AudioFlinger.cpp revision d8f178d613821c3f61a5c5e391eb275339e526a9
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#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// ---------------------------------------------------------------------------- 169 170#ifdef ADD_BATTERY_DATA 171// To collect the amplifier usage 172static void addBatteryData(uint32_t params) { 173 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 174 if (service == NULL) { 175 // it already logged 176 return; 177 } 178 179 service->addBatteryData(params); 180} 181#endif 182 183static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 184{ 185 const hw_module_t *mod; 186 int rc; 187 188 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 189 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 rc = audio_hw_device_open(mod, dev); 195 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 196 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 197 if (rc) { 198 goto out; 199 } 200 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 201 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 202 rc = BAD_VALUE; 203 goto out; 204 } 205 return 0; 206 207out: 208 *dev = NULL; 209 return rc; 210} 211 212// ---------------------------------------------------------------------------- 213 214AudioFlinger::AudioFlinger() 215 : BnAudioFlinger(), 216 mPrimaryHardwareDev(NULL), 217 mHardwareStatus(AUDIO_HW_IDLE), 218 mMasterVolume(1.0f), 219 mMasterVolumeSW(1.0f), 220 mMasterVolumeSupportLvl(MVS_NONE), 221 mMasterMute(false), 222 mMasterMuteSW(false), 223 mMasterMuteSupportLvl(MMS_NONE), 224 mNextUniqueId(1), 225 mMode(AUDIO_MODE_INVALID), 226 mBtNrecIsOff(false) 227{ 228} 229 230void AudioFlinger::onFirstRef() 231{ 232 int rc = 0; 233 234 Mutex::Autolock _l(mLock); 235 236 /* TODO: move all this work into an Init() function */ 237 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 238 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 239 uint32_t int_val; 240 if (1 == sscanf(val_str, "%u", &int_val)) { 241 mStandbyTimeInNsecs = milliseconds(int_val); 242 ALOGI("Using %u mSec as standby time.", int_val); 243 } else { 244 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 245 ALOGI("Using default %u mSec as standby time.", 246 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 247 } 248 } 249 250 mMode = AUDIO_MODE_NORMAL; 251} 252 253AudioFlinger::~AudioFlinger() 254{ 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327} 328 329 330void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 331{ 332 const size_t SIZE = 256; 333 char buffer[SIZE]; 334 String8 result; 335 hardware_call_state hardwareStatus = mHardwareStatus; 336 337 snprintf(buffer, SIZE, "Hardware status: %d\n" 338 "Standby Time mSec: %u\n", 339 hardwareStatus, 340 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 341 result.append(buffer); 342 write(fd, result.string(), result.size()); 343} 344 345void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 snprintf(buffer, SIZE, "Permission Denial: " 351 "can't dump AudioFlinger from pid=%d, uid=%d\n", 352 IPCThreadState::self()->getCallingPid(), 353 IPCThreadState::self()->getCallingUid()); 354 result.append(buffer); 355 write(fd, result.string(), result.size()); 356} 357 358static bool tryLock(Mutex& mutex) 359{ 360 bool locked = false; 361 for (int i = 0; i < kDumpLockRetries; ++i) { 362 if (mutex.tryLock() == NO_ERROR) { 363 locked = true; 364 break; 365 } 366 usleep(kDumpLockSleepUs); 367 } 368 return locked; 369} 370 371status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 372{ 373 if (!dumpAllowed()) { 374 dumpPermissionDenial(fd, args); 375 } else { 376 // get state of hardware lock 377 bool hardwareLocked = tryLock(mHardwareLock); 378 if (!hardwareLocked) { 379 String8 result(kHardwareLockedString); 380 write(fd, result.string(), result.size()); 381 } else { 382 mHardwareLock.unlock(); 383 } 384 385 bool locked = tryLock(mLock); 386 387 // failed to lock - AudioFlinger is probably deadlocked 388 if (!locked) { 389 String8 result(kDeadlockedString); 390 write(fd, result.string(), result.size()); 391 } 392 393 dumpClients(fd, args); 394 dumpInternals(fd, args); 395 396 // dump playback threads 397 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 398 mPlaybackThreads.valueAt(i)->dump(fd, args); 399 } 400 401 // dump record threads 402 for (size_t i = 0; i < mRecordThreads.size(); i++) { 403 mRecordThreads.valueAt(i)->dump(fd, args); 404 } 405 406 // dump all hardware devs 407 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 408 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 409 dev->dump(dev, fd); 410 } 411 if (locked) mLock.unlock(); 412 } 413 return NO_ERROR; 414} 415 416sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 417{ 418 // If pid is already in the mClients wp<> map, then use that entry 419 // (for which promote() is always != 0), otherwise create a new entry and Client. 420 sp<Client> client = mClients.valueFor(pid).promote(); 421 if (client == 0) { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 return client; 427} 428 429// IAudioFlinger interface 430 431 432sp<IAudioTrack> AudioFlinger::createTrack( 433 pid_t pid, 434 audio_stream_type_t streamType, 435 uint32_t sampleRate, 436 audio_format_t format, 437 audio_channel_mask_t channelMask, 438 int frameCount, 439 IAudioFlinger::track_flags_t flags, 440 const sp<IMemory>& sharedBuffer, 441 audio_io_handle_t output, 442 pid_t tid, 443 int *sessionId, 444 status_t *status) 445{ 446 sp<PlaybackThread::Track> track; 447 sp<TrackHandle> trackHandle; 448 sp<Client> client; 449 status_t lStatus; 450 int lSessionId; 451 452 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 453 // but if someone uses binder directly they could bypass that and cause us to crash 454 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 455 ALOGE("createTrack() invalid stream type %d", streamType); 456 lStatus = BAD_VALUE; 457 goto Exit; 458 } 459 460 { 461 Mutex::Autolock _l(mLock); 462 PlaybackThread *thread = checkPlaybackThread_l(output); 463 PlaybackThread *effectThread = NULL; 464 if (thread == NULL) { 465 ALOGE("unknown output thread"); 466 lStatus = BAD_VALUE; 467 goto Exit; 468 } 469 470 client = registerPid_l(pid); 471 472 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 473 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 474 // check if an effect chain with the same session ID is present on another 475 // output thread and move it here. 476 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 477 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 478 if (mPlaybackThreads.keyAt(i) != output) { 479 uint32_t sessions = t->hasAudioSession(*sessionId); 480 if (sessions & PlaybackThread::EFFECT_SESSION) { 481 effectThread = t.get(); 482 break; 483 } 484 } 485 } 486 lSessionId = *sessionId; 487 } else { 488 // if no audio session id is provided, create one here 489 lSessionId = nextUniqueId(); 490 if (sessionId != NULL) { 491 *sessionId = lSessionId; 492 } 493 } 494 ALOGV("createTrack() lSessionId: %d", lSessionId); 495 496 track = thread->createTrack_l(client, streamType, sampleRate, format, 497 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 498 499 // move effect chain to this output thread if an effect on same session was waiting 500 // for a track to be created 501 if (lStatus == NO_ERROR && effectThread != NULL) { 502 Mutex::Autolock _dl(thread->mLock); 503 Mutex::Autolock _sl(effectThread->mLock); 504 moveEffectChain_l(lSessionId, effectThread, thread, true); 505 } 506 507 // Look for sync events awaiting for a session to be used. 508 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 509 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 510 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 511 if (lStatus == NO_ERROR) { 512 track->setSyncEvent(mPendingSyncEvents[i]); 513 } else { 514 mPendingSyncEvents[i]->cancel(); 515 } 516 mPendingSyncEvents.removeAt(i); 517 i--; 518 } 519 } 520 } 521 } 522 if (lStatus == NO_ERROR) { 523 trackHandle = new TrackHandle(track); 524 } else { 525 // remove local strong reference to Client before deleting the Track so that the Client 526 // destructor is called by the TrackBase destructor with mLock held 527 client.clear(); 528 track.clear(); 529 } 530 531Exit: 532 if (status != NULL) { 533 *status = lStatus; 534 } 535 return trackHandle; 536} 537 538uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 539{ 540 Mutex::Autolock _l(mLock); 541 PlaybackThread *thread = checkPlaybackThread_l(output); 542 if (thread == NULL) { 543 ALOGW("sampleRate() unknown thread %d", output); 544 return 0; 545 } 546 return thread->sampleRate(); 547} 548 549int AudioFlinger::channelCount(audio_io_handle_t output) const 550{ 551 Mutex::Autolock _l(mLock); 552 PlaybackThread *thread = checkPlaybackThread_l(output); 553 if (thread == NULL) { 554 ALOGW("channelCount() unknown thread %d", output); 555 return 0; 556 } 557 return thread->channelCount(); 558} 559 560audio_format_t AudioFlinger::format(audio_io_handle_t output) const 561{ 562 Mutex::Autolock _l(mLock); 563 PlaybackThread *thread = checkPlaybackThread_l(output); 564 if (thread == NULL) { 565 ALOGW("format() unknown thread %d", output); 566 return AUDIO_FORMAT_INVALID; 567 } 568 return thread->format(); 569} 570 571size_t AudioFlinger::frameCount(audio_io_handle_t output) const 572{ 573 Mutex::Autolock _l(mLock); 574 PlaybackThread *thread = checkPlaybackThread_l(output); 575 if (thread == NULL) { 576 ALOGW("frameCount() unknown thread %d", output); 577 return 0; 578 } 579 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 580 // should examine all callers and fix them to handle smaller counts 581 return thread->frameCount(); 582} 583 584uint32_t AudioFlinger::latency(audio_io_handle_t output) const 585{ 586 Mutex::Autolock _l(mLock); 587 PlaybackThread *thread = checkPlaybackThread_l(output); 588 if (thread == NULL) { 589 ALOGW("latency() unknown thread %d", output); 590 return 0; 591 } 592 return thread->latency(); 593} 594 595status_t AudioFlinger::setMasterVolume(float value) 596{ 597 status_t ret = initCheck(); 598 if (ret != NO_ERROR) { 599 return ret; 600 } 601 602 // check calling permissions 603 if (!settingsAllowed()) { 604 return PERMISSION_DENIED; 605 } 606 607 float swmv = value; 608 609 Mutex::Autolock _l(mLock); 610 611 // when hw supports master volume, don't scale in sw mixer 612 if (MVS_NONE != mMasterVolumeSupportLvl) { 613 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 614 AutoMutex lock(mHardwareLock); 615 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 616 617 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 618 if (NULL != dev->set_master_volume) { 619 dev->set_master_volume(dev, value); 620 } 621 mHardwareStatus = AUDIO_HW_IDLE; 622 } 623 624 swmv = 1.0; 625 } 626 627 mMasterVolume = value; 628 mMasterVolumeSW = swmv; 629 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 630 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 631 632 return NO_ERROR; 633} 634 635status_t AudioFlinger::setMode(audio_mode_t mode) 636{ 637 status_t ret = initCheck(); 638 if (ret != NO_ERROR) { 639 return ret; 640 } 641 642 // check calling permissions 643 if (!settingsAllowed()) { 644 return PERMISSION_DENIED; 645 } 646 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 647 ALOGW("Illegal value: setMode(%d)", mode); 648 return BAD_VALUE; 649 } 650 651 { // scope for the lock 652 AutoMutex lock(mHardwareLock); 653 mHardwareStatus = AUDIO_HW_SET_MODE; 654 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 655 mHardwareStatus = AUDIO_HW_IDLE; 656 } 657 658 if (NO_ERROR == ret) { 659 Mutex::Autolock _l(mLock); 660 mMode = mode; 661 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 662 mPlaybackThreads.valueAt(i)->setMode(mode); 663 } 664 665 return ret; 666} 667 668status_t AudioFlinger::setMicMute(bool state) 669{ 670 status_t ret = initCheck(); 671 if (ret != NO_ERROR) { 672 return ret; 673 } 674 675 // check calling permissions 676 if (!settingsAllowed()) { 677 return PERMISSION_DENIED; 678 } 679 680 AutoMutex lock(mHardwareLock); 681 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 682 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 683 mHardwareStatus = AUDIO_HW_IDLE; 684 return ret; 685} 686 687bool AudioFlinger::getMicMute() const 688{ 689 status_t ret = initCheck(); 690 if (ret != NO_ERROR) { 691 return false; 692 } 693 694 bool state = AUDIO_MODE_INVALID; 695 AutoMutex lock(mHardwareLock); 696 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 697 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 698 mHardwareStatus = AUDIO_HW_IDLE; 699 return state; 700} 701 702status_t AudioFlinger::setMasterMute(bool muted) 703{ 704 status_t ret = initCheck(); 705 if (ret != NO_ERROR) { 706 return ret; 707 } 708 709 // check calling permissions 710 if (!settingsAllowed()) { 711 return PERMISSION_DENIED; 712 } 713 714 bool swmm = muted; 715 716 // when hw supports master mute, don't mute in sw mixer 717 if (MMS_NONE != mMasterMuteSupportLvl) { 718 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 719 AutoMutex lock(mHardwareLock); 720 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 721 722 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 723 if (NULL != dev->set_master_mute) { 724 dev->set_master_mute(dev, muted); 725 } 726 mHardwareStatus = AUDIO_HW_IDLE; 727 } 728 729 swmm = false; 730 } 731 732 Mutex::Autolock _l(mLock); 733 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 734 mMasterMute = muted; 735 mMasterMuteSW = swmm; 736 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 737 mPlaybackThreads.valueAt(i)->setMasterMute(swmm); 738 739 return NO_ERROR; 740} 741 742float AudioFlinger::masterVolume() const 743{ 744 Mutex::Autolock _l(mLock); 745 return masterVolume_l(); 746} 747 748float AudioFlinger::masterVolumeSW() const 749{ 750 Mutex::Autolock _l(mLock); 751 return masterVolumeSW_l(); 752} 753 754bool AudioFlinger::masterMute() const 755{ 756 Mutex::Autolock _l(mLock); 757 return masterMute_l(); 758} 759 760bool AudioFlinger::masterMuteSW() const 761{ 762 Mutex::Autolock _l(mLock); 763 return masterMuteSW_l(); 764} 765 766float AudioFlinger::masterVolume_l() const 767{ 768 if (MVS_FULL == mMasterVolumeSupportLvl) { 769 float ret_val; 770 AutoMutex lock(mHardwareLock); 771 772 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 773 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 774 (NULL != mPrimaryHardwareDev->get_master_volume), 775 "can't get master volume"); 776 777 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 778 mHardwareStatus = AUDIO_HW_IDLE; 779 return ret_val; 780 } 781 782 return mMasterVolume; 783} 784 785bool AudioFlinger::masterMute_l() const 786{ 787 if (MMS_FULL == mMasterMuteSupportLvl) { 788 bool ret_val; 789 AutoMutex lock(mHardwareLock); 790 791 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 792 assert(NULL != mPrimaryHardwareDev); 793 assert(NULL != mPrimaryHardwareDev->get_master_mute); 794 795 mPrimaryHardwareDev->get_master_mute(mPrimaryHardwareDev, &ret_val); 796 mHardwareStatus = AUDIO_HW_IDLE; 797 return ret_val; 798 } 799 800 return mMasterMute; 801} 802 803status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 804 audio_io_handle_t output) 805{ 806 // check calling permissions 807 if (!settingsAllowed()) { 808 return PERMISSION_DENIED; 809 } 810 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 ALOGE("setStreamVolume() invalid stream %d", stream); 813 return BAD_VALUE; 814 } 815 816 AutoMutex lock(mLock); 817 PlaybackThread *thread = NULL; 818 if (output) { 819 thread = checkPlaybackThread_l(output); 820 if (thread == NULL) { 821 return BAD_VALUE; 822 } 823 } 824 825 mStreamTypes[stream].volume = value; 826 827 if (thread == NULL) { 828 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 829 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 830 } 831 } else { 832 thread->setStreamVolume(stream, value); 833 } 834 835 return NO_ERROR; 836} 837 838status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 839{ 840 // check calling permissions 841 if (!settingsAllowed()) { 842 return PERMISSION_DENIED; 843 } 844 845 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 846 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 847 ALOGE("setStreamMute() invalid stream %d", stream); 848 return BAD_VALUE; 849 } 850 851 AutoMutex lock(mLock); 852 mStreamTypes[stream].mute = muted; 853 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 854 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 855 856 return NO_ERROR; 857} 858 859float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 860{ 861 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 862 return 0.0f; 863 } 864 865 AutoMutex lock(mLock); 866 float volume; 867 if (output) { 868 PlaybackThread *thread = checkPlaybackThread_l(output); 869 if (thread == NULL) { 870 return 0.0f; 871 } 872 volume = thread->streamVolume(stream); 873 } else { 874 volume = streamVolume_l(stream); 875 } 876 877 return volume; 878} 879 880bool AudioFlinger::streamMute(audio_stream_type_t stream) const 881{ 882 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 883 return true; 884 } 885 886 AutoMutex lock(mLock); 887 return streamMute_l(stream); 888} 889 890status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 891{ 892 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 893 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 894 // check calling permissions 895 if (!settingsAllowed()) { 896 return PERMISSION_DENIED; 897 } 898 899 // ioHandle == 0 means the parameters are global to the audio hardware interface 900 if (ioHandle == 0) { 901 Mutex::Autolock _l(mLock); 902 status_t final_result = NO_ERROR; 903 { 904 AutoMutex lock(mHardwareLock); 905 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 906 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 907 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 908 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 909 final_result = result ?: final_result; 910 } 911 mHardwareStatus = AUDIO_HW_IDLE; 912 } 913 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 914 AudioParameter param = AudioParameter(keyValuePairs); 915 String8 value; 916 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 917 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 918 if (mBtNrecIsOff != btNrecIsOff) { 919 for (size_t i = 0; i < mRecordThreads.size(); i++) { 920 sp<RecordThread> thread = mRecordThreads.valueAt(i); 921 audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL; 922 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 923 // collect all of the thread's session IDs 924 KeyedVector<int, bool> ids = thread->sessionIds(); 925 // suspend effects associated with those session IDs 926 for (size_t j = 0; j < ids.size(); ++j) { 927 int sessionId = ids.keyAt(j); 928 thread->setEffectSuspended(FX_IID_AEC, 929 suspend, 930 sessionId); 931 thread->setEffectSuspended(FX_IID_NS, 932 suspend, 933 sessionId); 934 } 935 } 936 mBtNrecIsOff = btNrecIsOff; 937 } 938 } 939 String8 screenState; 940 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 941 bool isOff = screenState == "off"; 942 if (isOff != (gScreenState & 1)) { 943 gScreenState = ((gScreenState & ~1) + 2) | isOff; 944 } 945 } 946 return final_result; 947 } 948 949 // hold a strong ref on thread in case closeOutput() or closeInput() is called 950 // and the thread is exited once the lock is released 951 sp<ThreadBase> thread; 952 { 953 Mutex::Autolock _l(mLock); 954 thread = checkPlaybackThread_l(ioHandle); 955 if (thread == 0) { 956 thread = checkRecordThread_l(ioHandle); 957 } else if (thread == primaryPlaybackThread_l()) { 958 // indicate output device change to all input threads for pre processing 959 AudioParameter param = AudioParameter(keyValuePairs); 960 int value; 961 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 962 (value != 0)) { 963 for (size_t i = 0; i < mRecordThreads.size(); i++) { 964 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 965 } 966 } 967 } 968 } 969 if (thread != 0) { 970 return thread->setParameters(keyValuePairs); 971 } 972 return BAD_VALUE; 973} 974 975String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 976{ 977// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 978// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 979 980 Mutex::Autolock _l(mLock); 981 982 if (ioHandle == 0) { 983 String8 out_s8; 984 985 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 986 char *s; 987 { 988 AutoMutex lock(mHardwareLock); 989 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 990 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 991 s = dev->get_parameters(dev, keys.string()); 992 mHardwareStatus = AUDIO_HW_IDLE; 993 } 994 out_s8 += String8(s ? s : ""); 995 free(s); 996 } 997 return out_s8; 998 } 999 1000 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1001 if (playbackThread != NULL) { 1002 return playbackThread->getParameters(keys); 1003 } 1004 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1005 if (recordThread != NULL) { 1006 return recordThread->getParameters(keys); 1007 } 1008 return String8(""); 1009} 1010 1011size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1012 audio_channel_mask_t channelMask) const 1013{ 1014 status_t ret = initCheck(); 1015 if (ret != NO_ERROR) { 1016 return 0; 1017 } 1018 1019 AutoMutex lock(mHardwareLock); 1020 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1021 struct audio_config config = { 1022 sample_rate: sampleRate, 1023 channel_mask: channelMask, 1024 format: format, 1025 }; 1026 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 1027 mHardwareStatus = AUDIO_HW_IDLE; 1028 return size; 1029} 1030 1031unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1032{ 1033 Mutex::Autolock _l(mLock); 1034 1035 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1036 if (recordThread != NULL) { 1037 return recordThread->getInputFramesLost(); 1038 } 1039 return 0; 1040} 1041 1042status_t AudioFlinger::setVoiceVolume(float value) 1043{ 1044 status_t ret = initCheck(); 1045 if (ret != NO_ERROR) { 1046 return ret; 1047 } 1048 1049 // check calling permissions 1050 if (!settingsAllowed()) { 1051 return PERMISSION_DENIED; 1052 } 1053 1054 AutoMutex lock(mHardwareLock); 1055 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1056 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1057 mHardwareStatus = AUDIO_HW_IDLE; 1058 1059 return ret; 1060} 1061 1062status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1063 audio_io_handle_t output) const 1064{ 1065 status_t status; 1066 1067 Mutex::Autolock _l(mLock); 1068 1069 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1070 if (playbackThread != NULL) { 1071 return playbackThread->getRenderPosition(halFrames, dspFrames); 1072 } 1073 1074 return BAD_VALUE; 1075} 1076 1077void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1078{ 1079 1080 Mutex::Autolock _l(mLock); 1081 1082 pid_t pid = IPCThreadState::self()->getCallingPid(); 1083 if (mNotificationClients.indexOfKey(pid) < 0) { 1084 sp<NotificationClient> notificationClient = new NotificationClient(this, 1085 client, 1086 pid); 1087 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1088 1089 mNotificationClients.add(pid, notificationClient); 1090 1091 sp<IBinder> binder = client->asBinder(); 1092 binder->linkToDeath(notificationClient); 1093 1094 // the config change is always sent from playback or record threads to avoid deadlock 1095 // with AudioSystem::gLock 1096 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1097 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1098 } 1099 1100 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1101 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1102 } 1103 } 1104} 1105 1106void AudioFlinger::removeNotificationClient(pid_t pid) 1107{ 1108 Mutex::Autolock _l(mLock); 1109 1110 mNotificationClients.removeItem(pid); 1111 1112 ALOGV("%d died, releasing its sessions", pid); 1113 size_t num = mAudioSessionRefs.size(); 1114 bool removed = false; 1115 for (size_t i = 0; i< num; ) { 1116 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1117 ALOGV(" pid %d @ %d", ref->mPid, i); 1118 if (ref->mPid == pid) { 1119 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1120 mAudioSessionRefs.removeAt(i); 1121 delete ref; 1122 removed = true; 1123 num--; 1124 } else { 1125 i++; 1126 } 1127 } 1128 if (removed) { 1129 purgeStaleEffects_l(); 1130 } 1131} 1132 1133// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1134void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1135{ 1136 size_t size = mNotificationClients.size(); 1137 for (size_t i = 0; i < size; i++) { 1138 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1139 param2); 1140 } 1141} 1142 1143// removeClient_l() must be called with AudioFlinger::mLock held 1144void AudioFlinger::removeClient_l(pid_t pid) 1145{ 1146 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1147 mClients.removeItem(pid); 1148} 1149 1150// getEffectThread_l() must be called with AudioFlinger::mLock held 1151sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1152{ 1153 sp<PlaybackThread> thread; 1154 1155 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1156 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1157 ALOG_ASSERT(thread == 0); 1158 thread = mPlaybackThreads.valueAt(i); 1159 } 1160 } 1161 1162 return thread; 1163} 1164 1165// ---------------------------------------------------------------------------- 1166 1167AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1168 audio_devices_t device, type_t type) 1169 : Thread(false), 1170 mType(type), 1171 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1172 // mChannelMask 1173 mChannelCount(0), 1174 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1175 mParamStatus(NO_ERROR), 1176 mStandby(false), mDevice(device), mId(id), 1177 mDeathRecipient(new PMDeathRecipient(this)) 1178{ 1179} 1180 1181AudioFlinger::ThreadBase::~ThreadBase() 1182{ 1183 mParamCond.broadcast(); 1184 // do not lock the mutex in destructor 1185 releaseWakeLock_l(); 1186 if (mPowerManager != 0) { 1187 sp<IBinder> binder = mPowerManager->asBinder(); 1188 binder->unlinkToDeath(mDeathRecipient); 1189 } 1190} 1191 1192void AudioFlinger::ThreadBase::exit() 1193{ 1194 ALOGV("ThreadBase::exit"); 1195 { 1196 // This lock prevents the following race in thread (uniprocessor for illustration): 1197 // if (!exitPending()) { 1198 // // context switch from here to exit() 1199 // // exit() calls requestExit(), what exitPending() observes 1200 // // exit() calls signal(), which is dropped since no waiters 1201 // // context switch back from exit() to here 1202 // mWaitWorkCV.wait(...); 1203 // // now thread is hung 1204 // } 1205 AutoMutex lock(mLock); 1206 requestExit(); 1207 mWaitWorkCV.signal(); 1208 } 1209 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1210 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1211 requestExitAndWait(); 1212} 1213 1214status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1215{ 1216 status_t status; 1217 1218 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1219 Mutex::Autolock _l(mLock); 1220 1221 mNewParameters.add(keyValuePairs); 1222 mWaitWorkCV.signal(); 1223 // wait condition with timeout in case the thread loop has exited 1224 // before the request could be processed 1225 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1226 status = mParamStatus; 1227 mWaitWorkCV.signal(); 1228 } else { 1229 status = TIMED_OUT; 1230 } 1231 return status; 1232} 1233 1234void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1235{ 1236 Mutex::Autolock _l(mLock); 1237 sendConfigEvent_l(event, param); 1238} 1239 1240// sendConfigEvent_l() must be called with ThreadBase::mLock held 1241void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1242{ 1243 ConfigEvent configEvent; 1244 configEvent.mEvent = event; 1245 configEvent.mParam = param; 1246 mConfigEvents.add(configEvent); 1247 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1248 mWaitWorkCV.signal(); 1249} 1250 1251void AudioFlinger::ThreadBase::processConfigEvents() 1252{ 1253 mLock.lock(); 1254 while (!mConfigEvents.isEmpty()) { 1255 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1256 ConfigEvent configEvent = mConfigEvents[0]; 1257 mConfigEvents.removeAt(0); 1258 // release mLock before locking AudioFlinger mLock: lock order is always 1259 // AudioFlinger then ThreadBase to avoid cross deadlock 1260 mLock.unlock(); 1261 mAudioFlinger->mLock.lock(); 1262 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1263 mAudioFlinger->mLock.unlock(); 1264 mLock.lock(); 1265 } 1266 mLock.unlock(); 1267} 1268 1269void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1270{ 1271 const size_t SIZE = 256; 1272 char buffer[SIZE]; 1273 String8 result; 1274 1275 bool locked = tryLock(mLock); 1276 if (!locked) { 1277 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1278 write(fd, buffer, strlen(buffer)); 1279 } 1280 1281 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1282 result.append(buffer); 1283 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1284 result.append(buffer); 1285 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1286 result.append(buffer); 1287 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1288 result.append(buffer); 1289 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1290 result.append(buffer); 1291 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1292 result.append(buffer); 1293 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1294 result.append(buffer); 1295 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1296 result.append(buffer); 1297 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1298 result.append(buffer); 1299 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1300 result.append(buffer); 1301 1302 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1303 result.append(buffer); 1304 result.append(" Index Command"); 1305 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1306 snprintf(buffer, SIZE, "\n %02d ", i); 1307 result.append(buffer); 1308 result.append(mNewParameters[i]); 1309 } 1310 1311 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1312 result.append(buffer); 1313 snprintf(buffer, SIZE, " Index event param\n"); 1314 result.append(buffer); 1315 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1316 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1317 result.append(buffer); 1318 } 1319 result.append("\n"); 1320 1321 write(fd, result.string(), result.size()); 1322 1323 if (locked) { 1324 mLock.unlock(); 1325 } 1326} 1327 1328void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1329{ 1330 const size_t SIZE = 256; 1331 char buffer[SIZE]; 1332 String8 result; 1333 1334 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1335 write(fd, buffer, strlen(buffer)); 1336 1337 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1338 sp<EffectChain> chain = mEffectChains[i]; 1339 if (chain != 0) { 1340 chain->dump(fd, args); 1341 } 1342 } 1343} 1344 1345void AudioFlinger::ThreadBase::acquireWakeLock() 1346{ 1347 Mutex::Autolock _l(mLock); 1348 acquireWakeLock_l(); 1349} 1350 1351void AudioFlinger::ThreadBase::acquireWakeLock_l() 1352{ 1353 if (mPowerManager == 0) { 1354 // use checkService() to avoid blocking if power service is not up yet 1355 sp<IBinder> binder = 1356 defaultServiceManager()->checkService(String16("power")); 1357 if (binder == 0) { 1358 ALOGW("Thread %s cannot connect to the power manager service", mName); 1359 } else { 1360 mPowerManager = interface_cast<IPowerManager>(binder); 1361 binder->linkToDeath(mDeathRecipient); 1362 } 1363 } 1364 if (mPowerManager != 0) { 1365 sp<IBinder> binder = new BBinder(); 1366 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1367 binder, 1368 String16(mName)); 1369 if (status == NO_ERROR) { 1370 mWakeLockToken = binder; 1371 } 1372 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1373 } 1374} 1375 1376void AudioFlinger::ThreadBase::releaseWakeLock() 1377{ 1378 Mutex::Autolock _l(mLock); 1379 releaseWakeLock_l(); 1380} 1381 1382void AudioFlinger::ThreadBase::releaseWakeLock_l() 1383{ 1384 if (mWakeLockToken != 0) { 1385 ALOGV("releaseWakeLock_l() %s", mName); 1386 if (mPowerManager != 0) { 1387 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1388 } 1389 mWakeLockToken.clear(); 1390 } 1391} 1392 1393void AudioFlinger::ThreadBase::clearPowerManager() 1394{ 1395 Mutex::Autolock _l(mLock); 1396 releaseWakeLock_l(); 1397 mPowerManager.clear(); 1398} 1399 1400void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1401{ 1402 sp<ThreadBase> thread = mThread.promote(); 1403 if (thread != 0) { 1404 thread->clearPowerManager(); 1405 } 1406 ALOGW("power manager service died !!!"); 1407} 1408 1409void AudioFlinger::ThreadBase::setEffectSuspended( 1410 const effect_uuid_t *type, bool suspend, int sessionId) 1411{ 1412 Mutex::Autolock _l(mLock); 1413 setEffectSuspended_l(type, suspend, sessionId); 1414} 1415 1416void AudioFlinger::ThreadBase::setEffectSuspended_l( 1417 const effect_uuid_t *type, bool suspend, int sessionId) 1418{ 1419 sp<EffectChain> chain = getEffectChain_l(sessionId); 1420 if (chain != 0) { 1421 if (type != NULL) { 1422 chain->setEffectSuspended_l(type, suspend); 1423 } else { 1424 chain->setEffectSuspendedAll_l(suspend); 1425 } 1426 } 1427 1428 updateSuspendedSessions_l(type, suspend, sessionId); 1429} 1430 1431void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1432{ 1433 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1434 if (index < 0) { 1435 return; 1436 } 1437 1438 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1439 mSuspendedSessions.valueAt(index); 1440 1441 for (size_t i = 0; i < sessionEffects.size(); i++) { 1442 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1443 for (int j = 0; j < desc->mRefCount; j++) { 1444 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1445 chain->setEffectSuspendedAll_l(true); 1446 } else { 1447 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1448 desc->mType.timeLow); 1449 chain->setEffectSuspended_l(&desc->mType, true); 1450 } 1451 } 1452 } 1453} 1454 1455void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1456 bool suspend, 1457 int sessionId) 1458{ 1459 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1460 1461 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1462 1463 if (suspend) { 1464 if (index >= 0) { 1465 sessionEffects = mSuspendedSessions.valueAt(index); 1466 } else { 1467 mSuspendedSessions.add(sessionId, sessionEffects); 1468 } 1469 } else { 1470 if (index < 0) { 1471 return; 1472 } 1473 sessionEffects = mSuspendedSessions.valueAt(index); 1474 } 1475 1476 1477 int key = EffectChain::kKeyForSuspendAll; 1478 if (type != NULL) { 1479 key = type->timeLow; 1480 } 1481 index = sessionEffects.indexOfKey(key); 1482 1483 sp<SuspendedSessionDesc> desc; 1484 if (suspend) { 1485 if (index >= 0) { 1486 desc = sessionEffects.valueAt(index); 1487 } else { 1488 desc = new SuspendedSessionDesc(); 1489 if (type != NULL) { 1490 desc->mType = *type; 1491 } 1492 sessionEffects.add(key, desc); 1493 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1494 } 1495 desc->mRefCount++; 1496 } else { 1497 if (index < 0) { 1498 return; 1499 } 1500 desc = sessionEffects.valueAt(index); 1501 if (--desc->mRefCount == 0) { 1502 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1503 sessionEffects.removeItemsAt(index); 1504 if (sessionEffects.isEmpty()) { 1505 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1506 sessionId); 1507 mSuspendedSessions.removeItem(sessionId); 1508 } 1509 } 1510 } 1511 if (!sessionEffects.isEmpty()) { 1512 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1513 } 1514} 1515 1516void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1517 bool enabled, 1518 int sessionId) 1519{ 1520 Mutex::Autolock _l(mLock); 1521 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1522} 1523 1524void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1525 bool enabled, 1526 int sessionId) 1527{ 1528 if (mType != RECORD) { 1529 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1530 // another session. This gives the priority to well behaved effect control panels 1531 // and applications not using global effects. 