 5517 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5518 mBufferQueue.add(pInBuffer); 5519 } else if (mActive) { 5520 stop(); 5521 } 5522 } 5523 5524 return outputBufferFull; 5525} 5526 5527status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5528{ 5529 int active; 5530 status_t result; 5531 audio_track_cblk_t* cblk = mCblk; 5532 uint32_t framesReq = buffer->frameCount; 5533 5534// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5535 buffer->frameCount = 0; 5536 5537 uint32_t framesAvail = cblk->framesAvailable(); 5538 5539 5540 if (framesAvail == 0) { 5541 Mutex::Autolock _l(cblk->lock); 5542 goto start_loop_here; 5543 while (framesAvail == 0) { 5544 active = mActive; 5545 if (CC_UNLIKELY(!active)) { 5546 ALOGV("Not active and NO_MORE_BUFFERS"); 5547 return NO_MORE_BUFFERS; 5548 } 5549 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5550 if (result != NO_ERROR) { 5551 return NO_MORE_BUFFERS; 5552 } 5553 // read the server count again 5554 start_loop_here: 5555 framesAvail = cblk->framesAvailable_l(); 5556 } 5557 } 5558 5559// if (framesAvail < framesReq) { 5560// return NO_MORE_BUFFERS; 5561// } 5562 5563 if (framesReq > framesAvail) { 5564 framesReq = framesAvail; 5565 } 5566 5567 uint32_t u = cblk->user; 5568 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5569 5570 if (framesReq > bufferEnd - u) { 5571 framesReq = bufferEnd - u; 5572 } 5573 5574 buffer->frameCount = framesReq; 5575 buffer->raw = (void *)cblk->buffer(u); 5576 return NO_ERROR; 5577} 5578 5579 5580void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5581{ 5582 size_t size = mBufferQueue.size(); 5583 5584 for (size_t i = 0; i < size; i++) { 5585 Buffer *pBuffer = mBufferQueue.itemAt(i); 5586 delete [] pBuffer->mBuffer; 5587 delete pBuffer; 5588 } 5589 mBufferQueue.clear(); 5590} 5591 5592// ---------------------------------------------------------------------------- 5593 5594AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5595 : RefBase(), 5596 mAudioFlinger(audioFlinger), 5597 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5598 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5599 mPid(pid), 5600 mTimedTrackCount(0) 5601{ 5602 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5603} 5604 5605// Client destructor must be called with AudioFlinger::mLock held 5606AudioFlinger::Client::~Client() 5607{ 5608 mAudioFlinger->removeClient_l(mPid); 5609} 5610 5611sp<MemoryDealer> AudioFlinger::Client::heap() const 5612{ 5613 return mMemoryDealer; 5614} 5615 5616// Reserve one of the limited slots for a timed audio track associated 5617// with this client 5618bool AudioFlinger::Client::reserveTimedTrack() 5619{ 5620 const int kMaxTimedTracksPerClient = 4; 5621 5622 Mutex::Autolock _l(mTimedTrackLock); 5623 5624 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5625 ALOGW("can not create timed track - pid %d has exceeded the limit", 5626 mPid); 5627 return false; 5628 } 5629 5630 mTimedTrackCount++; 5631 return true; 5632} 5633 5634// Release a slot for a timed audio track 5635void AudioFlinger::Client::releaseTimedTrack() 5636{ 5637 Mutex::Autolock _l(mTimedTrackLock); 5638 mTimedTrackCount--; 5639} 5640 5641// ---------------------------------------------------------------------------- 5642 5643AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5644 const sp<IAudioFlingerClient>& client, 5645 pid_t pid) 5646 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5647{ 5648} 5649 5650AudioFlinger::NotificationClient::~NotificationClient() 5651{ 5652} 5653 5654void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5655{ 5656 sp<NotificationClient> keep(this); 5657 mAudioFlinger->removeNotificationClient(mPid); 5658} 5659 5660// ---------------------------------------------------------------------------- 5661 5662AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5663 : BnAudioTrack(), 5664 mTrack(track) 5665{ 5666} 5667 5668AudioFlinger::TrackHandle::~TrackHandle() { 5669 // just stop the track on deletion, associated resources 5670 // will be freed from the main thread once all pending buffers have 5671 // been played. Unless it's not in the active track list, in which 5672 // case we free everything now... 5673 mTrack->destroy(); 5674} 5675 5676sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5677 return mTrack->getCblk(); 5678} 5679 5680status_t AudioFlinger::TrackHandle::start() { 5681 return mTrack->start(); 5682} 5683 5684void AudioFlinger::TrackHandle::stop() { 5685 mTrack->stop(); 5686} 5687 5688void AudioFlinger::TrackHandle::flush() { 5689 mTrack->flush(); 5690} 5691 5692void AudioFlinger::TrackHandle::mute(bool e) { 5693 mTrack->mute(e); 5694} 5695 5696void AudioFlinger::TrackHandle::pause() { 5697 mTrack->pause(); 5698} 5699 5700status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5701{ 5702 return mTrack->attachAuxEffect(EffectId); 5703} 5704 5705status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5706 sp<IMemory>* buffer) { 5707 if (!mTrack->isTimedTrack()) 5708 return INVALID_OPERATION; 5709 5710 PlaybackThread::TimedTrack* tt = 5711 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5712 return tt->allocateTimedBuffer(size, buffer); 5713} 5714 5715status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5716 int64_t pts) { 5717 if (!mTrack->isTimedTrack()) 5718 return INVALID_OPERATION; 5719 5720 PlaybackThread::TimedTrack* tt = 5721 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5722 return tt->queueTimedBuffer(buffer, pts); 5723} 5724 5725status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5726 const LinearTransform& xform, int target) { 5727 5728 if (!mTrack->isTimedTrack()) 5729 return INVALID_OPERATION; 5730 5731 PlaybackThread::TimedTrack* tt = 5732 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5733 return tt->setMediaTimeTransform( 5734 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5735} 5736 5737status_t AudioFlinger::TrackHandle::onTransact( 5738 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5739{ 5740 return BnAudioTrack::onTransact(code, data, reply, flags); 5741} 5742 5743// ---------------------------------------------------------------------------- 5744 5745sp<IAudioRecord> AudioFlinger::openRecord( 5746 pid_t pid, 5747 audio_io_handle_t input, 5748 uint32_t sampleRate, 5749 audio_format_t format, 5750 uint32_t channelMask, 5751 int frameCount, 5752 IAudioFlinger::track_flags_t flags, 5753 int *sessionId, 5754 status_t *status) 5755{ 5756 sp<RecordThread::RecordTrack> recordTrack; 5757 sp<RecordHandle> recordHandle; 5758 sp<Client> client; 5759 status_t lStatus; 5760 RecordThread *thread; 5761 size_t inFrameCount; 5762 int lSessionId; 5763 5764 // check calling permissions 5765 if (!recordingAllowed()) { 5766 lStatus = PERMISSION_DENIED; 5767 goto Exit; 5768 } 5769 5770 // add client to list 5771 { // scope for mLock 5772 Mutex::Autolock _l(mLock); 5773 thread = checkRecordThread_l(input); 5774 if (thread == NULL) { 5775 lStatus = BAD_VALUE; 5776 goto Exit; 5777 } 5778 5779 client = registerPid_l(pid); 5780 5781 // If no audio session id is provided, create one here 5782 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5783 lSessionId = *sessionId; 5784 } else { 5785 lSessionId = nextUniqueId(); 5786 if (sessionId != NULL) { 5787 *sessionId = lSessionId; 5788 } 5789 } 5790 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5791 recordTrack = thread->createRecordTrack_l(client, 5792 sampleRate, 5793 format, 5794 channelMask, 5795 frameCount, 5796 lSessionId, 5797 &lStatus); 5798 } 5799 if (lStatus != NO_ERROR) { 5800 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5801 // destructor is called by the TrackBase destructor with mLock held 5802 client.clear(); 5803 recordTrack.clear(); 5804 goto Exit; 5805 } 5806 5807 // return to handle to client 5808 recordHandle = new RecordHandle(recordTrack); 5809 lStatus = NO_ERROR; 5810 5811Exit: 5812 if (status) { 5813 *status = lStatus; 5814 } 5815 return recordHandle; 5816} 5817 5818// ---------------------------------------------------------------------------- 5819 5820AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5821 : BnAudioRecord(), 5822 mRecordTrack(recordTrack) 5823{ 5824} 5825 5826AudioFlinger::RecordHandle::~RecordHandle() { 5827 stop(); 5828} 5829 5830sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5831 return mRecordTrack->getCblk(); 5832} 5833 5834status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5835 ALOGV("RecordHandle::start()"); 5836 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5837} 5838 5839void AudioFlinger::RecordHandle::stop() { 5840 ALOGV("RecordHandle::stop()"); 5841 mRecordTrack->stop(); 5842} 5843 5844status_t AudioFlinger::RecordHandle::onTransact( 5845 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5846{ 5847 return BnAudioRecord::onTransact(code, data, reply, flags); 5848} 5849 5850// ---------------------------------------------------------------------------- 5851 5852AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5853 AudioStreamIn *input, 5854 uint32_t sampleRate, 5855 uint32_t channels, 5856 audio_io_handle_t id, 5857 uint32_t device) : 5858 ThreadBase(audioFlinger, id, device, RECORD), 5859 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5860 // mRsmpInIndex and mInputBytes set by readInputParameters() 5861 mReqChannelCount(popcount(channels)), 5862 mReqSampleRate(sampleRate) 5863 // mBytesRead is only meaningful while active, and so is cleared in start() 5864 // (but might be better to also clear here for dump?) 5865{ 5866 snprintf(mName, kNameLength, "AudioIn_%X", id); 5867 5868 readInputParameters(); 5869} 5870 5871 5872AudioFlinger::RecordThread::~RecordThread() 5873{ 5874 delete[] mRsmpInBuffer; 5875 delete mResampler; 5876 delete[] mRsmpOutBuffer; 5877} 5878 5879void AudioFlinger::RecordThread::onFirstRef() 5880{ 5881 run(mName, PRIORITY_URGENT_AUDIO); 5882} 5883 5884status_t AudioFlinger::RecordThread::readyToRun() 5885{ 5886 status_t status = initCheck(); 5887 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5888 return status; 5889} 5890 5891bool AudioFlinger::RecordThread::threadLoop() 5892{ 5893 AudioBufferProvider::Buffer buffer; 5894 sp<RecordTrack> activeTrack; 5895 Vector< sp<EffectChain> > effectChains; 5896 5897 nsecs_t lastWarning = 0; 5898 5899 acquireWakeLock(); 5900 5901 // start recording 5902 while (!exitPending()) { 5903 5904 processConfigEvents(); 5905 5906 { // scope for mLock 5907 Mutex::Autolock _l(mLock); 5908 checkForNewParameters_l(); 5909 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5910 if (!mStandby) { 5911 mInput->stream->common.standby(&mInput->stream->common); 5912 mStandby = true; 5913 } 5914 5915 if (exitPending()) break; 5916 5917 releaseWakeLock_l(); 5918 ALOGV("RecordThread: loop stopping"); 5919 // go to sleep 5920 mWaitWorkCV.wait(mLock); 5921 ALOGV("RecordThread: loop starting"); 5922 acquireWakeLock_l(); 5923 continue; 5924 } 5925 if (mActiveTrack != 0) { 5926 if (mActiveTrack->mState == TrackBase::PAUSING) { 5927 if (!mStandby) { 5928 mInput->stream->common.standby(&mInput->stream->common); 5929 mStandby = true; 5930 } 5931 mActiveTrack.clear(); 5932 mStartStopCond.broadcast(); 5933 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5934 if (mReqChannelCount != mActiveTrack->channelCount()) { 5935 mActiveTrack.clear(); 5936 mStartStopCond.broadcast(); 5937 } else if (mBytesRead != 0) { 5938 // record start succeeds only if first read from audio input 5939 // succeeds 5940 if (mBytesRead > 0) { 5941 mActiveTrack->mState = TrackBase::ACTIVE; 5942 } else { 5943 mActiveTrack.clear(); 5944 } 5945 mStartStopCond.broadcast(); 5946 } 5947 mStandby = false; 5948 } 5949 } 5950 lockEffectChains_l(effectChains); 5951 } 5952 5953 if (mActiveTrack != 0) { 5954 if (mActiveTrack->mState != TrackBase::ACTIVE && 5955 mActiveTrack->mState != TrackBase::RESUMING) { 5956 unlockEffectChains(effectChains); 5957 usleep(kRecordThreadSleepUs); 5958 continue; 5959 } 5960 for (size_t i = 0; i < effectChains.size(); i ++) { 5961 effectChains[i]->process_l(); 5962 } 5963 5964 buffer.frameCount = mFrameCount; 5965 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5966 size_t framesOut = buffer.frameCount; 5967 if (mResampler == NULL) { 5968 // no resampling 5969 while (framesOut) { 5970 size_t framesIn = mFrameCount - mRsmpInIndex; 5971 if (framesIn) { 5972 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5973 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5974 if (framesIn > framesOut) 5975 framesIn = framesOut; 5976 mRsmpInIndex += framesIn; 5977 framesOut -= framesIn; 5978 if ((int)mChannelCount == mReqChannelCount || 5979 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5980 memcpy(dst, src, framesIn * mFrameSize); 5981 } else { 5982 int16_t *src16 = (int16_t *)src; 5983 int16_t *dst16 = (int16_t *)dst; 5984 if (mChannelCount == 1) { 5985 while (framesIn--) { 5986 *dst16++ = *src16; 5987 *dst16++ = *src16++; 5988 } 5989 } else { 5990 while (framesIn--) { 5991 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5992 src16 += 2; 5993 } 5994 } 5995 } 5996 } 5997 if (framesOut && mFrameCount == mRsmpInIndex) { 5998 if (framesOut == mFrameCount && 5999 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6000 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6001 framesOut = 0; 6002 } else { 6003 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6004 mRsmpInIndex = 0; 6005 } 6006 if (mBytesRead < 0) { 6007 ALOGE("Error reading audio input"); 6008 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6009 // Force input into standby so that it tries to 6010 // recover at next read attempt 6011 mInput->stream->common.standby(&mInput->stream->common); 6012 usleep(kRecordThreadSleepUs); 6013 } 6014 mRsmpInIndex = mFrameCount; 6015 framesOut = 0; 6016 buffer.frameCount = 0; 6017 } 6018 } 6019 } 6020 } else { 6021 // resampling 6022 6023 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6024 // alter output frame count as if we were expecting stereo samples 6025 if (mChannelCount == 1 && mReqChannelCount == 1) { 6026 framesOut >>= 1; 6027 } 6028 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6029 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6030 // are 32 bit aligned which should be always true. 6031 if (mChannelCount == 2 && mReqChannelCount == 1) { 6032 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6033 // the resampler always outputs stereo samples: do post stereo to mono conversion 6034 int16_t *src = (int16_t *)mRsmpOutBuffer; 6035 int16_t *dst = buffer.i16; 6036 while (framesOut--) { 6037 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6038 src += 2; 6039 } 6040 } else { 6041 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6042 } 6043 6044 } 6045 if (mFramestoDrop == 0) { 6046 mActiveTrack->releaseBuffer(&buffer); 6047 } else { 6048 if (mFramestoDrop > 0) { 6049 mFramestoDrop -= buffer.frameCount; 6050 if (mFramestoDrop <= 0) { 6051 clearSyncStartEvent(); 6052 } 6053 } else { 6054 mFramestoDrop += buffer.frameCount; 6055 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6056 mSyncStartEvent->isCancelled()) { 6057 ALOGW("Synced record %s, session %d, trigger session %d", 6058 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6059 mActiveTrack->sessionId(), 6060 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6061 clearSyncStartEvent(); 6062 } 6063 } 6064 } 6065 mActiveTrack->overflow(); 6066 } 6067 // client isn't retrieving buffers fast enough 6068 else { 6069 if (!mActiveTrack->setOverflow()) { 6070 nsecs_t now = systemTime(); 6071 if ((now - lastWarning) > kWarningThrottleNs) { 6072 ALOGW("RecordThread: buffer overflow"); 6073 lastWarning = now; 6074 } 6075 } 6076 // Release the processor for a while before asking for a new buffer. 