AudioFlinger.cpp revision 254af180475346b6186b49c297f340c9c4817511
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// ---------------------------------------------------------------------------- 165 166#ifdef ADD_BATTERY_DATA 167// To collect the amplifier usage 168static void addBatteryData(uint32_t params) { 169 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 170 if (service == NULL) { 171 // it already logged 172 return; 173 } 174 175 service->addBatteryData(params); 176} 177#endif 178 179static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 180{ 181 const hw_module_t *mod; 182 int rc; 183 184 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 185 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 186 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 187 if (rc) { 188 goto out; 189 } 190 rc = audio_hw_device_open(mod, dev); 191 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 197 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 198 rc = BAD_VALUE; 199 goto out; 200 } 201 return 0; 202 203out: 204 *dev = NULL; 205 return rc; 206} 207 208// ---------------------------------------------------------------------------- 209 210AudioFlinger::AudioFlinger() 211 : BnAudioFlinger(), 212 mPrimaryHardwareDev(NULL), 213 mHardwareStatus(AUDIO_HW_IDLE), 214 mMasterVolume(1.0f), 215 mMasterVolumeSW(1.0f), 216 mMasterVolumeSupportLvl(MVS_NONE), 217 mMasterMute(false), 218 mNextUniqueId(1), 219 mMode(AUDIO_MODE_INVALID), 220 mBtNrecIsOff(false) 221{ 222} 223 224void AudioFlinger::onFirstRef() 225{ 226 int rc = 0; 227 228 Mutex::Autolock _l(mLock); 229 230 /* TODO: move all this work into an Init() function */ 231 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 232 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 233 uint32_t int_val; 234 if (1 == sscanf(val_str, "%u", &int_val)) { 235 mStandbyTimeInNsecs = milliseconds(int_val); 236 ALOGI("Using %u mSec as standby time.", int_val); 237 } else { 238 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 239 ALOGI("Using default %u mSec as standby time.", 240 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 241 } 242 } 243 244 mMode = AUDIO_MODE_NORMAL; 245} 246 247AudioFlinger::~AudioFlinger() 248{ 249 250 while (!mRecordThreads.isEmpty()) { 251 // closeInput() will remove first entry from mRecordThreads 252 closeInput(mRecordThreads.keyAt(0)); 253 } 254 while (!mPlaybackThreads.isEmpty()) { 255 // closeOutput() will remove first entry from mPlaybackThreads 256 closeOutput(mPlaybackThreads.keyAt(0)); 257 } 258 259 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 260 // no mHardwareLock needed, as there are no other references to this 261 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 262 delete mAudioHwDevs.valueAt(i); 263 } 264} 265 266static const char * const audio_interfaces[] = { 267 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 268 AUDIO_HARDWARE_MODULE_ID_A2DP, 269 AUDIO_HARDWARE_MODULE_ID_USB, 270}; 271#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 272 273audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 274{ 275 // if module is 0, the request comes from an old policy manager and we should load 276 // well known modules 277 if (module == 0) { 278 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 279 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 280 loadHwModule_l(audio_interfaces[i]); 281 } 282 } else { 283 // check a match for the requested module handle 284 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 285 if (audioHwdevice != NULL) { 286 return audioHwdevice->hwDevice(); 287 } 288 } 289 // then try to find a module supporting the requested device. 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 292 if ((dev->get_supported_devices(dev) & devices) == devices) 293 return dev; 294 } 295 296 return NULL; 297} 298 299status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 305 result.append("Clients:\n"); 306 for (size_t i = 0; i < mClients.size(); ++i) { 307 sp<Client> client = mClients.valueAt(i).promote(); 308 if (client != 0) { 309 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 310 result.append(buffer); 311 } 312 } 313 314 result.append("Global session refs:\n"); 315 result.append(" session pid count\n"); 316 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 317 AudioSessionRef *r = mAudioSessionRefs[i]; 318 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 319 result.append(buffer); 320 } 321 write(fd, result.string(), result.size()); 322 return NO_ERROR; 323} 324 325 326status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 327{ 328 const size_t SIZE = 256; 329 char buffer[SIZE]; 330 String8 result; 331 hardware_call_state hardwareStatus = mHardwareStatus; 332 333 snprintf(buffer, SIZE, "Hardware status: %d\n" 334 "Standby Time mSec: %u\n", 335 hardwareStatus, 336 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 337 result.append(buffer); 338 write(fd, result.string(), result.size()); 339 return NO_ERROR; 340} 341 342status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353 return NO_ERROR; 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 audio_channel_mask_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 // check if an effect chain with the same session ID is present on another 473 // output thread and move it here. 474 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 475 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 476 if (mPlaybackThreads.keyAt(i) != output) { 477 uint32_t sessions = t->hasAudioSession(*sessionId); 478 if (sessions & PlaybackThread::EFFECT_SESSION) { 479 effectThread = t.get(); 480 break; 481 } 482 } 483 } 484 lSessionId = *sessionId; 485 } else { 486 // if no audio session id is provided, create one here 487 lSessionId = nextUniqueId(); 488 if (sessionId != NULL) { 489 *sessionId = lSessionId; 490 } 491 } 492 ALOGV("createTrack() lSessionId: %d", lSessionId); 493 494 track = thread->createTrack_l(client, streamType, sampleRate, format, 495 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 496 497 // move effect chain to this output thread if an effect on same session was waiting 498 // for a track to be created 499 if (lStatus == NO_ERROR && effectThread != NULL) { 500 Mutex::Autolock _dl(thread->mLock); 501 Mutex::Autolock _sl(effectThread->mLock); 502 moveEffectChain_l(lSessionId, effectThread, thread, true); 503 } 504 505 // Look for sync events awaiting for a session to be used. 506 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 507 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 508 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 509 if (lStatus == NO_ERROR) { 510 track->setSyncEvent(mPendingSyncEvents[i]); 511 } else { 512 mPendingSyncEvents[i]->cancel(); 513 } 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 RecordThread::RecordTrack *track = thread->track(); 872 if (track != NULL) { 873 audio_devices_t device = (audio_devices_t)( 874 thread->device() & AUDIO_DEVICE_IN_ALL); 875 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 876 thread->setEffectSuspended(FX_IID_AEC, 877 suspend, 878 track->sessionId()); 879 thread->setEffectSuspended(FX_IID_NS, 880 suspend, 881 track->sessionId()); 882 } 883 } 884 mBtNrecIsOff = btNrecIsOff; 885 } 886 } 887 String8 screenState; 888 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 889 bool isOff = screenState == "off"; 890 if (isOff != (gScreenState & 1)) { 891 gScreenState = ((gScreenState & ~1) + 2) | isOff; 892 } 893 } 894 return final_result; 895 } 896 897 // hold a strong ref on thread in case closeOutput() or closeInput() is called 898 // and the thread is exited once the lock is released 899 sp<ThreadBase> thread; 900 { 901 Mutex::Autolock _l(mLock); 902 thread = checkPlaybackThread_l(ioHandle); 903 if (thread == 0) { 904 thread = checkRecordThread_l(ioHandle); 905 } else if (thread == primaryPlaybackThread_l()) { 906 // indicate output device change to all input threads for pre processing 907 AudioParameter param = AudioParameter(keyValuePairs); 908 int value; 909 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 910 (value != 0)) { 911 for (size_t i = 0; i < mRecordThreads.size(); i++) { 912 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 913 } 914 } 915 } 916 } 917 if (thread != 0) { 918 return thread->setParameters(keyValuePairs); 919 } 920 return BAD_VALUE; 921} 922 923String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 924{ 925// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 926// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 927 928 Mutex::Autolock _l(mLock); 929 930 if (ioHandle == 0) { 931 String8 out_s8; 932 933 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 934 char *s; 935 { 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 938 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 939 s = dev->get_parameters(dev, keys.string()); 940 mHardwareStatus = AUDIO_HW_IDLE; 941 } 942 out_s8 += String8(s ? s : ""); 943 free(s); 944 } 945 return out_s8; 946 } 947 948 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 949 if (playbackThread != NULL) { 950 return playbackThread->getParameters(keys); 951 } 952 RecordThread *recordThread = checkRecordThread_l(ioHandle); 953 if (recordThread != NULL) { 954 return recordThread->getParameters(keys); 955 } 956 return String8(""); 957} 958 959size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 960 audio_channel_mask_t channelMask) const 961{ 962 status_t ret = initCheck(); 963 if (ret != NO_ERROR) { 964 return 0; 965 } 966 967 AutoMutex lock(mHardwareLock); 968 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 969 struct audio_config config = { 970 sample_rate: sampleRate, 971 channel_mask: channelMask, 972 format: format, 973 }; 974 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 975 mHardwareStatus = AUDIO_HW_IDLE; 976 return size; 977} 978 979unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 980{ 981 if (ioHandle == 0) { 982 return 0; 983 } 984 985 Mutex::Autolock _l(mLock); 986 987 RecordThread *recordThread = checkRecordThread_l(ioHandle); 988 if (recordThread != NULL) { 989 return recordThread->getInputFramesLost(); 990 } 991 return 0; 992} 993 994status_t AudioFlinger::setVoiceVolume(float value) 995{ 996 status_t ret = initCheck(); 997 if (ret != NO_ERROR) { 998 return ret; 999 } 1000 1001 // check calling permissions 1002 if (!settingsAllowed()) { 1003 return PERMISSION_DENIED; 1004 } 1005 1006 AutoMutex lock(mHardwareLock); 1007 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1008 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1009 mHardwareStatus = AUDIO_HW_IDLE; 1010 1011 return ret; 1012} 1013 1014status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1015 audio_io_handle_t output) const 1016{ 1017 status_t status; 1018 1019 Mutex::Autolock _l(mLock); 1020 1021 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1022 if (playbackThread != NULL) { 1023 return playbackThread->getRenderPosition(halFrames, dspFrames); 1024 } 1025 1026 return BAD_VALUE; 1027} 1028 1029void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1030{ 1031 1032 Mutex::Autolock _l(mLock); 1033 1034 pid_t pid = IPCThreadState::self()->getCallingPid(); 1035 if (mNotificationClients.indexOfKey(pid) < 0) { 1036 sp<NotificationClient> notificationClient = new NotificationClient(this, 1037 client, 1038 pid); 1039 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1040 1041 mNotificationClients.add(pid, notificationClient); 1042 1043 sp<IBinder> binder = client->asBinder(); 1044 binder->linkToDeath(notificationClient); 1045 1046 // the config change is always sent from playback or record threads to avoid deadlock 1047 // with AudioSystem::gLock 1048 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1049 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1050 } 1051 1052 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1053 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1054 } 1055 } 1056} 1057 1058void AudioFlinger::removeNotificationClient(pid_t pid) 1059{ 1060 Mutex::Autolock _l(mLock); 1061 1062 mNotificationClients.removeItem(pid); 1063 1064 ALOGV("%d died, releasing its sessions", pid); 1065 size_t num = mAudioSessionRefs.size(); 1066 bool removed = false; 1067 for (size_t i = 0; i< num; ) { 1068 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1069 ALOGV(" pid %d @ %d", ref->mPid, i); 1070 if (ref->mPid == pid) { 1071 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1072 mAudioSessionRefs.removeAt(i); 1073 delete ref; 1074 removed = true; 1075 num--; 1076 } else { 1077 i++; 1078 } 1079 } 1080 if (removed) { 1081 purgeStaleEffects_l(); 1082 } 1083} 1084 1085// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1086void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1087{ 1088 size_t size = mNotificationClients.size(); 1089 for (size_t i = 0; i < size; i++) { 1090 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1091 param2); 1092 } 1093} 1094 1095// removeClient_l() must be called with AudioFlinger::mLock held 1096void AudioFlinger::removeClient_l(pid_t pid) 1097{ 1098 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1099 mClients.removeItem(pid); 1100} 1101 1102// getEffectThread_l() must be called with AudioFlinger::mLock held 1103sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1104{ 1105 sp<PlaybackThread> thread; 1106 1107 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1108 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1109 ALOG_ASSERT(thread == 0); 1110 thread = mPlaybackThreads.valueAt(i); 1111 } 1112 } 1113 1114 return thread; 1115} 1116 1117// ---------------------------------------------------------------------------- 1118 1119AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1120 uint32_t device, type_t type) 1121 : Thread(false), 1122 mType(type), 1123 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1124 // mChannelMask 1125 mChannelCount(0), 1126 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1127 mParamStatus(NO_ERROR), 1128 mStandby(false), mDevice((audio_devices_t) device), mId(id), 1129 mDeathRecipient(new PMDeathRecipient(this)) 1130{ 1131} 1132 1133AudioFlinger::ThreadBase::~ThreadBase() 1134{ 1135 mParamCond.broadcast(); 1136 // do not lock the mutex in destructor 1137 releaseWakeLock_l(); 1138 if (mPowerManager != 0) { 1139 sp<IBinder> binder = mPowerManager->asBinder(); 1140 binder->unlinkToDeath(mDeathRecipient); 1141 } 1142} 1143 1144void AudioFlinger::ThreadBase::exit() 1145{ 1146 ALOGV("ThreadBase::exit"); 1147 { 1148 // This lock prevents the following race in thread (uniprocessor for illustration): 1149 // if (!exitPending()) { 1150 // // context switch from here to exit() 1151 // // exit() calls requestExit(), what exitPending() observes 1152 // // exit() calls signal(), which is dropped since no waiters 1153 // // context switch back from exit() to here 1154 // mWaitWorkCV.wait(...); 1155 // // now thread is hung 1156 // } 1157 AutoMutex lock(mLock); 1158 requestExit(); 1159 mWaitWorkCV.signal(); 1160 } 1161 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1162 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1163 requestExitAndWait(); 1164} 1165 1166status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1167{ 1168 status_t status; 1169 1170 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1171 Mutex::Autolock _l(mLock); 1172 1173 mNewParameters.add(keyValuePairs); 1174 mWaitWorkCV.signal(); 1175 // wait condition with timeout in case the thread loop has exited 1176 // before the request could be processed 1177 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1178 status = mParamStatus; 1179 mWaitWorkCV.signal(); 1180 } else { 1181 status = TIMED_OUT; 1182 } 1183 return status; 1184} 1185 1186void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1187{ 1188 Mutex::Autolock _l(mLock); 1189 sendConfigEvent_l(event, param); 1190} 1191 1192// sendConfigEvent_l() must be called with ThreadBase::mLock held 1193void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1194{ 1195 ConfigEvent configEvent; 1196 configEvent.mEvent = event; 1197 configEvent.mParam = param; 1198 mConfigEvents.add(configEvent); 1199 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1200 mWaitWorkCV.signal(); 1201} 1202 1203void AudioFlinger::ThreadBase::processConfigEvents() 1204{ 1205 mLock.lock(); 1206 while (!mConfigEvents.isEmpty()) { 1207 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1208 ConfigEvent configEvent = mConfigEvents[0]; 1209 mConfigEvents.removeAt(0); 1210 // release mLock before locking AudioFlinger mLock: lock order is always 1211 // AudioFlinger then ThreadBase to avoid cross deadlock 1212 mLock.unlock(); 1213 mAudioFlinger->mLock.lock(); 1214 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1215 mAudioFlinger->mLock.unlock(); 1216 mLock.lock(); 1217 } 1218 mLock.unlock(); 1219} 1220 1221status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1222{ 1223 const size_t SIZE = 256; 1224 char buffer[SIZE]; 1225 String8 result; 1226 1227 bool locked = tryLock(mLock); 1228 if (!locked) { 1229 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1230 write(fd, buffer, strlen(buffer)); 1231 } 1232 1233 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1246 result.append(buffer); 1247 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1250 result.append(buffer); 1251 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1252 result.append(buffer); 1253 1254 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1255 result.append(buffer); 1256 result.append(" Index Command"); 1257 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1258 snprintf(buffer, SIZE, "\n %02d ", i); 1259 result.append(buffer); 1260 result.append(mNewParameters[i]); 1261 } 1262 1263 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1264 result.append(buffer); 1265 snprintf(buffer, SIZE, " Index event param\n"); 1266 result.append(buffer); 1267 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1268 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1269 result.append(buffer); 1270 } 1271 result.append("\n"); 1272 1273 write(fd, result.string(), result.size()); 1274 1275 if (locked) { 1276 mLock.unlock(); 1277 } 1278 return NO_ERROR; 1279} 1280 1281status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1282{ 1283 const size_t SIZE = 256; 1284 char buffer[SIZE]; 1285 String8 result; 1286 1287 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1288 write(fd, buffer, strlen(buffer)); 1289 1290 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1291 sp<EffectChain> chain = mEffectChains[i]; 1292 if (chain != 0) { 1293 chain->dump(fd, args); 1294 } 1295 } 1296 return NO_ERROR; 1297} 1298 1299void AudioFlinger::ThreadBase::acquireWakeLock() 1300{ 1301 Mutex::Autolock _l(mLock); 1302 acquireWakeLock_l(); 1303} 1304 1305void AudioFlinger::ThreadBase::acquireWakeLock_l() 1306{ 1307 if (mPowerManager == 0) { 1308 // use checkService() to avoid blocking if power service is not up yet 1309 sp<IBinder> binder = 1310 defaultServiceManager()->checkService(String16("power")); 1311 if (binder == 0) { 1312 ALOGW("Thread %s cannot connect to the power manager service", mName); 1313 } else { 1314 mPowerManager = interface_cast<IPowerManager>(binder); 1315 binder->linkToDeath(mDeathRecipient); 1316 } 1317 } 1318 if (mPowerManager != 0) { 1319 sp<IBinder> binder = new BBinder(); 1320 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1321 binder, 1322 String16(mName)); 1323 if (status == NO_ERROR) { 1324 mWakeLockToken = binder; 1325 } 1326 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1327 } 1328} 1329 1330void AudioFlinger::ThreadBase::releaseWakeLock() 1331{ 1332 Mutex::Autolock _l(mLock); 1333 releaseWakeLock_l(); 1334} 1335 1336void AudioFlinger::ThreadBase::releaseWakeLock_l() 1337{ 1338 if (mWakeLockToken != 0) { 1339 ALOGV("releaseWakeLock_l() %s", mName); 1340 if (mPowerManager != 0) { 1341 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1342 } 1343 mWakeLockToken.clear(); 1344 } 1345} 1346 1347void AudioFlinger::ThreadBase::clearPowerManager() 1348{ 1349 Mutex::Autolock _l(mLock); 1350 releaseWakeLock_l(); 1351 mPowerManager.clear(); 1352} 1353 1354void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1355{ 1356 sp<ThreadBase> thread = mThread.promote(); 1357 if (thread != 0) { 1358 thread->clearPowerManager(); 1359 } 1360 ALOGW("power manager service died !!!"); 1361} 1362 1363void AudioFlinger::ThreadBase::setEffectSuspended( 1364 const effect_uuid_t *type, bool suspend, int sessionId) 1365{ 1366 Mutex::Autolock _l(mLock); 1367 setEffectSuspended_l(type, suspend, sessionId); 1368} 1369 1370void AudioFlinger::ThreadBase::setEffectSuspended_l( 1371 const effect_uuid_t *type, bool suspend, int sessionId) 1372{ 1373 sp<EffectChain> chain = getEffectChain_l(sessionId); 1374 if (chain != 0) { 1375 if (type != NULL) { 1376 chain->setEffectSuspended_l(type, suspend); 1377 } else { 1378 chain->setEffectSuspendedAll_l(suspend); 1379 } 1380 } 1381 1382 updateSuspendedSessions_l(type, suspend, sessionId); 1383} 1384 1385void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1386{ 1387 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1388 if (index < 0) { 1389 return; 1390 } 1391 1392 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1393 mSuspendedSessions.editValueAt(index); 1394 1395 for (size_t i = 0; i < sessionEffects.size(); i++) { 1396 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1397 for (int j = 0; j < desc->mRefCount; j++) { 1398 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1399 chain->setEffectSuspendedAll_l(true); 1400 } else { 1401 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1402 desc->mType.timeLow); 1403 chain->setEffectSuspended_l(&desc->mType, true); 1404 } 1405 } 1406 } 1407} 1408 1409void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1410 bool suspend, 1411 int sessionId) 1412{ 1413 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1414 1415 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1416 1417 if (suspend) { 1418 if (index >= 0) { 1419 sessionEffects = mSuspendedSessions.editValueAt(index); 1420 } else { 1421 mSuspendedSessions.add(sessionId, sessionEffects); 1422 } 1423 } else { 1424 if (index < 0) { 1425 return; 1426 } 1427 sessionEffects = mSuspendedSessions.editValueAt(index); 1428 } 1429 1430 1431 int key = EffectChain::kKeyForSuspendAll; 1432 if (type != NULL) { 1433 key = type->timeLow; 1434 } 1435 index = sessionEffects.indexOfKey(key); 1436 1437 sp<SuspendedSessionDesc> desc; 1438 if (suspend) { 1439 if (index >= 0) { 1440 desc = sessionEffects.valueAt(index); 1441 } else { 1442 desc = new SuspendedSessionDesc(); 1443 if (type != NULL) { 1444 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1445 } 1446 sessionEffects.add(key, desc); 1447 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1448 } 1449 desc->mRefCount++; 1450 } else { 1451 if (index < 0) { 1452 return; 1453 } 1454 desc = sessionEffects.valueAt(index); 1455 if (--desc->mRefCount == 0) { 1456 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1457 sessionEffects.removeItemsAt(index); 1458 if (sessionEffects.isEmpty()) { 1459 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1460 sessionId); 1461 mSuspendedSessions.removeItem(sessionId); 1462 } 1463 } 1464 } 1465 if (!sessionEffects.isEmpty()) { 1466 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1467 } 1468} 1469 1470void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1471 bool enabled, 1472 int sessionId) 1473{ 1474 Mutex::Autolock _l(mLock); 1475 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1476} 1477 1478void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1479 bool enabled, 1480 int sessionId) 1481{ 1482 if (mType != RECORD) { 1483 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1484 // another session. This gives the priority to well behaved effect control panels 1485 // and applications not using global effects. 1486 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1487 // global effects 1488 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1489 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1490 } 1491 } 1492 1493 sp<EffectChain> chain = getEffectChain_l(sessionId); 1494 if (chain != 0) { 1495 chain->checkSuspendOnEffectEnabled(effect, enabled); 1496 } 1497} 1498 1499// ---------------------------------------------------------------------------- 1500 1501AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1502 AudioStreamOut* output, 1503 audio_io_handle_t id, 1504 uint32_t device, 1505 type_t type) 1506 : ThreadBase(audioFlinger, id, device, type), 1507 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1508 // Assumes constructor is called by AudioFlinger with it's mLock held, 1509 // but it would be safer to explicitly pass initial masterMute as parameter 1510 mMasterMute(audioFlinger->masterMute_l()), 1511 // mStreamTypes[] initialized in constructor body 1512 mOutput(output), 1513 // Assumes constructor is called by AudioFlinger with it's mLock held, 1514 // but it would be safer to explicitly pass initial masterVolume as parameter 1515 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1516 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1517 mMixerStatus(MIXER_IDLE), 1518 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1519 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1520 mScreenState(gScreenState), 1521 // index 0 is reserved for normal mixer's submix 1522 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1523{ 1524 snprintf(mName, kNameLength, "AudioOut_%X", id); 1525 1526 readOutputParameters(); 1527 1528 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1529 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1530 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1531 stream = (audio_stream_type_t) (stream + 1)) { 1532 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1533 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1534 } 1535 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1536 // because mAudioFlinger doesn't have one to copy from 1537} 1538 1539AudioFlinger::PlaybackThread::~PlaybackThread() 1540{ 1541 delete [] mMixBuffer; 1542} 1543 1544status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1545{ 1546 dumpInternals(fd, args); 1547 dumpTracks(fd, args); 1548 dumpEffectChains(fd, args); 1549 return NO_ERROR; 1550} 1551 1552status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1553{ 1554 const size_t SIZE = 256; 1555 char buffer[SIZE]; 1556 String8 result; 1557 1558 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1559 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1560 const stream_type_t *st = &mStreamTypes[i]; 1561 if (i > 0) { 1562 result.appendFormat(", "); 1563 } 1564 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1565 if (st->mute) { 1566 result.append("M"); 1567 } 1568 } 1569 result.append("\n"); 1570 write(fd, result.string(), result.length()); 1571 result.clear(); 1572 1573 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1574 result.append(buffer); 1575 Track::appendDumpHeader(result); 1576 for (size_t i = 0; i < mTracks.size(); ++i) { 1577 sp<Track> track = mTracks[i]; 1578 if (track != 0) { 1579 track->dump(buffer, SIZE); 1580 result.append(buffer); 1581 } 1582 } 1583 1584 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1585 result.append(buffer); 1586 Track::appendDumpHeader(result); 1587 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1588 sp<Track> track = mActiveTracks[i].promote(); 1589 if (track != 0) { 1590 track->dump(buffer, SIZE); 1591 result.append(buffer); 1592 } 1593 } 1594 write(fd, result.string(), result.size()); 1595 1596 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1597 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1598 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1599 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1600 1601 return NO_ERROR; 1602} 1603 1604status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1605{ 1606 const size_t SIZE = 256; 1607 char buffer[SIZE]; 1608 String8 result; 1609 1610 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1611 result.append(buffer); 1612 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1613 result.append(buffer); 1614 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1615 result.append(buffer); 1616 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1617 result.append(buffer); 1618 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1619 result.append(buffer); 1620 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1621 result.append(buffer); 1622 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1623 result.append(buffer); 1624 write(fd, result.string(), result.size()); 1625 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1626 1627 dumpBase(fd, args); 1628 1629 return NO_ERROR; 1630} 1631 1632// Thread virtuals 1633status_t AudioFlinger::PlaybackThread::readyToRun() 1634{ 1635 status_t status = initCheck(); 1636 if (status == NO_ERROR) { 1637 ALOGI("AudioFlinger's thread %p ready to run", this); 1638 } else { 1639 ALOGE("No working audio driver found."); 1640 } 1641 return status; 1642} 1643 1644void AudioFlinger::PlaybackThread::onFirstRef() 1645{ 1646 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1647} 1648 1649// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1650sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1651 const sp<AudioFlinger::Client>& client, 1652 audio_stream_type_t streamType, 1653 uint32_t sampleRate, 1654 audio_format_t format, 1655 audio_channel_mask_t channelMask, 1656 int frameCount, 1657 const sp<IMemory>& sharedBuffer, 1658 int sessionId, 1659 IAudioFlinger::track_flags_t flags, 1660 pid_t tid, 1661 status_t *status) 1662{ 1663 sp<Track> track; 1664 status_t lStatus; 1665 1666 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1667 1668 // client expresses a preference for FAST, but we get the final say 1669 if (flags & IAudioFlinger::TRACK_FAST) { 1670 if ( 1671 // not timed 1672 (!isTimed) && 1673 // either of these use cases: 1674 ( 1675 // use case 1: shared buffer with any frame count 1676 ( 1677 (sharedBuffer != 0) 1678 ) || 1679 // use case 2: callback handler and frame count is default or at least as large as HAL 1680 ( 1681 (tid != -1) && 1682 ((frameCount == 0) || 1683 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1684 ) 1685 ) && 1686 // PCM data 1687 audio_is_linear_pcm(format) && 1688 // mono or stereo 1689 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1690 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1691#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1692 // hardware sample rate 1693 (sampleRate == mSampleRate) && 1694#endif 1695 // normal mixer has an associated fast mixer 1696 hasFastMixer() && 1697 // there are sufficient fast track slots available 1698 (mFastTrackAvailMask != 0) 1699 // FIXME test that MixerThread for this fast track has a capable output HAL 1700 // FIXME add a permission test also? 1701 ) { 1702 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1703 if (frameCount == 0) { 1704 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1705 } 1706 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1707 frameCount, mFrameCount); 1708 } else { 1709 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1710 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1711 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1712 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1713 audio_is_linear_pcm(format), 1714 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1715 flags &= ~IAudioFlinger::TRACK_FAST; 1716 // For compatibility with AudioTrack calculation, buffer depth is forced 1717 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1718 // This is probably too conservative, but legacy application code may depend on it. 1719 // If you change this calculation, also review the start threshold which is related. 1720 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1721 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1722 if (minBufCount < 2) { 1723 minBufCount = 2; 1724 } 1725 int minFrameCount = mNormalFrameCount * minBufCount; 1726 if (frameCount < minFrameCount) { 1727 frameCount = minFrameCount; 1728 } 1729 } 1730 } 1731 1732 if (mType == DIRECT) { 1733 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1734 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1735 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1736 "for output %p with format %d", 1737 sampleRate, format, channelMask, mOutput, mFormat); 1738 lStatus = BAD_VALUE; 1739 goto Exit; 1740 } 1741 } 1742 } else { 1743 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1744 if (sampleRate > mSampleRate*2) { 1745 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1746 lStatus = BAD_VALUE; 1747 goto Exit; 1748 } 1749 } 1750 1751 lStatus = initCheck(); 1752 if (lStatus != NO_ERROR) { 1753 ALOGE("Audio driver not initialized."); 1754 goto Exit; 1755 } 1756 1757 { // scope for mLock 1758 Mutex::Autolock _l(mLock); 1759 1760 // all tracks in same audio session must share the same routing strategy otherwise 1761 // conflicts will happen when tracks are moved from one output to another by audio policy 1762 // manager 1763 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1764 for (size_t i = 0; i < mTracks.size(); ++i) { 1765 sp<Track> t = mTracks[i]; 1766 if (t != 0 && !t->isOutputTrack()) { 1767 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1768 if (sessionId == t->sessionId() && strategy != actual) { 1769 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1770 strategy, actual); 1771 lStatus = BAD_VALUE; 1772 goto Exit; 1773 } 1774 } 1775 } 1776 1777 if (!isTimed) { 1778 track = new Track(this, client, streamType, sampleRate, format, 1779 channelMask, frameCount, sharedBuffer, sessionId, flags); 1780 } else { 1781 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1782 channelMask, frameCount, sharedBuffer, sessionId); 1783 } 1784 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1785 lStatus = NO_MEMORY; 1786 goto Exit; 1787 } 1788 mTracks.add(track); 1789 1790 sp<EffectChain> chain = getEffectChain_l(sessionId); 1791 if (chain != 0) { 1792 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1793 track->setMainBuffer(chain->inBuffer()); 1794 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1795 chain->incTrackCnt(); 1796 } 1797 } 1798 1799 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1800 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1801 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1802 // so ask activity manager to do this on our behalf 1803 int err = requestPriority(callingPid, tid, 1); 1804 if (err != 0) { 1805 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1806 1, callingPid, tid, err); 1807 } 1808 } 1809 1810 lStatus = NO_ERROR; 1811 1812Exit: 1813 if (status) { 1814 *status = lStatus; 1815 } 1816 return track; 1817} 1818 1819uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1820{ 1821 if (mFastMixer != NULL) { 1822 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1823 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1824 } 1825 return latency; 1826} 1827 1828uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1829{ 1830 return latency; 1831} 1832 1833uint32_t AudioFlinger::PlaybackThread::latency() const 1834{ 1835 Mutex::Autolock _l(mLock); 1836 return latency_l(); 1837} 1838uint32_t AudioFlinger::PlaybackThread::latency_l() const 1839{ 1840 if (initCheck() == NO_ERROR) { 1841 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1842 } else { 1843 return 0; 1844 } 1845} 1846 1847void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1848{ 1849 Mutex::Autolock _l(mLock); 1850 mMasterVolume = value; 1851} 1852 1853void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1854{ 1855 Mutex::Autolock _l(mLock); 1856 setMasterMute_l(muted); 1857} 1858 1859void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1860{ 1861 Mutex::Autolock _l(mLock); 1862 mStreamTypes[stream].volume = value; 1863} 1864 1865void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1866{ 1867 Mutex::Autolock _l(mLock); 1868 mStreamTypes[stream].mute = muted; 1869} 1870 1871float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1872{ 1873 Mutex::Autolock _l(mLock); 1874 return mStreamTypes[stream].volume; 1875} 1876 1877// addTrack_l() must be called with ThreadBase::mLock held 1878status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1879{ 1880 status_t status = ALREADY_EXISTS; 1881 1882 // set retry count for buffer fill 1883 track->mRetryCount = kMaxTrackStartupRetries; 1884 if (mActiveTracks.indexOf(track) < 0) { 1885 // the track is newly added, make sure it fills up all its 1886 // buffers before playing. This is to ensure the client will 1887 // effectively get the latency it requested. 