1532 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1533 // global effects 1534 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1535 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1536 } 1537 } 1538 1539 sp<EffectChain> chain = getEffectChain_l(sessionId); 1540 if (chain != 0) { 1541 chain->checkSuspendOnEffectEnabled(effect, enabled); 1542 } 1543} 1544 1545// ---------------------------------------------------------------------------- 1546 1547AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1548 AudioStreamOut* output, 1549 audio_io_handle_t id, 1550 audio_devices_t device, 1551 type_t type) 1552 : ThreadBase(audioFlinger, id, device, type), 1553 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1554 // Assumes constructor is called by AudioFlinger with it's mLock held, 1555 // but it would be safer to explicitly pass initial masterMute as parameter 1556 mMasterMute(audioFlinger->masterMuteSW_l()), 1557 // mStreamTypes[] initialized in constructor body 1558 mOutput(output), 1559 // Assumes constructor is called by AudioFlinger with it's mLock held, 1560 // but it would be safer to explicitly pass initial masterVolume as parameter 1561 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1562 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1563 mMixerStatus(MIXER_IDLE), 1564 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1565 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1566 mScreenState(gScreenState), 1567 // index 0 is reserved for normal mixer's submix 1568 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1569{ 1570 snprintf(mName, kNameLength, "AudioOut_%X", id); 1571 1572 readOutputParameters(); 1573 1574 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1575 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1576 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1577 stream = (audio_stream_type_t) (stream + 1)) { 1578 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1579 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1580 } 1581 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1582 // because mAudioFlinger doesn't have one to copy from 1583} 1584 1585AudioFlinger::PlaybackThread::~PlaybackThread() 1586{ 1587 delete [] mMixBuffer; 1588} 1589 1590void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1591{ 1592 dumpInternals(fd, args); 1593 dumpTracks(fd, args); 1594 dumpEffectChains(fd, args); 1595} 1596 1597void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1598{ 1599 const size_t SIZE = 256; 1600 char buffer[SIZE]; 1601 String8 result; 1602 1603 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1604 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1605 const stream_type_t *st = &mStreamTypes[i]; 1606 if (i > 0) { 1607 result.appendFormat(", "); 1608 } 1609 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1610 if (st->mute) { 1611 result.append("M"); 1612 } 1613 } 1614 result.append("\n"); 1615 write(fd, result.string(), result.length()); 1616 result.clear(); 1617 1618 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1619 result.append(buffer); 1620 Track::appendDumpHeader(result); 1621 for (size_t i = 0; i < mTracks.size(); ++i) { 1622 sp<Track> track = mTracks[i]; 1623 if (track != 0) { 1624 track->dump(buffer, SIZE); 1625 result.append(buffer); 1626 } 1627 } 1628 1629 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1630 result.append(buffer); 1631 Track::appendDumpHeader(result); 1632 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1633 sp<Track> track = mActiveTracks[i].promote(); 1634 if (track != 0) { 1635 track->dump(buffer, SIZE); 1636 result.append(buffer); 1637 } 1638 } 1639 write(fd, result.string(), result.size()); 1640 1641 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1642 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1643 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1644 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1645} 1646 1647void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1648{ 1649 const size_t SIZE = 256; 1650 char buffer[SIZE]; 1651 String8 result; 1652 1653 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1654 result.append(buffer); 1655 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1656 result.append(buffer); 1657 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1658 result.append(buffer); 1659 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1660 result.append(buffer); 1661 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1662 result.append(buffer); 1663 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1664 result.append(buffer); 1665 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1666 result.append(buffer); 1667 write(fd, result.string(), result.size()); 1668 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1669 1670 dumpBase(fd, args); 1671} 1672 1673// Thread virtuals 1674status_t AudioFlinger::PlaybackThread::readyToRun() 1675{ 1676 status_t status = initCheck(); 1677 if (status == NO_ERROR) { 1678 ALOGI("AudioFlinger's thread %p ready to run", this); 1679 } else { 1680 ALOGE("No working audio driver found."); 1681 } 1682 return status; 1683} 1684 1685void AudioFlinger::PlaybackThread::onFirstRef() 1686{ 1687 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1688} 1689 1690// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1691sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1692 const sp<AudioFlinger::Client>& client, 1693 audio_stream_type_t streamType, 1694 uint32_t sampleRate, 1695 audio_format_t format, 1696 audio_channel_mask_t channelMask, 1697 int frameCount, 1698 const sp<IMemory>& sharedBuffer, 1699 int sessionId, 1700 IAudioFlinger::track_flags_t flags, 1701 pid_t tid, 1702 status_t *status) 1703{ 1704 sp<Track> track; 1705 status_t lStatus; 1706 1707 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1708 1709 // client expresses a preference for FAST, but we get the final say 1710 if (flags & IAudioFlinger::TRACK_FAST) { 1711 if ( 1712 // not timed 1713 (!isTimed) && 1714 // either of these use cases: 1715 ( 1716 // use case 1: shared buffer with any frame count 1717 ( 1718 (sharedBuffer != 0) 1719 ) || 1720 // use case 2: callback handler and frame count is default or at least as large as HAL 1721 ( 1722 (tid != -1) && 1723 ((frameCount == 0) || 1724 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1725 ) 1726 ) && 1727 // PCM data 1728 audio_is_linear_pcm(format) && 1729 // mono or stereo 1730 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1731 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1732#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1733 // hardware sample rate 1734 (sampleRate == mSampleRate) && 1735#endif 1736 // normal mixer has an associated fast mixer 1737 hasFastMixer() && 1738 // there are sufficient fast track slots available 1739 (mFastTrackAvailMask != 0) 1740 // FIXME test that MixerThread for this fast track has a capable output HAL 1741 // FIXME add a permission test also? 1742 ) { 1743 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1744 if (frameCount == 0) { 1745 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1746 } 1747 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1748 frameCount, mFrameCount); 1749 } else { 1750 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1751 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1752 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1753 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1754 audio_is_linear_pcm(format), 1755 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1756 flags &= ~IAudioFlinger::TRACK_FAST; 1757 // For compatibility with AudioTrack calculation, buffer depth is forced 1758 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1759 // This is probably too conservative, but legacy application code may depend on it. 1760 // If you change this calculation, also review the start threshold which is related. 1761 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1762 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1763 if (minBufCount < 2) { 1764 minBufCount = 2; 1765 } 1766 int minFrameCount = mNormalFrameCount * minBufCount; 1767 if (frameCount < minFrameCount) { 1768 frameCount = minFrameCount; 1769 } 1770 } 1771 } 1772 1773 if (mType == DIRECT) { 1774 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1775 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1776 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1777 "for output %p with format %d", 1778 sampleRate, format, channelMask, mOutput, mFormat); 1779 lStatus = BAD_VALUE; 1780 goto Exit; 1781 } 1782 } 1783 } else { 1784 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1785 if (sampleRate > mSampleRate*2) { 1786 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1787 lStatus = BAD_VALUE; 1788 goto Exit; 1789 } 1790 } 1791 1792 lStatus = initCheck(); 1793 if (lStatus != NO_ERROR) { 1794 ALOGE("Audio driver not initialized."); 1795 goto Exit; 1796 } 1797 1798 { // scope for mLock 1799 Mutex::Autolock _l(mLock); 1800 1801 // all tracks in same audio session must share the same routing strategy otherwise 1802 // conflicts will happen when tracks are moved from one output to another by audio policy 1803 // manager 1804 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1805 for (size_t i = 0; i < mTracks.size(); ++i) { 1806 sp<Track> t = mTracks[i]; 1807 if (t != 0 && !t->isOutputTrack()) { 1808 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1809 if (sessionId == t->sessionId() && strategy != actual) { 1810 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1811 strategy, actual); 1812 lStatus = BAD_VALUE; 1813 goto Exit; 1814 } 1815 } 1816 } 1817 1818 if (!isTimed) { 1819 track = new Track(this, client, streamType, sampleRate, format, 1820 channelMask, frameCount, sharedBuffer, sessionId, flags); 1821 } else { 1822 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1823 channelMask, frameCount, sharedBuffer, sessionId); 1824 } 1825 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1826 lStatus = NO_MEMORY; 1827 goto Exit; 1828 } 1829 mTracks.add(track); 1830 1831 sp<EffectChain> chain = getEffectChain_l(sessionId); 1832 if (chain != 0) { 1833 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1834 track->setMainBuffer(chain->inBuffer()); 1835 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1836 chain->incTrackCnt(); 1837 } 1838 } 1839 1840 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1841 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1842 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1843 // so ask activity manager to do this on our behalf 1844 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1845 if (err != 0) { 1846 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1847 kPriorityAudioApp, callingPid, tid, err); 1848 } 1849 } 1850 1851 lStatus = NO_ERROR; 1852 1853Exit: 1854 if (status) { 1855 *status = lStatus; 1856 } 1857 return track; 1858} 1859 1860uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1861{ 1862 if (mFastMixer != NULL) { 1863 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1864 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1865 } 1866 return latency; 1867} 1868 1869uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1870{ 1871 return latency; 1872} 1873 1874uint32_t AudioFlinger::PlaybackThread::latency() const 1875{ 1876 Mutex::Autolock _l(mLock); 1877 return latency_l(); 1878} 1879uint32_t AudioFlinger::PlaybackThread::latency_l() const 1880{ 1881 if (initCheck() == NO_ERROR) { 1882 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1883 } else { 1884 return 0; 1885 } 1886} 1887 1888void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1889{ 1890 Mutex::Autolock _l(mLock); 1891 mMasterVolume = value; 1892} 1893 1894void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1895{ 1896 Mutex::Autolock _l(mLock); 1897 setMasterMute_l(muted); 1898} 1899 1900void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1901{ 1902 Mutex::Autolock _l(mLock); 1903 mStreamTypes[stream].volume = value; 1904} 1905 1906void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1907{ 1908 Mutex::Autolock _l(mLock); 1909 mStreamTypes[stream].mute = muted; 1910} 1911 1912float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1913{ 1914 Mutex::Autolock _l(mLock); 1915 return mStreamTypes[stream].volume; 1916} 1917 1918// addTrack_l() must be called with ThreadBase::mLock held 1919status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1920{ 1921 status_t status = ALREADY_EXISTS; 1922 1923 // set retry count for buffer fill 1924 track->mRetryCount = kMaxTrackStartupRetries; 1925 if (mActiveTracks.indexOf(track) < 0) { 1926 // the track is newly added, make sure it fills up all its 1927 // buffers before playing. This is to ensure the client will 1928 // effectively get the latency it requested. 1929 track->mFillingUpStatus = Track::FS_FILLING; 1930 track->mResetDone = false; 1931 track->mPresentationCompleteFrames = 0; 1932 mActiveTracks.add(track); 1933 if (track->mainBuffer() != mMixBuffer) { 1934 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1935 if (chain != 0) { 1936 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1937 chain->incActiveTrackCnt(); 1938 } 1939 } 1940 1941 status = NO_ERROR; 1942 } 1943 1944 ALOGV("mWaitWorkCV.broadcast"); 1945 mWaitWorkCV.broadcast(); 1946 1947 return status; 1948} 1949 1950// destroyTrack_l() must be called with ThreadBase::mLock held 1951void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1952{ 1953 track->mState = TrackBase::TERMINATED; 1954 // active tracks are removed by threadLoop() 1955 if (mActiveTracks.indexOf(track) < 0) { 1956 removeTrack_l(track); 1957 } 1958} 1959 1960void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1961{ 1962 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1963 mTracks.remove(track); 1964 deleteTrackName_l(track->name()); 1965 // redundant as track is about to be destroyed, for dumpsys only 1966 track->mName = -1; 1967 if (track->isFastTrack()) { 1968 int index = track->mFastIndex; 1969 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1970 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1971 mFastTrackAvailMask |= 1 << index; 1972 // redundant as track is about to be destroyed, for dumpsys only 1973 track->mFastIndex = -1; 1974 } 1975 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1976 if (chain != 0) { 1977 chain->decTrackCnt(); 1978 } 1979} 1980 1981String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1982{ 1983 String8 out_s8 = String8(""); 1984 char *s; 1985 1986 Mutex::Autolock _l(mLock); 1987 if (initCheck() != NO_ERROR) { 1988 return out_s8; 1989 } 1990 1991 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1992 out_s8 = String8(s); 1993 free(s); 1994 return out_s8; 1995} 1996 1997// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1998void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1999 AudioSystem::OutputDescriptor desc; 2000 void *param2 = NULL; 2001 2002 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 2003 2004 switch (event) { 2005 case AudioSystem::OUTPUT_OPENED: 2006 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2007 desc.channels = mChannelMask; 2008 desc.samplingRate = mSampleRate; 2009 desc.format = mFormat; 2010 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 2011 desc.latency = latency(); 2012 param2 = &desc; 2013 break; 2014 2015 case AudioSystem::STREAM_CONFIG_CHANGED: 2016 param2 = ¶m; 2017 case AudioSystem::OUTPUT_CLOSED: 2018 default: 2019 break; 2020 } 2021 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2022} 2023 2024void AudioFlinger::PlaybackThread::readOutputParameters() 2025{ 2026 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2027 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2028 mChannelCount = (uint16_t)popcount(mChannelMask); 2029 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2030 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2031 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2032 if (mFrameCount & 15) { 2033 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2034 mFrameCount); 2035 } 2036 2037 // Calculate size of normal mix buffer relative to the HAL output buffer size 2038 double multiplier = 1.0; 2039 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2040 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2041 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2042 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2043 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2044 maxNormalFrameCount = maxNormalFrameCount & ~15; 2045 if (maxNormalFrameCount < minNormalFrameCount) { 2046 maxNormalFrameCount = minNormalFrameCount; 2047 } 2048 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2049 if (multiplier <= 1.0) { 2050 multiplier = 1.0; 2051 } else if (multiplier <= 2.0) { 2052 if (2 * mFrameCount <= maxNormalFrameCount) { 2053 multiplier = 2.0; 2054 } else { 2055 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2056 } 2057 } else { 2058 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2059 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2060 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2061 // FIXME this rounding up should not be done if no HAL SRC 2062 uint32_t truncMult = (uint32_t) multiplier; 2063 if ((truncMult & 1)) { 2064 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2065 ++truncMult; 2066 } 2067 } 2068 multiplier = (double) truncMult; 2069 } 2070 } 2071 mNormalFrameCount = multiplier * mFrameCount; 2072 // round up to nearest 16 frames to satisfy AudioMixer 2073 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2074 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2075 2076 delete[] mMixBuffer; 2077 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2078 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2079 2080 // force reconfiguration of effect chains and engines to take new buffer size and audio 2081 // parameters into account 2082 // Note that mLock is not held when readOutputParameters() is called from the constructor 2083 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2084 // matter. 2085 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2086 Vector< sp<EffectChain> > effectChains = mEffectChains; 2087 for (size_t i = 0; i < effectChains.size(); i ++) { 2088 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2089 } 2090} 2091 2092 2093status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2094{ 2095 if (halFrames == NULL || dspFrames == NULL) { 2096 return BAD_VALUE; 2097 } 2098 Mutex::Autolock _l(mLock); 2099 if (initCheck() != NO_ERROR) { 2100 return INVALID_OPERATION; 2101 } 2102 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2103 2104 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2105} 2106 2107uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2108{ 2109 Mutex::Autolock _l(mLock); 2110 uint32_t result = 0; 2111 if (getEffectChain_l(sessionId) != 0) { 2112 result = EFFECT_SESSION; 2113 } 2114 2115 for (size_t i = 0; i < mTracks.size(); ++i) { 2116 sp<Track> track = mTracks[i]; 2117 if (sessionId == track->sessionId() && 2118 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2119 result |= TRACK_SESSION; 2120 break; 2121 } 2122 } 2123 2124 return result; 2125} 2126 2127uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2128{ 2129 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2130 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2131 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2132 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2133 } 2134 for (size_t i = 0; i < mTracks.size(); i++) { 2135 sp<Track> track = mTracks[i]; 2136 if (sessionId == track->sessionId() && 2137 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2138 return AudioSystem::getStrategyForStream(track->streamType()); 2139 } 2140 } 2141 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2142} 2143 2144 2145AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2146{ 2147 Mutex::Autolock _l(mLock); 2148 return mOutput; 2149} 2150 2151AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2152{ 2153 Mutex::Autolock _l(mLock); 2154 AudioStreamOut *output = mOutput; 2155 mOutput = NULL; 2156 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2157 // must push a NULL and wait for ack 2158 mOutputSink.clear(); 2159 mPipeSink.clear(); 2160 mNormalSink.clear(); 2161 return output; 2162} 2163 2164// this method must always be called either with ThreadBase mLock held or inside the thread loop 2165audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2166{ 2167 if (mOutput == NULL) { 2168 return NULL; 2169 } 2170 return &mOutput->stream->common; 2171} 2172 2173uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2174{ 2175 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2176} 2177 2178status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2179{ 2180 if (!isValidSyncEvent(event)) { 2181 return BAD_VALUE; 2182 } 2183 2184 Mutex::Autolock _l(mLock); 2185 2186 for (size_t i = 0; i < mTracks.size(); ++i) { 2187 sp<Track> track = mTracks[i]; 2188 if (event->triggerSession() == track->sessionId()) { 2189 track->setSyncEvent(event); 2190 return NO_ERROR; 2191 } 2192 } 2193 2194 return NAME_NOT_FOUND; 2195} 2196 2197bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2198{ 2199 switch (event->type()) { 2200 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2201 return true; 2202 default: 2203 break; 2204 } 2205 return false; 2206} 2207 2208void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2209{ 2210 size_t count = tracksToRemove.size(); 2211 if (CC_UNLIKELY(count)) { 2212 for (size_t i = 0 ; i < count ; i++) { 2213 const sp<Track>& track = tracksToRemove.itemAt(i); 2214 if ((track->sharedBuffer() != 0) && 2215 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2216 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2217 } 2218 } 2219 } 2220 2221} 2222 2223// ---------------------------------------------------------------------------- 2224 2225AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2226 audio_io_handle_t id, audio_devices_t device, type_t type) 2227 : PlaybackThread(audioFlinger, output, id, device, type), 2228 // mAudioMixer below 2229 // mFastMixer below 2230 mFastMixerFutex(0) 2231 // mOutputSink below 2232 // mPipeSink below 2233 // mNormalSink below 2234{ 2235 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2236 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2237 "mFrameCount=%d, mNormalFrameCount=%d", 2238 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2239 mNormalFrameCount); 2240 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2241 2242 // FIXME - Current mixer implementation only supports stereo output 2243 if (mChannelCount != FCC_2) { 2244 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2245 } 2246 2247 // create an NBAIO sink for the HAL output stream, and negotiate 2248 mOutputSink = new AudioStreamOutSink(output->stream); 2249 size_t numCounterOffers = 0; 2250 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2251 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2252 ALOG_ASSERT(index == 0); 2253 2254 // initialize fast mixer depending on configuration 2255 bool initFastMixer; 2256 switch (kUseFastMixer) { 2257 case FastMixer_Never: 2258 initFastMixer = false; 2259 break; 2260 case FastMixer_Always: 2261 initFastMixer = true; 2262 break; 2263 case FastMixer_Static: 2264 case FastMixer_Dynamic: 2265 initFastMixer = mFrameCount < mNormalFrameCount; 2266 break; 2267 } 2268 if (initFastMixer) { 2269 2270 // create a MonoPipe to connect our submix to FastMixer 2271 NBAIO_Format format = mOutputSink->format(); 2272 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2273 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2274 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2275 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2276 const NBAIO_Format offers[1] = {format}; 2277 size_t numCounterOffers = 0; 2278 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2279 ALOG_ASSERT(index == 0); 2280 monoPipe->setAvgFrames((mScreenState & 1) ? 2281 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2282 mPipeSink = monoPipe; 2283 2284#ifdef TEE_SINK_FRAMES 2285 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2286 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2287 numCounterOffers = 0; 2288 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2289 ALOG_ASSERT(index == 0); 2290 mTeeSink = teeSink; 2291 PipeReader *teeSource = new PipeReader(*teeSink); 2292 numCounterOffers = 0; 2293 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2294 ALOG_ASSERT(index == 0); 2295 mTeeSource = teeSource; 2296#endif 2297 2298 // create fast mixer and configure it initially with just one fast track for our submix 2299 mFastMixer = new FastMixer(); 2300 FastMixerStateQueue *sq = mFastMixer->sq(); 2301#ifdef STATE_QUEUE_DUMP 2302 sq->setObserverDump(&mStateQueueObserverDump); 2303 sq->setMutatorDump(&mStateQueueMutatorDump); 2304#endif 2305 FastMixerState *state = sq->begin(); 2306 FastTrack *fastTrack = &state->mFastTracks[0]; 2307 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2308 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2309 fastTrack->mVolumeProvider = NULL; 2310 fastTrack->mGeneration++; 2311 state->mFastTracksGen++; 2312 state->mTrackMask = 1; 2313 // fast mixer will use the HAL output sink 2314 state->mOutputSink = mOutputSink.get(); 2315 state->mOutputSinkGen++; 2316 state->mFrameCount = mFrameCount; 2317 state->mCommand = FastMixerState::COLD_IDLE; 2318 // already done in constructor initialization list 2319 //mFastMixerFutex = 0; 2320 state->mColdFutexAddr = &mFastMixerFutex; 2321 state->mColdGen++; 2322 state->mDumpState = &mFastMixerDumpState; 2323 state->mTeeSink = mTeeSink.get(); 2324 sq->end(); 2325 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2326 2327 // start the fast mixer 2328 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2329 pid_t tid = mFastMixer->getTid(); 2330 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2331 if (err != 0) { 2332 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2333 kPriorityFastMixer, getpid_cached, tid, err); 2334 } 2335 2336#ifdef AUDIO_WATCHDOG 2337 // create and start the watchdog 2338 mAudioWatchdog = new AudioWatchdog(); 2339 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2340 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2341 tid = mAudioWatchdog->getTid(); 2342 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2343 if (err != 0) { 2344 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2345 kPriorityFastMixer, getpid_cached, tid, err); 2346 } 2347#endif 2348 2349 } else { 2350 mFastMixer = NULL; 2351 } 2352 2353 switch (kUseFastMixer) { 2354 case FastMixer_Never: 2355 case FastMixer_Dynamic: 2356 mNormalSink = mOutputSink; 2357 break; 2358 case FastMixer_Always: 2359 mNormalSink = mPipeSink; 2360 break; 2361 case FastMixer_Static: 2362 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2363 break; 2364 } 2365} 2366 2367AudioFlinger::MixerThread::~MixerThread() 2368{ 2369 if (mFastMixer != NULL) { 2370 FastMixerStateQueue *sq = mFastMixer->sq(); 2371 FastMixerState *state = sq->begin(); 2372 if (state->mCommand == FastMixerState::COLD_IDLE) { 2373 int32_t old = android_atomic_inc(&mFastMixerFutex); 2374 if (old == -1) { 2375 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2376 } 2377 } 2378 state->mCommand = FastMixerState::EXIT; 2379 sq->end(); 2380 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2381 mFastMixer->join(); 2382 // Though the fast mixer thread has exited, it's state queue is still valid. 2383 // We'll use that extract the final state which contains one remaining fast track 2384 // corresponding to our sub-mix. 2385 state = sq->begin(); 2386 ALOG_ASSERT(state->mTrackMask == 1); 2387 FastTrack *fastTrack = &state->mFastTracks[0]; 2388 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2389 delete fastTrack->mBufferProvider; 2390 sq->end(false /*didModify*/); 2391 delete mFastMixer; 2392 if (mAudioWatchdog != 0) { 2393 mAudioWatchdog->requestExit(); 2394 mAudioWatchdog->requestExitAndWait(); 2395 mAudioWatchdog.clear(); 2396 } 2397 } 2398 delete mAudioMixer; 2399} 2400 2401class CpuStats { 2402public: 2403 CpuStats(); 2404 void sample(const String8 &title); 2405#ifdef DEBUG_CPU_USAGE 2406private: 2407 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2408 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2409 2410 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2411 2412 int mCpuNum; // thread's current CPU number 2413 int mCpukHz; // frequency of thread's current CPU in kHz 2414#endif 2415}; 2416 2417CpuStats::CpuStats() 2418#ifdef DEBUG_CPU_USAGE 2419 : mCpuNum(-1), mCpukHz(-1) 2420#endif 2421{ 2422} 2423 2424void CpuStats::sample(const String8 &title) { 2425#ifdef DEBUG_CPU_USAGE 2426 // get current thread's delta CPU time in wall clock ns 2427 double wcNs; 2428 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2429 2430 // record sample for wall clock statistics 2431 if (valid) { 2432 mWcStats.sample(wcNs); 2433 } 2434 2435 // get the current CPU number 2436 int cpuNum = sched_getcpu(); 2437 2438 // get the current CPU frequency in kHz 2439 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2440 2441 // check if either CPU number or frequency changed 2442 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2443 mCpuNum = cpuNum; 2444 mCpukHz = cpukHz; 2445 // ignore sample for purposes of cycles 2446 valid = false; 2447 } 2448 2449 // if no change in CPU number or frequency, then record sample for cycle statistics 2450 if (valid && mCpukHz > 0) { 2451 double cycles = wcNs * cpukHz * 0.000001; 2452 mHzStats.sample(cycles); 2453 } 2454 2455 unsigned n = mWcStats.n(); 2456 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2457 if ((n & 127) == 1) { 2458 long long elapsed = mCpuUsage.elapsed(); 2459 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2460 double perLoop = elapsed / (double) n; 2461 double perLoop100 = perLoop * 0.01; 2462 double perLoop1k = perLoop * 0.001; 2463 double mean = mWcStats.mean(); 2464 double stddev = mWcStats.stddev(); 2465 double minimum = mWcStats.minimum(); 2466 double maximum = mWcStats.maximum(); 2467 double meanCycles = mHzStats.mean(); 2468 double stddevCycles = mHzStats.stddev(); 2469 double minCycles = mHzStats.minimum(); 2470 double maxCycles = mHzStats.maximum(); 2471 mCpuUsage.resetElapsed(); 2472 mWcStats.reset(); 2473 mHzStats.reset(); 2474 ALOGD("CPU usage for %s over past %.1f secs\n" 2475 " (%u mixer loops at %.1f mean ms per loop):\n" 2476 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2477 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2478 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2479 title.string(), 2480 elapsed * .000000001, n, perLoop * .000001, 2481 mean * .001, 2482 stddev * .001, 2483 minimum * .001, 2484 maximum * .001, 2485 mean / perLoop100, 2486 stddev / perLoop100, 2487 minimum / perLoop100, 2488 maximum / perLoop100, 2489 meanCycles / perLoop1k, 2490 stddevCycles / perLoop1k, 2491 minCycles / perLoop1k, 2492 maxCycles / perLoop1k); 2493 2494 } 2495 } 2496#endif 2497}; 2498 2499void AudioFlinger::PlaybackThread::checkSilentMode_l() 2500{ 2501 if (!mMasterMute) { 2502 char value[PROPERTY_VALUE_MAX]; 2503 if (property_get("ro.audio.silent", value, "0") > 0) { 2504 char *endptr; 2505 unsigned long ul = strtoul(value, &endptr, 0); 2506 if (*endptr == '\0' && ul != 0) { 2507 ALOGD("Silence is golden"); 2508 // The setprop command will not allow a property to be changed after 2509 // the first time it is set, so we don't have to worry about un-muting. 2510 setMasterMute_l(true); 2511 } 2512 } 2513 } 2514} 2515 2516bool AudioFlinger::PlaybackThread::threadLoop() 2517{ 2518 Vector< sp<Track> > tracksToRemove; 2519 2520 standbyTime = systemTime(); 2521 2522 // MIXER 2523 nsecs_t lastWarning = 0; 2524 2525 // DUPLICATING 2526 // FIXME could this be made local to while loop? 2527 writeFrames = 0; 2528 2529 cacheParameters_l(); 2530 sleepTime = idleSleepTime; 2531 2532 if (mType == MIXER) { 2533 sleepTimeShift = 0; 2534 } 2535 2536 CpuStats cpuStats; 2537 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2538 2539 acquireWakeLock(); 2540 2541 while (!exitPending()) 2542 { 2543 cpuStats.sample(myName); 2544 2545 Vector< sp<EffectChain> > effectChains; 2546 2547 processConfigEvents(); 2548 2549 { // scope for mLock 2550 2551 Mutex::Autolock _l(mLock); 2552 2553 if (checkForNewParameters_l()) { 2554 cacheParameters_l(); 2555 } 2556 2557 saveOutputTracks(); 2558 2559 // put audio hardware into standby after short delay 2560 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2561 isSuspended())) { 2562 if (!mStandby) { 2563 2564 threadLoop_standby(); 2565 2566 mStandby = true; 2567 mBytesWritten = 0; 2568 } 2569 2570 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2571 // we're about to wait, flush the binder command buffer 2572 IPCThreadState::self()->flushCommands(); 2573 2574 clearOutputTracks(); 2575 2576 if (exitPending()) break; 2577 2578 releaseWakeLock_l(); 2579 // wait until we have something to do... 2580 ALOGV("%s going to sleep", myName.string()); 2581 mWaitWorkCV.wait(mLock); 2582 ALOGV("%s waking up", myName.string()); 2583 acquireWakeLock_l(); 2584 2585 mMixerStatus = MIXER_IDLE; 2586 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2587 2588 checkSilentMode_l(); 2589 2590 standbyTime = systemTime() + standbyDelay; 2591 sleepTime = idleSleepTime; 2592 if (mType == MIXER) { 2593 sleepTimeShift = 0; 2594 } 2595 2596 continue; 2597 } 2598 } 2599 2600 // mMixerStatusIgnoringFastTracks is also updated internally 2601 mMixerStatus = prepareTracks_l(&tracksToRemove); 2602 2603 // prevent any changes in effect chain list and in each effect chain 2604 // during mixing and effect process as the audio buffers could be deleted 2605 // or modified if an effect is created or deleted 2606 lockEffectChains_l(effectChains); 2607 } 2608 2609 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2610 threadLoop_mix(); 2611 } else { 2612 threadLoop_sleepTime(); 2613 } 2614 2615 if (isSuspended()) { 2616 sleepTime = suspendSleepTimeUs(); 2617 } 2618 2619 // only process effects if we're going to write 2620 if (sleepTime == 0) { 2621 for (size_t i = 0; i < effectChains.size(); i ++) { 2622 effectChains[i]->process_l(); 2623 } 2624 } 2625 2626 // enable changes in effect chain 2627 unlockEffectChains(effectChains); 2628 2629 // sleepTime == 0 means we must write to audio hardware 2630 if (sleepTime == 0) { 2631 2632 threadLoop_write(); 2633 2634if (mType == MIXER) { 2635 // write blocked detection 2636 nsecs_t now = systemTime(); 2637 nsecs_t delta = now - mLastWriteTime; 2638 if (!mStandby && delta > maxPeriod) { 2639 mNumDelayedWrites++; 2640 if ((now - lastWarning) > kWarningThrottleNs) { 2641#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2642 ScopedTrace st(ATRACE_TAG, "underrun"); 2643#endif 2644 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2645 ns2ms(delta), mNumDelayedWrites, this); 2646 lastWarning = now; 2647 } 2648 } 2649} 2650 2651 mStandby = false; 2652 } else { 2653 usleep(sleepTime); 2654 } 2655 2656 // Finally let go of removed track(s), without the lock held 2657 // since we can't guarantee the destructors won't acquire that 2658 // same lock. This will also mutate and push a new fast mixer state. 2659 threadLoop_removeTracks(tracksToRemove); 2660 tracksToRemove.clear(); 2661 2662 // FIXME I don't understand the need for this here; 2663 // it was in the original code but maybe the 2664 // assignment in saveOutputTracks() makes this unnecessary? 2665 clearOutputTracks(); 2666 2667 // Effect chains will be actually deleted here if they were removed from 2668 // mEffectChains list during mixing or effects processing 2669 effectChains.clear(); 2670 2671 // FIXME Note that the above .clear() is no longer necessary since effectChains 2672 // is now local to this block, but will keep it for now (at least until merge done). 