AudioFlinger.cpp revision 6648821933dc06c0b09ab2c8b32135edddcd4291
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), // see also onFirstRef() 214 mMasterVolume(1.0f), 215 mMasterVolumeSupportLvl(MVS_NONE), 216 mMasterMute(false), 217 mNextUniqueId(1), 218 mMode(AUDIO_MODE_INVALID), 219 mBtNrecIsOff(false) 220{ 221} 222 223void AudioFlinger::onFirstRef() 224{ 225 int rc = 0; 226 227 Mutex::Autolock _l(mLock); 228 229 /* TODO: move all this work into an Init() function */ 230 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 231 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 232 uint32_t int_val; 233 if (1 == sscanf(val_str, "%u", &int_val)) { 234 mStandbyTimeInNsecs = milliseconds(int_val); 235 ALOGI("Using %u mSec as standby time.", int_val); 236 } else { 237 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 238 ALOGI("Using default %u mSec as standby time.", 239 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 240 } 241 } 242 243 mMode = AUDIO_MODE_NORMAL; 244 mMasterVolumeSW = 1.0; 245 mMasterVolume = 1.0; 246 mHardwareStatus = AUDIO_HW_IDLE; 247} 248 249AudioFlinger::~AudioFlinger() 250{ 251 252 while (!mRecordThreads.isEmpty()) { 253 // closeInput() will remove first entry from mRecordThreads 254 closeInput(mRecordThreads.keyAt(0)); 255 } 256 while (!mPlaybackThreads.isEmpty()) { 257 // closeOutput() will remove first entry from mPlaybackThreads 258 closeOutput(mPlaybackThreads.keyAt(0)); 259 } 260 261 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 262 // no mHardwareLock needed, as there are no other references to this 263 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 264 delete mAudioHwDevs.valueAt(i); 265 } 266} 267 268static const char * const audio_interfaces[] = { 269 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 270 AUDIO_HARDWARE_MODULE_ID_A2DP, 271 AUDIO_HARDWARE_MODULE_ID_USB, 272}; 273#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 274 275audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 276{ 277 // if module is 0, the request comes from an old policy manager and we should load 278 // well known modules 279 if (module == 0) { 280 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 281 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 282 loadHwModule_l(audio_interfaces[i]); 283 } 284 } else { 285 // check a match for the requested module handle 286 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 287 if (audioHwdevice != NULL) { 288 return audioHwdevice->hwDevice(); 289 } 290 } 291 // then try to find a module supporting the requested device. 292 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 293 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 294 if ((dev->get_supported_devices(dev) & devices) == devices) 295 return dev; 296 } 297 298 return NULL; 299} 300 301status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 302{ 303 const size_t SIZE = 256; 304 char buffer[SIZE]; 305 String8 result; 306 307 result.append("Clients:\n"); 308 for (size_t i = 0; i < mClients.size(); ++i) { 309 sp<Client> client = mClients.valueAt(i).promote(); 310 if (client != 0) { 311 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 312 result.append(buffer); 313 } 314 } 315 316 result.append("Global session refs:\n"); 317 result.append(" session pid count\n"); 318 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 319 AudioSessionRef *r = mAudioSessionRefs[i]; 320 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 321 result.append(buffer); 322 } 323 write(fd, result.string(), result.size()); 324 return NO_ERROR; 325} 326 327 328status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 329{ 330 const size_t SIZE = 256; 331 char buffer[SIZE]; 332 String8 result; 333 hardware_call_state hardwareStatus = mHardwareStatus; 334 335 snprintf(buffer, SIZE, "Hardware status: %d\n" 336 "Standby Time mSec: %u\n", 337 hardwareStatus, 338 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 339 result.append(buffer); 340 write(fd, result.string(), result.size()); 341 return NO_ERROR; 342} 343 344status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 345{ 346 const size_t SIZE = 256; 347 char buffer[SIZE]; 348 String8 result; 349 snprintf(buffer, SIZE, "Permission Denial: " 350 "can't dump AudioFlinger from pid=%d, uid=%d\n", 351 IPCThreadState::self()->getCallingPid(), 352 IPCThreadState::self()->getCallingUid()); 353 result.append(buffer); 354 write(fd, result.string(), result.size()); 355 return NO_ERROR; 356} 357 358static bool tryLock(Mutex& mutex) 359{ 360 bool locked = false; 361 for (int i = 0; i < kDumpLockRetries; ++i) { 362 if (mutex.tryLock() == NO_ERROR) { 363 locked = true; 364 break; 365 } 366 usleep(kDumpLockSleepUs); 367 } 368 return locked; 369} 370 371status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 372{ 373 if (!dumpAllowed()) { 374 dumpPermissionDenial(fd, args); 375 } else { 376 // get state of hardware lock 377 bool hardwareLocked = tryLock(mHardwareLock); 378 if (!hardwareLocked) { 379 String8 result(kHardwareLockedString); 380 write(fd, result.string(), result.size()); 381 } else { 382 mHardwareLock.unlock(); 383 } 384 385 bool locked = tryLock(mLock); 386 387 // failed to lock - AudioFlinger is probably deadlocked 388 if (!locked) { 389 String8 result(kDeadlockedString); 390 write(fd, result.string(), result.size()); 391 } 392 393 dumpClients(fd, args); 394 dumpInternals(fd, args); 395 396 // dump playback threads 397 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 398 mPlaybackThreads.valueAt(i)->dump(fd, args); 399 } 400 401 // dump record threads 402 for (size_t i = 0; i < mRecordThreads.size(); i++) { 403 mRecordThreads.valueAt(i)->dump(fd, args); 404 } 405 406 // dump all hardware devs 407 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 408 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 409 dev->dump(dev, fd); 410 } 411 if (locked) mLock.unlock(); 412 } 413 return NO_ERROR; 414} 415 416sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 417{ 418 // If pid is already in the mClients wp<> map, then use that entry 419 // (for which promote() is always != 0), otherwise create a new entry and Client. 