6077 // This will give the application more chance to read from the buffer and 6078 // clear the overflow. 6079 usleep(kRecordThreadSleepUs); 6080 } 6081 } 6082 // enable changes in effect chain 6083 unlockEffectChains(effectChains); 6084 effectChains.clear(); 6085 } 6086 6087 if (!mStandby) { 6088 mInput->stream->common.standby(&mInput->stream->common); 6089 } 6090 mActiveTrack.clear(); 6091 6092 mStartStopCond.broadcast(); 6093 6094 releaseWakeLock(); 6095 6096 ALOGV("RecordThread %p exiting", this); 6097 return false; 6098} 6099 6100 6101sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6102 const sp<AudioFlinger::Client>& client, 6103 uint32_t sampleRate, 6104 audio_format_t format, 6105 int channelMask, 6106 int frameCount, 6107 int sessionId, 6108 status_t *status) 6109{ 6110 sp<RecordTrack> track; 6111 status_t lStatus; 6112 6113 lStatus = initCheck(); 6114 if (lStatus != NO_ERROR) { 6115 ALOGE("Audio driver not initialized."); 6116 goto Exit; 6117 } 6118 6119 { // scope for mLock 6120 Mutex::Autolock _l(mLock); 6121 6122 track = new RecordTrack(this, client, sampleRate, 6123 format, channelMask, frameCount, sessionId); 6124 6125 if (track->getCblk() == 0) { 6126 lStatus = NO_MEMORY; 6127 goto Exit; 6128 } 6129 6130 mTrack = track.get(); 6131 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6132 bool suspend = audio_is_bluetooth_sco_device( 6133 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6134 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6135 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6136 } 6137 lStatus = NO_ERROR; 6138 6139Exit: 6140 if (status) { 6141 *status = lStatus; 6142 } 6143 return track; 6144} 6145 6146status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6147 AudioSystem::sync_event_t event, 6148 int triggerSession) 6149{ 6150 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6151 sp<ThreadBase> strongMe = this; 6152 status_t status = NO_ERROR; 6153 6154 if (event == AudioSystem::SYNC_EVENT_NONE) { 6155 clearSyncStartEvent(); 6156 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6157 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6158 triggerSession, 6159 recordTrack->sessionId(), 6160 syncStartEventCallback, 6161 this); 6162 // Sync event can be cancelled by the trigger session if the track is not in a 6163 // compatible state in which case we start record immediately 6164 if (mSyncStartEvent->isCancelled()) { 6165 clearSyncStartEvent(); 6166 } else { 6167 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6168 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6169 } 6170 } 6171 6172 { 6173 AutoMutex lock(mLock); 6174 if (mActiveTrack != 0) { 6175 if (recordTrack != mActiveTrack.get()) { 6176 status = -EBUSY; 6177 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6178 mActiveTrack->mState = TrackBase::ACTIVE; 6179 } 6180 return status; 6181 } 6182 6183 recordTrack->mState = TrackBase::IDLE; 6184 mActiveTrack = recordTrack; 6185 mLock.unlock(); 6186 status_t status = AudioSystem::startInput(mId); 6187 mLock.lock(); 6188 if (status != NO_ERROR) { 6189 mActiveTrack.clear(); 6190 clearSyncStartEvent(); 6191 return status; 6192 } 6193 mRsmpInIndex = mFrameCount; 6194 mBytesRead = 0; 6195 if (mResampler != NULL) { 6196 mResampler->reset(); 6197 } 6198 mActiveTrack->mState = TrackBase::RESUMING; 6199 // signal thread to start 6200 ALOGV("Signal record thread"); 6201 mWaitWorkCV.signal(); 6202 // do not wait for mStartStopCond if exiting 6203 if (exitPending()) { 6204 mActiveTrack.clear(); 6205 status = INVALID_OPERATION; 6206 goto startError; 6207 } 6208 mStartStopCond.wait(mLock); 6209 if (mActiveTrack == 0) { 6210 ALOGV("Record failed to start"); 6211 status = BAD_VALUE; 6212 goto startError; 6213 } 6214 ALOGV("Record started OK"); 6215 return status; 6216 } 6217startError: 6218 AudioSystem::stopInput(mId); 6219 clearSyncStartEvent(); 6220 return status; 6221} 6222 6223void AudioFlinger::RecordThread::clearSyncStartEvent() 6224{ 6225 if (mSyncStartEvent != 0) { 6226 mSyncStartEvent->cancel(); 6227 } 6228 mSyncStartEvent.clear(); 6229 mFramestoDrop = 0; 6230} 6231 6232void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6233{ 6234 sp<SyncEvent> strongEvent = event.promote(); 6235 6236 if (strongEvent != 0) { 6237 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6238 me->handleSyncStartEvent(strongEvent); 6239 } 6240} 6241 6242void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6243{ 6244 if (event == mSyncStartEvent) { 6245 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6246 // from audio HAL 6247 mFramestoDrop = mFrameCount * 2; 6248 } 6249} 6250 6251void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6252 ALOGV("RecordThread::stop"); 6253 sp<ThreadBase> strongMe = this; 6254 { 6255 AutoMutex lock(mLock); 6256 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6257 mActiveTrack->mState = TrackBase::PAUSING; 6258 // do not wait for mStartStopCond if exiting 6259 if (exitPending()) { 6260 return; 6261 } 6262 mStartStopCond.wait(mLock); 6263 // if we have been restarted, recordTrack == mActiveTrack.get() here 6264 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6265 mLock.unlock(); 6266 AudioSystem::stopInput(mId); 6267 mLock.lock(); 6268 ALOGV("Record stopped OK"); 6269 } 6270 } 6271 } 6272} 6273 6274bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6275{ 6276 return false; 6277} 6278 6279status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6280{ 6281 if (!isValidSyncEvent(event)) { 6282 return BAD_VALUE; 6283 } 6284 6285 Mutex::Autolock _l(mLock); 6286 6287 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6288 mTrack->setSyncEvent(event); 6289 return NO_ERROR; 6290 } 6291 return NAME_NOT_FOUND; 6292} 6293 6294status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6295{ 6296 const size_t SIZE = 256; 6297 char buffer[SIZE]; 6298 String8 result; 6299 6300 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6301 result.append(buffer); 6302 6303 if (mActiveTrack != 0) { 6304 result.append("Active Track:\n"); 6305 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6306 mActiveTrack->dump(buffer, SIZE); 6307 result.append(buffer); 6308 6309 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6310 result.append(buffer); 6311 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6312 result.append(buffer); 6313 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6314 result.append(buffer); 6315 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6316 result.append(buffer); 6317 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6318 result.append(buffer); 6319 6320 6321 } else { 6322 result.append("No record client\n"); 6323 } 6324 write(fd, result.string(), result.size()); 6325 6326 dumpBase(fd, args); 6327 dumpEffectChains(fd, args); 6328 6329 return NO_ERROR; 6330} 6331 6332// AudioBufferProvider interface 6333status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6334{ 6335 size_t framesReq = buffer->frameCount; 6336 size_t framesReady = mFrameCount - mRsmpInIndex; 6337 int channelCount; 6338 6339 if (framesReady == 0) { 6340 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6341 if (mBytesRead < 0) { 6342 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6343 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6344 // Force input into standby so that it tries to 6345 // recover at next read attempt 6346 mInput->stream->common.standby(&mInput->stream->common); 6347 usleep(kRecordThreadSleepUs); 6348 } 6349 buffer->raw = NULL; 6350 buffer->frameCount = 0; 6351 return NOT_ENOUGH_DATA; 6352 } 6353 mRsmpInIndex = 0; 6354 framesReady = mFrameCount; 6355 } 6356 6357 if (framesReq > framesReady) { 6358 framesReq = framesReady; 6359 } 6360 6361 if (mChannelCount == 1 && mReqChannelCount == 2) { 6362 channelCount = 1; 6363 } else { 6364 channelCount = 2; 6365 } 6366 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6367 buffer->frameCount = framesReq; 6368 return NO_ERROR; 6369} 6370 6371// AudioBufferProvider interface 6372void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6373{ 6374 mRsmpInIndex += buffer->frameCount; 6375 buffer->frameCount = 0; 6376} 6377 6378bool AudioFlinger::RecordThread::checkForNewParameters_l() 6379{ 6380 bool reconfig = false; 6381 6382 while (!mNewParameters.isEmpty()) { 6383 status_t status = NO_ERROR; 6384 String8 keyValuePair = mNewParameters[0]; 6385 AudioParameter param = AudioParameter(keyValuePair); 6386 int value; 6387 audio_format_t reqFormat = mFormat; 6388 int reqSamplingRate = mReqSampleRate; 6389 int reqChannelCount = mReqChannelCount; 6390 6391 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6392 reqSamplingRate = value; 6393 reconfig = true; 6394 } 6395 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6396 reqFormat = (audio_format_t) value; 6397 reconfig = true; 6398 } 6399 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6400 reqChannelCount = popcount(value); 6401 reconfig = true; 6402 } 6403 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6404 // do not accept frame count changes if tracks are open as the track buffer 6405 // size depends on frame count and correct behavior would not be guaranteed 6406 // if frame count is changed after track creation 6407 if (mActiveTrack != 0) { 6408 status = INVALID_OPERATION; 6409 } else { 6410 reconfig = true; 6411 } 6412 } 6413 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6414 // forward device change to effects that have requested to be 6415 // aware of attached audio device. 6416 for (size_t i = 0; i < mEffectChains.size(); i++) { 6417 mEffectChains[i]->setDevice_l(value); 6418 } 6419 // store input device and output device but do not forward output device to audio HAL. 6420 // Note that status is ignored by the caller for output device 6421 // (see AudioFlinger::setParameters() 6422 if (value & AUDIO_DEVICE_OUT_ALL) { 6423 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6424 status = BAD_VALUE; 6425 } else { 6426 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6427 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6428 if (mTrack != NULL) { 6429 bool suspend = audio_is_bluetooth_sco_device( 6430 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6431 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6432 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6433 } 6434 } 6435 mDevice |= (uint32_t)value; 6436 } 6437 if (status == NO_ERROR) { 6438 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6439 if (status == INVALID_OPERATION) { 6440 mInput->stream->common.standby(&mInput->stream->common); 6441 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6442 keyValuePair.string()); 6443 } 6444 if (reconfig) { 6445 if (status == BAD_VALUE && 6446 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6447 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6448 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6449 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6450 (reqChannelCount <= FCC_2)) { 6451 status = NO_ERROR; 6452 } 6453 if (status == NO_ERROR) { 6454 readInputParameters(); 6455 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6456 } 6457 } 6458 } 6459 6460 mNewParameters.removeAt(0); 6461 6462 mParamStatus = status; 6463 mParamCond.signal(); 6464 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6465 // already timed out waiting for the status and will never signal the condition. 6466 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6467 } 6468 return reconfig; 6469} 6470 6471String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6472{ 6473 char *s; 6474 String8 out_s8 = String8(); 6475 6476 Mutex::Autolock _l(mLock); 6477 if (initCheck() != NO_ERROR) { 6478 return out_s8; 6479 } 6480 6481 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6482 out_s8 = String8(s); 6483 free(s); 6484 return out_s8; 6485} 6486 6487void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6488 AudioSystem::OutputDescriptor desc; 6489 void *param2 = NULL; 6490 6491 switch (event) { 6492 case AudioSystem::INPUT_OPENED: 6493 case AudioSystem::INPUT_CONFIG_CHANGED: 6494 desc.channels = mChannelMask; 6495 desc.samplingRate = mSampleRate; 6496 desc.format = mFormat; 6497 desc.frameCount = mFrameCount; 6498 desc.latency = 0; 6499 param2 = &desc; 6500 break; 6501 6502 case AudioSystem::INPUT_CLOSED: 6503 default: 6504 break; 6505 } 6506 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6507} 6508 6509void AudioFlinger::RecordThread::readInputParameters() 6510{ 6511 delete mRsmpInBuffer; 6512 // mRsmpInBuffer is always assigned a new[] below 6513 delete mRsmpOutBuffer; 6514 mRsmpOutBuffer = NULL; 6515 delete mResampler; 6516 mResampler = NULL; 6517 6518 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6519 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6520 mChannelCount = (uint16_t)popcount(mChannelMask); 6521 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6522 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6523 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6524 mFrameCount = mInputBytes / mFrameSize; 6525 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6526 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6527 6528 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6529 { 6530 int channelCount; 6531 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6532 // stereo to mono post process as the resampler always outputs stereo. 