2673 } 2674 2675 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2676 if (mType == MIXER || mType == DIRECT) { 2677 // put output stream into standby mode 2678 if (!mStandby) { 2679 mOutput->stream->common.standby(&mOutput->stream->common); 2680 } 2681 } 2682 2683 releaseWakeLock(); 2684 2685 ALOGV("Thread %p type %d exiting", this, mType); 2686 return false; 2687} 2688 2689void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2690{ 2691 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2692} 2693 2694void AudioFlinger::MixerThread::threadLoop_write() 2695{ 2696 // FIXME we should only do one push per cycle; confirm this is true 2697 // Start the fast mixer if it's not already running 2698 if (mFastMixer != NULL) { 2699 FastMixerStateQueue *sq = mFastMixer->sq(); 2700 FastMixerState *state = sq->begin(); 2701 if (state->mCommand != FastMixerState::MIX_WRITE && 2702 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2703 if (state->mCommand == FastMixerState::COLD_IDLE) { 2704 int32_t old = android_atomic_inc(&mFastMixerFutex); 2705 if (old == -1) { 2706 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2707 } 2708 if (mAudioWatchdog != 0) { 2709 mAudioWatchdog->resume(); 2710 } 2711 } 2712 state->mCommand = FastMixerState::MIX_WRITE; 2713 sq->end(); 2714 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2715 if (kUseFastMixer == FastMixer_Dynamic) { 2716 mNormalSink = mPipeSink; 2717 } 2718 } else { 2719 sq->end(false /*didModify*/); 2720 } 2721 } 2722 PlaybackThread::threadLoop_write(); 2723} 2724 2725// shared by MIXER and DIRECT, overridden by DUPLICATING 2726void AudioFlinger::PlaybackThread::threadLoop_write() 2727{ 2728 // FIXME rewrite to reduce number of system calls 2729 mLastWriteTime = systemTime(); 2730 mInWrite = true; 2731 int bytesWritten; 2732 2733 // If an NBAIO sink is present, use it to write the normal mixer's submix 2734 if (mNormalSink != 0) { 2735#define mBitShift 2 // FIXME 2736 size_t count = mixBufferSize >> mBitShift; 2737#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2738 Tracer::traceBegin(ATRACE_TAG, "write"); 2739#endif 2740 // update the setpoint when gScreenState changes 2741 uint32_t screenState = gScreenState; 2742 if (screenState != mScreenState) { 2743 mScreenState = screenState; 2744 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2745 if (pipe != NULL) { 2746 pipe->setAvgFrames((mScreenState & 1) ? 2747 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2748 } 2749 } 2750 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2751#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2752 Tracer::traceEnd(ATRACE_TAG); 2753#endif 2754 if (framesWritten > 0) { 2755 bytesWritten = framesWritten << mBitShift; 2756 } else { 2757 bytesWritten = framesWritten; 2758 } 2759 // otherwise use the HAL / AudioStreamOut directly 2760 } else { 2761 // Direct output thread. 2762 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2763 } 2764 2765 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2766 mNumWrites++; 2767 mInWrite = false; 2768} 2769 2770void AudioFlinger::MixerThread::threadLoop_standby() 2771{ 2772 // Idle the fast mixer if it's currently running 2773 if (mFastMixer != NULL) { 2774 FastMixerStateQueue *sq = mFastMixer->sq(); 2775 FastMixerState *state = sq->begin(); 2776 if (!(state->mCommand & FastMixerState::IDLE)) { 2777 state->mCommand = FastMixerState::COLD_IDLE; 2778 state->mColdFutexAddr = &mFastMixerFutex; 2779 state->mColdGen++; 2780 mFastMixerFutex = 0; 2781 sq->end(); 2782 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2783 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2784 if (kUseFastMixer == FastMixer_Dynamic) { 2785 mNormalSink = mOutputSink; 2786 } 2787 if (mAudioWatchdog != 0) { 2788 mAudioWatchdog->pause(); 2789 } 2790 } else { 2791 sq->end(false /*didModify*/); 2792 } 2793 } 2794 PlaybackThread::threadLoop_standby(); 2795} 2796 2797// shared by MIXER and DIRECT, overridden by DUPLICATING 2798void AudioFlinger::PlaybackThread::threadLoop_standby() 2799{ 2800 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2801 mOutput->stream->common.standby(&mOutput->stream->common); 2802} 2803 2804void AudioFlinger::MixerThread::threadLoop_mix() 2805{ 2806 // obtain the presentation timestamp of the next output buffer 2807 int64_t pts; 2808 status_t status = INVALID_OPERATION; 2809 2810 if (NULL != mOutput->stream->get_next_write_timestamp) { 2811 status = mOutput->stream->get_next_write_timestamp( 2812 mOutput->stream, &pts); 2813 } 2814 2815 if (status != NO_ERROR) { 2816 pts = AudioBufferProvider::kInvalidPTS; 2817 } 2818 2819 // mix buffers... 2820 mAudioMixer->process(pts); 2821 // increase sleep time progressively when application underrun condition clears. 2822 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2823 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2824 // such that we would underrun the audio HAL. 2825 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2826 sleepTimeShift--; 2827 } 2828 sleepTime = 0; 2829 standbyTime = systemTime() + standbyDelay; 2830 //TODO: delay standby when effects have a tail 2831} 2832 2833void AudioFlinger::MixerThread::threadLoop_sleepTime() 2834{ 2835 // If no tracks are ready, sleep once for the duration of an output 2836 // buffer size, then write 0s to the output 2837 if (sleepTime == 0) { 2838 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2839 sleepTime = activeSleepTime >> sleepTimeShift; 2840 if (sleepTime < kMinThreadSleepTimeUs) { 2841 sleepTime = kMinThreadSleepTimeUs; 2842 } 2843 // reduce sleep time in case of consecutive application underruns to avoid 2844 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2845 // duration we would end up writing less data than needed by the audio HAL if 2846 // the condition persists. 2847 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2848 sleepTimeShift++; 2849 } 2850 } else { 2851 sleepTime = idleSleepTime; 2852 } 2853 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2854 memset (mMixBuffer, 0, mixBufferSize); 2855 sleepTime = 0; 2856 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2857 } 2858 // TODO add standby time extension fct of effect tail 2859} 2860 2861// prepareTracks_l() must be called with ThreadBase::mLock held 2862AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2863 Vector< sp<Track> > *tracksToRemove) 2864{ 2865 2866 mixer_state mixerStatus = MIXER_IDLE; 2867 // find out which tracks need to be processed 2868 size_t count = mActiveTracks.size(); 2869 size_t mixedTracks = 0; 2870 size_t tracksWithEffect = 0; 2871 // counts only _active_ fast tracks 2872 size_t fastTracks = 0; 2873 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2874 2875 float masterVolume = mMasterVolume; 2876 bool masterMute = mMasterMute; 2877 2878 if (masterMute) { 2879 masterVolume = 0; 2880 } 2881 // Delegate master volume control to effect in output mix effect chain if needed 2882 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2883 if (chain != 0) { 2884 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2885 chain->setVolume_l(&v, &v); 2886 masterVolume = (float)((v + (1 << 23)) >> 24); 2887 chain.clear(); 2888 } 2889 2890 // prepare a new state to push 2891 FastMixerStateQueue *sq = NULL; 2892 FastMixerState *state = NULL; 2893 bool didModify = false; 2894 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2895 if (mFastMixer != NULL) { 2896 sq = mFastMixer->sq(); 2897 state = sq->begin(); 2898 } 2899 2900 for (size_t i=0 ; i<count ; i++) { 2901 sp<Track> t = mActiveTracks[i].promote(); 2902 if (t == 0) continue; 2903 2904 // this const just means the local variable doesn't change 2905 Track* const track = t.get(); 2906 2907 // process fast tracks 2908 if (track->isFastTrack()) { 2909 2910 // It's theoretically possible (though unlikely) for a fast track to be created 2911 // and then removed within the same normal mix cycle. This is not a problem, as 2912 // the track never becomes active so it's fast mixer slot is never touched. 2913 // The converse, of removing an (active) track and then creating a new track 2914 // at the identical fast mixer slot within the same normal mix cycle, 2915 // is impossible because the slot isn't marked available until the end of each cycle. 2916 int j = track->mFastIndex; 2917 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2918 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2919 FastTrack *fastTrack = &state->mFastTracks[j]; 2920 2921 // Determine whether the track is currently in underrun condition, 2922 // and whether it had a recent underrun. 2923 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2924 FastTrackUnderruns underruns = ftDump->mUnderruns; 2925 uint32_t recentFull = (underruns.mBitFields.mFull - 2926 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2927 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2928 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2929 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2930 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2931 uint32_t recentUnderruns = recentPartial + recentEmpty; 2932 track->mObservedUnderruns = underruns; 2933 // don't count underruns that occur while stopping or pausing 2934 // or stopped which can occur when flush() is called while active 2935 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2936 track->mUnderrunCount += recentUnderruns; 2937 } 2938 2939 // This is similar to the state machine for normal tracks, 2940 // with a few modifications for fast tracks. 2941 bool isActive = true; 2942 switch (track->mState) { 2943 case TrackBase::STOPPING_1: 2944 // track stays active in STOPPING_1 state until first underrun 2945 if (recentUnderruns > 0) { 2946 track->mState = TrackBase::STOPPING_2; 2947 } 2948 break; 2949 case TrackBase::PAUSING: 2950 // ramp down is not yet implemented 2951 track->setPaused(); 2952 break; 2953 case TrackBase::RESUMING: 2954 // ramp up is not yet implemented 2955 track->mState = TrackBase::ACTIVE; 2956 break; 2957 case TrackBase::ACTIVE: 2958 if (recentFull > 0 || recentPartial > 0) { 2959 // track has provided at least some frames recently: reset retry count 2960 track->mRetryCount = kMaxTrackRetries; 2961 } 2962 if (recentUnderruns == 0) { 2963 // no recent underruns: stay active 2964 break; 2965 } 2966 // there has recently been an underrun of some kind 2967 if (track->sharedBuffer() == 0) { 2968 // were any of the recent underruns "empty" (no frames available)? 2969 if (recentEmpty == 0) { 2970 // no, then ignore the partial underruns as they are allowed indefinitely 2971 break; 2972 } 2973 // there has recently been an "empty" underrun: decrement the retry counter 2974 if (--(track->mRetryCount) > 0) { 2975 break; 2976 } 2977 // indicate to client process that the track was disabled because of underrun; 2978 // it will then automatically call start() when data is available 2979 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2980 // remove from active list, but state remains ACTIVE [confusing but true] 2981 isActive = false; 2982 break; 2983 } 2984 // fall through 2985 case TrackBase::STOPPING_2: 2986 case TrackBase::PAUSED: 2987 case TrackBase::TERMINATED: 2988 case TrackBase::STOPPED: 2989 case TrackBase::FLUSHED: // flush() while active 2990 // Check for presentation complete if track is inactive 2991 // We have consumed all the buffers of this track. 2992 // This would be incomplete if we auto-paused on underrun 2993 { 2994 size_t audioHALFrames = 2995 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2996 size_t framesWritten = 2997 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2998 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2999 // track stays in active list until presentation is complete 3000 break; 3001 } 3002 } 3003 if (track->isStopping_2()) { 3004 track->mState = TrackBase::STOPPED; 3005 } 3006 if (track->isStopped()) { 3007 // Can't reset directly, as fast mixer is still polling this track 3008 // track->reset(); 3009 // So instead mark this track as needing to be reset after push with ack 3010 resetMask |= 1 << i; 3011 } 3012 isActive = false; 3013 break; 3014 case TrackBase::IDLE: 3015 default: 3016 LOG_FATAL("unexpected track state %d", track->mState); 3017 } 3018 3019 if (isActive) { 3020 // was it previously inactive? 3021 if (!(state->mTrackMask & (1 << j))) { 3022 ExtendedAudioBufferProvider *eabp = track; 3023 VolumeProvider *vp = track; 3024 fastTrack->mBufferProvider = eabp; 3025 fastTrack->mVolumeProvider = vp; 3026 fastTrack->mSampleRate = track->mSampleRate; 3027 fastTrack->mChannelMask = track->mChannelMask; 3028 fastTrack->mGeneration++; 3029 state->mTrackMask |= 1 << j; 3030 didModify = true; 3031 // no acknowledgement required for newly active tracks 3032 } 3033 // cache the combined master volume and stream type volume for fast mixer; this 3034 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3035 track->mCachedVolume = track->isMuted() ? 3036 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3037 ++fastTracks; 3038 } else { 3039 // was it previously active? 3040 if (state->mTrackMask & (1 << j)) { 3041 fastTrack->mBufferProvider = NULL; 3042 fastTrack->mGeneration++; 3043 state->mTrackMask &= ~(1 << j); 3044 didModify = true; 3045 // If any fast tracks were removed, we must wait for acknowledgement 3046 // because we're about to decrement the last sp<> on those tracks. 3047 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3048 } else { 3049 LOG_FATAL("fast track %d should have been active", j); 3050 } 3051 tracksToRemove->add(track); 3052 // Avoids a misleading display in dumpsys 3053 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3054 } 3055 continue; 3056 } 3057 3058 { // local variable scope to avoid goto warning 3059 3060 audio_track_cblk_t* cblk = track->cblk(); 3061 3062 // The first time a track is added we wait 3063 // for all its buffers to be filled before processing it 3064 int name = track->name(); 3065 // make sure that we have enough frames to mix one full buffer. 3066 // enforce this condition only once to enable draining the buffer in case the client 3067 // app does not call stop() and relies on underrun to stop: 3068 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3069 // during last round 3070 uint32_t minFrames = 1; 3071 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3072 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3073 if (t->sampleRate() == (int)mSampleRate) { 3074 minFrames = mNormalFrameCount; 3075 } else { 3076 // +1 for rounding and +1 for additional sample needed for interpolation 3077 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3078 // add frames already consumed but not yet released by the resampler 3079 // because cblk->framesReady() will include these frames 3080 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3081 // the minimum track buffer size is normally twice the number of frames necessary 3082 // to fill one buffer and the resampler should not leave more than one buffer worth 3083 // of unreleased frames after each pass, but just in case... 3084 ALOG_ASSERT(minFrames <= cblk->frameCount); 3085 } 3086 } 3087 if ((track->framesReady() >= minFrames) && track->isReady() && 3088 !track->isPaused() && !track->isTerminated()) 3089 { 3090 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3091 3092 mixedTracks++; 3093 3094 // track->mainBuffer() != mMixBuffer means there is an effect chain 3095 // connected to the track 3096 chain.clear(); 3097 if (track->mainBuffer() != mMixBuffer) { 3098 chain = getEffectChain_l(track->sessionId()); 3099 // Delegate volume control to effect in track effect chain if needed 3100 if (chain != 0) { 3101 tracksWithEffect++; 3102 } else { 3103 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3104 name, track->sessionId()); 3105 } 3106 } 3107 3108 3109 int param = AudioMixer::VOLUME; 3110 if (track->mFillingUpStatus == Track::FS_FILLED) { 3111 // no ramp for the first volume setting 3112 track->mFillingUpStatus = Track::FS_ACTIVE; 3113 if (track->mState == TrackBase::RESUMING) { 3114 track->mState = TrackBase::ACTIVE; 3115 param = AudioMixer::RAMP_VOLUME; 3116 } 3117 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3118 } else if (cblk->server != 0) { 3119 // If the track is stopped before the first frame was mixed, 3120 // do not apply ramp 3121 param = AudioMixer::RAMP_VOLUME; 3122 } 3123 3124 // compute volume for this track 3125 uint32_t vl, vr, va; 3126 if (track->isMuted() || track->isPausing() || 3127 mStreamTypes[track->streamType()].mute) { 3128 vl = vr = va = 0; 3129 if (track->isPausing()) { 3130 track->setPaused(); 3131 } 3132 } else { 3133 3134 // read original volumes with volume control 3135 float typeVolume = mStreamTypes[track->streamType()].volume; 3136 float v = masterVolume * typeVolume; 3137 uint32_t vlr = cblk->getVolumeLR(); 3138 vl = vlr & 0xFFFF; 3139 vr = vlr >> 16; 3140 // track volumes come from shared memory, so can't be trusted and must be clamped 3141 if (vl > MAX_GAIN_INT) { 3142 ALOGV("Track left volume out of range: %04X", vl); 3143 vl = MAX_GAIN_INT; 3144 } 3145 if (vr > MAX_GAIN_INT) { 3146 ALOGV("Track right volume out of range: %04X", vr); 3147 vr = MAX_GAIN_INT; 3148 } 3149 // now apply the master volume and stream type volume 3150 vl = (uint32_t)(v * vl) << 12; 3151 vr = (uint32_t)(v * vr) << 12; 3152 // assuming master volume and stream type volume each go up to 1.0, 3153 // vl and vr are now in 8.24 format 3154 3155 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3156 // send level comes from shared memory and so may be corrupt 3157 if (sendLevel > MAX_GAIN_INT) { 3158 ALOGV("Track send level out of range: %04X", sendLevel); 3159 sendLevel = MAX_GAIN_INT; 3160 } 3161 va = (uint32_t)(v * sendLevel); 3162 } 3163 // Delegate volume control to effect in track effect chain if needed 3164 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3165 // Do not ramp volume if volume is controlled by effect 3166 param = AudioMixer::VOLUME; 3167 track->mHasVolumeController = true; 3168 } else { 3169 // force no volume ramp when volume controller was just disabled or removed 3170 // from effect chain to avoid volume spike 3171 if (track->mHasVolumeController) { 3172 param = AudioMixer::VOLUME; 3173 } 3174 track->mHasVolumeController = false; 3175 } 3176 3177 // Convert volumes from 8.24 to 4.12 format 3178 // This additional clamping is needed in case chain->setVolume_l() overshot 3179 vl = (vl + (1 << 11)) >> 12; 3180 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3181 vr = (vr + (1 << 11)) >> 12; 3182 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3183 3184 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3185 3186 // XXX: these things DON'T need to be done each time 3187 mAudioMixer->setBufferProvider(name, track); 3188 mAudioMixer->enable(name); 3189 3190 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3191 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3192 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3193 mAudioMixer->setParameter( 3194 name, 3195 AudioMixer::TRACK, 3196 AudioMixer::FORMAT, (void *)track->format()); 3197 mAudioMixer->setParameter( 3198 name, 3199 AudioMixer::TRACK, 3200 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3201 mAudioMixer->setParameter( 3202 name, 3203 AudioMixer::RESAMPLE, 3204 AudioMixer::SAMPLE_RATE, 3205 (void *)(cblk->sampleRate)); 3206 mAudioMixer->setParameter( 3207 name, 3208 AudioMixer::TRACK, 3209 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3210 mAudioMixer->setParameter( 3211 name, 3212 AudioMixer::TRACK, 3213 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3214 3215 // reset retry count 3216 track->mRetryCount = kMaxTrackRetries; 3217 3218 // If one track is ready, set the mixer ready if: 3219 // - the mixer was not ready during previous round OR 3220 // - no other track is not ready 3221 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3222 mixerStatus != MIXER_TRACKS_ENABLED) { 3223 mixerStatus = MIXER_TRACKS_READY; 3224 } 3225 } else { 3226 // clear effect chain input buffer if an active track underruns to avoid sending 3227 // previous audio buffer again to effects 3228 chain = getEffectChain_l(track->sessionId()); 3229 if (chain != 0) { 3230 chain->clearInputBuffer(); 3231 } 3232 3233 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3234 if ((track->sharedBuffer() != 0) || 3235 track->isStopped() || track->isPaused()) { 3236 // We have consumed all the buffers of this track. 3237 // Remove it from the list of active tracks. 3238 // TODO: use actual buffer filling status instead of latency when available from 3239 // audio HAL 3240 size_t audioHALFrames = 3241 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3242 size_t framesWritten = 3243 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3244 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3245 if (track->isStopped()) { 3246 track->reset(); 3247 } 3248 tracksToRemove->add(track); 3249 } 3250 } else { 3251 track->mUnderrunCount++; 3252 // No buffers for this track. Give it a few chances to 3253 // fill a buffer, then remove it from active list. 3254 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3255 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3256 tracksToRemove->add(track); 3257 // indicate to client process that the track was disabled because of underrun; 3258 // it will then automatically call start() when data is available 3259 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3260 // If one track is not ready, mark the mixer also not ready if: 3261 // - the mixer was ready during previous round OR 3262 // - no other track is ready 3263 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3264 mixerStatus != MIXER_TRACKS_READY) { 3265 mixerStatus = MIXER_TRACKS_ENABLED; 3266 } 3267 } 3268 mAudioMixer->disable(name); 3269 } 3270 3271 } // local variable scope to avoid goto warning 3272track_is_ready: ; 3273 3274 } 3275 3276 // Push the new FastMixer state if necessary 3277 bool pauseAudioWatchdog = false; 3278 if (didModify) { 3279 state->mFastTracksGen++; 3280 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3281 if (kUseFastMixer == FastMixer_Dynamic && 3282 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3283 state->mCommand = FastMixerState::COLD_IDLE; 3284 state->mColdFutexAddr = &mFastMixerFutex; 3285 state->mColdGen++; 3286 mFastMixerFutex = 0; 3287 if (kUseFastMixer == FastMixer_Dynamic) { 3288 mNormalSink = mOutputSink; 3289 } 3290 // If we go into cold idle, need to wait for acknowledgement 3291 // so that fast mixer stops doing I/O. 3292 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3293 pauseAudioWatchdog = true; 3294 } 3295 sq->end(); 3296 } 3297 if (sq != NULL) { 3298 sq->end(didModify); 3299 sq->push(block); 3300 } 3301 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3302 mAudioWatchdog->pause(); 3303 } 3304 3305 // Now perform the deferred reset on fast tracks that have stopped 3306 while (resetMask != 0) { 3307 size_t i = __builtin_ctz(resetMask); 3308 ALOG_ASSERT(i < count); 3309 resetMask &= ~(1 << i); 3310 sp<Track> t = mActiveTracks[i].promote(); 3311 if (t == 0) continue; 3312 Track* track = t.get(); 3313 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3314 track->reset(); 3315 } 3316 3317 // remove all the tracks that need to be... 3318 count = tracksToRemove->size(); 3319 if (CC_UNLIKELY(count)) { 3320 for (size_t i=0 ; i<count ; i++) { 3321 const sp<Track>& track = tracksToRemove->itemAt(i); 3322 mActiveTracks.remove(track); 3323 if (track->mainBuffer() != mMixBuffer) { 3324 chain = getEffectChain_l(track->sessionId()); 3325 if (chain != 0) { 3326 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3327 chain->decActiveTrackCnt(); 3328 } 3329 } 3330 if (track->isTerminated()) { 3331 removeTrack_l(track); 3332 } 3333 } 3334 } 3335 3336 // mix buffer must be cleared if all tracks are connected to an 3337 // effect chain as in this case the mixer will not write to 3338 // mix buffer and track effects will accumulate into it 3339 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3340 // FIXME as a performance optimization, should remember previous zero status 3341 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3342 } 3343 3344 // if any fast tracks, then status is ready 3345 mMixerStatusIgnoringFastTracks = mixerStatus; 3346 if (fastTracks > 0) { 3347 mixerStatus = MIXER_TRACKS_READY; 3348 } 3349 return mixerStatus; 3350} 3351 3352/* 3353The derived values that are cached: 3354 - mixBufferSize from frame count * frame size 3355 - activeSleepTime from activeSleepTimeUs() 3356 - idleSleepTime from idleSleepTimeUs() 3357 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3358 - maxPeriod from frame count and sample rate (MIXER only) 3359 3360The parameters that affect these derived values are: 3361 - frame count 3362 - frame size 3363 - sample rate 3364 - device type: A2DP or not 3365 - device latency 3366 - format: PCM or not 3367 - active sleep time 3368 - idle sleep time 3369*/ 3370 3371void AudioFlinger::PlaybackThread::cacheParameters_l() 3372{ 3373 mixBufferSize = mNormalFrameCount * mFrameSize; 3374 activeSleepTime = activeSleepTimeUs(); 3375 idleSleepTime = idleSleepTimeUs(); 3376} 3377 3378void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3379{ 3380 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3381 this, streamType, mTracks.size()); 3382 Mutex::Autolock _l(mLock); 3383 3384 size_t size = mTracks.size(); 3385 for (size_t i = 0; i < size; i++) { 3386 sp<Track> t = mTracks[i]; 3387 if (t->streamType() == streamType) { 3388 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3389 t->mCblk->cv.signal(); 3390 } 3391 } 3392} 3393 3394// getTrackName_l() must be called with ThreadBase::mLock held 3395int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3396{ 3397 return mAudioMixer->getTrackName(channelMask); 3398} 3399 3400// deleteTrackName_l() must be called with ThreadBase::mLock held 3401void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3402{ 3403 ALOGV("remove track (%d) and delete from mixer", name); 3404 mAudioMixer->deleteTrackName(name); 3405} 3406 3407// checkForNewParameters_l() must be called with ThreadBase::mLock held 3408bool AudioFlinger::MixerThread::checkForNewParameters_l() 3409{ 3410 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3411 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3412 bool reconfig = false; 3413 3414 while (!mNewParameters.isEmpty()) { 3415 3416 if (mFastMixer != NULL) { 3417 FastMixerStateQueue *sq = mFastMixer->sq(); 3418 FastMixerState *state = sq->begin(); 3419 if (!(state->mCommand & FastMixerState::IDLE)) { 3420 previousCommand = state->mCommand; 3421 state->mCommand = FastMixerState::HOT_IDLE; 3422 sq->end(); 3423 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3424 } else { 3425 sq->end(false /*didModify*/); 3426 } 3427 } 3428 3429 status_t status = NO_ERROR; 3430 String8 keyValuePair = mNewParameters[0]; 3431 AudioParameter param = AudioParameter(keyValuePair); 3432 int value; 3433 3434 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3435 reconfig = true; 3436 } 3437 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3438 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3439 status = BAD_VALUE; 3440 } else { 3441 reconfig = true; 3442 } 3443 } 3444 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3445 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3446 status = BAD_VALUE; 3447 } else { 3448 reconfig = true; 3449 } 3450 } 3451 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3452 // do not accept frame count changes if tracks are open as the track buffer 3453 // size depends on frame count and correct behavior would not be guaranteed 3454 // if frame count is changed after track creation 3455 if (!mTracks.isEmpty()) { 3456 status = INVALID_OPERATION; 3457 } else { 3458 reconfig = true; 3459 } 3460 } 3461 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3462#ifdef ADD_BATTERY_DATA 3463 // when changing the audio output device, call addBatteryData to notify 3464 // the change 3465 if (mDevice != value) { 3466 uint32_t params = 0; 3467 // check whether speaker is on 3468 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3469 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3470 } 3471 3472 audio_devices_t deviceWithoutSpeaker 3473 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3474 // check if any other device (except speaker) is on 3475 if (value & deviceWithoutSpeaker ) { 3476 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3477 } 3478 3479 if (params != 0) { 3480 addBatteryData(params); 3481 } 3482 } 3483#endif 3484 3485 // forward device change to effects that have requested to be 3486 // aware of attached audio device. 3487 mDevice = value; 3488 for (size_t i = 0; i < mEffectChains.size(); i++) { 3489 mEffectChains[i]->setDevice_l(mDevice); 3490 } 3491 } 3492 3493 if (status == NO_ERROR) { 3494 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3495 keyValuePair.string()); 3496 if (!mStandby && status == INVALID_OPERATION) { 3497 mOutput->stream->common.standby(&mOutput->stream->common); 3498 mStandby = true; 3499 mBytesWritten = 0; 3500 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3501 keyValuePair.string()); 3502 } 3503 if (status == NO_ERROR && reconfig) { 3504 delete mAudioMixer; 3505 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3506 mAudioMixer = NULL; 3507 readOutputParameters(); 3508 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3509 for (size_t i = 0; i < mTracks.size() ; i++) { 3510 int name = getTrackName_l(mTracks[i]->mChannelMask); 3511 if (name < 0) break; 3512 mTracks[i]->mName = name; 3513 // limit track sample rate to 2 x new output sample rate 3514 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3515 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3516 } 3517 } 3518 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3519 } 3520 } 3521 3522 mNewParameters.removeAt(0); 3523 3524 mParamStatus = status; 3525 mParamCond.signal(); 3526 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3527 // already timed out waiting for the status and will never signal the condition. 3528 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3529 } 3530 3531 if (!(previousCommand & FastMixerState::IDLE)) { 3532 ALOG_ASSERT(mFastMixer != NULL); 3533 FastMixerStateQueue *sq = mFastMixer->sq(); 3534 FastMixerState *state = sq->begin(); 3535 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3536 state->mCommand = previousCommand; 3537 sq->end(); 3538 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3539 } 3540 3541 return reconfig; 3542} 3543 3544void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3545{ 3546 const size_t SIZE = 256; 3547 char buffer[SIZE]; 3548 String8 result; 3549 3550 PlaybackThread::dumpInternals(fd, args); 3551 3552 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3553 result.append(buffer); 3554 write(fd, result.string(), result.size()); 3555 3556 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3557 FastMixerDumpState copy = mFastMixerDumpState; 3558 copy.dump(fd); 3559 3560#ifdef STATE_QUEUE_DUMP 3561 // Similar for state queue 3562 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3563 observerCopy.dump(fd); 3564 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3565 mutatorCopy.dump(fd); 3566#endif 3567 3568 // Write the tee output to a .wav file 3569 NBAIO_Source *teeSource = mTeeSource.get(); 3570 if (teeSource != NULL) { 3571 char teePath[64]; 3572 struct timeval tv; 3573 gettimeofday(&tv, NULL); 3574 struct tm tm; 3575 localtime_r(&tv.tv_sec, &tm); 3576 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3577 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3578 if (teeFd >= 0) { 3579 char wavHeader[44]; 3580 memcpy(wavHeader, 3581 "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", 3582 sizeof(wavHeader)); 3583 NBAIO_Format format = teeSource->format(); 3584 unsigned channelCount = Format_channelCount(format); 3585 ALOG_ASSERT(channelCount <= FCC_2); 3586 unsigned sampleRate = Format_sampleRate(format); 3587 wavHeader[22] = channelCount; // number of channels 3588 wavHeader[24] = sampleRate; // sample rate 3589 wavHeader[25] = sampleRate >> 8; 3590 wavHeader[32] = channelCount * 2; // block alignment 3591 write(teeFd, wavHeader, sizeof(wavHeader)); 3592 size_t total = 0; 3593 bool firstRead = true; 3594 for (;;) { 3595#define TEE_SINK_READ 1024 3596 short buffer[TEE_SINK_READ * FCC_2]; 3597 size_t count = TEE_SINK_READ; 3598 ssize_t actual = teeSource->read(buffer, count); 3599 bool wasFirstRead = firstRead; 3600 firstRead = false; 3601 if (actual <= 0) { 3602 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3603 continue; 3604 } 3605 break; 3606 } 3607 ALOG_ASSERT(actual <= (ssize_t)count); 3608 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3609 total += actual; 3610 } 3611 lseek(teeFd, (off_t) 4, SEEK_SET); 3612 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3613 write(teeFd, &temp, sizeof(temp)); 3614 lseek(teeFd, (off_t) 40, SEEK_SET); 3615 temp = total * channelCount * sizeof(short); 3616 write(teeFd, &temp, sizeof(temp)); 3617 close(teeFd); 3618 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3619 } else { 3620 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3621 } 3622 } 3623 3624 if (mAudioWatchdog != 0) { 3625 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3626 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3627 wdCopy.dump(fd); 3628 } 3629} 3630 3631uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3632{ 3633 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3634} 3635 3636uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3637{ 3638 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3639} 3640 3641void AudioFlinger::MixerThread::cacheParameters_l() 3642{ 3643 PlaybackThread::cacheParameters_l(); 3644 3645 // FIXME: Relaxed timing because of a certain device that can't meet latency 3646 // Should be reduced to 2x after the vendor fixes the driver issue 3647 // increase threshold again due to low power audio mode. The way this warning 3648 // threshold is calculated and its usefulness should be reconsidered anyway. 3649 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3650} 3651 3652// ---------------------------------------------------------------------------- 3653AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3654 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3655 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3656 // mLeftVolFloat, mRightVolFloat 3657{ 3658} 3659 3660AudioFlinger::DirectOutputThread::~DirectOutputThread() 3661{ 3662} 3663 3664AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3665 Vector< sp<Track> > *tracksToRemove 3666) 3667{ 3668 sp<Track> trackToRemove; 3669 3670 mixer_state mixerStatus = MIXER_IDLE; 3671 3672 // find out which tracks need to be processed 3673 if (mActiveTracks.size() != 0) { 3674 sp<Track> t = mActiveTracks[0].promote(); 3675 // The track died recently 3676 if (t == 0) return MIXER_IDLE; 3677 3678 Track* const track = t.get(); 3679 audio_track_cblk_t* cblk = track->cblk(); 3680 3681 // The first time a track is added we wait 3682 // for all its buffers to be filled before processing it 3683 uint32_t minFrames; 3684 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3685 minFrames = mNormalFrameCount; 3686 } else { 3687 minFrames = 1; 3688 } 3689 if ((track->framesReady() >= minFrames) && track->isReady() && 3690 !track->isPaused() && !track->isTerminated()) 3691 { 3692 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3693 3694 if (track->mFillingUpStatus == Track::FS_FILLED) { 3695 track->mFillingUpStatus = Track::FS_ACTIVE; 3696 mLeftVolFloat = mRightVolFloat = 0; 3697 if (track->mState == TrackBase::RESUMING) { 3698 track->mState = TrackBase::ACTIVE; 3699 } 3700 } 3701 3702 // compute volume for this track 3703 float left, right; 3704 if (track->isMuted() || mMasterMute || track->isPausing() || 3705 mStreamTypes[track->streamType()].mute) { 3706 left = right = 0; 3707 if (track->isPausing()) { 3708 track->setPaused(); 3709 } 3710 } else { 3711 float typeVolume = mStreamTypes[track->streamType()].volume; 3712 float v = mMasterVolume * typeVolume; 3713 uint32_t vlr = cblk->getVolumeLR(); 3714 float v_clamped = v * (vlr & 0xFFFF); 3715 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3716 left = v_clamped/MAX_GAIN; 3717 v_clamped = v * (vlr >> 16); 3718 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3719 right = v_clamped/MAX_GAIN; 3720 } 3721 3722 if (left != mLeftVolFloat || right != mRightVolFloat) { 3723 mLeftVolFloat = left; 3724 mRightVolFloat = right; 3725 3726 // Convert volumes from float to 8.24 3727 uint32_t vl = (uint32_t)(left * (1 << 24)); 3728 uint32_t vr = (uint32_t)(right * (1 << 24)); 3729 3730 // Delegate volume control to effect in track effect chain if needed 3731 // only one effect chain can be present on DirectOutputThread, so if 3732 // there is one, the track is connected to it 3733 if (!mEffectChains.isEmpty()) { 3734 // Do not ramp volume if volume is controlled by effect 3735 mEffectChains[0]->setVolume_l(&vl, &vr); 3736 left = (float)vl / (1 << 24); 3737 right = (float)vr / (1 << 24); 3738 } 3739 mOutput->stream->set_volume(mOutput->stream, left, right); 3740 } 3741 3742 // reset retry count 3743 track->mRetryCount = kMaxTrackRetriesDirect; 3744 mActiveTrack = t; 3745 mixerStatus = MIXER_TRACKS_READY; 3746 } else { 3747 // clear effect chain input buffer if an active track underruns to avoid sending 3748 // previous audio buffer again to effects 3749 if (!mEffectChains.isEmpty()) { 3750 mEffectChains[0]->clearInputBuffer(); 3751 } 3752 3753 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3754 if ((track->sharedBuffer() != 0) || 3755 track->isStopped() || track->isPaused()) { 3756 // We have consumed all the buffers of this track. 