420 sp<Client> client = mClients.valueFor(pid).promote(); 421 if (client == 0) { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 return client; 427} 428 429// IAudioFlinger interface 430 431 432sp<IAudioTrack> AudioFlinger::createTrack( 433 pid_t pid, 434 audio_stream_type_t streamType, 435 uint32_t sampleRate, 436 audio_format_t format, 437 uint32_t channelMask, 438 int frameCount, 439 IAudioFlinger::track_flags_t flags, 440 const sp<IMemory>& sharedBuffer, 441 audio_io_handle_t output, 442 pid_t tid, 443 int *sessionId, 444 status_t *status) 445{ 446 sp<PlaybackThread::Track> track; 447 sp<TrackHandle> trackHandle; 448 sp<Client> client; 449 status_t lStatus; 450 int lSessionId; 451 452 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 453 // but if someone uses binder directly they could bypass that and cause us to crash 454 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 455 ALOGE("createTrack() invalid stream type %d", streamType); 456 lStatus = BAD_VALUE; 457 goto Exit; 458 } 459 460 { 461 Mutex::Autolock _l(mLock); 462 PlaybackThread *thread = checkPlaybackThread_l(output); 463 PlaybackThread *effectThread = NULL; 464 if (thread == NULL) { 465 ALOGE("unknown output thread"); 466 lStatus = BAD_VALUE; 467 goto Exit; 468 } 469 470 client = registerPid_l(pid); 471 472 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 473 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 474 // check if an effect chain with the same session ID is present on another 475 // output thread and move it here. 476 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 477 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 478 if (mPlaybackThreads.keyAt(i) != output) { 479 uint32_t sessions = t->hasAudioSession(*sessionId); 480 if (sessions & PlaybackThread::EFFECT_SESSION) { 481 effectThread = t.get(); 482 break; 483 } 484 } 485 } 486 lSessionId = *sessionId; 487 } else { 488 // if no audio session id is provided, create one here 489 lSessionId = nextUniqueId(); 490 if (sessionId != NULL) { 491 *sessionId = lSessionId; 492 } 493 } 494 ALOGV("createTrack() lSessionId: %d", lSessionId); 495 496 track = thread->createTrack_l(client, streamType, sampleRate, format, 497 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 498 499 // move effect chain to this output thread if an effect on same session was waiting 500 // for a track to be created 501 if (lStatus == NO_ERROR && effectThread != NULL) { 502 Mutex::Autolock _dl(thread->mLock); 503 Mutex::Autolock _sl(effectThread->mLock); 504 moveEffectChain_l(lSessionId, effectThread, thread, true); 505 } 506 507 // Look for sync events awaiting for a session to be used. 508 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 509 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 510 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 511 if (lStatus == NO_ERROR) { 512 track->setSyncEvent(mPendingSyncEvents[i]); 513 } else { 514 mPendingSyncEvents[i]->cancel(); 515 } 516 mPendingSyncEvents.removeAt(i); 517 i--; 518 } 519 } 520 } 521 } 522 if (lStatus == NO_ERROR) { 523 trackHandle = new TrackHandle(track); 524 } else { 525 // remove local strong reference to Client before deleting the Track so that the Client 526 // destructor is called by the TrackBase destructor with mLock held 527 client.clear(); 528 track.clear(); 529 } 530 531Exit: 532 if (status != NULL) { 533 *status = lStatus; 534 } 535 return trackHandle; 536} 537 538uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 539{ 540 Mutex::Autolock _l(mLock); 541 PlaybackThread *thread = checkPlaybackThread_l(output); 542 if (thread == NULL) { 543 ALOGW("sampleRate() unknown thread %d", output); 544 return 0; 545 } 546 return thread->sampleRate(); 547} 548 549int AudioFlinger::channelCount(audio_io_handle_t output) const 550{ 551 Mutex::Autolock _l(mLock); 552 PlaybackThread *thread = checkPlaybackThread_l(output); 553 if (thread == NULL) { 554 ALOGW("channelCount() unknown thread %d", output); 555 return 0; 556 } 557 return thread->channelCount(); 558} 559 560audio_format_t AudioFlinger::format(audio_io_handle_t output) const 561{ 562 Mutex::Autolock _l(mLock); 563 PlaybackThread *thread = checkPlaybackThread_l(output); 564 if (thread == NULL) { 565 ALOGW("format() unknown thread %d", output); 566 return AUDIO_FORMAT_INVALID; 567 } 568 return thread->format(); 569} 570 571size_t AudioFlinger::frameCount(audio_io_handle_t output) const 572{ 573 Mutex::Autolock _l(mLock); 574 PlaybackThread *thread = checkPlaybackThread_l(output); 575 if (thread == NULL) { 576 ALOGW("frameCount() unknown thread %d", output); 577 return 0; 578 } 579 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 580 // should examine all callers and fix them to handle smaller counts 581 return thread->frameCount(); 582} 583 584uint32_t AudioFlinger::latency(audio_io_handle_t output) const 585{ 586 Mutex::Autolock _l(mLock); 587 PlaybackThread *thread = checkPlaybackThread_l(output); 588 if (thread == NULL) { 589 ALOGW("latency() unknown thread %d", output); 590 return 0; 591 } 592 return thread->latency(); 593} 594 595status_t AudioFlinger::setMasterVolume(float value) 596{ 597 status_t ret = initCheck(); 598 if (ret != NO_ERROR) { 599 return ret; 600 } 601 602 // check calling permissions 603 if (!settingsAllowed()) { 604 return PERMISSION_DENIED; 605 } 606 607 float swmv = value; 608 609 Mutex::Autolock _l(mLock); 610 611 // when hw supports master volume, don't scale in sw mixer 612 if (MVS_NONE != mMasterVolumeSupportLvl) { 613 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 614 AutoMutex lock(mHardwareLock); 615 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 616 617 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 618 if (NULL != dev->set_master_volume) { 619 dev->set_master_volume(dev, value); 620 } 621 mHardwareStatus = AUDIO_HW_IDLE; 622 } 623 624 swmv = 1.0; 625 } 626 627 mMasterVolume = value; 628 mMasterVolumeSW = swmv; 629 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 630 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 631 632 return NO_ERROR; 633} 634 635status_t AudioFlinger::setMode(audio_mode_t mode) 636{ 637 status_t ret = initCheck(); 638 if (ret != NO_ERROR) { 639 return ret; 640 } 641 642 // check calling permissions 643 if (!settingsAllowed()) { 644 return PERMISSION_DENIED; 645 } 646 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 647 ALOGW("Illegal value: setMode(%d)", mode); 648 return BAD_VALUE; 649 } 650 651 { // scope for the lock 652 AutoMutex lock(mHardwareLock); 653 mHardwareStatus = AUDIO_HW_SET_MODE; 654 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 655 mHardwareStatus = AUDIO_HW_IDLE; 656 } 657 658 if (NO_ERROR == ret) { 659 Mutex::Autolock _l(mLock); 660 mMode = mode; 661 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 662 mPlaybackThreads.valueAt(i)->setMode(mode); 663 } 664 665 return ret; 666} 667 668status_t AudioFlinger::setMicMute(bool state) 669{ 670 status_t ret = initCheck(); 671 if (ret != NO_ERROR) { 672 return ret; 673 } 674 675 // check calling permissions 676 if (!settingsAllowed()) { 677 return PERMISSION_DENIED; 678 } 679 680 AutoMutex lock(mHardwareLock); 681 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 682 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 683 mHardwareStatus = AUDIO_HW_IDLE; 684 return ret; 685} 686 687bool AudioFlinger::getMicMute() const 688{ 689 status_t ret = initCheck(); 690 if (ret != NO_ERROR) { 691 return false; 692 } 693 694 bool state = AUDIO_MODE_INVALID; 695 AutoMutex lock(mHardwareLock); 696 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 697 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 698 mHardwareStatus = AUDIO_HW_IDLE; 699 return state; 700} 701 702status_t AudioFlinger::setMasterMute(bool muted) 703{ 704 // check calling permissions 705 if (!settingsAllowed()) { 706 return PERMISSION_DENIED; 707 } 708 709 Mutex::Autolock _l(mLock); 710 