6533 if (mChannelCount == 1 && mReqChannelCount == 2) { 6534 channelCount = 1; 6535 } else { 6536 channelCount = 2; 6537 } 6538 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6539 mResampler->setSampleRate(mSampleRate); 6540 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6541 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6542 6543 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6544 if (mChannelCount == 1 && mReqChannelCount == 1) { 6545 mFrameCount >>= 1; 6546 } 6547 6548 } 6549 mRsmpInIndex = mFrameCount; 6550} 6551 6552unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6553{ 6554 Mutex::Autolock _l(mLock); 6555 if (initCheck() != NO_ERROR) { 6556 return 0; 6557 } 6558 6559 return mInput->stream->get_input_frames_lost(mInput->stream); 6560} 6561 6562uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6563{ 6564 Mutex::Autolock _l(mLock); 6565 uint32_t result = 0; 6566 if (getEffectChain_l(sessionId) != 0) { 6567 result = EFFECT_SESSION; 6568 } 6569 6570 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6571 result |= TRACK_SESSION; 6572 } 6573 6574 return result; 6575} 6576 6577AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6578{ 6579 Mutex::Autolock _l(mLock); 6580 return mTrack; 6581} 6582 6583AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6584{ 6585 Mutex::Autolock _l(mLock); 6586 return mInput; 6587} 6588 6589AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6590{ 6591 Mutex::Autolock _l(mLock); 6592 AudioStreamIn *input = mInput; 6593 mInput = NULL; 6594 return input; 6595} 6596 6597// this method must always be called either with ThreadBase mLock held or inside the thread loop 6598audio_stream_t* AudioFlinger::RecordThread::stream() const 6599{ 6600 if (mInput == NULL) { 6601 return NULL; 6602 } 6603 return &mInput->stream->common; 6604} 6605 6606 6607// ---------------------------------------------------------------------------- 6608 6609audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6610{ 6611 if (!settingsAllowed()) { 6612 return 0; 6613 } 6614 Mutex::Autolock _l(mLock); 6615 return loadHwModule_l(name); 6616} 6617 6618// loadHwModule_l() must be called with AudioFlinger::mLock held 6619audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6620{ 6621 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6622 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6623 ALOGW("loadHwModule() module %s already loaded", name); 6624 return mAudioHwDevs.keyAt(i); 6625 } 6626 } 6627 6628 audio_hw_device_t *dev; 6629 6630 int rc = load_audio_interface(name, &dev); 6631 if (rc) { 6632 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6633 return 0; 6634 } 6635 6636 mHardwareStatus = AUDIO_HW_INIT; 6637 rc = dev->init_check(dev); 6638 mHardwareStatus = AUDIO_HW_IDLE; 6639 if (rc) { 6640 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6641 return 0; 6642 } 6643 6644 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6645 (NULL != dev->set_master_volume)) { 6646 AutoMutex lock(mHardwareLock); 6647 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6648 dev->set_master_volume(dev, mMasterVolume); 6649 mHardwareStatus = AUDIO_HW_IDLE; 6650 } 6651 6652 audio_module_handle_t handle = nextUniqueId(); 6653 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6654 6655 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6656 name, dev->common.module->name, dev->common.module->id, handle); 6657 6658 return handle; 6659 6660} 6661 6662audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6663 audio_devices_t *pDevices, 6664 uint32_t *pSamplingRate, 6665 audio_format_t *pFormat, 6666 audio_channel_mask_t *pChannelMask, 6667 uint32_t *pLatencyMs, 6668 audio_output_flags_t flags) 6669{ 6670 status_t status; 6671 PlaybackThread *thread = NULL; 6672 struct audio_config config = { 6673 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6674 channel_mask: pChannelMask ? *pChannelMask : 0, 6675 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6676 }; 6677 audio_stream_out_t *outStream = NULL; 6678 audio_hw_device_t *outHwDev; 6679 6680 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6681 module, 6682 (pDevices != NULL) ? (int)*pDevices : 0, 6683 config.sample_rate, 6684 config.format, 6685 config.channel_mask, 6686 flags); 6687 6688 if (pDevices == NULL || *pDevices == 0) { 6689 return 0; 6690 } 6691 6692 Mutex::Autolock _l(mLock); 6693 6694 outHwDev = findSuitableHwDev_l(module, *pDevices); 6695 if (outHwDev == NULL) 6696 return 0; 6697 6698 audio_io_handle_t id = nextUniqueId(); 6699 6700 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6701 6702 status = outHwDev->open_output_stream(outHwDev, 6703 id, 6704 *pDevices, 6705 (audio_output_flags_t)flags, 6706 &config, 6707 &outStream); 6708 6709 mHardwareStatus = AUDIO_HW_IDLE; 6710 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6711 outStream, 6712 config.sample_rate, 6713 config.format, 6714 config.channel_mask, 6715 status); 6716 6717 if (status == NO_ERROR && outStream != NULL) { 6718 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6719 6720 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6721 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6722 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6723 thread = new DirectOutputThread(this, output, id, *pDevices); 6724 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6725 } else { 6726 thread = new MixerThread(this, output, id, *pDevices); 6727 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6728 } 6729 mPlaybackThreads.add(id, thread); 6730 6731 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6732 if (pFormat != NULL) *pFormat = config.format; 6733 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6734 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6735 6736 // notify client processes of the new output creation 6737 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6738 6739 // the first primary output opened designates the primary hw device 6740 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6741 ALOGI("Using module %d has the primary audio interface", module); 6742 mPrimaryHardwareDev = outHwDev; 6743 6744 AutoMutex lock(mHardwareLock); 6745 mHardwareStatus = AUDIO_HW_SET_MODE; 6746 outHwDev->set_mode(outHwDev, mMode); 6747 6748 // Determine the level of master volume support the primary audio HAL has, 6749 // and set the initial master volume at the same time. 6750 float initialVolume = 1.0; 6751 mMasterVolumeSupportLvl = MVS_NONE; 6752 6753 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6754 if ((NULL != outHwDev->get_master_volume) && 6755 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6756 mMasterVolumeSupportLvl = MVS_FULL; 6757 } else { 6758 mMasterVolumeSupportLvl = MVS_SETONLY; 6759 initialVolume = 1.0; 6760 } 6761 6762 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6763 if ((NULL == outHwDev->set_master_volume) || 6764 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6765 mMasterVolumeSupportLvl = MVS_NONE; 6766 } 6767 // now that we have a primary device, initialize master volume on other devices 6768 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6769 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6770 6771 if ((dev != mPrimaryHardwareDev) && 6772 (NULL != dev->set_master_volume)) { 6773 dev->set_master_volume(dev, initialVolume); 6774 } 6775 } 6776 mHardwareStatus = AUDIO_HW_IDLE; 6777 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6778 ? initialVolume 6779 : 1.0; 6780 mMasterVolume = initialVolume; 6781 } 6782 return id; 6783 } 6784 6785 return 0; 6786} 6787 6788audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6789 audio_io_handle_t output2) 6790{ 6791 Mutex::Autolock _l(mLock); 6792 MixerThread *thread1 = checkMixerThread_l(output1); 6793 MixerThread *thread2 = checkMixerThread_l(output2); 6794 6795 if (thread1 == NULL || thread2 == NULL) { 6796 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6797 return 0; 6798 } 6799 6800 audio_io_handle_t id = nextUniqueId(); 6801 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6802 thread->addOutputTrack(thread2); 6803 mPlaybackThreads.add(id, thread); 6804 // notify client processes of the new output creation 6805 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6806 return id; 6807} 6808 6809status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6810{ 6811 // keep strong reference on the playback thread so that 6812 // it is not destroyed while exit() is executed 6813 sp<PlaybackThread> thread; 6814 { 6815 Mutex::Autolock _l(mLock); 6816 thread = checkPlaybackThread_l(output); 6817 if (thread == NULL) { 6818 return BAD_VALUE; 6819 } 6820 6821 ALOGV("closeOutput() %d", output); 6822 6823 if (thread->type() == ThreadBase::MIXER) { 6824 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6825 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6826 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6827 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6828 } 6829 } 6830 } 6831 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6832 mPlaybackThreads.removeItem(output); 6833 } 6834 thread->exit(); 6835 // The thread entity (active unit of execution) is no longer running here, 6836 // but the ThreadBase container still exists. 6837 6838 if (thread->type() != ThreadBase::DUPLICATING) { 6839 AudioStreamOut *out = thread->clearOutput(); 6840 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6841 // from now on thread->mOutput is NULL 6842 out->hwDev->close_output_stream(out->hwDev, out->stream); 6843 delete out; 6844 } 6845 return NO_ERROR; 6846} 6847 6848status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6849{ 6850 Mutex::Autolock _l(mLock); 6851 PlaybackThread *thread = checkPlaybackThread_l(output); 6852 6853 if (thread == NULL) { 6854 return BAD_VALUE; 6855 } 6856 6857 ALOGV("suspendOutput() %d", output); 6858 thread->suspend(); 6859 6860 return NO_ERROR; 6861} 6862 6863status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6864{ 6865 Mutex::Autolock _l(mLock); 6866 PlaybackThread *thread = checkPlaybackThread_l(output); 6867 6868 if (thread == NULL) { 6869 return BAD_VALUE; 6870 } 6871 6872 ALOGV("restoreOutput() %d", output); 6873 6874 thread->restore(); 6875 6876 return NO_ERROR; 6877} 6878 6879audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6880 audio_devices_t *pDevices, 6881 uint32_t *pSamplingRate, 6882 audio_format_t *pFormat, 6883 uint32_t *pChannelMask) 6884{ 6885 status_t status; 6886 RecordThread *thread = NULL; 6887 struct audio_config config = { 6888 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6889 channel_mask: pChannelMask ? *pChannelMask : 0, 6890 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6891 }; 6892 uint32_t reqSamplingRate = config.sample_rate; 6893 audio_format_t reqFormat = config.format; 6894 audio_channel_mask_t reqChannels = config.channel_mask; 6895 audio_stream_in_t *inStream = NULL; 6896 audio_hw_device_t *inHwDev; 6897 6898 if (pDevices == NULL || *pDevices == 0) { 6899 return 0; 6900 } 6901 6902 Mutex::Autolock _l(mLock); 6903 6904 inHwDev = findSuitableHwDev_l(module, *pDevices); 6905 if (inHwDev == NULL) 6906 return 0; 6907 6908 audio_io_handle_t id = nextUniqueId(); 6909 6910 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6911 &inStream); 6912 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6913 inStream, 6914 config.sample_rate, 6915 config.format, 6916 config.channel_mask, 6917 status); 6918 6919 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6920 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6921 // or stereo to mono conversions on 16 bit PCM inputs. 6922 if (status == BAD_VALUE && 6923 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6924 (config.sample_rate <= 2 * reqSamplingRate) && 6925 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6926 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6927 inStream = NULL; 6928 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6929 } 6930 6931 if (status == NO_ERROR && inStream != NULL) { 6932 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6933 6934 // Start record thread 6935 // RecorThread require both input and output device indication to forward to audio 6936 // pre processing modules 6937 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6938 thread = new RecordThread(this, 6939 input, 6940 reqSamplingRate, 6941 reqChannels, 6942 id, 6943 device); 6944 mRecordThreads.add(id, thread); 6945 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6946 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6947 if (pFormat != NULL) *pFormat = config.format; 6948 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6949 6950 input->stream->common.standby(&input->stream->common); 6951 6952 // notify client processes of the new input creation 6953 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6954 return id; 6955 } 6956 6957 return 0; 6958} 6959 6960status_t AudioFlinger::closeInput(audio_io_handle_t input) 6961{ 6962 // keep strong reference on the record thread so that 6963 // it is not destroyed while exit() is executed 6964 sp<RecordThread> thread; 6965 { 6966 Mutex::Autolock _l(mLock); 6967 thread = checkRecordThread_l(input); 6968 if (thread == NULL) { 6969 return BAD_VALUE; 6970 } 6971 6972 ALOGV("closeInput() %d", input); 6973 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6974 mRecordThreads.removeItem(input); 6975 } 6976 thread->exit(); 6977 // The thread entity (active unit of execution) is no longer running here, 6978 // but the ThreadBase container still exists. 