3757 // Remove it from the list of active tracks. 3758 // TODO: implement behavior for compressed audio 3759 size_t audioHALFrames = 3760 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3761 size_t framesWritten = 3762 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3763 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3764 if (track->isStopped()) { 3765 track->reset(); 3766 } 3767 trackToRemove = track; 3768 } 3769 } else { 3770 // No buffers for this track. Give it a few chances to 3771 // fill a buffer, then remove it from active list. 3772 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3773 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3774 trackToRemove = track; 3775 } else { 3776 mixerStatus = MIXER_TRACKS_ENABLED; 3777 } 3778 } 3779 } 3780 } 3781 3782 // FIXME merge this with similar code for removing multiple tracks 3783 // remove all the tracks that need to be... 3784 if (CC_UNLIKELY(trackToRemove != 0)) { 3785 tracksToRemove->add(trackToRemove); 3786 mActiveTracks.remove(trackToRemove); 3787 if (!mEffectChains.isEmpty()) { 3788 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3789 trackToRemove->sessionId()); 3790 mEffectChains[0]->decActiveTrackCnt(); 3791 } 3792 if (trackToRemove->isTerminated()) { 3793 removeTrack_l(trackToRemove); 3794 } 3795 } 3796 3797 return mixerStatus; 3798} 3799 3800void AudioFlinger::DirectOutputThread::threadLoop_mix() 3801{ 3802 AudioBufferProvider::Buffer buffer; 3803 size_t frameCount = mFrameCount; 3804 int8_t *curBuf = (int8_t *)mMixBuffer; 3805 // output audio to hardware 3806 while (frameCount) { 3807 buffer.frameCount = frameCount; 3808 mActiveTrack->getNextBuffer(&buffer); 3809 if (CC_UNLIKELY(buffer.raw == NULL)) { 3810 memset(curBuf, 0, frameCount * mFrameSize); 3811 break; 3812 } 3813 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3814 frameCount -= buffer.frameCount; 3815 curBuf += buffer.frameCount * mFrameSize; 3816 mActiveTrack->releaseBuffer(&buffer); 3817 } 3818 sleepTime = 0; 3819 standbyTime = systemTime() + standbyDelay; 3820 mActiveTrack.clear(); 3821 3822} 3823 3824void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3825{ 3826 if (sleepTime == 0) { 3827 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3828 sleepTime = activeSleepTime; 3829 } else { 3830 sleepTime = idleSleepTime; 3831 } 3832 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3833 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3834 sleepTime = 0; 3835 } 3836} 3837 3838// getTrackName_l() must be called with ThreadBase::mLock held 3839int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3840{ 3841 return 0; 3842} 3843 3844// deleteTrackName_l() must be called with ThreadBase::mLock held 3845void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3846{ 3847} 3848 3849// checkForNewParameters_l() must be called with ThreadBase::mLock held 3850bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3851{ 3852 bool reconfig = false; 3853 3854 while (!mNewParameters.isEmpty()) { 3855 status_t status = NO_ERROR; 3856 String8 keyValuePair = mNewParameters[0]; 3857 AudioParameter param = AudioParameter(keyValuePair); 3858 int value; 3859 3860 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3861 // do not accept frame count changes if tracks are open as the track buffer 3862 // size depends on frame count and correct behavior would not be garantied 3863 // if frame count is changed after track creation 3864 if (!mTracks.isEmpty()) { 3865 status = INVALID_OPERATION; 3866 } else { 3867 reconfig = true; 3868 } 3869 } 3870 if (status == NO_ERROR) { 3871 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3872 keyValuePair.string()); 3873 if (!mStandby && status == INVALID_OPERATION) { 3874 mOutput->stream->common.standby(&mOutput->stream->common); 3875 mStandby = true; 3876 mBytesWritten = 0; 3877 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3878 keyValuePair.string()); 3879 } 3880 if (status == NO_ERROR && reconfig) { 3881 readOutputParameters(); 3882 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3883 } 3884 } 3885 3886 mNewParameters.removeAt(0); 3887 3888 mParamStatus = status; 3889 mParamCond.signal(); 3890 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3891 // already timed out waiting for the status and will never signal the condition. 3892 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3893 } 3894 return reconfig; 3895} 3896 3897uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3898{ 3899 uint32_t time; 3900 if (audio_is_linear_pcm(mFormat)) { 3901 time = PlaybackThread::activeSleepTimeUs(); 3902 } else { 3903 time = 10000; 3904 } 3905 return time; 3906} 3907 3908uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3909{ 3910 uint32_t time; 3911 if (audio_is_linear_pcm(mFormat)) { 3912 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3913 } else { 3914 time = 10000; 3915 } 3916 return time; 3917} 3918 3919uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3920{ 3921 uint32_t time; 3922 if (audio_is_linear_pcm(mFormat)) { 3923 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3924 } else { 3925 time = 10000; 3926 } 3927 return time; 3928} 3929 3930void AudioFlinger::DirectOutputThread::cacheParameters_l() 3931{ 3932 PlaybackThread::cacheParameters_l(); 3933 3934 // use shorter standby delay as on normal output to release 3935 // hardware resources as soon as possible 3936 standbyDelay = microseconds(activeSleepTime*2); 3937} 3938 3939// ---------------------------------------------------------------------------- 3940 3941AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3942 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3943 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3944 mWaitTimeMs(UINT_MAX) 3945{ 3946 addOutputTrack(mainThread); 3947} 3948 3949AudioFlinger::DuplicatingThread::~DuplicatingThread() 3950{ 3951 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3952 mOutputTracks[i]->destroy(); 3953 } 3954} 3955 3956void AudioFlinger::DuplicatingThread::threadLoop_mix() 3957{ 3958 // mix buffers... 3959 if (outputsReady(outputTracks)) { 3960 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3961 } else { 3962 memset(mMixBuffer, 0, mixBufferSize); 3963 } 3964 sleepTime = 0; 3965 writeFrames = mNormalFrameCount; 3966 standbyTime = systemTime() + standbyDelay; 3967} 3968 3969void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3970{ 3971 if (sleepTime == 0) { 3972 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3973 sleepTime = activeSleepTime; 3974 } else { 3975 sleepTime = idleSleepTime; 3976 } 3977 } else if (mBytesWritten != 0) { 3978 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3979 writeFrames = mNormalFrameCount; 3980 memset(mMixBuffer, 0, mixBufferSize); 3981 } else { 3982 // flush remaining overflow buffers in output tracks 3983 writeFrames = 0; 3984 } 3985 sleepTime = 0; 3986 } 3987} 3988 3989void AudioFlinger::DuplicatingThread::threadLoop_write() 3990{ 3991 for (size_t i = 0; i < outputTracks.size(); i++) { 3992 outputTracks[i]->write(mMixBuffer, writeFrames); 3993 } 3994 mBytesWritten += mixBufferSize; 3995} 3996 3997void AudioFlinger::DuplicatingThread::threadLoop_standby() 3998{ 3999 // DuplicatingThread implements standby by stopping all tracks 4000 for (size_t i = 0; i < outputTracks.size(); i++) { 4001 outputTracks[i]->stop(); 4002 } 4003} 4004 4005void AudioFlinger::DuplicatingThread::saveOutputTracks() 4006{ 4007 outputTracks = mOutputTracks; 4008} 4009 4010void AudioFlinger::DuplicatingThread::clearOutputTracks() 4011{ 4012 outputTracks.clear(); 4013} 4014 4015void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4016{ 4017 Mutex::Autolock _l(mLock); 4018 // FIXME explain this formula 4019 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4020 OutputTrack *outputTrack = new OutputTrack(thread, 4021 this, 4022 mSampleRate, 4023 mFormat, 4024 mChannelMask, 4025 frameCount); 4026 if (outputTrack->cblk() != NULL) { 4027 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4028 mOutputTracks.add(outputTrack); 4029 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4030 updateWaitTime_l(); 4031 } 4032} 4033 4034void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4035{ 4036 Mutex::Autolock _l(mLock); 4037 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4038 if (mOutputTracks[i]->thread() == thread) { 4039 mOutputTracks[i]->destroy(); 4040 mOutputTracks.removeAt(i); 4041 updateWaitTime_l(); 4042 return; 4043 } 4044 } 4045 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4046} 4047 4048// caller must hold mLock 4049void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4050{ 4051 mWaitTimeMs = UINT_MAX; 4052 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4053 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4054 if (strong != 0) { 4055 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4056 if (waitTimeMs < mWaitTimeMs) { 4057 mWaitTimeMs = waitTimeMs; 4058 } 4059 } 4060 } 4061} 4062 4063 4064bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4065{ 4066 for (size_t i = 0; i < outputTracks.size(); i++) { 4067 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4068 if (thread == 0) { 4069 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4070 return false; 4071 } 4072 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4073 // see note at standby() declaration 4074 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4075 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4076 return false; 4077 } 4078 } 4079 return true; 4080} 4081 4082uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4083{ 4084 return (mWaitTimeMs * 1000) / 2; 4085} 4086 4087void AudioFlinger::DuplicatingThread::cacheParameters_l() 4088{ 4089 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4090 updateWaitTime_l(); 4091 4092 MixerThread::cacheParameters_l(); 4093} 4094 4095// ---------------------------------------------------------------------------- 4096 4097// TrackBase constructor must be called with AudioFlinger::mLock held 4098AudioFlinger::ThreadBase::TrackBase::TrackBase( 4099 ThreadBase *thread, 4100 const sp<Client>& client, 4101 uint32_t sampleRate, 4102 audio_format_t format, 4103 audio_channel_mask_t channelMask, 4104 int frameCount, 4105 const sp<IMemory>& sharedBuffer, 4106 int sessionId) 4107 : RefBase(), 4108 mThread(thread), 4109 mClient(client), 4110 mCblk(NULL), 4111 // mBuffer 4112 // mBufferEnd 4113 mFrameCount(0), 4114 mState(IDLE), 4115 mSampleRate(sampleRate), 4116 mFormat(format), 4117 mStepServerFailed(false), 4118 mSessionId(sessionId) 4119 // mChannelCount 4120 // mChannelMask 4121{ 4122 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4123 4124 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4125 size_t size = sizeof(audio_track_cblk_t); 4126 uint8_t channelCount = popcount(channelMask); 4127 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4128 if (sharedBuffer == 0) { 4129 size += bufferSize; 4130 } 4131 4132 if (client != NULL) { 4133 mCblkMemory = client->heap()->allocate(size); 4134 if (mCblkMemory != 0) { 4135 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4136 if (mCblk != NULL) { // construct the shared structure in-place. 4137 new(mCblk) audio_track_cblk_t(); 4138 // clear all buffers 4139 mCblk->frameCount = frameCount; 4140 mCblk->sampleRate = sampleRate; 4141// uncomment the following lines to quickly test 32-bit wraparound 4142// mCblk->user = 0xffff0000; 4143// mCblk->server = 0xffff0000; 4144// mCblk->userBase = 0xffff0000; 4145// mCblk->serverBase = 0xffff0000; 4146 mChannelCount = channelCount; 4147 mChannelMask = channelMask; 4148 if (sharedBuffer == 0) { 4149 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4150 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4151 // Force underrun condition to avoid false underrun callback until first data is 4152 // written to buffer (other flags are cleared) 4153 mCblk->flags = CBLK_UNDERRUN_ON; 4154 } else { 4155 mBuffer = sharedBuffer->pointer(); 4156 } 4157 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4158 } 4159 } else { 4160 ALOGE("not enough memory for AudioTrack size=%u", size); 4161 client->heap()->dump("AudioTrack"); 4162 return; 4163 } 4164 } else { 4165 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4166 // construct the shared structure in-place. 4167 new(mCblk) audio_track_cblk_t(); 4168 // clear all buffers 4169 mCblk->frameCount = frameCount; 4170 mCblk->sampleRate = sampleRate; 4171// uncomment the following lines to quickly test 32-bit wraparound 4172// mCblk->user = 0xffff0000; 4173// mCblk->server = 0xffff0000; 4174// mCblk->userBase = 0xffff0000; 4175// mCblk->serverBase = 0xffff0000; 4176 mChannelCount = channelCount; 4177 mChannelMask = channelMask; 4178 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4179 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4180 // Force underrun condition to avoid false underrun callback until first data is 4181 // written to buffer (other flags are cleared) 4182 mCblk->flags = CBLK_UNDERRUN_ON; 4183 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4184 } 4185} 4186 4187AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4188{ 4189 if (mCblk != NULL) { 4190 if (mClient == 0) { 4191 delete mCblk; 4192 } else { 4193 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4194 } 4195 } 4196 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4197 if (mClient != 0) { 4198 // Client destructor must run with AudioFlinger mutex locked 4199 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4200 // If the client's reference count drops to zero, the associated destructor 4201 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4202 // relying on the automatic clear() at end of scope. 4203 mClient.clear(); 4204 } 4205} 4206 4207// AudioBufferProvider interface 4208// getNextBuffer() = 0; 4209// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4210void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4211{ 4212 buffer->raw = NULL; 4213 mFrameCount = buffer->frameCount; 4214 // FIXME See note at getNextBuffer() 4215 (void) step(); // ignore return value of step() 4216 buffer->frameCount = 0; 4217} 4218 4219bool AudioFlinger::ThreadBase::TrackBase::step() { 4220 bool result; 4221 audio_track_cblk_t* cblk = this->cblk(); 4222 4223 result = cblk->stepServer(mFrameCount); 4224 if (!result) { 4225 ALOGV("stepServer failed acquiring cblk mutex"); 4226 mStepServerFailed = true; 4227 } 4228 return result; 4229} 4230 4231void AudioFlinger::ThreadBase::TrackBase::reset() { 4232 audio_track_cblk_t* cblk = this->cblk(); 4233 4234 cblk->user = 0; 4235 cblk->server = 0; 4236 cblk->userBase = 0; 4237 cblk->serverBase = 0; 4238 mStepServerFailed = false; 4239 ALOGV("TrackBase::reset"); 4240} 4241 4242int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4243 return (int)mCblk->sampleRate; 4244} 4245 4246void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4247 audio_track_cblk_t* cblk = this->cblk(); 4248 size_t frameSize = cblk->frameSize; 4249 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4250 int8_t *bufferEnd = bufferStart + frames * frameSize; 4251 4252 // Check validity of returned pointer in case the track control block would have been corrupted. 4253 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4254 "TrackBase::getBuffer buffer out of range:\n" 4255 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4256 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4257 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4258 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4259 4260 return bufferStart; 4261} 4262 4263status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4264{ 4265 mSyncEvents.add(event); 4266 return NO_ERROR; 4267} 4268 4269// ---------------------------------------------------------------------------- 4270 4271// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4272AudioFlinger::PlaybackThread::Track::Track( 4273 PlaybackThread *thread, 4274 const sp<Client>& client, 4275 audio_stream_type_t streamType, 4276 uint32_t sampleRate, 4277 audio_format_t format, 4278 audio_channel_mask_t channelMask, 4279 int frameCount, 4280 const sp<IMemory>& sharedBuffer, 4281 int sessionId, 4282 IAudioFlinger::track_flags_t flags) 4283 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4284 mMute(false), 4285 mFillingUpStatus(FS_INVALID), 4286 // mRetryCount initialized later when needed 4287 mSharedBuffer(sharedBuffer), 4288 mStreamType(streamType), 4289 mName(-1), // see note below 4290 mMainBuffer(thread->mixBuffer()), 4291 mAuxBuffer(NULL), 4292 mAuxEffectId(0), mHasVolumeController(false), 4293 mPresentationCompleteFrames(0), 4294 mFlags(flags), 4295 mFastIndex(-1), 4296 mUnderrunCount(0), 4297 mCachedVolume(1.0) 4298{ 4299 if (mCblk != NULL) { 4300 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4301 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4302 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4303 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4304 mName = thread->getTrackName_l(channelMask); 4305 mCblk->mName = mName; 4306 if (mName < 0) { 4307 ALOGE("no more track names available"); 4308 return; 4309 } 4310 // only allocate a fast track index if we were able to allocate a normal track name 4311 if (flags & IAudioFlinger::TRACK_FAST) { 4312 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4313 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4314 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4315 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4316 // FIXME This is too eager. We allocate a fast track index before the 4317 // fast track becomes active. Since fast tracks are a scarce resource, 4318 // this means we are potentially denying other more important fast tracks from 4319 // being created. It would be better to allocate the index dynamically. 4320 mFastIndex = i; 4321 mCblk->mName = i; 4322 // Read the initial underruns because this field is never cleared by the fast mixer 4323 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4324 thread->mFastTrackAvailMask &= ~(1 << i); 4325 } 4326 } 4327 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4328} 4329 4330AudioFlinger::PlaybackThread::Track::~Track() 4331{ 4332 ALOGV("PlaybackThread::Track destructor"); 4333} 4334 4335void AudioFlinger::PlaybackThread::Track::destroy() 4336{ 4337 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4338 // by removing it from mTracks vector, so there is a risk that this Tracks's 4339 // destructor is called. As the destructor needs to lock mLock, 4340 // we must acquire a strong reference on this Track before locking mLock 4341 // here so that the destructor is called only when exiting this function. 4342 // On the other hand, as long as Track::destroy() is only called by 4343 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4344 // this Track with its member mTrack. 4345 sp<Track> keep(this); 4346 { // scope for mLock 4347 sp<ThreadBase> thread = mThread.promote(); 4348 if (thread != 0) { 4349 if (!isOutputTrack()) { 4350 if (mState == ACTIVE || mState == RESUMING) { 4351 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4352 4353#ifdef ADD_BATTERY_DATA 4354 // to track the speaker usage 4355 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4356#endif 4357 } 4358 AudioSystem::releaseOutput(thread->id()); 4359 } 4360 Mutex::Autolock _l(thread->mLock); 4361 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4362 playbackThread->destroyTrack_l(this); 4363 } 4364 } 4365} 4366 4367/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4368{ 4369 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4370 " Server User Main buf Aux Buf Flags Underruns\n"); 4371} 4372 4373void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4374{ 4375 uint32_t vlr = mCblk->getVolumeLR(); 4376 if (isFastTrack()) { 4377 sprintf(buffer, " F %2d", mFastIndex); 4378 } else { 4379 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4380 } 4381 track_state state = mState; 4382 char stateChar; 4383 switch (state) { 4384 case IDLE: 4385 stateChar = 'I'; 4386 break; 4387 case TERMINATED: 4388 stateChar = 'T'; 4389 break; 4390 case STOPPING_1: 4391 stateChar = 's'; 4392 break; 4393 case STOPPING_2: 4394 stateChar = '5'; 4395 break; 4396 case STOPPED: 4397 stateChar = 'S'; 4398 break; 4399 case RESUMING: 4400 stateChar = 'R'; 4401 break; 4402 case ACTIVE: 4403 stateChar = 'A'; 4404 break; 4405 case PAUSING: 4406 stateChar = 'p'; 4407 break; 4408 case PAUSED: 4409 stateChar = 'P'; 4410 break; 4411 case FLUSHED: 4412 stateChar = 'F'; 4413 break; 4414 default: 4415 stateChar = '?'; 4416 break; 4417 } 4418 char nowInUnderrun; 4419 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4420 case UNDERRUN_FULL: 4421 nowInUnderrun = ' '; 4422 break; 4423 case UNDERRUN_PARTIAL: 4424 nowInUnderrun = '<'; 4425 break; 4426 case UNDERRUN_EMPTY: 4427 nowInUnderrun = '*'; 4428 break; 4429 default: 4430 nowInUnderrun = '?'; 4431 break; 4432 } 4433 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4434 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4435 (mClient == 0) ? getpid_cached : mClient->pid(), 4436 mStreamType, 4437 mFormat, 4438 mChannelMask, 4439 mSessionId, 4440 mFrameCount, 4441 mCblk->frameCount, 4442 stateChar, 4443 mMute, 4444 mFillingUpStatus, 4445 mCblk->sampleRate, 4446 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4447 20.0 * log10((vlr >> 16) / 4096.0), 4448 mCblk->server, 4449 mCblk->user, 4450 (int)mMainBuffer, 4451 (int)mAuxBuffer, 4452 mCblk->flags, 4453 mUnderrunCount, 4454 nowInUnderrun); 4455} 4456 4457// AudioBufferProvider interface 4458status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4459 AudioBufferProvider::Buffer* buffer, int64_t pts) 4460{ 4461 audio_track_cblk_t* cblk = this->cblk(); 4462 uint32_t framesReady; 4463 uint32_t framesReq = buffer->frameCount; 4464 4465 // Check if last stepServer failed, try to step now 4466 if (mStepServerFailed) { 4467 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4468 // Since the fast mixer is higher priority than client callback thread, 4469 // it does not result in priority inversion for client. 4470 // But a non-blocking solution would be preferable to avoid 4471 // fast mixer being unable to tryLock(), and 4472 // to avoid the extra context switches if the client wakes up, 4473 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4474 if (!step()) goto getNextBuffer_exit; 4475 ALOGV("stepServer recovered"); 4476 mStepServerFailed = false; 4477 } 4478 4479 // FIXME Same as above 4480 framesReady = cblk->framesReady(); 4481 4482 if (CC_LIKELY(framesReady)) { 4483 uint32_t s = cblk->server; 4484 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4485 4486 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4487 if (framesReq > framesReady) { 4488 framesReq = framesReady; 4489 } 4490 if (framesReq > bufferEnd - s) { 4491 framesReq = bufferEnd - s; 4492 } 4493 4494 buffer->raw = getBuffer(s, framesReq); 4495 buffer->frameCount = framesReq; 4496 return NO_ERROR; 4497 } 4498 4499getNextBuffer_exit: 4500 buffer->raw = NULL; 4501 buffer->frameCount = 0; 4502 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4503 return NOT_ENOUGH_DATA; 4504} 4505 4506// Note that framesReady() takes a mutex on the control block using tryLock(). 4507// This could result in priority inversion if framesReady() is called by the normal mixer, 4508// as the normal mixer thread runs at lower 4509// priority than the client's callback thread: there is a short window within framesReady() 4510// during which the normal mixer could be preempted, and the client callback would block. 4511// Another problem can occur if framesReady() is called by the fast mixer: 4512// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4513// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4514size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4515 return mCblk->framesReady(); 4516} 4517 4518// Don't call for fast tracks; the framesReady() could result in priority inversion 4519bool AudioFlinger::PlaybackThread::Track::isReady() const { 4520 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4521 4522 if (framesReady() >= mCblk->frameCount || 4523 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4524 mFillingUpStatus = FS_FILLED; 4525 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4526 return true; 4527 } 4528 return false; 4529} 4530 4531status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4532 int triggerSession) 4533{ 4534 status_t status = NO_ERROR; 4535 ALOGV("start(%d), calling pid %d session %d", 4536 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4537 4538 sp<ThreadBase> thread = mThread.promote(); 4539 if (thread != 0) { 4540 Mutex::Autolock _l(thread->mLock); 4541 track_state state = mState; 4542 // here the track could be either new, or restarted 4543 // in both cases "unstop" the track 4544 if (mState == PAUSED) { 4545 mState = TrackBase::RESUMING; 4546 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4547 } else { 4548 mState = TrackBase::ACTIVE; 4549 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4550 } 4551 4552 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4553 thread->mLock.unlock(); 4554 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4555 thread->mLock.lock(); 4556 4557#ifdef ADD_BATTERY_DATA 4558 // to track the speaker usage 4559 if (status == NO_ERROR) { 4560 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4561 } 4562#endif 4563 } 4564 if (status == NO_ERROR) { 4565 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4566 playbackThread->addTrack_l(this); 4567 } else { 4568 mState = state; 4569 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4570 } 4571 } else { 4572 status = BAD_VALUE; 4573 } 4574 return status; 4575} 4576 4577void AudioFlinger::PlaybackThread::Track::stop() 4578{ 4579 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4580 sp<ThreadBase> thread = mThread.promote(); 4581 if (thread != 0) { 4582 Mutex::Autolock _l(thread->mLock); 4583 track_state state = mState; 4584 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4585 // If the track is not active (PAUSED and buffers full), flush buffers 4586 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4587 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4588 reset(); 4589 mState = STOPPED; 4590 } else if (!isFastTrack()) { 4591 mState = STOPPED; 4592 } else { 4593 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4594 // and then to STOPPED and reset() when presentation is complete 4595 mState = STOPPING_1; 4596 } 4597 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4598 } 4599 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4600 thread->mLock.unlock(); 4601 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4602 thread->mLock.lock(); 4603 4604#ifdef ADD_BATTERY_DATA 4605 // to track the speaker usage 4606 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4607#endif 4608 } 4609 } 4610} 4611 4612void AudioFlinger::PlaybackThread::Track::pause() 4613{ 4614 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4615 sp<ThreadBase> thread = mThread.promote(); 4616 if (thread != 0) { 4617 Mutex::Autolock _l(thread->mLock); 4618 if (mState == ACTIVE || mState == RESUMING) { 4619 mState = PAUSING; 4620 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4621 if (!isOutputTrack()) { 4622 thread->mLock.unlock(); 4623 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4624 thread->mLock.lock(); 4625 4626#ifdef ADD_BATTERY_DATA 4627 // to track the speaker usage 4628 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4629#endif 4630 } 4631 } 4632 } 4633} 4634 4635void AudioFlinger::PlaybackThread::Track::flush() 4636{ 4637 ALOGV("flush(%d)", mName); 4638 sp<ThreadBase> thread = mThread.promote(); 4639 if (thread != 0) { 4640 Mutex::Autolock _l(thread->mLock); 4641 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4642 mState != PAUSING) { 4643 return; 4644 } 4645 // No point remaining in PAUSED state after a flush => go to 4646 // FLUSHED state 4647 mState = FLUSHED; 4648 // do not reset the track if it is still in the process of being stopped or paused. 4649 // this will be done by prepareTracks_l() when the track is stopped. 4650 // prepareTracks_l() will see mState == FLUSHED, then 4651 // remove from active track list, reset(), and trigger presentation complete 4652 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4653 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4654 reset(); 4655 } 4656 } 4657} 4658 4659void AudioFlinger::PlaybackThread::Track::reset() 4660{ 4661 // Do not reset twice to avoid discarding data written just after a flush and before 4662 // the audioflinger thread detects the track is stopped. 4663 if (!mResetDone) { 4664 TrackBase::reset(); 4665 // Force underrun condition to avoid false underrun callback until first data is 4666 // written to buffer 4667 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4668 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4669 mFillingUpStatus = FS_FILLING; 4670 mResetDone = true; 4671 if (mState == FLUSHED) { 4672 mState = IDLE; 4673 } 4674 } 4675} 4676 4677void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4678{ 4679 mMute = muted; 4680} 4681 4682status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4683{ 4684 status_t status = DEAD_OBJECT; 4685 sp<ThreadBase> thread = mThread.promote(); 4686 if (thread != 0) { 4687 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4688 sp<AudioFlinger> af = mClient->audioFlinger(); 4689 4690 Mutex::Autolock _l(af->mLock); 4691 4692 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4693 4694 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4695 Mutex::Autolock _dl(playbackThread->mLock); 4696 Mutex::Autolock _sl(srcThread->mLock); 4697 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4698 if (chain == 0) { 4699 return INVALID_OPERATION; 4700 } 4701 4702 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4703 if (effect == 0) { 4704 return INVALID_OPERATION; 4705 } 4706 srcThread->removeEffect_l(effect); 4707 playbackThread->addEffect_l(effect); 4708 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4709 if (effect->state() == EffectModule::ACTIVE || 4710 effect->state() == EffectModule::STOPPING) { 4711 effect->start(); 4712 } 4713 4714 sp<EffectChain> dstChain = effect->chain().promote(); 4715 if (dstChain == 0) { 4716 srcThread->addEffect_l(effect); 4717 return INVALID_OPERATION; 4718 } 4719 AudioSystem::unregisterEffect(effect->id()); 4720 AudioSystem::registerEffect(&effect->desc(), 4721 srcThread->id(), 4722 dstChain->strategy(), 4723 AUDIO_SESSION_OUTPUT_MIX, 4724 effect->id()); 4725 } 4726 status = playbackThread->attachAuxEffect(this, EffectId); 4727 } 4728 return status; 4729} 4730 4731void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4732{ 4733 mAuxEffectId = EffectId; 4734 mAuxBuffer = buffer; 4735} 4736 4737bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4738 size_t audioHalFrames) 4739{ 4740 // a track is considered presented when the total number of frames written to audio HAL 4741 // corresponds to the number of frames written when presentationComplete() is called for the 4742 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4743 if (mPresentationCompleteFrames == 0) { 4744 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4745 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4746 mPresentationCompleteFrames, audioHalFrames); 4747 } 4748 if (framesWritten >= mPresentationCompleteFrames) { 4749 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4750 mSessionId, framesWritten); 4751 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4752 return true; 4753 } 4754 return false; 4755} 4756 4757void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4758{ 4759 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4760 if (mSyncEvents[i]->type() == type) { 4761 mSyncEvents[i]->trigger(); 4762 mSyncEvents.removeAt(i); 4763 i--; 4764 } 4765 } 4766} 4767 4768// implement VolumeBufferProvider interface 4769 4770uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4771{ 4772 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4773 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4774 uint32_t vlr = mCblk->getVolumeLR(); 4775 uint32_t vl = vlr & 0xFFFF; 4776 uint32_t vr = vlr >> 16; 4777 // track volumes come from shared memory, so can't be trusted and must be clamped 4778 if (vl > MAX_GAIN_INT) { 4779 vl = MAX_GAIN_INT; 4780 } 4781 if (vr > MAX_GAIN_INT) { 4782 vr = MAX_GAIN_INT; 4783 } 4784 // now apply the cached master volume and stream type volume; 4785 // this is trusted but lacks any synchronization or barrier so may be stale 4786 float v = mCachedVolume; 4787 vl *= v; 4788 vr *= v; 4789 // re-combine into U4.16 4790 vlr = (vr << 16) | (vl & 0xFFFF); 4791 // FIXME look at mute, pause, and stop flags 4792 return vlr; 4793} 4794 4795status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4796{ 4797 if (mState == TERMINATED || mState == PAUSED || 4798 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4799 (mState == STOPPED)))) { 4800 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4801 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4802 event->cancel(); 4803 return INVALID_OPERATION; 4804 } 4805 TrackBase::setSyncEvent(event); 4806 return NO_ERROR; 4807} 4808 4809// timed audio tracks 4810 4811sp<AudioFlinger::PlaybackThread::TimedTrack> 4812AudioFlinger::PlaybackThread::TimedTrack::create( 4813 PlaybackThread *thread, 4814 const sp<Client>& client, 4815 audio_stream_type_t streamType, 4816 uint32_t sampleRate, 4817 audio_format_t format, 4818 audio_channel_mask_t channelMask, 4819 int frameCount, 4820 const sp<IMemory>& sharedBuffer, 4821 int sessionId) { 4822 if (!client->reserveTimedTrack()) 4823 return 0; 4824 4825 return new TimedTrack( 4826 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4827 sharedBuffer, sessionId); 4828} 4829 4830AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4831 PlaybackThread *thread, 4832 const sp<Client>& client, 4833 audio_stream_type_t streamType, 4834 uint32_t sampleRate, 4835 audio_format_t format, 4836 audio_channel_mask_t channelMask, 4837 int frameCount, 4838 const sp<IMemory>& sharedBuffer, 4839 int sessionId) 4840 : Track(thread, client, streamType, sampleRate, format, channelMask, 4841 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4842 mQueueHeadInFlight(false), 4843 mTrimQueueHeadOnRelease(false), 4844 mFramesPendingInQueue(0), 4845 mTimedSilenceBuffer(NULL), 4846 mTimedSilenceBufferSize(0), 4847 mTimedAudioOutputOnTime(false), 4848 mMediaTimeTransformValid(false) 4849{ 4850 LocalClock lc; 4851 mLocalTimeFreq = lc.getLocalFreq(); 4852 4853 mLocalTimeToSampleTransform.a_zero = 0; 4854 mLocalTimeToSampleTransform.b_zero = 0; 4855 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4856 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4857 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4858 &mLocalTimeToSampleTransform.a_to_b_denom); 4859 4860 mMediaTimeToSampleTransform.a_zero = 0; 4861 mMediaTimeToSampleTransform.b_zero = 0; 4862 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4863 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4864 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4865 &mMediaTimeToSampleTransform.a_to_b_denom); 4866} 4867 4868AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4869 mClient->releaseTimedTrack(); 4870 delete [] mTimedSilenceBuffer; 4871} 4872 4873status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4874 size_t size, sp<IMemory>* buffer) { 4875 4876 Mutex::Autolock _l(mTimedBufferQueueLock); 4877 4878 trimTimedBufferQueue_l(); 4879 4880 // lazily initialize the shared memory heap for timed buffers 4881 if (mTimedMemoryDealer == NULL) { 4882 const int kTimedBufferHeapSize = 512 << 10; 4883 4884 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4885 "AudioFlingerTimed"); 4886 if (mTimedMemoryDealer == NULL) 4887 return NO_MEMORY; 4888 } 4889 4890 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4891 if (newBuffer == NULL) { 4892 newBuffer = mTimedMemoryDealer->allocate(size); 4893 if (newBuffer == NULL) 4894 return NO_MEMORY; 4895 } 4896 4897 *buffer = newBuffer; 4898 return NO_ERROR; 4899} 4900 4901// caller must hold mTimedBufferQueueLock 4902void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4903 int64_t mediaTimeNow; 4904 { 4905 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4906 if (!mMediaTimeTransformValid) 4907 return; 4908 4909 int64_t targetTimeNow; 4910 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4911 ? mCCHelper.getCommonTime(&targetTimeNow) 4912 : mCCHelper.getLocalTime(&targetTimeNow); 4913 4914 if (OK != res) 4915 return; 4916 4917 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4918 &mediaTimeNow)) { 4919 return; 4920 } 4921 } 4922 4923 size_t trimEnd; 4924 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4925 int64_t bufEnd; 4926 4927 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4928 // We have a next buffer. Just use its PTS as the PTS of the frame 4929 // following the last frame in this buffer. If the stream is sparse 4930 // (ie, there are deliberate gaps left in the stream which should be 4931 // filled with silence by the TimedAudioTrack), then this can result 4932 // in one extra buffer being left un-trimmed when it could have 4933 // been. In general, this is not typical, and we would rather 4934 // optimized away the TS calculation below for the more common case 4935 // where PTSes are contiguous. 