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 711 mMasterMute = muted; 712 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 713 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 714 715 return NO_ERROR; 716} 717 718float AudioFlinger::masterVolume() const 719{ 720 Mutex::Autolock _l(mLock); 721 return masterVolume_l(); 722} 723 724float AudioFlinger::masterVolumeSW() const 725{ 726 Mutex::Autolock _l(mLock); 727 return masterVolumeSW_l(); 728} 729 730bool AudioFlinger::masterMute() const 731{ 732 Mutex::Autolock _l(mLock); 733 return masterMute_l(); 734} 735 736float AudioFlinger::masterVolume_l() const 737{ 738 if (MVS_FULL == mMasterVolumeSupportLvl) { 739 float ret_val; 740 AutoMutex lock(mHardwareLock); 741 742 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 743 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 744 (NULL != mPrimaryHardwareDev->get_master_volume), 745 "can't get master volume"); 746 747 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 748 mHardwareStatus = AUDIO_HW_IDLE; 749 return ret_val; 750 } 751 752 return mMasterVolume; 753} 754 755status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 756 audio_io_handle_t output) 757{ 758 // check calling permissions 759 if (!settingsAllowed()) { 760 return PERMISSION_DENIED; 761 } 762 763 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 764 ALOGE("setStreamVolume() invalid stream %d", stream); 765 return BAD_VALUE; 766 } 767 768 AutoMutex lock(mLock); 769 PlaybackThread *thread = NULL; 770 if (output) { 771 thread = checkPlaybackThread_l(output); 772 if (thread == NULL) { 773 return BAD_VALUE; 774 } 775 } 776 777 mStreamTypes[stream].volume = value; 778 779 if (thread == NULL) { 780 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 781 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 782 } 783 } else { 784 thread->setStreamVolume(stream, value); 785 } 786 787 return NO_ERROR; 788} 789 790status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 791{ 792 // check calling permissions 793 if (!settingsAllowed()) { 794 return PERMISSION_DENIED; 795 } 796 797 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 798 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 799 ALOGE("setStreamMute() invalid stream %d", stream); 800 return BAD_VALUE; 801 } 802 803 AutoMutex lock(mLock); 804 mStreamTypes[stream].mute = muted; 805 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 806 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 807 808 return NO_ERROR; 809} 810 811float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 812{ 813 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 814 return 0.0f; 815 } 816 817 AutoMutex lock(mLock); 818 float volume; 819 if (output) { 820 PlaybackThread *thread = checkPlaybackThread_l(output); 821 if (thread == NULL) { 822 return 0.0f; 823 } 824 volume = thread->streamVolume(stream); 825 } else { 826 volume = streamVolume_l(stream); 827 } 828 829 return volume; 830} 831 832bool AudioFlinger::streamMute(audio_stream_type_t stream) const 833{ 834 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 835 return true; 836 } 837 838 AutoMutex lock(mLock); 839 return streamMute_l(stream); 840} 841 842status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 843{ 844 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 845 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 846 // check calling permissions 847 if (!settingsAllowed()) { 848 return PERMISSION_DENIED; 849 } 850 851 // ioHandle == 0 means the parameters are global to the audio hardware interface 852 if (ioHandle == 0) { 853 Mutex::Autolock _l(mLock); 854 status_t final_result = NO_ERROR; 855 { 856 AutoMutex lock(mHardwareLock); 857 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 858 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 859 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 860 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 861 final_result = result ?: final_result; 862 } 863 mHardwareStatus = AUDIO_HW_IDLE; 864 } 865 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 866 AudioParameter param = AudioParameter(keyValuePairs); 867 String8 value; 868 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 869 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 870 if (mBtNrecIsOff != btNrecIsOff) { 871 for (size_t i = 0; i < mRecordThreads.size(); i++) { 872 sp<RecordThread> thread = mRecordThreads.valueAt(i); 873 RecordThread::RecordTrack *track = thread->track(); 874 if (track != NULL) { 875 audio_devices_t device = (audio_devices_t)( 876 thread->device() & AUDIO_DEVICE_IN_ALL); 877 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 878 thread->setEffectSuspended(FX_IID_AEC, 879 suspend, 880 track->sessionId()); 881 thread->setEffectSuspended(FX_IID_NS, 882 suspend, 883 track->sessionId()); 884 } 885 } 886 mBtNrecIsOff = btNrecIsOff; 887 } 888 } 889 String8 screenState; 890 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 891 bool isOff = screenState == "off"; 892 if (isOff != (gScreenState & 1)) { 893 gScreenState = ((gScreenState & ~1) + 2) | isOff; 894 } 895 } 896 return final_result; 897 } 898 899 // hold a strong ref on thread in case closeOutput() or closeInput() is called 900 // and the thread is exited once the lock is released 901 sp<ThreadBase> thread; 902 { 903 Mutex::Autolock _l(mLock); 904 thread = checkPlaybackThread_l(ioHandle); 905 if (thread == 0) { 906 thread = checkRecordThread_l(ioHandle); 907 } else if (thread == primaryPlaybackThread_l()) { 908 // indicate output device change to all input threads for pre processing 909 AudioParameter param = AudioParameter(keyValuePairs); 910 int value; 911 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 912 (value != 0)) { 913 for (size_t i = 0; i < mRecordThreads.size(); i++) { 914 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 915 } 916 } 917 } 918 } 919 if (thread != 0) { 920 return thread->setParameters(keyValuePairs); 921 } 922 return BAD_VALUE; 923} 924 925String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 926{ 927// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 928// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 929 930 Mutex::Autolock _l(mLock); 931 932 if (ioHandle == 0) { 933 String8 out_s8; 934 935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 936 char *s; 937 { 938 AutoMutex lock(mHardwareLock); 939 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 940 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 941 s = dev->get_parameters(dev, keys.string()); 942 mHardwareStatus = AUDIO_HW_IDLE; 943 } 944 out_s8 += String8(s ? s : ""); 945 free(s); 946 } 947 return out_s8; 948 } 949 950 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 951 if (playbackThread != NULL) { 952 return playbackThread->getParameters(keys); 953 } 954 RecordThread *recordThread = checkRecordThread_l(ioHandle); 955 if (recordThread != NULL) { 956 return recordThread->getParameters(keys); 957 } 958 return String8(""); 959} 960 961size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 