1888 track->mFillingUpStatus = Track::FS_FILLING; 1889 track->mResetDone = false; 1890 track->mPresentationCompleteFrames = 0; 1891 mActiveTracks.add(track); 1892 if (track->mainBuffer() != mMixBuffer) { 1893 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1894 if (chain != 0) { 1895 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1896 chain->incActiveTrackCnt(); 1897 } 1898 } 1899 1900 status = NO_ERROR; 1901 } 1902 1903 ALOGV("mWaitWorkCV.broadcast"); 1904 mWaitWorkCV.broadcast(); 1905 1906 return status; 1907} 1908 1909// destroyTrack_l() must be called with ThreadBase::mLock held 1910void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1911{ 1912 track->mState = TrackBase::TERMINATED; 1913 // active tracks are removed by threadLoop() 1914 if (mActiveTracks.indexOf(track) < 0) { 1915 removeTrack_l(track); 1916 } 1917} 1918 1919void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1920{ 1921 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1922 mTracks.remove(track); 1923 deleteTrackName_l(track->name()); 1924 // redundant as track is about to be destroyed, for dumpsys only 1925 track->mName = -1; 1926 if (track->isFastTrack()) { 1927 int index = track->mFastIndex; 1928 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1929 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1930 mFastTrackAvailMask |= 1 << index; 1931 // redundant as track is about to be destroyed, for dumpsys only 1932 track->mFastIndex = -1; 1933 } 1934 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1935 if (chain != 0) { 1936 chain->decTrackCnt(); 1937 } 1938} 1939 1940String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1941{ 1942 String8 out_s8 = String8(""); 1943 char *s; 1944 1945 Mutex::Autolock _l(mLock); 1946 if (initCheck() != NO_ERROR) { 1947 return out_s8; 1948 } 1949 1950 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1951 out_s8 = String8(s); 1952 free(s); 1953 return out_s8; 1954} 1955 1956// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1957void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1958 AudioSystem::OutputDescriptor desc; 1959 void *param2 = NULL; 1960 1961 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1962 1963 switch (event) { 1964 case AudioSystem::OUTPUT_OPENED: 1965 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1966 desc.channels = mChannelMask; 1967 desc.samplingRate = mSampleRate; 1968 desc.format = mFormat; 1969 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1970 desc.latency = latency(); 1971 param2 = &desc; 1972 break; 1973 1974 case AudioSystem::STREAM_CONFIG_CHANGED: 1975 param2 = ¶m; 1976 case AudioSystem::OUTPUT_CLOSED: 1977 default: 1978 break; 1979 } 1980 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1981} 1982 1983void AudioFlinger::PlaybackThread::readOutputParameters() 1984{ 1985 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1986 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1987 mChannelCount = (uint16_t)popcount(mChannelMask); 1988 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1989 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1990 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1991 if (mFrameCount & 15) { 1992 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1993 mFrameCount); 1994 } 1995 1996 // Calculate size of normal mix buffer relative to the HAL output buffer size 1997 double multiplier = 1.0; 1998 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1999 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2000 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2001 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2002 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2003 maxNormalFrameCount = maxNormalFrameCount & ~15; 2004 if (maxNormalFrameCount < minNormalFrameCount) { 2005 maxNormalFrameCount = minNormalFrameCount; 2006 } 2007 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2008 if (multiplier <= 1.0) { 2009 multiplier = 1.0; 2010 } else if (multiplier <= 2.0) { 2011 if (2 * mFrameCount <= maxNormalFrameCount) { 2012 multiplier = 2.0; 2013 } else { 2014 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2015 } 2016 } else { 2017 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2018 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2019 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2020 // FIXME this rounding up should not be done if no HAL SRC 2021 uint32_t truncMult = (uint32_t) multiplier; 2022 if ((truncMult & 1)) { 2023 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2024 ++truncMult; 2025 } 2026 } 2027 multiplier = (double) truncMult; 2028 } 2029 } 2030 mNormalFrameCount = multiplier * mFrameCount; 2031 // round up to nearest 16 frames to satisfy AudioMixer 2032 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2033 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2034 2035 delete[] mMixBuffer; 2036 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2037 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2038 2039 // force reconfiguration of effect chains and engines to take new buffer size and audio 2040 // parameters into account 2041 // Note that mLock is not held when readOutputParameters() is called from the constructor 2042 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2043 // matter. 2044 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2045 Vector< sp<EffectChain> > effectChains = mEffectChains; 2046 for (size_t i = 0; i < effectChains.size(); i ++) { 2047 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2048 } 2049} 2050 2051 2052status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2053{ 2054 if (halFrames == NULL || dspFrames == NULL) { 2055 return BAD_VALUE; 2056 } 2057 Mutex::Autolock _l(mLock); 2058 if (initCheck() != NO_ERROR) { 2059 return INVALID_OPERATION; 2060 } 2061 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2062 2063 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2064} 2065 2066uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2067{ 2068 Mutex::Autolock _l(mLock); 2069 uint32_t result = 0; 2070 if (getEffectChain_l(sessionId) != 0) { 2071 result = EFFECT_SESSION; 2072 } 2073 2074 for (size_t i = 0; i < mTracks.size(); ++i) { 2075 sp<Track> track = mTracks[i]; 2076 if (sessionId == track->sessionId() && 2077 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2078 result |= TRACK_SESSION; 2079 break; 2080 } 2081 } 2082 2083 return result; 2084} 2085 2086uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2087{ 2088 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2089 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2090 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2091 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2092 } 2093 for (size_t i = 0; i < mTracks.size(); i++) { 2094 sp<Track> track = mTracks[i]; 2095 if (sessionId == track->sessionId() && 2096 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2097 return AudioSystem::getStrategyForStream(track->streamType()); 2098 } 2099 } 2100 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2101} 2102 2103 2104AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2105{ 2106 Mutex::Autolock _l(mLock); 2107 return mOutput; 2108} 2109 2110AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2111{ 2112 Mutex::Autolock _l(mLock); 2113 AudioStreamOut *output = mOutput; 2114 mOutput = NULL; 2115 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2116 // must push a NULL and wait for ack 2117 mOutputSink.clear(); 2118 mPipeSink.clear(); 2119 mNormalSink.clear(); 2120 return output; 2121} 2122 2123// this method must always be called either with ThreadBase mLock held or inside the thread loop 2124audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2125{ 2126 if (mOutput == NULL) { 2127 return NULL; 2128 } 2129 return &mOutput->stream->common; 2130} 2131 2132uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2133{ 2134 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2135} 2136 2137status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2138{ 2139 if (!isValidSyncEvent(event)) { 2140 return BAD_VALUE; 2141 } 2142 2143 Mutex::Autolock _l(mLock); 2144 2145 for (size_t i = 0; i < mTracks.size(); ++i) { 2146 sp<Track> track = mTracks[i]; 2147 if (event->triggerSession() == track->sessionId()) { 2148 track->setSyncEvent(event); 2149 return NO_ERROR; 2150 } 2151 } 2152 2153 return NAME_NOT_FOUND; 2154} 2155 2156bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2157{ 2158 switch (event->type()) { 2159 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2160 return true; 2161 default: 2162 break; 2163 } 2164 return false; 2165} 2166 2167void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2168{ 2169 size_t count = tracksToRemove.size(); 2170 if (CC_UNLIKELY(count)) { 2171 for (size_t i = 0 ; i < count ; i++) { 2172 const sp<Track>& track = tracksToRemove.itemAt(i); 2173 if ((track->sharedBuffer() != 0) && 2174 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2175 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2176 } 2177 } 2178 } 2179 2180} 2181 2182// ---------------------------------------------------------------------------- 2183 2184AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2185 audio_io_handle_t id, uint32_t device, type_t type) 2186 : PlaybackThread(audioFlinger, output, id, device, type), 2187 // mAudioMixer below 2188 // mFastMixer below 2189 mFastMixerFutex(0) 2190 // mOutputSink below 2191 // mPipeSink below 2192 // mNormalSink below 2193{ 2194 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2195 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2196 "mFrameCount=%d, mNormalFrameCount=%d", 2197 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2198 mNormalFrameCount); 2199 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2200 2201 // FIXME - Current mixer implementation only supports stereo output 2202 if (mChannelCount == 1) { 2203 ALOGE("Invalid audio hardware channel count"); 2204 } 2205 2206 // create an NBAIO sink for the HAL output stream, and negotiate 2207 mOutputSink = new AudioStreamOutSink(output->stream); 2208 size_t numCounterOffers = 0; 2209 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2210 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2211 ALOG_ASSERT(index == 0); 2212 2213 // initialize fast mixer depending on configuration 2214 bool initFastMixer; 2215 switch (kUseFastMixer) { 2216 case FastMixer_Never: 2217 initFastMixer = false; 2218 break; 2219 case FastMixer_Always: 2220 initFastMixer = true; 2221 break; 2222 case FastMixer_Static: 2223 case FastMixer_Dynamic: 2224 initFastMixer = mFrameCount < mNormalFrameCount; 2225 break; 2226 } 2227 if (initFastMixer) { 2228 2229 // create a MonoPipe to connect our submix to FastMixer 2230 NBAIO_Format format = mOutputSink->format(); 2231 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2232 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2233 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2234 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2235 const NBAIO_Format offers[1] = {format}; 2236 size_t numCounterOffers = 0; 2237 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2238 ALOG_ASSERT(index == 0); 2239 monoPipe->setAvgFrames((mScreenState & 1) ? 2240 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2241 mPipeSink = monoPipe; 2242 2243#ifdef TEE_SINK_FRAMES 2244 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2245 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2246 numCounterOffers = 0; 2247 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2248 ALOG_ASSERT(index == 0); 2249 mTeeSink = teeSink; 2250 PipeReader *teeSource = new PipeReader(*teeSink); 2251 numCounterOffers = 0; 2252 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2253 ALOG_ASSERT(index == 0); 2254 mTeeSource = teeSource; 2255#endif 2256 2257 // create fast mixer and configure it initially with just one fast track for our submix 2258 mFastMixer = new FastMixer(); 2259 FastMixerStateQueue *sq = mFastMixer->sq(); 2260#ifdef STATE_QUEUE_DUMP 2261 sq->setObserverDump(&mStateQueueObserverDump); 2262 sq->setMutatorDump(&mStateQueueMutatorDump); 2263#endif 2264 FastMixerState *state = sq->begin(); 2265 FastTrack *fastTrack = &state->mFastTracks[0]; 2266 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2267 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2268 fastTrack->mVolumeProvider = NULL; 2269 fastTrack->mGeneration++; 2270 state->mFastTracksGen++; 2271 state->mTrackMask = 1; 2272 // fast mixer will use the HAL output sink 2273 state->mOutputSink = mOutputSink.get(); 2274 state->mOutputSinkGen++; 2275 state->mFrameCount = mFrameCount; 2276 state->mCommand = FastMixerState::COLD_IDLE; 2277 // already done in constructor initialization list 2278 //mFastMixerFutex = 0; 2279 state->mColdFutexAddr = &mFastMixerFutex; 2280 state->mColdGen++; 2281 state->mDumpState = &mFastMixerDumpState; 2282 state->mTeeSink = mTeeSink.get(); 2283 sq->end(); 2284 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2285 2286 // start the fast mixer 2287 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2288 pid_t tid = mFastMixer->getTid(); 2289 int err = requestPriority(getpid_cached, tid, 2); 2290 if (err != 0) { 2291 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2292 2, getpid_cached, tid, err); 2293 } 2294 2295#ifdef AUDIO_WATCHDOG 2296 // create and start the watchdog 2297 mAudioWatchdog = new AudioWatchdog(); 2298 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2299 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2300 tid = mAudioWatchdog->getTid(); 2301 err = requestPriority(getpid_cached, tid, 1); 2302 if (err != 0) { 2303 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2304 1, getpid_cached, tid, err); 2305 } 2306#endif 2307 2308 } else { 2309 mFastMixer = NULL; 2310 } 2311 2312 switch (kUseFastMixer) { 2313 case FastMixer_Never: 2314 case FastMixer_Dynamic: 2315 mNormalSink = mOutputSink; 2316 break; 2317 case FastMixer_Always: 2318 mNormalSink = mPipeSink; 2319 break; 2320 case FastMixer_Static: 2321 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2322 break; 2323 } 2324} 2325 2326AudioFlinger::MixerThread::~MixerThread() 2327{ 2328 if (mFastMixer != NULL) { 2329 FastMixerStateQueue *sq = mFastMixer->sq(); 2330 FastMixerState *state = sq->begin(); 2331 if (state->mCommand == FastMixerState::COLD_IDLE) { 2332 int32_t old = android_atomic_inc(&mFastMixerFutex); 2333 if (old == -1) { 2334 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2335 } 2336 } 2337 state->mCommand = FastMixerState::EXIT; 2338 sq->end(); 2339 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2340 mFastMixer->join(); 2341 // Though the fast mixer thread has exited, it's state queue is still valid. 2342 // We'll use that extract the final state which contains one remaining fast track 2343 // corresponding to our sub-mix. 2344 state = sq->begin(); 2345 ALOG_ASSERT(state->mTrackMask == 1); 2346 FastTrack *fastTrack = &state->mFastTracks[0]; 2347 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2348 delete fastTrack->mBufferProvider; 2349 sq->end(false /*didModify*/); 2350 delete mFastMixer; 2351 if (mAudioWatchdog != 0) { 2352 mAudioWatchdog->requestExit(); 2353 mAudioWatchdog->requestExitAndWait(); 2354 mAudioWatchdog.clear(); 2355 } 2356 } 2357 delete mAudioMixer; 2358} 2359 2360class CpuStats { 2361public: 2362 CpuStats(); 2363 void sample(const String8 &title); 2364#ifdef DEBUG_CPU_USAGE 2365private: 2366 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2367 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2368 2369 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2370 2371 int mCpuNum; // thread's current CPU number 2372 int mCpukHz; // frequency of thread's current CPU in kHz 2373#endif 2374}; 2375 2376CpuStats::CpuStats() 2377#ifdef DEBUG_CPU_USAGE 2378 : mCpuNum(-1), mCpukHz(-1) 2379#endif 2380{ 2381} 2382 2383void CpuStats::sample(const String8 &title) { 2384#ifdef DEBUG_CPU_USAGE 2385 // get current thread's delta CPU time in wall clock ns 2386 double wcNs; 2387 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2388 2389 // record sample for wall clock statistics 2390 if (valid) { 2391 mWcStats.sample(wcNs); 2392 } 2393 2394 // get the current CPU number 2395 int cpuNum = sched_getcpu(); 2396 2397 // get the current CPU frequency in kHz 2398 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2399 2400 // check if either CPU number or frequency changed 2401 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2402 mCpuNum = cpuNum; 2403 mCpukHz = cpukHz; 2404 // ignore sample for purposes of cycles 2405 valid = false; 2406 } 2407 2408 // if no change in CPU number or frequency, then record sample for cycle statistics 2409 if (valid && mCpukHz > 0) { 2410 double cycles = wcNs * cpukHz * 0.000001; 2411 mHzStats.sample(cycles); 2412 } 2413 2414 unsigned n = mWcStats.n(); 2415 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2416 if ((n & 127) == 1) { 2417 long long elapsed = mCpuUsage.elapsed(); 2418 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2419 double perLoop = elapsed / (double) n; 2420 double perLoop100 = perLoop * 0.01; 2421 double perLoop1k = perLoop * 0.001; 2422 double mean = mWcStats.mean(); 2423 double stddev = mWcStats.stddev(); 2424 double minimum = mWcStats.minimum(); 2425 double maximum = mWcStats.maximum(); 2426 double meanCycles = mHzStats.mean(); 2427 double stddevCycles = mHzStats.stddev(); 2428 double minCycles = mHzStats.minimum(); 2429 double maxCycles = mHzStats.maximum(); 2430 mCpuUsage.resetElapsed(); 2431 mWcStats.reset(); 2432 mHzStats.reset(); 2433 ALOGD("CPU usage for %s over past %.1f secs\n" 2434 " (%u mixer loops at %.1f mean ms per loop):\n" 2435 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2436 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2437 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2438 title.string(), 2439 elapsed * .000000001, n, perLoop * .000001, 2440 mean * .001, 2441 stddev * .001, 2442 minimum * .001, 2443 maximum * .001, 2444 mean / perLoop100, 2445 stddev / perLoop100, 2446 minimum / perLoop100, 2447 maximum / perLoop100, 2448 meanCycles / perLoop1k, 2449 stddevCycles / perLoop1k, 2450 minCycles / perLoop1k, 2451 maxCycles / perLoop1k); 2452 2453 } 2454 } 2455#endif 2456}; 2457 2458void AudioFlinger::PlaybackThread::checkSilentMode_l() 2459{ 2460 if (!mMasterMute) { 2461 char value[PROPERTY_VALUE_MAX]; 2462 if (property_get("ro.audio.silent", value, "0") > 0) { 2463 char *endptr; 2464 unsigned long ul = strtoul(value, &endptr, 0); 2465 if (*endptr == '\0' && ul != 0) { 2466 ALOGD("Silence is golden"); 2467 // The setprop command will not allow a property to be changed after 2468 // the first time it is set, so we don't have to worry about un-muting. 2469 setMasterMute_l(true); 2470 } 2471 } 2472 } 2473} 2474 2475bool AudioFlinger::PlaybackThread::threadLoop() 2476{ 2477 Vector< sp<Track> > tracksToRemove; 2478 2479 standbyTime = systemTime(); 2480 2481 // MIXER 2482 nsecs_t lastWarning = 0; 2483 2484 // DUPLICATING 2485 // FIXME could this be made local to while loop? 2486 writeFrames = 0; 2487 2488 cacheParameters_l(); 2489 sleepTime = idleSleepTime; 2490 2491 if (mType == MIXER) { 2492 sleepTimeShift = 0; 2493 } 2494 2495 CpuStats cpuStats; 2496 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2497 2498 acquireWakeLock(); 2499 2500 while (!exitPending()) 2501 { 2502 cpuStats.sample(myName); 2503 2504 Vector< sp<EffectChain> > effectChains; 2505 2506 processConfigEvents(); 2507 2508 { // scope for mLock 2509 2510 Mutex::Autolock _l(mLock); 2511 2512 if (checkForNewParameters_l()) { 2513 cacheParameters_l(); 2514 } 2515 2516 saveOutputTracks(); 2517 2518 // put audio hardware into standby after short delay 2519 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2520 mSuspended > 0)) { 2521 if (!mStandby) { 2522 2523 threadLoop_standby(); 2524 2525 mStandby = true; 2526 mBytesWritten = 0; 2527 } 2528 2529 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2530 // we're about to wait, flush the binder command buffer 2531 IPCThreadState::self()->flushCommands(); 2532 2533 clearOutputTracks(); 2534 2535 if (exitPending()) break; 2536 2537 releaseWakeLock_l(); 2538 // wait until we have something to do... 2539 ALOGV("%s going to sleep", myName.string()); 2540 mWaitWorkCV.wait(mLock); 2541 ALOGV("%s waking up", myName.string()); 2542 acquireWakeLock_l(); 2543 2544 mMixerStatus = MIXER_IDLE; 2545 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2546 2547 checkSilentMode_l(); 2548 2549 standbyTime = systemTime() + standbyDelay; 2550 sleepTime = idleSleepTime; 2551 if (mType == MIXER) { 2552 sleepTimeShift = 0; 2553 } 2554 2555 continue; 2556 } 2557 } 2558 2559 // mMixerStatusIgnoringFastTracks is also updated internally 2560 mMixerStatus = prepareTracks_l(&tracksToRemove); 2561 2562 // prevent any changes in effect chain list and in each effect chain 2563 // during mixing and effect process as the audio buffers could be deleted 2564 // or modified if an effect is created or deleted 2565 lockEffectChains_l(effectChains); 2566 } 2567 2568 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2569 threadLoop_mix(); 2570 } else { 2571 threadLoop_sleepTime(); 2572 } 2573 2574 if (mSuspended > 0) { 2575 sleepTime = suspendSleepTimeUs(); 2576 } 2577 2578 // only process effects if we're going to write 2579 if (sleepTime == 0) { 2580 for (size_t i = 0; i < effectChains.size(); i ++) { 2581 effectChains[i]->process_l(); 2582 } 2583 } 2584 2585 // enable changes in effect chain 2586 unlockEffectChains(effectChains); 2587 2588 // sleepTime == 0 means we must write to audio hardware 2589 if (sleepTime == 0) { 2590 2591 threadLoop_write(); 2592 2593if (mType == MIXER) { 2594 // write blocked detection 2595 nsecs_t now = systemTime(); 2596 nsecs_t delta = now - mLastWriteTime; 2597 if (!mStandby && delta > maxPeriod) { 2598 mNumDelayedWrites++; 2599 if ((now - lastWarning) > kWarningThrottleNs) { 2600#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2601 ScopedTrace st(ATRACE_TAG, "underrun"); 2602#endif 2603 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2604 ns2ms(delta), mNumDelayedWrites, this); 2605 lastWarning = now; 2606 } 2607 } 2608} 2609 2610 mStandby = false; 2611 } else { 2612 usleep(sleepTime); 2613 } 2614 2615 // Finally let go of removed track(s), without the lock held 2616 // since we can't guarantee the destructors won't acquire that 2617 // same lock. This will also mutate and push a new fast mixer state. 2618 threadLoop_removeTracks(tracksToRemove); 2619 tracksToRemove.clear(); 2620 2621 // FIXME I don't understand the need for this here; 2622 // it was in the original code but maybe the 2623 // assignment in saveOutputTracks() makes this unnecessary? 2624 clearOutputTracks(); 2625 2626 // Effect chains will be actually deleted here if they were removed from 2627 // mEffectChains list during mixing or effects processing 2628 effectChains.clear(); 2629 2630 // FIXME Note that the above .clear() is no longer necessary since effectChains 2631 // is now local to this block, but will keep it for now (at least until merge done). 2632 } 2633 2634 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2635 if (mType == MIXER || mType == DIRECT) { 2636 // put output stream into standby mode 2637 if (!mStandby) { 2638 mOutput->stream->common.standby(&mOutput->stream->common); 2639 } 2640 } 2641 2642 releaseWakeLock(); 2643 2644 ALOGV("Thread %p type %d exiting", this, mType); 2645 return false; 2646} 2647 2648void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2649{ 2650 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2651} 2652 2653void AudioFlinger::MixerThread::threadLoop_write() 2654{ 2655 // FIXME we should only do one push per cycle; confirm this is true 2656 // Start the fast mixer if it's not already running 2657 if (mFastMixer != NULL) { 2658 FastMixerStateQueue *sq = mFastMixer->sq(); 2659 FastMixerState *state = sq->begin(); 2660 if (state->mCommand != FastMixerState::MIX_WRITE && 2661 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2662 if (state->mCommand == FastMixerState::COLD_IDLE) { 2663 int32_t old = android_atomic_inc(&mFastMixerFutex); 2664 if (old == -1) { 2665 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2666 } 2667 if (mAudioWatchdog != 0) { 2668 mAudioWatchdog->resume(); 2669 } 2670 } 2671 state->mCommand = FastMixerState::MIX_WRITE; 2672 sq->end(); 2673 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2674 if (kUseFastMixer == FastMixer_Dynamic) { 2675 mNormalSink = mPipeSink; 2676 } 2677 } else { 2678 sq->end(false /*didModify*/); 2679 } 2680 } 2681 PlaybackThread::threadLoop_write(); 2682} 2683 2684// shared by MIXER and DIRECT, overridden by DUPLICATING 2685void AudioFlinger::PlaybackThread::threadLoop_write() 2686{ 2687 // FIXME rewrite to reduce number of system calls 2688 mLastWriteTime = systemTime(); 2689 mInWrite = true; 2690 int bytesWritten; 2691 2692 // If an NBAIO sink is present, use it to write the normal mixer's submix 2693 if (mNormalSink != 0) { 2694#define mBitShift 2 // FIXME 2695 size_t count = mixBufferSize >> mBitShift; 2696#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2697 Tracer::traceBegin(ATRACE_TAG, "write"); 2698#endif 2699 // update the setpoint when gScreenState changes 2700 uint32_t screenState = gScreenState; 2701 if (screenState != mScreenState) { 2702 mScreenState = screenState; 2703 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2704 if (pipe != NULL) { 2705 pipe->setAvgFrames((mScreenState & 1) ? 2706 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2707 } 2708 } 2709 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2710#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2711 Tracer::traceEnd(ATRACE_TAG); 2712#endif 2713 if (framesWritten > 0) { 2714 bytesWritten = framesWritten << mBitShift; 2715 } else { 2716 bytesWritten = framesWritten; 2717 } 2718 // otherwise use the HAL / AudioStreamOut directly 2719 } else { 2720 // Direct output thread. 2721 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2722 } 2723 2724 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2725 mNumWrites++; 2726 mInWrite = false; 2727} 2728 2729void AudioFlinger::MixerThread::threadLoop_standby() 2730{ 2731 // Idle the fast mixer if it's currently running 2732 if (mFastMixer != NULL) { 2733 FastMixerStateQueue *sq = mFastMixer->sq(); 2734 FastMixerState *state = sq->begin(); 2735 if (!(state->mCommand & FastMixerState::IDLE)) { 2736 state->mCommand = FastMixerState::COLD_IDLE; 2737 state->mColdFutexAddr = &mFastMixerFutex; 2738 state->mColdGen++; 2739 mFastMixerFutex = 0; 2740 sq->end(); 2741 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2742 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2743 if (kUseFastMixer == FastMixer_Dynamic) { 2744 mNormalSink = mOutputSink; 2745 } 2746 if (mAudioWatchdog != 0) { 2747 mAudioWatchdog->pause(); 2748 } 2749 } else { 2750 sq->end(false /*didModify*/); 2751 } 2752 } 2753 PlaybackThread::threadLoop_standby(); 2754} 2755 2756// shared by MIXER and DIRECT, overridden by DUPLICATING 2757void AudioFlinger::PlaybackThread::threadLoop_standby() 2758{ 2759 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2760 mOutput->stream->common.standby(&mOutput->stream->common); 2761} 2762 2763void AudioFlinger::MixerThread::threadLoop_mix() 2764{ 2765 // obtain the presentation timestamp of the next output buffer 2766 int64_t pts; 2767 status_t status = INVALID_OPERATION; 2768 2769 if (NULL != mOutput->stream->get_next_write_timestamp) { 2770 status = mOutput->stream->get_next_write_timestamp( 2771 mOutput->stream, &pts); 2772 } 2773 2774 if (status != NO_ERROR) { 2775 pts = AudioBufferProvider::kInvalidPTS; 2776 } 2777 2778 // mix buffers... 2779 mAudioMixer->process(pts); 2780 // increase sleep time progressively when application underrun condition clears. 2781 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2782 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2783 // such that we would underrun the audio HAL. 2784 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2785 sleepTimeShift--; 2786 } 2787 sleepTime = 0; 2788 standbyTime = systemTime() + standbyDelay; 2789 //TODO: delay standby when effects have a tail 2790} 2791 2792void AudioFlinger::MixerThread::threadLoop_sleepTime() 2793{ 2794 // If no tracks are ready, sleep once for the duration of an output 2795 // buffer size, then write 0s to the output 2796 if (sleepTime == 0) { 2797 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2798 sleepTime = activeSleepTime >> sleepTimeShift; 2799 if (sleepTime < kMinThreadSleepTimeUs) { 2800 sleepTime = kMinThreadSleepTimeUs; 2801 } 2802 // reduce sleep time in case of consecutive application underruns to avoid 2803 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2804 // duration we would end up writing less data than needed by the audio HAL if 2805 // the condition persists. 2806 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2807 sleepTimeShift++; 2808 } 2809 } else { 2810 sleepTime = idleSleepTime; 2811 } 2812 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2813 memset (mMixBuffer, 0, mixBufferSize); 2814 sleepTime = 0; 2815 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2816 } 2817 // TODO add standby time extension fct of effect tail 2818} 2819 2820// prepareTracks_l() must be called with ThreadBase::mLock held 2821AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2822 Vector< sp<Track> > *tracksToRemove) 2823{ 2824 2825 mixer_state mixerStatus = MIXER_IDLE; 2826 // find out which tracks need to be processed 2827 size_t count = mActiveTracks.size(); 2828 size_t mixedTracks = 0; 2829 size_t tracksWithEffect = 0; 2830 // counts only _active_ fast tracks 2831 size_t fastTracks = 0; 2832 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2833 2834 float masterVolume = mMasterVolume; 2835 bool masterMute = mMasterMute; 2836 2837 if (masterMute) { 2838 masterVolume = 0; 2839 } 2840 // Delegate master volume control to effect in output mix effect chain if needed 2841 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2842 if (chain != 0) { 2843 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2844 chain->setVolume_l(&v, &v); 2845 masterVolume = (float)((v + (1 << 23)) >> 24); 2846 chain.clear(); 2847 } 2848 2849 // prepare a new state to push 2850 FastMixerStateQueue *sq = NULL; 2851 FastMixerState *state = NULL; 2852 bool didModify = false; 2853 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2854 if (mFastMixer != NULL) { 2855 sq = mFastMixer->sq(); 2856 state = sq->begin(); 2857 } 2858 2859 for (size_t i=0 ; i<count ; i++) { 2860 sp<Track> t = mActiveTracks[i].promote(); 2861 if (t == 0) continue; 2862 2863 // this const just means the local variable doesn't change 2864 Track* const track = t.get(); 2865 2866 // process fast tracks 2867 if (track->isFastTrack()) { 2868 2869 // It's theoretically possible (though unlikely) for a fast track to be created 2870 // and then removed within the same normal mix cycle. This is not a problem, as 2871 // the track never becomes active so it's fast mixer slot is never touched. 2872 // The converse, of removing an (active) track and then creating a new track 2873 // at the identical fast mixer slot within the same normal mix cycle, 2874 // is impossible because the slot isn't marked available until the end of each cycle. 2875 int j = track->mFastIndex; 2876 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2877 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2878 FastTrack *fastTrack = &state->mFastTracks[j]; 2879 2880 // Determine whether the track is currently in underrun condition, 2881 // and whether it had a recent underrun. 2882 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2883 FastTrackUnderruns underruns = ftDump->mUnderruns; 2884 uint32_t recentFull = (underruns.mBitFields.mFull - 2885 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2886 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2887 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2888 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2889 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2890 uint32_t recentUnderruns = recentPartial + recentEmpty; 2891 track->mObservedUnderruns = underruns; 2892 // don't count underruns that occur while stopping or pausing 2893 // or stopped which can occur when flush() is called while active 2894 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2895 track->mUnderrunCount += recentUnderruns; 2896 } 2897 2898 // This is similar to the state machine for normal tracks, 2899 // with a few modifications for fast tracks. 