4936 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4937 } else { 4938 // We have no next buffer. Compute the PTS of the frame following 4939 // the last frame in this buffer by computing the duration of of 4940 // this frame in media time units and adding it to the PTS of the 4941 // buffer. 4942 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4943 / mCblk->frameSize; 4944 4945 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4946 &bufEnd)) { 4947 ALOGE("Failed to convert frame count of %lld to media time" 4948 " duration" " (scale factor %d/%u) in %s", 4949 frameCount, 4950 mMediaTimeToSampleTransform.a_to_b_numer, 4951 mMediaTimeToSampleTransform.a_to_b_denom, 4952 __PRETTY_FUNCTION__); 4953 break; 4954 } 4955 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4956 } 4957 4958 if (bufEnd > mediaTimeNow) 4959 break; 4960 4961 // Is the buffer we want to use in the middle of a mix operation right 4962 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4963 // from the mixer which should be coming back shortly. 4964 if (!trimEnd && mQueueHeadInFlight) { 4965 mTrimQueueHeadOnRelease = true; 4966 } 4967 } 4968 4969 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4970 if (trimStart < trimEnd) { 4971 // Update the bookkeeping for framesReady() 4972 for (size_t i = trimStart; i < trimEnd; ++i) { 4973 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4974 } 4975 4976 // Now actually remove the buffers from the queue. 4977 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4978 } 4979} 4980 4981void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4982 const char* logTag) { 4983 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4984 "%s called (reason \"%s\"), but timed buffer queue has no" 4985 " elements to trim.", __FUNCTION__, logTag); 4986 4987 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4988 mTimedBufferQueue.removeAt(0); 4989} 4990 4991void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4992 const TimedBuffer& buf, 4993 const char* logTag) { 4994 uint32_t bufBytes = buf.buffer()->size(); 4995 uint32_t consumedAlready = buf.position(); 4996 4997 ALOG_ASSERT(consumedAlready <= bufBytes, 4998 "Bad bookkeeping while updating frames pending. Timed buffer is" 4999 " only %u bytes long, but claims to have consumed %u" 5000 " bytes. (update reason: \"%s\")", 5001 bufBytes, consumedAlready, logTag); 5002 5003 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 5004 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5005 "Bad bookkeeping while updating frames pending. Should have at" 5006 " least %u queued frames, but we think we have only %u. (update" 5007 " reason: \"%s\")", 5008 bufFrames, mFramesPendingInQueue, logTag); 5009 5010 mFramesPendingInQueue -= bufFrames; 5011} 5012 5013status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5014 const sp<IMemory>& buffer, int64_t pts) { 5015 5016 { 5017 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5018 if (!mMediaTimeTransformValid) 5019 return INVALID_OPERATION; 5020 } 5021 5022 Mutex::Autolock _l(mTimedBufferQueueLock); 5023 5024 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5025 mFramesPendingInQueue += bufFrames; 5026 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5027 5028 return NO_ERROR; 5029} 5030 5031status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5032 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5033 5034 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5035 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5036 target); 5037 5038 if (!(target == TimedAudioTrack::LOCAL_TIME || 5039 target == TimedAudioTrack::COMMON_TIME)) { 5040 return BAD_VALUE; 5041 } 5042 5043 Mutex::Autolock lock(mMediaTimeTransformLock); 5044 mMediaTimeTransform = xform; 5045 mMediaTimeTransformTarget = target; 5046 mMediaTimeTransformValid = true; 5047 5048 return NO_ERROR; 5049} 5050 5051#define min(a, b) ((a) < (b) ? (a) : (b)) 5052 5053// implementation of getNextBuffer for tracks whose buffers have timestamps 5054status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5055 AudioBufferProvider::Buffer* buffer, int64_t pts) 5056{ 5057 if (pts == AudioBufferProvider::kInvalidPTS) { 5058 buffer->raw = NULL; 5059 buffer->frameCount = 0; 5060 mTimedAudioOutputOnTime = false; 5061 return INVALID_OPERATION; 5062 } 5063 5064 Mutex::Autolock _l(mTimedBufferQueueLock); 5065 5066 ALOG_ASSERT(!mQueueHeadInFlight, 5067 "getNextBuffer called without releaseBuffer!"); 5068 5069 while (true) { 5070 5071 // if we have no timed buffers, then fail 5072 if (mTimedBufferQueue.isEmpty()) { 5073 buffer->raw = NULL; 5074 buffer->frameCount = 0; 5075 return NOT_ENOUGH_DATA; 5076 } 5077 5078 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5079 5080 // calculate the PTS of the head of the timed buffer queue expressed in 5081 // local time 5082 int64_t headLocalPTS; 5083 { 5084 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5085 5086 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5087 5088 if (mMediaTimeTransform.a_to_b_denom == 0) { 5089 // the transform represents a pause, so yield silence 5090 timedYieldSilence_l(buffer->frameCount, buffer); 5091 return NO_ERROR; 5092 } 5093 5094 int64_t transformedPTS; 5095 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5096 &transformedPTS)) { 5097 // the transform failed. this shouldn't happen, but if it does 5098 // then just drop this buffer 5099 ALOGW("timedGetNextBuffer transform failed"); 5100 buffer->raw = NULL; 5101 buffer->frameCount = 0; 5102 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5103 return NO_ERROR; 5104 } 5105 5106 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5107 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5108 &headLocalPTS)) { 5109 buffer->raw = NULL; 5110 buffer->frameCount = 0; 5111 return INVALID_OPERATION; 5112 } 5113 } else { 5114 headLocalPTS = transformedPTS; 5115 } 5116 } 5117 5118 // adjust the head buffer's PTS to reflect the portion of the head buffer 5119 // that has already been consumed 5120 int64_t effectivePTS = headLocalPTS + 5121 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5122 5123 // Calculate the delta in samples between the head of the input buffer 5124 // queue and the start of the next output buffer that will be written. 5125 // If the transformation fails because of over or underflow, it means 5126 // that the sample's position in the output stream is so far out of 5127 // whack that it should just be dropped. 5128 int64_t sampleDelta; 5129 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5130 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5131 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5132 " mix"); 5133 continue; 5134 } 5135 if (!mLocalTimeToSampleTransform.doForwardTransform( 5136 (effectivePTS - pts) << 32, &sampleDelta)) { 5137 ALOGV("*** too late during sample rate transform: dropped buffer"); 5138 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5139 continue; 5140 } 5141 5142 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5143 " sampleDelta=[%d.%08x]", 5144 head.pts(), head.position(), pts, 5145 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5146 + (sampleDelta >> 32)), 5147 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5148 5149 // if the delta between the ideal placement for the next input sample and 5150 // the current output position is within this threshold, then we will 5151 // concatenate the next input samples to the previous output 5152 const int64_t kSampleContinuityThreshold = 5153 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5154 5155 // if this is the first buffer of audio that we're emitting from this track 5156 // then it should be almost exactly on time. 5157 const int64_t kSampleStartupThreshold = 1LL << 32; 5158 5159 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5160 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5161 // the next input is close enough to being on time, so concatenate it 5162 // with the last output 5163 timedYieldSamples_l(buffer); 5164 5165 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5166 head.position(), buffer->frameCount); 5167 return NO_ERROR; 5168 } 5169 5170 // Looks like our output is not on time. Reset our on timed status. 5171 // Next time we mix samples from our input queue, then should be within 5172 // the StartupThreshold. 5173 mTimedAudioOutputOnTime = false; 5174 if (sampleDelta > 0) { 5175 // the gap between the current output position and the proper start of 5176 // the next input sample is too big, so fill it with silence 5177 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5178 5179 timedYieldSilence_l(framesUntilNextInput, buffer); 5180 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5181 return NO_ERROR; 5182 } else { 5183 // the next input sample is late 5184 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5185 size_t onTimeSamplePosition = 5186 head.position() + lateFrames * mCblk->frameSize; 5187 5188 if (onTimeSamplePosition > head.buffer()->size()) { 5189 // all the remaining samples in the head are too late, so 5190 // drop it and move on 5191 ALOGV("*** too late: dropped buffer"); 5192 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5193 continue; 5194 } else { 5195 // skip over the late samples 5196 head.setPosition(onTimeSamplePosition); 5197 5198 // yield the available samples 5199 timedYieldSamples_l(buffer); 5200 5201 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5202 return NO_ERROR; 5203 } 5204 } 5205 } 5206} 5207 5208// Yield samples from the timed buffer queue head up to the given output 5209// buffer's capacity. 5210// 5211// Caller must hold mTimedBufferQueueLock 5212void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5213 AudioBufferProvider::Buffer* buffer) { 5214 5215 const TimedBuffer& head = mTimedBufferQueue[0]; 5216 5217 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5218 head.position()); 5219 5220 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5221 mCblk->frameSize); 5222 size_t framesRequested = buffer->frameCount; 5223 buffer->frameCount = min(framesLeftInHead, framesRequested); 5224 5225 mQueueHeadInFlight = true; 5226 mTimedAudioOutputOnTime = true; 5227} 5228 5229// Yield samples of silence up to the given output buffer's capacity 5230// 5231// Caller must hold mTimedBufferQueueLock 5232void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5233 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5234 5235 // lazily allocate a buffer filled with silence 5236 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5237 delete [] mTimedSilenceBuffer; 5238 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5239 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5240 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5241 } 5242 5243 buffer->raw = mTimedSilenceBuffer; 5244 size_t framesRequested = buffer->frameCount; 5245 buffer->frameCount = min(numFrames, framesRequested); 5246 5247 mTimedAudioOutputOnTime = false; 5248} 5249 5250// AudioBufferProvider interface 5251void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5252 AudioBufferProvider::Buffer* buffer) { 5253 5254 Mutex::Autolock _l(mTimedBufferQueueLock); 5255 5256 // If the buffer which was just released is part of the buffer at the head 5257 // of the queue, be sure to update the amt of the buffer which has been 5258 // consumed. If the buffer being returned is not part of the head of the 5259 // queue, its either because the buffer is part of the silence buffer, or 5260 // because the head of the timed queue was trimmed after the mixer called 5261 // getNextBuffer but before the mixer called releaseBuffer. 5262 if (buffer->raw == mTimedSilenceBuffer) { 5263 ALOG_ASSERT(!mQueueHeadInFlight, 5264 "Queue head in flight during release of silence buffer!"); 5265 goto done; 5266 } 5267 5268 ALOG_ASSERT(mQueueHeadInFlight, 5269 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5270 " head in flight."); 5271 5272 if (mTimedBufferQueue.size()) { 5273 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5274 5275 void* start = head.buffer()->pointer(); 5276 void* end = reinterpret_cast<void*>( 5277 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5278 + head.buffer()->size()); 5279 5280 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5281 "released buffer not within the head of the timed buffer" 5282 " queue; qHead = [%p, %p], released buffer = %p", 5283 start, end, buffer->raw); 5284 5285 head.setPosition(head.position() + 5286 (buffer->frameCount * mCblk->frameSize)); 5287 mQueueHeadInFlight = false; 5288 5289 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5290 "Bad bookkeeping during releaseBuffer! Should have at" 5291 " least %u queued frames, but we think we have only %u", 5292 buffer->frameCount, mFramesPendingInQueue); 5293 5294 mFramesPendingInQueue -= buffer->frameCount; 5295 5296 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5297 || mTrimQueueHeadOnRelease) { 5298 trimTimedBufferQueueHead_l("releaseBuffer"); 5299 mTrimQueueHeadOnRelease = false; 5300 } 5301 } else { 5302 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5303 " buffers in the timed buffer queue"); 5304 } 5305 5306done: 5307 buffer->raw = 0; 5308 buffer->frameCount = 0; 5309} 5310 5311size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5312 Mutex::Autolock _l(mTimedBufferQueueLock); 5313 return mFramesPendingInQueue; 5314} 5315 5316AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5317 : mPTS(0), mPosition(0) {} 5318 5319AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5320 const sp<IMemory>& buffer, int64_t pts) 5321 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5322 5323// ---------------------------------------------------------------------------- 5324 5325// RecordTrack constructor must be called with AudioFlinger::mLock held 5326AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5327 RecordThread *thread, 5328 const sp<Client>& client, 5329 uint32_t sampleRate, 5330 audio_format_t format, 5331 audio_channel_mask_t channelMask, 5332 int frameCount, 5333 int sessionId) 5334 : TrackBase(thread, client, sampleRate, format, 5335 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5336 mOverflow(false) 5337{ 5338 if (mCblk != NULL) { 5339 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5340 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5341 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5342 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5343 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5344 } else { 5345 mCblk->frameSize = sizeof(int8_t); 5346 } 5347 } 5348} 5349 5350AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5351{ 5352 ALOGV("%s", __func__); 5353} 5354 5355// AudioBufferProvider interface 5356status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5357{ 5358 audio_track_cblk_t* cblk = this->cblk(); 5359 uint32_t framesAvail; 5360 uint32_t framesReq = buffer->frameCount; 5361 5362 // Check if last stepServer failed, try to step now 5363 if (mStepServerFailed) { 5364 if (!step()) goto getNextBuffer_exit; 5365 ALOGV("stepServer recovered"); 5366 mStepServerFailed = false; 5367 } 5368 5369 framesAvail = cblk->framesAvailable_l(); 5370 5371 if (CC_LIKELY(framesAvail)) { 5372 uint32_t s = cblk->server; 5373 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5374 5375 if (framesReq > framesAvail) { 5376 framesReq = framesAvail; 5377 } 5378 if (framesReq > bufferEnd - s) { 5379 framesReq = bufferEnd - s; 5380 } 5381 5382 buffer->raw = getBuffer(s, framesReq); 5383 buffer->frameCount = framesReq; 5384 return NO_ERROR; 5385 } 5386 5387getNextBuffer_exit: 5388 buffer->raw = NULL; 5389 buffer->frameCount = 0; 5390 return NOT_ENOUGH_DATA; 5391} 5392 5393status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5394 int triggerSession) 5395{ 5396 sp<ThreadBase> thread = mThread.promote(); 5397 if (thread != 0) { 5398 RecordThread *recordThread = (RecordThread *)thread.get(); 5399 return recordThread->start(this, event, triggerSession); 5400 } else { 5401 return BAD_VALUE; 5402 } 5403} 5404 5405void AudioFlinger::RecordThread::RecordTrack::stop() 5406{ 5407 sp<ThreadBase> thread = mThread.promote(); 5408 if (thread != 0) { 5409 RecordThread *recordThread = (RecordThread *)thread.get(); 5410 recordThread->mLock.lock(); 5411 bool doStop = recordThread->stop_l(this); 5412 if (doStop) { 5413 TrackBase::reset(); 5414 // Force overrun condition to avoid false overrun callback until first data is 5415 // read from buffer 5416 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5417 } 5418 recordThread->mLock.unlock(); 5419 if (doStop) { 5420 AudioSystem::stopInput(recordThread->id()); 5421 } 5422 } 5423} 5424 5425/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5426{ 5427 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5428} 5429 5430void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5431{ 5432 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5433 (mClient == 0) ? getpid_cached : mClient->pid(), 5434 mFormat, 5435 mChannelMask, 5436 mSessionId, 5437 mFrameCount, 5438 mState, 5439 mCblk->sampleRate, 5440 mCblk->server, 5441 mCblk->user); 5442} 5443 5444 5445// ---------------------------------------------------------------------------- 5446 5447AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5448 PlaybackThread *playbackThread, 5449 DuplicatingThread *sourceThread, 5450 uint32_t sampleRate, 5451 audio_format_t format, 5452 audio_channel_mask_t channelMask, 5453 int frameCount) 5454 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5455 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5456 mActive(false), mSourceThread(sourceThread) 5457{ 5458 5459 if (mCblk != NULL) { 5460 mCblk->flags |= CBLK_DIRECTION_OUT; 5461 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5462 mOutBuffer.frameCount = 0; 5463 playbackThread->mTracks.add(this); 5464 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5465 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5466 mCblk, mBuffer, mCblk->buffers, 5467 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5468 } else { 5469 ALOGW("Error creating output track on thread %p", playbackThread); 5470 } 5471} 5472 5473AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5474{ 5475 clearBufferQueue(); 5476} 5477 5478status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5479 int triggerSession) 5480{ 5481 status_t status = Track::start(event, triggerSession); 5482 if (status != NO_ERROR) { 5483 return status; 5484 } 5485 5486 mActive = true; 5487 mRetryCount = 127; 5488 return status; 5489} 5490 5491void AudioFlinger::PlaybackThread::OutputTrack::stop() 5492{ 5493 Track::stop(); 5494 clearBufferQueue(); 5495 mOutBuffer.frameCount = 0; 5496 mActive = false; 5497} 5498 5499bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5500{ 5501 Buffer *pInBuffer; 5502 Buffer inBuffer; 5503 uint32_t channelCount = mChannelCount; 5504 bool outputBufferFull = false; 5505 inBuffer.frameCount = frames; 5506 inBuffer.i16 = data; 5507 5508 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5509 5510 if (!mActive && frames != 0) { 5511 start(); 5512 sp<ThreadBase> thread = mThread.promote(); 5513 if (thread != 0) { 5514 MixerThread *mixerThread = (MixerThread *)thread.get(); 5515 if (mCblk->frameCount > frames){ 5516 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5517 uint32_t startFrames = (mCblk->frameCount - frames); 5518 pInBuffer = new Buffer; 5519 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5520 pInBuffer->frameCount = startFrames; 5521 pInBuffer->i16 = pInBuffer->mBuffer; 5522 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5523 mBufferQueue.add(pInBuffer); 5524 } else { 5525 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5526 } 5527 } 5528 } 5529 } 5530 5531 while (waitTimeLeftMs) { 5532 // First write pending buffers, then new data 5533 if (mBufferQueue.size()) { 5534 pInBuffer = mBufferQueue.itemAt(0); 5535 } else { 5536 pInBuffer = &inBuffer; 5537 } 5538 5539 if (pInBuffer->frameCount == 0) { 5540 break; 5541 } 5542 5543 if (mOutBuffer.frameCount == 0) { 5544 mOutBuffer.frameCount = pInBuffer->frameCount; 5545 nsecs_t startTime = systemTime(); 5546 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5547 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5548 outputBufferFull = true; 5549 break; 5550 } 5551 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5552 if (waitTimeLeftMs >= waitTimeMs) { 5553 waitTimeLeftMs -= waitTimeMs; 5554 } else { 5555 waitTimeLeftMs = 0; 5556 } 5557 } 5558 5559 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5560 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5561 mCblk->stepUser(outFrames); 5562 pInBuffer->frameCount -= outFrames; 5563 pInBuffer->i16 += outFrames * channelCount; 5564 mOutBuffer.frameCount -= outFrames; 5565 mOutBuffer.i16 += outFrames * channelCount; 5566 5567 if (pInBuffer->frameCount == 0) { 5568 if (mBufferQueue.size()) { 5569 mBufferQueue.removeAt(0); 5570 delete [] pInBuffer->mBuffer; 5571 delete pInBuffer; 5572 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5573 } else { 5574 break; 5575 } 5576 } 5577 } 5578 5579 // If we could not write all frames, allocate a buffer and queue it for next time. 5580 if (inBuffer.frameCount) { 5581 sp<ThreadBase> thread = mThread.promote(); 5582 if (thread != 0 && !thread->standby()) { 5583 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5584 pInBuffer = new Buffer; 5585 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5586 pInBuffer->frameCount = inBuffer.frameCount; 5587 pInBuffer->i16 = pInBuffer->mBuffer; 5588 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5589 mBufferQueue.add(pInBuffer); 5590 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5591 } else { 5592 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5593 } 5594 } 5595 } 5596 5597 // Calling write() with a 0 length buffer, means that no more data will be written: 5598 // If no more buffers are pending, fill output track buffer to make sure it is started 5599 // by output mixer. 5600 if (frames == 0 && mBufferQueue.size() == 0) { 5601 if (mCblk->user < mCblk->frameCount) { 5602 frames = mCblk->frameCount - mCblk->user; 5603 pInBuffer = new Buffer; 5604 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5605 pInBuffer->frameCount = frames; 5606 pInBuffer->i16 = pInBuffer->mBuffer; 5607 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5608 mBufferQueue.add(pInBuffer); 5609 } else if (mActive) { 5610 stop(); 5611 } 5612 } 5613 5614 return outputBufferFull; 5615} 5616 5617status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5618{ 5619 int active; 5620 status_t result; 5621 audio_track_cblk_t* cblk = mCblk; 5622 uint32_t framesReq = buffer->frameCount; 5623 5624// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5625 buffer->frameCount = 0; 5626 5627 uint32_t framesAvail = cblk->framesAvailable(); 5628 5629 5630 if (framesAvail == 0) { 5631 Mutex::Autolock _l(cblk->lock); 5632 goto start_loop_here; 5633 while (framesAvail == 0) { 5634 active = mActive; 5635 if (CC_UNLIKELY(!active)) { 5636 ALOGV("Not active and NO_MORE_BUFFERS"); 5637 return NO_MORE_BUFFERS; 5638 } 5639 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5640 if (result != NO_ERROR) { 5641 return NO_MORE_BUFFERS; 5642 } 5643 // read the server count again 5644 start_loop_here: 5645 framesAvail = cblk->framesAvailable_l(); 5646 } 5647 } 5648 5649// if (framesAvail < framesReq) { 5650// return NO_MORE_BUFFERS; 5651// } 5652 5653 if (framesReq > framesAvail) { 5654 framesReq = framesAvail; 5655 } 5656 5657 uint32_t u = cblk->user; 5658 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5659 5660 if (framesReq > bufferEnd - u) { 5661 framesReq = bufferEnd - u; 5662 } 5663 5664 buffer->frameCount = framesReq; 5665 buffer->raw = (void *)cblk->buffer(u); 5666 return NO_ERROR; 5667} 5668 5669 5670void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5671{ 5672 size_t size = mBufferQueue.size(); 5673 5674 for (size_t i = 0; i < size; i++) { 5675 Buffer *pBuffer = mBufferQueue.itemAt(i); 5676 delete [] pBuffer->mBuffer; 5677 delete pBuffer; 5678 } 5679 mBufferQueue.clear(); 5680} 5681 5682// ---------------------------------------------------------------------------- 5683 5684AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5685 : RefBase(), 5686 mAudioFlinger(audioFlinger), 5687 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5688 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5689 mPid(pid), 5690 mTimedTrackCount(0) 5691{ 5692 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5693} 5694 5695// Client destructor must be called with AudioFlinger::mLock held 5696AudioFlinger::Client::~Client() 5697{ 5698 mAudioFlinger->removeClient_l(mPid); 5699} 5700 5701sp<MemoryDealer> AudioFlinger::Client::heap() const 5702{ 5703 return mMemoryDealer; 5704} 5705 5706// Reserve one of the limited slots for a timed audio track associated 5707// with this client 5708bool AudioFlinger::Client::reserveTimedTrack() 5709{ 5710 const int kMaxTimedTracksPerClient = 4; 5711 5712 Mutex::Autolock _l(mTimedTrackLock); 5713 5714 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5715 ALOGW("can not create timed track - pid %d has exceeded the limit", 5716 mPid); 5717 return false; 5718 } 5719 5720 mTimedTrackCount++; 5721 return true; 5722} 5723 5724// Release a slot for a timed audio track 5725void AudioFlinger::Client::releaseTimedTrack() 5726{ 5727 Mutex::Autolock _l(mTimedTrackLock); 5728 mTimedTrackCount--; 5729} 5730 5731// ---------------------------------------------------------------------------- 5732 5733AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5734 const sp<IAudioFlingerClient>& client, 5735 pid_t pid) 5736 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5737{ 5738} 5739 5740AudioFlinger::NotificationClient::~NotificationClient() 5741{ 5742} 5743 5744void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5745{ 5746 sp<NotificationClient> keep(this); 5747 mAudioFlinger->removeNotificationClient(mPid); 5748} 5749 5750// ---------------------------------------------------------------------------- 5751 5752AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5753 : BnAudioTrack(), 5754 mTrack(track) 5755{ 5756} 5757 5758AudioFlinger::TrackHandle::~TrackHandle() { 5759 // just stop the track on deletion, associated resources 5760 // will be freed from the main thread once all pending buffers have 5761 // been played. Unless it's not in the active track list, in which 5762 // case we free everything now... 5763 mTrack->destroy(); 5764} 5765 5766sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5767 return mTrack->getCblk(); 5768} 5769 5770status_t AudioFlinger::TrackHandle::start() { 5771 return mTrack->start(); 5772} 5773 5774void AudioFlinger::TrackHandle::stop() { 5775 mTrack->stop(); 5776} 5777 5778void AudioFlinger::TrackHandle::flush() { 5779 mTrack->flush(); 5780} 5781 5782void AudioFlinger::TrackHandle::mute(bool e) { 5783 mTrack->mute(e); 5784} 5785 5786void AudioFlinger::TrackHandle::pause() { 5787 mTrack->pause(); 5788} 5789 5790status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5791{ 5792 return mTrack->attachAuxEffect(EffectId); 5793} 5794 5795status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5796 sp<IMemory>* buffer) { 5797 if (!mTrack->isTimedTrack()) 5798 return INVALID_OPERATION; 5799 5800 PlaybackThread::TimedTrack* tt = 5801 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5802 return tt->allocateTimedBuffer(size, buffer); 5803} 5804 5805status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5806 int64_t pts) { 5807 if (!mTrack->isTimedTrack()) 5808 return INVALID_OPERATION; 5809 5810 PlaybackThread::TimedTrack* tt = 5811 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5812 return tt->queueTimedBuffer(buffer, pts); 5813} 5814 5815status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5816 const LinearTransform& xform, int target) { 5817 5818 if (!mTrack->isTimedTrack()) 5819 return INVALID_OPERATION; 5820 5821 PlaybackThread::TimedTrack* tt = 5822 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5823 return tt->setMediaTimeTransform( 5824 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5825} 5826 5827status_t AudioFlinger::TrackHandle::onTransact( 5828 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5829{ 5830 return BnAudioTrack::onTransact(code, data, reply, flags); 5831} 5832 5833// ---------------------------------------------------------------------------- 5834 5835sp<IAudioRecord> AudioFlinger::openRecord( 5836 pid_t pid, 5837 audio_io_handle_t input, 5838 uint32_t sampleRate, 5839 audio_format_t format, 5840 audio_channel_mask_t channelMask, 5841 int frameCount, 5842 IAudioFlinger::track_flags_t flags, 5843 pid_t tid, 5844 int *sessionId, 5845 status_t *status) 5846{ 5847 sp<RecordThread::RecordTrack> recordTrack; 5848 sp<RecordHandle> recordHandle; 5849 sp<Client> client; 5850 status_t lStatus; 5851 RecordThread *thread; 5852 size_t inFrameCount; 5853 int lSessionId; 5854 5855 // check calling permissions 5856 if (!recordingAllowed()) { 5857 lStatus = PERMISSION_DENIED; 5858 goto Exit; 5859 } 5860 5861 // add client to list 5862 { // scope for mLock 5863 Mutex::Autolock _l(mLock); 5864 thread = checkRecordThread_l(input); 5865 if (thread == NULL) { 5866 lStatus = BAD_VALUE; 5867 goto Exit; 5868 } 5869 5870 client = registerPid_l(pid); 5871 5872 // If no audio session id is provided, create one here 5873 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5874 lSessionId = *sessionId; 5875 } else { 5876 lSessionId = nextUniqueId(); 5877 if (sessionId != NULL) { 5878 *sessionId = lSessionId; 5879 } 5880 } 5881 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5882 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5883 frameCount, lSessionId, flags, tid, &lStatus); 5884 } 5885 if (lStatus != NO_ERROR) { 5886 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5887 // destructor is called by the TrackBase destructor with mLock held 5888 client.clear(); 5889 recordTrack.clear(); 5890 goto Exit; 5891 } 5892 5893 // return to handle to client 5894 recordHandle = new RecordHandle(recordTrack); 5895 lStatus = NO_ERROR; 5896 5897Exit: 5898 if (status) { 5899 *status = lStatus; 5900 } 5901 return recordHandle; 5902} 5903 5904// ---------------------------------------------------------------------------- 5905 5906AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5907 : BnAudioRecord(), 5908 mRecordTrack(recordTrack) 5909{ 5910} 5911 5912AudioFlinger::RecordHandle::~RecordHandle() { 5913 stop_nonvirtual(); 5914 mRecordTrack->destroy(); 5915} 5916 5917sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5918 return mRecordTrack->getCblk(); 5919} 5920 5921status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5922 ALOGV("RecordHandle::start()"); 5923 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5924} 5925 5926void AudioFlinger::RecordHandle::stop() { 5927 stop_nonvirtual(); 5928} 5929 5930void AudioFlinger::RecordHandle::stop_nonvirtual() { 5931 ALOGV("RecordHandle::stop()"); 5932 mRecordTrack->stop(); 5933} 5934 5935status_t AudioFlinger::RecordHandle::onTransact( 5936 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5937{ 5938 return BnAudioRecord::onTransact(code, data, reply, flags); 5939} 5940 5941// ---------------------------------------------------------------------------- 5942 5943AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5944 AudioStreamIn *input, 5945 uint32_t sampleRate, 5946 audio_channel_mask_t channelMask, 5947 audio_io_handle_t id, 5948 audio_devices_t device) : 5949 ThreadBase(audioFlinger, id, device, RECORD), 5950 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5951 // mRsmpInIndex and mInputBytes set by readInputParameters() 5952 mReqChannelCount(popcount(channelMask)), 5953 mReqSampleRate(sampleRate) 5954 // mBytesRead is only meaningful while active, and so is cleared in start() 5955 // (but might be better to also clear here for dump?) 5956{ 5957 snprintf(mName, kNameLength, "AudioIn_%X", id); 5958 5959 readInputParameters(); 5960} 5961 5962 5963AudioFlinger::RecordThread::~RecordThread() 5964{ 5965 delete[] mRsmpInBuffer; 5966 delete mResampler; 5967 delete[] mRsmpOutBuffer; 5968} 5969 5970void AudioFlinger::RecordThread::onFirstRef() 5971{ 5972 run(mName, PRIORITY_URGENT_AUDIO); 5973} 5974 5975status_t AudioFlinger::RecordThread::readyToRun() 5976{ 5977 status_t status = initCheck(); 5978 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5979 return status; 5980} 5981 5982bool AudioFlinger::RecordThread::threadLoop() 5983{ 5984 AudioBufferProvider::Buffer buffer; 5985 sp<RecordTrack> activeTrack; 5986 Vector< sp<EffectChain> > effectChains; 5987 5988 nsecs_t lastWarning = 0; 5989 5990 inputStandBy(); 5991 acquireWakeLock(); 5992 5993 // start recording 5994 while (!exitPending()) { 5995 5996 processConfigEvents(); 5997 5998 { // scope for mLock 5999 Mutex::Autolock _l(mLock); 6000 checkForNewParameters_l(); 6001 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6002 standby(); 6003 6004 if (exitPending()) break; 6005 6006 releaseWakeLock_l(); 6007 ALOGV("RecordThread: loop stopping"); 6008 // go to sleep 6009 mWaitWorkCV.wait(mLock); 6010 ALOGV("RecordThread: loop starting"); 6011 acquireWakeLock_l(); 6012 continue; 6013 } 6014 if (mActiveTrack != 0) { 6015 if (mActiveTrack->mState == TrackBase::PAUSING) { 6016 standby(); 6017 mActiveTrack.clear(); 6018 mStartStopCond.broadcast(); 6019 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6020 if (mReqChannelCount != mActiveTrack->channelCount()) { 6021 mActiveTrack.clear(); 6022 mStartStopCond.broadcast(); 6023 } else if (mBytesRead != 0) { 6024 // record start succeeds only if first read from audio input 6025 // succeeds 6026 if (mBytesRead > 0) { 6027 mActiveTrack->mState = TrackBase::ACTIVE; 6028 } else { 6029 mActiveTrack.clear(); 6030 } 6031 mStartStopCond.broadcast(); 6032 } 6033 mStandby = false; 6034 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6035 removeTrack_l(mActiveTrack); 6036 mActiveTrack.clear(); 6037 } 6038 } 6039 lockEffectChains_l(effectChains); 6040 } 6041 6042 if (mActiveTrack != 0) { 6043 if (mActiveTrack->mState != TrackBase::ACTIVE && 6044 mActiveTrack->mState != TrackBase::RESUMING) { 6045 unlockEffectChains(effectChains); 6046 usleep(kRecordThreadSleepUs); 6047 continue; 6048 } 6049 for (size_t i = 0; i < effectChains.size(); i ++) { 6050 effectChains[i]->process_l(); 6051 } 6052 6053 buffer.frameCount = mFrameCount; 6054 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6055 size_t framesOut = buffer.frameCount; 6056 if (mResampler == NULL) { 6057 // no resampling 6058 while (framesOut) { 6059 size_t framesIn = mFrameCount - mRsmpInIndex; 6060 if (framesIn) { 6061 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6062 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6063 if (framesIn > framesOut) 6064 framesIn = framesOut; 6065 mRsmpInIndex += framesIn; 6066 framesOut -= framesIn; 6067 if ((int)mChannelCount == mReqChannelCount || 6068 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6069 memcpy(dst, src, framesIn * mFrameSize); 6070 } else { 6071 if (mChannelCount == 1) { 6072 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6073 (int16_t *)src, framesIn); 6074 } else { 6075 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6076 (int16_t *)src, framesIn); 6077 } 6078 } 6079 } 6080 if (framesOut && mFrameCount == mRsmpInIndex) { 6081 if (framesOut == mFrameCount && 6082 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6083 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6084 framesOut = 0; 6085 } else { 6086 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6087 mRsmpInIndex = 0; 6088 } 6089 if (mBytesRead < 0) { 6090 ALOGE("Error reading audio input"); 6091 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6092 // Force input into standby so that it tries to 6093 // recover at next read attempt 6094 inputStandBy(); 6095 usleep(kRecordThreadSleepUs); 6096 } 6097 mRsmpInIndex = mFrameCount; 6098 framesOut = 0; 6099 buffer.frameCount = 0; 6100 } 6101 } 6102 } 6103 } else { 6104 // resampling 6105 6106 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6107 // alter output frame count as if we were expecting stereo samples 6108 if (mChannelCount == 1 && mReqChannelCount == 1) { 6109 framesOut >>= 1; 6110 } 6111 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6112 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6113 // are 32 bit aligned which should be always true. 6114 if (mChannelCount == 2 && mReqChannelCount == 1) { 6115 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6116 // the resampler always outputs stereo samples: do post stereo to mono conversion 6117 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6118 framesOut); 6119 } else { 6120 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6121 } 6122 6123 } 6124 if (mFramestoDrop == 0) { 6125 mActiveTrack->releaseBuffer(&buffer); 6126 } else { 6127 if (mFramestoDrop > 0) { 6128 mFramestoDrop -= buffer.frameCount; 6129 if (mFramestoDrop <= 0) { 6130 clearSyncStartEvent(); 6131 } 6132 } else { 6133 mFramestoDrop += buffer.frameCount; 6134 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6135 mSyncStartEvent->isCancelled()) { 6136 ALOGW("Synced record %s, session %d, trigger session %d", 6137 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6138 mActiveTrack->sessionId(), 6139 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6140 clearSyncStartEvent(); 6141 } 6142 } 6143 } 6144 mActiveTrack->clearOverflow(); 6145 } 6146 // client isn't retrieving buffers fast enough 6147 else { 6148 if (!mActiveTrack->setOverflow()) { 6149 nsecs_t now = systemTime(); 6150 if ((now - lastWarning) > kWarningThrottleNs) { 6151 ALOGW("RecordThread: buffer overflow"); 6152 lastWarning = now; 6153 } 6154 } 6155 // Release the processor for a while before asking for a new buffer. 