962 audio_channel_mask_t channelMask) const 963{ 964 status_t ret = initCheck(); 965 if (ret != NO_ERROR) { 966 return 0; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 971 struct audio_config config = { 972 sample_rate: sampleRate, 973 channel_mask: channelMask, 974 format: format, 975 }; 976 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 977 mHardwareStatus = AUDIO_HW_IDLE; 978 return size; 979} 980 981unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 982{ 983 if (ioHandle == 0) { 984 return 0; 985 } 986 987 Mutex::Autolock _l(mLock); 988 989 RecordThread *recordThread = checkRecordThread_l(ioHandle); 990 if (recordThread != NULL) { 991 return recordThread->getInputFramesLost(); 992 } 993 return 0; 994} 995 996status_t AudioFlinger::setVoiceVolume(float value) 997{ 998 status_t ret = initCheck(); 999 if (ret != NO_ERROR) { 1000 return ret; 1001 } 1002 1003 // check calling permissions 1004 if (!settingsAllowed()) { 1005 return PERMISSION_DENIED; 1006 } 1007 1008 AutoMutex lock(mHardwareLock); 1009 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1010 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1011 mHardwareStatus = AUDIO_HW_IDLE; 1012 1013 return ret; 1014} 1015 1016status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1017 audio_io_handle_t output) const 1018{ 1019 status_t status; 1020 1021 Mutex::Autolock _l(mLock); 1022 1023 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1024 if (playbackThread != NULL) { 1025 return playbackThread->getRenderPosition(halFrames, dspFrames); 1026 } 1027 1028 return BAD_VALUE; 1029} 1030 1031void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1032{ 1033 1034 Mutex::Autolock _l(mLock); 1035 1036 pid_t pid = IPCThreadState::self()->getCallingPid(); 1037 if (mNotificationClients.indexOfKey(pid) < 0) { 1038 sp<NotificationClient> notificationClient = new NotificationClient(this, 1039 client, 1040 pid); 1041 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1042 1043 mNotificationClients.add(pid, notificationClient); 1044 1045 sp<IBinder> binder = client->asBinder(); 1046 binder->linkToDeath(notificationClient); 1047 1048 // the config change is always sent from playback or record threads to avoid deadlock 1049 // with AudioSystem::gLock 1050 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1051 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1052 } 1053 1054 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1055 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1056 } 1057 } 1058} 1059 1060void AudioFlinger::removeNotificationClient(pid_t pid) 1061{ 1062 Mutex::Autolock _l(mLock); 1063 1064 mNotificationClients.removeItem(pid); 1065 1066 ALOGV("%d died, releasing its sessions", pid); 1067 size_t num = mAudioSessionRefs.size(); 1068 bool removed = false; 1069 for (size_t i = 0; i< num; ) { 1070 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1071 ALOGV(" pid %d @ %d", ref->mPid, i); 1072 if (ref->mPid == pid) { 1073 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1074 mAudioSessionRefs.removeAt(i); 1075 delete ref; 1076 removed = true; 1077 num--; 1078 } else { 1079 i++; 1080 } 1081 } 1082 if (removed) { 1083 purgeStaleEffects_l(); 1084 } 1085} 1086 1087// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1089{ 1090 size_t size = mNotificationClients.size(); 1091 for (size_t i = 0; i < size; i++) { 1092 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1093 param2); 1094 } 1095} 1096 1097// removeClient_l() must be called with AudioFlinger::mLock held 1098void AudioFlinger::removeClient_l(pid_t pid) 1099{ 1100 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1101 mClients.removeItem(pid); 1102} 1103 1104// getEffectThread_l() must be called with AudioFlinger::mLock held 1105sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1106{ 1107 sp<PlaybackThread> thread; 1108 1109 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1110 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1111 ALOG_ASSERT(thread == 0); 1112 thread = mPlaybackThreads.valueAt(i); 1113 } 1114 } 1115 1116 return thread; 1117} 1118 1119// ---------------------------------------------------------------------------- 1120 1121AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1122 uint32_t device, type_t type) 1123 : Thread(false), 1124 mType(type), 1125 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1126 // mChannelMask 1127 mChannelCount(0), 1128 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1129 mParamStatus(NO_ERROR), 1130 mStandby(false), mDevice((audio_devices_t) device), mId(id), 1131 mDeathRecipient(new PMDeathRecipient(this)) 1132{ 1133} 1134 1135AudioFlinger::ThreadBase::~ThreadBase() 1136{ 1137 mParamCond.broadcast(); 1138 // do not lock the mutex in destructor 1139 releaseWakeLock_l(); 1140 if (mPowerManager != 0) { 1141 sp<IBinder> binder = mPowerManager->asBinder(); 1142 binder->unlinkToDeath(mDeathRecipient); 1143 } 1144} 1145 1146void AudioFlinger::ThreadBase::exit() 1147{ 1148 ALOGV("ThreadBase::exit"); 1149 { 1150 // This lock prevents the following race in thread (uniprocessor for illustration): 1151 // if (!exitPending()) { 1152 // // context switch from here to exit() 1153 // // exit() calls requestExit(), what exitPending() observes 1154 // // exit() calls signal(), which is dropped since no waiters 1155 // // context switch back from exit() to here 1156 // mWaitWorkCV.wait(...); 1157 // // now thread is hung 1158 // } 1159 AutoMutex lock(mLock); 1160 requestExit(); 1161 mWaitWorkCV.signal(); 1162 } 1163 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1164 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1165 requestExitAndWait(); 1166} 1167 1168status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1169{ 1170 status_t status; 1171 1172 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1173 Mutex::Autolock _l(mLock); 1174 1175 mNewParameters.add(keyValuePairs); 1176 mWaitWorkCV.signal(); 1177 // wait condition with timeout in case the thread loop has exited 1178 // before the request could be processed 1179 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1180 status = mParamStatus; 1181 mWaitWorkCV.signal(); 1182 } else { 1183 status = TIMED_OUT; 1184 } 1185 return status; 1186} 1187 1188void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1189{ 1190 Mutex::Autolock _l(mLock); 1191 sendConfigEvent_l(event, param); 1192} 1193 1194// sendConfigEvent_l() must be called with ThreadBase::mLock held 1195void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1196{ 1197 ConfigEvent configEvent; 1198 configEvent.mEvent = event; 1199 configEvent.mParam = param; 1200 mConfigEvents.add(configEvent); 1201 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1202 mWaitWorkCV.signal(); 1203} 1204 1205void AudioFlinger::ThreadBase::processConfigEvents() 1206{ 1207 mLock.lock(); 1208 while (!mConfigEvents.isEmpty()) { 1209 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1210 ConfigEvent configEvent = mConfigEvents[0]; 1211 mConfigEvents.removeAt(0); 1212 // release mLock