2900 bool isActive = true; 2901 switch (track->mState) { 2902 case TrackBase::STOPPING_1: 2903 // track stays active in STOPPING_1 state until first underrun 2904 if (recentUnderruns > 0) { 2905 track->mState = TrackBase::STOPPING_2; 2906 } 2907 break; 2908 case TrackBase::PAUSING: 2909 // ramp down is not yet implemented 2910 track->setPaused(); 2911 break; 2912 case TrackBase::RESUMING: 2913 // ramp up is not yet implemented 2914 track->mState = TrackBase::ACTIVE; 2915 break; 2916 case TrackBase::ACTIVE: 2917 if (recentFull > 0 || recentPartial > 0) { 2918 // track has provided at least some frames recently: reset retry count 2919 track->mRetryCount = kMaxTrackRetries; 2920 } 2921 if (recentUnderruns == 0) { 2922 // no recent underruns: stay active 2923 break; 2924 } 2925 // there has recently been an underrun of some kind 2926 if (track->sharedBuffer() == 0) { 2927 // were any of the recent underruns "empty" (no frames available)? 2928 if (recentEmpty == 0) { 2929 // no, then ignore the partial underruns as they are allowed indefinitely 2930 break; 2931 } 2932 // there has recently been an "empty" underrun: decrement the retry counter 2933 if (--(track->mRetryCount) > 0) { 2934 break; 2935 } 2936 // indicate to client process that the track was disabled because of underrun; 2937 // it will then automatically call start() when data is available 2938 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2939 // remove from active list, but state remains ACTIVE [confusing but true] 2940 isActive = false; 2941 break; 2942 } 2943 // fall through 2944 case TrackBase::STOPPING_2: 2945 case TrackBase::PAUSED: 2946 case TrackBase::TERMINATED: 2947 case TrackBase::STOPPED: 2948 case TrackBase::FLUSHED: // flush() while active 2949 // Check for presentation complete if track is inactive 2950 // We have consumed all the buffers of this track. 2951 // This would be incomplete if we auto-paused on underrun 2952 { 2953 size_t audioHALFrames = 2954 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2955 size_t framesWritten = 2956 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2957 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2958 // track stays in active list until presentation is complete 2959 break; 2960 } 2961 } 2962 if (track->isStopping_2()) { 2963 track->mState = TrackBase::STOPPED; 2964 } 2965 if (track->isStopped()) { 2966 // Can't reset directly, as fast mixer is still polling this track 2967 // track->reset(); 2968 // So instead mark this track as needing to be reset after push with ack 2969 resetMask |= 1 << i; 2970 } 2971 isActive = false; 2972 break; 2973 case TrackBase::IDLE: 2974 default: 2975 LOG_FATAL("unexpected track state %d", track->mState); 2976 } 2977 2978 if (isActive) { 2979 // was it previously inactive? 2980 if (!(state->mTrackMask & (1 << j))) { 2981 ExtendedAudioBufferProvider *eabp = track; 2982 VolumeProvider *vp = track; 2983 fastTrack->mBufferProvider = eabp; 2984 fastTrack->mVolumeProvider = vp; 2985 fastTrack->mSampleRate = track->mSampleRate; 2986 fastTrack->mChannelMask = track->mChannelMask; 2987 fastTrack->mGeneration++; 2988 state->mTrackMask |= 1 << j; 2989 didModify = true; 2990 // no acknowledgement required for newly active tracks 2991 } 2992 // cache the combined master volume and stream type volume for fast mixer; this 2993 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2994 track->mCachedVolume = track->isMuted() ? 2995 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2996 ++fastTracks; 2997 } else { 2998 // was it previously active? 2999 if (state->mTrackMask & (1 << j)) { 3000 fastTrack->mBufferProvider = NULL; 3001 fastTrack->mGeneration++; 3002 state->mTrackMask &= ~(1 << j); 3003 didModify = true; 3004 // If any fast tracks were removed, we must wait for acknowledgement 3005 // because we're about to decrement the last sp<> on those tracks. 3006 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3007 } else { 3008 LOG_FATAL("fast track %d should have been active", j); 3009 } 3010 tracksToRemove->add(track); 3011 // Avoids a misleading display in dumpsys 3012 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3013 } 3014 continue; 3015 } 3016 3017 { // local variable scope to avoid goto warning 3018 3019 audio_track_cblk_t* cblk = track->cblk(); 3020 3021 // The first time a track is added we wait 3022 // for all its buffers to be filled before processing it 3023 int name = track->name(); 3024 // make sure that we have enough frames to mix one full buffer. 3025 // enforce this condition only once to enable draining the buffer in case the client 3026 // app does not call stop() and relies on underrun to stop: 3027 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3028 // during last round 3029 uint32_t minFrames = 1; 3030 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3031 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3032 if (t->sampleRate() == (int)mSampleRate) { 3033 minFrames = mNormalFrameCount; 3034 } else { 3035 // +1 for rounding and +1 for additional sample needed for interpolation 3036 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3037 // add frames already consumed but not yet released by the resampler 3038 // because cblk->framesReady() will include these frames 3039 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3040 // the minimum track buffer size is normally twice the number of frames necessary 3041 // to fill one buffer and the resampler should not leave more than one buffer worth 3042 // of unreleased frames after each pass, but just in case... 3043 ALOG_ASSERT(minFrames <= cblk->frameCount); 3044 } 3045 } 3046 if ((track->framesReady() >= minFrames) && track->isReady() && 3047 !track->isPaused() && !track->isTerminated()) 3048 { 3049 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3050 3051 mixedTracks++; 3052 3053 // track->mainBuffer() != mMixBuffer means there is an effect chain 3054 // connected to the track 3055 chain.clear(); 3056 if (track->mainBuffer() != mMixBuffer) { 3057 chain = getEffectChain_l(track->sessionId()); 3058 // Delegate volume control to effect in track effect chain if needed 3059 if (chain != 0) { 3060 tracksWithEffect++; 3061 } else { 3062 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3063 name, track->sessionId()); 3064 } 3065 } 3066 3067 3068 int param = AudioMixer::VOLUME; 3069 if (track->mFillingUpStatus == Track::FS_FILLED) { 3070 // no ramp for the first volume setting 3071 track->mFillingUpStatus = Track::FS_ACTIVE; 3072 if (track->mState == TrackBase::RESUMING) { 3073 track->mState = TrackBase::ACTIVE; 3074 param = AudioMixer::RAMP_VOLUME; 3075 } 3076 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3077 } else if (cblk->server != 0) { 3078 // If the track is stopped before the first frame was mixed, 3079 // do not apply ramp 3080 param = AudioMixer::RAMP_VOLUME; 3081 } 3082 3083 // compute volume for this track 3084 uint32_t vl, vr, va; 3085 if (track->isMuted() || track->isPausing() || 3086 mStreamTypes[track->streamType()].mute) { 3087 vl = vr = va = 0; 3088 if (track->isPausing()) { 3089 track->setPaused(); 3090 } 3091 } else { 3092 3093 // read original volumes with volume control 3094 float typeVolume = mStreamTypes[track->streamType()].volume; 3095 float v = masterVolume * typeVolume; 3096 uint32_t vlr = cblk->getVolumeLR(); 3097 vl = vlr & 0xFFFF; 3098 vr = vlr >> 16; 3099 // track volumes come from shared memory, so can't be trusted and must be clamped 3100 if (vl > MAX_GAIN_INT) { 3101 ALOGV("Track left volume out of range: %04X", vl); 3102 vl = MAX_GAIN_INT; 3103 } 3104 if (vr > MAX_GAIN_INT) { 3105 ALOGV("Track right volume out of range: %04X", vr); 3106 vr = MAX_GAIN_INT; 3107 } 3108 // now apply the master volume and stream type volume 3109 vl = (uint32_t)(v * vl) << 12; 3110 vr = (uint32_t)(v * vr) << 12; 3111 // assuming master volume and stream type volume each go up to 1.0, 3112 // vl and vr are now in 8.24 format 3113 3114 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3115 // send level comes from shared memory and so may be corrupt 3116 if (sendLevel > MAX_GAIN_INT) { 3117 ALOGV("Track send level out of range: %04X", sendLevel); 3118 sendLevel = MAX_GAIN_INT; 3119 } 3120 va = (uint32_t)(v * sendLevel); 3121 } 3122 // Delegate volume control to effect in track effect chain if needed 3123 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3124 // Do not ramp volume if volume is controlled by effect 3125 param = AudioMixer::VOLUME; 3126 track->mHasVolumeController = true; 3127 } else { 3128 // force no volume ramp when volume controller was just disabled or removed 3129 // from effect chain to avoid volume spike 3130 if (track->mHasVolumeController) { 3131 param = AudioMixer::VOLUME; 3132 } 3133 track->mHasVolumeController = false; 3134 } 3135 3136 // Convert volumes from 8.24 to 4.12 format 3137 // This additional clamping is needed in case chain->setVolume_l() overshot 3138 vl = (vl + (1 << 11)) >> 12; 3139 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3140 vr = (vr + (1 << 11)) >> 12; 3141 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3142 3143 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3144 3145 // XXX: these things DON'T need to be done each time 3146 mAudioMixer->setBufferProvider(name, track); 3147 mAudioMixer->enable(name); 3148 3149 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3150 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3151 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3152 mAudioMixer->setParameter( 3153 name, 3154 AudioMixer::TRACK, 3155 AudioMixer::FORMAT, (void *)track->format()); 3156 mAudioMixer->setParameter( 3157 name, 3158 AudioMixer::TRACK, 3159 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3160 mAudioMixer->setParameter( 3161 name, 3162 AudioMixer::RESAMPLE, 3163 AudioMixer::SAMPLE_RATE, 3164 (void *)(cblk->sampleRate)); 3165 mAudioMixer->setParameter( 3166 name, 3167 AudioMixer::TRACK, 3168 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3169 mAudioMixer->setParameter( 3170 name, 3171 AudioMixer::TRACK, 3172 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3173 3174 // reset retry count 3175 track->mRetryCount = kMaxTrackRetries; 3176 3177 // If one track is ready, set the mixer ready if: 3178 // - the mixer was not ready during previous round OR 3179 // - no other track is not ready 3180 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3181 mixerStatus != MIXER_TRACKS_ENABLED) { 3182 mixerStatus = MIXER_TRACKS_READY; 3183 } 3184 } else { 3185 // clear effect chain input buffer if an active track underruns to avoid sending 3186 // previous audio buffer again to effects 3187 chain = getEffectChain_l(track->sessionId()); 3188 if (chain != 0) { 3189 chain->clearInputBuffer(); 3190 } 3191 3192 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3193 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3194 track->isStopped() || track->isPaused()) { 3195 // We have consumed all the buffers of this track. 3196 // Remove it from the list of active tracks. 3197 // TODO: use actual buffer filling status instead of latency when available from 3198 // audio HAL 3199 size_t audioHALFrames = 3200 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3201 size_t framesWritten = 3202 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3203 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3204 if (track->isStopped()) { 3205 track->reset(); 3206 } 3207 tracksToRemove->add(track); 3208 } 3209 } else { 3210 track->mUnderrunCount++; 3211 // No buffers for this track. Give it a few chances to 3212 // fill a buffer, then remove it from active list. 3213 if (--(track->mRetryCount) <= 0) { 3214 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3215 tracksToRemove->add(track); 3216 // indicate to client process that the track was disabled because of underrun; 3217 // it will then automatically call start() when data is available 3218 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3219 // If one track is not ready, mark the mixer also not ready if: 3220 // - the mixer was ready during previous round OR 3221 // - no other track is ready 3222 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3223 mixerStatus != MIXER_TRACKS_READY) { 3224 mixerStatus = MIXER_TRACKS_ENABLED; 3225 } 3226 } 3227 mAudioMixer->disable(name); 3228 } 3229 3230 } // local variable scope to avoid goto warning 3231track_is_ready: ; 3232 3233 } 3234 3235 // Push the new FastMixer state if necessary 3236 bool pauseAudioWatchdog = false; 3237 if (didModify) { 3238 state->mFastTracksGen++; 3239 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3240 if (kUseFastMixer == FastMixer_Dynamic && 3241 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3242 state->mCommand = FastMixerState::COLD_IDLE; 3243 state->mColdFutexAddr = &mFastMixerFutex; 3244 state->mColdGen++; 3245 mFastMixerFutex = 0; 3246 if (kUseFastMixer == FastMixer_Dynamic) { 3247 mNormalSink = mOutputSink; 3248 } 3249 // If we go into cold idle, need to wait for acknowledgement 3250 // so that fast mixer stops doing I/O. 3251 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3252 pauseAudioWatchdog = true; 3253 } 3254 sq->end(); 3255 } 3256 if (sq != NULL) { 3257 sq->end(didModify); 3258 sq->push(block); 3259 } 3260 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3261 mAudioWatchdog->pause(); 3262 } 3263 3264 // Now perform the deferred reset on fast tracks that have stopped 3265 while (resetMask != 0) { 3266 size_t i = __builtin_ctz(resetMask); 3267 ALOG_ASSERT(i < count); 3268 resetMask &= ~(1 << i); 3269 sp<Track> t = mActiveTracks[i].promote(); 3270 if (t == 0) continue; 3271 Track* track = t.get(); 3272 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3273 track->reset(); 3274 } 3275 3276 // remove all the tracks that need to be... 3277 count = tracksToRemove->size(); 3278 if (CC_UNLIKELY(count)) { 3279 for (size_t i=0 ; i<count ; i++) { 3280 const sp<Track>& track = tracksToRemove->itemAt(i); 3281 mActiveTracks.remove(track); 3282 if (track->mainBuffer() != mMixBuffer) { 3283 chain = getEffectChain_l(track->sessionId()); 3284 if (chain != 0) { 3285 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3286 chain->decActiveTrackCnt(); 3287 } 3288 } 3289 if (track->isTerminated()) { 3290 removeTrack_l(track); 3291 } 3292 } 3293 } 3294 3295 // mix buffer must be cleared if all tracks are connected to an 3296 // effect chain as in this case the mixer will not write to 3297 // mix buffer and track effects will accumulate into it 3298 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3299 // FIXME as a performance optimization, should remember previous zero status 3300 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3301 } 3302 3303 // if any fast tracks, then status is ready 3304 mMixerStatusIgnoringFastTracks = mixerStatus; 3305 if (fastTracks > 0) { 3306 mixerStatus = MIXER_TRACKS_READY; 3307 } 3308 return mixerStatus; 3309} 3310 3311/* 3312The derived values that are cached: 3313 - mixBufferSize from frame count * frame size 3314 - activeSleepTime from activeSleepTimeUs() 3315 - idleSleepTime from idleSleepTimeUs() 3316 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3317 - maxPeriod from frame count and sample rate (MIXER only) 3318 3319The parameters that affect these derived values are: 3320 - frame count 3321 - frame size 3322 - sample rate 3323 - device type: A2DP or not 3324 - device latency 3325 - format: PCM or not 3326 - active sleep time 3327 - idle sleep time 3328*/ 3329 3330void AudioFlinger::PlaybackThread::cacheParameters_l() 3331{ 3332 mixBufferSize = mNormalFrameCount * mFrameSize; 3333 activeSleepTime = activeSleepTimeUs(); 3334 idleSleepTime = idleSleepTimeUs(); 3335} 3336 3337void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3338{ 3339 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3340 this, streamType, mTracks.size()); 3341 Mutex::Autolock _l(mLock); 3342 3343 size_t size = mTracks.size(); 3344 for (size_t i = 0; i < size; i++) { 3345 sp<Track> t = mTracks[i]; 3346 if (t->streamType() == streamType) { 3347 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3348 t->mCblk->cv.signal(); 3349 } 3350 } 3351} 3352 3353// getTrackName_l() must be called with ThreadBase::mLock held 3354int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3355{ 3356 return mAudioMixer->getTrackName(channelMask); 3357} 3358 3359// deleteTrackName_l() must be called with ThreadBase::mLock held 3360void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3361{ 3362 ALOGV("remove track (%d) and delete from mixer", name); 3363 mAudioMixer->deleteTrackName(name); 3364} 3365 3366// checkForNewParameters_l() must be called with ThreadBase::mLock held 3367bool AudioFlinger::MixerThread::checkForNewParameters_l() 3368{ 3369 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3370 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3371 bool reconfig = false; 3372 3373 while (!mNewParameters.isEmpty()) { 3374 3375 if (mFastMixer != NULL) { 3376 FastMixerStateQueue *sq = mFastMixer->sq(); 3377 FastMixerState *state = sq->begin(); 3378 if (!(state->mCommand & FastMixerState::IDLE)) { 3379 previousCommand = state->mCommand; 3380 state->mCommand = FastMixerState::HOT_IDLE; 3381 sq->end(); 3382 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3383 } else { 3384 sq->end(false /*didModify*/); 3385 } 3386 } 3387 3388 status_t status = NO_ERROR; 3389 String8 keyValuePair = mNewParameters[0]; 3390 AudioParameter param = AudioParameter(keyValuePair); 3391 int value; 3392 3393 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3394 reconfig = true; 3395 } 3396 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3397 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3398 status = BAD_VALUE; 3399 } else { 3400 reconfig = true; 3401 } 3402 } 3403 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3404 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3405 status = BAD_VALUE; 3406 } else { 3407 reconfig = true; 3408 } 3409 } 3410 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3411 // do not accept frame count changes if tracks are open as the track buffer 3412 // size depends on frame count and correct behavior would not be guaranteed 3413 // if frame count is changed after track creation 3414 if (!mTracks.isEmpty()) { 3415 status = INVALID_OPERATION; 3416 } else { 3417 reconfig = true; 3418 } 3419 } 3420 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3421#ifdef ADD_BATTERY_DATA 3422 // when changing the audio output device, call addBatteryData to notify 3423 // the change 3424 if ((int)mDevice != value) { 3425 uint32_t params = 0; 3426 // check whether speaker is on 3427 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3428 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3429 } 3430 3431 int deviceWithoutSpeaker 3432 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3433 // check if any other device (except speaker) is on 3434 if (value & deviceWithoutSpeaker ) { 3435 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3436 } 3437 3438 if (params != 0) { 3439 addBatteryData(params); 3440 } 3441 } 3442#endif 3443 3444 // forward device change to effects that have requested to be 3445 // aware of attached audio device. 3446 mDevice = (audio_devices_t) value; 3447 for (size_t i = 0; i < mEffectChains.size(); i++) { 3448 mEffectChains[i]->setDevice_l(mDevice); 3449 } 3450 } 3451 3452 if (status == NO_ERROR) { 3453 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3454 keyValuePair.string()); 3455 if (!mStandby && status == INVALID_OPERATION) { 3456 mOutput->stream->common.standby(&mOutput->stream->common); 3457 mStandby = true; 3458 mBytesWritten = 0; 3459 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3460 keyValuePair.string()); 3461 } 3462 if (status == NO_ERROR && reconfig) { 3463 delete mAudioMixer; 3464 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3465 mAudioMixer = NULL; 3466 readOutputParameters(); 3467 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3468 for (size_t i = 0; i < mTracks.size() ; i++) { 3469 int name = getTrackName_l(mTracks[i]->mChannelMask); 3470 if (name < 0) break; 3471 mTracks[i]->mName = name; 3472 // limit track sample rate to 2 x new output sample rate 3473 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3474 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3475 } 3476 } 3477 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3478 } 3479 } 3480 3481 mNewParameters.removeAt(0); 3482 3483 mParamStatus = status; 3484 mParamCond.signal(); 3485 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3486 // already timed out waiting for the status and will never signal the condition. 3487 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3488 } 3489 3490 if (!(previousCommand & FastMixerState::IDLE)) { 3491 ALOG_ASSERT(mFastMixer != NULL); 3492 FastMixerStateQueue *sq = mFastMixer->sq(); 3493 FastMixerState *state = sq->begin(); 3494 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3495 state->mCommand = previousCommand; 3496 sq->end(); 3497 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3498 } 3499 3500 return reconfig; 3501} 3502 3503status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3504{ 3505 const size_t SIZE = 256; 3506 char buffer[SIZE]; 3507 String8 result; 3508 3509 PlaybackThread::dumpInternals(fd, args); 3510 3511 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3512 result.append(buffer); 3513 write(fd, result.string(), result.size()); 3514 3515 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3516 FastMixerDumpState copy = mFastMixerDumpState; 3517 copy.dump(fd); 3518 3519#ifdef STATE_QUEUE_DUMP 3520 // Similar for state queue 3521 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3522 observerCopy.dump(fd); 3523 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3524 mutatorCopy.dump(fd); 3525#endif 3526 3527 // Write the tee output to a .wav file 3528 NBAIO_Source *teeSource = mTeeSource.get(); 3529 if (teeSource != NULL) { 3530 char teePath[64]; 3531 struct timeval tv; 3532 gettimeofday(&tv, NULL); 3533 struct tm tm; 3534 localtime_r(&tv.tv_sec, &tm); 3535 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3536 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3537 if (teeFd >= 0) { 3538 char wavHeader[44]; 3539 memcpy(wavHeader, 3540 "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", 3541 sizeof(wavHeader)); 3542 NBAIO_Format format = teeSource->format(); 3543 unsigned channelCount = Format_channelCount(format); 3544 ALOG_ASSERT(channelCount <= FCC_2); 3545 unsigned sampleRate = Format_sampleRate(format); 3546 wavHeader[22] = channelCount; // number of channels 3547 wavHeader[24] = sampleRate; // sample rate 3548 wavHeader[25] = sampleRate >> 8; 3549 wavHeader[32] = channelCount * 2; // block alignment 3550 write(teeFd, wavHeader, sizeof(wavHeader)); 3551 size_t total = 0; 3552 bool firstRead = true; 3553 for (;;) { 3554#define TEE_SINK_READ 1024 3555 short buffer[TEE_SINK_READ * FCC_2]; 3556 size_t count = TEE_SINK_READ; 3557 ssize_t actual = teeSource->read(buffer, count); 3558 bool wasFirstRead = firstRead; 3559 firstRead = false; 3560 if (actual <= 0) { 3561 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3562 continue; 3563 } 3564 break; 3565 } 3566 ALOG_ASSERT(actual <= (ssize_t)count); 3567 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3568 total += actual; 3569 } 3570 lseek(teeFd, (off_t) 4, SEEK_SET); 3571 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3572 write(teeFd, &temp, sizeof(temp)); 3573 lseek(teeFd, (off_t) 40, SEEK_SET); 3574 temp = total * channelCount * sizeof(short); 3575 write(teeFd, &temp, sizeof(temp)); 3576 close(teeFd); 3577 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3578 } else { 3579 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3580 } 3581 } 3582 3583 if (mAudioWatchdog != 0) { 3584 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3585 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3586 wdCopy.dump(fd); 3587 } 3588 3589 return NO_ERROR; 3590} 3591 3592uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3593{ 3594 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3595} 3596 3597uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3598{ 3599 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3600} 3601 3602void AudioFlinger::MixerThread::cacheParameters_l() 3603{ 3604 PlaybackThread::cacheParameters_l(); 3605 3606 // FIXME: Relaxed timing because of a certain device that can't meet latency 3607 // Should be reduced to 2x after the vendor fixes the driver issue 3608 // increase threshold again due to low power audio mode. The way this warning 3609 // threshold is calculated and its usefulness should be reconsidered anyway. 3610 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3611} 3612 3613// ---------------------------------------------------------------------------- 3614AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3615 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3616 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3617 // mLeftVolFloat, mRightVolFloat 3618{ 3619} 3620 3621AudioFlinger::DirectOutputThread::~DirectOutputThread() 3622{ 3623} 3624 3625AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3626 Vector< sp<Track> > *tracksToRemove 3627) 3628{ 3629 sp<Track> trackToRemove; 3630 3631 mixer_state mixerStatus = MIXER_IDLE; 3632 3633 // find out which tracks need to be processed 3634 if (mActiveTracks.size() != 0) { 3635 sp<Track> t = mActiveTracks[0].promote(); 3636 // The track died recently 3637 if (t == 0) return MIXER_IDLE; 3638 3639 Track* const track = t.get(); 3640 audio_track_cblk_t* cblk = track->cblk(); 3641 3642 // The first time a track is added we wait 3643 // for all its buffers to be filled before processing it 3644 uint32_t minFrames; 3645 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3646 minFrames = mNormalFrameCount; 3647 } else { 3648 minFrames = 1; 3649 } 3650 if ((track->framesReady() >= minFrames) && track->isReady() && 3651 !track->isPaused() && !track->isTerminated()) 3652 { 3653 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3654 3655 if (track->mFillingUpStatus == Track::FS_FILLED) { 3656 track->mFillingUpStatus = Track::FS_ACTIVE; 3657 mLeftVolFloat = mRightVolFloat = 0; 3658 if (track->mState == TrackBase::RESUMING) { 3659 track->mState = TrackBase::ACTIVE; 3660 } 3661 } 3662 3663 // compute volume for this track 3664 float left, right; 3665 if (track->isMuted() || mMasterMute || track->isPausing() || 3666 mStreamTypes[track->streamType()].mute) { 3667 left = right = 0; 3668 if (track->isPausing()) { 3669 track->setPaused(); 3670 } 3671 } else { 3672 float typeVolume = mStreamTypes[track->streamType()].volume; 3673 float v = mMasterVolume * typeVolume; 3674 uint32_t vlr = cblk->getVolumeLR(); 3675 float v_clamped = v * (vlr & 0xFFFF); 3676 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3677 left = v_clamped/MAX_GAIN; 3678 v_clamped = v * (vlr >> 16); 3679 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3680 right = v_clamped/MAX_GAIN; 3681 } 3682 3683 if (left != mLeftVolFloat || right != mRightVolFloat) { 3684 mLeftVolFloat = left; 3685 mRightVolFloat = right; 3686 3687 // Convert volumes from float to 8.24 3688 uint32_t vl = (uint32_t)(left * (1 << 24)); 3689 uint32_t vr = (uint32_t)(right * (1 << 24)); 3690 3691 // Delegate volume control to effect in track effect chain if needed 3692 // only one effect chain can be present on DirectOutputThread, so if 3693 // there is one, the track is connected to it 3694 if (!mEffectChains.isEmpty()) { 3695 // Do not ramp volume if volume is controlled by effect 3696 mEffectChains[0]->setVolume_l(&vl, &vr); 3697 left = (float)vl / (1 << 24); 3698 right = (float)vr / (1 << 24); 3699 } 3700 mOutput->stream->set_volume(mOutput->stream, left, right); 3701 } 3702 3703 // reset retry count 3704 track->mRetryCount = kMaxTrackRetriesDirect; 3705 mActiveTrack = t; 3706 mixerStatus = MIXER_TRACKS_READY; 3707 } else { 3708 // clear effect chain input buffer if an active track underruns to avoid sending 3709 // previous audio buffer again to effects 3710 if (!mEffectChains.isEmpty()) { 3711 mEffectChains[0]->clearInputBuffer(); 3712 } 3713 3714 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3715 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3716 track->isStopped() || track->isPaused()) { 3717 // We have consumed all the buffers of this track. 3718 // Remove it from the list of active tracks. 3719 // TODO: implement behavior for compressed audio 3720 size_t audioHALFrames = 3721 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3722 size_t framesWritten = 3723 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3724 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3725 if (track->isStopped()) { 3726 track->reset(); 3727 } 3728 trackToRemove = track; 3729 } 3730 } else { 3731 // No buffers for this track. Give it a few chances to 3732 // fill a buffer, then remove it from active list. 3733 if (--(track->mRetryCount) <= 0) { 3734 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3735 trackToRemove = track; 3736 } else { 3737 mixerStatus = MIXER_TRACKS_ENABLED; 3738 } 3739 } 3740 } 3741 } 3742 3743 // FIXME merge this with similar code for removing multiple tracks 3744 // remove all the tracks that need to be... 3745 if (CC_UNLIKELY(trackToRemove != 0)) { 3746 tracksToRemove->add(trackToRemove); 3747 mActiveTracks.remove(trackToRemove); 3748 if (!mEffectChains.isEmpty()) { 3749 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3750 trackToRemove->sessionId()); 3751 mEffectChains[0]->decActiveTrackCnt(); 3752 } 3753 if (trackToRemove->isTerminated()) { 3754 removeTrack_l(trackToRemove); 3755 } 3756 } 3757 3758 return mixerStatus; 3759} 3760 3761void AudioFlinger::DirectOutputThread::threadLoop_mix() 3762{ 3763 AudioBufferProvider::Buffer buffer; 3764 size_t frameCount = mFrameCount; 3765 int8_t *curBuf = (int8_t *)mMixBuffer; 3766 // output audio to hardware 3767 while (frameCount) { 3768 buffer.frameCount = frameCount; 3769 mActiveTrack->getNextBuffer(&buffer); 3770 if (CC_UNLIKELY(buffer.raw == NULL)) { 3771 memset(curBuf, 0, frameCount * mFrameSize); 3772 break; 3773 } 3774 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3775 frameCount -= buffer.frameCount; 3776 curBuf += buffer.frameCount * mFrameSize; 3777 mActiveTrack->releaseBuffer(&buffer); 3778 } 3779 sleepTime = 0; 3780 standbyTime = systemTime() + standbyDelay; 3781 mActiveTrack.clear(); 3782 3783} 3784 3785void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3786{ 3787 if (sleepTime == 0) { 3788 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3789 sleepTime = activeSleepTime; 3790 } else { 3791 sleepTime = idleSleepTime; 3792 } 3793 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3794 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3795 sleepTime = 0; 3796 } 3797} 3798 3799// getTrackName_l() must be called with ThreadBase::mLock held 3800int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3801{ 3802 return 0; 3803} 3804 3805// deleteTrackName_l() must be called with ThreadBase::mLock held 3806void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3807{ 3808} 3809 3810// checkForNewParameters_l() must be called with ThreadBase::mLock held 3811bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3812{ 3813 bool reconfig = false; 3814 3815 while (!mNewParameters.isEmpty()) { 3816 status_t status = NO_ERROR; 3817 String8 keyValuePair = mNewParameters[0]; 3818 AudioParameter param = AudioParameter(keyValuePair); 3819 int value; 3820 3821 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3822 // do not accept frame count changes if tracks are open as the track buffer 3823 // size depends on frame count and correct behavior would not be garantied 3824 // if frame count is changed after track creation 3825 if (!mTracks.isEmpty()) { 3826 status = INVALID_OPERATION; 3827 } else { 3828 reconfig = true; 3829 } 3830 } 3831 if (status == NO_ERROR) { 3832 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3833 keyValuePair.string()); 3834 if (!mStandby && status == INVALID_OPERATION) { 3835 mOutput->stream->common.standby(&mOutput->stream->common); 3836 mStandby = true; 3837 mBytesWritten = 0; 3838 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3839 keyValuePair.string()); 3840 } 3841 if (status == NO_ERROR && reconfig) { 3842 readOutputParameters(); 3843 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3844 } 3845 } 3846 3847 mNewParameters.removeAt(0); 3848 3849 mParamStatus = status; 3850 mParamCond.signal(); 3851 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3852 // already timed out waiting for the status and will never signal the condition. 3853 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3854 } 3855 return reconfig; 3856} 3857 3858uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3859{ 3860 uint32_t time; 3861 if (audio_is_linear_pcm(mFormat)) { 3862 time = PlaybackThread::activeSleepTimeUs(); 3863 } else { 3864 time = 10000; 3865 } 3866 return time; 3867} 3868 3869uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3870{ 3871 uint32_t time; 3872 if (audio_is_linear_pcm(mFormat)) { 3873 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3874 } else { 3875 time = 10000; 3876 } 3877 return time; 3878} 3879 3880uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3881{ 3882 uint32_t time; 3883 if (audio_is_linear_pcm(mFormat)) { 3884 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3885 } else { 3886 time = 10000; 3887 } 3888 return time; 3889} 3890 3891void AudioFlinger::DirectOutputThread::cacheParameters_l() 3892{ 3893 PlaybackThread::cacheParameters_l(); 3894 3895 // use shorter standby delay as on normal output to release 3896 // hardware resources as soon as possible 3897 standbyDelay = microseconds(activeSleepTime*2); 3898} 3899 3900// ---------------------------------------------------------------------------- 3901 3902AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3903 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3904 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3905 mWaitTimeMs(UINT_MAX) 3906{ 3907 addOutputTrack(mainThread); 3908} 3909 3910AudioFlinger::DuplicatingThread::~DuplicatingThread() 3911{ 3912 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3913 mOutputTracks[i]->destroy(); 3914 } 3915} 3916 3917void AudioFlinger::DuplicatingThread::threadLoop_mix() 3918{ 3919 // mix buffers... 