6979 6980 AudioStreamIn *in = thread->clearInput(); 6981 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6982 // from now on thread->mInput is NULL 6983 in->hwDev->close_input_stream(in->hwDev, in->stream); 6984 delete in; 6985 6986 return NO_ERROR; 6987} 6988 6989status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6990{ 6991 Mutex::Autolock _l(mLock); 6992 MixerThread *dstThread = checkMixerThread_l(output); 6993 if (dstThread == NULL) { 6994 ALOGW("setStreamOutput() bad output id %d", output); 6995 return BAD_VALUE; 6996 } 6997 6998 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6999 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7000 7001 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7002 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7003 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7004 MixerThread *srcThread = (MixerThread *)thread; 7005 srcThread->invalidateTracks(stream); 7006 } 7007 } 7008 7009 return NO_ERROR; 7010} 7011 7012 7013int AudioFlinger::newAudioSessionId() 7014{ 7015 return nextUniqueId(); 7016} 7017 7018void AudioFlinger::acquireAudioSessionId(int audioSession) 7019{ 7020 Mutex::Autolock _l(mLock); 7021 pid_t caller = IPCThreadState::self()->getCallingPid(); 7022 ALOGV("acquiring %d from %d", audioSession, caller); 7023 size_t num = mAudioSessionRefs.size(); 7024 for (size_t i = 0; i< num; i++) { 7025 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7026 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7027 ref->mCnt++; 7028 ALOGV(" incremented refcount to %d", ref->mCnt); 7029 return; 7030 } 7031 } 7032 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7033 ALOGV(" added new entry for %d", audioSession); 7034} 7035 7036void AudioFlinger::releaseAudioSessionId(int audioSession) 7037{ 7038 Mutex::Autolock _l(mLock); 7039 pid_t caller = IPCThreadState::self()->getCallingPid(); 7040 ALOGV("releasing %d from %d", audioSession, caller); 7041 size_t num = mAudioSessionRefs.size(); 7042 for (size_t i = 0; i< num; i++) { 7043 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7044 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7045 ref->mCnt--; 7046 ALOGV(" decremented refcount to %d", ref->mCnt); 7047 if (ref->mCnt == 0) { 7048 mAudioSessionRefs.removeAt(i); 7049 delete ref; 7050 purgeStaleEffects_l(); 7051 } 7052 return; 7053 } 7054 } 7055 ALOGW("session id %d not found for pid %d", audioSession, caller); 7056} 7057 7058void AudioFlinger::purgeStaleEffects_l() { 7059 7060 ALOGV("purging stale effects"); 7061 7062 Vector< sp<EffectChain> > chains; 7063 7064 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7065 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7066 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7067 sp<EffectChain> ec = t->mEffectChains[j]; 7068 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7069 chains.push(ec); 7070 } 7071 } 7072 } 7073 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7074 sp<RecordThread> t = mRecordThreads.valueAt(i); 7075 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7076 sp<EffectChain> ec = t->mEffectChains[j]; 7077 chains.push(ec); 7078 } 7079 } 7080 7081 for (size_t i = 0; i < chains.size(); i++) { 7082 sp<EffectChain> ec = chains[i]; 7083 int sessionid = ec->sessionId(); 7084 sp<ThreadBase> t = ec->mThread.promote(); 7085 if (t == 0) { 7086 continue; 7087 } 7088 size_t numsessionrefs = mAudioSessionRefs.size(); 7089 bool found = false; 7090 for (size_t k = 0; k < numsessionrefs; k++) { 7091 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7092 if (ref->mSessionid == sessionid) { 7093 ALOGV(" session %d still exists for %d with %d refs", 7094 sessionid, ref->mPid, ref->mCnt); 7095 found = true; 7096 break; 7097 } 7098 } 7099 if (!found) { 7100 // remove all effects from the chain 7101 while (ec->mEffects.size()) { 7102 sp<EffectModule> effect = ec->mEffects[0]; 7103 effect->unPin(); 7104 Mutex::Autolock _l (t->mLock); 7105 t->removeEffect_l(effect); 7106 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7107 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7108 if (handle != 0) { 7109 handle->mEffect.clear(); 7110 if (handle->mHasControl && handle->mEnabled) { 7111 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7112 } 7113 } 7114 } 7115 AudioSystem::unregisterEffect(effect->id()); 7116 } 7117 } 7118 } 7119 return; 7120} 7121 7122// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7123AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7124{ 7125 return mPlaybackThreads.valueFor(output).get(); 7126} 7127 7128// checkMixerThread_l() must be called with AudioFlinger::mLock held 7129AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7130{ 7131 PlaybackThread *thread = checkPlaybackThread_l(output); 7132 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7133} 7134 7135// checkRecordThread_l() must be called with AudioFlinger::mLock held 7136AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7137{ 7138 return mRecordThreads.valueFor(input).get(); 7139} 7140 7141uint32_t AudioFlinger::nextUniqueId() 7142{ 7143 return android_atomic_inc(&mNextUniqueId); 7144} 7145 7146AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7147{ 7148 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7149 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7150 AudioStreamOut *output = thread->getOutput(); 7151 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7152 return thread; 7153 } 7154 } 7155 return NULL; 7156} 7157 7158uint32_t AudioFlinger::primaryOutputDevice_l() const 7159{ 7160 PlaybackThread *thread = primaryPlaybackThread_l(); 7161 7162 if (thread == NULL) { 7163 return 0; 7164 } 7165 7166 return thread->device(); 7167} 7168 7169sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7170 int triggerSession, 7171 int listenerSession, 7172 sync_event_callback_t callBack, 7173 void *cookie) 7174{ 7175 Mutex::Autolock _l(mLock); 7176 7177 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7178 status_t playStatus = NAME_NOT_FOUND; 7179 status_t recStatus = NAME_NOT_FOUND; 7180 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7181 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7182 if (playStatus == NO_ERROR) { 7183 return event; 7184 } 7185 } 7186 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7187 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7188 if (recStatus == NO_ERROR) { 7189 return event; 7190 } 7191 } 7192 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7193 mPendingSyncEvents.add(event); 7194 } else { 7195 ALOGV("createSyncEvent() invalid event %d", event->type()); 7196 event.clear(); 7197 } 7198 return event; 7199} 7200 7201// ---------------------------------------------------------------------------- 7202// Effect management 7203// ---------------------------------------------------------------------------- 7204 7205 7206status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7207{ 7208 Mutex::Autolock _l(mLock); 7209 return EffectQueryNumberEffects(numEffects); 7210} 7211 7212status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7213{ 7214 Mutex::Autolock _l(mLock); 7215 return EffectQueryEffect(index, descriptor); 7216} 7217 7218status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7219 effect_descriptor_t *descriptor) const 7220{ 7221 Mutex::Autolock _l(mLock); 7222 return EffectGetDescriptor(pUuid, descriptor); 7223} 7224 7225 7226sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7227 effect_descriptor_t *pDesc, 7228 const sp<IEffectClient>& effectClient, 7229 int32_t priority, 7230 audio_io_handle_t io, 7231 int sessionId, 7232 status_t *status, 7233 int *id, 7234 int *enabled) 7235{ 7236 status_t lStatus = NO_ERROR; 7237 sp<EffectHandle> handle; 7238 effect_descriptor_t desc; 7239 7240 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7241 pid, effectClient.get(), priority, sessionId, io); 7242 7243 if (pDesc == NULL) { 7244 lStatus = BAD_VALUE; 7245 goto Exit; 7246 } 7247 7248 // check audio settings permission for global effects 7249 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7250 lStatus = PERMISSION_DENIED; 7251 goto Exit; 7252 } 7253 7254 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7255 // that can only be created by audio policy manager (running in same process) 7256 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7257 lStatus = PERMISSION_DENIED; 7258 goto Exit; 7259 } 7260 7261 if (io == 0) { 7262 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7263 // output must be specified by AudioPolicyManager when using session 7264 // AUDIO_SESSION_OUTPUT_STAGE 7265 lStatus = BAD_VALUE; 7266 goto Exit; 7267 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7268 // if the output returned by getOutputForEffect() is removed before we lock the 7269 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7270 // and we will exit safely 7271 io = AudioSystem::getOutputForEffect(&desc); 7272 } 7273 } 7274 7275 { 7276 Mutex::Autolock _l(mLock); 7277 7278 7279 if (!EffectIsNullUuid(&pDesc->uuid)) { 7280 // if uuid is specified, request effect descriptor 7281 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7282 if (lStatus < 0) { 7283 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7284 goto Exit; 7285 } 7286 } else { 7287 // if uuid is not specified, look for an available implementation 7288 // of the required type in effect factory 7289 if (EffectIsNullUuid(&pDesc->type)) { 7290 ALOGW("createEffect() no effect type"); 7291 lStatus = BAD_VALUE; 7292 goto Exit; 7293 } 7294 uint32_t numEffects = 0; 7295 effect_descriptor_t d; 7296 d.flags = 0; // prevent compiler warning 7297 bool found = false; 7298 7299 lStatus = EffectQueryNumberEffects(&numEffects); 7300 if (lStatus < 0) { 7301 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7302 goto Exit; 7303 } 7304 for (uint32_t i = 0; i < numEffects; i++) { 7305 lStatus = EffectQueryEffect(i, &desc); 7306 if (lStatus < 0) { 7307 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7308 continue; 7309 } 7310 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7311 // If matching type found save effect descriptor. If the session is 7312 // 0 and the effect is not auxiliary, continue enumeration in case 7313 // an auxiliary version of this effect type is available 7314 found = true; 7315 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7316 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7317 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7318 break; 7319 } 7320 } 7321 } 7322 if (!found) { 7323 lStatus = BAD_VALUE; 7324 ALOGW("createEffect() effect not found"); 7325 goto Exit; 7326 } 7327 // For same effect type, chose auxiliary version over insert version if 7328 // connect to output mix (Compliance to OpenSL ES) 7329 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7330 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7331 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7332 } 7333 } 7334 7335 // Do not allow auxiliary effects on a session different from 0 (output mix) 7336 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7337 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7338 lStatus = INVALID_OPERATION; 7339 goto Exit; 7340 } 7341 7342 // check recording permission for visualizer 7343 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7344 !recordingAllowed()) { 7345 lStatus = PERMISSION_DENIED; 7346 goto Exit; 7347 } 7348 7349 // return effect descriptor 7350 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7351 7352 // If output is not specified try to find a matching audio session ID in one of the 7353 // output threads. 7354 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7355 // because of code checking output when entering the function. 7356 // Note: io is never 0 when creating an effect on an input 7357 if (io == 0) { 7358 // look for the thread where the specified audio session is present 7359 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7360 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7361 io = mPlaybackThreads.keyAt(i); 7362 break; 7363 } 7364 } 7365 if (io == 0) { 7366 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7367 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7368 io = mRecordThreads.keyAt(i); 7369 break; 7370 } 7371 } 7372 } 7373 // If no output thread contains the requested session ID, default to 7374 // first output. The effect chain will be moved to the correct output 7375 // thread when a track with the same session ID is created 7376 if (io == 0 && mPlaybackThreads.size()) { 7377 io = mPlaybackThreads.keyAt(0); 7378 } 7379 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7380 } 7381 ThreadBase *thread = checkRecordThread_l(io); 7382 if (thread == NULL) { 7383 thread = checkPlaybackThread_l(io); 7384 if (thread == NULL) { 7385 ALOGE("createEffect() unknown output thread"); 7386 lStatus = BAD_VALUE; 7387 goto Exit; 7388 } 7389 } 7390 7391 sp<Client> client = registerPid_l(pid); 7392 7393 // create effect on selected output thread 7394 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7395 &desc, enabled, &lStatus); 7396 if (handle != 0 && id != NULL) { 7397 *id = handle->id(); 7398 } 7399 } 7400 7401Exit: 7402 if (status != NULL) { 7403 *status = lStatus; 7404 } 7405 return handle; 7406} 7407 7408status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7409 audio_io_handle_t dstOutput) 7410{ 7411 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7412 sessionId, srcOutput, dstOutput); 7413 Mutex::Autolock _l(mLock); 7414 if (srcOutput == dstOutput) { 7415 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7416 return NO_ERROR; 7417 } 7418 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7419 if (srcThread == NULL) { 7420 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7421 return BAD_VALUE; 7422 } 7423 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7424 if (dstThread == NULL) { 7425 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7426 return BAD_VALUE; 7427 } 7428 7429 Mutex::Autolock _dl(dstThread->mLock); 7430 Mutex::Autolock _sl(srcThread->mLock); 7431 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7432 7433 return NO_ERROR; 7434} 7435 7436// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7437status_t AudioFlinger::moveEffectChain_l(int sessionId, 7438 AudioFlinger::PlaybackThread *srcThread, 7439 AudioFlinger::PlaybackThread *dstThread, 7440 bool reRegister) 7441{ 7442 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7443 sessionId, srcThread, dstThread); 7444 7445 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7446 if (chain == 0) { 7447 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7448 sessionId, srcThread); 7449 return INVALID_OPERATION; 7450 } 7451 7452 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7453 // so that a new chain is created with correct parameters when first effect is added. This is 7454 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7455 // removed. 