before locking AudioFlinger mLock: lock order is always 1213 // AudioFlinger then ThreadBase to avoid cross deadlock 1214 mLock.unlock(); 1215 mAudioFlinger->mLock.lock(); 1216 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1217 mAudioFlinger->mLock.unlock(); 1218 mLock.lock(); 1219 } 1220 mLock.unlock(); 1221} 1222 1223status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1224{ 1225 const size_t SIZE = 256; 1226 char buffer[SIZE]; 1227 String8 result; 1228 1229 bool locked = tryLock(mLock); 1230 if (!locked) { 1231 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1232 write(fd, buffer, strlen(buffer)); 1233 } 1234 1235 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1246 result.append(buffer); 1247 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1250 result.append(buffer); 1251 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1252 result.append(buffer); 1253 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1254 result.append(buffer); 1255 1256 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1257 result.append(buffer); 1258 result.append(" Index Command"); 1259 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1260 snprintf(buffer, SIZE, "\n %02d ", i); 1261 result.append(buffer); 1262 result.append(mNewParameters[i]); 1263 } 1264 1265 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1266 result.append(buffer); 1267 snprintf(buffer, SIZE, " Index event param\n"); 1268 result.append(buffer); 1269 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1270 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1271 result.append(buffer); 1272 } 1273 result.append("\n"); 1274 1275 write(fd, result.string(), result.size()); 1276 1277 if (locked) { 1278 mLock.unlock(); 1279 } 1280 return NO_ERROR; 1281} 1282 1283status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1284{ 1285 const size_t SIZE = 256; 1286 char buffer[SIZE]; 1287 String8 result; 1288 1289 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1290 write(fd, buffer, strlen(buffer)); 1291 1292 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1293 sp<EffectChain> chain = mEffectChains[i]; 1294 if (chain != 0) { 1295 chain->dump(fd, args); 1296 } 1297 } 1298 return NO_ERROR; 1299} 1300 1301void AudioFlinger::ThreadBase::acquireWakeLock() 1302{ 1303 Mutex::Autolock _l(mLock); 1304 acquireWakeLock_l(); 1305} 1306 1307void AudioFlinger::ThreadBase::acquireWakeLock_l() 1308{ 1309 if (mPowerManager == 0) { 1310 // use checkService() to avoid blocking if power service is not up yet 1311 sp<IBinder> binder = 1312 defaultServiceManager()->checkService(String16("power")); 1313 if (binder == 0) { 1314 ALOGW("Thread %s cannot connect to the power manager service", mName); 1315 } else { 1316 mPowerManager = interface_cast<IPowerManager>(binder); 1317 binder->linkToDeath(mDeathRecipient); 1318 } 1319 } 1320 if (mPowerManager != 0) { 1321 sp<IBinder> binder = new BBinder(); 1322 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1323 binder, 1324 String16(mName)); 1325 if (status == NO_ERROR) { 1326 mWakeLockToken = binder; 1327 } 1328 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1329 } 1330} 1331 1332void AudioFlinger::ThreadBase::releaseWakeLock() 1333{ 1334 Mutex::Autolock _l(mLock); 1335 releaseWakeLock_l(); 1336} 1337 1338void AudioFlinger::ThreadBase::releaseWakeLock_l() 1339{ 1340 if (mWakeLockToken != 0) { 1341 ALOGV("releaseWakeLock_l() %s", mName); 1342 if (mPowerManager != 0) { 1343 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1344 } 1345 mWakeLockToken.clear(); 1346 } 1347} 1348 1349void AudioFlinger::ThreadBase::clearPowerManager() 1350{ 1351 Mutex::Autolock _l(mLock); 1352 releaseWakeLock_l(); 1353 mPowerManager.clear(); 1354} 1355 1356void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1357{ 1358 sp<ThreadBase> thread = mThread.promote(); 1359 if (thread != 0) { 1360 thread->clearPowerManager(); 1361 } 1362 ALOGW("power manager service died !!!"); 1363} 1364 1365void AudioFlinger::ThreadBase::setEffectSuspended( 1366 const effect_uuid_t *type, bool suspend, int sessionId) 1367{ 1368 Mutex::Autolock _l(mLock); 1369 setEffectSuspended_l(type, suspend, sessionId); 1370} 1371 1372void AudioFlinger::ThreadBase::setEffectSuspended_l( 1373 const effect_uuid_t *type, bool suspend, int sessionId) 1374{ 1375 sp<EffectChain> chain = getEffectChain_l(sessionId); 1376 if (chain != 0) { 1377 if (type != NULL) { 1378 chain->setEffectSuspended_l(type, suspend); 1379 } else { 1380 chain->setEffectSuspendedAll_l(suspend); 1381 } 1382 } 1383 1384 updateSuspendedSessions_l(type, suspend, sessionId); 1385} 1386 1387void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1388{ 1389 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1390 if (index < 0) { 1391 return; 1392 } 1393 1394 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1395 mSuspendedSessions.editValueAt(index); 1396 1397 for (size_t i = 0; i < sessionEffects.size(); i++) { 1398 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1399 for (int j = 0; j < desc->mRefCount; j++) { 1400 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1401 chain->setEffectSuspendedAll_l(true); 1402 } else { 1403 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1404 desc->mType.timeLow); 1405 chain->setEffectSuspended_l(&desc->mType, true); 1406 } 1407 } 1408 } 1409} 1410 1411void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1412 bool suspend, 1413 int sessionId) 1414{ 1415 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1416 1417 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1418 1419 if (suspend) { 1420 if (index >= 0) { 1421 sessionEffects = mSuspendedSessions.editValueAt(index); 1422 } else { 1423 mSuspendedSessions.add(sessionId, sessionEffects); 1424 } 1425 } else { 1426 if (index < 0) { 1427 return; 1428 } 1429 sessionEffects = mSuspendedSessions.editValueAt(index); 1430 } 1431 1432 1433 int key = EffectChain::kKeyForSuspendAll; 1434 if (type != NULL) { 1435 key = type->timeLow; 1436 } 1437 index = sessionEffects.indexOfKey(key); 1438 1439 sp<SuspendedSessionDesc> desc; 1440 if (suspend) { 1441 if (index >= 0) { 1442 desc = sessionEffects.valueAt(index); 1443 } else { 1444 desc = new SuspendedSessionDesc(); 1445 if (type != NULL) { 1446 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1447 } 1448 sessionEffects.add(key, desc); 1449 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1450 } 1451 desc->mRefCount++; 1452 } else { 1453 if (index < 0) { 1454 return; 1455 } 1456 desc = sessionEffects.valueAt(index); 1457 if (--desc->mRefCount == 0) { 1458 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1459 sessionEffects.removeItemsAt(index); 1460 if (sessionEffects.isEmpty()) { 1461 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1462 sessionId); 1463 mSuspendedSessions.removeItem(sessionId); 1464 } 1465 } 1466 } 1467 if (!sessionEffects.isEmpty()) { 1468 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1469 } 1470} 1471 1472void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1473 bool enabled, 1474 int sessionId) 1475{ 1476 Mutex::Autolock _l(mLock); 1477 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1478} 1479 1480void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1481 bool enabled, 1482 int sessionId) 1483{ 1484 if (mType != RECORD) { 1485 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1486 // another session. This gives the priority to well behaved effect control panels 1487 // and applications not using global effects. 