3920 if (outputsReady(outputTracks)) { 3921 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3922 } else { 3923 memset(mMixBuffer, 0, mixBufferSize); 3924 } 3925 sleepTime = 0; 3926 writeFrames = mNormalFrameCount; 3927 standbyTime = systemTime() + standbyDelay; 3928} 3929 3930void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3931{ 3932 if (sleepTime == 0) { 3933 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3934 sleepTime = activeSleepTime; 3935 } else { 3936 sleepTime = idleSleepTime; 3937 } 3938 } else if (mBytesWritten != 0) { 3939 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3940 writeFrames = mNormalFrameCount; 3941 memset(mMixBuffer, 0, mixBufferSize); 3942 } else { 3943 // flush remaining overflow buffers in output tracks 3944 writeFrames = 0; 3945 } 3946 sleepTime = 0; 3947 } 3948} 3949 3950void AudioFlinger::DuplicatingThread::threadLoop_write() 3951{ 3952 for (size_t i = 0; i < outputTracks.size(); i++) { 3953 outputTracks[i]->write(mMixBuffer, writeFrames); 3954 } 3955 mBytesWritten += mixBufferSize; 3956} 3957 3958void AudioFlinger::DuplicatingThread::threadLoop_standby() 3959{ 3960 // DuplicatingThread implements standby by stopping all tracks 3961 for (size_t i = 0; i < outputTracks.size(); i++) { 3962 outputTracks[i]->stop(); 3963 } 3964} 3965 3966void AudioFlinger::DuplicatingThread::saveOutputTracks() 3967{ 3968 outputTracks = mOutputTracks; 3969} 3970 3971void AudioFlinger::DuplicatingThread::clearOutputTracks() 3972{ 3973 outputTracks.clear(); 3974} 3975 3976void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3977{ 3978 Mutex::Autolock _l(mLock); 3979 // FIXME explain this formula 3980 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3981 OutputTrack *outputTrack = new OutputTrack(thread, 3982 this, 3983 mSampleRate, 3984 mFormat, 3985 mChannelMask, 3986 frameCount); 3987 if (outputTrack->cblk() != NULL) { 3988 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3989 mOutputTracks.add(outputTrack); 3990 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3991 updateWaitTime_l(); 3992 } 3993} 3994 3995void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3996{ 3997 Mutex::Autolock _l(mLock); 3998 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3999 if (mOutputTracks[i]->thread() == thread) { 4000 mOutputTracks[i]->destroy(); 4001 mOutputTracks.removeAt(i); 4002 updateWaitTime_l(); 4003 return; 4004 } 4005 } 4006 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4007} 4008 4009// caller must hold mLock 4010void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4011{ 4012 mWaitTimeMs = UINT_MAX; 4013 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4014 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4015 if (strong != 0) { 4016 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4017 if (waitTimeMs < mWaitTimeMs) { 4018 mWaitTimeMs = waitTimeMs; 4019 } 4020 } 4021 } 4022} 4023 4024 4025bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4026{ 4027 for (size_t i = 0; i < outputTracks.size(); i++) { 4028 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4029 if (thread == 0) { 4030 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4031 return false; 4032 } 4033 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4034 // see note at standby() declaration 4035 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4036 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4037 return false; 4038 } 4039 } 4040 return true; 4041} 4042 4043uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4044{ 4045 return (mWaitTimeMs * 1000) / 2; 4046} 4047 4048void AudioFlinger::DuplicatingThread::cacheParameters_l() 4049{ 4050 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4051 updateWaitTime_l(); 4052 4053 MixerThread::cacheParameters_l(); 4054} 4055 4056// ---------------------------------------------------------------------------- 4057 4058// TrackBase constructor must be called with AudioFlinger::mLock held 4059AudioFlinger::ThreadBase::TrackBase::TrackBase( 4060 ThreadBase *thread, 4061 const sp<Client>& client, 4062 uint32_t sampleRate, 4063 audio_format_t format, 4064 audio_channel_mask_t channelMask, 4065 int frameCount, 4066 const sp<IMemory>& sharedBuffer, 4067 int sessionId) 4068 : RefBase(), 4069 mThread(thread), 4070 mClient(client), 4071 mCblk(NULL), 4072 // mBuffer 4073 // mBufferEnd 4074 mFrameCount(0), 4075 mState(IDLE), 4076 mSampleRate(sampleRate), 4077 mFormat(format), 4078 mStepServerFailed(false), 4079 mSessionId(sessionId) 4080 // mChannelCount 4081 // mChannelMask 4082{ 4083 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4084 4085 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4086 size_t size = sizeof(audio_track_cblk_t); 4087 uint8_t channelCount = popcount(channelMask); 4088 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4089 if (sharedBuffer == 0) { 4090 size += bufferSize; 4091 } 4092 4093 if (client != NULL) { 4094 mCblkMemory = client->heap()->allocate(size); 4095 if (mCblkMemory != 0) { 4096 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4097 if (mCblk != NULL) { // construct the shared structure in-place. 4098 new(mCblk) audio_track_cblk_t(); 4099 // clear all buffers 4100 mCblk->frameCount = frameCount; 4101 mCblk->sampleRate = sampleRate; 4102// uncomment the following lines to quickly test 32-bit wraparound 4103// mCblk->user = 0xffff0000; 4104// mCblk->server = 0xffff0000; 4105// mCblk->userBase = 0xffff0000; 4106// mCblk->serverBase = 0xffff0000; 4107 mChannelCount = channelCount; 4108 mChannelMask = channelMask; 4109 if (sharedBuffer == 0) { 4110 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4111 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4112 // Force underrun condition to avoid false underrun callback until first data is 4113 // written to buffer (other flags are cleared) 4114 mCblk->flags = CBLK_UNDERRUN_ON; 4115 } else { 4116 mBuffer = sharedBuffer->pointer(); 4117 } 4118 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4119 } 4120 } else { 4121 ALOGE("not enough memory for AudioTrack size=%u", size); 4122 client->heap()->dump("AudioTrack"); 4123 return; 4124 } 4125 } else { 4126 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4127 // construct the shared structure in-place. 4128 new(mCblk) audio_track_cblk_t(); 4129 // clear all buffers 4130 mCblk->frameCount = frameCount; 4131 mCblk->sampleRate = sampleRate; 4132// uncomment the following lines to quickly test 32-bit wraparound 4133// mCblk->user = 0xffff0000; 4134// mCblk->server = 0xffff0000; 4135// mCblk->userBase = 0xffff0000; 4136// mCblk->serverBase = 0xffff0000; 4137 mChannelCount = channelCount; 4138 mChannelMask = channelMask; 4139 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4140 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4141 // Force underrun condition to avoid false underrun callback until first data is 4142 // written to buffer (other flags are cleared) 4143 mCblk->flags = CBLK_UNDERRUN_ON; 4144 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4145 } 4146} 4147 4148AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4149{ 4150 if (mCblk != NULL) { 4151 if (mClient == 0) { 4152 delete mCblk; 4153 } else { 4154 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4155 } 4156 } 4157 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4158 if (mClient != 0) { 4159 // Client destructor must run with AudioFlinger mutex locked 4160 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4161 // If the client's reference count drops to zero, the associated destructor 4162 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4163 // relying on the automatic clear() at end of scope. 4164 mClient.clear(); 4165 } 4166} 4167 4168// AudioBufferProvider interface 4169// getNextBuffer() = 0; 4170// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4171void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4172{ 4173 buffer->raw = NULL; 4174 mFrameCount = buffer->frameCount; 4175 // FIXME See note at getNextBuffer() 4176 (void) step(); // ignore return value of step() 4177 buffer->frameCount = 0; 4178} 4179 4180bool AudioFlinger::ThreadBase::TrackBase::step() { 4181 bool result; 4182 audio_track_cblk_t* cblk = this->cblk(); 4183 4184 result = cblk->stepServer(mFrameCount); 4185 if (!result) { 4186 ALOGV("stepServer failed acquiring cblk mutex"); 4187 mStepServerFailed = true; 4188 } 4189 return result; 4190} 4191 4192void AudioFlinger::ThreadBase::TrackBase::reset() { 4193 audio_track_cblk_t* cblk = this->cblk(); 4194 4195 cblk->user = 0; 4196 cblk->server = 0; 4197 cblk->userBase = 0; 4198 cblk->serverBase = 0; 4199 mStepServerFailed = false; 4200 ALOGV("TrackBase::reset"); 4201} 4202 4203int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4204 return (int)mCblk->sampleRate; 4205} 4206 4207void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4208 audio_track_cblk_t* cblk = this->cblk(); 4209 size_t frameSize = cblk->frameSize; 4210 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4211 int8_t *bufferEnd = bufferStart + frames * frameSize; 4212 4213 // Check validity of returned pointer in case the track control block would have been corrupted. 4214 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4215 "TrackBase::getBuffer buffer out of range:\n" 4216 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4217 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4218 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4219 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4220 4221 return bufferStart; 4222} 4223 4224status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4225{ 4226 mSyncEvents.add(event); 4227 return NO_ERROR; 4228} 4229 4230// ---------------------------------------------------------------------------- 4231 4232// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4233AudioFlinger::PlaybackThread::Track::Track( 4234 PlaybackThread *thread, 4235 const sp<Client>& client, 4236 audio_stream_type_t streamType, 4237 uint32_t sampleRate, 4238 audio_format_t format, 4239 audio_channel_mask_t channelMask, 4240 int frameCount, 4241 const sp<IMemory>& sharedBuffer, 4242 int sessionId, 4243 IAudioFlinger::track_flags_t flags) 4244 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4245 mMute(false), 4246 mFillingUpStatus(FS_INVALID), 4247 // mRetryCount initialized later when needed 4248 mSharedBuffer(sharedBuffer), 4249 mStreamType(streamType), 4250 mName(-1), // see note below 4251 mMainBuffer(thread->mixBuffer()), 4252 mAuxBuffer(NULL), 4253 mAuxEffectId(0), mHasVolumeController(false), 4254 mPresentationCompleteFrames(0), 4255 mFlags(flags), 4256 mFastIndex(-1), 4257 mUnderrunCount(0), 4258 mCachedVolume(1.0) 4259{ 4260 if (mCblk != NULL) { 4261 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4262 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4263 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4264 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4265 mName = thread->getTrackName_l(channelMask); 4266 mCblk->mName = mName; 4267 if (mName < 0) { 4268 ALOGE("no more track names available"); 4269 return; 4270 } 4271 // only allocate a fast track index if we were able to allocate a normal track name 4272 if (flags & IAudioFlinger::TRACK_FAST) { 4273 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4274 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4275 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4276 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4277 // FIXME This is too eager. We allocate a fast track index before the 4278 // fast track becomes active. Since fast tracks are a scarce resource, 4279 // this means we are potentially denying other more important fast tracks from 4280 // being created. It would be better to allocate the index dynamically. 4281 mFastIndex = i; 4282 mCblk->mName = i; 4283 // Read the initial underruns because this field is never cleared by the fast mixer 4284 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4285 thread->mFastTrackAvailMask &= ~(1 << i); 4286 } 4287 } 4288 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4289} 4290 4291AudioFlinger::PlaybackThread::Track::~Track() 4292{ 4293 ALOGV("PlaybackThread::Track destructor"); 4294 sp<ThreadBase> thread = mThread.promote(); 4295 if (thread != 0) { 4296 Mutex::Autolock _l(thread->mLock); 4297 mState = TERMINATED; 4298 } 4299} 4300 4301void AudioFlinger::PlaybackThread::Track::destroy() 4302{ 4303 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4304 // by removing it from mTracks vector, so there is a risk that this Tracks's 4305 // destructor is called. As the destructor needs to lock mLock, 4306 // we must acquire a strong reference on this Track before locking mLock 4307 // here so that the destructor is called only when exiting this function. 4308 // On the other hand, as long as Track::destroy() is only called by 4309 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4310 // this Track with its member mTrack. 4311 sp<Track> keep(this); 4312 { // scope for mLock 4313 sp<ThreadBase> thread = mThread.promote(); 4314 if (thread != 0) { 4315 if (!isOutputTrack()) { 4316 if (mState == ACTIVE || mState == RESUMING) { 4317 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4318 4319#ifdef ADD_BATTERY_DATA 4320 // to track the speaker usage 4321 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4322#endif 4323 } 4324 AudioSystem::releaseOutput(thread->id()); 4325 } 4326 Mutex::Autolock _l(thread->mLock); 4327 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4328 playbackThread->destroyTrack_l(this); 4329 } 4330 } 4331} 4332 4333/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4334{ 4335 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4336 " Server User Main buf Aux Buf Flags Underruns\n"); 4337} 4338 4339void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4340{ 4341 uint32_t vlr = mCblk->getVolumeLR(); 4342 if (isFastTrack()) { 4343 sprintf(buffer, " F %2d", mFastIndex); 4344 } else { 4345 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4346 } 4347 track_state state = mState; 4348 char stateChar; 4349 switch (state) { 4350 case IDLE: 4351 stateChar = 'I'; 4352 break; 4353 case TERMINATED: 4354 stateChar = 'T'; 4355 break; 4356 case STOPPING_1: 4357 stateChar = 's'; 4358 break; 4359 case STOPPING_2: 4360 stateChar = '5'; 4361 break; 4362 case STOPPED: 4363 stateChar = 'S'; 4364 break; 4365 case RESUMING: 4366 stateChar = 'R'; 4367 break; 4368 case ACTIVE: 4369 stateChar = 'A'; 4370 break; 4371 case PAUSING: 4372 stateChar = 'p'; 4373 break; 4374 case PAUSED: 4375 stateChar = 'P'; 4376 break; 4377 case FLUSHED: 4378 stateChar = 'F'; 4379 break; 4380 default: 4381 stateChar = '?'; 4382 break; 4383 } 4384 char nowInUnderrun; 4385 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4386 case UNDERRUN_FULL: 4387 nowInUnderrun = ' '; 4388 break; 4389 case UNDERRUN_PARTIAL: 4390 nowInUnderrun = '<'; 4391 break; 4392 case UNDERRUN_EMPTY: 4393 nowInUnderrun = '*'; 4394 break; 4395 default: 4396 nowInUnderrun = '?'; 4397 break; 4398 } 4399 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4400 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4401 (mClient == 0) ? getpid_cached : mClient->pid(), 4402 mStreamType, 4403 mFormat, 4404 mChannelMask, 4405 mSessionId, 4406 mFrameCount, 4407 mCblk->frameCount, 4408 stateChar, 4409 mMute, 4410 mFillingUpStatus, 4411 mCblk->sampleRate, 4412 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4413 20.0 * log10((vlr >> 16) / 4096.0), 4414 mCblk->server, 4415 mCblk->user, 4416 (int)mMainBuffer, 4417 (int)mAuxBuffer, 4418 mCblk->flags, 4419 mUnderrunCount, 4420 nowInUnderrun); 4421} 4422 4423// AudioBufferProvider interface 4424status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4425 AudioBufferProvider::Buffer* buffer, int64_t pts) 4426{ 4427 audio_track_cblk_t* cblk = this->cblk(); 4428 uint32_t framesReady; 4429 uint32_t framesReq = buffer->frameCount; 4430 4431 // Check if last stepServer failed, try to step now 4432 if (mStepServerFailed) { 4433 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4434 // Since the fast mixer is higher priority than client callback thread, 4435 // it does not result in priority inversion for client. 4436 // But a non-blocking solution would be preferable to avoid 4437 // fast mixer being unable to tryLock(), and 4438 // to avoid the extra context switches if the client wakes up, 4439 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4440 if (!step()) goto getNextBuffer_exit; 4441 ALOGV("stepServer recovered"); 4442 mStepServerFailed = false; 4443 } 4444 4445 // FIXME Same as above 4446 framesReady = cblk->framesReady(); 4447 4448 if (CC_LIKELY(framesReady)) { 4449 uint32_t s = cblk->server; 4450 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4451 4452 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4453 if (framesReq > framesReady) { 4454 framesReq = framesReady; 4455 } 4456 if (framesReq > bufferEnd - s) { 4457 framesReq = bufferEnd - s; 4458 } 4459 4460 buffer->raw = getBuffer(s, framesReq); 4461 buffer->frameCount = framesReq; 4462 return NO_ERROR; 4463 } 4464 4465getNextBuffer_exit: 4466 buffer->raw = NULL; 4467 buffer->frameCount = 0; 4468 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4469 return NOT_ENOUGH_DATA; 4470} 4471 4472// Note that framesReady() takes a mutex on the control block using tryLock(). 4473// This could result in priority inversion if framesReady() is called by the normal mixer, 4474// as the normal mixer thread runs at lower 4475// priority than the client's callback thread: there is a short window within framesReady() 4476// during which the normal mixer could be preempted, and the client callback would block. 4477// Another problem can occur if framesReady() is called by the fast mixer: 4478// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4479// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4480size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4481 return mCblk->framesReady(); 4482} 4483 4484// Don't call for fast tracks; the framesReady() could result in priority inversion 4485bool AudioFlinger::PlaybackThread::Track::isReady() const { 4486 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4487 4488 if (framesReady() >= mCblk->frameCount || 4489 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4490 mFillingUpStatus = FS_FILLED; 4491 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4492 return true; 4493 } 4494 return false; 4495} 4496 4497status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4498 int triggerSession) 4499{ 4500 status_t status = NO_ERROR; 4501 ALOGV("start(%d), calling pid %d session %d", 4502 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4503 4504 sp<ThreadBase> thread = mThread.promote(); 4505 if (thread != 0) { 4506 Mutex::Autolock _l(thread->mLock); 4507 track_state state = mState; 4508 // here the track could be either new, or restarted 4509 // in both cases "unstop" the track 4510 if (mState == PAUSED) { 4511 mState = TrackBase::RESUMING; 4512 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4513 } else { 4514 mState = TrackBase::ACTIVE; 4515 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4516 } 4517 4518 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4519 thread->mLock.unlock(); 4520 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4521 thread->mLock.lock(); 4522 4523#ifdef ADD_BATTERY_DATA 4524 // to track the speaker usage 4525 if (status == NO_ERROR) { 4526 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4527 } 4528#endif 4529 } 4530 if (status == NO_ERROR) { 4531 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4532 playbackThread->addTrack_l(this); 4533 } else { 4534 mState = state; 4535 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4536 } 4537 } else { 4538 status = BAD_VALUE; 4539 } 4540 return status; 4541} 4542 4543void AudioFlinger::PlaybackThread::Track::stop() 4544{ 4545 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4546 sp<ThreadBase> thread = mThread.promote(); 4547 if (thread != 0) { 4548 Mutex::Autolock _l(thread->mLock); 4549 track_state state = mState; 4550 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4551 // If the track is not active (PAUSED and buffers full), flush buffers 4552 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4553 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4554 reset(); 4555 mState = STOPPED; 4556 } else if (!isFastTrack()) { 4557 mState = STOPPED; 4558 } else { 4559 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4560 // and then to STOPPED and reset() when presentation is complete 4561 mState = STOPPING_1; 4562 } 4563 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4564 } 4565 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4566 thread->mLock.unlock(); 4567 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4568 thread->mLock.lock(); 4569 4570#ifdef ADD_BATTERY_DATA 4571 // to track the speaker usage 4572 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4573#endif 4574 } 4575 } 4576} 4577 4578void AudioFlinger::PlaybackThread::Track::pause() 4579{ 4580 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4581 sp<ThreadBase> thread = mThread.promote(); 4582 if (thread != 0) { 4583 Mutex::Autolock _l(thread->mLock); 4584 if (mState == ACTIVE || mState == RESUMING) { 4585 mState = PAUSING; 4586 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4587 if (!isOutputTrack()) { 4588 thread->mLock.unlock(); 4589 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4590 thread->mLock.lock(); 4591 4592#ifdef ADD_BATTERY_DATA 4593 // to track the speaker usage 4594 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4595#endif 4596 } 4597 } 4598 } 4599} 4600 4601void AudioFlinger::PlaybackThread::Track::flush() 4602{ 4603 ALOGV("flush(%d)", mName); 4604 sp<ThreadBase> thread = mThread.promote(); 4605 if (thread != 0) { 4606 Mutex::Autolock _l(thread->mLock); 4607 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4608 mState != PAUSING) { 4609 return; 4610 } 4611 // No point remaining in PAUSED state after a flush => go to 4612 // FLUSHED state 4613 mState = FLUSHED; 4614 // do not reset the track if it is still in the process of being stopped or paused. 4615 // this will be done by prepareTracks_l() when the track is stopped. 4616 // prepareTracks_l() will see mState == FLUSHED, then 4617 // remove from active track list, reset(), and trigger presentation complete 4618 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4619 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4620 reset(); 4621 } 4622 } 4623} 4624 4625void AudioFlinger::PlaybackThread::Track::reset() 4626{ 4627 // Do not reset twice to avoid discarding data written just after a flush and before 4628 // the audioflinger thread detects the track is stopped. 4629 if (!mResetDone) { 4630 TrackBase::reset(); 4631 // Force underrun condition to avoid false underrun callback until first data is 4632 // written to buffer 4633 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4634 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4635 mFillingUpStatus = FS_FILLING; 4636 mResetDone = true; 4637 if (mState == FLUSHED) { 4638 mState = IDLE; 4639 } 4640 } 4641} 4642 4643void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4644{ 4645 mMute = muted; 4646} 4647 4648status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4649{ 4650 status_t status = DEAD_OBJECT; 4651 sp<ThreadBase> thread = mThread.promote(); 4652 if (thread != 0) { 4653 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4654 sp<AudioFlinger> af = mClient->audioFlinger(); 4655 4656 Mutex::Autolock _l(af->mLock); 4657 4658 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4659 4660 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4661 Mutex::Autolock _dl(playbackThread->mLock); 4662 Mutex::Autolock _sl(srcThread->mLock); 4663 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4664 if (chain == 0) { 4665 return INVALID_OPERATION; 4666 } 4667 4668 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4669 if (effect == 0) { 4670 return INVALID_OPERATION; 4671 } 4672 srcThread->removeEffect_l(effect); 4673 playbackThread->addEffect_l(effect); 4674 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4675 if (effect->state() == EffectModule::ACTIVE || 4676 effect->state() == EffectModule::STOPPING) { 4677 effect->start(); 4678 } 4679 4680 sp<EffectChain> dstChain = effect->chain().promote(); 4681 if (dstChain == 0) { 4682 srcThread->addEffect_l(effect); 4683 return INVALID_OPERATION; 4684 } 4685 AudioSystem::unregisterEffect(effect->id()); 4686 AudioSystem::registerEffect(&effect->desc(), 4687 srcThread->id(), 4688 dstChain->strategy(), 4689 AUDIO_SESSION_OUTPUT_MIX, 4690 effect->id()); 4691 } 4692 status = playbackThread->attachAuxEffect(this, EffectId); 4693 } 4694 return status; 4695} 4696 4697void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4698{ 4699 mAuxEffectId = EffectId; 4700 mAuxBuffer = buffer; 4701} 4702 4703bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4704 size_t audioHalFrames) 4705{ 4706 // a track is considered presented when the total number of frames written to audio HAL 4707 // corresponds to the number of frames written when presentationComplete() is called for the 4708 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4709 if (mPresentationCompleteFrames == 0) { 4710 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4711 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4712 mPresentationCompleteFrames, audioHalFrames); 4713 } 4714 if (framesWritten >= mPresentationCompleteFrames) { 4715 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4716 mSessionId, framesWritten); 4717 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4718 return true; 4719 } 4720 return false; 4721} 4722 4723void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4724{ 4725 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4726 if (mSyncEvents[i]->type() == type) { 4727 mSyncEvents[i]->trigger(); 4728 mSyncEvents.removeAt(i); 4729 i--; 4730 } 4731 } 4732} 4733 4734// implement VolumeBufferProvider interface 4735 4736uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4737{ 4738 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4739 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4740 uint32_t vlr = mCblk->getVolumeLR(); 4741 uint32_t vl = vlr & 0xFFFF; 4742 uint32_t vr = vlr >> 16; 4743 // track volumes come from shared memory, so can't be trusted and must be clamped 4744 if (vl > MAX_GAIN_INT) { 4745 vl = MAX_GAIN_INT; 4746 } 4747 if (vr > MAX_GAIN_INT) { 4748 vr = MAX_GAIN_INT; 4749 } 4750 // now apply the cached master volume and stream type volume; 4751 // this is trusted but lacks any synchronization or barrier so may be stale 4752 float v = mCachedVolume; 4753 vl *= v; 4754 vr *= v; 4755 // re-combine into U4.16 4756 vlr = (vr << 16) | (vl & 0xFFFF); 4757 // FIXME look at mute, pause, and stop flags 4758 return vlr; 4759} 4760 4761status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4762{ 4763 if (mState == TERMINATED || mState == PAUSED || 4764 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4765 (mState == STOPPED)))) { 4766 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4767 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4768 event->cancel(); 4769 return INVALID_OPERATION; 4770 } 4771 TrackBase::setSyncEvent(event); 4772 return NO_ERROR; 4773} 4774 4775// timed audio tracks 4776 4777sp<AudioFlinger::PlaybackThread::TimedTrack> 4778AudioFlinger::PlaybackThread::TimedTrack::create( 4779 PlaybackThread *thread, 4780 const sp<Client>& client, 4781 audio_stream_type_t streamType, 4782 uint32_t sampleRate, 4783 audio_format_t format, 4784 audio_channel_mask_t channelMask, 4785 int frameCount, 4786 const sp<IMemory>& sharedBuffer, 4787 int sessionId) { 4788 if (!client->reserveTimedTrack()) 4789 return 0; 4790 4791 return new TimedTrack( 4792 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4793 sharedBuffer, sessionId); 4794} 4795 4796AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4797 PlaybackThread *thread, 4798 const sp<Client>& client, 4799 audio_stream_type_t streamType, 4800 uint32_t sampleRate, 4801 audio_format_t format, 4802 audio_channel_mask_t channelMask, 4803 int frameCount, 4804 const sp<IMemory>& sharedBuffer, 4805 int sessionId) 4806 : Track(thread, client, streamType, sampleRate, format, channelMask, 4807 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4808 mQueueHeadInFlight(false), 4809 mTrimQueueHeadOnRelease(false), 4810 mFramesPendingInQueue(0), 4811 mTimedSilenceBuffer(NULL), 4812 mTimedSilenceBufferSize(0), 4813 mTimedAudioOutputOnTime(false), 4814 mMediaTimeTransformValid(false) 4815{ 4816 LocalClock lc; 4817 mLocalTimeFreq = lc.getLocalFreq(); 4818 4819 mLocalTimeToSampleTransform.a_zero = 0; 4820 mLocalTimeToSampleTransform.b_zero = 0; 4821 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4822 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4823 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4824 &mLocalTimeToSampleTransform.a_to_b_denom); 4825 4826 mMediaTimeToSampleTransform.a_zero = 0; 4827 mMediaTimeToSampleTransform.b_zero = 0; 4828 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4829 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4830 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4831 &mMediaTimeToSampleTransform.a_to_b_denom); 4832} 4833 4834AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4835 mClient->releaseTimedTrack(); 4836 delete [] mTimedSilenceBuffer; 4837} 4838 4839status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4840 size_t size, sp<IMemory>* buffer) { 4841 4842 Mutex::Autolock _l(mTimedBufferQueueLock); 4843 4844 trimTimedBufferQueue_l(); 4845 4846 // lazily initialize the shared memory heap for timed buffers 4847 if (mTimedMemoryDealer == NULL) { 4848 const int kTimedBufferHeapSize = 512 << 10; 4849 4850 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4851 "AudioFlingerTimed"); 4852 if (mTimedMemoryDealer == NULL) 4853 return NO_MEMORY; 4854 } 4855 4856 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4857 if (newBuffer == NULL) { 4858 newBuffer = mTimedMemoryDealer->allocate(size); 4859 if (newBuffer == NULL) 4860 return NO_MEMORY; 4861 } 4862 4863 *buffer = newBuffer; 4864 return NO_ERROR; 4865} 4866 4867// caller must hold mTimedBufferQueueLock 4868void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4869 int64_t mediaTimeNow; 4870 { 4871 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4872 if (!mMediaTimeTransformValid) 4873 return; 4874 4875 int64_t targetTimeNow; 4876 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4877 ? mCCHelper.getCommonTime(&targetTimeNow) 4878 : mCCHelper.getLocalTime(&targetTimeNow); 4879 4880 if (OK != res) 4881 return; 4882 4883 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4884 &mediaTimeNow)) { 4885 return; 4886 } 4887 } 4888 4889 size_t trimEnd; 4890 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4891 int64_t bufEnd; 4892 4893 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4894 // We have a next buffer. Just use its PTS as the PTS of the frame 4895 // following the last frame in this buffer. If the stream is sparse 4896 // (ie, there are deliberate gaps left in the stream which should be 4897 // filled with silence by the TimedAudioTrack), then this can result 4898 // in one extra buffer being left un-trimmed when it could have 4899 // been. In general, this is not typical, and we would rather 4900 // optimized away the TS calculation below for the more common case 4901 // where PTSes are contiguous. 4902 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4903 } else { 4904 // We have no next buffer. Compute the PTS of the frame following 4905 // the last frame in this buffer by computing the duration of of 4906 // this frame in media time units and adding it to the PTS of the 4907 // buffer. 4908 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4909 / mCblk->frameSize; 4910 4911 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4912 &bufEnd)) { 4913 ALOGE("Failed to convert frame count of %lld to media time" 4914 " duration" " (scale factor %d/%u) in %s", 4915 frameCount, 4916 mMediaTimeToSampleTransform.a_to_b_numer, 4917 mMediaTimeToSampleTransform.a_to_b_denom, 4918 __PRETTY_FUNCTION__); 4919 break; 4920 } 4921 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4922 } 4923 4924 if (bufEnd > mediaTimeNow) 4925 break; 4926 4927 // Is the buffer we want to use in the middle of a mix operation right 4928 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4929 // from the mixer which should be coming back shortly. 4930 if (!trimEnd && mQueueHeadInFlight) { 4931 mTrimQueueHeadOnRelease = true; 4932 } 4933 } 4934 4935 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4936 if (trimStart < trimEnd) { 4937 // Update the bookkeeping for framesReady() 4938 for (size_t i = trimStart; i < trimEnd; ++i) { 4939 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4940 } 4941 4942 // Now actually remove the buffers from the queue. 4943 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4944 } 4945} 4946 4947void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4948 const char* logTag) { 4949 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4950 "%s called (reason \"%s\"), but timed buffer queue has no" 4951 " elements to trim.", __FUNCTION__, logTag); 4952 4953 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4954 mTimedBufferQueue.removeAt(0); 4955} 4956 4957void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4958 const TimedBuffer& buf, 4959 const char* logTag) { 4960 uint32_t bufBytes = buf.buffer()->size(); 4961 uint32_t consumedAlready = buf.position(); 4962 4963 ALOG_ASSERT(consumedAlready <= bufBytes, 4964 "Bad bookkeeping while updating frames pending. Timed buffer is" 4965 " only %u bytes long, but claims to have consumed %u" 4966 " bytes. (update reason: \"%s\")", 4967 bufBytes, consumedAlready, logTag); 4968 4969 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4970 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4971 "Bad bookkeeping while updating frames pending. Should have at" 4972 " least %u queued frames, but we think we have only %u. (update" 4973 " reason: \"%s\")", 4974 bufFrames, mFramesPendingInQueue, logTag); 4975 4976 mFramesPendingInQueue -= bufFrames; 4977} 4978 4979status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4980 const sp<IMemory>& buffer, int64_t pts) { 4981 4982 { 4983 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4984 if (!mMediaTimeTransformValid) 4985 return INVALID_OPERATION; 4986 } 4987 4988 Mutex::Autolock _l(mTimedBufferQueueLock); 4989 4990 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4991 mFramesPendingInQueue += bufFrames; 4992 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4993 4994 return NO_ERROR; 4995} 4996 4997status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4998 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4999 5000 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5001 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5002 target); 5003 5004 if (!