6156 // This will give the application more chance to read from the buffer and 6157 // clear the overflow. 6158 usleep(kRecordThreadSleepUs); 6159 } 6160 } 6161 // enable changes in effect chain 6162 unlockEffectChains(effectChains); 6163 effectChains.clear(); 6164 } 6165 6166 standby(); 6167 6168 { 6169 Mutex::Autolock _l(mLock); 6170 mActiveTrack.clear(); 6171 mStartStopCond.broadcast(); 6172 } 6173 6174 releaseWakeLock(); 6175 6176 ALOGV("RecordThread %p exiting", this); 6177 return false; 6178} 6179 6180void AudioFlinger::RecordThread::standby() 6181{ 6182 if (!mStandby) { 6183 inputStandBy(); 6184 mStandby = true; 6185 } 6186} 6187 6188void AudioFlinger::RecordThread::inputStandBy() 6189{ 6190 mInput->stream->common.standby(&mInput->stream->common); 6191} 6192 6193sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6194 const sp<AudioFlinger::Client>& client, 6195 uint32_t sampleRate, 6196 audio_format_t format, 6197 audio_channel_mask_t channelMask, 6198 int frameCount, 6199 int sessionId, 6200 IAudioFlinger::track_flags_t flags, 6201 pid_t tid, 6202 status_t *status) 6203{ 6204 sp<RecordTrack> track; 6205 status_t lStatus; 6206 6207 lStatus = initCheck(); 6208 if (lStatus != NO_ERROR) { 6209 ALOGE("Audio driver not initialized."); 6210 goto Exit; 6211 } 6212 6213 // FIXME use flags and tid similar to createTrack_l() 6214 6215 { // scope for mLock 6216 Mutex::Autolock _l(mLock); 6217 6218 track = new RecordTrack(this, client, sampleRate, 6219 format, channelMask, frameCount, sessionId); 6220 6221 if (track->getCblk() == 0) { 6222 lStatus = NO_MEMORY; 6223 goto Exit; 6224 } 6225 mTracks.add(track); 6226 6227 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6228 bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) && 6229 mAudioFlinger->btNrecIsOff(); 6230 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6231 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6232 } 6233 lStatus = NO_ERROR; 6234 6235Exit: 6236 if (status) { 6237 *status = lStatus; 6238 } 6239 return track; 6240} 6241 6242status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6243 AudioSystem::sync_event_t event, 6244 int triggerSession) 6245{ 6246 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6247 sp<ThreadBase> strongMe = this; 6248 status_t status = NO_ERROR; 6249 6250 if (event == AudioSystem::SYNC_EVENT_NONE) { 6251 clearSyncStartEvent(); 6252 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6253 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6254 triggerSession, 6255 recordTrack->sessionId(), 6256 syncStartEventCallback, 6257 this); 6258 // Sync event can be cancelled by the trigger session if the track is not in a 6259 // compatible state in which case we start record immediately 6260 if (mSyncStartEvent->isCancelled()) { 6261 clearSyncStartEvent(); 6262 } else { 6263 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6264 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6265 } 6266 } 6267 6268 { 6269 AutoMutex lock(mLock); 6270 if (mActiveTrack != 0) { 6271 if (recordTrack != mActiveTrack.get()) { 6272 status = -EBUSY; 6273 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6274 mActiveTrack->mState = TrackBase::ACTIVE; 6275 } 6276 return status; 6277 } 6278 6279 recordTrack->mState = TrackBase::IDLE; 6280 mActiveTrack = recordTrack; 6281 mLock.unlock(); 6282 status_t status = AudioSystem::startInput(mId); 6283 mLock.lock(); 6284 if (status != NO_ERROR) { 6285 mActiveTrack.clear(); 6286 clearSyncStartEvent(); 6287 return status; 6288 } 6289 mRsmpInIndex = mFrameCount; 6290 mBytesRead = 0; 6291 if (mResampler != NULL) { 6292 mResampler->reset(); 6293 } 6294 mActiveTrack->mState = TrackBase::RESUMING; 6295 // signal thread to start 6296 ALOGV("Signal record thread"); 6297 mWaitWorkCV.signal(); 6298 // do not wait for mStartStopCond if exiting 6299 if (exitPending()) { 6300 mActiveTrack.clear(); 6301 status = INVALID_OPERATION; 6302 goto startError; 6303 } 6304 mStartStopCond.wait(mLock); 6305 if (mActiveTrack == 0) { 6306 ALOGV("Record failed to start"); 6307 status = BAD_VALUE; 6308 goto startError; 6309 } 6310 ALOGV("Record started OK"); 6311 return status; 6312 } 6313startError: 6314 AudioSystem::stopInput(mId); 6315 clearSyncStartEvent(); 6316 return status; 6317} 6318 6319void AudioFlinger::RecordThread::clearSyncStartEvent() 6320{ 6321 if (mSyncStartEvent != 0) { 6322 mSyncStartEvent->cancel(); 6323 } 6324 mSyncStartEvent.clear(); 6325 mFramestoDrop = 0; 6326} 6327 6328void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6329{ 6330 sp<SyncEvent> strongEvent = event.promote(); 6331 6332 if (strongEvent != 0) { 6333 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6334 me->handleSyncStartEvent(strongEvent); 6335 } 6336} 6337 6338void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6339{ 6340 if (event == mSyncStartEvent) { 6341 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6342 // from audio HAL 6343 mFramestoDrop = mFrameCount * 2; 6344 } 6345} 6346 6347bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6348 ALOGV("RecordThread::stop"); 6349 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6350 return false; 6351 } 6352 recordTrack->mState = TrackBase::PAUSING; 6353 // do not wait for mStartStopCond if exiting 6354 if (exitPending()) { 6355 return true; 6356 } 6357 mStartStopCond.wait(mLock); 6358 // if we have been restarted, recordTrack == mActiveTrack.get() here 6359 if (exitPending() || recordTrack != mActiveTrack.get()) { 6360 ALOGV("Record stopped OK"); 6361 return true; 6362 } 6363 return false; 6364} 6365 6366bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6367{ 6368 return false; 6369} 6370 6371status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6372{ 6373 if (!isValidSyncEvent(event)) { 6374 return BAD_VALUE; 6375 } 6376 6377 int eventSession = event->triggerSession(); 6378 status_t ret = NAME_NOT_FOUND; 6379 6380 Mutex::Autolock _l(mLock); 6381 6382 for (size_t i = 0; i < mTracks.size(); i++) { 6383 sp<RecordTrack> track = mTracks[i]; 6384 if (eventSession == track->sessionId()) { 6385 track->setSyncEvent(event); 6386 ret = NO_ERROR; 6387 } 6388 } 6389 return ret; 6390} 6391 6392void AudioFlinger::RecordThread::RecordTrack::destroy() 6393{ 6394 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6395 sp<RecordTrack> keep(this); 6396 { 6397 sp<ThreadBase> thread = mThread.promote(); 6398 if (thread != 0) { 6399 if (mState == ACTIVE || mState == RESUMING) { 6400 AudioSystem::stopInput(thread->id()); 6401 } 6402 AudioSystem::releaseInput(thread->id()); 6403 Mutex::Autolock _l(thread->mLock); 6404 RecordThread *recordThread = (RecordThread *) thread.get(); 6405 recordThread->destroyTrack_l(this); 6406 } 6407 } 6408} 6409 6410// destroyTrack_l() must be called with ThreadBase::mLock held 6411void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6412{ 6413 track->mState = TrackBase::TERMINATED; 6414 // active tracks are removed by threadLoop() 6415 if (mActiveTrack != track) { 6416 removeTrack_l(track); 6417 } 6418} 6419 6420void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6421{ 6422 mTracks.remove(track); 6423 // need anything related to effects here? 6424} 6425 6426void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6427{ 6428 dumpInternals(fd, args); 6429 dumpTracks(fd, args); 6430 dumpEffectChains(fd, args); 6431} 6432 6433void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6434{ 6435 const size_t SIZE = 256; 6436 char buffer[SIZE]; 6437 String8 result; 6438 6439 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6440 result.append(buffer); 6441 6442 if (mActiveTrack != 0) { 6443 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6444 result.append(buffer); 6445 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6446 result.append(buffer); 6447 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6448 result.append(buffer); 6449 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6450 result.append(buffer); 6451 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6452 result.append(buffer); 6453 } else { 6454 result.append("No active record client\n"); 6455 } 6456 6457 write(fd, result.string(), result.size()); 6458 6459 dumpBase(fd, args); 6460} 6461 6462void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6463{ 6464 const size_t SIZE = 256; 6465 char buffer[SIZE]; 6466 String8 result; 6467 6468 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6469 result.append(buffer); 6470 RecordTrack::appendDumpHeader(result); 6471 for (size_t i = 0; i < mTracks.size(); ++i) { 6472 sp<RecordTrack> track = mTracks[i]; 6473 if (track != 0) { 6474 track->dump(buffer, SIZE); 6475 result.append(buffer); 6476 } 6477 } 6478 6479 if (mActiveTrack != 0) { 6480 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6481 result.append(buffer); 6482 RecordTrack::appendDumpHeader(result); 6483 mActiveTrack->dump(buffer, SIZE); 6484 result.append(buffer); 6485 6486 } 6487 write(fd, result.string(), result.size()); 6488} 6489 6490// AudioBufferProvider interface 6491status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6492{ 6493 size_t framesReq = buffer->frameCount; 6494 size_t framesReady = mFrameCount - mRsmpInIndex; 6495 int channelCount; 6496 6497 if (framesReady == 0) { 6498 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6499 if (mBytesRead < 0) { 6500 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6501 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6502 // Force input into standby so that it tries to 6503 // recover at next read attempt 6504 inputStandBy(); 6505 usleep(kRecordThreadSleepUs); 6506 } 6507 buffer->raw = NULL; 6508 buffer->frameCount = 0; 6509 return NOT_ENOUGH_DATA; 6510 } 6511 mRsmpInIndex = 0; 6512 framesReady = mFrameCount; 6513 } 6514 6515 if (framesReq > framesReady) { 6516 framesReq = framesReady; 6517 } 6518 6519 if (mChannelCount == 1 && mReqChannelCount == 2) { 6520 channelCount = 1; 6521 } else { 6522 channelCount = 2; 6523 } 6524 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6525 buffer->frameCount = framesReq; 6526 return NO_ERROR; 6527} 6528 6529// AudioBufferProvider interface 6530void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6531{ 6532 mRsmpInIndex += buffer->frameCount; 6533 buffer->frameCount = 0; 6534} 6535 6536bool AudioFlinger::RecordThread::checkForNewParameters_l() 6537{ 6538 bool reconfig = false; 6539 6540 while (!mNewParameters.isEmpty()) { 6541 status_t status = NO_ERROR; 6542 String8 keyValuePair = mNewParameters[0]; 6543 AudioParameter param = AudioParameter(keyValuePair); 6544 int value; 6545 audio_format_t reqFormat = mFormat; 6546 int reqSamplingRate = mReqSampleRate; 6547 int reqChannelCount = mReqChannelCount; 6548 6549 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6550 reqSamplingRate = value; 6551 reconfig = true; 6552 } 6553 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6554 reqFormat = (audio_format_t) value; 6555 reconfig = true; 6556 } 6557 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6558 reqChannelCount = popcount(value); 6559 reconfig = true; 6560 } 6561 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6562 // do not accept frame count changes if tracks are open as the track buffer 6563 // size depends on frame count and correct behavior would not be guaranteed 6564 // if frame count is changed after track creation 6565 if (mActiveTrack != 0) { 6566 status = INVALID_OPERATION; 6567 } else { 6568 reconfig = true; 6569 } 6570 } 6571 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6572 // forward device change to effects that have requested to be 6573 // aware of attached audio device. 6574 for (size_t i = 0; i < mEffectChains.size(); i++) { 6575 mEffectChains[i]->setDevice_l(value); 6576 } 6577 // store input device and output device but do not forward output device to audio HAL. 6578 // Note that status is ignored by the caller for output device 6579 // (see AudioFlinger::setParameters() 6580 audio_devices_t newDevice = mDevice; 6581 if (value & AUDIO_DEVICE_OUT_ALL) { 6582 newDevice &= ~(value & AUDIO_DEVICE_OUT_ALL); 6583 status = BAD_VALUE; 6584 } else { 6585 newDevice &= ~(value & AUDIO_DEVICE_IN_ALL); 6586 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6587 if (mTracks.size() > 0) { 6588 bool suspend = audio_is_bluetooth_sco_device( 6589 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6590 for (size_t i = 0; i < mTracks.size(); i++) { 6591 sp<RecordTrack> track = mTracks[i]; 6592 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6593 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6594 } 6595 } 6596 } 6597 newDevice |= value; 6598 mDevice = newDevice; // since mDevice is read by other threads, only write to it once 6599 } 6600 if (status == NO_ERROR) { 6601 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6602 if (status == INVALID_OPERATION) { 6603 inputStandBy(); 6604 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6605 keyValuePair.string()); 6606 } 6607 if (reconfig) { 6608 if (status == BAD_VALUE && 6609 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6610 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6611 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6612 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6613 (reqChannelCount <= FCC_2)) { 6614 status = NO_ERROR; 6615 } 6616 if (status == NO_ERROR) { 6617 readInputParameters(); 6618 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6619 } 6620 } 6621 } 6622 6623 mNewParameters.removeAt(0); 6624 6625 mParamStatus = status; 6626 mParamCond.signal(); 6627 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6628 // already timed out waiting for the status and will never signal the condition. 6629 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6630 } 6631 return reconfig; 6632} 6633 6634String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6635{ 6636 char *s; 6637 String8 out_s8 = String8(); 6638 6639 Mutex::Autolock _l(mLock); 6640 if (initCheck() != NO_ERROR) { 6641 return out_s8; 6642 } 6643 6644 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6645 out_s8 = String8(s); 6646 free(s); 6647 return out_s8; 6648} 6649 6650void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6651 AudioSystem::OutputDescriptor desc; 6652 void *param2 = NULL; 6653 6654 switch (event) { 6655 case AudioSystem::INPUT_OPENED: 6656 case AudioSystem::INPUT_CONFIG_CHANGED: 6657 desc.channels = mChannelMask; 6658 desc.samplingRate = mSampleRate; 6659 desc.format = mFormat; 6660 desc.frameCount = mFrameCount; 6661 desc.latency = 0; 6662 param2 = &desc; 6663 break; 6664 6665 case AudioSystem::INPUT_CLOSED: 6666 default: 6667 break; 6668 } 6669 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6670} 6671 6672void AudioFlinger::RecordThread::readInputParameters() 6673{ 6674 delete mRsmpInBuffer; 6675 // mRsmpInBuffer is always assigned a new[] below 6676 delete mRsmpOutBuffer; 6677 mRsmpOutBuffer = NULL; 6678 delete mResampler; 6679 mResampler = NULL; 6680 6681 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6682 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6683 mChannelCount = (uint16_t)popcount(mChannelMask); 6684 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6685 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6686 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6687 mFrameCount = mInputBytes / mFrameSize; 6688 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6689 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6690 6691 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6692 { 6693 int channelCount; 6694 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6695 // stereo to mono post process as the resampler always outputs stereo. 6696 if (mChannelCount == 1 && mReqChannelCount == 2) { 6697 channelCount = 1; 6698 } else { 6699 channelCount = 2; 6700 } 6701 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6702 mResampler->setSampleRate(mSampleRate); 6703 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6704 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6705 6706 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6707 if (mChannelCount == 1 && mReqChannelCount == 1) { 6708 mFrameCount >>= 1; 6709 } 6710 6711 } 6712 mRsmpInIndex = mFrameCount; 6713} 6714 6715unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6716{ 6717 Mutex::Autolock _l(mLock); 6718 if (initCheck() != NO_ERROR) { 6719 return 0; 6720 } 6721 6722 return mInput->stream->get_input_frames_lost(mInput->stream); 6723} 6724 6725uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6726{ 6727 Mutex::Autolock _l(mLock); 6728 uint32_t result = 0; 6729 if (getEffectChain_l(sessionId) != 0) { 6730 result = EFFECT_SESSION; 6731 } 6732 6733 for (size_t i = 0; i < mTracks.size(); ++i) { 6734 if (sessionId == mTracks[i]->sessionId()) { 6735 result |= TRACK_SESSION; 6736 break; 6737 } 6738 } 6739 6740 return result; 6741} 6742 6743KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() 6744{ 6745 KeyedVector<int, bool> ids; 6746 Mutex::Autolock _l(mLock); 6747 for (size_t j = 0; j < mTracks.size(); ++j) { 6748 sp<RecordThread::RecordTrack> track = mTracks[j]; 6749 int sessionId = track->sessionId(); 6750 if (ids.indexOfKey(sessionId) < 0) { 6751 ids.add(sessionId, true); 6752 } 6753 } 6754 return ids; 6755} 6756 6757AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6758{ 6759 Mutex::Autolock _l(mLock); 6760 AudioStreamIn *input = mInput; 6761 mInput = NULL; 6762 return input; 6763} 6764 6765// this method must always be called either with ThreadBase mLock held or inside the thread loop 6766audio_stream_t* AudioFlinger::RecordThread::stream() const 6767{ 6768 if (mInput == NULL) { 6769 return NULL; 6770 } 6771 return &mInput->stream->common; 6772} 6773 6774 6775// ---------------------------------------------------------------------------- 6776 6777audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6778{ 6779 if (!settingsAllowed()) { 6780 return 0; 6781 } 6782 Mutex::Autolock _l(mLock); 6783 return loadHwModule_l(name); 6784} 6785 6786// loadHwModule_l() must be called with AudioFlinger::mLock held 6787audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6788{ 6789 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6790 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6791 ALOGW("loadHwModule() module %s already loaded", name); 6792 return mAudioHwDevs.keyAt(i); 6793 } 6794 } 6795 6796 audio_hw_device_t *dev; 6797 6798 int rc = load_audio_interface(name, &dev); 6799 if (rc) { 6800 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6801 return 0; 6802 } 6803 6804 mHardwareStatus = AUDIO_HW_INIT; 6805 rc = dev->init_check(dev); 6806 mHardwareStatus = AUDIO_HW_IDLE; 6807 if (rc) { 6808 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6809 return 0; 6810 } 6811 6812 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6813 (NULL != dev->set_master_volume)) { 6814 AutoMutex lock(mHardwareLock); 6815 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6816 dev->set_master_volume(dev, mMasterVolume); 6817 mHardwareStatus = AUDIO_HW_IDLE; 6818 } 6819 6820 audio_module_handle_t handle = nextUniqueId(); 6821 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6822 6823 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6824 name, dev->common.module->name, dev->common.module->id, handle); 6825 6826 return handle; 6827 6828} 6829 6830audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6831 audio_devices_t *pDevices, 6832 uint32_t *pSamplingRate, 6833 audio_format_t *pFormat, 6834 audio_channel_mask_t *pChannelMask, 6835 uint32_t *pLatencyMs, 6836 audio_output_flags_t flags) 6837{ 6838 status_t status; 6839 PlaybackThread *thread = NULL; 6840 struct audio_config config = { 6841 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6842 channel_mask: pChannelMask ? *pChannelMask : 0, 6843 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6844 }; 6845 audio_stream_out_t *outStream = NULL; 6846 audio_hw_device_t *outHwDev; 6847 6848 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6849 module, 6850 (pDevices != NULL) ? *pDevices : 0, 6851 config.sample_rate, 6852 config.format, 6853 config.channel_mask, 6854 flags); 6855 6856 if (pDevices == NULL || *pDevices == 0) { 6857 return 0; 6858 } 6859 6860 Mutex::Autolock _l(mLock); 6861 6862 outHwDev = findSuitableHwDev_l(module, *pDevices); 6863 if (outHwDev == NULL) 6864 return 0; 6865 6866 audio_io_handle_t id = nextUniqueId(); 6867 6868 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6869 6870 status = outHwDev->open_output_stream(outHwDev, 6871 id, 6872 *pDevices, 6873 (audio_output_flags_t)flags, 6874 &config, 6875 &outStream); 6876 6877 mHardwareStatus = AUDIO_HW_IDLE; 6878 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6879 outStream, 6880 config.sample_rate, 6881 config.format, 6882 config.channel_mask, 6883 status); 6884 6885 if (status == NO_ERROR && outStream != NULL) { 6886 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6887 6888 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6889 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6890 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6891 thread = new DirectOutputThread(this, output, id, *pDevices); 6892 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6893 } else { 6894 thread = new MixerThread(this, output, id, *pDevices); 6895 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6896 } 6897 mPlaybackThreads.add(id, thread); 6898 6899 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6900 if (pFormat != NULL) *pFormat = config.format; 6901 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6902 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6903 6904 // notify client processes of the new output creation 6905 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6906 6907 // the first primary output opened designates the primary hw device 6908 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6909 ALOGI("Using module %d has the primary audio interface", module); 6910 mPrimaryHardwareDev = outHwDev; 6911 6912 AutoMutex lock(mHardwareLock); 6913 mHardwareStatus = AUDIO_HW_SET_MODE; 6914 outHwDev->set_mode(outHwDev, mMode); 6915 6916 // Determine the level of master volume/master mute support the primary 6917 // audio HAL has, and set the initial master volume/mute state at the same 6918 // time. 6919 float initialVolume = 1.0; 6920 bool initialMute = false; 6921 mMasterVolumeSupportLvl = MVS_NONE; 6922 mMasterMuteSupportLvl = MMS_NONE; 6923 6924 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6925 if ((NULL != outHwDev->get_master_volume) && 6926 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6927 mMasterVolumeSupportLvl = MVS_FULL; 6928 } else { 6929 mMasterVolumeSupportLvl = MVS_SETONLY; 6930 initialVolume = 1.0; 6931 } 6932 6933 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6934 if ((NULL == outHwDev->set_master_volume) || 6935 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6936 mMasterVolumeSupportLvl = MVS_NONE; 6937 } 6938 6939 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6940 if ((NULL != outHwDev->get_master_mute) && 6941 (NO_ERROR == outHwDev->get_master_mute(outHwDev, &initialMute))) { 6942 mMasterMuteSupportLvl = MMS_FULL; 6943 } else { 6944 mMasterMuteSupportLvl = MMS_SETONLY; 6945 initialMute = 0; 6946 } 6947 6948 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6949 if ((NULL == outHwDev->set_master_mute) || 6950 (NO_ERROR != outHwDev->set_master_mute(outHwDev, initialMute))) { 6951 mMasterMuteSupportLvl = MMS_NONE; 6952 } 6953 6954 // now that we have a primary device, initialize master volume/mute 6955 // on other devices 6956 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6957 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6958 6959 if ((dev != mPrimaryHardwareDev) && 6960 (NULL != dev->set_master_volume)) { 6961 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6962 dev->set_master_volume(dev, initialVolume); 6963 } 6964 6965 if (NULL != dev->set_master_mute) { 6966 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6967 dev->set_master_mute(dev, initialMute); 6968 } 6969 } 6970 6971 mHardwareStatus = AUDIO_HW_IDLE; 6972 mMasterVolumeSW = initialVolume; 6973 mMasterVolume = initialVolume; 6974 mMasterMuteSW = initialMute; 6975 mMasterMute = initialMute; 6976 } 6977 return id; 6978 } 6979 6980 return 0; 6981} 6982 6983audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6984 audio_io_handle_t output2) 6985{ 6986 Mutex::Autolock _l(mLock); 6987 MixerThread *thread1 = checkMixerThread_l(output1); 6988 MixerThread *thread2 = checkMixerThread_l(output2); 6989 6990 if (thread1 == NULL || thread2 == NULL) { 6991 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6992 return 0; 6993 } 6994 6995 audio_io_handle_t id = nextUniqueId(); 6996 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6997 thread->addOutputTrack(thread2); 6998 mPlaybackThreads.add(id, thread); 6999 // notify client processes of the new output creation 7000 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7001 return id; 7002} 7003 7004status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7005{ 7006 return closeOutput_nonvirtual(output); 7007} 7008 7009status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7010{ 7011 // keep strong reference on the playback thread so that 7012 // it is not destroyed while exit() is executed 7013 sp<PlaybackThread> thread; 7014 { 7015 Mutex::Autolock _l(mLock); 7016 thread = checkPlaybackThread_l(output); 7017 if (thread == NULL) { 7018 return BAD_VALUE; 7019 } 7020 7021 ALOGV("closeOutput() %d", output); 7022 7023 if (thread->type() == ThreadBase::MIXER) { 7024 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7025 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7026 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7027 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7028 } 7029 } 7030 } 7031 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7032 mPlaybackThreads.removeItem(output); 7033 } 7034 thread->exit(); 7035 // The thread entity (active unit of execution) is no longer running here, 7036 // but the ThreadBase container still exists. 7037 7038 if (thread->type() != ThreadBase::DUPLICATING) { 7039 AudioStreamOut *out = thread->clearOutput(); 7040 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7041 // from now on thread->mOutput is NULL 7042 out->hwDev->close_output_stream(out->hwDev, out->stream); 7043 delete out; 7044 } 7045 return NO_ERROR; 7046} 7047 7048status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7049{ 7050 Mutex::Autolock _l(mLock); 7051 PlaybackThread *thread = checkPlaybackThread_l(output); 7052 7053 if (thread == NULL) { 7054 return BAD_VALUE; 7055 } 7056 7057 ALOGV("suspendOutput() %d", output); 7058 thread->suspend(); 7059 7060 return NO_ERROR; 7061} 7062 7063status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7064{ 7065 Mutex::Autolock _l(mLock); 7066 PlaybackThread *thread = checkPlaybackThread_l(output); 7067 7068 if (thread == NULL) { 7069 return BAD_VALUE; 7070 } 7071 7072 ALOGV("restoreOutput() %d", output); 7073 7074 thread->restore(); 7075 7076 return NO_ERROR; 7077} 7078 7079audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7080 audio_devices_t *pDevices, 7081 uint32_t *pSamplingRate, 7082 audio_format_t *pFormat, 7083 audio_channel_mask_t *pChannelMask) 7084{ 7085 status_t status; 7086 RecordThread *thread = NULL; 7087 struct audio_config config = { 7088 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7089 channel_mask: pChannelMask ? *pChannelMask : 0, 7090 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7091 }; 7092 uint32_t reqSamplingRate = config.sample_rate; 7093 audio_format_t reqFormat = config.format; 7094 audio_channel_mask_t reqChannels = config.channel_mask; 7095 audio_stream_in_t *inStream = NULL; 7096 audio_hw_device_t *inHwDev; 7097 7098 if (pDevices == NULL || *pDevices == 0) { 7099 return 0; 7100 } 7101 7102 Mutex::Autolock _l(mLock); 7103 7104 inHwDev = findSuitableHwDev_l(module, *pDevices); 7105 if (inHwDev == NULL) 7106 return 0; 7107 7108 audio_io_handle_t id = nextUniqueId(); 7109 7110 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 7111 &inStream); 7112 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7113 inStream, 7114 config.sample_rate, 7115 config.format, 7116 config.channel_mask, 7117 status); 7118 7119 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7120 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7121 // or stereo to mono conversions on 16 bit PCM inputs. 7122 if (status == BAD_VALUE && 7123 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7124 (config.sample_rate <= 2 * reqSamplingRate) && 7125 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7126 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7127 inStream = NULL; 7128 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 7129 } 7130 7131 if (status == NO_ERROR && inStream != NULL) { 7132 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7133 7134 // Start record thread 7135 // RecorThread require both input and output device indication to forward to audio 7136 // pre processing modules 7137 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7138 thread = new RecordThread(this, 7139 input, 7140 reqSamplingRate, 7141 reqChannels, 7142 id, 7143 device); 7144 mRecordThreads.add(id, thread); 7145 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7146 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7147 if (pFormat != NULL) *pFormat = config.format; 7148 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7149 7150 // notify client processes of the new input creation 7151 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7152 return id; 7153 } 7154 7155 return 0; 7156} 7157 7158status_t AudioFlinger::closeInput(audio_io_handle_t input) 7159{ 7160 return closeInput_nonvirtual(input); 7161} 7162 7163status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7164{ 7165 // keep strong reference on the record thread so that 7166 // it is not destroyed while exit() is executed 7167 sp<RecordThread> thread; 7168 { 7169 Mutex::Autolock _l(mLock); 7170 thread = checkRecordThread_l(input); 7171 if (thread == 0) { 7172 return BAD_VALUE; 7173 } 7174 7175 ALOGV("closeInput() %d", input); 7176 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7177 mRecordThreads.removeItem(input); 7178 } 7179 thread->exit(); 7180 // The thread entity (active unit of execution) is no longer running here, 7181 // but the ThreadBase container still exists. 