1488 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1489 // global effects 1490 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1491 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1492 } 1493 } 1494 1495 sp<EffectChain> chain = getEffectChain_l(sessionId); 1496 if (chain != 0) { 1497 chain->checkSuspendOnEffectEnabled(effect, enabled); 1498 } 1499} 1500 1501// ---------------------------------------------------------------------------- 1502 1503AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1504 AudioStreamOut* output, 1505 audio_io_handle_t id, 1506 uint32_t device, 1507 type_t type) 1508 : ThreadBase(audioFlinger, id, device, type), 1509 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1510 // Assumes constructor is called by AudioFlinger with it's mLock held, 1511 // but it would be safer to explicitly pass initial masterMute as parameter 1512 mMasterMute(audioFlinger->masterMute_l()), 1513 // mStreamTypes[] initialized in constructor body 1514 mOutput(output), 1515 // Assumes constructor is called by AudioFlinger with it's mLock held, 1516 // but it would be safer to explicitly pass initial masterVolume as parameter 1517 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1518 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1519 mMixerStatus(MIXER_IDLE), 1520 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1521 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1522 mScreenState(gScreenState), 1523 // index 0 is reserved for normal mixer's submix 1524 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1525{ 1526 snprintf(mName, kNameLength, "AudioOut_%X", id); 1527 1528 readOutputParameters(); 1529 1530 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1531 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1532 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1533 stream = (audio_stream_type_t) (stream + 1)) { 1534 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1535 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1536 } 1537 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1538 // because mAudioFlinger doesn't have one to copy from 1539} 1540 1541AudioFlinger::PlaybackThread::~PlaybackThread() 1542{ 1543 delete [] mMixBuffer; 1544} 1545 1546status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1547{ 1548 dumpInternals(fd, args); 1549 dumpTracks(fd, args); 1550 dumpEffectChains(fd, args); 1551 return NO_ERROR; 1552} 1553 1554status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1555{ 1556 const size_t SIZE = 256; 1557 char buffer[SIZE]; 1558 String8 result; 1559 1560 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1561 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1562 const stream_type_t *st = &mStreamTypes[i]; 1563 if (i > 0) { 1564 result.appendFormat(", "); 1565 } 1566 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1567 if (st->mute) { 1568 result.append("M"); 1569 } 1570 } 1571 result.append("\n"); 1572 write(fd, result.string(), result.length()); 1573 result.clear(); 1574 1575 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1576 result.append(buffer); 1577 Track::appendDumpHeader(result); 1578 for (size_t i = 0; i < mTracks.size(); ++i) { 1579 sp<Track> track = mTracks[i]; 1580 if (track != 0) { 1581 track->dump(buffer, SIZE); 1582 result.append(buffer); 1583 } 1584 } 1585 1586 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1587 result.append(buffer); 1588 Track::appendDumpHeader(result); 1589 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1590 sp<Track> track = mActiveTracks[i].promote(); 1591 if (track != 0) { 1592 track->dump(buffer, SIZE); 1593 result.append(buffer); 1594 } 1595 } 1596 write(fd, result.string(), result.size()); 1597 1598 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1599 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1600 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1601 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1602 1603 return NO_ERROR; 1604} 1605 1606status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1607{ 1608 const size_t SIZE = 256; 1609 char buffer[SIZE]; 1610 String8 result; 1611 1612 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1613 result.append(buffer); 1614 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1615 result.append(buffer); 1616 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1617 result.append(buffer); 1618 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1619 result.append(buffer); 1620 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1621 result.append(buffer); 1622 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1623 result.append(buffer); 1624 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1625 result.append(buffer); 1626 write(fd, result.string(), result.size()); 1627 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1628 1629 dumpBase(fd, args); 1630 1631 return NO_ERROR; 1632} 1633 1634// Thread virtuals 1635status_t AudioFlinger::PlaybackThread::readyToRun() 1636{ 1637 status_t status = initCheck(); 1638 if (status == NO_ERROR) { 1639 ALOGI("AudioFlinger's thread %p ready to run", this); 1640 } else { 1641 ALOGE("No working audio driver found."); 1642 } 1643 return status; 1644} 1645 1646void AudioFlinger::PlaybackThread::onFirstRef() 1647{ 1648 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1649} 1650 1651// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1652sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1653 const sp<AudioFlinger::Client>& client, 1654 audio_stream_type_t streamType, 1655 uint32_t sampleRate, 1656 audio_format_t format, 1657 uint32_t channelMask, 1658 int frameCount, 1659 const sp<IMemory>& sharedBuffer, 1660 int sessionId, 1661 IAudioFlinger::track_flags_t flags, 1662 pid_t tid, 1663 status_t *status) 1664{ 1665 sp<Track> track; 1666 status_t lStatus; 1667 1668 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1669 1670 // client expresses a preference for FAST, but we get the final say 1671 if (flags & IAudioFlinger::TRACK_FAST) { 1672 if ( 1673 // not timed 1674 (!isTimed) && 1675 // either of these use cases: 1676 ( 1677 // use case 1: shared buffer with any frame count 1678 ( 1679 (sharedBuffer != 0) 1680 ) || 1681 // use case 2: callback handler and frame count is default or at least as large as HAL 1682 ( 1683 (tid != -1) && 1684 ((frameCount == 0) || 1685 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1686 ) 1687 ) && 1688 // PCM data 1689 audio_is_linear_pcm(format) && 1690 // mono or stereo 1691 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1692 