(target == TimedAudioTrack::LOCAL_TIME || 5005 target == TimedAudioTrack::COMMON_TIME)) { 5006 return BAD_VALUE; 5007 } 5008 5009 Mutex::Autolock lock(mMediaTimeTransformLock); 5010 mMediaTimeTransform = xform; 5011 mMediaTimeTransformTarget = target; 5012 mMediaTimeTransformValid = true; 5013 5014 return NO_ERROR; 5015} 5016 5017#define min(a, b) ((a) < (b) ? (a) : (b)) 5018 5019// implementation of getNextBuffer for tracks whose buffers have timestamps 5020status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5021 AudioBufferProvider::Buffer* buffer, int64_t pts) 5022{ 5023 if (pts == AudioBufferProvider::kInvalidPTS) { 5024 buffer->raw = NULL; 5025 buffer->frameCount = 0; 5026 mTimedAudioOutputOnTime = false; 5027 return INVALID_OPERATION; 5028 } 5029 5030 Mutex::Autolock _l(mTimedBufferQueueLock); 5031 5032 ALOG_ASSERT(!mQueueHeadInFlight, 5033 "getNextBuffer called without releaseBuffer!"); 5034 5035 while (true) { 5036 5037 // if we have no timed buffers, then fail 5038 if (mTimedBufferQueue.isEmpty()) { 5039 buffer->raw = NULL; 5040 buffer->frameCount = 0; 5041 return NOT_ENOUGH_DATA; 5042 } 5043 5044 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5045 5046 // calculate the PTS of the head of the timed buffer queue expressed in 5047 // local time 5048 int64_t headLocalPTS; 5049 { 5050 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5051 5052 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5053 5054 if (mMediaTimeTransform.a_to_b_denom == 0) { 5055 // the transform represents a pause, so yield silence 5056 timedYieldSilence_l(buffer->frameCount, buffer); 5057 return NO_ERROR; 5058 } 5059 5060 int64_t transformedPTS; 5061 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5062 &transformedPTS)) { 5063 // the transform failed. this shouldn't happen, but if it does 5064 // then just drop this buffer 5065 ALOGW("timedGetNextBuffer transform failed"); 5066 buffer->raw = NULL; 5067 buffer->frameCount = 0; 5068 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5069 return NO_ERROR; 5070 } 5071 5072 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5073 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5074 &headLocalPTS)) { 5075 buffer->raw = NULL; 5076 buffer->frameCount = 0; 5077 return INVALID_OPERATION; 5078 } 5079 } else { 5080 headLocalPTS = transformedPTS; 5081 } 5082 } 5083 5084 // adjust the head buffer's PTS to reflect the portion of the head buffer 5085 // that has already been consumed 5086 int64_t effectivePTS = headLocalPTS + 5087 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5088 5089 // Calculate the delta in samples between the head of the input buffer 5090 // queue and the start of the next output buffer that will be written. 5091 // If the transformation fails because of over or underflow, it means 5092 // that the sample's position in the output stream is so far out of 5093 // whack that it should just be dropped. 5094 int64_t sampleDelta; 5095 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5096 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5097 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5098 " mix"); 5099 continue; 5100 } 5101 if (!mLocalTimeToSampleTransform.doForwardTransform( 5102 (effectivePTS - pts) << 32, &sampleDelta)) { 5103 ALOGV("*** too late during sample rate transform: dropped buffer"); 5104 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5105 continue; 5106 } 5107 5108 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5109 " sampleDelta=[%d.%08x]", 5110 head.pts(), head.position(), pts, 5111 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5112 + (sampleDelta >> 32)), 5113 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5114 5115 // if the delta between the ideal placement for the next input sample and 5116 // the current output position is within this threshold, then we will 5117 // concatenate the next input samples to the previous output 5118 const int64_t kSampleContinuityThreshold = 5119 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5120 5121 // if this is the first buffer of audio that we're emitting from this track 5122 // then it should be almost exactly on time. 5123 const int64_t kSampleStartupThreshold = 1LL << 32; 5124 5125 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5126 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5127 // the next input is close enough to being on time, so concatenate it 5128 // with the last output 5129 timedYieldSamples_l(buffer); 5130 5131 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5132 head.position(), buffer->frameCount); 5133 return NO_ERROR; 5134 } 5135 5136 // Looks like our output is not on time. Reset our on timed status. 5137 // Next time we mix samples from our input queue, then should be within 5138 // the StartupThreshold. 5139 mTimedAudioOutputOnTime = false; 5140 if (sampleDelta > 0) { 5141 // the gap between the current output position and the proper start of 5142 // the next input sample is too big, so fill it with silence 5143 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5144 5145 timedYieldSilence_l(framesUntilNextInput, buffer); 5146 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5147 return NO_ERROR; 5148 } else { 5149 // the next input sample is late 5150 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5151 size_t onTimeSamplePosition = 5152 head.position() + lateFrames * mCblk->frameSize; 5153 5154 if (onTimeSamplePosition > head.buffer()->size()) { 5155 // all the remaining samples in the head are too late, so 5156 // drop it and move on 5157 ALOGV("*** too late: dropped buffer"); 5158 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5159 continue; 5160 } else { 5161 // skip over the late samples 5162 head.setPosition(onTimeSamplePosition); 5163 5164 // yield the available samples 5165 timedYieldSamples_l(buffer); 5166 5167 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5168 return NO_ERROR; 5169 } 5170 } 5171 } 5172} 5173 5174// Yield samples from the timed buffer queue head up to the given output 5175// buffer's capacity. 5176// 5177// Caller must hold mTimedBufferQueueLock 5178void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5179 AudioBufferProvider::Buffer* buffer) { 5180 5181 const TimedBuffer& head = mTimedBufferQueue[0]; 5182 5183 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5184 head.position()); 5185 5186 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5187 mCblk->frameSize); 5188 size_t framesRequested = buffer->frameCount; 5189 buffer->frameCount = min(framesLeftInHead, framesRequested); 5190 5191 mQueueHeadInFlight = true; 5192 mTimedAudioOutputOnTime = true; 5193} 5194 5195// Yield samples of silence up to the given output buffer's capacity 5196// 5197// Caller must hold mTimedBufferQueueLock 5198void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5199 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5200 5201 // lazily allocate a buffer filled with silence 5202 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5203 delete [] mTimedSilenceBuffer; 5204 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5205 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5206 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5207 } 5208 5209 buffer->raw = mTimedSilenceBuffer; 5210 size_t framesRequested = buffer->frameCount; 5211 buffer->frameCount = min(numFrames, framesRequested); 5212 5213 mTimedAudioOutputOnTime = false; 5214} 5215 5216// AudioBufferProvider interface 5217void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5218 AudioBufferProvider::Buffer* buffer) { 5219 5220 Mutex::Autolock _l(mTimedBufferQueueLock); 5221 5222 // If the buffer which was just released is part of the buffer at the head 5223 // of the queue, be sure to update the amt of the buffer which has been 5224 // consumed. If the buffer being returned is not part of the head of the 5225 // queue, its either because the buffer is part of the silence buffer, or 5226 // because the head of the timed queue was trimmed after the mixer called 5227 // getNextBuffer but before the mixer called releaseBuffer. 5228 if (buffer->raw == mTimedSilenceBuffer) { 5229 ALOG_ASSERT(!mQueueHeadInFlight, 5230 "Queue head in flight during release of silence buffer!"); 5231 goto done; 5232 } 5233 5234 ALOG_ASSERT(mQueueHeadInFlight, 5235 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5236 " head in flight."); 5237 5238 if (mTimedBufferQueue.size()) { 5239 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5240 5241 void* start = head.buffer()->pointer(); 5242 void* end = reinterpret_cast<void*>( 5243 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5244 + head.buffer()->size()); 5245 5246 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5247 "released buffer not within the head of the timed buffer" 5248 " queue; qHead = [%p, %p], released buffer = %p", 5249 start, end, buffer->raw); 5250 5251 head.setPosition(head.position() + 5252 (buffer->frameCount * mCblk->frameSize)); 5253 mQueueHeadInFlight = false; 5254 5255 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5256 "Bad bookkeeping during releaseBuffer! Should have at" 5257 " least %u queued frames, but we think we have only %u", 5258 buffer->frameCount, mFramesPendingInQueue); 5259 5260 mFramesPendingInQueue -= buffer->frameCount; 5261 5262 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5263 || mTrimQueueHeadOnRelease) { 5264 trimTimedBufferQueueHead_l("releaseBuffer"); 5265 mTrimQueueHeadOnRelease = false; 5266 } 5267 } else { 5268 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5269 " buffers in the timed buffer queue"); 5270 } 5271 5272done: 5273 buffer->raw = 0; 5274 buffer->frameCount = 0; 5275} 5276 5277size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5278 Mutex::Autolock _l(mTimedBufferQueueLock); 5279 return mFramesPendingInQueue; 5280} 5281 5282AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5283 : mPTS(0), mPosition(0) {} 5284 5285AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5286 const sp<IMemory>& buffer, int64_t pts) 5287 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5288 5289// ---------------------------------------------------------------------------- 5290 5291// RecordTrack constructor must be called with AudioFlinger::mLock held 5292AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5293 RecordThread *thread, 5294 const sp<Client>& client, 5295 uint32_t sampleRate, 5296 audio_format_t format, 5297 audio_channel_mask_t channelMask, 5298 int frameCount, 5299 int sessionId) 5300 : TrackBase(thread, client, sampleRate, format, 5301 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5302 mOverflow(false) 5303{ 5304 if (mCblk != NULL) { 5305 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5306 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5307 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5308 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5309 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5310 } else { 5311 mCblk->frameSize = sizeof(int8_t); 5312 } 5313 } 5314} 5315 5316AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5317{ 5318 sp<ThreadBase> thread = mThread.promote(); 5319 if (thread != 0) { 5320 AudioSystem::releaseInput(thread->id()); 5321 } 5322} 5323 5324// AudioBufferProvider interface 5325status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5326{ 5327 audio_track_cblk_t* cblk = this->cblk(); 5328 uint32_t framesAvail; 5329 uint32_t framesReq = buffer->frameCount; 5330 5331 // Check if last stepServer failed, try to step now 5332 if (mStepServerFailed) { 5333 if (!step()) goto getNextBuffer_exit; 5334 ALOGV("stepServer recovered"); 5335 mStepServerFailed = false; 5336 } 5337 5338 framesAvail = cblk->framesAvailable_l(); 5339 5340 if (CC_LIKELY(framesAvail)) { 5341 uint32_t s = cblk->server; 5342 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5343 5344 if (framesReq > framesAvail) { 5345 framesReq = framesAvail; 5346 } 5347 if (framesReq > bufferEnd - s) { 5348 framesReq = bufferEnd - s; 5349 } 5350 5351 buffer->raw = getBuffer(s, framesReq); 5352 buffer->frameCount = framesReq; 5353 return NO_ERROR; 5354 } 5355 5356getNextBuffer_exit: 5357 buffer->raw = NULL; 5358 buffer->frameCount = 0; 5359 return NOT_ENOUGH_DATA; 5360} 5361 5362status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5363 int triggerSession) 5364{ 5365 sp<ThreadBase> thread = mThread.promote(); 5366 if (thread != 0) { 5367 RecordThread *recordThread = (RecordThread *)thread.get(); 5368 return recordThread->start(this, event, triggerSession); 5369 } else { 5370 return BAD_VALUE; 5371 } 5372} 5373 5374void AudioFlinger::RecordThread::RecordTrack::stop() 5375{ 5376 sp<ThreadBase> thread = mThread.promote(); 5377 if (thread != 0) { 5378 RecordThread *recordThread = (RecordThread *)thread.get(); 5379 recordThread->stop(this); 5380 TrackBase::reset(); 5381 // Force overrun condition to avoid false overrun callback until first data is 5382 // read from buffer 5383 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5384 } 5385} 5386 5387void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5388{ 5389 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5390 (mClient == 0) ? getpid_cached : mClient->pid(), 5391 mFormat, 5392 mChannelMask, 5393 mSessionId, 5394 mFrameCount, 5395 mState, 5396 mCblk->sampleRate, 5397 mCblk->server, 5398 mCblk->user); 5399} 5400 5401 5402// ---------------------------------------------------------------------------- 5403 5404AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5405 PlaybackThread *playbackThread, 5406 DuplicatingThread *sourceThread, 5407 uint32_t sampleRate, 5408 audio_format_t format, 5409 audio_channel_mask_t channelMask, 5410 int frameCount) 5411 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5412 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5413 mActive(false), mSourceThread(sourceThread) 5414{ 5415 5416 if (mCblk != NULL) { 5417 mCblk->flags |= CBLK_DIRECTION_OUT; 5418 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5419 mOutBuffer.frameCount = 0; 5420 playbackThread->mTracks.add(this); 5421 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5422 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5423 mCblk, mBuffer, mCblk->buffers, 5424 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5425 } else { 5426 ALOGW("Error creating output track on thread %p", playbackThread); 5427 } 5428} 5429 5430AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5431{ 5432 clearBufferQueue(); 5433} 5434 5435status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5436 int triggerSession) 5437{ 5438 status_t status = Track::start(event, triggerSession); 5439 if (status != NO_ERROR) { 5440 return status; 5441 } 5442 5443 mActive = true; 5444 mRetryCount = 127; 5445 return status; 5446} 5447 5448void AudioFlinger::PlaybackThread::OutputTrack::stop() 5449{ 5450 Track::stop(); 5451 clearBufferQueue(); 5452 mOutBuffer.frameCount = 0; 5453 mActive = false; 5454} 5455 5456bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5457{ 5458 Buffer *pInBuffer; 5459 Buffer inBuffer; 5460 uint32_t channelCount = mChannelCount; 5461 bool outputBufferFull = false; 5462 inBuffer.frameCount = frames; 5463 inBuffer.i16 = data; 5464 5465 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5466 5467 if (!mActive && frames != 0) { 5468 start(); 5469 sp<ThreadBase> thread = mThread.promote(); 5470 if (thread != 0) { 5471 MixerThread *mixerThread = (MixerThread *)thread.get(); 5472 if (mCblk->frameCount > frames){ 5473 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5474 uint32_t startFrames = (mCblk->frameCount - frames); 5475 pInBuffer = new Buffer; 5476 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5477 pInBuffer->frameCount = startFrames; 5478 pInBuffer->i16 = pInBuffer->mBuffer; 5479 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5480 mBufferQueue.add(pInBuffer); 5481 } else { 5482 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5483 } 5484 } 5485 } 5486 } 5487 5488 while (waitTimeLeftMs) { 5489 // First write pending buffers, then new data 5490 if (mBufferQueue.size()) { 5491 pInBuffer = mBufferQueue.itemAt(0); 5492 } else { 5493 pInBuffer = &inBuffer; 5494 } 5495 5496 if (pInBuffer->frameCount == 0) { 5497 break; 5498 } 5499 5500 if (mOutBuffer.frameCount == 0) { 5501 mOutBuffer.frameCount = pInBuffer->frameCount; 5502 nsecs_t startTime = systemTime(); 5503 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5504 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5505 outputBufferFull = true; 5506 break; 5507 } 5508 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5509 if (waitTimeLeftMs >= waitTimeMs) { 5510 waitTimeLeftMs -= waitTimeMs; 5511 } else { 5512 waitTimeLeftMs = 0; 5513 } 5514 } 5515 5516 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5517 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5518 mCblk->stepUser(outFrames); 5519 pInBuffer->frameCount -= outFrames; 5520 pInBuffer->i16 += outFrames * channelCount; 5521 mOutBuffer.frameCount -= outFrames; 5522 mOutBuffer.i16 += outFrames * channelCount; 5523 5524 if (pInBuffer->frameCount == 0) { 5525 if (mBufferQueue.size()) { 5526 mBufferQueue.removeAt(0); 5527 delete [] pInBuffer->mBuffer; 5528 delete pInBuffer; 5529 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5530 } else { 5531 break; 5532 } 5533 } 5534 } 5535 5536 // If we could not write all frames, allocate a buffer and queue it for next time. 5537 if (inBuffer.frameCount) { 5538 sp<ThreadBase> thread = mThread.promote(); 5539 if (thread != 0 && !thread->standby()) { 5540 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5541 pInBuffer = new Buffer; 5542 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5543 pInBuffer->frameCount = inBuffer.frameCount; 5544 pInBuffer->i16 = pInBuffer->mBuffer; 5545 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5546 mBufferQueue.add(pInBuffer); 5547 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5548 } else { 5549 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5550 } 5551 } 5552 } 5553 5554 // Calling write() with a 0 length buffer, means that no more data will be written: 5555 // If no more buffers are pending, fill output track buffer to make sure it is started 5556 // by output mixer. 5557 if (frames == 0 && mBufferQueue.size() == 0) { 5558 if (mCblk->user < mCblk->frameCount) { 5559 frames = mCblk->frameCount - mCblk->user; 5560 pInBuffer = new Buffer; 5561 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5562 pInBuffer->frameCount = frames; 5563 pInBuffer->i16 = pInBuffer->mBuffer; 5564 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5565 mBufferQueue.add(pInBuffer); 5566 } else if (mActive) { 5567 stop(); 5568 } 5569 } 5570 5571 return outputBufferFull; 5572} 5573 5574status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5575{ 5576 int active; 5577 status_t result; 5578 audio_track_cblk_t* cblk = mCblk; 5579 uint32_t framesReq = buffer->frameCount; 5580 5581// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5582 buffer->frameCount = 0; 5583 5584 uint32_t framesAvail = cblk->framesAvailable(); 5585 5586 5587 if (framesAvail == 0) { 5588 Mutex::Autolock _l(cblk->lock); 5589 goto start_loop_here; 5590 while (framesAvail == 0) { 5591 active = mActive; 5592 if (CC_UNLIKELY(!active)) { 5593 ALOGV("Not active and NO_MORE_BUFFERS"); 5594 return NO_MORE_BUFFERS; 5595 } 5596 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5597 if (result != NO_ERROR) { 5598 return NO_MORE_BUFFERS; 5599 } 5600 // read the server count again 5601 start_loop_here: 5602 framesAvail = cblk->framesAvailable_l(); 5603 } 5604 } 5605 5606// if (framesAvail < framesReq) { 5607// return NO_MORE_BUFFERS; 5608// } 5609 5610 if (framesReq > framesAvail) { 5611 framesReq = framesAvail; 5612 } 5613 5614 uint32_t u = cblk->user; 5615 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5616 5617 if (framesReq > bufferEnd - u) { 5618 framesReq = bufferEnd - u; 5619 } 5620 5621 buffer->frameCount = framesReq; 5622 buffer->raw = (void *)cblk->buffer(u); 5623 return NO_ERROR; 5624} 5625 5626 5627void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5628{ 5629 size_t size = mBufferQueue.size(); 5630 5631 for (size_t i = 0; i < size; i++) { 5632 Buffer *pBuffer = mBufferQueue.itemAt(i); 5633 delete [] pBuffer->mBuffer; 5634 delete pBuffer; 5635 } 5636 mBufferQueue.clear(); 5637} 5638 5639// ---------------------------------------------------------------------------- 5640 5641AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5642 : RefBase(), 5643 mAudioFlinger(audioFlinger), 5644 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5645 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5646 mPid(pid), 5647 mTimedTrackCount(0) 5648{ 5649 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5650} 5651 5652// Client destructor must be called with AudioFlinger::mLock held 5653AudioFlinger::Client::~Client() 5654{ 5655 mAudioFlinger->removeClient_l(mPid); 5656} 5657 5658sp<MemoryDealer> AudioFlinger::Client::heap() const 5659{ 5660 return mMemoryDealer; 5661} 5662 5663// Reserve one of the limited slots for a timed audio track associated 5664// with this client 5665bool AudioFlinger::Client::reserveTimedTrack() 5666{ 5667 const int kMaxTimedTracksPerClient = 4; 5668 5669 Mutex::Autolock _l(mTimedTrackLock); 5670 5671 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5672 ALOGW("can not create timed track - pid %d has exceeded the limit", 5673 mPid); 5674 return false; 5675 } 5676 5677 mTimedTrackCount++; 5678 return true; 5679} 5680 5681// Release a slot for a timed audio track 5682void AudioFlinger::Client::releaseTimedTrack() 5683{ 5684 Mutex::Autolock _l(mTimedTrackLock); 5685 mTimedTrackCount--; 5686} 5687 5688// ---------------------------------------------------------------------------- 5689 5690AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5691 const sp<IAudioFlingerClient>& client, 5692 pid_t pid) 5693 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5694{ 5695} 5696 5697AudioFlinger::NotificationClient::~NotificationClient() 5698{ 5699} 5700 5701void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5702{ 5703 sp<NotificationClient> keep(this); 5704 mAudioFlinger->removeNotificationClient(mPid); 5705} 5706 5707// ---------------------------------------------------------------------------- 5708 5709AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5710 : BnAudioTrack(), 5711 mTrack(track) 5712{ 5713} 5714 5715AudioFlinger::TrackHandle::~TrackHandle() { 5716 // just stop the track on deletion, associated resources 5717 // will be freed from the main thread once all pending buffers have 5718 // been played. Unless it's not in the active track list, in which 5719 // case we free everything now... 5720 mTrack->destroy(); 5721} 5722 5723sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5724 return mTrack->getCblk(); 5725} 5726 5727status_t AudioFlinger::TrackHandle::start() { 5728 return mTrack->start(); 5729} 5730 5731void AudioFlinger::TrackHandle::stop() { 5732 mTrack->stop(); 5733} 5734 5735void AudioFlinger::TrackHandle::flush() { 5736 mTrack->flush(); 5737} 5738 5739void AudioFlinger::TrackHandle::mute(bool e) { 5740 mTrack->mute(e); 5741} 5742 5743void AudioFlinger::TrackHandle::pause() { 5744 mTrack->pause(); 5745} 5746 5747status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5748{ 5749 return mTrack->attachAuxEffect(EffectId); 5750} 5751 5752status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5753 sp<IMemory>* buffer) { 5754 if (!mTrack->isTimedTrack()) 5755 return INVALID_OPERATION; 5756 5757 PlaybackThread::TimedTrack* tt = 5758 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5759 return tt->allocateTimedBuffer(size, buffer); 5760} 5761 5762status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5763 int64_t pts) { 5764 if (!mTrack->isTimedTrack()) 5765 return INVALID_OPERATION; 5766 5767 PlaybackThread::TimedTrack* tt = 5768 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5769 return tt->queueTimedBuffer(buffer, pts); 5770} 5771 5772status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5773 const LinearTransform& xform, int target) { 5774 5775 if (!mTrack->isTimedTrack()) 5776 return INVALID_OPERATION; 5777 5778 PlaybackThread::TimedTrack* tt = 5779 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5780 return tt->setMediaTimeTransform( 5781 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5782} 5783 5784status_t AudioFlinger::TrackHandle::onTransact( 5785 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5786{ 5787 return BnAudioTrack::onTransact(code, data, reply, flags); 5788} 5789 5790// ---------------------------------------------------------------------------- 5791 5792sp<IAudioRecord> AudioFlinger::openRecord( 5793 pid_t pid, 5794 audio_io_handle_t input, 5795 uint32_t sampleRate, 5796 audio_format_t format, 5797 audio_channel_mask_t channelMask, 5798 int frameCount, 5799 IAudioFlinger::track_flags_t flags, 5800 int *sessionId, 5801 status_t *status) 5802{ 5803 sp<RecordThread::RecordTrack> recordTrack; 5804 sp<RecordHandle> recordHandle; 5805 sp<Client> client; 5806 status_t lStatus; 5807 RecordThread *thread; 5808 size_t inFrameCount; 5809 int lSessionId; 5810 5811 // check calling permissions 5812 if (!recordingAllowed()) { 5813 lStatus = PERMISSION_DENIED; 5814 goto Exit; 5815 } 5816 5817 // add client to list 5818 { // scope for mLock 5819 Mutex::Autolock _l(mLock); 5820 thread = checkRecordThread_l(input); 5821 if (thread == NULL) { 5822 lStatus = BAD_VALUE; 5823 goto Exit; 5824 } 5825 5826 client = registerPid_l(pid); 5827 5828 // If no audio session id is provided, create one here 5829 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5830 lSessionId = *sessionId; 5831 } else { 5832 lSessionId = nextUniqueId(); 5833 if (sessionId != NULL) { 5834 *sessionId = lSessionId; 5835 } 5836 } 5837 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5838 recordTrack = thread->createRecordTrack_l(client, 5839 sampleRate, 5840 format, 5841 channelMask, 5842 frameCount, 5843 lSessionId, 5844 &lStatus); 5845 } 5846 if (lStatus != NO_ERROR) { 5847 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5848 // destructor is called by the TrackBase destructor with mLock held 5849 client.clear(); 5850 recordTrack.clear(); 5851 goto Exit; 5852 } 5853 5854 // return to handle to client 5855 recordHandle = new RecordHandle(recordTrack); 5856 lStatus = NO_ERROR; 5857 5858Exit: 5859 if (status) { 5860 *status = lStatus; 5861 } 5862 return recordHandle; 5863} 5864 5865// ---------------------------------------------------------------------------- 5866 5867AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5868 : BnAudioRecord(), 5869 mRecordTrack(recordTrack) 5870{ 5871} 5872 5873AudioFlinger::RecordHandle::~RecordHandle() { 5874 stop(); 5875} 5876 5877sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5878 return mRecordTrack->getCblk(); 5879} 5880 5881status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5882 ALOGV("RecordHandle::start()"); 5883 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5884} 5885 5886void AudioFlinger::RecordHandle::stop() { 5887 ALOGV("RecordHandle::stop()"); 5888 mRecordTrack->stop(); 5889} 5890 5891status_t AudioFlinger::RecordHandle::onTransact( 5892 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5893{ 5894 return BnAudioRecord::onTransact(code, data, reply, flags); 5895} 5896 5897// ---------------------------------------------------------------------------- 5898 5899AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5900 AudioStreamIn *input, 5901 uint32_t sampleRate, 5902 audio_channel_mask_t channelMask, 5903 audio_io_handle_t id, 5904 uint32_t device) : 5905 ThreadBase(audioFlinger, id, device, RECORD), 5906 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5907 // mRsmpInIndex and mInputBytes set by readInputParameters() 5908 mReqChannelCount(popcount(channelMask)), 5909 mReqSampleRate(sampleRate) 5910 // mBytesRead is only meaningful while active, and so is cleared in start() 5911 // (but might be better to also clear here for dump?) 5912{ 5913 snprintf(mName, kNameLength, "AudioIn_%X", id); 5914 5915 readInputParameters(); 5916} 5917 5918 5919AudioFlinger::RecordThread::~RecordThread() 5920{ 5921 delete[] mRsmpInBuffer; 5922 delete mResampler; 5923 delete[] mRsmpOutBuffer; 5924} 5925 5926void AudioFlinger::RecordThread::onFirstRef() 5927{ 5928 run(mName, PRIORITY_URGENT_AUDIO); 5929} 5930 5931status_t AudioFlinger::RecordThread::readyToRun() 5932{ 5933 status_t status = initCheck(); 5934 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5935 return status; 5936} 5937 5938bool AudioFlinger::RecordThread::threadLoop() 5939{ 5940 AudioBufferProvider::Buffer buffer; 5941 sp<RecordTrack> activeTrack; 5942 Vector< sp<EffectChain> > effectChains; 5943 5944 nsecs_t lastWarning = 0; 5945 5946 acquireWakeLock(); 5947 5948 // start recording 5949 while (!exitPending()) { 5950 5951 processConfigEvents(); 5952 5953 { // scope for mLock 5954 Mutex::Autolock _l(mLock); 5955 checkForNewParameters_l(); 5956 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5957 if (!mStandby) { 5958 mInput->stream->common.standby(&mInput->stream->common); 5959 mStandby = true; 5960 } 5961 5962 if (exitPending()) break; 5963 5964 releaseWakeLock_l(); 5965 ALOGV("RecordThread: loop stopping"); 5966 // go to sleep 5967 mWaitWorkCV.wait(mLock); 5968 ALOGV("RecordThread: loop starting"); 5969 acquireWakeLock_l(); 5970 continue; 5971 } 5972 if (mActiveTrack != 0) { 5973 if (mActiveTrack->mState == TrackBase::PAUSING) { 5974 if (!mStandby) { 5975 mInput->stream->common.standby(&mInput->stream->common); 5976 mStandby = true; 5977 } 5978 mActiveTrack.clear(); 5979 mStartStopCond.broadcast(); 5980 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5981 if (mReqChannelCount != mActiveTrack->channelCount()) { 5982 mActiveTrack.clear(); 5983 mStartStopCond.broadcast(); 5984 } else if (mBytesRead != 0) { 5985 // record start succeeds only if first read from audio input 5986 // succeeds 5987 if (mBytesRead > 0) { 5988 mActiveTrack->mState = TrackBase::ACTIVE; 5989 } else { 5990 mActiveTrack.clear(); 5991 } 5992 mStartStopCond.broadcast(); 5993 } 5994 mStandby = false; 5995 } 5996 } 5997 lockEffectChains_l(effectChains); 5998 } 5999 6000 if (mActiveTrack != 0) { 6001 if (mActiveTrack->mState != TrackBase::ACTIVE && 6002 mActiveTrack->mState != TrackBase::RESUMING) { 6003 unlockEffectChains(effectChains); 6004 usleep(kRecordThreadSleepUs); 6005 continue; 6006 } 6007 for (size_t i = 0; i < effectChains.size(); i ++) { 6008 effectChains[i]->process_l(); 6009 } 6010 6011 buffer.frameCount = mFrameCount; 6012 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6013 size_t framesOut = buffer.frameCount; 6014 if (mResampler == NULL) { 6015 // no resampling 6016 while (framesOut) { 6017 size_t framesIn = mFrameCount - mRsmpInIndex; 6018 if (framesIn) { 6019 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6020 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6021 if (framesIn > framesOut) 6022 framesIn = framesOut; 6023 mRsmpInIndex += framesIn; 6024 framesOut -= framesIn; 6025 if ((int)mChannelCount == mReqChannelCount || 6026 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6027 memcpy(dst, src, framesIn * mFrameSize); 6028 } else { 6029 int16_t *src16 = (int16_t *)src; 6030 int16_t *dst16 = (int16_t *)dst; 6031 if (mChannelCount == 1) { 6032 while (framesIn--) { 6033 *dst16++ = *src16; 6034 *dst16++ = *src16++; 6035 } 6036 } else { 6037 while (framesIn--) { 6038 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6039 src16 += 2; 6040 } 6041 } 6042 } 6043 } 6044 if (framesOut && mFrameCount == mRsmpInIndex) { 6045 if (framesOut == mFrameCount && 6046 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6047 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6048 framesOut = 0; 6049 } else { 6050 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6051 mRsmpInIndex = 0; 6052 } 6053 if (mBytesRead < 0) { 6054 ALOGE("Error reading audio input"); 6055 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6056 // Force input into standby so that it tries to 6057 // recover at next read attempt 6058 mInput->stream->common.standby(&mInput->stream->common); 6059 usleep(kRecordThreadSleepUs); 6060 } 6061 mRsmpInIndex = mFrameCount; 6062 framesOut = 0; 6063 buffer.frameCount = 0; 6064 } 6065 } 6066 } 6067 } else { 6068 // resampling 6069 6070 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6071 // alter output frame count as if we were expecting stereo samples 6072 if (mChannelCount == 1 && mReqChannelCount == 1) { 6073 framesOut >>= 1; 6074 } 6075 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6076 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6077 // are 32 bit aligned which should be always true. 6078 if (mChannelCount == 2 && mReqChannelCount == 1) { 6079 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6080 // the resampler always outputs stereo samples: do post stereo to mono conversion 6081 int16_t *src = (int16_t *)mRsmpOutBuffer; 6082 int16_t *dst = buffer.i16; 6083 while (framesOut--) { 6084 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6085 src += 2; 6086 } 6087 } else { 6088 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6089 } 6090 6091 } 6092 if (mFramestoDrop == 0) { 6093 mActiveTrack->releaseBuffer(&buffer); 6094 } else { 6095 if (mFramestoDrop > 0) { 6096 mFramestoDrop -= buffer.frameCount; 6097 if (mFramestoDrop <= 0) { 6098 clearSyncStartEvent(); 6099 } 6100 } else { 6101 mFramestoDrop += buffer.frameCount; 6102 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6103 mSyncStartEvent->isCancelled()) { 6104 ALOGW("Synced record %s, session %d, trigger session %d", 6105 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6106 mActiveTrack->sessionId(), 6107 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6108 clearSyncStartEvent(); 6109 } 6110 } 6111 } 6112 mActiveTrack->overflow(); 6113 } 6114 // client isn't retrieving buffers fast enough 6115 else { 6116 if (!mActiveTrack->setOverflow()) { 6117 nsecs_t now = systemTime(); 6118 if ((now - lastWarning) > kWarningThrottleNs) { 6119 ALOGW("RecordThread: buffer overflow"); 6120 lastWarning = now; 6121 } 6122 } 6123 // Release the processor for a while before asking for a new buffer. 