7182 7183 AudioStreamIn *in = thread->clearInput(); 7184 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7185 // from now on thread->mInput is NULL 7186 in->hwDev->close_input_stream(in->hwDev, in->stream); 7187 delete in; 7188 7189 return NO_ERROR; 7190} 7191 7192status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7193{ 7194 Mutex::Autolock _l(mLock); 7195 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7196 7197 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7198 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7199 thread->invalidateTracks(stream); 7200 } 7201 7202 return NO_ERROR; 7203} 7204 7205 7206int AudioFlinger::newAudioSessionId() 7207{ 7208 return nextUniqueId(); 7209} 7210 7211void AudioFlinger::acquireAudioSessionId(int audioSession) 7212{ 7213 Mutex::Autolock _l(mLock); 7214 pid_t caller = IPCThreadState::self()->getCallingPid(); 7215 ALOGV("acquiring %d from %d", audioSession, caller); 7216 size_t num = mAudioSessionRefs.size(); 7217 for (size_t i = 0; i< num; i++) { 7218 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7219 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7220 ref->mCnt++; 7221 ALOGV(" incremented refcount to %d", ref->mCnt); 7222 return; 7223 } 7224 } 7225 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7226 ALOGV(" added new entry for %d", audioSession); 7227} 7228 7229void AudioFlinger::releaseAudioSessionId(int audioSession) 7230{ 7231 Mutex::Autolock _l(mLock); 7232 pid_t caller = IPCThreadState::self()->getCallingPid(); 7233 ALOGV("releasing %d from %d", audioSession, caller); 7234 size_t num = mAudioSessionRefs.size(); 7235 for (size_t i = 0; i< num; i++) { 7236 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7237 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7238 ref->mCnt--; 7239 ALOGV(" decremented refcount to %d", ref->mCnt); 7240 if (ref->mCnt == 0) { 7241 mAudioSessionRefs.removeAt(i); 7242 delete ref; 7243 purgeStaleEffects_l(); 7244 } 7245 return; 7246 } 7247 } 7248 ALOGW("session id %d not found for pid %d", audioSession, caller); 7249} 7250 7251void AudioFlinger::purgeStaleEffects_l() { 7252 7253 ALOGV("purging stale effects"); 7254 7255 Vector< sp<EffectChain> > chains; 7256 7257 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7258 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7259 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7260 sp<EffectChain> ec = t->mEffectChains[j]; 7261 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7262 chains.push(ec); 7263 } 7264 } 7265 } 7266 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7267 sp<RecordThread> t = mRecordThreads.valueAt(i); 7268 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7269 sp<EffectChain> ec = t->mEffectChains[j]; 7270 chains.push(ec); 7271 } 7272 } 7273 7274 for (size_t i = 0; i < chains.size(); i++) { 7275 sp<EffectChain> ec = chains[i]; 7276 int sessionid = ec->sessionId(); 7277 sp<ThreadBase> t = ec->mThread.promote(); 7278 if (t == 0) { 7279 continue; 7280 } 7281 size_t numsessionrefs = mAudioSessionRefs.size(); 7282 bool found = false; 7283 for (size_t k = 0; k < numsessionrefs; k++) { 7284 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7285 if (ref->mSessionid == sessionid) { 7286 ALOGV(" session %d still exists for %d with %d refs", 7287 sessionid, ref->mPid, ref->mCnt); 7288 found = true; 7289 break; 7290 } 7291 } 7292 if (!found) { 7293 Mutex::Autolock _l (t->mLock); 7294 // remove all effects from the chain 7295 while (ec->mEffects.size()) { 7296 sp<EffectModule> effect = ec->mEffects[0]; 7297 effect->unPin(); 7298 t->removeEffect_l(effect); 7299 if (effect->purgeHandles()) { 7300 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7301 } 7302 AudioSystem::unregisterEffect(effect->id()); 7303 } 7304 } 7305 } 7306 return; 7307} 7308 7309// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7310AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7311{ 7312 return mPlaybackThreads.valueFor(output).get(); 7313} 7314 7315// checkMixerThread_l() must be called with AudioFlinger::mLock held 7316AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7317{ 7318 PlaybackThread *thread = checkPlaybackThread_l(output); 7319 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7320} 7321 7322// checkRecordThread_l() must be called with AudioFlinger::mLock held 7323AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7324{ 7325 return mRecordThreads.valueFor(input).get(); 7326} 7327 7328uint32_t AudioFlinger::nextUniqueId() 7329{ 7330 return android_atomic_inc(&mNextUniqueId); 7331} 7332 7333AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7334{ 7335 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7336 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7337 AudioStreamOut *output = thread->getOutput(); 7338 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7339 return thread; 7340 } 7341 } 7342 return NULL; 7343} 7344 7345audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7346{ 7347 PlaybackThread *thread = primaryPlaybackThread_l(); 7348 7349 if (thread == NULL) { 7350 return 0; 7351 } 7352 7353 return thread->device(); 7354} 7355 7356sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7357 int triggerSession, 7358 int listenerSession, 7359 sync_event_callback_t callBack, 7360 void *cookie) 7361{ 7362 Mutex::Autolock _l(mLock); 7363 7364 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7365 status_t playStatus = NAME_NOT_FOUND; 7366 status_t recStatus = NAME_NOT_FOUND; 7367 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7368 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7369 if (playStatus == NO_ERROR) { 7370 return event; 7371 } 7372 } 7373 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7374 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7375 if (recStatus == NO_ERROR) { 7376 return event; 7377 } 7378 } 7379 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7380 mPendingSyncEvents.add(event); 7381 } else { 7382 ALOGV("createSyncEvent() invalid event %d", event->type()); 7383 event.clear(); 7384 } 7385 return event; 7386} 7387 7388// ---------------------------------------------------------------------------- 7389// Effect management 7390// ---------------------------------------------------------------------------- 7391 7392 7393status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7394{ 7395 Mutex::Autolock _l(mLock); 7396 return EffectQueryNumberEffects(numEffects); 7397} 7398 7399status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7400{ 7401 Mutex::Autolock _l(mLock); 7402 return EffectQueryEffect(index, descriptor); 7403} 7404 7405status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7406 effect_descriptor_t *descriptor) const 7407{ 7408 Mutex::Autolock _l(mLock); 7409 return EffectGetDescriptor(pUuid, descriptor); 7410} 7411 7412 7413sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7414 effect_descriptor_t *pDesc, 7415 const sp<IEffectClient>& effectClient, 7416 int32_t priority, 7417 audio_io_handle_t io, 7418 int sessionId, 7419 status_t *status, 7420 int *id, 7421 int *enabled) 7422{ 7423 status_t lStatus = NO_ERROR; 7424 sp<EffectHandle> handle; 7425 effect_descriptor_t desc; 7426 7427 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7428 pid, effectClient.get(), priority, sessionId, io); 7429 7430 if (pDesc == NULL) { 7431 lStatus = BAD_VALUE; 7432 goto Exit; 7433 } 7434 7435 // check audio settings permission for global effects 7436 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7437 lStatus = PERMISSION_DENIED; 7438 goto Exit; 7439 } 7440 7441 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7442 // that can only be created by audio policy manager (running in same process) 7443 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7444 lStatus = PERMISSION_DENIED; 7445 goto Exit; 7446 } 7447 7448 if (io == 0) { 7449 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7450 // output must be specified by AudioPolicyManager when using session 7451 // AUDIO_SESSION_OUTPUT_STAGE 7452 lStatus = BAD_VALUE; 7453 goto Exit; 7454 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7455 // if the output returned by getOutputForEffect() is removed before we lock the 7456 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7457 // and we will exit safely 7458 io = AudioSystem::getOutputForEffect(&desc); 7459 } 7460 } 7461 7462 { 7463 Mutex::Autolock _l(mLock); 7464 7465 7466 if (!EffectIsNullUuid(&pDesc->uuid)) { 7467 // if uuid is specified, request effect descriptor 7468 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7469 if (lStatus < 0) { 7470 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7471 goto Exit; 7472 } 7473 } else { 7474 // if uuid is not specified, look for an available implementation 7475 // of the required type in effect factory 7476 if (EffectIsNullUuid(&pDesc->type)) { 7477 ALOGW("createEffect() no effect type"); 7478 lStatus = BAD_VALUE; 7479 goto Exit; 7480 } 7481 uint32_t numEffects = 0; 7482 effect_descriptor_t d; 7483 d.flags = 0; // prevent compiler warning 7484 bool found = false; 7485 7486 lStatus = EffectQueryNumberEffects(&numEffects); 7487 if (lStatus < 0) { 7488 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7489 goto Exit; 7490 } 7491 for (uint32_t i = 0; i < numEffects; i++) { 7492 lStatus = EffectQueryEffect(i, &desc); 7493 if (lStatus < 0) { 7494 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7495 continue; 7496 } 7497 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7498 // If matching type found save effect descriptor. If the session is 7499 // 0 and the effect is not auxiliary, continue enumeration in case 7500 // an auxiliary version of this effect type is available 7501 found = true; 7502 d = desc; 7503 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7504 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7505 break; 7506 } 7507 } 7508 } 7509 if (!found) { 7510 lStatus = BAD_VALUE; 7511 ALOGW("createEffect() effect not found"); 7512 goto Exit; 7513 } 7514 // For same effect type, chose auxiliary version over insert version if 7515 // connect to output mix (Compliance to OpenSL ES) 7516 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7517 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7518 desc = d; 7519 } 7520 } 7521 7522 // Do not allow auxiliary effects on a session different from 0 (output mix) 7523 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7524 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7525 lStatus = INVALID_OPERATION; 7526 goto Exit; 7527 } 7528 7529 // check recording permission for visualizer 7530 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7531 !recordingAllowed()) { 7532 lStatus = PERMISSION_DENIED; 7533 goto Exit; 7534 } 7535 7536 // return effect descriptor 7537 *pDesc = desc; 7538 7539 // If output is not specified try to find a matching audio session ID in one of the 7540 // output threads. 7541 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7542 // because of code checking output when entering the function. 7543 // Note: io is never 0 when creating an effect on an input 7544 if (io == 0) { 7545 // look for the thread where the specified audio session is present 7546 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7547 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7548 io = mPlaybackThreads.keyAt(i); 7549 break; 7550 } 7551 } 7552 if (io == 0) { 7553 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7554 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7555 io = mRecordThreads.keyAt(i); 7556 break; 7557 } 7558 } 7559 } 7560 // If no output thread contains the requested session ID, default to 7561 // first output. The effect chain will be moved to the correct output 7562 // thread when a track with the same session ID is created 7563 if (io == 0 && mPlaybackThreads.size()) { 7564 io = mPlaybackThreads.keyAt(0); 7565 } 7566 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7567 } 7568 ThreadBase *thread = checkRecordThread_l(io); 7569 if (thread == NULL) { 7570 thread = checkPlaybackThread_l(io); 7571 if (thread == NULL) { 7572 ALOGE("createEffect() unknown output thread"); 7573 lStatus = BAD_VALUE; 7574 goto Exit; 7575 } 7576 } 7577 7578 sp<Client> client = registerPid_l(pid); 7579 7580 // create effect on selected output thread 7581 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7582 &desc, enabled, &lStatus); 7583 if (handle != 0 && id != NULL) { 7584 *id = handle->id(); 7585 } 7586 } 7587 7588Exit: 7589 if (status != NULL) { 7590 *status = lStatus; 7591 } 7592 return handle; 7593} 7594 7595status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7596 audio_io_handle_t dstOutput) 7597{ 7598 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7599 sessionId, srcOutput, dstOutput); 7600 Mutex::Autolock _l(mLock); 7601 if (srcOutput == dstOutput) { 7602 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7603 return NO_ERROR; 7604 } 7605 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7606 if (srcThread == NULL) { 7607 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7608 return BAD_VALUE; 7609 } 7610 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7611 if (dstThread == NULL) { 7612 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7613 return BAD_VALUE; 7614 } 7615 7616 Mutex::Autolock _dl(dstThread->mLock); 7617 Mutex::Autolock _sl(srcThread->mLock); 7618 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7619 7620 return NO_ERROR; 7621} 7622 7623// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7624status_t AudioFlinger::moveEffectChain_l(int sessionId, 7625 AudioFlinger::PlaybackThread *srcThread, 7626 AudioFlinger::PlaybackThread *dstThread, 7627 bool reRegister) 7628{ 7629 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7630 sessionId, srcThread, dstThread); 7631 7632 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7633 if (chain == 0) { 7634 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7635 sessionId, srcThread); 7636 return INVALID_OPERATION; 7637 } 7638 7639 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7640 // so that a new chain is created with correct parameters when first effect is added. This is 7641 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7642 // removed. 7643 srcThread->removeEffectChain_l(chain); 7644 7645 // transfer all effects one by one so that new effect chain is created on new thread with 7646 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7647 audio_io_handle_t dstOutput = dstThread->id(); 7648 sp<EffectChain> dstChain; 7649 uint32_t strategy = 0; // prevent compiler warning 7650 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7651 while (effect != 0) { 7652 srcThread->removeEffect_l(effect); 7653 dstThread->addEffect_l(effect); 7654 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7655 if (effect->state() == EffectModule::ACTIVE || 7656 effect->state() == EffectModule::STOPPING) { 7657 effect->start(); 7658 } 7659 // if the move request is not received from audio policy manager, the effect must be 7660 // re-registered with the new strategy and output 7661 if (dstChain == 0) { 7662 dstChain = effect->chain().promote(); 7663 if (dstChain == 0) { 7664 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7665 srcThread->addEffect_l(effect); 7666 return NO_INIT; 7667 } 7668 strategy = dstChain->strategy(); 7669 } 7670 if (reRegister) { 7671 AudioSystem::unregisterEffect(effect->id()); 7672 AudioSystem::registerEffect(&effect->desc(), 7673 dstOutput, 7674 strategy, 7675 sessionId, 7676 effect->id()); 7677 } 7678 effect = chain->getEffectFromId_l(0); 7679 } 7680 7681 return NO_ERROR; 7682} 7683 7684 7685// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7686sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7687 const sp<AudioFlinger::Client>& client, 7688 const sp<IEffectClient>& effectClient, 7689 int32_t priority, 7690 int sessionId, 7691 effect_descriptor_t *desc, 7692 int *enabled, 7693 status_t *status 7694 ) 7695{ 7696 sp<EffectModule> effect; 7697 sp<EffectHandle> handle; 7698 status_t lStatus; 7699 sp<EffectChain> chain; 7700 bool chainCreated = false; 7701 bool effectCreated = false; 7702 bool effectRegistered = false; 7703 7704 lStatus = initCheck(); 7705 if (lStatus != NO_ERROR) { 7706 ALOGW("createEffect_l() Audio driver not initialized."); 7707 goto Exit; 7708 } 7709 7710 // Do not allow effects with session ID 0 on direct output or duplicating threads 7711 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7712 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7713 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7714 desc->name, sessionId); 7715 lStatus = BAD_VALUE; 7716 goto Exit; 7717 } 7718 // Only Pre processor effects are allowed on input threads and only on input threads 7719 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7720 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7721 desc->name, desc->flags, mType); 7722 lStatus = BAD_VALUE; 7723 goto Exit; 7724 } 7725 7726 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7727 7728 { // scope for mLock 7729 Mutex::Autolock _l(mLock); 7730 7731 // check for existing effect chain with the requested audio session 7732 chain = getEffectChain_l(sessionId); 7733 if (chain == 0) { 7734 // create a new chain for this session 7735 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7736 chain = new EffectChain(this, sessionId); 7737 addEffectChain_l(chain); 7738 chain->setStrategy(getStrategyForSession_l(sessionId)); 7739 chainCreated = true; 7740 } else { 7741 effect = chain->getEffectFromDesc_l(desc); 7742 } 7743 7744 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7745 7746 if (effect == 0) { 7747 int id = mAudioFlinger->nextUniqueId(); 7748 // Check CPU and memory usage 7749 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7750 if (lStatus != NO_ERROR) { 7751 goto Exit; 7752 } 7753 effectRegistered = true; 7754 // create a new effect module if none present in the chain 7755 effect = new EffectModule(this, chain, desc, id, sessionId); 7756 lStatus = effect->status(); 7757 if (lStatus != NO_ERROR) { 7758 goto Exit; 7759 } 7760 lStatus = chain->addEffect_l(effect); 7761 if (lStatus != NO_ERROR) { 7762 goto Exit; 7763 } 7764 effectCreated = true; 7765 7766 effect->setDevice(mDevice); 7767 effect->setMode(mAudioFlinger->getMode()); 7768 } 7769 // create effect handle and connect it to effect module 7770 handle = new EffectHandle(effect, client, effectClient, priority); 7771 lStatus = effect->addHandle(handle.get()); 7772 if (enabled != NULL) { 7773 *enabled = (int)effect->isEnabled(); 7774 } 7775 } 7776 7777Exit: 7778 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7779 Mutex::Autolock _l(mLock); 7780 if (effectCreated) { 7781 chain->removeEffect_l(effect); 7782 } 7783 if (effectRegistered) { 7784 AudioSystem::unregisterEffect(effect->id()); 7785 } 7786 if (chainCreated) { 7787 removeEffectChain_l(chain); 7788 } 7789 handle.clear(); 7790 } 7791 7792 if (status != NULL) { 7793 *status = lStatus; 7794 } 7795 return handle; 7796} 7797 7798sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7799{ 7800 Mutex::Autolock _l(mLock); 7801 return getEffect_l(sessionId, effectId); 7802} 7803 7804sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7805{ 7806 sp<EffectChain> chain = getEffectChain_l(sessionId); 7807 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7808} 7809 7810// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7811// PlaybackThread::mLock held 7812status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7813{ 7814 // check for existing effect chain with the requested audio session 7815 int sessionId = effect->sessionId(); 7816 sp<EffectChain> chain = getEffectChain_l(sessionId); 7817 bool chainCreated = false; 7818 7819 if (chain == 0) { 7820 // create a new chain for this session 7821 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7822 chain = new EffectChain(this, sessionId); 7823 addEffectChain_l(chain); 7824 chain->setStrategy(getStrategyForSession_l(sessionId)); 7825 chainCreated = true; 7826 } 7827 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7828 7829 if (chain->getEffectFromId_l(effect->id()) != 0) { 7830 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7831 this, effect->desc().name, chain.get()); 7832 return BAD_VALUE; 7833 } 7834 7835 status_t status = chain->addEffect_l(effect); 7836 if (status != NO_ERROR) { 7837 if (chainCreated) { 7838 removeEffectChain_l(chain); 7839 } 7840 return status; 7841 } 7842 7843 effect->setDevice(mDevice); 7844 effect->setMode(mAudioFlinger->getMode()); 7845 return NO_ERROR; 7846} 7847 7848void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7849 7850 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7851 effect_descriptor_t desc = effect->desc(); 7852 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7853 detachAuxEffect_l(effect->id()); 7854 } 7855 7856 sp<EffectChain> chain = effect->chain().promote(); 7857 if (chain != 0) { 7858 // remove effect chain if removing last effect 7859 if (chain->removeEffect_l(effect) == 0) { 7860 removeEffectChain_l(chain); 7861 } 7862 } else { 7863 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7864 } 7865} 7866 7867void AudioFlinger::ThreadBase::lockEffectChains_l( 7868 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7869{ 7870 effectChains = mEffectChains; 7871 for (size_t i = 0; i < mEffectChains.size(); i++) { 7872 mEffectChains[i]->lock(); 7873 } 7874} 7875 7876void AudioFlinger::ThreadBase::unlockEffectChains( 7877 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7878{ 7879 for (size_t i = 0; i < effectChains.size(); i++) { 7880 effectChains[i]->unlock(); 7881 } 7882} 7883 7884sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7885{ 7886 Mutex::Autolock _l(mLock); 7887 return getEffectChain_l(sessionId); 7888} 7889 7890sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7891{ 7892 size_t size = mEffectChains.size(); 7893 for (size_t i = 0; i < size; i++) { 7894 if (mEffectChains[i]->sessionId() == sessionId) { 7895 return mEffectChains[i]; 7896 } 7897 } 7898 return 0; 7899} 7900 7901void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7902{ 7903 Mutex::Autolock _l(mLock); 7904 size_t size = mEffectChains.size(); 7905 for (size_t i = 0; i < size; i++) { 7906 mEffectChains[i]->setMode_l(mode); 7907 } 7908} 7909 7910void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7911 EffectHandle *handle, 7912 bool unpinIfLast) { 7913 7914 Mutex::Autolock _l(mLock); 7915 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7916 // delete the effect module if removing last handle on it 7917 if (effect->removeHandle(handle) == 0) { 7918 if (!effect->isPinned() || unpinIfLast) { 7919 removeEffect_l(effect); 7920 AudioSystem::unregisterEffect(effect->id()); 7921 } 7922 } 7923} 7924 7925status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7926{ 7927 int session = chain->sessionId(); 7928 int16_t *buffer = mMixBuffer; 7929 bool ownsBuffer = false; 7930 7931 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7932 if (session > 0) { 7933 // Only one effect chain can be present in direct output thread and it uses 7934 // the mix buffer as input 7935 if (mType != DIRECT) { 7936 size_t numSamples = mNormalFrameCount * mChannelCount; 7937 buffer = new int16_t[numSamples]; 7938 memset(buffer, 0, numSamples * sizeof(int16_t)); 7939 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7940 ownsBuffer = true; 7941 } 7942 7943 // Attach all tracks with same session ID to this chain. 7944 for (size_t i = 0; i < mTracks.size(); ++i) { 7945 sp<Track> track = mTracks[i]; 7946 if (session == track->sessionId()) { 7947 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7948 track->setMainBuffer(buffer); 7949 chain->incTrackCnt(); 7950 } 7951 } 7952 7953 // indicate all active tracks in the chain 7954 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7955 sp<Track> track = mActiveTracks[i].promote(); 7956 if (track == 0) continue; 7957 if (session == track->sessionId()) { 7958 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7959 chain->incActiveTrackCnt(); 7960 } 7961 } 7962 } 7963 7964 chain->setInBuffer(buffer, ownsBuffer); 7965 chain->setOutBuffer(mMixBuffer); 7966 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7967 // chains list in order to be processed last as it contains output stage effects 7968 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7969 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7970 // after track specific effects and before output stage 7971 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7972 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7973 // Effect chain for other sessions are inserted at beginning of effect 7974 // chains list to be processed before output mix effects. Relative order between other 7975 // sessions is not important 7976 size_t size = mEffectChains.size(); 7977 size_t i = 0; 7978 for (i = 0; i < size; i++) { 7979 if (mEffectChains[i]->sessionId() < session) break; 7980 } 7981 mEffectChains.insertAt(chain, i); 7982 checkSuspendOnAddEffectChain_l(chain); 7983 7984 return NO_ERROR; 7985} 7986 7987size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7988{ 7989 int session = chain->sessionId(); 7990 7991 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7992 7993 for (size_t i = 0; i < mEffectChains.size(); i++) { 7994 if (chain == mEffectChains[i]) { 7995 mEffectChains.removeAt(i); 7996 // detach all active tracks from the chain 7997 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7998 sp<Track> track = mActiveTracks[i].promote(); 7999 if (track == 0) continue; 8000 if (session == track->sessionId()) { 8001 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8002 chain.get(), session); 8003 chain->decActiveTrackCnt(); 8004 } 8005 } 8006 8007 // detach all tracks with same session ID from this chain 8008 for (size_t i = 0; i < mTracks.size(); ++i) { 8009 sp<Track> track = mTracks[i]; 8010 if (session == track->sessionId()) { 8011 track->setMainBuffer(mMixBuffer); 8012 chain->decTrackCnt(); 8013 } 8014 } 8015 break; 8016 } 8017 } 8018 return mEffectChains.size(); 8019} 8020 8021status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8022 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8023{ 8024 Mutex::Autolock _l(mLock); 8025 return attachAuxEffect_l(track, EffectId); 8026} 8027 8028status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8029 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8030{ 8031 status_t status = NO_ERROR; 8032 8033 if (EffectId == 0) { 8034 track->setAuxBuffer(0, NULL); 8035 } else { 8036 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8037 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8038 if (effect != 0) { 8039 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8040 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8041 } else { 8042 status = INVALID_OPERATION; 8043 } 8044 } else { 8045 status = BAD_VALUE; 8046 } 8047 } 8048 return status; 8049} 8050 8051void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8052{ 8053 for (size_t i = 0; i < mTracks.size(); ++i) { 8054 sp<Track> track = mTracks[i]; 8055 if (track->auxEffectId() == effectId) { 8056 attachAuxEffect_l(track, 0); 8057 } 8058 } 8059} 8060 8061status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8062{ 8063 // only one chain per input thread 8064 if (mEffectChains.size() != 0) { 8065 return INVALID_OPERATION; 8066 } 8067 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8068 8069 chain->setInBuffer(NULL); 8070 chain->setOutBuffer(NULL); 8071 8072 checkSuspendOnAddEffectChain_l(chain); 8073 8074 mEffectChains.add(chain); 8075 8076 return NO_ERROR; 8077} 8078 8079size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8080{ 8081 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8082 ALOGW_IF(mEffectChains.size() != 1, 8083 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8084 chain.get(), mEffectChains.size(), this); 8085 if (mEffectChains.size() == 1) { 8086 mEffectChains.removeAt(0); 8087 } 8088 return 0; 8089} 8090 8091// ---------------------------------------------------------------------------- 8092// EffectModule implementation 8093// ---------------------------------------------------------------------------- 8094 8095#undef LOG_TAG 8096#define LOG_TAG "AudioFlinger::EffectModule" 8097 8098AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8099 const wp<AudioFlinger::EffectChain>& chain, 8100 effect_descriptor_t *desc, 8101 int id, 8102 int sessionId) 8103 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8104 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8105 mDescriptor(*desc), 8106 // mConfig is set by configure() and not used before then 8107 mEffectInterface(NULL), 8108 mStatus(NO_INIT), mState(IDLE), 8109 // mMaxDisableWaitCnt is set by configure() and not used before then 8110 // mDisableWaitCnt is set by process() and updateState() and not used before then 8111 mSuspended(false) 8112{ 8113 ALOGV("Constructor %p", this); 8114 int lStatus; 8115 8116 // create effect engine from effect factory 8117 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8118 8119 if (mStatus != NO_ERROR) { 8120 return; 8121 } 8122 lStatus = init(); 8123 if (lStatus < 0) { 8124 mStatus = lStatus; 8125 goto Error; 8126 } 8127 8128 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8129 return; 8130Error: 8131 EffectRelease(mEffectInterface); 8132 mEffectInterface = NULL; 8133 ALOGV("Constructor Error %d", mStatus); 8134} 8135 8136AudioFlinger::EffectModule::~EffectModule() 8137{ 8138 ALOGV("Destructor %p", this); 8139 if (mEffectInterface != NULL) { 8140 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8141 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8142 sp<ThreadBase> thread = mThread.promote(); 8143 if (thread != 0) { 8144 audio_stream_t *stream = thread->stream(); 8145 if (stream != NULL) { 8146 stream->remove_audio_effect(stream, mEffectInterface); 8147 } 8148 } 8149 } 8150 // release effect engine 8151 EffectRelease(mEffectInterface); 8152 } 8153} 8154 8155status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8156{ 8157 status_t status; 8158 8159 Mutex::Autolock _l(mLock); 8160 int priority = handle->priority(); 8161 size_t size = mHandles.size(); 8162 EffectHandle *controlHandle = NULL; 8163 size_t i; 8164 for (i = 0; i < size; i++) { 8165 EffectHandle *h = mHandles[i]; 8166 if (h == NULL || h->destroyed_l()) continue; 8167 // first non destroyed handle is considered in control 8168 if (controlHandle == NULL) 8169 controlHandle = h; 8170 if (h->priority() <= priority) break; 8171 } 8172 // if inserted in first place, move effect control from previous owner to this handle 8173 if (i == 0) { 8174 bool enabled = false; 8175 if (controlHandle != NULL) { 8176 enabled = controlHandle->enabled(); 8177 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8178 } 8179 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8180 status = NO_ERROR; 8181 } else { 8182 status = ALREADY_EXISTS; 8183 } 8184 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8185 mHandles.insertAt(handle, i); 8186 return status; 8187} 8188 8189size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8190{ 8191 Mutex::Autolock _l(mLock); 8192 size_t size = mHandles.size(); 8193 size_t i; 8194 for (i = 0; i < size; i++) { 8195 if (mHandles[i] == handle) break; 8196 } 8197 if (i == size) { 8198 return size; 8199 } 8200 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8201 8202 mHandles.removeAt(i); 8203 // if removed from first place, move effect control from this handle to next in line 8204 if (i == 0) { 8205 EffectHandle *h = controlHandle_l(); 8206 if (h != NULL) { 8207 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8208 } 8209 } 8210 8211 // Prevent calls to process() and other functions on effect interface from now on. 8212 // The effect engine will be released by the destructor when the last strong reference on 8213 // this object is released which can happen after next process is called. 8214 if (mHandles.size() == 0 && !mPinned) { 8215 mState = DESTROYED; 8216 } 8217 8218 return mHandles.size(); 8219} 8220 8221// must be called with EffectModule::mLock held 8222AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8223{ 8224 // the first valid handle in the list has control over the module 8225 for (size_t i = 0; i < mHandles.size(); i++) { 8226 EffectHandle *h = mHandles[i]; 8227 if (h != NULL && !h->destroyed_l()) { 8228 return h; 8229 } 8230 } 8231 8232 return NULL; 8233} 8234 8235size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8236{ 8237 ALOGV("disconnect() %p handle %p", this, handle); 8238 // keep a strong reference on this EffectModule to avoid calling the 8239 // destructor before we exit 8240 sp<EffectModule> keep(this); 8241 { 8242 sp<ThreadBase> thread = mThread.promote(); 8243 if (thread != 0) { 8244 thread->disconnectEffect(keep, handle, unpinIfLast); 8245 } 8246 } 8247 return mHandles.size(); 8248} 8249 8250void AudioFlinger::EffectModule::updateState() { 8251 Mutex::Autolock _l(mLock); 8252 8253 switch (mState) { 8254 case RESTART: 8255 reset_l(); 8256 // FALL THROUGH 8257 8258 case STARTING: 8259 // clear auxiliary effect input buffer for next accumulation 8260 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8261 memset(mConfig.inputCfg.buffer.raw, 8262 0, 8263 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8264 } 8265 start_l(); 8266 mState = ACTIVE; 8267 break; 8268 case STOPPING: 8269 stop_l(); 8270 mDisableWaitCnt = mMaxDisableWaitCnt; 8271 mState = STOPPED; 8272 break; 8273 case STOPPED: 8274 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8275 // turn off sequence. 8276 if (--mDisableWaitCnt == 0) { 8277 reset_l(); 8278 mState = IDLE; 8279 } 8280 break; 8281 default: //IDLE , ACTIVE, DESTROYED 8282 break; 8283 } 8284} 8285 8286void AudioFlinger::EffectModule::process() 8287{ 8288 Mutex::Autolock _l(mLock); 8289 8290 if (mState == DESTROYED || mEffectInterface == NULL || 8291 mConfig.inputCfg.buffer.raw == NULL || 8292 mConfig.outputCfg.buffer.raw == NULL) { 8293 return; 8294 } 8295 8296 if (isProcessEnabled()) { 8297 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8298 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8299 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8300 mConfig.inputCfg.buffer.s32, 8301 mConfig.inputCfg.buffer.frameCount/2); 8302 } 8303 8304 // do the actual processing in the effect engine 8305 int ret = (*mEffectInterface)->process(mEffectInterface, 8306 &mConfig.inputCfg.buffer, 8307 &mConfig.outputCfg.buffer); 8308 8309 // force transition to IDLE state when engine is ready 8310 if (mState == STOPPED && ret == -ENODATA) { 8311 mDisableWaitCnt = 1; 8312 } 8313 8314 // clear auxiliary effect input buffer for next accumulation 8315 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8316 memset(mConfig.inputCfg.buffer.raw, 0, 8317 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8318 } 8319 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8320 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8321 // If an insert effect is idle and input buffer is different from output buffer, 8322 // accumulate input onto output 8323 sp<EffectChain> chain = mChain.promote(); 8324 if (chain != 0 && chain->activeTrackCnt() != 0) { 8325 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8326 int16_t *in = mConfig.inputCfg.buffer.s16; 8327 int16_t *out = mConfig.outputCfg.buffer.s16; 8328 for (size_t i = 0; i < frameCnt; i++) { 8329 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8330 } 8331 } 8332 } 8333} 8334 8335void AudioFlinger::EffectModule::reset_l() 8336{ 8337 if (mEffectInterface == NULL) { 8338 return; 8339 } 8340 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8341} 8342 8343status_t AudioFlinger::EffectModule::configure() 8344{ 8345 if (mEffectInterface == NULL) { 8346 return NO_INIT; 8347 } 8348 8349 sp<ThreadBase> thread = mThread.promote(); 8350 if (thread == 0) { 8351 return DEAD_OBJECT; 8352 } 8353 8354 // TODO: handle configuration of effects replacing track process 8355 audio_channel_mask_t channelMask = thread->channelMask(); 8356 8357 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8358 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8359 } else { 8360 mConfig.inputCfg.channels = channelMask; 8361 } 8362 mConfig.outputCfg.channels = channelMask; 8363 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8364 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8365 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8366 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8367 mConfig.inputCfg.bufferProvider.cookie = NULL; 8368 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8369 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8370 mConfig.outputCfg.bufferProvider.cookie = NULL; 8371 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8372 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8373 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8374 // Insert effect: 8375 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8376 // always overwrites output buffer: input buffer == output buffer 8377 // - in other sessions: 8378 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8379 // other effect: overwrites output buffer: input buffer == output buffer 8380 // Auxiliary effect: 8381 // accumulates in output buffer: input buffer != output buffer 8382 // Therefore: accumulate <=> input buffer != output buffer 8383 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8384 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8385 } else { 8386 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8387 } 8388 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8389 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8390 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8391 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8392 8393 ALOGV("configure() %p thread %p buffer %p framecount %d", 8394 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8395 8396 status_t cmdStatus; 8397 uint32_t size = sizeof(int); 8398 status_t status = (*mEffectInterface)->command(mEffectInterface, 8399 EFFECT_CMD_SET_CONFIG, 8400 sizeof(effect_config_t), 8401 &mConfig, 8402 &size, 8403 &cmdStatus); 8404 if (status == 0) { 8405 status = cmdStatus; 8406 } 8407 8408 if (status == 0 && 8409 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8410 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8411 effect_param_t *p = (effect_param_t *)buf32; 8412 8413 p->psize = sizeof(uint32_t); 8414 p->vsize = sizeof(uint32_t); 8415 size = sizeof(int); 8416 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8417 8418 uint32_t latency = 0; 8419 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8420 if (pbt != NULL) { 8421 latency = pbt->latency_l(); 8422 } 8423 8424 *((int32_t *)p->data + 1)= latency; 8425 (*mEffectInterface)->command(mEffectInterface, 8426 EFFECT_CMD_SET_PARAM, 8427 sizeof(effect_param_t) + 8, 8428 &buf32, 8429 &size, 8430 &cmdStatus); 8431 } 8432 8433 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8434 (1000 * mConfig.outputCfg.buffer.frameCount); 8435 8436 return status; 8437} 8438 8439status_t AudioFlinger::EffectModule::init() 8440{ 8441 Mutex::Autolock _l(mLock); 8442 if (mEffectInterface == NULL) { 8443 return NO_INIT; 8444 } 8445 status_t cmdStatus; 8446 uint32_t size = sizeof(status_t); 8447 status_t status = (*mEffectInterface)->command(mEffectInterface, 8448 EFFECT_CMD_INIT, 8449 0, 8450 NULL, 8451 &size, 8452 &cmdStatus); 8453 if (status == 0) { 8454 status = cmdStatus; 8455 } 8456 return status; 8457} 8458 8459status_t AudioFlinger::EffectModule::start() 8460{ 8461 Mutex::Autolock _l(mLock); 8462 return start_l(); 8463} 8464 8465status_t AudioFlinger::EffectModule::start_l() 8466{ 8467 if (mEffectInterface == NULL) { 8468 return NO_INIT; 8469 } 8470 status_t cmdStatus; 8471 uint32_t size = sizeof(status_t); 8472 status_t status = (*mEffectInterface)->command(mEffectInterface, 8473 EFFECT_CMD_ENABLE, 8474 0, 8475 NULL, 8476 &size, 8477 &cmdStatus); 8478 if (status == 0) { 8479 status = cmdStatus; 8480 } 8481 if (status == 0 && 8482 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8483 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8484 sp<ThreadBase> thread = mThread.promote(); 8485 if (thread != 0) { 8486 audio_stream_t *stream = thread->stream(); 8487 if (stream != NULL) { 8488 stream->add_audio_effect(stream, mEffectInterface); 8489 } 8490 } 8491 } 8492 return status; 8493} 8494 8495status_t AudioFlinger::EffectModule::stop() 8496{ 8497 Mutex::Autolock _l(mLock); 8498 return stop_l(); 8499} 8500 8501status_t AudioFlinger::EffectModule::stop_l() 8502{ 8503 if (mEffectInterface == NULL) { 8504 return NO_INIT; 8505 } 8506 status_t cmdStatus; 8507 uint32_t size = sizeof(status_t); 8508 status_t status = (*mEffectInterface)->command(mEffectInterface, 8509 EFFECT_CMD_DISABLE, 8510 0, 8511 NULL, 8512 &size, 8513 &cmdStatus); 8514 if (status == 0) { 8515 status = cmdStatus; 8516 } 8517 if (status == 0 && 8518 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8519 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8520 sp<ThreadBase> thread = mThread.promote(); 8521 if (thread != 0) { 8522 audio_stream_t *stream = thread->stream(); 8523 if (stream != NULL) { 8524 stream->remove_audio_effect(stream, mEffectInterface); 8525 } 8526 } 8527 } 8528 return status; 8529} 8530 8531status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8532 uint32_t cmdSize, 8533 void *pCmdData, 8534 uint32_t *replySize, 8535 void *pReplyData) 8536{ 8537 Mutex::Autolock _l(mLock); 8538// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8539 8540 if (mState == DESTROYED || mEffectInterface == NULL) { 8541 return NO_INIT; 8542 } 8543 status_t status = (*mEffectInterface)->command(mEffectInterface, 8544 cmdCode, 8545 cmdSize, 8546 pCmdData, 8547 replySize, 8548 pReplyData); 8549 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8550 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8551 for (size_t i = 1; i < mHandles.size(); i++) { 8552 EffectHandle *h = mHandles[i]; 8553 if (h != NULL && !h->destroyed_l()) { 8554 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8555 } 8556 } 8557 } 8558 return status; 8559} 8560 8561status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8562{ 8563 Mutex::Autolock _l(mLock); 8564 return setEnabled_l(enabled); 8565} 8566 8567// must be called with EffectModule::mLock held 8568status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8569{ 8570 8571 ALOGV("setEnabled %p enabled %d", this, enabled); 8572 8573 if (enabled != isEnabled()) { 8574 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8575 if (enabled && status != NO_ERROR) { 8576 return status; 8577 } 8578 8579 switch (mState) { 8580 // going from disabled to enabled 8581 case IDLE: 8582 mState = STARTING; 8583 break; 8584 case STOPPED: 8585 mState = RESTART; 8586 break; 8587 case STOPPING: 8588 mState = ACTIVE; 8589 break; 8590 8591 // going from enabled to disabled 8592 case RESTART: 8593 mState = STOPPED; 8594 break; 8595 case STARTING: 8596 mState = IDLE; 8597 break; 8598 case ACTIVE: 8599 mState = STOPPING; 8600 break; 8601 case DESTROYED: 8602 return NO_ERROR; // simply ignore as we are being destroyed 8603 } 8604 for (size_t i = 1; i < mHandles.size(); i++) { 8605 EffectHandle *h = mHandles[i]; 8606 if (h != NULL && !h->destroyed_l()) { 8607 h->setEnabled(enabled); 8608 } 8609 } 8610 } 8611 return NO_ERROR; 8612} 8613 8614bool AudioFlinger::EffectModule::isEnabled() const 8615{ 8616 switch (mState) { 8617 case RESTART: 8618 case STARTING: 8619 case ACTIVE: 8620 return true; 8621 case IDLE: 8622 case STOPPING: 8623 case STOPPED: 8624 case DESTROYED: 8625 default: 8626 return false; 8627 } 8628} 8629 8630bool AudioFlinger::EffectModule::isProcessEnabled() const 8631{ 8632 switch (mState) { 8633 case RESTART: 8634 case ACTIVE: 8635 case STOPPING: 8636 case STOPPED: 8637 return true; 8638 case IDLE: 8639 case STARTING: 8640 case DESTROYED: 8641 default: 8642 return false; 8643 } 8644} 8645 8646status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8647{ 8648 Mutex::Autolock _l(mLock); 8649 status_t status = NO_ERROR; 8650 8651 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8652 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8653 if (isProcessEnabled() && 8654 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8655 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8656 status_t cmdStatus; 8657 uint32_t volume[2]; 8658 uint32_t *pVolume = NULL; 8659 uint32_t size = sizeof(volume); 8660 volume[0] = *left; 8661 volume[1] = *right; 8662 if (controller) { 8663 pVolume = volume; 8664 } 8665 status = (*mEffectInterface)->command(mEffectInterface, 8666 EFFECT_CMD_SET_VOLUME, 8667 size, 8668 volume, 8669 &size, 8670 pVolume); 8671 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8672 *left = volume[0]; 8673 *right = volume[1]; 8674 } 8675 } 8676 return status; 8677} 8678 8679status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8680{ 8681 Mutex::Autolock _l(mLock); 8682 status_t status = NO_ERROR; 8683 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8684 // audio pre processing modules on RecordThread can receive both output and 8685 // input device indication in the same call 8686 audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL; 8687 if (dev) { 8688 status_t cmdStatus; 8689 uint32_t size = sizeof(status_t); 8690 8691 status = (*mEffectInterface)->command(mEffectInterface, 8692 EFFECT_CMD_SET_DEVICE, 8693 sizeof(uint32_t), 8694 &dev, 8695 &size, 8696 &cmdStatus); 8697 if (status == NO_ERROR) { 8698 status = cmdStatus; 8699 } 8700 } 8701 dev = device & AUDIO_DEVICE_IN_ALL; 8702 if (dev) { 8703 status_t cmdStatus; 8704 uint32_t size = sizeof(status_t); 8705 8706 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8707 EFFECT_CMD_SET_INPUT_DEVICE, 8708 sizeof(uint32_t), 8709 &dev, 8710 &size, 8711 &cmdStatus); 8712 if (status2 == NO_ERROR) { 8713 status2 = cmdStatus; 8714 } 8715 if (status == NO_ERROR) { 8716 status = status2; 8717 } 8718 } 8719 } 8720 return status; 8721} 8722 8723status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8724{ 8725 Mutex::Autolock _l(mLock); 8726 status_t status = NO_ERROR; 8727 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8728 status_t cmdStatus; 8729 uint32_t size = sizeof(status_t); 8730 status = (*mEffectInterface)->command(mEffectInterface, 8731 EFFECT_CMD_SET_AUDIO_MODE, 8732 sizeof(audio_mode_t), 8733 &mode, 8734 &size, 8735 &cmdStatus); 8736 if (status == NO_ERROR) { 8737 status = cmdStatus; 8738 } 8739 } 8740 return status; 8741} 8742 8743void AudioFlinger::EffectModule::setSuspended(bool suspended) 8744{ 8745 Mutex::Autolock _l(mLock); 8746 mSuspended = suspended; 8747} 8748 8749bool AudioFlinger::EffectModule::suspended() const 8750{ 8751 Mutex::Autolock _l(mLock); 8752 return mSuspended; 8753} 8754 8755bool AudioFlinger::EffectModule::purgeHandles() 8756{ 8757 bool enabled = false; 8758 Mutex::Autolock _l(mLock); 8759 for (size_t i = 0; i < mHandles.size(); i++) { 8760 EffectHandle *handle = mHandles[i]; 8761 if (handle != NULL && !handle->destroyed_l()) { 8762 handle->effect().clear(); 8763 if (handle->hasControl()) { 8764 enabled = handle->enabled(); 8765 } 8766 } 8767 } 8768 return enabled; 8769} 8770 8771void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8772{ 8773 const size_t SIZE = 256; 8774 char buffer[SIZE]; 8775 String8 result; 8776 8777 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8778 result.append(buffer); 8779 8780 bool locked = tryLock(mLock); 8781 // failed to lock - AudioFlinger is probably deadlocked 8782 if (!locked) { 8783 result.append("\t\tCould not lock Fx mutex:\n"); 8784 } 8785 8786 result.append("\t\tSession Status State Engine:\n"); 8787 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8788 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8789 result.append(buffer); 8790 8791 result.append("\t\tDescriptor:\n"); 8792 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8793 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8794 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8795 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8796 result.append(buffer); 8797 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8798 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8799 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8800 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8801 result.append(buffer); 8802 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8803 mDescriptor.apiVersion, 8804 mDescriptor.flags); 8805 result.append(buffer); 8806 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8807 mDescriptor.name); 8808 result.append(buffer); 8809 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8810 mDescriptor.implementor); 8811 result.append(buffer); 8812 8813 result.append("\t\t- Input configuration:\n"); 8814 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8815 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8816 (uint32_t)mConfig.inputCfg.buffer.raw, 8817 mConfig.inputCfg.buffer.frameCount, 8818 mConfig.inputCfg.samplingRate, 8819 mConfig.inputCfg.channels, 8820 mConfig.inputCfg.format); 8821 result.append(buffer); 8822 8823 result.append("\t\t- Output configuration:\n"); 8824 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8825 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8826 (uint32_t)mConfig.outputCfg.buffer.raw, 8827 mConfig.outputCfg.buffer.frameCount, 8828 mConfig.outputCfg.samplingRate, 8829 mConfig.outputCfg.channels, 8830 mConfig.outputCfg.format); 8831 result.append(buffer); 8832 8833 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8834 result.append(buffer); 8835 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8836 for (size_t i = 0; i < mHandles.size(); ++i) { 8837 EffectHandle *handle = mHandles[i]; 8838 if (handle != NULL && !handle->destroyed_l()) { 8839 handle->dump(buffer, SIZE); 8840 result.append(buffer); 8841 } 8842 } 8843 8844 result.append("\n"); 8845 8846 write(fd, result.string(), result.length()); 8847 8848 if (locked) { 8849 mLock.unlock(); 8850 } 8851} 8852 8853// ---------------------------------------------------------------------------- 8854// EffectHandle implementation 8855// ---------------------------------------------------------------------------- 8856 8857#undef LOG_TAG 8858#define LOG_TAG "AudioFlinger::EffectHandle" 8859 8860AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8861 const sp<AudioFlinger::Client>& client, 8862 const sp<IEffectClient>& effectClient, 8863 int32_t priority) 8864 : BnEffect(), 8865 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8866 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8867{ 8868 ALOGV("constructor %p", this); 8869 8870 if (client == 0) { 8871 return; 8872 } 8873 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8874 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8875 if (mCblkMemory != 0) { 8876 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8877 8878 if (mCblk != NULL) { 8879 new(mCblk) effect_param_cblk_t(); 8880 mBuffer = (uint8_t *)mCblk + bufOffset; 8881 } 8882 } else { 8883 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8884 return; 8885 } 8886} 8887 8888AudioFlinger::EffectHandle::~EffectHandle() 8889{ 8890 ALOGV("Destructor %p", this); 8891 8892 if (mEffect == 0) { 8893 mDestroyed = true; 8894 return; 8895 } 8896 mEffect->lock(); 8897 mDestroyed = true; 8898 mEffect->unlock(); 8899 disconnect(false); 8900} 8901 8902status_t AudioFlinger::EffectHandle::enable() 8903{ 8904 ALOGV("enable %p", this); 8905 if (!mHasControl) return INVALID_OPERATION; 8906 if (mEffect == 0) return DEAD_OBJECT; 8907 8908 if (mEnabled) { 8909 return NO_ERROR; 8910 } 8911 8912 mEnabled = true; 8913 8914 sp<ThreadBase> thread = mEffect->thread().promote(); 8915 if (thread != 0) { 8916 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8917 } 8918 8919 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8920 if (mEffect->suspended()) { 8921 return NO_ERROR; 8922 } 8923 8924 status_t status = mEffect->setEnabled(true); 8925 if (status != NO_ERROR) { 8926 if (thread != 0) { 8927 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8928 } 8929 mEnabled = false; 8930 } 8931 return status; 8932} 8933 8934status_t AudioFlinger::EffectHandle::disable() 8935{ 8936 ALOGV("disable %p", this); 8937 if (!mHasControl) return INVALID_OPERATION; 8938 if (mEffect == 0) return DEAD_OBJECT; 8939 8940 if (!mEnabled) { 8941 return NO_ERROR; 8942 } 8943 mEnabled = false; 8944 8945 if (mEffect->suspended()) { 8946 return NO_ERROR; 8947 } 8948 8949 status_t status = mEffect->setEnabled(false); 8950 8951 sp<ThreadBase> thread = mEffect->thread().promote(); 8952 if (thread != 0) { 8953 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8954 } 8955 8956 return status; 8957} 8958 8959void AudioFlinger::EffectHandle::disconnect() 8960{ 8961 disconnect(true); 8962} 8963 8964void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8965{ 8966 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8967 if (mEffect == 0) { 8968 return; 8969 } 8970 // restore suspended effects if the disconnected handle was enabled and the last one. 8971 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8972 sp<ThreadBase> thread = mEffect->thread().promote(); 8973 if (thread != 0) { 8974 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8975 } 8976 } 8977 8978 // release sp on module => module destructor can be called now 8979 mEffect.clear(); 8980 if (mClient != 0) { 8981 if (mCblk != NULL) { 8982 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8983 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8984 } 8985 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8986 // Client destructor must run with AudioFlinger mutex locked 8987 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8988 mClient.clear(); 8989 } 8990} 8991 8992status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8993 uint32_t cmdSize, 8994 void *pCmdData, 8995 uint32_t *replySize, 8996 void *pReplyData) 8997{ 8998// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8999// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9000 9001 // only get parameter command is permitted for applications not controlling the effect 9002 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9003 return INVALID_OPERATION; 9004 } 9005 if (mEffect == 0) return DEAD_OBJECT; 9006 if (mClient == 0) return INVALID_OPERATION; 9007 9008 // handle commands that are not forwarded transparently to effect engine 9009 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9010 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 9011 // no risk to block the whole media server process or mixer threads is we are stuck here 9012 Mutex::Autolock _l(mCblk->lock); 9013 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9014 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9015 mCblk->serverIndex = 0; 9016 mCblk->clientIndex = 0; 9017 return BAD_VALUE; 9018 } 9019 status_t status = NO_ERROR; 9020 while (mCblk->serverIndex < mCblk->clientIndex) { 9021 int reply; 9022 uint32_t rsize = sizeof(int); 9023 int *p = (int *)(mBuffer + mCblk->serverIndex); 9024 int size = *p++; 9025 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9026 ALOGW("command(): invalid parameter block size"); 9027 break; 9028 } 9029 effect_param_t *param = (effect_param_t *)p; 9030 if (param->psize == 0 || param->vsize == 0) { 9031 ALOGW("command(): null parameter or value size"); 9032 mCblk->serverIndex += size; 9033 continue; 9034 } 9035 uint32_t psize = sizeof(effect_param_t) + 9036 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9037 param->vsize; 9038 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9039 psize, 9040 p, 9041 &rsize, 9042 &reply); 9043 // stop at first error encountered 9044 if (ret != NO_ERROR) { 9045 status = ret; 9046 *(int *)pReplyData = reply; 9047 break; 9048 } else if (reply != NO_ERROR) { 9049 *(int *)pReplyData = reply; 9050 break; 9051 } 9052 mCblk->serverIndex += size; 9053 } 9054 mCblk->serverIndex = 0; 9055 mCblk->clientIndex = 0; 9056 return status; 9057 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9058 *(int *)pReplyData = NO_ERROR; 9059 return enable(); 9060 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9061 *(int *)pReplyData = NO_ERROR; 9062 return disable(); 9063 } 9064 9065 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9066} 9067 9068void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9069{ 9070 ALOGV("setControl %p control %d", this, hasControl); 9071 9072 mHasControl = hasControl; 9073 mEnabled = enabled; 9074 9075 if (signal && mEffectClient != 0) { 9076 mEffectClient->controlStatusChanged(hasControl); 9077 } 9078} 9079 9080void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9081 uint32_t cmdSize, 9082 void *pCmdData, 9083 uint32_t replySize, 9084 void *pReplyData) 9085{ 9086 if (mEffectClient != 0) { 9087 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9088 } 9089} 9090 9091 9092 9093void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9094{ 9095 if (mEffectClient != 0) { 9096 mEffectClient->enableStatusChanged(enabled); 9097 } 9098} 9099 9100status_t AudioFlinger::EffectHandle::onTransact( 9101 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9102{ 9103 return BnEffect::onTransact(code, data, reply, flags); 9104} 9105 9106 9107void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9108{ 9109 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9110 9111 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9112 (mClient == 0) ? getpid_cached : mClient->pid(), 9113 mPriority, 9114 mHasControl, 9115 !locked, 9116 mCblk ? mCblk->clientIndex : 0, 9117 mCblk ? mCblk->serverIndex : 0 9118 ); 9119 9120 if (locked) { 9121 mCblk->lock.unlock(); 9122 } 9123} 9124 9125#undef LOG_TAG 9126#define LOG_TAG "AudioFlinger::EffectChain" 9127 9128AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9129 int sessionId) 9130 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9131 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9132 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9133{ 9134 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9135 if (thread == NULL) { 9136 return; 9137 } 9138 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9139 thread->frameCount(); 9140} 9141 9142AudioFlinger::EffectChain::~EffectChain() 9143{ 9144 if (mOwnInBuffer) { 9145 delete mInBuffer; 9146 } 9147 9148} 9149 9150// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9151sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9152{ 9153 size_t size = mEffects.size(); 9154 9155 for (size_t i = 0; i < size; i++) { 9156 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9157 return mEffects[i]; 9158 } 9159 } 9160 return 0; 9161} 9162 9163// getEffectFromId_l() must be called with ThreadBase::mLock held 9164sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9165{ 9166 size_t size = mEffects.size(); 9167 9168 for (size_t i = 0; i < size; i++) { 9169 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9170 if (id == 0 || mEffects[i]->id() == id) { 9171 return mEffects[i]; 9172 } 9173 } 9174 return 0; 9175} 9176 9177// getEffectFromType_l() must be called with ThreadBase::mLock held 9178sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9179 const effect_uuid_t *type) 9180{ 9181 size_t size = mEffects.size(); 9182 9183 for (size_t i = 0; i < size; i++) { 9184 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9185 return mEffects[i]; 9186 } 9187 } 9188 return 0; 9189} 9190 9191void AudioFlinger::EffectChain::clearInputBuffer() 9192{ 9193 Mutex::Autolock _l(mLock); 9194 sp<ThreadBase> thread = mThread.promote(); 9195 if (thread == 0) { 9196 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9197 return; 9198 } 9199 clearInputBuffer_l(thread); 9200} 9201 9202// Must be called with EffectChain::mLock locked 9203void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9204{ 9205 size_t numSamples = thread->frameCount() * thread->channelCount(); 9206 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9207 9208} 9209 9210// Must be called with EffectChain::mLock locked 9211void AudioFlinger::EffectChain::process_l() 9212{ 9213 sp<ThreadBase> thread = mThread.promote(); 9214 if (thread == 0) { 9215 ALOGW("process_l(): cannot promote mixer thread"); 9216 return; 9217 } 9218 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9219 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9220 // always process effects unless no more tracks are on the session and the effect tail 9221 // has been rendered 9222 bool doProcess = true; 9223 if (!isGlobalSession) { 9224 bool tracksOnSession = (trackCnt() != 0); 9225 9226 if (!tracksOnSession && mTailBufferCount == 0) { 9227 doProcess = false; 9228 } 9229 9230 if (activeTrackCnt() == 0) { 9231 // if no track is active and the effect tail has not been rendered, 9232 // the input buffer must be cleared here as the mixer process will not do it 9233 if (tracksOnSession || mTailBufferCount > 0) { 9234 clearInputBuffer_l(thread); 9235 if (mTailBufferCount > 0) { 9236 mTailBufferCount--; 9237 } 9238 } 9239 } 9240 } 9241 9242 size_t size = mEffects.size(); 9243 if (doProcess) { 9244 for (size_t i = 0; i < size; i++) { 9245 mEffects[i]->process(); 9246 } 9247 } 9248 for (size_t i = 0; i < size; i++) { 9249 mEffects[i]->updateState(); 9250 } 9251} 9252 9253// addEffect_l() must be called with PlaybackThread::mLock held 9254status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9255{ 9256 effect_descriptor_t desc = effect->desc(); 9257 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9258 9259 Mutex::Autolock _l(mLock); 9260 effect->setChain(this); 9261 sp<ThreadBase> thread = mThread.promote(); 9262 if (thread == 0) { 9263 return NO_INIT; 9264 } 9265 effect->setThread(thread); 9266 9267 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9268 // Auxiliary effects are inserted at the beginning of mEffects vector as 9269 // they are processed first and accumulated in chain input buffer 9270 mEffects.insertAt(effect, 0); 9271 9272 // the input buffer for auxiliary effect contains mono samples in 9273 // 32 bit format. This is to avoid saturation in AudoMixer 9274 // accumulation stage. Saturation is done in EffectModule::process() before 9275 // calling the process in effect engine 9276 size_t numSamples = thread->frameCount(); 9277 int32_t *buffer = new int32_t[numSamples]; 9278 memset(buffer, 0, numSamples * sizeof(int32_t)); 9279 effect->setInBuffer((int16_t *)buffer); 9280 // auxiliary effects output samples to chain input buffer for further processing 9281 // by insert effects 9282 effect->setOutBuffer(mInBuffer); 9283 } else { 9284 // Insert effects are inserted at the end of mEffects vector as they are processed 9285 // after track and auxiliary effects. 9286 // Insert effect order as a function of indicated preference: 9287 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9288 // another effect is present 9289 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9290 // last effect claiming first position 9291 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9292 // first effect claiming last position 9293 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9294 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9295 // already present 9296 9297 size_t size = mEffects.size(); 9298 size_t idx_insert = size; 9299 ssize_t idx_insert_first = -1; 9300 ssize_t idx_insert_last = -1; 9301 9302 for (size_t i = 0; i < size; i++) { 9303 effect_descriptor_t d = mEffects[i]->desc(); 9304 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9305 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9306 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9307 // check invalid effect chaining combinations 9308 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9309 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9310 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9311 return INVALID_OPERATION; 9312 } 9313 // remember position of first insert effect and by default 9314 // select this as insert position for new effect 9315 if (idx_insert == size) { 9316 idx_insert = i; 9317 } 9318 // remember position of last insert effect claiming 9319 // first position 9320 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9321 idx_insert_first = i; 9322 } 9323 // remember position of first insert effect claiming 9324 // last position 9325 if (iPref == EFFECT_FLAG_INSERT_LAST && 9326 idx_insert_last == -1) { 9327 idx_insert_last = i; 9328 } 9329 } 9330 } 9331 9332 // modify idx_insert from first position if needed 9333 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9334 if (idx_insert_last != -1) { 9335 idx_insert = idx_insert_last; 9336 } else { 9337 idx_insert = size; 9338 } 9339 } else { 9340 if (idx_insert_first != -1) { 9341 idx_insert = idx_insert_first + 1; 9342 } 9343 } 9344 9345 // always read samples from chain input buffer 9346 effect->setInBuffer(mInBuffer); 9347 9348 // if last effect in the chain, output samples to chain 9349 // output buffer, otherwise to chain input buffer 9350 if (idx_insert == size) { 9351 if (idx_insert != 0) { 9352 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9353 mEffects[idx_insert-1]->configure(); 9354 } 9355 effect->setOutBuffer(mOutBuffer); 9356 } else { 9357 effect->setOutBuffer(mInBuffer); 9358 } 9359 mEffects.insertAt(effect, idx_insert); 9360 9361 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9362 } 9363 effect->configure(); 9364 return NO_ERROR; 9365} 9366 9367// removeEffect_l() must be called with PlaybackThread::mLock held 9368size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9369{ 9370 Mutex::Autolock _l(mLock); 9371 size_t size = mEffects.size(); 9372 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9373 9374 for (size_t i = 0; i < size; i++) { 9375 if (effect == mEffects[i]) { 9376 // calling stop here will remove pre-processing effect from the audio HAL. 9377 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9378 // the middle of a read from audio HAL 9379 if (mEffects[i]->state() == EffectModule::ACTIVE || 9380 mEffects[i]->state() == EffectModule::STOPPING) { 9381 mEffects[i]->stop(); 9382 } 9383 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9384 delete[] effect->inBuffer(); 9385 } else { 9386 if (i == size - 1 && i != 0) { 9387 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9388 mEffects[i - 1]->configure(); 9389 } 9390 } 9391 mEffects.removeAt(i); 9392 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9393 break; 9394 } 9395 } 9396 9397 return mEffects.size(); 9398} 9399 9400// setDevice_l() must be called with PlaybackThread::mLock held 9401void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9402{ 9403 size_t size = mEffects.size(); 9404 for (size_t i = 0; i < size; i++) { 9405 mEffects[i]->setDevice(device); 9406 } 9407} 9408 9409// setMode_l() must be called with PlaybackThread::mLock held 9410void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9411{ 9412 size_t size = mEffects.size(); 9413 for (size_t i = 0; i < size; i++) { 9414 mEffects[i]->setMode(mode); 9415 } 9416} 9417 9418// setVolume_l() must be called with PlaybackThread::mLock held 9419bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9420{ 9421 uint32_t newLeft = *left; 9422 uint32_t newRight = *right; 9423 bool hasControl = false; 9424 int ctrlIdx = -1; 9425 size_t size = mEffects.size(); 9426 9427 // first update volume controller 9428 for (size_t i = size; i > 0; i--) { 9429 if (mEffects[i - 1]->isProcessEnabled() && 9430 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9431 ctrlIdx = i - 1; 9432 hasControl = true; 9433 break; 9434 } 9435 } 9436 9437 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9438 if (hasControl) { 9439 *left = mNewLeftVolume; 9440 *right = mNewRightVolume; 9441 } 9442 return hasControl; 9443 } 9444 9445 mVolumeCtrlIdx = ctrlIdx; 9446 mLeftVolume = newLeft; 9447 mRightVolume = newRight; 9448 9449 // second get volume update from volume controller 9450 if (ctrlIdx >= 0) { 9451 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9452 mNewLeftVolume = newLeft; 9453 mNewRightVolume = newRight; 9454 } 9455 // then indicate volume to all other effects in chain. 9456 // Pass altered volume to effects before volume controller 9457 // and requested volume to effects after controller 9458 uint32_t lVol = newLeft; 9459 uint32_t rVol = newRight; 9460 9461 for (size_t i = 0; i < size; i++) { 9462 if ((int)i == ctrlIdx) continue; 9463 // this also works for ctrlIdx == -1 when there is no volume controller 9464 if ((int)i > ctrlIdx) { 9465 lVol = *left; 9466 rVol = *right; 9467 } 9468 mEffects[i]->setVolume(&lVol, &rVol, false); 9469 } 9470 *left = newLeft; 9471 *right = newRight; 9472 9473 return hasControl; 9474} 9475 9476void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9477{ 9478 const size_t SIZE = 256; 9479 char buffer[SIZE]; 9480 String8 result; 9481 9482 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9483 result.append(buffer); 9484 9485 bool locked = tryLock(mLock); 9486 // failed to lock - AudioFlinger is probably deadlocked 9487 if (!locked) { 9488 result.append("\tCould not lock mutex:\n"); 9489 } 9490 9491 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9492 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9493 mEffects.size(), 9494 (uint32_t)mInBuffer, 9495 (uint32_t)mOutBuffer, 9496 mActiveTrackCnt); 9497 result.append(buffer); 9498 write(fd, result.string(), result.size()); 9499 9500 for (size_t i = 0; i < mEffects.size(); ++i) { 9501 sp<EffectModule> effect = mEffects[i]; 9502 if (effect != 0) { 9503 effect->dump(fd, args); 9504 } 9505 } 9506 9507 if (locked) { 9508 mLock.unlock(); 9509 } 9510} 9511 9512// must be called with ThreadBase::mLock held 9513void AudioFlinger::EffectChain::setEffectSuspended_l( 9514 const effect_uuid_t *type, bool suspend) 9515{ 9516 sp<SuspendedEffectDesc> desc; 9517 // use effect type UUID timelow as key as there is no real risk of identical 9518 // timeLow fields among effect type UUIDs. 9519 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9520 if (suspend) { 9521 if (index >= 0) { 9522 desc = mSuspendedEffects.valueAt(index); 9523 } else { 9524 desc = new SuspendedEffectDesc(); 9525 desc->mType = *type; 9526 mSuspendedEffects.add(type->timeLow, desc); 9527 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9528 } 9529 if (desc->mRefCount++ == 0) { 9530 sp<EffectModule> effect = getEffectIfEnabled(type); 9531 if (effect != 0) { 9532 desc->mEffect = effect; 9533 effect->setSuspended(true); 9534 effect->setEnabled(false); 9535 } 9536 } 9537 } else { 9538 if (index < 0) { 9539 return; 9540 } 9541 desc = mSuspendedEffects.valueAt(index); 9542 if (desc->mRefCount <= 0) { 9543 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9544 desc->mRefCount = 1; 9545 } 9546 if (--desc->mRefCount == 0) { 9547 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9548 if (desc->mEffect != 0) { 9549 sp<EffectModule> effect = desc->mEffect.promote(); 9550 if (effect != 0) { 9551 effect->setSuspended(false); 9552 effect->lock(); 9553 EffectHandle *handle = effect->controlHandle_l(); 9554 if (handle != NULL && !handle->destroyed_l()) { 9555 effect->setEnabled_l(handle->enabled()); 9556 } 9557 effect->unlock(); 9558 } 9559 desc->mEffect.clear(); 9560 } 9561 mSuspendedEffects.removeItemsAt(index); 9562 } 9563 } 9564} 9565 9566// must be called with ThreadBase::mLock held 9567void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9568{ 9569 sp<SuspendedEffectDesc> desc; 9570 9571 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9572 if (suspend) { 9573 if (index >= 0) { 9574 desc = mSuspendedEffects.valueAt(index); 9575 } else { 9576 desc = new SuspendedEffectDesc(); 9577 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9578 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9579 } 9580 if (desc->mRefCount++ == 0) { 9581 Vector< sp<EffectModule> > effects; 9582 getSuspendEligibleEffects(effects); 9583 for (size_t i = 0; i < effects.size(); i++) { 9584 setEffectSuspended_l(&effects[i]->desc().type, true); 9585 } 9586 } 9587 } else { 9588 if (index < 0) { 9589 return; 9590 } 9591 desc = mSuspendedEffects.valueAt(index); 9592 if (desc->mRefCount <= 0) { 9593 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9594 desc->mRefCount = 1; 9595 } 9596 if (--desc->mRefCount == 0) { 9597 Vector<const effect_uuid_t *> types; 9598 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9599 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9600 continue; 9601 } 9602 types.add(&mSuspendedEffects.valueAt(i)->mType); 9603 } 9604 for (size_t i = 0; i < types.size(); i++) { 9605 setEffectSuspended_l(types[i], false); 9606 } 9607 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9608 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9609 } 9610 } 9611} 9612 9613 9614// The volume effect is used for automated tests only 9615#ifndef OPENSL_ES_H_ 9616static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9617 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9618const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9619#endif //OPENSL_ES_H_ 9620 9621bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9622{ 9623 // auxiliary effects and visualizer are never suspended on output mix 9624 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9625 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9626 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9627 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9628 return false; 9629 } 9630 return true; 9631} 9632 9633void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9634{ 9635 effects.clear(); 9636 for (size_t i = 0; i < mEffects.size(); i++) { 9637 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9638 effects.add(mEffects[i]); 9639 } 9640 } 9641} 9642 9643sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9644 const effect_uuid_t *type) 9645{ 9646 sp<EffectModule> effect = getEffectFromType_l(type); 9647 return effect != 0 && effect->isEnabled() ? effect : 0; 9648} 9649 9650void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9651 bool enabled) 9652{ 9653 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9654 if (enabled) { 9655 if (index < 0) { 9656 // if the effect is not suspend check if all effects are suspended 9657 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9658 if (index < 0) { 9659 return; 9660 } 9661 if (!isEffectEligibleForSuspend(effect->desc())) { 9662 return; 9663 } 9664 setEffectSuspended_l(&effect->desc().type, enabled); 9665 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9666 if (index < 0) { 9667 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9668 return; 9669 } 9670 } 9671 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9672 effect->desc().type.timeLow); 9673 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9674 // if effect is requested to suspended but was not yet enabled, supend it now. 9675 if (desc->mEffect == 0) { 9676 desc->mEffect = effect; 9677 effect->setEnabled(false); 9678 effect->setSuspended(true); 9679 } 9680 } else { 9681 if (index < 0) { 9682 return; 9683 } 9684 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9685 effect->desc().type.timeLow); 9686 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9687 desc->mEffect.clear(); 9688 effect->setSuspended(false); 9689 } 9690} 9691 9692#undef LOG_TAG 9693#define LOG_TAG "AudioFlinger" 9694 9695// ---------------------------------------------------------------------------- 9696 9697status_t AudioFlinger::onTransact( 9698 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9699{ 9700 return BnAudioFlinger::onTransact(code, data, reply, flags); 9701} 9702 9703}; // namespace android 9704