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1693#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1694 // hardware sample rate 1695 (sampleRate == mSampleRate) && 1696#endif 1697 // normal mixer has an associated fast mixer 1698 hasFastMixer() && 1699 // there are sufficient fast track slots available 1700 (mFastTrackAvailMask != 0) 1701 // FIXME test that MixerThread for this fast track has a capable output HAL 1702 // FIXME add a permission test also? 1703 ) { 1704 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1705 if (frameCount == 0) { 1706 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1707 } 1708 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1709 frameCount, mFrameCount); 1710 } else { 1711 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1712 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1713 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1714 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1715 audio_is_linear_pcm(format), 1716 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1717 flags &= ~IAudioFlinger::TRACK_FAST; 1718 // For compatibility with AudioTrack calculation, buffer depth is forced 1719 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1720 // This is probably too conservative, but legacy application code may depend on it. 1721 // If you change this calculation, also review the start threshold which is related. 1722 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1723 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1724 if (minBufCount < 2) { 1725 minBufCount = 2; 1726 } 1727 int minFrameCount = mNormalFrameCount * minBufCount; 1728 if (frameCount < minFrameCount) { 1729 frameCount = minFrameCount; 1730 } 1731 } 1732 } 1733 1734 if (mType == DIRECT) { 1735 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1736 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1737 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1738 "for output %p with format %d", 1739 sampleRate, format, channelMask, mOutput, mFormat); 1740 lStatus = BAD_VALUE; 1741 goto Exit; 1742 } 1743 } 1744 } else { 1745 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1746 if (sampleRate > mSampleRate*2) { 1747 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1748 lStatus = BAD_VALUE; 1749 goto Exit; 1750 } 1751 } 1752 1753 lStatus = initCheck(); 1754 if (lStatus != NO_ERROR) { 1755 ALOGE("Audio driver not initialized."); 1756 goto Exit; 1757 } 1758 1759 { // scope for mLock 1760 Mutex::Autolock _l(mLock); 1761 1762 // all tracks in same audio session must share the same routing strategy otherwise 1763 // conflicts will happen when tracks are moved from one output to another by audio policy 1764 // manager 1765 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1766 for (size_t i = 0; i < mTracks.size(); ++i) { 1767 sp<Track> t = mTracks[i]; 1768 if (t != 0 && !t->isOutputTrack()) { 1769 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1770 if (sessionId == t->sessionId() && strategy != actual) { 1771 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1772 strategy, actual); 1773 lStatus = BAD_VALUE; 1774 goto Exit; 1775 } 1776 } 1777 } 1778 1779 if (!isTimed) { 1780 track = new Track(this, client, streamType, sampleRate, format, 1781 channelMask, frameCount, sharedBuffer, sessionId, flags); 1782 } else { 1783 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1784 channelMask, frameCount, sharedBuffer, sessionId); 1785 } 1786 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1787 lStatus = NO_MEMORY; 1788 goto Exit; 1789 } 1790 mTracks.add(track); 1791 1792 sp<EffectChain> chain = getEffectChain_l(sessionId); 1793 if (chain != 0) { 1794 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1795 track->setMainBuffer(chain->inBuffer()); 1796 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1797 chain->incTrackCnt(); 1798 } 1799 } 1800 1801 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1802 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1803 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1804 // so ask activity manager to do this on our behalf 1805 int err = requestPriority(callingPid, tid, 1); 1806 if (err != 0) { 1807 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1808 1, callingPid, tid, err); 1809 } 1810 } 1811 1812 lStatus = NO_ERROR; 1813 1814Exit: 1815 if (status) { 1816 *status = lStatus; 1817 } 1818 return track; 1819} 1820 1821uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1822{ 1823 if (mFastMixer != NULL) { 1824 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1825 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1826 } 1827 return latency; 1828} 1829 1830uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1831{ 1832 return latency; 1833} 1834 1835uint32_t AudioFlinger::PlaybackThread::latency() const 1836{ 1837 Mutex::Autolock _l(mLock); 1838 return latency_l(); 1839} 1840uint32_t AudioFlinger::PlaybackThread::latency_l() const 1841{ 1842 if (initCheck() == NO_ERROR) { 1843 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1844 } else { 1845 return 0; 1846 } 1847} 1848 1849void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 mMasterVolume = value; 1853} 1854 1855void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1856{ 1857 Mutex::Autolock _l(mLock); 1858 setMasterMute_l(muted); 1859} 1860 1861void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1862{ 1863 Mutex::Autolock _l(mLock); 1864 mStreamTypes[stream].volume = value; 1865} 1866 1867void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1868{ 1869 Mutex::Autolock _l(mLock); 1870 mStreamTypes[stream].mute = muted; 1871} 1872 1873float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1874{ 1875 Mutex::Autolock _l(mLock); 1876 return mStreamTypes[stream].volume; 1877} 1878 1879// addTrack_l() must be called with ThreadBase::mLock held 1880status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1881{ 1882 status_t status = ALREADY_EXISTS; 1883 1884 // set retry count for buffer fill 1885 track->mRetryCount = kMaxTrackStartupRetries; 1886 if (mActiveTracks.indexOf(track) < 0) { 1887 // the track is newly added, make sure it fills up all its 1888 // buffers before playing. This is to ensure the client will 1889 // effectively get the latency it requested. 