7456 srcThread->removeEffectChain_l(chain); 7457 7458 // transfer all effects one by one so that new effect chain is created on new thread with 7459 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7460 audio_io_handle_t dstOutput = dstThread->id(); 7461 sp<EffectChain> dstChain; 7462 uint32_t strategy = 0; // prevent compiler warning 7463 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7464 while (effect != 0) { 7465 srcThread->removeEffect_l(effect); 7466 dstThread->addEffect_l(effect); 7467 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7468 if (effect->state() == EffectModule::ACTIVE || 7469 effect->state() == EffectModule::STOPPING) { 7470 effect->start(); 7471 } 7472 // if the move request is not received from audio policy manager, the effect must be 7473 // re-registered with the new strategy and output 7474 if (dstChain == 0) { 7475 dstChain = effect->chain().promote(); 7476 if (dstChain == 0) { 7477 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7478 srcThread->addEffect_l(effect); 7479 return NO_INIT; 7480 } 7481 strategy = dstChain->strategy(); 7482 } 7483 if (reRegister) { 7484 AudioSystem::unregisterEffect(effect->id()); 7485 AudioSystem::registerEffect(&effect->desc(), 7486 dstOutput, 7487 strategy, 7488 sessionId, 7489 effect->id()); 7490 } 7491 effect = chain->getEffectFromId_l(0); 7492 } 7493 7494 return NO_ERROR; 7495} 7496 7497 7498// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7499sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7500 const sp<AudioFlinger::Client>& client, 7501 const sp<IEffectClient>& effectClient, 7502 int32_t priority, 7503 int sessionId, 7504 effect_descriptor_t *desc, 7505 int *enabled, 7506 status_t *status 7507 ) 7508{ 7509 sp<EffectModule> effect; 7510 sp<EffectHandle> handle; 7511 status_t lStatus; 7512 sp<EffectChain> chain; 7513 bool chainCreated = false; 7514 bool effectCreated = false; 7515 bool effectRegistered = false; 7516 7517 lStatus = initCheck(); 7518 if (lStatus != NO_ERROR) { 7519 ALOGW("createEffect_l() Audio driver not initialized."); 7520 goto Exit; 7521 } 7522 7523 // Do not allow effects with session ID 0 on direct output or duplicating threads 7524 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7525 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7526 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7527 desc->name, sessionId); 7528 lStatus = BAD_VALUE; 7529 goto Exit; 7530 } 7531 // Only Pre processor effects are allowed on input threads and only on input threads 7532 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7533 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7534 desc->name, desc->flags, mType); 7535 lStatus = BAD_VALUE; 7536 goto Exit; 7537 } 7538 7539 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7540 7541 { // scope for mLock 7542 Mutex::Autolock _l(mLock); 7543 7544 // check for existing effect chain with the requested audio session 7545 chain = getEffectChain_l(sessionId); 7546 if (chain == 0) { 7547 // create a new chain for this session 7548 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7549 chain = new EffectChain(this, sessionId); 7550 addEffectChain_l(chain); 7551 chain->setStrategy(getStrategyForSession_l(sessionId)); 7552 chainCreated = true; 7553 } else { 7554 effect = chain->getEffectFromDesc_l(desc); 7555 } 7556 7557 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7558 7559 if (effect == 0) { 7560 int id = mAudioFlinger->nextUniqueId(); 7561 // Check CPU and memory usage 7562 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7563 if (lStatus != NO_ERROR) { 7564 goto Exit; 7565 } 7566 effectRegistered = true; 7567 // create a new effect module if none present in the chain 7568 effect = new EffectModule(this, chain, desc, id, sessionId); 7569 lStatus = effect->status(); 7570 if (lStatus != NO_ERROR) { 7571 goto Exit; 7572 } 7573 lStatus = chain->addEffect_l(effect); 7574 if (lStatus != NO_ERROR) { 7575 goto Exit; 7576 } 7577 effectCreated = true; 7578 7579 effect->setDevice(mDevice); 7580 effect->setMode(mAudioFlinger->getMode()); 7581 } 7582 // create effect handle and connect it to effect module 7583 handle = new EffectHandle(effect, client, effectClient, priority); 7584 lStatus = effect->addHandle(handle); 7585 if (enabled != NULL) { 7586 *enabled = (int)effect->isEnabled(); 7587 } 7588 } 7589 7590Exit: 7591 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7592 Mutex::Autolock _l(mLock); 7593 if (effectCreated) { 7594 chain->removeEffect_l(effect); 7595 } 7596 if (effectRegistered) { 7597 AudioSystem::unregisterEffect(effect->id()); 7598 } 7599 if (chainCreated) { 7600 removeEffectChain_l(chain); 7601 } 7602 handle.clear(); 7603 } 7604 7605 if (status != NULL) { 7606 *status = lStatus; 7607 } 7608 return handle; 7609} 7610 7611sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7612{ 7613 sp<EffectChain> chain = getEffectChain_l(sessionId); 7614 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7615} 7616 7617// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7618// PlaybackThread::mLock held 7619status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7620{ 7621 // check for existing effect chain with the requested audio session 7622 int sessionId = effect->sessionId(); 7623 sp<EffectChain> chain = getEffectChain_l(sessionId); 7624 bool chainCreated = false; 7625 7626 if (chain == 0) { 7627 // create a new chain for this session 7628 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7629 chain = new EffectChain(this, sessionId); 7630 addEffectChain_l(chain); 7631 chain->setStrategy(getStrategyForSession_l(sessionId)); 7632 chainCreated = true; 7633 } 7634 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7635 7636 if (chain->getEffectFromId_l(effect->id()) != 0) { 7637 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7638 this, effect->desc().name, chain.get()); 7639 return BAD_VALUE; 7640 } 7641 7642 status_t status = chain->addEffect_l(effect); 7643 if (status != NO_ERROR) { 7644 if (chainCreated) { 7645 removeEffectChain_l(chain); 7646 } 7647 return status; 7648 } 7649 7650 effect->setDevice(mDevice); 7651 effect->setMode(mAudioFlinger->getMode()); 7652 return NO_ERROR; 7653} 7654 7655void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7656 7657 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7658 effect_descriptor_t desc = effect->desc(); 7659 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7660 detachAuxEffect_l(effect->id()); 7661 } 7662 7663 sp<EffectChain> chain = effect->chain().promote(); 7664 if (chain != 0) { 7665 // remove effect chain if removing last effect 7666 if (chain->removeEffect_l(effect) == 0) { 7667 removeEffectChain_l(chain); 7668 } 7669 } else { 7670 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7671 } 7672} 7673 7674void AudioFlinger::ThreadBase::lockEffectChains_l( 7675 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7676{ 7677 effectChains = mEffectChains; 7678 for (size_t i = 0; i < mEffectChains.size(); i++) { 7679 mEffectChains[i]->lock(); 7680 } 7681} 7682 7683void AudioFlinger::ThreadBase::unlockEffectChains( 7684 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7685{ 7686 for (size_t i = 0; i < effectChains.size(); i++) { 7687 effectChains[i]->unlock(); 7688 } 7689} 7690 7691sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7692{ 7693 Mutex::Autolock _l(mLock); 7694 return getEffectChain_l(sessionId); 7695} 7696 7697sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7698{ 7699 size_t size = mEffectChains.size(); 7700 for (size_t i = 0; i < size; i++) { 7701 if (mEffectChains[i]->sessionId() == sessionId) { 7702 return mEffectChains[i]; 7703 } 7704 } 7705 return 0; 7706} 7707 7708void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7709{ 7710 Mutex::Autolock _l(mLock); 7711 size_t size = mEffectChains.size(); 7712 for (size_t i = 0; i < size; i++) { 7713 mEffectChains[i]->setMode_l(mode); 7714 } 7715} 7716 7717void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7718 const wp<EffectHandle>& handle, 7719 bool unpinIfLast) { 7720 7721 Mutex::Autolock _l(mLock); 7722 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7723 // delete the effect module if removing last handle on it 7724 if (effect->removeHandle(handle) == 0) { 7725 if (!effect->isPinned() || unpinIfLast) { 7726 removeEffect_l(effect); 7727 AudioSystem::unregisterEffect(effect->id()); 7728 } 7729 } 7730} 7731 7732status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7733{ 7734 int session = chain->sessionId(); 7735 int16_t *buffer = mMixBuffer; 7736 bool ownsBuffer = false; 7737 7738 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7739 if (session > 0) { 7740 // Only one effect chain can be present in direct output thread and it uses 7741 // the mix buffer as input 7742 if (mType != DIRECT) { 7743 size_t numSamples = mNormalFrameCount * mChannelCount; 7744 buffer = new int16_t[numSamples]; 7745 memset(buffer, 0, numSamples * sizeof(int16_t)); 7746 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7747 ownsBuffer = true; 7748 } 7749 7750 // Attach all tracks with same session ID to this chain. 7751 for (size_t i = 0; i < mTracks.size(); ++i) { 7752 sp<Track> track = mTracks[i]; 7753 if (session == track->sessionId()) { 7754 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7755 track->setMainBuffer(buffer); 7756 chain->incTrackCnt(); 7757 } 7758 } 7759 7760 // indicate all active tracks in the chain 7761 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7762 sp<Track> track = mActiveTracks[i].promote(); 7763 if (track == 0) continue; 7764 if (session == track->sessionId()) { 7765 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7766 chain->incActiveTrackCnt(); 7767 } 7768 } 7769 } 7770 7771 chain->setInBuffer(buffer, ownsBuffer); 7772 chain->setOutBuffer(mMixBuffer); 7773 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7774 // chains list in order to be processed last as it contains output stage effects 7775 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7776 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7777 // after track specific effects and before output stage 7778 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7779 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7780 // Effect chain for other sessions are inserted at beginning of effect 7781 // chains list to be processed before output mix effects. Relative order between other 7782 // sessions is not important 7783 size_t size = mEffectChains.size(); 7784 size_t i = 0; 7785 for (i = 0; i < size; i++) { 7786 if (mEffectChains[i]->sessionId() < session) break; 7787 } 7788 mEffectChains.insertAt(chain, i); 7789 checkSuspendOnAddEffectChain_l(chain); 7790 7791 return NO_ERROR; 7792} 7793 7794size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7795{ 7796 int session = chain->sessionId(); 7797 7798 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7799 7800 for (size_t i = 0; i < mEffectChains.size(); i++) { 7801 if (chain == mEffectChains[i]) { 7802 mEffectChains.removeAt(i); 7803 // detach all active tracks from the chain 7804 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7805 sp<Track> track = mActiveTracks[i].promote(); 7806 if (track == 0) continue; 7807 if (session == track->sessionId()) { 7808 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7809 chain.get(), session); 7810 chain->decActiveTrackCnt(); 7811 } 7812 } 7813 7814 // detach all tracks with same session ID from this chain 7815 for (size_t i = 0; i < mTracks.size(); ++i) { 7816 sp<Track> track = mTracks[i]; 7817 if (session == track->sessionId()) { 7818 track->setMainBuffer(mMixBuffer); 7819 chain->decTrackCnt(); 7820 } 7821 } 7822 break; 7823 } 7824 } 7825 return mEffectChains.size(); 7826} 7827 7828status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7829 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7830{ 7831 Mutex::Autolock _l(mLock); 7832 return attachAuxEffect_l(track, EffectId); 7833} 7834 7835status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7836 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7837{ 7838 status_t status = NO_ERROR; 7839 7840 if (EffectId == 0) { 7841 track->setAuxBuffer(0, NULL); 7842 } else { 7843 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7844 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7845 if (effect != 0) { 7846 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7847 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7848 } else { 7849 status = INVALID_OPERATION; 7850 } 7851 } else { 7852 status = BAD_VALUE; 7853 } 7854 } 7855 return status; 7856} 7857 7858void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7859{ 7860 for (size_t i = 0; i < mTracks.size(); ++i) { 7861 sp<Track> track = mTracks[i]; 7862 if (track->auxEffectId() == effectId) { 7863 attachAuxEffect_l(track, 0); 7864 } 7865 } 7866} 7867 7868status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7869{ 7870 // only one chain per input thread 7871 if (mEffectChains.size() != 0) { 7872 return INVALID_OPERATION; 7873 } 7874 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7875 7876 chain->setInBuffer(NULL); 7877 chain->setOutBuffer(NULL); 7878 7879 checkSuspendOnAddEffectChain_l(chain); 7880 7881 mEffectChains.add(chain); 7882 7883 return NO_ERROR; 7884} 7885 7886size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7887{ 7888 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7889 ALOGW_IF(mEffectChains.size() != 1, 7890 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7891 chain.get(), mEffectChains.size(), this); 7892 if (mEffectChains.size() == 1) { 7893 mEffectChains.removeAt(0); 7894 } 7895 return 0; 7896} 7897 7898// ---------------------------------------------------------------------------- 7899// EffectModule implementation 7900// ---------------------------------------------------------------------------- 7901 7902#undef LOG_TAG 7903#define LOG_TAG "AudioFlinger::EffectModule" 7904 7905AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7906 const wp<AudioFlinger::EffectChain>& chain, 7907 effect_descriptor_t *desc, 7908 int id, 7909 int sessionId) 7910 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7911 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7912{ 7913 ALOGV("Constructor %p", this); 7914 int lStatus; 7915 if (thread == NULL) { 7916 return; 7917 } 7918 7919 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7920 7921 // create effect engine from effect factory 7922 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7923 7924 if (mStatus != NO_ERROR) { 7925 return; 7926 } 7927 lStatus = init(); 7928 if (lStatus < 0) { 7929 mStatus = lStatus; 7930 goto Error; 7931 } 7932 7933 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7934 mPinned = true; 7935 } 7936 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7937 return; 7938Error: 7939 EffectRelease(mEffectInterface); 7940 mEffectInterface = NULL; 7941 ALOGV("Constructor Error %d", mStatus); 7942} 7943 7944AudioFlinger::EffectModule::~EffectModule() 7945{ 7946 ALOGV("Destructor %p", this); 7947 if (mEffectInterface != NULL) { 7948 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7949 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7950 sp<ThreadBase> thread = mThread.promote(); 7951 if (thread != 0) { 7952 audio_stream_t *stream = thread->stream(); 7953 if (stream != NULL) { 7954 stream->remove_audio_effect(stream, mEffectInterface); 7955 } 7956 } 7957 } 7958 // release effect engine 7959 EffectRelease(mEffectInterface); 7960 } 7961} 7962 7963status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7964{ 7965 status_t status; 7966 7967 Mutex::Autolock _l(mLock); 7968 int priority = handle->priority(); 7969 size_t size = mHandles.size(); 7970 sp<EffectHandle> h; 7971 size_t i; 7972 for (i = 0; i < size; i++) { 7973 h = mHandles[i].promote(); 7974 if (h == 0) continue; 7975 if (h->priority() <= priority) break; 7976 } 7977 // if inserted in first place, move effect control from previous owner to this handle 7978 if (i == 0) { 7979 bool enabled = false; 7980 if (h != 0) { 7981 enabled = h->enabled(); 7982 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7983 } 7984 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7985 status = NO_ERROR; 7986 } else { 7987 status = ALREADY_EXISTS; 7988 } 7989 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7990 mHandles.insertAt(handle, i); 7991 return status; 7992} 7993 7994size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7995{ 7996 Mutex::Autolock _l(mLock); 7997 size_t size = mHandles.size(); 7998 size_t i; 7999 for (i = 0; i < size; i++) { 8000 if (mHandles[i] == handle) break; 8001 } 8002 if (i == size) { 8003 return size; 8004 } 8005 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8006 8007 bool enabled = false; 8008 EffectHandle *hdl = handle.unsafe_get(); 8009 if (hdl != NULL) { 8010 ALOGV("removeHandle() unsafe_get OK"); 8011 enabled = hdl->enabled(); 8012 } 8013 mHandles.removeAt(i); 8014 size = mHandles.size(); 8015 // if removed from first place, move effect control from this handle to next in line 8016 if (i == 0 && size != 0) { 8017 sp<EffectHandle> h = mHandles[0].promote(); 8018 if (h != 0) { 8019 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8020 } 8021 } 8022 8023 // Prevent calls to process() and other functions on effect interface from now on. 8024 // The effect engine will be released by the destructor when the last strong reference on 8025 // this object is released which can happen after next process is called. 