6124 // This will give the application more chance to read from the buffer and 6125 // clear the overflow. 6126 usleep(kRecordThreadSleepUs); 6127 } 6128 } 6129 // enable changes in effect chain 6130 unlockEffectChains(effectChains); 6131 effectChains.clear(); 6132 } 6133 6134 if (!mStandby) { 6135 mInput->stream->common.standby(&mInput->stream->common); 6136 } 6137 mActiveTrack.clear(); 6138 6139 mStartStopCond.broadcast(); 6140 6141 releaseWakeLock(); 6142 6143 ALOGV("RecordThread %p exiting", this); 6144 return false; 6145} 6146 6147 6148sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6149 const sp<AudioFlinger::Client>& client, 6150 uint32_t sampleRate, 6151 audio_format_t format, 6152 audio_channel_mask_t channelMask, 6153 int frameCount, 6154 int sessionId, 6155 status_t *status) 6156{ 6157 sp<RecordTrack> track; 6158 status_t lStatus; 6159 6160 lStatus = initCheck(); 6161 if (lStatus != NO_ERROR) { 6162 ALOGE("Audio driver not initialized."); 6163 goto Exit; 6164 } 6165 6166 { // scope for mLock 6167 Mutex::Autolock _l(mLock); 6168 6169 track = new RecordTrack(this, client, sampleRate, 6170 format, channelMask, frameCount, sessionId); 6171 6172 if (track->getCblk() == 0) { 6173 lStatus = NO_MEMORY; 6174 goto Exit; 6175 } 6176 6177 mTrack = track.get(); 6178 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6179 bool suspend = audio_is_bluetooth_sco_device( 6180 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6181 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6182 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6183 } 6184 lStatus = NO_ERROR; 6185 6186Exit: 6187 if (status) { 6188 *status = lStatus; 6189 } 6190 return track; 6191} 6192 6193status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6194 AudioSystem::sync_event_t event, 6195 int triggerSession) 6196{ 6197 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6198 sp<ThreadBase> strongMe = this; 6199 status_t status = NO_ERROR; 6200 6201 if (event == AudioSystem::SYNC_EVENT_NONE) { 6202 clearSyncStartEvent(); 6203 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6204 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6205 triggerSession, 6206 recordTrack->sessionId(), 6207 syncStartEventCallback, 6208 this); 6209 // Sync event can be cancelled by the trigger session if the track is not in a 6210 // compatible state in which case we start record immediately 6211 if (mSyncStartEvent->isCancelled()) { 6212 clearSyncStartEvent(); 6213 } else { 6214 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6215 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6216 } 6217 } 6218 6219 { 6220 AutoMutex lock(mLock); 6221 if (mActiveTrack != 0) { 6222 if (recordTrack != mActiveTrack.get()) { 6223 status = -EBUSY; 6224 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6225 mActiveTrack->mState = TrackBase::ACTIVE; 6226 } 6227 return status; 6228 } 6229 6230 recordTrack->mState = TrackBase::IDLE; 6231 mActiveTrack = recordTrack; 6232 mLock.unlock(); 6233 status_t status = AudioSystem::startInput(mId); 6234 mLock.lock(); 6235 if (status != NO_ERROR) { 6236 mActiveTrack.clear(); 6237 clearSyncStartEvent(); 6238 return status; 6239 } 6240 mRsmpInIndex = mFrameCount; 6241 mBytesRead = 0; 6242 if (mResampler != NULL) { 6243 mResampler->reset(); 6244 } 6245 mActiveTrack->mState = TrackBase::RESUMING; 6246 // signal thread to start 6247 ALOGV("Signal record thread"); 6248 mWaitWorkCV.signal(); 6249 // do not wait for mStartStopCond if exiting 6250 if (exitPending()) { 6251 mActiveTrack.clear(); 6252 status = INVALID_OPERATION; 6253 goto startError; 6254 } 6255 mStartStopCond.wait(mLock); 6256 if (mActiveTrack == 0) { 6257 ALOGV("Record failed to start"); 6258 status = BAD_VALUE; 6259 goto startError; 6260 } 6261 ALOGV("Record started OK"); 6262 return status; 6263 } 6264startError: 6265 AudioSystem::stopInput(mId); 6266 clearSyncStartEvent(); 6267 return status; 6268} 6269 6270void AudioFlinger::RecordThread::clearSyncStartEvent() 6271{ 6272 if (mSyncStartEvent != 0) { 6273 mSyncStartEvent->cancel(); 6274 } 6275 mSyncStartEvent.clear(); 6276 mFramestoDrop = 0; 6277} 6278 6279void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6280{ 6281 sp<SyncEvent> strongEvent = event.promote(); 6282 6283 if (strongEvent != 0) { 6284 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6285 me->handleSyncStartEvent(strongEvent); 6286 } 6287} 6288 6289void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6290{ 6291 if (event == mSyncStartEvent) { 6292 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6293 // from audio HAL 6294 mFramestoDrop = mFrameCount * 2; 6295 } 6296} 6297 6298void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6299 ALOGV("RecordThread::stop"); 6300 sp<ThreadBase> strongMe = this; 6301 { 6302 AutoMutex lock(mLock); 6303 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6304 mActiveTrack->mState = TrackBase::PAUSING; 6305 // do not wait for mStartStopCond if exiting 6306 if (exitPending()) { 6307 return; 6308 } 6309 mStartStopCond.wait(mLock); 6310 // if we have been restarted, recordTrack == mActiveTrack.get() here 6311 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6312 mLock.unlock(); 6313 AudioSystem::stopInput(mId); 6314 mLock.lock(); 6315 ALOGV("Record stopped OK"); 6316 } 6317 } 6318 } 6319} 6320 6321bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6322{ 6323 return false; 6324} 6325 6326status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6327{ 6328 if (!isValidSyncEvent(event)) { 6329 return BAD_VALUE; 6330 } 6331 6332 Mutex::Autolock _l(mLock); 6333 6334 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6335 mTrack->setSyncEvent(event); 6336 return NO_ERROR; 6337 } 6338 return NAME_NOT_FOUND; 6339} 6340 6341status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6342{ 6343 const size_t SIZE = 256; 6344 char buffer[SIZE]; 6345 String8 result; 6346 6347 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6348 result.append(buffer); 6349 6350 if (mActiveTrack != 0) { 6351 result.append("Active Track:\n"); 6352 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6353 mActiveTrack->dump(buffer, SIZE); 6354 result.append(buffer); 6355 6356 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6357 result.append(buffer); 6358 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6359 result.append(buffer); 6360 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6361 result.append(buffer); 6362 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6363 result.append(buffer); 6364 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6365 result.append(buffer); 6366 6367 6368 } else { 6369 result.append("No record client\n"); 6370 } 6371 write(fd, result.string(), result.size()); 6372 6373 dumpBase(fd, args); 6374 dumpEffectChains(fd, args); 6375 6376 return NO_ERROR; 6377} 6378 6379// AudioBufferProvider interface 6380status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6381{ 6382 size_t framesReq = buffer->frameCount; 6383 size_t framesReady = mFrameCount - mRsmpInIndex; 6384 int channelCount; 6385 6386 if (framesReady == 0) { 6387 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6388 if (mBytesRead < 0) { 6389 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6390 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6391 // Force input into standby so that it tries to 6392 // recover at next read attempt 6393 mInput->stream->common.standby(&mInput->stream->common); 6394 usleep(kRecordThreadSleepUs); 6395 } 6396 buffer->raw = NULL; 6397 buffer->frameCount = 0; 6398 return NOT_ENOUGH_DATA; 6399 } 6400 mRsmpInIndex = 0; 6401 framesReady = mFrameCount; 6402 } 6403 6404 if (framesReq > framesReady) { 6405 framesReq = framesReady; 6406 } 6407 6408 if (mChannelCount == 1 && mReqChannelCount == 2) { 6409 channelCount = 1; 6410 } else { 6411 channelCount = 2; 6412 } 6413 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6414 buffer->frameCount = framesReq; 6415 return NO_ERROR; 6416} 6417 6418// AudioBufferProvider interface 6419void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6420{ 6421 mRsmpInIndex += buffer->frameCount; 6422 buffer->frameCount = 0; 6423} 6424 6425bool AudioFlinger::RecordThread::checkForNewParameters_l() 6426{ 6427 bool reconfig = false; 6428 6429 while (!mNewParameters.isEmpty()) { 6430 status_t status = NO_ERROR; 6431 String8 keyValuePair = mNewParameters[0]; 6432 AudioParameter param = AudioParameter(keyValuePair); 6433 int value; 6434 audio_format_t reqFormat = mFormat; 6435 int reqSamplingRate = mReqSampleRate; 6436 int reqChannelCount = mReqChannelCount; 6437 6438 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6439 reqSamplingRate = value; 6440 reconfig = true; 6441 } 6442 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6443 reqFormat = (audio_format_t) value; 6444 reconfig = true; 6445 } 6446 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6447 reqChannelCount = popcount(value); 6448 reconfig = true; 6449 } 6450 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6451 // do not accept frame count changes if tracks are open as the track buffer 6452 // size depends on frame count and correct behavior would not be guaranteed 6453 // if frame count is changed after track creation 6454 if (mActiveTrack != 0) { 6455 status = INVALID_OPERATION; 6456 } else { 6457 reconfig = true; 6458 } 6459 } 6460 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6461 // forward device change to effects that have requested to be 6462 // aware of attached audio device. 6463 for (size_t i = 0; i < mEffectChains.size(); i++) { 6464 mEffectChains[i]->setDevice_l(value); 6465 } 6466 // store input device and output device but do not forward output device to audio HAL. 6467 // Note that status is ignored by the caller for output device 6468 // (see AudioFlinger::setParameters() 6469 uint32_t /*audio_devices_t*/ newDevice = mDevice; 6470 if (value & AUDIO_DEVICE_OUT_ALL) { 6471 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6472 status = BAD_VALUE; 6473 } else { 6474 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6475 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6476 if (mTrack != NULL) { 6477 bool suspend = audio_is_bluetooth_sco_device( 6478 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6479 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6480 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6481 } 6482 } 6483 newDevice |= value; 6484 mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once 6485 } 6486 if (status == NO_ERROR) { 6487 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6488 if (status == INVALID_OPERATION) { 6489 mInput->stream->common.standby(&mInput->stream->common); 6490 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6491 keyValuePair.string()); 6492 } 6493 if (reconfig) { 6494 if (status == BAD_VALUE && 6495 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6496 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6497 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6498 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6499 (reqChannelCount <= FCC_2)) { 6500 status = NO_ERROR; 6501 } 6502 if (status == NO_ERROR) { 6503 readInputParameters(); 6504 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6505 } 6506 } 6507 } 6508 6509 mNewParameters.removeAt(0); 6510 6511 mParamStatus = status; 6512 mParamCond.signal(); 6513 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6514 // already timed out waiting for the status and will never signal the condition. 6515 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6516 } 6517 return reconfig; 6518} 6519 6520String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6521{ 6522 char *s; 6523 String8 out_s8 = String8(); 6524 6525 Mutex::Autolock _l(mLock); 6526 if (initCheck() != NO_ERROR) { 6527 return out_s8; 6528 } 6529 6530 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6531 out_s8 = String8(s); 6532 free(s); 6533 return out_s8; 6534} 6535 6536void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6537 AudioSystem::OutputDescriptor desc; 6538 void *param2 = NULL; 6539 6540 switch (event) { 6541 case AudioSystem::INPUT_OPENED: 6542 case AudioSystem::INPUT_CONFIG_CHANGED: 6543 desc.channels = mChannelMask; 6544 desc.samplingRate = mSampleRate; 6545 desc.format = mFormat; 6546 desc.frameCount = mFrameCount; 6547 desc.latency = 0; 6548 param2 = &desc; 6549 break; 6550 6551 case AudioSystem::INPUT_CLOSED: 6552 default: 6553 break; 6554 } 6555 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6556} 6557 6558void AudioFlinger::RecordThread::readInputParameters() 6559{ 6560 delete mRsmpInBuffer; 6561 // mRsmpInBuffer is always assigned a new[] below 6562 delete mRsmpOutBuffer; 6563 mRsmpOutBuffer = NULL; 6564 delete mResampler; 6565 mResampler = NULL; 6566 6567 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6568 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6569 mChannelCount = (uint16_t)popcount(mChannelMask); 6570 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6571 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6572 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6573 mFrameCount = mInputBytes / mFrameSize; 6574 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6575 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6576 6577 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6578 { 6579 int channelCount; 6580 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6581 // stereo to mono post process as the resampler always outputs stereo. 6582 if (mChannelCount == 1 && mReqChannelCount == 2) { 6583 channelCount = 1; 6584 } else { 6585 channelCount = 2; 6586 } 6587 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6588 mResampler->setSampleRate(mSampleRate); 6589 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6590 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6591 6592 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6593 if (mChannelCount == 1 && mReqChannelCount == 1) { 6594 mFrameCount >>= 1; 6595 } 6596 6597 } 6598 mRsmpInIndex = mFrameCount; 6599} 6600 6601unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6602{ 6603 Mutex::Autolock _l(mLock); 6604 if (initCheck() != NO_ERROR) { 6605 return 0; 6606 } 6607 6608 return mInput->stream->get_input_frames_lost(mInput->stream); 6609} 6610 6611uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6612{ 6613 Mutex::Autolock _l(mLock); 6614 uint32_t result = 0; 6615 if (getEffectChain_l(sessionId) != 0) { 6616 result = EFFECT_SESSION; 6617 } 6618 6619 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6620 result |= TRACK_SESSION; 6621 } 6622 6623 return result; 6624} 6625 6626AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6627{ 6628 Mutex::Autolock _l(mLock); 6629 return mTrack; 6630} 6631 6632AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6633{ 6634 Mutex::Autolock _l(mLock); 6635 return mInput; 6636} 6637 6638AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6639{ 6640 Mutex::Autolock _l(mLock); 6641 AudioStreamIn *input = mInput; 6642 mInput = NULL; 6643 return input; 6644} 6645 6646// this method must always be called either with ThreadBase mLock held or inside the thread loop 6647audio_stream_t* AudioFlinger::RecordThread::stream() const 6648{ 6649 if (mInput == NULL) { 6650 return NULL; 6651 } 6652 return &mInput->stream->common; 6653} 6654 6655 6656// ---------------------------------------------------------------------------- 6657 6658audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6659{ 6660 if (!settingsAllowed()) { 6661 return 0; 6662 } 6663 Mutex::Autolock _l(mLock); 6664 return loadHwModule_l(name); 6665} 6666 6667// loadHwModule_l() must be called with AudioFlinger::mLock held 6668audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6669{ 6670 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6671 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6672 ALOGW("loadHwModule() module %s already loaded", name); 6673 return mAudioHwDevs.keyAt(i); 6674 } 6675 } 6676 6677 audio_hw_device_t *dev; 6678 6679 int rc = load_audio_interface(name, &dev); 6680 if (rc) { 6681 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6682 return 0; 6683 } 6684 6685 mHardwareStatus = AUDIO_HW_INIT; 6686 rc = dev->init_check(dev); 6687 mHardwareStatus = AUDIO_HW_IDLE; 6688 if (rc) { 6689 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6690 return 0; 6691 } 6692 6693 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6694 (NULL != dev->set_master_volume)) { 6695 AutoMutex lock(mHardwareLock); 6696 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6697 dev->set_master_volume(dev, mMasterVolume); 6698 mHardwareStatus = AUDIO_HW_IDLE; 6699 } 6700 6701 audio_module_handle_t handle = nextUniqueId(); 6702 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6703 6704 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6705 name, dev->common.module->name, dev->common.module->id, handle); 6706 6707 return handle; 6708 6709} 6710 6711audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6712 audio_devices_t *pDevices, 6713 uint32_t *pSamplingRate, 6714 audio_format_t *pFormat, 6715 audio_channel_mask_t *pChannelMask, 6716 uint32_t *pLatencyMs, 6717 audio_output_flags_t flags) 6718{ 6719 status_t status; 6720 PlaybackThread *thread = NULL; 6721 struct audio_config config = { 6722 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6723 channel_mask: pChannelMask ? *pChannelMask : 0, 6724 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6725 }; 6726 audio_stream_out_t *outStream = NULL; 6727 audio_hw_device_t *outHwDev; 6728 6729 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6730 module, 6731 (pDevices != NULL) ? (int)*pDevices : 0, 6732 config.sample_rate, 6733 config.format, 6734 config.channel_mask, 6735 flags); 6736 6737 if (pDevices == NULL || *pDevices == 0) { 6738 return 0; 6739 } 6740 6741 Mutex::Autolock _l(mLock); 6742 6743 outHwDev = findSuitableHwDev_l(module, *pDevices); 6744 if (outHwDev == NULL) 6745 return 0; 6746 6747 audio_io_handle_t id = nextUniqueId(); 6748 6749 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6750 6751 status = outHwDev->open_output_stream(outHwDev, 6752 id, 6753 *pDevices, 6754 (audio_output_flags_t)flags, 6755 &config, 6756 &outStream); 6757 6758 mHardwareStatus = AUDIO_HW_IDLE; 6759 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6760 outStream, 6761 config.sample_rate, 6762 config.format, 6763 config.channel_mask, 6764 status); 6765 6766 if (status == NO_ERROR && outStream != NULL) { 6767 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6768 6769 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6770 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6771 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6772 thread = new DirectOutputThread(this, output, id, *pDevices); 6773 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6774 } else { 6775 thread = new MixerThread(this, output, id, *pDevices); 6776 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6777 } 6778 mPlaybackThreads.add(id, thread); 6779 6780 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6781 if (pFormat != NULL) *pFormat = config.format; 6782 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6783 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6784 6785 // notify client processes of the new output creation 6786 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6787 6788 // the first primary output opened designates the primary hw device 6789 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6790 ALOGI("Using module %d has the primary audio interface", module); 6791 mPrimaryHardwareDev = outHwDev; 6792 6793 AutoMutex lock(mHardwareLock); 6794 mHardwareStatus = AUDIO_HW_SET_MODE; 6795 outHwDev->set_mode(outHwDev, mMode); 6796 6797 // Determine the level of master volume support the primary audio HAL has, 6798 // and set the initial master volume at the same time. 6799 float initialVolume = 1.0; 6800 mMasterVolumeSupportLvl = MVS_NONE; 6801 6802 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6803 if ((NULL != outHwDev->get_master_volume) && 6804 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6805 mMasterVolumeSupportLvl = MVS_FULL; 6806 } else { 6807 mMasterVolumeSupportLvl = MVS_SETONLY; 6808 initialVolume = 1.0; 6809 } 6810 6811 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6812 if ((NULL == outHwDev->set_master_volume) || 6813 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6814 mMasterVolumeSupportLvl = MVS_NONE; 6815 } 6816 // now that we have a primary device, initialize master volume on other devices 6817 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6818 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6819 6820 if ((dev != mPrimaryHardwareDev) && 6821 (NULL != dev->set_master_volume)) { 6822 dev->set_master_volume(dev, initialVolume); 6823 } 6824 } 6825 mHardwareStatus = AUDIO_HW_IDLE; 6826 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6827 ? initialVolume 6828 : 1.0; 6829 mMasterVolume = initialVolume; 6830 } 6831 return id; 6832 } 6833 6834 return 0; 6835} 6836 6837audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6838 audio_io_handle_t output2) 6839{ 6840 Mutex::Autolock _l(mLock); 6841 MixerThread *thread1 = checkMixerThread_l(output1); 6842 MixerThread *thread2 = checkMixerThread_l(output2); 6843 6844 if (thread1 == NULL || thread2 == NULL) { 6845 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6846 return 0; 6847 } 6848 6849 audio_io_handle_t id = nextUniqueId(); 6850 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6851 thread->addOutputTrack(thread2); 6852 mPlaybackThreads.add(id, thread); 6853 // notify client processes of the new output creation 6854 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6855 return id; 6856} 6857 6858status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6859{ 6860 // keep strong reference on the playback thread so that 6861 // it is not destroyed while exit() is executed 6862 sp<PlaybackThread> thread; 6863 { 6864 Mutex::Autolock _l(mLock); 6865 thread = checkPlaybackThread_l(output); 6866 if (thread == NULL) { 6867 return BAD_VALUE; 6868 } 6869 6870 ALOGV("closeOutput() %d", output); 6871 6872 if (thread->type() == ThreadBase::MIXER) { 6873 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6874 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6875 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6876 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6877 } 6878 } 6879 } 6880 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6881 mPlaybackThreads.removeItem(output); 6882 } 6883 thread->exit(); 6884 // The thread entity (active unit of execution) is no longer running here, 6885 // but the ThreadBase container still exists. 6886 6887 if (thread->type() != ThreadBase::DUPLICATING) { 6888 AudioStreamOut *out = thread->clearOutput(); 6889 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6890 // from now on thread->mOutput is NULL 6891 out->hwDev->close_output_stream(out->hwDev, out->stream); 6892 delete out; 6893 } 6894 return NO_ERROR; 6895} 6896 6897status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6898{ 6899 Mutex::Autolock _l(mLock); 6900 PlaybackThread *thread = checkPlaybackThread_l(output); 6901 6902 if (thread == NULL) { 6903 return BAD_VALUE; 6904 } 6905 6906 ALOGV("suspendOutput() %d", output); 6907 thread->suspend(); 6908 6909 return NO_ERROR; 6910} 6911 6912status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6913{ 6914 Mutex::Autolock _l(mLock); 6915 PlaybackThread *thread = checkPlaybackThread_l(output); 6916 6917 if (thread == NULL) { 6918 return BAD_VALUE; 6919 } 6920 6921 ALOGV("restoreOutput() %d", output); 6922 6923 thread->restore(); 6924 6925 return NO_ERROR; 6926} 6927 6928audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6929 audio_devices_t *pDevices, 6930 uint32_t *pSamplingRate, 6931 audio_format_t *pFormat, 6932 audio_channel_mask_t *pChannelMask) 6933{ 6934 status_t status; 6935 RecordThread *thread = NULL; 6936 struct audio_config config = { 6937 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6938 channel_mask: pChannelMask ? *pChannelMask : 0, 6939 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6940 }; 6941 uint32_t reqSamplingRate = config.sample_rate; 6942 audio_format_t reqFormat = config.format; 6943 audio_channel_mask_t reqChannels = config.channel_mask; 6944 audio_stream_in_t *inStream = NULL; 6945 audio_hw_device_t *inHwDev; 6946 6947 if (pDevices == NULL || *pDevices == 0) { 6948 return 0; 6949 } 6950 6951 Mutex::Autolock _l(mLock); 6952 6953 inHwDev = findSuitableHwDev_l(module, *pDevices); 6954 if (inHwDev == NULL) 6955 return 0; 6956 6957 audio_io_handle_t id = nextUniqueId(); 6958 6959 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6960 &inStream); 6961 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6962 inStream, 6963 config.sample_rate, 6964 config.format, 6965 config.channel_mask, 6966 status); 6967 6968 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6969 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6970 // or stereo to mono conversions on 16 bit PCM inputs. 6971 if (status == BAD_VALUE && 6972 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6973 (config.sample_rate <= 2 * reqSamplingRate) && 6974 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6975 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 6976 inStream = NULL; 6977 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6978 } 6979 6980 if (status == NO_ERROR && inStream != NULL) { 6981 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6982 6983 // Start record thread 6984 // RecorThread require both input and output device indication to forward to audio 6985 // pre processing modules 6986 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6987 thread = new RecordThread(this, 6988 input, 6989 reqSamplingRate, 6990 reqChannels, 6991 id, 6992 device); 6993 mRecordThreads.add(id, thread); 6994 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6995 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6996 if (pFormat != NULL) *pFormat = config.format; 6997 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6998 6999 input->stream->common.standby(&input->stream->common); 7000 7001 // notify client processes of the new input creation 7002 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7003 return id; 7004 } 7005 7006 return 0; 7007} 7008 7009status_t AudioFlinger::closeInput(audio_io_handle_t input) 7010{ 7011 // keep strong reference on the record thread so that 7012 // it is not destroyed while exit() is executed 7013 sp<RecordThread> thread; 7014 { 7015 Mutex::Autolock _l(mLock); 7016 thread = checkRecordThread_l(input); 7017 if (thread == 0) { 7018 return BAD_VALUE; 7019 } 7020 7021 ALOGV("closeInput() %d", input); 7022 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7023 mRecordThreads.removeItem(input); 7024 } 7025 thread->exit(); 7026 // The thread entity (active unit of execution) is no longer running here, 7027 // but the ThreadBase container still exists. 7028 7029 AudioStreamIn *in = thread->clearInput(); 7030 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7031 // from now on thread->mInput is NULL 7032 in->hwDev->close_input_stream(in->hwDev, in->stream); 7033 delete in; 7034 7035 return NO_ERROR; 7036} 7037 7038status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7039{ 7040 Mutex::Autolock _l(mLock); 7041 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7042 7043 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7044 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7045 thread->invalidateTracks(stream); 7046 } 7047 7048 return NO_ERROR; 7049} 7050 7051 7052int AudioFlinger::newAudioSessionId() 7053{ 7054 return nextUniqueId(); 7055} 7056 7057void AudioFlinger::acquireAudioSessionId(int audioSession) 7058{ 7059 Mutex::Autolock _l(mLock); 7060 pid_t caller = IPCThreadState::self()->getCallingPid(); 7061 ALOGV("acquiring %d from %d", audioSession, caller); 7062 size_t num = mAudioSessionRefs.size(); 7063 for (size_t i = 0; i< num; i++) { 7064 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7065 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7066 ref->mCnt++; 7067 ALOGV(" incremented refcount to %d", ref->mCnt); 7068 return; 7069 } 7070 } 7071 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7072 ALOGV(" added new entry for %d", audioSession); 7073} 7074 7075void AudioFlinger::releaseAudioSessionId(int audioSession) 7076{ 7077 Mutex::Autolock _l(mLock); 7078 pid_t caller = IPCThreadState::self()->getCallingPid(); 7079 ALOGV("releasing %d from %d", audioSession, caller); 7080 size_t num = mAudioSessionRefs.size(); 7081 for (size_t i = 0; i< num; i++) { 7082 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7083 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7084 ref->mCnt--; 7085 ALOGV(" decremented refcount to %d", ref->mCnt); 7086 if (ref->mCnt == 0) { 7087 mAudioSessionRefs.removeAt(i); 7088 delete ref; 7089 purgeStaleEffects_l(); 7090 } 7091 return; 7092 } 7093 } 7094 ALOGW("session id %d not found for pid %d", audioSession, caller); 7095} 7096 7097void AudioFlinger::purgeStaleEffects_l() { 7098 7099 ALOGV("purging stale effects"); 7100 7101 Vector< sp<EffectChain> > chains; 7102 7103 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7104 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7105 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7106 sp<EffectChain> ec = t->mEffectChains[j]; 7107 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7108 chains.push(ec); 7109 } 7110 } 7111 } 7112 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7113 sp<RecordThread> t = mRecordThreads.valueAt(i); 7114 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7115 sp<EffectChain> ec = t->mEffectChains[j]; 7116 chains.push(ec); 7117 } 7118 } 7119 7120 for (size_t i = 0; i < chains.size(); i++) { 7121 sp<EffectChain> ec = chains[i]; 7122 int sessionid = ec->sessionId(); 7123 sp<ThreadBase> t = ec->mThread.promote(); 7124 if (t == 0) { 7125 continue; 7126 } 7127 size_t numsessionrefs = mAudioSessionRefs.size(); 7128 bool found = false; 7129 for (size_t k = 0; k < numsessionrefs; k++) { 7130 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7131 if (ref->mSessionid == sessionid) { 7132 ALOGV(" session %d still exists for %d with %d refs", 7133 sessionid, ref->mPid, ref->mCnt); 7134 found = true; 7135 break; 7136 } 7137 } 7138 if (!found) { 7139 Mutex::Autolock _l (t->mLock); 7140 // remove all effects from the chain 7141 while (ec->mEffects.size()) { 7142 sp<EffectModule> effect = ec->mEffects[0]; 7143 effect->unPin(); 7144 t->removeEffect_l(effect); 7145 if (effect->purgeHandles()) { 7146 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7147 } 7148 AudioSystem::unregisterEffect(effect->id()); 7149 } 7150 } 7151 } 7152 return; 7153} 7154 7155// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7156AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7157{ 7158 return mPlaybackThreads.valueFor(output).get(); 7159} 7160 7161// checkMixerThread_l() must be called with AudioFlinger::mLock held 7162AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7163{ 7164 PlaybackThread *thread = checkPlaybackThread_l(output); 7165 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7166} 7167 7168// checkRecordThread_l() must be called with AudioFlinger::mLock held 7169AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7170{ 7171 return mRecordThreads.valueFor(input).get(); 7172} 7173 7174uint32_t AudioFlinger::nextUniqueId() 7175{ 7176 return android_atomic_inc(&mNextUniqueId); 7177} 7178 7179AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7180{ 7181 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7182 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7183 AudioStreamOut *output = thread->getOutput(); 7184 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7185 return thread; 7186 } 7187 } 7188 return NULL; 7189} 7190 7191uint32_t AudioFlinger::primaryOutputDevice_l() const 7192{ 7193 PlaybackThread *thread = primaryPlaybackThread_l(); 7194 7195 if (thread == NULL) { 7196 return 0; 7197 } 7198 7199 return thread->device(); 7200} 7201 7202sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7203 int triggerSession, 7204 int listenerSession, 7205 sync_event_callback_t callBack, 7206 void *cookie) 7207{ 7208 Mutex::Autolock _l(mLock); 7209 7210 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7211 status_t playStatus = NAME_NOT_FOUND; 7212 status_t recStatus = NAME_NOT_FOUND; 7213 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7214 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7215 if (playStatus == NO_ERROR) { 7216 return event; 7217 } 7218 } 7219 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7220 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7221 if (recStatus == NO_ERROR) { 7222 return event; 7223 } 7224 } 7225 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7226 mPendingSyncEvents.add(event); 7227 } else { 7228 ALOGV("createSyncEvent() invalid event %d", event->type()); 7229 event.clear(); 7230 } 7231 return event; 7232} 7233 7234// ---------------------------------------------------------------------------- 7235// Effect management 7236// ---------------------------------------------------------------------------- 7237 7238 7239status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7240{ 7241 Mutex::Autolock _l(mLock); 7242 return EffectQueryNumberEffects(numEffects); 7243} 7244 7245status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7246{ 7247 Mutex::Autolock _l(mLock); 7248 return EffectQueryEffect(index, descriptor); 7249} 7250 7251status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7252 effect_descriptor_t *descriptor) const 7253{ 7254 Mutex::Autolock _l(mLock); 7255 return EffectGetDescriptor(pUuid, descriptor); 7256} 7257 7258 7259sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7260 effect_descriptor_t *pDesc, 7261 const sp<IEffectClient>& effectClient, 7262 int32_t priority, 7263 audio_io_handle_t io, 7264 int sessionId, 7265 status_t *status, 7266 int *id, 7267 int *enabled) 7268{ 7269 status_t lStatus = NO_ERROR; 7270 sp<EffectHandle> handle; 7271 effect_descriptor_t desc; 7272 7273 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7274 pid, effectClient.get(), priority, sessionId, io); 7275 7276 if (pDesc == NULL) { 7277 lStatus = BAD_VALUE; 7278 goto Exit; 7279 } 7280 7281 // check audio settings permission for global effects 7282 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7283 lStatus = PERMISSION_DENIED; 7284 goto Exit; 7285 } 7286 7287 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7288 // that can only be created by audio policy manager (running in same process) 7289 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7290 lStatus = PERMISSION_DENIED; 7291 goto Exit; 7292 } 7293 7294 if (io == 0) { 7295 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7296 // output must be specified by AudioPolicyManager when using session 7297 // AUDIO_SESSION_OUTPUT_STAGE 7298 lStatus = BAD_VALUE; 7299 goto Exit; 7300 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7301 // if the output returned by getOutputForEffect() is removed before we lock the 7302 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7303 // and we will exit safely 7304 io = AudioSystem::getOutputForEffect(&desc); 7305 } 7306 } 7307 7308 { 7309 Mutex::Autolock _l(mLock); 7310 7311 7312 if (!EffectIsNullUuid(&pDesc->uuid)) { 7313 // if uuid is specified, request effect descriptor 7314 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7315 if (lStatus < 0) { 7316 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7317 goto Exit; 7318 } 7319 } else { 7320 // if uuid is not specified, look for an available implementation 7321 // of the required type in effect factory 7322 if (EffectIsNullUuid(&pDesc->type)) { 7323 ALOGW("createEffect() no effect type"); 7324 lStatus = BAD_VALUE; 7325 goto Exit; 7326 } 7327 uint32_t numEffects = 0; 7328 effect_descriptor_t d; 7329 d.flags = 0; // prevent compiler warning 7330 bool found = false; 7331 7332 lStatus = EffectQueryNumberEffects(&numEffects); 7333 if (lStatus < 0) { 7334 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7335 goto Exit; 7336 } 7337 for (uint32_t i = 0; i < numEffects; i++) { 7338 lStatus = EffectQueryEffect(i, &desc); 7339 if (lStatus < 0) { 7340 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7341 continue; 7342 } 7343 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7344 // If matching type found save effect descriptor. If the session is 7345 // 0 and the effect is not auxiliary, continue enumeration in case 7346 // an auxiliary version of this effect type is available 7347 found = true; 7348 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7349 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7350 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7351 break; 7352 } 7353 } 7354 } 7355 if (!found) { 7356 lStatus = BAD_VALUE; 7357 ALOGW("createEffect() effect not found"); 7358 goto Exit; 7359 } 7360 // For same effect type, chose auxiliary version over insert version if 7361 // connect to output mix (Compliance to OpenSL ES) 7362 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7363 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7364 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7365 } 7366 } 7367 7368 // Do not allow auxiliary effects on a session different from 0 (output mix) 7369 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7370 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7371 lStatus = INVALID_OPERATION; 7372 goto Exit; 7373 } 7374 7375 // check recording permission for visualizer 7376 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7377 !recordingAllowed()) { 7378 lStatus = PERMISSION_DENIED; 7379 goto Exit; 7380 } 7381 7382 // return effect descriptor 7383 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7384 7385 // If output is not specified try to find a matching audio session ID in one of the 7386 // output threads. 