8026 if (size == 0 && !mPinned) { 8027 mState = DESTROYED; 8028 } 8029 8030 return size; 8031} 8032 8033sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8034{ 8035 Mutex::Autolock _l(mLock); 8036 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8037} 8038 8039void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8040{ 8041 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8042 // keep a strong reference on this EffectModule to avoid calling the 8043 // destructor before we exit 8044 sp<EffectModule> keep(this); 8045 { 8046 sp<ThreadBase> thread = mThread.promote(); 8047 if (thread != 0) { 8048 thread->disconnectEffect(keep, handle, unpinIfLast); 8049 } 8050 } 8051} 8052 8053void AudioFlinger::EffectModule::updateState() { 8054 Mutex::Autolock _l(mLock); 8055 8056 switch (mState) { 8057 case RESTART: 8058 reset_l(); 8059 // FALL THROUGH 8060 8061 case STARTING: 8062 // clear auxiliary effect input buffer for next accumulation 8063 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8064 memset(mConfig.inputCfg.buffer.raw, 8065 0, 8066 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8067 } 8068 start_l(); 8069 mState = ACTIVE; 8070 break; 8071 case STOPPING: 8072 stop_l(); 8073 mDisableWaitCnt = mMaxDisableWaitCnt; 8074 mState = STOPPED; 8075 break; 8076 case STOPPED: 8077 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8078 // turn off sequence. 8079 if (--mDisableWaitCnt == 0) { 8080 reset_l(); 8081 mState = IDLE; 8082 } 8083 break; 8084 default: //IDLE , ACTIVE, DESTROYED 8085 break; 8086 } 8087} 8088 8089void AudioFlinger::EffectModule::process() 8090{ 8091 Mutex::Autolock _l(mLock); 8092 8093 if (mState == DESTROYED || mEffectInterface == NULL || 8094 mConfig.inputCfg.buffer.raw == NULL || 8095 mConfig.outputCfg.buffer.raw == NULL) { 8096 return; 8097 } 8098 8099 if (isProcessEnabled()) { 8100 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8101 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8102 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8103 mConfig.inputCfg.buffer.s32, 8104 mConfig.inputCfg.buffer.frameCount/2); 8105 } 8106 8107 // do the actual processing in the effect engine 8108 int ret = (*mEffectInterface)->process(mEffectInterface, 8109 &mConfig.inputCfg.buffer, 8110 &mConfig.outputCfg.buffer); 8111 8112 // force transition to IDLE state when engine is ready 8113 if (mState == STOPPED && ret == -ENODATA) { 8114 mDisableWaitCnt = 1; 8115 } 8116 8117 // clear auxiliary effect input buffer for next accumulation 8118 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8119 memset(mConfig.inputCfg.buffer.raw, 0, 8120 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8121 } 8122 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8123 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8124 // If an insert effect is idle and input buffer is different from output buffer, 8125 // accumulate input onto output 8126 sp<EffectChain> chain = mChain.promote(); 8127 if (chain != 0 && chain->activeTrackCnt() != 0) { 8128 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8129 int16_t *in = mConfig.inputCfg.buffer.s16; 8130 int16_t *out = mConfig.outputCfg.buffer.s16; 8131 for (size_t i = 0; i < frameCnt; i++) { 8132 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8133 } 8134 } 8135 } 8136} 8137 8138void AudioFlinger::EffectModule::reset_l() 8139{ 8140 if (mEffectInterface == NULL) { 8141 return; 8142 } 8143 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8144} 8145 8146status_t AudioFlinger::EffectModule::configure() 8147{ 8148 uint32_t channels; 8149 if (mEffectInterface == NULL) { 8150 return NO_INIT; 8151 } 8152 8153 sp<ThreadBase> thread = mThread.promote(); 8154 if (thread == 0) { 8155 return DEAD_OBJECT; 8156 } 8157 8158 // TODO: handle configuration of effects replacing track process 8159 if (thread->channelCount() == 1) { 8160 channels = AUDIO_CHANNEL_OUT_MONO; 8161 } else { 8162 channels = AUDIO_CHANNEL_OUT_STEREO; 8163 } 8164 8165 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8166 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8167 } else { 8168 mConfig.inputCfg.channels = channels; 8169 } 8170 mConfig.outputCfg.channels = channels; 8171 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8172 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8173 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8174 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8175 mConfig.inputCfg.bufferProvider.cookie = NULL; 8176 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8177 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8178 mConfig.outputCfg.bufferProvider.cookie = NULL; 8179 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8180 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8181 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8182 // Insert effect: 8183 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8184 // always overwrites output buffer: input buffer == output buffer 8185 // - in other sessions: 8186 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8187 // other effect: overwrites output buffer: input buffer == output buffer 8188 // Auxiliary effect: 8189 // accumulates in output buffer: input buffer != output buffer 8190 // Therefore: accumulate <=> input buffer != output buffer 8191 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8192 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8193 } else { 8194 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8195 } 8196 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8197 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8198 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8199 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8200 8201 ALOGV("configure() %p thread %p buffer %p framecount %d", 8202 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8203 8204 status_t cmdStatus; 8205 uint32_t size = sizeof(int); 8206 status_t status = (*mEffectInterface)->command(mEffectInterface, 8207 EFFECT_CMD_SET_CONFIG, 8208 sizeof(effect_config_t), 8209 &mConfig, 8210 &size, 8211 &cmdStatus); 8212 if (status == 0) { 8213 status = cmdStatus; 8214 } 8215 8216 if (status == 0 && 8217 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8218 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8219 effect_param_t *p = (effect_param_t *)buf32; 8220 8221 p->psize = sizeof(uint32_t); 8222 p->vsize = sizeof(uint32_t); 8223 size = sizeof(int); 8224 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8225 8226 uint32_t latency = 0; 8227 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8228 if (pbt != NULL) { 8229 latency = pbt->latency_l(); 8230 } 8231 8232 *((int32_t *)p->data + 1)= latency; 8233 (*mEffectInterface)->command(mEffectInterface, 8234 EFFECT_CMD_SET_PARAM, 8235 sizeof(effect_param_t) + 8, 8236 &buf32, 8237 &size, 8238 &cmdStatus); 8239 } 8240 8241 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8242 (1000 * mConfig.outputCfg.buffer.frameCount); 8243 8244 return status; 8245} 8246 8247status_t AudioFlinger::EffectModule::init() 8248{ 8249 Mutex::Autolock _l(mLock); 8250 if (mEffectInterface == NULL) { 8251 return NO_INIT; 8252 } 8253 status_t cmdStatus; 8254 uint32_t size = sizeof(status_t); 8255 status_t status = (*mEffectInterface)->command(mEffectInterface, 8256 EFFECT_CMD_INIT, 8257 0, 8258 NULL, 8259 &size, 8260 &cmdStatus); 8261 if (status == 0) { 8262 status = cmdStatus; 8263 } 8264 return status; 8265} 8266 8267status_t AudioFlinger::EffectModule::start() 8268{ 8269 Mutex::Autolock _l(mLock); 8270 return start_l(); 8271} 8272 8273status_t AudioFlinger::EffectModule::start_l() 8274{ 8275 if (mEffectInterface == NULL) { 8276 return NO_INIT; 8277 } 8278 status_t cmdStatus; 8279 uint32_t size = sizeof(status_t); 8280 status_t status = (*mEffectInterface)->command(mEffectInterface, 8281 EFFECT_CMD_ENABLE, 8282 0, 8283 NULL, 8284 &size, 8285 &cmdStatus); 8286 if (status == 0) { 8287 status = cmdStatus; 8288 } 8289 if (status == 0 && 8290 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8291 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8292 sp<ThreadBase> thread = mThread.promote(); 8293 if (thread != 0) { 8294 audio_stream_t *stream = thread->stream(); 8295 if (stream != NULL) { 8296 stream->add_audio_effect(stream, mEffectInterface); 8297 } 8298 } 8299 } 8300 return status; 8301} 8302 8303status_t AudioFlinger::EffectModule::stop() 8304{ 8305 Mutex::Autolock _l(mLock); 8306 return stop_l(); 8307} 8308 8309status_t AudioFlinger::EffectModule::stop_l() 8310{ 8311 if (mEffectInterface == NULL) { 8312 return NO_INIT; 8313 } 8314 status_t cmdStatus; 8315 uint32_t size = sizeof(status_t); 8316 status_t status = (*mEffectInterface)->command(mEffectInterface, 8317 EFFECT_CMD_DISABLE, 8318 0, 8319 NULL, 8320 &size, 8321 &cmdStatus); 8322 if (status == 0) { 8323 status = cmdStatus; 8324 } 8325 if (status == 0 && 8326 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8327 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8328 sp<ThreadBase> thread = mThread.promote(); 8329 if (thread != 0) { 8330 audio_stream_t *stream = thread->stream(); 8331 if (stream != NULL) { 8332 stream->remove_audio_effect(stream, mEffectInterface); 8333 } 8334 } 8335 } 8336 return status; 8337} 8338 8339status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8340 uint32_t cmdSize, 8341 void *pCmdData, 8342 uint32_t *replySize, 8343 void *pReplyData) 8344{ 8345 Mutex::Autolock _l(mLock); 8346// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8347 8348 if (mState == DESTROYED || mEffectInterface == NULL) { 8349 return NO_INIT; 8350 } 8351 status_t status = (*mEffectInterface)->command(mEffectInterface, 8352 cmdCode, 8353 cmdSize, 8354 pCmdData, 8355 replySize, 8356 pReplyData); 8357 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8358 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8359 for (size_t i = 1; i < mHandles.size(); i++) { 8360 sp<EffectHandle> h = mHandles[i].promote(); 8361 if (h != 0) { 8362 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8363 } 8364 } 8365 } 8366 return status; 8367} 8368 8369status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8370{ 8371 8372 Mutex::Autolock _l(mLock); 8373 ALOGV("setEnabled %p enabled %d", this, enabled); 8374 8375 if (enabled != isEnabled()) { 8376 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8377 if (enabled && status != NO_ERROR) { 8378 return status; 8379 } 8380 8381 switch (mState) { 8382 // going from disabled to enabled 8383 case IDLE: 8384 mState = STARTING; 8385 break; 8386 case STOPPED: 8387 mState = RESTART; 8388 break; 8389 case STOPPING: 8390 mState = ACTIVE; 8391 break; 8392 8393 // going from enabled to disabled 8394 case RESTART: 8395 mState = STOPPED; 8396 break; 8397 case STARTING: 8398 mState = IDLE; 8399 break; 8400 case ACTIVE: 8401 mState = STOPPING; 8402 break; 8403 case DESTROYED: 8404 return NO_ERROR; // simply ignore as we are being destroyed 8405 } 8406 for (size_t i = 1; i < mHandles.size(); i++) { 8407 sp<EffectHandle> h = mHandles[i].promote(); 8408 if (h != 0) { 8409 h->setEnabled(enabled); 8410 } 8411 } 8412 } 8413 return NO_ERROR; 8414} 8415 8416bool AudioFlinger::EffectModule::isEnabled() const 8417{ 8418 switch (mState) { 8419 case RESTART: 8420 case STARTING: 8421 case ACTIVE: 8422 return true; 8423 case IDLE: 8424 case STOPPING: 8425 case STOPPED: 8426 case DESTROYED: 8427 default: 8428 return false; 8429 } 8430} 8431 8432bool AudioFlinger::EffectModule::isProcessEnabled() const 8433{ 8434 switch (mState) { 8435 case RESTART: 8436 case ACTIVE: 8437 case STOPPING: 8438 case STOPPED: 8439 return true; 8440 case IDLE: 8441 case STARTING: 8442 case DESTROYED: 8443 default: 8444 return false; 8445 } 8446} 8447 8448status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8449{ 8450 Mutex::Autolock _l(mLock); 8451 status_t status = NO_ERROR; 8452 8453 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8454 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8455 if (isProcessEnabled() && 8456 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8457 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8458 status_t cmdStatus; 8459 uint32_t volume[2]; 8460 uint32_t *pVolume = NULL; 8461 uint32_t size = sizeof(volume); 8462 volume[0] = *left; 8463 volume[1] = *right; 8464 if (controller) { 8465 pVolume = volume; 8466 } 8467 status = (*mEffectInterface)->command(mEffectInterface, 8468 EFFECT_CMD_SET_VOLUME, 8469 size, 8470 volume, 8471 &size, 8472 pVolume); 8473 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8474 *left = volume[0]; 8475 *right = volume[1]; 8476 } 8477 } 8478 return status; 8479} 8480 8481status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8482{ 8483 Mutex::Autolock _l(mLock); 8484 status_t status = NO_ERROR; 8485 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8486 // audio pre processing modules on RecordThread can receive both output and 8487 // input device indication in the same call 8488 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8489 if (dev) { 8490 status_t cmdStatus; 8491 uint32_t size = sizeof(status_t); 8492 8493 status = (*mEffectInterface)->command(mEffectInterface, 8494 EFFECT_CMD_SET_DEVICE, 8495 sizeof(uint32_t), 8496 &dev, 8497 &size, 8498 &cmdStatus); 8499 if (status == NO_ERROR) { 8500 status = cmdStatus; 8501 } 8502 } 8503 dev = device & AUDIO_DEVICE_IN_ALL; 8504 if (dev) { 8505 status_t cmdStatus; 8506 uint32_t size = sizeof(status_t); 8507 8508 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8509 EFFECT_CMD_SET_INPUT_DEVICE, 8510 sizeof(uint32_t), 8511 &dev, 8512 &size, 8513 &cmdStatus); 8514 if (status2 == NO_ERROR) { 8515 status2 = cmdStatus; 8516 } 8517 if (status == NO_ERROR) { 8518 status = status2; 8519 } 8520 } 8521 } 8522 return status; 8523} 8524 8525status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8526{ 8527 Mutex::Autolock _l(mLock); 8528 status_t status = NO_ERROR; 8529 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8530 status_t cmdStatus; 8531 uint32_t size = sizeof(status_t); 8532 status = (*mEffectInterface)->command(mEffectInterface, 8533 EFFECT_CMD_SET_AUDIO_MODE, 8534 sizeof(audio_mode_t), 8535 &mode, 8536 &size, 8537 &cmdStatus); 8538 if (status == NO_ERROR) { 8539 status = cmdStatus; 8540 } 8541 } 8542 return status; 8543} 8544 8545void AudioFlinger::EffectModule::setSuspended(bool suspended) 8546{ 8547 Mutex::Autolock _l(mLock); 8548 mSuspended = suspended; 8549} 8550 8551bool AudioFlinger::EffectModule::suspended() const 8552{ 8553 Mutex::Autolock _l(mLock); 8554 return mSuspended; 8555} 8556 8557status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8558{ 8559 const size_t SIZE = 256; 8560 char buffer[SIZE]; 8561 String8 result; 8562 8563 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8564 result.append(buffer); 8565 8566 bool locked = tryLock(mLock); 8567 // failed to lock - AudioFlinger is probably deadlocked 8568 if (!locked) { 8569 result.append("\t\tCould not lock Fx mutex:\n"); 8570 } 8571 8572 result.append("\t\tSession Status State Engine:\n"); 8573 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8574 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8575 result.append(buffer); 8576 8577 result.append("\t\tDescriptor:\n"); 8578 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8579 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8580 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8581 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8582 result.append(buffer); 8583 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8584 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8585 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8586 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8587 result.append(buffer); 8588 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8589 mDescriptor.apiVersion, 8590 mDescriptor.flags); 8591 result.append(buffer); 8592 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8593 mDescriptor.name); 8594 result.append(buffer); 8595 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8596 mDescriptor.implementor); 8597 result.append(buffer); 8598 8599 result.append("\t\t- Input configuration:\n"); 8600 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8601 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8602 (uint32_t)mConfig.inputCfg.buffer.raw, 8603 mConfig.inputCfg.buffer.frameCount, 8604 mConfig.inputCfg.samplingRate, 8605 mConfig.inputCfg.channels, 8606 mConfig.inputCfg.format); 8607 result.append(buffer); 8608 8609 result.append("\t\t- Output configuration:\n"); 8610 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8611 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8612 (uint32_t)mConfig.outputCfg.buffer.raw, 8613 mConfig.outputCfg.buffer.frameCount, 8614 mConfig.outputCfg.samplingRate, 8615 mConfig.outputCfg.channels, 8616 mConfig.outputCfg.format); 8617 result.append(buffer); 8618 8619 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8620 result.append(buffer); 8621 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8622 for (size_t i = 0; i < mHandles.size(); ++i) { 8623 sp<EffectHandle> handle = mHandles[i].promote(); 8624 if (handle != 0) { 8625 handle->dump(buffer, SIZE); 8626 result.append(buffer); 8627 } 8628 } 8629 8630 result.append("\n"); 8631 8632 write(fd, result.string(), result.length()); 8633 8634 if (locked) { 8635 mLock.unlock(); 8636 } 8637 8638 return NO_ERROR; 8639} 8640 8641// ---------------------------------------------------------------------------- 8642// EffectHandle implementation 8643// ---------------------------------------------------------------------------- 8644 8645#undef LOG_TAG 8646#define LOG_TAG "AudioFlinger::EffectHandle" 8647 8648AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8649 const sp<AudioFlinger::Client>& client, 8650 const sp<IEffectClient>& effectClient, 8651 int32_t priority) 8652 : BnEffect(), 8653 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8654 mPriority(priority), mHasControl(false), mEnabled(false) 8655{ 8656 ALOGV("constructor %p", this); 8657 8658 if (client == 0) { 8659 return; 8660 } 8661 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8662 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8663 if (mCblkMemory != 0) { 8664 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8665 8666 if (mCblk != NULL) { 8667 new(mCblk) effect_param_cblk_t(); 8668 mBuffer = (uint8_t *)mCblk + bufOffset; 8669 } 8670 } else { 8671 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8672 return; 8673 } 8674} 8675 8676AudioFlinger::EffectHandle::~EffectHandle() 8677{ 8678 ALOGV("Destructor %p", this); 8679 disconnect(false); 8680 ALOGV("Destructor DONE %p", this); 8681} 8682 8683status_t AudioFlinger::EffectHandle::enable() 8684{ 8685 ALOGV("enable %p", this); 8686 if (!mHasControl) return INVALID_OPERATION; 8687 if (mEffect == 0) return DEAD_OBJECT; 8688 8689 if (mEnabled) { 8690 return NO_ERROR; 8691 } 8692 8693 mEnabled = true; 8694 8695 sp<ThreadBase> thread = mEffect->thread().promote(); 8696 if (thread != 0) { 8697 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8698 } 8699 8700 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8701 if (mEffect->suspended()) { 8702 return NO_ERROR; 8703 } 8704 8705 status_t status = mEffect->setEnabled(true); 8706 if (status != NO_ERROR) { 8707 if (thread != 0) { 8708 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8709 } 8710 mEnabled = false; 8711 } 8712 return status; 8713} 8714 8715status_t AudioFlinger::EffectHandle::disable() 8716{ 8717 ALOGV("disable %p", this); 8718 if (!mHasControl) return INVALID_OPERATION; 8719 if (mEffect == 0) return DEAD_OBJECT; 8720 8721 if (!mEnabled) { 8722 return NO_ERROR; 8723 } 8724 mEnabled = false; 8725 8726 if (mEffect->suspended()) { 8727 return NO_ERROR; 8728 } 8729 8730 status_t status = mEffect->setEnabled(false); 8731 8732 sp<ThreadBase> thread = mEffect->thread().promote(); 8733 if (thread != 0) { 8734 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8735 } 8736 8737 return status; 8738} 8739 8740void AudioFlinger::EffectHandle::disconnect() 8741{ 8742 disconnect(true); 8743} 8744 8745void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8746{ 8747 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8748 if (mEffect == 0) { 8749 return; 8750 } 8751 mEffect->disconnect(this, unpinIfLast); 8752 8753 if (mHasControl && mEnabled) { 8754 sp<ThreadBase> thread = mEffect->thread().promote(); 8755 if (thread != 0) { 8756 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8757 } 8758 } 8759 8760 // release sp on module => module destructor can be called now 8761 mEffect.clear(); 8762 if (mClient != 0) { 8763 if (mCblk != NULL) { 8764 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8765 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8766 } 8767 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8768 // Client destructor must run with AudioFlinger mutex locked 8769 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8770 mClient.clear(); 8771 } 8772} 8773 8774status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8775 uint32_t cmdSize, 8776 void *pCmdData, 8777 uint32_t *replySize, 8778 void *pReplyData) 8779{ 8780// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8781// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8782 8783 // only get parameter command is permitted for applications not controlling the effect 8784 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8785 return INVALID_OPERATION; 8786 } 8787 if (mEffect == 0) return DEAD_OBJECT; 8788 if (mClient == 0) return INVALID_OPERATION; 8789 8790 // handle commands that are not forwarded transparently to effect engine 8791 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8792 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8793 // no risk to block the whole media server process or mixer threads is we are stuck here 8794 Mutex::Autolock _l(mCblk->lock); 8795 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8796 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8797 mCblk->serverIndex = 0; 8798 mCblk->clientIndex = 0; 8799 return BAD_VALUE; 8800 } 8801 status_t status = NO_ERROR; 8802 while (mCblk->serverIndex < mCblk->clientIndex) { 8803 int reply; 8804 uint32_t rsize = sizeof(int); 8805 int *p = (int *)(mBuffer + mCblk->serverIndex); 8806 int size = *p++; 8807 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8808 ALOGW("command(): invalid parameter block size"); 8809 break; 8810 } 8811 effect_param_t *param = (effect_param_t *)p; 8812 if (param->psize == 0 || param->vsize == 0) { 8813 ALOGW("command(): null parameter or value size"); 8814 mCblk->serverIndex += size; 8815 continue; 8816 } 8817 uint32_t psize = sizeof(effect_param_t) + 8818 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8819 param->vsize; 8820 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8821 psize, 8822 p, 8823 &rsize, 8824 &reply); 8825 // stop at first error encountered 8826 if (ret != NO_ERROR) { 8827 status = ret; 8828 *(int *)pReplyData = reply; 8829 break; 8830 } else if (reply != NO_ERROR) { 8831 *(int *)pReplyData = reply; 8832 break; 8833 } 8834 mCblk->serverIndex += size; 8835 } 8836 mCblk->serverIndex = 0; 8837 mCblk->clientIndex = 0; 8838 return status; 8839 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8840 *(int *)pReplyData = NO_ERROR; 8841 return enable(); 8842 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8843 *(int *)pReplyData = NO_ERROR; 8844 return disable(); 8845 } 8846 8847 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8848} 8849 8850void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8851{ 8852 ALOGV("setControl %p control %d", this, hasControl); 8853 8854 mHasControl = hasControl; 8855 mEnabled = enabled; 8856 8857 if (signal && mEffectClient != 0) { 8858 mEffectClient->controlStatusChanged(hasControl); 8859 } 8860} 8861 8862void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8863 uint32_t cmdSize, 8864 void *pCmdData, 8865 uint32_t replySize, 8866 void *pReplyData) 8867{ 8868 if (mEffectClient != 0) { 8869 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8870 } 8871} 8872 8873 8874 8875void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8876{ 8877 if (mEffectClient != 0) { 8878 mEffectClient->enableStatusChanged(enabled); 8879 } 8880} 8881 8882status_t AudioFlinger::EffectHandle::onTransact( 8883 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8884{ 8885 return BnEffect::onTransact(code, data, reply, flags); 8886} 8887 8888 8889void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8890{ 8891 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8892 8893 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8894 (mClient == 0) ? getpid_cached : mClient->pid(), 8895 mPriority, 8896 mHasControl, 8897 !locked, 8898 mCblk ? mCblk->clientIndex : 0, 8899 mCblk ? mCblk->serverIndex : 0 8900 ); 8901 8902 if (locked) { 8903 mCblk->lock.unlock(); 8904 } 8905} 8906 8907#undef LOG_TAG 8908#define LOG_TAG "AudioFlinger::EffectChain" 8909 8910AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8911 int sessionId) 8912 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8913 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8914 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8915{ 8916 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8917 if (thread == NULL) { 8918 return; 8919 } 8920 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8921 thread->frameCount(); 8922} 8923 8924AudioFlinger::EffectChain::~EffectChain() 8925{ 8926 if (mOwnInBuffer) { 8927 delete mInBuffer; 8928 } 8929 8930} 8931 8932// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8933sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8934{ 8935 size_t size = mEffects.size(); 8936 8937 for (size_t i = 0; i < size; i++) { 8938 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8939 return mEffects[i]; 8940 } 8941 } 8942 return 0; 8943} 8944 8945// getEffectFromId_l() must be called with ThreadBase::mLock held 8946sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8947{ 8948 size_t size = mEffects.size(); 8949 8950 for (size_t i = 0; i < size; i++) { 8951 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8952 if (id == 0 || mEffects[i]->id() == id) { 8953 return mEffects[i]; 8954 } 8955 } 8956 return 0; 8957} 8958 8959// getEffectFromType_l() must be called with ThreadBase::mLock held 8960sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8961 const effect_uuid_t *type) 8962{ 8963 size_t size = mEffects.size(); 8964 8965 for (size_t i = 0; i < size; i++) { 8966 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8967 return mEffects[i]; 8968 } 8969 } 8970 return 0; 8971} 8972 8973void AudioFlinger::EffectChain::clearInputBuffer() 8974{ 8975 Mutex::Autolock _l(mLock); 8976 sp<ThreadBase> thread = mThread.promote(); 8977 if (thread == 0) { 8978 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 8979 return; 8980 } 8981 clearInputBuffer_l(thread); 8982} 8983 8984// Must be called with EffectChain::mLock locked 8985void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 8986{ 8987 size_t numSamples = thread->frameCount() * thread->channelCount(); 8988 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8989 8990} 8991 8992// Must be called with EffectChain::mLock locked 8993void AudioFlinger::EffectChain::process_l() 8994{ 8995 sp<ThreadBase> thread = mThread.promote(); 8996 if (thread == 0) { 8997 ALOGW("process_l(): cannot promote mixer thread"); 8998 return; 8999 } 9000 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9001 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9002 // always process effects unless no more tracks are on the session and the effect tail 9003 // has been rendered 9004 bool doProcess = true; 9005 if (!isGlobalSession) { 9006 bool tracksOnSession = (trackCnt() != 0); 9007 9008 if (!tracksOnSession && mTailBufferCount == 0) { 9009 doProcess = false; 9010 } 9011 9012 if (activeTrackCnt() == 0) { 9013 // if no track is active and the effect tail has not been rendered, 9014 // the input buffer must be cleared here as the mixer process will not do it 9015 if (tracksOnSession || mTailBufferCount > 0) { 9016 clearInputBuffer_l(thread); 9017 if (mTailBufferCount > 0) { 9018 mTailBufferCount--; 9019 } 9020 } 9021 } 9022 } 9023 9024 size_t size = mEffects.size(); 9025 if (doProcess) { 9026 for (size_t i = 0; i < size; i++) { 9027 mEffects[i]->process(); 9028 } 9029 } 9030 for (size_t i = 0; i < size; i++) { 9031 mEffects[i]->updateState(); 9032 } 9033} 9034 9035// addEffect_l() must be called with PlaybackThread::mLock held 9036status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9037{ 9038 effect_descriptor_t desc = effect->desc(); 9039 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9040 9041 Mutex::Autolock _l(mLock); 9042 effect->setChain(this); 9043 sp<ThreadBase> thread = mThread.promote(); 9044 if (thread == 0) { 9045 return NO_INIT; 9046 } 9047 effect->setThread(thread); 9048 9049 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9050 // Auxiliary effects are inserted at the beginning of mEffects vector as 9051 // they are processed first and accumulated in chain input buffer 9052 mEffects.insertAt(effect, 0); 9053 9054 // the input buffer for auxiliary effect contains mono samples in 9055 // 32 bit format. This is to avoid saturation in AudoMixer 9056 // accumulation stage. Saturation is done in EffectModule::process() before 9057 // calling the process in effect engine 9058 size_t numSamples = thread->frameCount(); 9059 int32_t *buffer = new int32_t[numSamples]; 9060 memset(buffer, 0, numSamples * sizeof(int32_t)); 9061 effect->setInBuffer((int16_t *)buffer); 9062 // auxiliary effects output samples to chain input buffer for further processing 9063 // by insert effects 9064 effect->setOutBuffer(mInBuffer); 9065 } else { 9066 // Insert effects are inserted at the end of mEffects vector as they are processed 9067 // after track and auxiliary effects. 