7387 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7388 // because of code checking output when entering the function. 7389 // Note: io is never 0 when creating an effect on an input 7390 if (io == 0) { 7391 // look for the thread where the specified audio session is present 7392 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7393 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7394 io = mPlaybackThreads.keyAt(i); 7395 break; 7396 } 7397 } 7398 if (io == 0) { 7399 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7400 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7401 io = mRecordThreads.keyAt(i); 7402 break; 7403 } 7404 } 7405 } 7406 // If no output thread contains the requested session ID, default to 7407 // first output. The effect chain will be moved to the correct output 7408 // thread when a track with the same session ID is created 7409 if (io == 0 && mPlaybackThreads.size()) { 7410 io = mPlaybackThreads.keyAt(0); 7411 } 7412 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7413 } 7414 ThreadBase *thread = checkRecordThread_l(io); 7415 if (thread == NULL) { 7416 thread = checkPlaybackThread_l(io); 7417 if (thread == NULL) { 7418 ALOGE("createEffect() unknown output thread"); 7419 lStatus = BAD_VALUE; 7420 goto Exit; 7421 } 7422 } 7423 7424 sp<Client> client = registerPid_l(pid); 7425 7426 // create effect on selected output thread 7427 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7428 &desc, enabled, &lStatus); 7429 if (handle != 0 && id != NULL) { 7430 *id = handle->id(); 7431 } 7432 } 7433 7434Exit: 7435 if (status != NULL) { 7436 *status = lStatus; 7437 } 7438 return handle; 7439} 7440 7441status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7442 audio_io_handle_t dstOutput) 7443{ 7444 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7445 sessionId, srcOutput, dstOutput); 7446 Mutex::Autolock _l(mLock); 7447 if (srcOutput == dstOutput) { 7448 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7449 return NO_ERROR; 7450 } 7451 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7452 if (srcThread == NULL) { 7453 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7454 return BAD_VALUE; 7455 } 7456 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7457 if (dstThread == NULL) { 7458 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7459 return BAD_VALUE; 7460 } 7461 7462 Mutex::Autolock _dl(dstThread->mLock); 7463 Mutex::Autolock _sl(srcThread->mLock); 7464 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7465 7466 return NO_ERROR; 7467} 7468 7469// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7470status_t AudioFlinger::moveEffectChain_l(int sessionId, 7471 AudioFlinger::PlaybackThread *srcThread, 7472 AudioFlinger::PlaybackThread *dstThread, 7473 bool reRegister) 7474{ 7475 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7476 sessionId, srcThread, dstThread); 7477 7478 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7479 if (chain == 0) { 7480 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7481 sessionId, srcThread); 7482 return INVALID_OPERATION; 7483 } 7484 7485 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7486 // so that a new chain is created with correct parameters when first effect is added. This is 7487 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7488 // removed. 7489 srcThread->removeEffectChain_l(chain); 7490 7491 // transfer all effects one by one so that new effect chain is created on new thread with 7492 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7493 audio_io_handle_t dstOutput = dstThread->id(); 7494 sp<EffectChain> dstChain; 7495 uint32_t strategy = 0; // prevent compiler warning 7496 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7497 while (effect != 0) { 7498 srcThread->removeEffect_l(effect); 7499 dstThread->addEffect_l(effect); 7500 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7501 if (effect->state() == EffectModule::ACTIVE || 7502 effect->state() == EffectModule::STOPPING) { 7503 effect->start(); 7504 } 7505 // if the move request is not received from audio policy manager, the effect must be 7506 // re-registered with the new strategy and output 7507 if (dstChain == 0) { 7508 dstChain = effect->chain().promote(); 7509 if (dstChain == 0) { 7510 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7511 srcThread->addEffect_l(effect); 7512 return NO_INIT; 7513 } 7514 strategy = dstChain->strategy(); 7515 } 7516 if (reRegister) { 7517 AudioSystem::unregisterEffect(effect->id()); 7518 AudioSystem::registerEffect(&effect->desc(), 7519 dstOutput, 7520 strategy, 7521 sessionId, 7522 effect->id()); 7523 } 7524 effect = chain->getEffectFromId_l(0); 7525 } 7526 7527 return NO_ERROR; 7528} 7529 7530 7531// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7532sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7533 const sp<AudioFlinger::Client>& client, 7534 const sp<IEffectClient>& effectClient, 7535 int32_t priority, 7536 int sessionId, 7537 effect_descriptor_t *desc, 7538 int *enabled, 7539 status_t *status 7540 ) 7541{ 7542 sp<EffectModule> effect; 7543 sp<EffectHandle> handle; 7544 status_t lStatus; 7545 sp<EffectChain> chain; 7546 bool chainCreated = false; 7547 bool effectCreated = false; 7548 bool effectRegistered = false; 7549 7550 lStatus = initCheck(); 7551 if (lStatus != NO_ERROR) { 7552 ALOGW("createEffect_l() Audio driver not initialized."); 7553 goto Exit; 7554 } 7555 7556 // Do not allow effects with session ID 0 on direct output or duplicating threads 7557 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7558 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7559 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7560 desc->name, sessionId); 7561 lStatus = BAD_VALUE; 7562 goto Exit; 7563 } 7564 // Only Pre processor effects are allowed on input threads and only on input threads 7565 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7566 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7567 desc->name, desc->flags, mType); 7568 lStatus = BAD_VALUE; 7569 goto Exit; 7570 } 7571 7572 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7573 7574 { // scope for mLock 7575 Mutex::Autolock _l(mLock); 7576 7577 // check for existing effect chain with the requested audio session 7578 chain = getEffectChain_l(sessionId); 7579 if (chain == 0) { 7580 // create a new chain for this session 7581 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7582 chain = new EffectChain(this, sessionId); 7583 addEffectChain_l(chain); 7584 chain->setStrategy(getStrategyForSession_l(sessionId)); 7585 chainCreated = true; 7586 } else { 7587 effect = chain->getEffectFromDesc_l(desc); 7588 } 7589 7590 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7591 7592 if (effect == 0) { 7593 int id = mAudioFlinger->nextUniqueId(); 7594 // Check CPU and memory usage 7595 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7596 if (lStatus != NO_ERROR) { 7597 goto Exit; 7598 } 7599 effectRegistered = true; 7600 // create a new effect module if none present in the chain 7601 effect = new EffectModule(this, chain, desc, id, sessionId); 7602 lStatus = effect->status(); 7603 if (lStatus != NO_ERROR) { 7604 goto Exit; 7605 } 7606 lStatus = chain->addEffect_l(effect); 7607 if (lStatus != NO_ERROR) { 7608 goto Exit; 7609 } 7610 effectCreated = true; 7611 7612 effect->setDevice(mDevice); 7613 effect->setMode(mAudioFlinger->getMode()); 7614 } 7615 // create effect handle and connect it to effect module 7616 handle = new EffectHandle(effect, client, effectClient, priority); 7617 lStatus = effect->addHandle(handle.get()); 7618 if (enabled != NULL) { 7619 *enabled = (int)effect->isEnabled(); 7620 } 7621 } 7622 7623Exit: 7624 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7625 Mutex::Autolock _l(mLock); 7626 if (effectCreated) { 7627 chain->removeEffect_l(effect); 7628 } 7629 if (effectRegistered) { 7630 AudioSystem::unregisterEffect(effect->id()); 7631 } 7632 if (chainCreated) { 7633 removeEffectChain_l(chain); 7634 } 7635 handle.clear(); 7636 } 7637 7638 if (status != NULL) { 7639 *status = lStatus; 7640 } 7641 return handle; 7642} 7643 7644sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7645{ 7646 Mutex::Autolock _l(mLock); 7647 return getEffect_l(sessionId, effectId); 7648} 7649 7650sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7651{ 7652 sp<EffectChain> chain = getEffectChain_l(sessionId); 7653 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7654} 7655 7656// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7657// PlaybackThread::mLock held 7658status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7659{ 7660 // check for existing effect chain with the requested audio session 7661 int sessionId = effect->sessionId(); 7662 sp<EffectChain> chain = getEffectChain_l(sessionId); 7663 bool chainCreated = false; 7664 7665 if (chain == 0) { 7666 // create a new chain for this session 7667 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7668 chain = new EffectChain(this, sessionId); 7669 addEffectChain_l(chain); 7670 chain->setStrategy(getStrategyForSession_l(sessionId)); 7671 chainCreated = true; 7672 } 7673 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7674 7675 if (chain->getEffectFromId_l(effect->id()) != 0) { 7676 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7677 this, effect->desc().name, chain.get()); 7678 return BAD_VALUE; 7679 } 7680 7681 status_t status = chain->addEffect_l(effect); 7682 if (status != NO_ERROR) { 7683 if (chainCreated) { 7684 removeEffectChain_l(chain); 7685 } 7686 return status; 7687 } 7688 7689 effect->setDevice(mDevice); 7690 effect->setMode(mAudioFlinger->getMode()); 7691 return NO_ERROR; 7692} 7693 7694void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7695 7696 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7697 effect_descriptor_t desc = effect->desc(); 7698 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7699 detachAuxEffect_l(effect->id()); 7700 } 7701 7702 sp<EffectChain> chain = effect->chain().promote(); 7703 if (chain != 0) { 7704 // remove effect chain if removing last effect 7705 if (chain->removeEffect_l(effect) == 0) { 7706 removeEffectChain_l(chain); 7707 } 7708 } else { 7709 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7710 } 7711} 7712 7713void AudioFlinger::ThreadBase::lockEffectChains_l( 7714 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7715{ 7716 effectChains = mEffectChains; 7717 for (size_t i = 0; i < mEffectChains.size(); i++) { 7718 mEffectChains[i]->lock(); 7719 } 7720} 7721 7722void AudioFlinger::ThreadBase::unlockEffectChains( 7723 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7724{ 7725 for (size_t i = 0; i < effectChains.size(); i++) { 7726 effectChains[i]->unlock(); 7727 } 7728} 7729 7730sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7731{ 7732 Mutex::Autolock _l(mLock); 7733 return getEffectChain_l(sessionId); 7734} 7735 7736sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7737{ 7738 size_t size = mEffectChains.size(); 7739 for (size_t i = 0; i < size; i++) { 7740 if (mEffectChains[i]->sessionId() == sessionId) { 7741 return mEffectChains[i]; 7742 } 7743 } 7744 return 0; 7745} 7746 7747void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7748{ 7749 Mutex::Autolock _l(mLock); 7750 size_t size = mEffectChains.size(); 7751 for (size_t i = 0; i < size; i++) { 7752 mEffectChains[i]->setMode_l(mode); 7753 } 7754} 7755 7756void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7757 EffectHandle *handle, 7758 bool unpinIfLast) { 7759 7760 Mutex::Autolock _l(mLock); 7761 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7762 // delete the effect module if removing last handle on it 7763 if (effect->removeHandle(handle) == 0) { 7764 if (!effect->isPinned() || unpinIfLast) { 7765 removeEffect_l(effect); 7766 AudioSystem::unregisterEffect(effect->id()); 7767 } 7768 } 7769} 7770 7771status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7772{ 7773 int session = chain->sessionId(); 7774 int16_t *buffer = mMixBuffer; 7775 bool ownsBuffer = false; 7776 7777 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7778 if (session > 0) { 7779 // Only one effect chain can be present in direct output thread and it uses 7780 // the mix buffer as input 7781 if (mType != DIRECT) { 7782 size_t numSamples = mNormalFrameCount * mChannelCount; 7783 buffer = new int16_t[numSamples]; 7784 memset(buffer, 0, numSamples * sizeof(int16_t)); 7785 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7786 ownsBuffer = true; 7787 } 7788 7789 // Attach all tracks with same session ID to this chain. 7790 for (size_t i = 0; i < mTracks.size(); ++i) { 7791 sp<Track> track = mTracks[i]; 7792 if (session == track->sessionId()) { 7793 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7794 track->setMainBuffer(buffer); 7795 chain->incTrackCnt(); 7796 } 7797 } 7798 7799 // indicate all active tracks in the chain 7800 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7801 sp<Track> track = mActiveTracks[i].promote(); 7802 if (track == 0) continue; 7803 if (session == track->sessionId()) { 7804 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7805 chain->incActiveTrackCnt(); 7806 } 7807 } 7808 } 7809 7810 chain->setInBuffer(buffer, ownsBuffer); 7811 chain->setOutBuffer(mMixBuffer); 7812 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7813 // chains list in order to be processed last as it contains output stage effects 7814 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7815 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7816 // after track specific effects and before output stage 7817 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7818 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7819 // Effect chain for other sessions are inserted at beginning of effect 7820 // chains list to be processed before output mix effects. Relative order between other 7821 // sessions is not important 7822 size_t size = mEffectChains.size(); 7823 size_t i = 0; 7824 for (i = 0; i < size; i++) { 7825 if (mEffectChains[i]->sessionId() < session) break; 7826 } 7827 mEffectChains.insertAt(chain, i); 7828 checkSuspendOnAddEffectChain_l(chain); 7829 7830 return NO_ERROR; 7831} 7832 7833size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7834{ 7835 int session = chain->sessionId(); 7836 7837 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7838 7839 for (size_t i = 0; i < mEffectChains.size(); i++) { 7840 if (chain == mEffectChains[i]) { 7841 mEffectChains.removeAt(i); 7842 // detach all active tracks from the chain 7843 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7844 sp<Track> track = mActiveTracks[i].promote(); 7845 if (track == 0) continue; 7846 if (session == track->sessionId()) { 7847 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7848 chain.get(), session); 7849 chain->decActiveTrackCnt(); 7850 } 7851 } 7852 7853 // detach all tracks with same session ID from this chain 7854 for (size_t i = 0; i < mTracks.size(); ++i) { 7855 sp<Track> track = mTracks[i]; 7856 if (session == track->sessionId()) { 7857 track->setMainBuffer(mMixBuffer); 7858 chain->decTrackCnt(); 7859 } 7860 } 7861 break; 7862 } 7863 } 7864 return mEffectChains.size(); 7865} 7866 7867status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7868 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7869{ 7870 Mutex::Autolock _l(mLock); 7871 return attachAuxEffect_l(track, EffectId); 7872} 7873 7874status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7875 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7876{ 7877 status_t status = NO_ERROR; 7878 7879 if (EffectId == 0) { 7880 track->setAuxBuffer(0, NULL); 7881 } else { 7882 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7883 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7884 if (effect != 0) { 7885 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7886 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7887 } else { 7888 status = INVALID_OPERATION; 7889 } 7890 } else { 7891 status = BAD_VALUE; 7892 } 7893 } 7894 return status; 7895} 7896 7897void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7898{ 7899 for (size_t i = 0; i < mTracks.size(); ++i) { 7900 sp<Track> track = mTracks[i]; 7901 if (track->auxEffectId() == effectId) { 7902 attachAuxEffect_l(track, 0); 7903 } 7904 } 7905} 7906 7907status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7908{ 7909 // only one chain per input thread 7910 if (mEffectChains.size() != 0) { 7911 return INVALID_OPERATION; 7912 } 7913 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7914 7915 chain->setInBuffer(NULL); 7916 chain->setOutBuffer(NULL); 7917 7918 checkSuspendOnAddEffectChain_l(chain); 7919 7920 mEffectChains.add(chain); 7921 7922 return NO_ERROR; 7923} 7924 7925size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7926{ 7927 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7928 ALOGW_IF(mEffectChains.size() != 1, 7929 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7930 chain.get(), mEffectChains.size(), this); 7931 if (mEffectChains.size() == 1) { 7932 mEffectChains.removeAt(0); 7933 } 7934 return 0; 7935} 7936 7937// ---------------------------------------------------------------------------- 7938// EffectModule implementation 7939// ---------------------------------------------------------------------------- 7940 7941#undef LOG_TAG 7942#define LOG_TAG "AudioFlinger::EffectModule" 7943 7944AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7945 const wp<AudioFlinger::EffectChain>& chain, 7946 effect_descriptor_t *desc, 7947 int id, 7948 int sessionId) 7949 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7950 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7951 // mDescriptor is set below 7952 // mConfig is set by configure() and not used before then 7953 mEffectInterface(NULL), 7954 mStatus(NO_INIT), mState(IDLE), 7955 // mMaxDisableWaitCnt is set by configure() and not used before then 7956 // mDisableWaitCnt is set by process() and updateState() and not used before then 7957 mSuspended(false) 7958{ 7959 ALOGV("Constructor %p", this); 7960 int lStatus; 7961 if (thread == NULL) { 7962 return; 7963 } 7964 7965 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7966 7967 // create effect engine from effect factory 7968 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7969 7970 if (mStatus != NO_ERROR) { 7971 return; 7972 } 7973 lStatus = init(); 7974 if (lStatus < 0) { 7975 mStatus = lStatus; 7976 goto Error; 7977 } 7978 7979 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7980 return; 7981Error: 7982 EffectRelease(mEffectInterface); 7983 mEffectInterface = NULL; 7984 ALOGV("Constructor Error %d", mStatus); 7985} 7986 7987AudioFlinger::EffectModule::~EffectModule() 7988{ 7989 ALOGV("Destructor %p", this); 7990 if (mEffectInterface != NULL) { 7991 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7992 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7993 sp<ThreadBase> thread = mThread.promote(); 7994 if (thread != 0) { 7995 audio_stream_t *stream = thread->stream(); 7996 if (stream != NULL) { 7997 stream->remove_audio_effect(stream, mEffectInterface); 7998 } 7999 } 8000 } 8001 // release effect engine 8002 EffectRelease(mEffectInterface); 8003 } 8004} 8005 8006status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8007{ 8008 status_t status; 8009 8010 Mutex::Autolock _l(mLock); 8011 int priority = handle->priority(); 8012 size_t size = mHandles.size(); 8013 EffectHandle *controlHandle = NULL; 8014 size_t i; 8015 for (i = 0; i < size; i++) { 8016 EffectHandle *h = mHandles[i]; 8017 if (h == NULL || h->destroyed_l()) continue; 8018 // first non destroyed handle is considered in control 8019 if (controlHandle == NULL) 8020 controlHandle = h; 8021 if (h->priority() <= priority) break; 8022 } 8023 // if inserted in first place, move effect control from previous owner to this handle 8024 if (i == 0) { 8025 bool enabled = false; 8026 if (controlHandle != NULL) { 8027 enabled = controlHandle->enabled(); 8028 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8029 } 8030 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8031 status = NO_ERROR; 8032 } else { 8033 status = ALREADY_EXISTS; 8034 } 8035 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8036 mHandles.insertAt(handle, i); 8037 return status; 8038} 8039 8040size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8041{ 8042 Mutex::Autolock _l(mLock); 8043 size_t size = mHandles.size(); 8044 size_t i; 8045 for (i = 0; i < size; i++) { 8046 if (mHandles[i] == handle) break; 8047 } 8048 if (i == size) { 8049 return size; 8050 } 8051 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8052 8053 mHandles.removeAt(i); 8054 // if removed from first place, move effect control from this handle to next in line 8055 if (i == 0) { 8056 EffectHandle *h = controlHandle_l(); 8057 if (h != NULL) { 8058 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8059 } 8060 } 8061 8062 // Prevent calls to process() and other functions on effect interface from now on. 8063 // The effect engine will be released by the destructor when the last strong reference on 8064 // this object is released which can happen after next process is called. 8065 if (mHandles.size() == 0 && !mPinned) { 8066 mState = DESTROYED; 8067 } 8068 8069 return size; 8070} 8071 8072// must be called with EffectModule::mLock held 8073AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8074{ 8075 // the first valid handle in the list has control over the module 8076 for (size_t i = 0; i < mHandles.size(); i++) { 8077 EffectHandle *h = mHandles[i]; 8078 if (h != NULL && !h->destroyed_l()) { 8079 return h; 8080 } 8081 } 8082 8083 return NULL; 8084} 8085 8086size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8087{ 8088 ALOGV("disconnect() %p handle %p", this, handle); 8089 // keep a strong reference on this EffectModule to avoid calling the 8090 // destructor before we exit 8091 sp<EffectModule> keep(this); 8092 { 8093 sp<ThreadBase> thread = mThread.promote(); 8094 if (thread != 0) { 8095 thread->disconnectEffect(keep, handle, unpinIfLast); 8096 } 8097 } 8098 return mHandles.size(); 8099} 8100 8101void AudioFlinger::EffectModule::updateState() { 8102 Mutex::Autolock _l(mLock); 8103 8104 switch (mState) { 8105 case RESTART: 8106 reset_l(); 8107 // FALL THROUGH 8108 8109 case STARTING: 8110 // clear auxiliary effect input buffer for next accumulation 8111 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8112 memset(mConfig.inputCfg.buffer.raw, 8113 0, 8114 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8115 } 8116 start_l(); 8117 mState = ACTIVE; 8118 break; 8119 case STOPPING: 8120 stop_l(); 8121 mDisableWaitCnt = mMaxDisableWaitCnt; 8122 mState = STOPPED; 8123 break; 8124 case STOPPED: 8125 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8126 // turn off sequence. 8127 if (--mDisableWaitCnt == 0) { 8128 reset_l(); 8129 mState = IDLE; 8130 } 8131 break; 8132 default: //IDLE , ACTIVE, DESTROYED 8133 break; 8134 } 8135} 8136 8137void AudioFlinger::EffectModule::process() 8138{ 8139 Mutex::Autolock _l(mLock); 8140 8141 if (mState == DESTROYED || mEffectInterface == NULL || 8142 mConfig.inputCfg.buffer.raw == NULL || 8143 mConfig.outputCfg.buffer.raw == NULL) { 8144 return; 8145 } 8146 8147 if (isProcessEnabled()) { 8148 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8149 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8150 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8151 mConfig.inputCfg.buffer.s32, 8152 mConfig.inputCfg.buffer.frameCount/2); 8153 } 8154 8155 // do the actual processing in the effect engine 8156 int ret = (*mEffectInterface)->process(mEffectInterface, 8157 &mConfig.inputCfg.buffer, 8158 &mConfig.outputCfg.buffer); 8159 8160 // force transition to IDLE state when engine is ready 8161 if (mState == STOPPED && ret == -ENODATA) { 8162 mDisableWaitCnt = 1; 8163 } 8164 8165 // clear auxiliary effect input buffer for next accumulation 8166 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8167 memset(mConfig.inputCfg.buffer.raw, 0, 8168 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8169 } 8170 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8171 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8172 // If an insert effect is idle and input buffer is different from output buffer, 8173 // accumulate input onto output 8174 sp<EffectChain> chain = mChain.promote(); 8175 if (chain != 0 && chain->activeTrackCnt() != 0) { 8176 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8177 int16_t *in = mConfig.inputCfg.buffer.s16; 8178 int16_t *out = mConfig.outputCfg.buffer.s16; 8179 for (size_t i = 0; i < frameCnt; i++) { 8180 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8181 } 8182 } 8183 } 8184} 8185 8186void AudioFlinger::EffectModule::reset_l() 8187{ 8188 if (mEffectInterface == NULL) { 8189 return; 8190 } 8191 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8192} 8193 8194status_t AudioFlinger::EffectModule::configure() 8195{ 8196 if (mEffectInterface == NULL) { 8197 return NO_INIT; 8198 } 8199 8200 sp<ThreadBase> thread = mThread.promote(); 8201 if (thread == 0) { 8202 return DEAD_OBJECT; 8203 } 8204 8205 // TODO: handle configuration of effects replacing track process 8206 audio_channel_mask_t channelMask = thread->channelMask(); 8207 8208 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8209 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8210 } else { 8211 mConfig.inputCfg.channels = channelMask; 8212 } 8213 mConfig.outputCfg.channels = channelMask; 8214 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8215 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8216 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8217 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8218 mConfig.inputCfg.bufferProvider.cookie = NULL; 8219 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8220 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8221 mConfig.outputCfg.bufferProvider.cookie = NULL; 8222 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8223 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8224 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8225 // Insert effect: 8226 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8227 // always overwrites output buffer: input buffer == output buffer 8228 // - in other sessions: 8229 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8230 // other effect: overwrites output buffer: input buffer == output buffer 8231 // Auxiliary effect: 8232 // accumulates in output buffer: input buffer != output buffer 8233 // Therefore: accumulate <=> input buffer != output buffer 8234 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8235 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8236 } else { 8237 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8238 } 8239 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8240 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8241 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8242 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8243 8244 ALOGV("configure() %p thread %p buffer %p framecount %d", 8245 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8246 8247 status_t cmdStatus; 8248 uint32_t size = sizeof(int); 8249 status_t status = (*mEffectInterface)->command(mEffectInterface, 8250 EFFECT_CMD_SET_CONFIG, 8251 sizeof(effect_config_t), 8252 &mConfig, 8253 &size, 8254 &cmdStatus); 8255 if (status == 0) { 8256 status = cmdStatus; 8257 } 8258 8259 if (status == 0 && 8260 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8261 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8262 effect_param_t *p = (effect_param_t *)buf32; 8263 8264 p->psize = sizeof(uint32_t); 8265 p->vsize = sizeof(uint32_t); 8266 size = sizeof(int); 8267 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8268 8269 uint32_t latency = 0; 8270 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8271 if (pbt != NULL) { 8272 latency = pbt->latency_l(); 8273 } 8274 8275 *((int32_t *)p->data + 1)= latency; 8276 (*mEffectInterface)->command(mEffectInterface, 8277 EFFECT_CMD_SET_PARAM, 8278 sizeof(effect_param_t) + 8, 8279 &buf32, 8280 &size, 8281 &cmdStatus); 8282 } 8283 8284 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8285 (1000 * mConfig.outputCfg.buffer.frameCount); 8286 8287 return status; 8288} 8289 8290status_t AudioFlinger::EffectModule::init() 8291{ 8292 Mutex::Autolock _l(mLock); 8293 if (mEffectInterface == NULL) { 8294 return NO_INIT; 8295 } 8296 status_t cmdStatus; 8297 uint32_t size = sizeof(status_t); 8298 status_t status = (*mEffectInterface)->command(mEffectInterface, 8299 EFFECT_CMD_INIT, 8300 0, 8301 NULL, 8302 &size, 8303 &cmdStatus); 8304 if (status == 0) { 8305 status = cmdStatus; 8306 } 8307 return status; 8308} 8309 8310status_t AudioFlinger::EffectModule::start() 8311{ 8312 Mutex::Autolock _l(mLock); 8313 return start_l(); 8314} 8315 8316status_t AudioFlinger::EffectModule::start_l() 8317{ 8318 if (mEffectInterface == NULL) { 8319 return NO_INIT; 8320 } 8321 status_t cmdStatus; 8322 uint32_t size = sizeof(status_t); 8323 status_t status = (*mEffectInterface)->command(mEffectInterface, 8324 EFFECT_CMD_ENABLE, 8325 0, 8326 NULL, 8327 &size, 8328 &cmdStatus); 8329 if (status == 0) { 8330 status = cmdStatus; 8331 } 8332 if (status == 0 && 8333 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8334 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8335 sp<ThreadBase> thread = mThread.promote(); 8336 if (thread != 0) { 8337 audio_stream_t *stream = thread->stream(); 8338 if (stream != NULL) { 8339 stream->add_audio_effect(stream, mEffectInterface); 8340 } 8341 } 8342 } 8343 return status; 8344} 8345 8346status_t AudioFlinger::EffectModule::stop() 8347{ 8348 Mutex::Autolock _l(mLock); 8349 return stop_l(); 8350} 8351 8352status_t AudioFlinger::EffectModule::stop_l() 8353{ 8354 if (mEffectInterface == NULL) { 8355 return NO_INIT; 8356 } 8357 status_t cmdStatus; 8358 uint32_t size = sizeof(status_t); 8359 status_t status = (*mEffectInterface)->command(mEffectInterface, 8360 EFFECT_CMD_DISABLE, 8361 0, 8362 NULL, 8363 &size, 8364 &cmdStatus); 8365 if (status == 0) { 8366 status = cmdStatus; 8367 } 8368 if (status == 0 && 8369 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8370 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8371 sp<ThreadBase> thread = mThread.promote(); 8372 if (thread != 0) { 8373 audio_stream_t *stream = thread->stream(); 8374 if (stream != NULL) { 8375 