9068 // Insert effect order as a function of indicated preference: 9069 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9070 // another effect is present 9071 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9072 // last effect claiming first position 9073 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9074 // first effect claiming last position 9075 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9076 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9077 // already present 9078 9079 size_t size = mEffects.size(); 9080 size_t idx_insert = size; 9081 ssize_t idx_insert_first = -1; 9082 ssize_t idx_insert_last = -1; 9083 9084 for (size_t i = 0; i < size; i++) { 9085 effect_descriptor_t d = mEffects[i]->desc(); 9086 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9087 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9088 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9089 // check invalid effect chaining combinations 9090 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9091 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9092 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9093 return INVALID_OPERATION; 9094 } 9095 // remember position of first insert effect and by default 9096 // select this as insert position for new effect 9097 if (idx_insert == size) { 9098 idx_insert = i; 9099 } 9100 // remember position of last insert effect claiming 9101 // first position 9102 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9103 idx_insert_first = i; 9104 } 9105 // remember position of first insert effect claiming 9106 // last position 9107 if (iPref == EFFECT_FLAG_INSERT_LAST && 9108 idx_insert_last == -1) { 9109 idx_insert_last = i; 9110 } 9111 } 9112 } 9113 9114 // modify idx_insert from first position if needed 9115 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9116 if (idx_insert_last != -1) { 9117 idx_insert = idx_insert_last; 9118 } else { 9119 idx_insert = size; 9120 } 9121 } else { 9122 if (idx_insert_first != -1) { 9123 idx_insert = idx_insert_first + 1; 9124 } 9125 } 9126 9127 // always read samples from chain input buffer 9128 effect->setInBuffer(mInBuffer); 9129 9130 // if last effect in the chain, output samples to chain 9131 // output buffer, otherwise to chain input buffer 9132 if (idx_insert == size) { 9133 if (idx_insert != 0) { 9134 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9135 mEffects[idx_insert-1]->configure(); 9136 } 9137 effect->setOutBuffer(mOutBuffer); 9138 } else { 9139 effect->setOutBuffer(mInBuffer); 9140 } 9141 mEffects.insertAt(effect, idx_insert); 9142 9143 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9144 } 9145 effect->configure(); 9146 return NO_ERROR; 9147} 9148 9149// removeEffect_l() must be called with PlaybackThread::mLock held 9150size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9151{ 9152 Mutex::Autolock _l(mLock); 9153 size_t size = mEffects.size(); 9154 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9155 9156 for (size_t i = 0; i < size; i++) { 9157 if (effect == mEffects[i]) { 9158 // calling stop here will remove pre-processing effect from the audio HAL. 9159 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9160 // the middle of a read from audio HAL 9161 if (mEffects[i]->state() == EffectModule::ACTIVE || 9162 mEffects[i]->state() == EffectModule::STOPPING) { 9163 mEffects[i]->stop(); 9164 } 9165 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9166 delete[] effect->inBuffer(); 9167 } else { 9168 if (i == size - 1 && i != 0) { 9169 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9170 mEffects[i - 1]->configure(); 9171 } 9172 } 9173 mEffects.removeAt(i); 9174 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9175 break; 9176 } 9177 } 9178 9179 return mEffects.size(); 9180} 9181 9182// setDevice_l() must be called with PlaybackThread::mLock held 9183void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9184{ 9185 size_t size = mEffects.size(); 9186 for (size_t i = 0; i < size; i++) { 9187 mEffects[i]->setDevice(device); 9188 } 9189} 9190 9191// setMode_l() must be called with PlaybackThread::mLock held 9192void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9193{ 9194 size_t size = mEffects.size(); 9195 for (size_t i = 0; i < size; i++) { 9196 mEffects[i]->setMode(mode); 9197 } 9198} 9199 9200// setVolume_l() must be called with PlaybackThread::mLock held 9201bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9202{ 9203 uint32_t newLeft = *left; 9204 uint32_t newRight = *right; 9205 bool hasControl = false; 9206 int ctrlIdx = -1; 9207 size_t size = mEffects.size(); 9208 9209 // first update volume controller 9210 for (size_t i = size; i > 0; i--) { 9211 if (mEffects[i - 1]->isProcessEnabled() && 9212 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9213 ctrlIdx = i - 1; 9214 hasControl = true; 9215 break; 9216 } 9217 } 9218 9219 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9220 if (hasControl) { 9221 *left = mNewLeftVolume; 9222 *right = mNewRightVolume; 9223 } 9224 return hasControl; 9225 } 9226 9227 mVolumeCtrlIdx = ctrlIdx; 9228 mLeftVolume = newLeft; 9229 mRightVolume = newRight; 9230 9231 // second get volume update from volume controller 9232 if (ctrlIdx >= 0) { 9233 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9234 mNewLeftVolume = newLeft; 9235 mNewRightVolume = newRight; 9236 } 9237 // then indicate volume to all other effects in chain. 9238 // Pass altered volume to effects before volume controller 9239 // and requested volume to effects after controller 9240 uint32_t lVol = newLeft; 9241 uint32_t rVol = newRight; 9242 9243 for (size_t i = 0; i < size; i++) { 9244 if ((int)i == ctrlIdx) continue; 9245 // this also works for ctrlIdx == -1 when there is no volume controller 9246 if ((int)i > ctrlIdx) { 9247 lVol = *left; 9248 rVol = *right; 9249 } 9250 mEffects[i]->setVolume(&lVol, &rVol, false); 9251 } 9252 *left = newLeft; 9253 *right = newRight; 9254 9255 return hasControl; 9256} 9257 9258status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9259{ 9260 const size_t SIZE = 256; 9261 char buffer[SIZE]; 9262 String8 result; 9263 9264 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9265 result.append(buffer); 9266 9267 bool locked = tryLock(mLock); 9268 // failed to lock - AudioFlinger is probably deadlocked 9269 if (!locked) { 9270 result.append("\tCould not lock mutex:\n"); 9271 } 9272 9273 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9274 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9275 mEffects.size(), 9276 (uint32_t)mInBuffer, 9277 (uint32_t)mOutBuffer, 9278 mActiveTrackCnt); 9279 result.append(buffer); 9280 write(fd, result.string(), result.size()); 9281 9282 for (size_t i = 0; i < mEffects.size(); ++i) { 9283 sp<EffectModule> effect = mEffects[i]; 9284 if (effect != 0) { 9285 effect->dump(fd, args); 9286 } 9287 } 9288 9289 if (locked) { 9290 mLock.unlock(); 9291 } 9292 9293 return NO_ERROR; 9294} 9295 9296// must be called with ThreadBase::mLock held 9297void AudioFlinger::EffectChain::setEffectSuspended_l( 9298 const effect_uuid_t *type, bool suspend) 9299{ 9300 sp<SuspendedEffectDesc> desc; 9301 // use effect type UUID timelow as key as there is no real risk of identical 9302 // timeLow fields among effect type UUIDs. 9303 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9304 if (suspend) { 9305 if (index >= 0) { 9306 desc = mSuspendedEffects.valueAt(index); 9307 } else { 9308 desc = new SuspendedEffectDesc(); 9309 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9310 mSuspendedEffects.add(type->timeLow, desc); 9311 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9312 } 9313 if (desc->mRefCount++ == 0) { 9314 sp<EffectModule> effect = getEffectIfEnabled(type); 9315 if (effect != 0) { 9316 desc->mEffect = effect; 9317 effect->setSuspended(true); 9318 effect->setEnabled(false); 9319 } 9320 } 9321 } else { 9322 if (index < 0) { 9323 return; 9324 } 9325 desc = mSuspendedEffects.valueAt(index); 9326 if (desc->mRefCount <= 0) { 9327 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9328 desc->mRefCount = 1; 9329 } 9330 if (--desc->mRefCount == 0) { 9331 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9332 if (desc->mEffect != 0) { 9333 sp<EffectModule> effect = desc->mEffect.promote(); 9334 if (effect != 0) { 9335 effect->setSuspended(false); 9336 sp<EffectHandle> handle = effect->controlHandle(); 9337 if (handle != 0) { 9338 effect->setEnabled(handle->enabled()); 9339 } 9340 } 9341 desc->mEffect.clear(); 9342 } 9343 mSuspendedEffects.removeItemsAt(index); 9344 } 9345 } 9346} 9347 9348// must be called with ThreadBase::mLock held 9349void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9350{ 9351 sp<SuspendedEffectDesc> desc; 9352 9353 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9354 if (suspend) { 9355 if (index >= 0) { 9356 desc = mSuspendedEffects.valueAt(index); 9357 } else { 9358 desc = new SuspendedEffectDesc(); 9359 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9360 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9361 } 9362 if (desc->mRefCount++ == 0) { 9363 Vector< sp<EffectModule> > effects; 9364 getSuspendEligibleEffects(effects); 9365 for (size_t i = 0; i < effects.size(); i++) { 9366 setEffectSuspended_l(&effects[i]->desc().type, true); 9367 } 9368 } 9369 } else { 9370 if (index < 0) { 9371 return; 9372 } 9373 desc = mSuspendedEffects.valueAt(index); 9374 if (desc->mRefCount <= 0) { 9375 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9376 desc->mRefCount = 1; 9377 } 9378 if (--desc->mRefCount == 0) { 9379 Vector<const effect_uuid_t *> types; 9380 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9381 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9382 continue; 9383 } 9384 types.add(&mSuspendedEffects.valueAt(i)->mType); 9385 } 9386 for (size_t i = 0; i < types.size(); i++) { 9387 setEffectSuspended_l(types[i], false); 9388 } 9389 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9390 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9391 } 9392 } 9393} 9394 9395 9396// The volume effect is used for automated tests only 9397#ifndef OPENSL_ES_H_ 9398static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9399 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9400const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9401#endif //OPENSL_ES_H_ 9402 9403bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9404{ 9405 // auxiliary effects and visualizer are never suspended on output mix 9406 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9407 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9408 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9409 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9410 return false; 9411 } 9412 return true; 9413} 9414 9415void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9416{ 9417 effects.clear(); 9418 for (size_t i = 0; i < mEffects.size(); i++) { 9419 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9420 effects.add(mEffects[i]); 9421 } 9422 } 9423} 9424 9425sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9426 const effect_uuid_t *type) 9427{ 9428 sp<EffectModule> effect = getEffectFromType_l(type); 9429 return effect != 0 && effect->isEnabled() ? effect : 0; 9430} 9431 9432void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9433 bool enabled) 9434{ 9435 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9436 if (enabled) { 9437 if (index < 0) { 9438 // if the effect is not suspend check if all effects are suspended 9439 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9440 if (index < 0) { 9441 return; 9442 } 9443 if (!isEffectEligibleForSuspend(effect->desc())) { 9444 return; 9445 } 9446 setEffectSuspended_l(&effect->desc().type, enabled); 9447 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9448 if (index < 0) { 9449 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9450 return; 9451 } 9452 } 9453 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9454 effect->desc().type.timeLow); 9455 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9456 // if effect is requested to suspended but was not yet enabled, supend it now. 9457 if (desc->mEffect == 0) { 9458 desc->mEffect = effect; 9459 effect->setEnabled(false); 9460 effect->setSuspended(true); 9461 } 9462 } else { 9463 if (index < 0) { 9464 return; 9465 } 9466 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9467 effect->desc().type.timeLow); 9468 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9469 desc->mEffect.clear(); 9470 effect->setSuspended(false); 9471 } 9472} 9473 9474#undef LOG_TAG 9475#define LOG_TAG "AudioFlinger" 9476 9477// ---------------------------------------------------------------------------- 9478 9479status_t AudioFlinger::onTransact( 9480 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9481{ 9482 return BnAudioFlinger::onTransact(code, data, reply, flags); 9483} 9484 9485}; // namespace android 9486