stream->remove_audio_effect(stream, mEffectInterface); 8376 } 8377 } 8378 } 8379 return status; 8380} 8381 8382status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8383 uint32_t cmdSize, 8384 void *pCmdData, 8385 uint32_t *replySize, 8386 void *pReplyData) 8387{ 8388 Mutex::Autolock _l(mLock); 8389// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8390 8391 if (mState == DESTROYED || mEffectInterface == NULL) { 8392 return NO_INIT; 8393 } 8394 status_t status = (*mEffectInterface)->command(mEffectInterface, 8395 cmdCode, 8396 cmdSize, 8397 pCmdData, 8398 replySize, 8399 pReplyData); 8400 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8401 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8402 for (size_t i = 1; i < mHandles.size(); i++) { 8403 EffectHandle *h = mHandles[i]; 8404 if (h != NULL && !h->destroyed_l()) { 8405 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8406 } 8407 } 8408 } 8409 return status; 8410} 8411 8412status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8413{ 8414 Mutex::Autolock _l(mLock); 8415 return setEnabled_l(enabled); 8416} 8417 8418// must be called with EffectModule::mLock held 8419status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8420{ 8421 8422 ALOGV("setEnabled %p enabled %d", this, enabled); 8423 8424 if (enabled != isEnabled()) { 8425 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8426 if (enabled && status != NO_ERROR) { 8427 return status; 8428 } 8429 8430 switch (mState) { 8431 // going from disabled to enabled 8432 case IDLE: 8433 mState = STARTING; 8434 break; 8435 case STOPPED: 8436 mState = RESTART; 8437 break; 8438 case STOPPING: 8439 mState = ACTIVE; 8440 break; 8441 8442 // going from enabled to disabled 8443 case RESTART: 8444 mState = STOPPED; 8445 break; 8446 case STARTING: 8447 mState = IDLE; 8448 break; 8449 case ACTIVE: 8450 mState = STOPPING; 8451 break; 8452 case DESTROYED: 8453 return NO_ERROR; // simply ignore as we are being destroyed 8454 } 8455 for (size_t i = 1; i < mHandles.size(); i++) { 8456 EffectHandle *h = mHandles[i]; 8457 if (h != NULL && !h->destroyed_l()) { 8458 h->setEnabled(enabled); 8459 } 8460 } 8461 } 8462 return NO_ERROR; 8463} 8464 8465bool AudioFlinger::EffectModule::isEnabled() const 8466{ 8467 switch (mState) { 8468 case RESTART: 8469 case STARTING: 8470 case ACTIVE: 8471 return true; 8472 case IDLE: 8473 case STOPPING: 8474 case STOPPED: 8475 case DESTROYED: 8476 default: 8477 return false; 8478 } 8479} 8480 8481bool AudioFlinger::EffectModule::isProcessEnabled() const 8482{ 8483 switch (mState) { 8484 case RESTART: 8485 case ACTIVE: 8486 case STOPPING: 8487 case STOPPED: 8488 return true; 8489 case IDLE: 8490 case STARTING: 8491 case DESTROYED: 8492 default: 8493 return false; 8494 } 8495} 8496 8497status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8498{ 8499 Mutex::Autolock _l(mLock); 8500 status_t status = NO_ERROR; 8501 8502 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8503 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8504 if (isProcessEnabled() && 8505 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8506 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8507 status_t cmdStatus; 8508 uint32_t volume[2]; 8509 uint32_t *pVolume = NULL; 8510 uint32_t size = sizeof(volume); 8511 volume[0] = *left; 8512 volume[1] = *right; 8513 if (controller) { 8514 pVolume = volume; 8515 } 8516 status = (*mEffectInterface)->command(mEffectInterface, 8517 EFFECT_CMD_SET_VOLUME, 8518 size, 8519 volume, 8520 &size, 8521 pVolume); 8522 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8523 *left = volume[0]; 8524 *right = volume[1]; 8525 } 8526 } 8527 return status; 8528} 8529 8530status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8531{ 8532 Mutex::Autolock _l(mLock); 8533 status_t status = NO_ERROR; 8534 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8535 // audio pre processing modules on RecordThread can receive both output and 8536 // input device indication in the same call 8537 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8538 if (dev) { 8539 status_t cmdStatus; 8540 uint32_t size = sizeof(status_t); 8541 8542 status = (*mEffectInterface)->command(mEffectInterface, 8543 EFFECT_CMD_SET_DEVICE, 8544 sizeof(uint32_t), 8545 &dev, 8546 &size, 8547 &cmdStatus); 8548 if (status == NO_ERROR) { 8549 status = cmdStatus; 8550 } 8551 } 8552 dev = device & AUDIO_DEVICE_IN_ALL; 8553 if (dev) { 8554 status_t cmdStatus; 8555 uint32_t size = sizeof(status_t); 8556 8557 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8558 EFFECT_CMD_SET_INPUT_DEVICE, 8559 sizeof(uint32_t), 8560 &dev, 8561 &size, 8562 &cmdStatus); 8563 if (status2 == NO_ERROR) { 8564 status2 = cmdStatus; 8565 } 8566 if (status == NO_ERROR) { 8567 status = status2; 8568 } 8569 } 8570 } 8571 return status; 8572} 8573 8574status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8575{ 8576 Mutex::Autolock _l(mLock); 8577 status_t status = NO_ERROR; 8578 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8579 status_t cmdStatus; 8580 uint32_t size = sizeof(status_t); 8581 status = (*mEffectInterface)->command(mEffectInterface, 8582 EFFECT_CMD_SET_AUDIO_MODE, 8583 sizeof(audio_mode_t), 8584 &mode, 8585 &size, 8586 &cmdStatus); 8587 if (status == NO_ERROR) { 8588 status = cmdStatus; 8589 } 8590 } 8591 return status; 8592} 8593 8594void AudioFlinger::EffectModule::setSuspended(bool suspended) 8595{ 8596 Mutex::Autolock _l(mLock); 8597 mSuspended = suspended; 8598} 8599 8600bool AudioFlinger::EffectModule::suspended() const 8601{ 8602 Mutex::Autolock _l(mLock); 8603 return mSuspended; 8604} 8605 8606bool AudioFlinger::EffectModule::purgeHandles() 8607{ 8608 bool enabled = false; 8609 Mutex::Autolock _l(mLock); 8610 for (size_t i = 0; i < mHandles.size(); i++) { 8611 EffectHandle *handle = mHandles[i]; 8612 if (handle != NULL && !handle->destroyed_l()) { 8613 handle->effect().clear(); 8614 if (handle->hasControl()) { 8615 enabled = handle->enabled(); 8616 } 8617 } 8618 } 8619 return enabled; 8620} 8621 8622status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8623{ 8624 const size_t SIZE = 256; 8625 char buffer[SIZE]; 8626 String8 result; 8627 8628 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8629 result.append(buffer); 8630 8631 bool locked = tryLock(mLock); 8632 // failed to lock - AudioFlinger is probably deadlocked 8633 if (!locked) { 8634 result.append("\t\tCould not lock Fx mutex:\n"); 8635 } 8636 8637 result.append("\t\tSession Status State Engine:\n"); 8638 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8639 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8640 result.append(buffer); 8641 8642 result.append("\t\tDescriptor:\n"); 8643 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8644 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8645 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8646 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8647 result.append(buffer); 8648 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8649 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8650 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8651 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8652 result.append(buffer); 8653 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8654 mDescriptor.apiVersion, 8655 mDescriptor.flags); 8656 result.append(buffer); 8657 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8658 mDescriptor.name); 8659 result.append(buffer); 8660 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8661 mDescriptor.implementor); 8662 result.append(buffer); 8663 8664 result.append("\t\t- Input configuration:\n"); 8665 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8666 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8667 (uint32_t)mConfig.inputCfg.buffer.raw, 8668 mConfig.inputCfg.buffer.frameCount, 8669 mConfig.inputCfg.samplingRate, 8670 mConfig.inputCfg.channels, 8671 mConfig.inputCfg.format); 8672 result.append(buffer); 8673 8674 result.append("\t\t- Output configuration:\n"); 8675 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8676 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8677 (uint32_t)mConfig.outputCfg.buffer.raw, 8678 mConfig.outputCfg.buffer.frameCount, 8679 mConfig.outputCfg.samplingRate, 8680 mConfig.outputCfg.channels, 8681 mConfig.outputCfg.format); 8682 result.append(buffer); 8683 8684 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8685 result.append(buffer); 8686 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8687 for (size_t i = 0; i < mHandles.size(); ++i) { 8688 EffectHandle *handle = mHandles[i]; 8689 if (handle != NULL && !handle->destroyed_l()) { 8690 handle->dump(buffer, SIZE); 8691 result.append(buffer); 8692 } 8693 } 8694 8695 result.append("\n"); 8696 8697 write(fd, result.string(), result.length()); 8698 8699 if (locked) { 8700 mLock.unlock(); 8701 } 8702 8703 return NO_ERROR; 8704} 8705 8706// ---------------------------------------------------------------------------- 8707// EffectHandle implementation 8708// ---------------------------------------------------------------------------- 8709 8710#undef LOG_TAG 8711#define LOG_TAG "AudioFlinger::EffectHandle" 8712 8713AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8714 const sp<AudioFlinger::Client>& client, 8715 const sp<IEffectClient>& effectClient, 8716 int32_t priority) 8717 : BnEffect(), 8718 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8719 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8720{ 8721 ALOGV("constructor %p", this); 8722 8723 if (client == 0) { 8724 return; 8725 } 8726 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8727 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8728 if (mCblkMemory != 0) { 8729 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8730 8731 if (mCblk != NULL) { 8732 new(mCblk) effect_param_cblk_t(); 8733 mBuffer = (uint8_t *)mCblk + bufOffset; 8734 } 8735 } else { 8736 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8737 return; 8738 } 8739} 8740 8741AudioFlinger::EffectHandle::~EffectHandle() 8742{ 8743 ALOGV("Destructor %p", this); 8744 8745 if (mEffect == 0) { 8746 mDestroyed = true; 8747 return; 8748 } 8749 mEffect->lock(); 8750 mDestroyed = true; 8751 mEffect->unlock(); 8752 disconnect(false); 8753} 8754 8755status_t AudioFlinger::EffectHandle::enable() 8756{ 8757 ALOGV("enable %p", this); 8758 if (!mHasControl) return INVALID_OPERATION; 8759 if (mEffect == 0) return DEAD_OBJECT; 8760 8761 if (mEnabled) { 8762 return NO_ERROR; 8763 } 8764 8765 mEnabled = true; 8766 8767 sp<ThreadBase> thread = mEffect->thread().promote(); 8768 if (thread != 0) { 8769 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8770 } 8771 8772 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8773 if (mEffect->suspended()) { 8774 return NO_ERROR; 8775 } 8776 8777 status_t status = mEffect->setEnabled(true); 8778 if (status != NO_ERROR) { 8779 if (thread != 0) { 8780 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8781 } 8782 mEnabled = false; 8783 } 8784 return status; 8785} 8786 8787status_t AudioFlinger::EffectHandle::disable() 8788{ 8789 ALOGV("disable %p", this); 8790 if (!mHasControl) return INVALID_OPERATION; 8791 if (mEffect == 0) return DEAD_OBJECT; 8792 8793 if (!mEnabled) { 8794 return NO_ERROR; 8795 } 8796 mEnabled = false; 8797 8798 if (mEffect->suspended()) { 8799 return NO_ERROR; 8800 } 8801 8802 status_t status = mEffect->setEnabled(false); 8803 8804 sp<ThreadBase> thread = mEffect->thread().promote(); 8805 if (thread != 0) { 8806 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8807 } 8808 8809 return status; 8810} 8811 8812void AudioFlinger::EffectHandle::disconnect() 8813{ 8814 disconnect(true); 8815} 8816 8817void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8818{ 8819 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8820 if (mEffect == 0) { 8821 return; 8822 } 8823 // restore suspended effects if the disconnected handle was enabled and the last one. 8824 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8825 sp<ThreadBase> thread = mEffect->thread().promote(); 8826 if (thread != 0) { 8827 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8828 } 8829 } 8830 8831 // release sp on module => module destructor can be called now 8832 mEffect.clear(); 8833 if (mClient != 0) { 8834 if (mCblk != NULL) { 8835 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8836 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8837 } 8838 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8839 // Client destructor must run with AudioFlinger mutex locked 8840 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8841 mClient.clear(); 8842 } 8843} 8844 8845status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8846 uint32_t cmdSize, 8847 void *pCmdData, 8848 uint32_t *replySize, 8849 void *pReplyData) 8850{ 8851// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8852// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8853 8854 // only get parameter command is permitted for applications not controlling the effect 8855 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8856 return INVALID_OPERATION; 8857 } 8858 if (mEffect == 0) return DEAD_OBJECT; 8859 if (mClient == 0) return INVALID_OPERATION; 8860 8861 // handle commands that are not forwarded transparently to effect engine 8862 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8863 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8864 // no risk to block the whole media server process or mixer threads is we are stuck here 8865 Mutex::Autolock _l(mCblk->lock); 8866 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8867 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8868 mCblk->serverIndex = 0; 8869 mCblk->clientIndex = 0; 8870 return BAD_VALUE; 8871 } 8872 status_t status = NO_ERROR; 8873 while (mCblk->serverIndex < mCblk->clientIndex) { 8874 int reply; 8875 uint32_t rsize = sizeof(int); 8876 int *p = (int *)(mBuffer + mCblk->serverIndex); 8877 int size = *p++; 8878 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8879 ALOGW("command(): invalid parameter block size"); 8880 break; 8881 } 8882 effect_param_t *param = (effect_param_t *)p; 8883 if (param->psize == 0 || param->vsize == 0) { 8884 ALOGW("command(): null parameter or value size"); 8885 mCblk->serverIndex += size; 8886 continue; 8887 } 8888 uint32_t psize = sizeof(effect_param_t) + 8889 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8890 param->vsize; 8891 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8892 psize, 8893 p, 8894 &rsize, 8895 &reply); 8896 // stop at first error encountered 8897 if (ret != NO_ERROR) { 8898 status = ret; 8899 *(int *)pReplyData = reply; 8900 break; 8901 } else if (reply != NO_ERROR) { 8902 *(int *)pReplyData = reply; 8903 break; 8904 } 8905 mCblk->serverIndex += size; 8906 } 8907 mCblk->serverIndex = 0; 8908 mCblk->clientIndex = 0; 8909 return status; 8910 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8911 *(int *)pReplyData = NO_ERROR; 8912 return enable(); 8913 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8914 *(int *)pReplyData = NO_ERROR; 8915 return disable(); 8916 } 8917 8918 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8919} 8920 8921void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8922{ 8923 ALOGV("setControl %p control %d", this, hasControl); 8924 8925 mHasControl = hasControl; 8926 mEnabled = enabled; 8927 8928 if (signal && mEffectClient != 0) { 8929 mEffectClient->controlStatusChanged(hasControl); 8930 } 8931} 8932 8933void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8934 uint32_t cmdSize, 8935 void *pCmdData, 8936 uint32_t replySize, 8937 void *pReplyData) 8938{ 8939 if (mEffectClient != 0) { 8940 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8941 } 8942} 8943 8944 8945 8946void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8947{ 8948 if (mEffectClient != 0) { 8949 mEffectClient->enableStatusChanged(enabled); 8950 } 8951} 8952 8953status_t AudioFlinger::EffectHandle::onTransact( 8954 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8955{ 8956 return BnEffect::onTransact(code, data, reply, flags); 8957} 8958 8959 8960void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8961{ 8962 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8963 8964 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8965 (mClient == 0) ? getpid_cached : mClient->pid(), 8966 mPriority, 8967 mHasControl, 8968 !locked, 8969 mCblk ? mCblk->clientIndex : 0, 8970 mCblk ? mCblk->serverIndex : 0 8971 ); 8972 8973 if (locked) { 8974 mCblk->lock.unlock(); 8975 } 8976} 8977 8978#undef LOG_TAG 8979#define LOG_TAG "AudioFlinger::EffectChain" 8980 8981AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8982 int sessionId) 8983 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8984 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8985 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8986{ 8987 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8988 if (thread == NULL) { 8989 return; 8990 } 8991 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8992 thread->frameCount(); 8993} 8994 8995AudioFlinger::EffectChain::~EffectChain() 8996{ 8997 if (mOwnInBuffer) { 8998 delete mInBuffer; 8999 } 9000 9001} 9002 9003// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9004sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9005{ 9006 size_t size = mEffects.size(); 9007 9008 for (size_t i = 0; i < size; i++) { 9009 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9010 return mEffects[i]; 9011 } 9012 } 9013 return 0; 9014} 9015 9016// getEffectFromId_l() must be called with ThreadBase::mLock held 9017sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9018{ 9019 size_t size = mEffects.size(); 9020 9021 for (size_t i = 0; i < size; i++) { 9022 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9023 if (id == 0 || mEffects[i]->id() == id) { 9024 return mEffects[i]; 9025 } 9026 } 9027 return 0; 9028} 9029 9030// getEffectFromType_l() must be called with ThreadBase::mLock held 9031sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9032 const effect_uuid_t *type) 9033{ 9034 size_t size = mEffects.size(); 9035 9036 for (size_t i = 0; i < size; i++) { 9037 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9038 return mEffects[i]; 9039 } 9040 } 9041 return 0; 9042} 9043 9044void AudioFlinger::EffectChain::clearInputBuffer() 9045{ 9046 Mutex::Autolock _l(mLock); 9047 sp<ThreadBase> thread = mThread.promote(); 9048 if (thread == 0) { 9049 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9050 return; 9051 } 9052 clearInputBuffer_l(thread); 9053} 9054 9055// Must be called with EffectChain::mLock locked 9056void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9057{ 9058 size_t numSamples = thread->frameCount() * thread->channelCount(); 9059 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9060 9061} 9062 9063// Must be called with EffectChain::mLock locked 9064void AudioFlinger::EffectChain::process_l() 9065{ 9066 sp<ThreadBase> thread = mThread.promote(); 9067 if (thread == 0) { 9068 ALOGW("process_l(): cannot promote mixer thread"); 9069 return; 9070 } 9071 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9072 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9073 // always process effects unless no more tracks are on the session and the effect tail 9074 // has been rendered 9075 bool doProcess = true; 9076 if (!isGlobalSession) { 9077 bool tracksOnSession = (trackCnt() != 0); 9078 9079 if (!tracksOnSession && mTailBufferCount == 0) { 9080 doProcess = false; 9081 } 9082 9083 if (activeTrackCnt() == 0) { 9084 // if no track is active and the effect tail has not been rendered, 9085 // the input buffer must be cleared here as the mixer process will not do it 9086 if (tracksOnSession || mTailBufferCount > 0) { 9087 clearInputBuffer_l(thread); 9088 if (mTailBufferCount > 0) { 9089 mTailBufferCount--; 9090 } 9091 } 9092 } 9093 } 9094 9095 size_t size = mEffects.size(); 9096 if (doProcess) { 9097 for (size_t i = 0; i < size; i++) { 9098 mEffects[i]->process(); 9099 } 9100 } 9101 for (size_t i = 0; i < size; i++) { 9102 mEffects[i]->updateState(); 9103 } 9104} 9105 9106// addEffect_l() must be called with PlaybackThread::mLock held 9107status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9108{ 9109 effect_descriptor_t desc = effect->desc(); 9110 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9111 9112 Mutex::Autolock _l(mLock); 9113 effect->setChain(this); 9114 sp<ThreadBase> thread = mThread.promote(); 9115 if (thread == 0) { 9116 return NO_INIT; 9117 } 9118 effect->setThread(thread); 9119 9120 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9121 // Auxiliary effects are inserted at the beginning of mEffects vector as 9122 // they are processed first and accumulated in chain input buffer 9123 mEffects.insertAt(effect, 0); 9124 9125 // the input buffer for auxiliary effect contains mono samples in 9126 // 32 bit format. This is to avoid saturation in AudoMixer 9127 // accumulation stage. Saturation is done in EffectModule::process() before 9128 // calling the process in effect engine 9129 size_t numSamples = thread->frameCount(); 9130 int32_t *buffer = new int32_t[numSamples]; 9131 memset(buffer, 0, numSamples * sizeof(int32_t)); 9132 effect->setInBuffer((int16_t *)buffer); 9133 // auxiliary effects output samples to chain input buffer for further processing 9134 // by insert effects 9135 effect->setOutBuffer(mInBuffer); 9136 } else { 9137 // Insert effects are inserted at the end of mEffects vector as they are processed 9138 // after track and auxiliary effects. 9139 // Insert effect order as a function of indicated preference: 9140 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9141 // another effect is present 9142 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9143 // last effect claiming first position 9144 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9145 // first effect claiming last position 9146 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9147 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9148 // already present 9149 9150 size_t size = mEffects.size(); 9151 size_t idx_insert = size; 9152 ssize_t idx_insert_first = -1; 9153 ssize_t idx_insert_last = -1; 9154 9155 for (size_t i = 0; i < size; i++) { 9156 effect_descriptor_t d = mEffects[i]->desc(); 9157 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9158 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9159 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9160 // check invalid effect chaining combinations 9161 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9162 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9163 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9164 return INVALID_OPERATION; 9165 } 9166 // remember position of first insert effect and by default 9167 // select this as insert position for new effect 9168 if (idx_insert == size) { 9169 idx_insert = i; 9170 } 9171 // remember position of last insert effect claiming 9172 // first position 9173 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9174 idx_insert_first = i; 9175 } 9176 // remember position of first insert effect claiming 9177 // last position 9178 if (iPref == EFFECT_FLAG_INSERT_LAST && 9179 idx_insert_last == -1) { 9180 idx_insert_last = i; 9181 } 9182 } 9183 } 9184 9185 // modify idx_insert from first position if needed 9186 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9187 if (idx_insert_last != -1) { 9188 idx_insert = idx_insert_last; 9189 } else { 9190 idx_insert = size; 9191 } 9192 } else { 9193 if (idx_insert_first != -1) { 9194 idx_insert = idx_insert_first + 1; 9195 } 9196 } 9197 9198 // always read samples from chain input buffer 9199 effect->setInBuffer(mInBuffer); 9200 9201 // if last effect in the chain, output samples to chain 9202 // output buffer, otherwise to chain input buffer 9203 if (idx_insert == size) { 9204 if (idx_insert != 0) { 9205 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9206 mEffects[idx_insert-1]->configure(); 9207 } 9208 effect->setOutBuffer(mOutBuffer); 9209 } else { 9210 effect->setOutBuffer(mInBuffer); 9211 } 9212 mEffects.insertAt(effect, idx_insert); 9213 9214 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9215 } 9216 effect->configure(); 9217 return NO_ERROR; 9218} 9219 9220// removeEffect_l() must be called with PlaybackThread::mLock held 9221size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9222{ 9223 Mutex::Autolock _l(mLock); 9224 size_t size = mEffects.size(); 9225 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9226 9227 for (size_t i = 0; i < size; i++) { 9228 if (effect == mEffects[i]) { 9229 // calling stop here will remove pre-processing effect from the audio HAL. 9230 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9231 // the middle of a read from audio HAL 9232 if (mEffects[i]->state() == EffectModule::ACTIVE || 9233 mEffects[i]->state() == EffectModule::STOPPING) { 9234 mEffects[i]->stop(); 9235 } 9236 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9237 delete[] effect->inBuffer(); 9238 } else { 9239 if (i == size - 1 && i != 0) { 9240 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9241 mEffects[i - 1]->configure(); 9242 } 9243 } 9244 mEffects.removeAt(i); 9245 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9246 break; 9247 } 9248 } 9249 9250 return mEffects.size(); 9251} 9252 9253// setDevice_l() must be called with PlaybackThread::mLock held 9254void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9255{ 9256 size_t size = mEffects.size(); 9257 for (size_t i = 0; i < size; i++) { 9258 mEffects[i]->setDevice(device); 9259 } 9260} 9261 9262// setMode_l() must be called with PlaybackThread::mLock held 9263void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9264{ 9265 size_t size = mEffects.size(); 9266 for (size_t i = 0; i < size; i++) { 9267 mEffects[i]->setMode(mode); 9268 } 9269} 9270 9271// setVolume_l() must be called with PlaybackThread::mLock held 9272bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9273{ 9274 uint32_t newLeft = *left; 9275 uint32_t newRight = *right; 9276 bool hasControl = false; 9277 int ctrlIdx = -1; 9278 size_t size = mEffects.size(); 9279 9280 // first update volume controller 9281 for (size_t i = size; i > 0; i--) { 9282 if (mEffects[i - 1]->isProcessEnabled() && 9283 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9284 ctrlIdx = i - 1; 9285 hasControl = true; 9286 break; 9287 } 9288 } 9289 9290 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9291 if (hasControl) { 9292 *left = mNewLeftVolume; 9293 *right = mNewRightVolume; 9294 } 9295 return hasControl; 9296 } 9297 9298 mVolumeCtrlIdx = ctrlIdx; 9299 mLeftVolume = newLeft; 9300 mRightVolume = newRight; 9301 9302 // second get volume update from volume controller 9303 if (ctrlIdx >= 0) { 9304 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9305 mNewLeftVolume = newLeft; 9306 mNewRightVolume = newRight; 9307 } 9308 // then indicate volume to all other effects in chain. 9309 // Pass altered volume to effects before volume controller 9310 // and requested volume to effects after controller 9311 uint32_t lVol = newLeft; 9312 uint32_t rVol = newRight; 9313 9314 for (size_t i = 0; i < size; i++) { 9315 if ((int)i == ctrlIdx) continue; 9316 // this also works for ctrlIdx == -1 when there is no volume controller 9317 if ((int)i > ctrlIdx) { 9318 lVol = *left; 9319 rVol = *right; 9320 } 9321 mEffects[i]->setVolume(&lVol, &rVol, false); 9322 } 9323 *left = newLeft; 9324 *right = newRight; 9325 9326 return hasControl; 9327} 9328 9329status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9330{ 9331 const size_t SIZE = 256; 9332 char buffer[SIZE]; 9333 String8 result; 9334 9335 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9336 result.append(buffer); 9337 9338 bool locked = tryLock(mLock); 9339 // failed to lock - AudioFlinger is probably deadlocked 9340 if (!locked) { 9341 result.append("\tCould not lock mutex:\n"); 9342 } 9343 9344 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9345 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9346 mEffects.size(), 9347 (uint32_t)mInBuffer, 9348 (uint32_t)mOutBuffer, 9349 mActiveTrackCnt); 9350 result.append(buffer); 9351 write(fd, result.string(), result.size()); 9352 9353 for (size_t i = 0; i < mEffects.size(); ++i) { 9354 sp<EffectModule> effect = mEffects[i]; 9355 if (effect != 0) { 9356 effect->dump(fd, args); 9357 } 9358 } 9359 9360 if (locked) { 9361 mLock.unlock(); 9362 } 9363 9364 return NO_ERROR; 9365} 9366 9367// must be called with ThreadBase::mLock held 9368void AudioFlinger::EffectChain::setEffectSuspended_l( 9369 const effect_uuid_t *type, bool suspend) 9370{ 9371 sp<SuspendedEffectDesc> desc; 9372 // use effect type UUID timelow as key as there is no real risk of identical 9373 // timeLow fields among effect type UUIDs. 9374 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9375 if (suspend) { 9376 if (index >= 0) { 9377 desc = mSuspendedEffects.valueAt(index); 9378 } else { 9379 desc = new SuspendedEffectDesc(); 9380 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9381 mSuspendedEffects.add(type->timeLow, desc); 9382 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9383 } 9384 if (desc->mRefCount++ == 0) { 9385 sp<EffectModule> effect = getEffectIfEnabled(type); 9386 if (effect != 0) { 9387 desc->mEffect = effect; 9388 effect->setSuspended(true); 9389 effect->setEnabled(false); 9390 } 9391 } 9392 } else { 9393 if (index < 0) { 9394 return; 9395 } 9396 desc = mSuspendedEffects.valueAt(index); 9397 if (desc->mRefCount <= 0) { 9398 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9399 desc->mRefCount = 1; 9400 } 9401 if (--desc->mRefCount == 0) { 9402 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9403 if (desc->mEffect != 0) { 9404 sp<EffectModule> effect = desc->mEffect.promote(); 9405 if (effect != 0) { 9406 effect->setSuspended(false); 9407 effect->lock(); 9408 EffectHandle *handle = effect->controlHandle_l(); 9409 if (handle != NULL && !handle->destroyed_l()) { 9410 effect->setEnabled_l(handle->enabled()); 9411 } 9412 effect->unlock(); 9413 } 9414 desc->mEffect.clear(); 9415 } 9416 mSuspendedEffects.removeItemsAt(index); 9417 } 9418 } 9419} 9420 9421// must be called with ThreadBase::mLock held 9422void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9423{ 9424 sp<SuspendedEffectDesc> desc; 9425 9426 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9427 if (suspend) { 9428 if (index >= 0) { 9429 desc = mSuspendedEffects.valueAt(index); 9430 } else { 9431 desc = new SuspendedEffectDesc(); 9432 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9433 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9434 } 9435 if (desc->mRefCount++ == 0) { 9436 Vector< sp<EffectModule> > effects; 9437 getSuspendEligibleEffects(effects); 9438 for (size_t i = 0; i < effects.size(); i++) { 9439 setEffectSuspended_l(&effects[i]->desc().type, true); 9440 } 9441 } 9442 } else { 9443 if (index < 0) { 9444 return; 9445 } 9446 desc = mSuspendedEffects.valueAt(index); 9447 if (desc->mRefCount <= 0) { 9448 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9449 desc->mRefCount = 1; 9450 } 9451 if (--desc->mRefCount == 0) { 9452 Vector<const effect_uuid_t *> types; 9453 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9454 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9455 continue; 9456 } 9457 types.add(&mSuspendedEffects.valueAt(i)->mType); 9458 } 9459 for (size_t i = 0; i < types.size(); i++) { 9460 setEffectSuspended_l(types[i], false); 9461 } 9462 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9463 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9464 } 9465 } 9466} 9467 9468 9469// The volume effect is used for automated tests only 9470#ifndef OPENSL_ES_H_ 9471static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9472 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9473const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9474#endif //OPENSL_ES_H_ 9475 9476bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9477{ 9478 // auxiliary effects and visualizer are never suspended on output mix 9479 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9480 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9481 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9482 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9483 return false; 9484 } 9485 return true; 9486} 9487 9488void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9489{ 9490 effects.clear(); 9491 for (size_t i = 0; i < mEffects.size(); i++) { 9492 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9493 effects.add(mEffects[i]); 9494 } 9495 } 9496} 9497 9498sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9499 const effect_uuid_t *type) 9500{ 9501 sp<EffectModule> effect = getEffectFromType_l(type); 9502 return effect != 0 && effect->isEnabled() ? effect : 0; 9503} 9504 9505void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9506 bool enabled) 9507{ 9508 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9509 if (enabled) { 9510 if (index < 0) { 9511 // if the effect is not suspend check if all effects are suspended 9512 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9513 if (index < 0) { 9514 return; 9515 } 9516 if (!isEffectEligibleForSuspend(effect->desc())) { 9517 return; 9518 } 9519 setEffectSuspended_l(&effect->desc().type, enabled); 9520 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9521 if (index < 0) { 9522 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9523 return; 9524 } 9525 } 9526 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9527 effect->desc().type.timeLow); 9528 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9529 // if effect is requested to suspended but was not yet enabled, supend it now. 9530 if (desc->mEffect == 0) { 9531 desc->mEffect = effect; 9532 effect->setEnabled(false); 9533 effect->setSuspended(true); 9534 } 9535 } else { 9536 if (index < 0) { 9537 return; 9538 } 9539 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9540 effect->desc().type.timeLow); 9541 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9542 desc->mEffect.clear(); 9543 effect->setSuspended(false); 9544 } 9545} 9546 9547#undef LOG_TAG 9548#define LOG_TAG "AudioFlinger" 9549 9550// ---------------------------------------------------------------------------- 9551 9552status_t AudioFlinger::onTransact( 9553 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9554{ 9555 return BnAudioFlinger::onTransact(code, data, reply, flags); 9556} 9557 9558}; // namespace android 9559