1890 track->mFillingUpStatus = Track::FS_FILLING; 1891 track->mResetDone = false; 1892 track->mPresentationCompleteFrames = 0; 1893 mActiveTracks.add(track); 1894 if (track->mainBuffer() != mMixBuffer) { 1895 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1896 if (chain != 0) { 1897 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1898 chain->incActiveTrackCnt(); 1899 } 1900 } 1901 1902 status = NO_ERROR; 1903 } 1904 1905 ALOGV("mWaitWorkCV.broadcast"); 1906 mWaitWorkCV.broadcast(); 1907 1908 return status; 1909} 1910 1911// destroyTrack_l() must be called with ThreadBase::mLock held 1912void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1913{ 1914 track->mState = TrackBase::TERMINATED; 1915 // active tracks are removed by threadLoop() 1916 if (mActiveTracks.indexOf(track) < 0) { 1917 removeTrack_l(track); 1918 } 1919} 1920 1921void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1922{ 1923 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1924 mTracks.remove(track); 1925 deleteTrackName_l(track->name()); 1926 // redundant as track is about to be destroyed, for dumpsys only 1927 track->mName = -1; 1928 if (track->isFastTrack()) { 1929 int index = track->mFastIndex; 1930 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1931 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1932 mFastTrackAvailMask |= 1 << index; 1933 // redundant as track is about to be destroyed, for dumpsys only 1934 track->mFastIndex = -1; 1935 } 1936 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1937 if (chain != 0) { 1938 chain->decTrackCnt(); 1939 } 1940} 1941 1942String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1943{ 1944 String8 out_s8 = String8(""); 1945 char *s; 1946 1947 Mutex::Autolock _l(mLock); 1948 if (initCheck() != NO_ERROR) { 1949 return out_s8; 1950 } 1951 1952 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1953 out_s8 = String8(s); 1954 free(s); 1955 return out_s8; 1956} 1957 1958// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1959void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1960 AudioSystem::OutputDescriptor desc; 1961 void *param2 = NULL; 1962 1963 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1964 1965 switch (event) { 1966 case AudioSystem::OUTPUT_OPENED: 1967 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1968 desc.channels = mChannelMask; 1969 desc.samplingRate = mSampleRate; 1970 desc.format = mFormat; 1971 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1972 desc.latency = latency(); 1973 param2 = &desc; 1974 break; 1975 1976 case AudioSystem::STREAM_CONFIG_CHANGED: 1977 param2 = ¶m; 1978 case AudioSystem::OUTPUT_CLOSED: 1979 default: 1980 break; 1981 } 1982 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1983} 1984 1985void AudioFlinger::PlaybackThread::readOutputParameters() 1986{ 1987 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1988 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1989 mChannelCount = (uint16_t)popcount(mChannelMask); 1990 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1991 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1992 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1993 if (mFrameCount & 15) { 1994 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1995 mFrameCount); 1996 } 1997 1998 // Calculate size of normal mix buffer relative to the HAL output buffer size 1999 double multiplier = 1.0; 2000 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2001 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2002 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2003 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2004 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2005 maxNormalFrameCount = maxNormalFrameCount & ~15; 2006 if (maxNormalFrameCount < minNormalFrameCount) { 2007 maxNormalFrameCount = minNormalFrameCount; 2008 } 2009 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2010 if (multiplier <= 1.0) { 2011 multiplier = 1.0; 2012 } else if (multiplier <= 2.0) { 2013 if (2 * mFrameCount <= maxNormalFrameCount) { 2014 multiplier = 2.0; 2015 } else { 2016 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2017 } 2018 } else { 2019 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2020 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2021 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2022 // FIXME this rounding up should not be done if no HAL SRC 2023 uint32_t truncMult = (uint32_t) multiplier; 2024 if ((truncMult & 1)) { 2025 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2026 ++truncMult; 2027 } 2028 } 2029 multiplier = (double) truncMult; 2030 } 2031 } 2032 mNormalFrameCount = multiplier * mFrameCount; 2033 // round up to nearest 16 frames to satisfy AudioMixer 2034 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2035 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2036 2037 delete[] mMixBuffer; 2038 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2039 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2040 2041 // force reconfiguration of effect chains and engines to take new buffer size and audio 2042 // parameters into account 2043 // Note that mLock is not held when readOutputParameters() is called from the constructor 2044 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2045 // matter. 2046 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2047 Vector< sp<EffectChain> > effectChains = mEffectChains; 2048 for (size_t i = 0; i < effectChains.size(); i ++) { 2049 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2050 } 2051} 2052 2053 2054status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2055{ 2056 if (halFrames == NULL || dspFrames == NULL) { 2057 return BAD_VALUE; 2058 } 2059 Mutex::Autolock _l(mLock); 2060 if (initCheck() != NO_ERROR) { 2061 return INVALID_OPERATION; 2062 } 2063 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2064 2065 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2066} 2067 2068uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2069{ 2070 Mutex::Autolock _l(mLock); 2071 uint32_t result = 0; 2072 if (getEffectChain_l(sessionId) != 0) { 2073 result = EFFECT_SESSION; 2074 } 2075 2076 for (size_t i = 0; i < mTracks.size(); ++i) { 2077 sp<Track> track = mTracks[i]; 2078 if (sessionId == track->sessionId() && 2079 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2080 result |= TRACK_SESSION; 2081 break; 2082 } 2083 } 2084 2085 return result; 2086} 2087 2088uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2089{ 2090 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2091 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2092 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2093 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2094 } 2095 for (size_t i = 0; i < mTracks.size(); i++) { 2096 sp<Track> track = mTracks[i]; 2097 if (sessionId == track->sessionId() && 2098 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2099 return AudioSystem::getStrategyForStream(track->streamType()); 2100 } 2101 } 2102 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2103} 2104 2105 2106AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2107{ 2108 Mutex::Autolock _l(mLock); 2109 return mOutput; 2110} 2111 2112AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2113{ 2114 Mutex::Autolock _l(mLock); 2115 AudioStreamOut *output = mOutput; 2116 mOutput = NULL; 2117 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2118 // must push a NULL and wait for ack 2119 mOutputSink.clear(); 2120 mPipeSink.clear(); 2121 mNormalSink.clear(); 2122 return output; 2123} 2124 2125// this method must always be called either with ThreadBase mLock held or inside the thread loop 2126audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2127{ 2128 if (mOutput == NULL) { 2129 return NULL; 2130 } 2131 return &mOutput->stream->common; 2132} 2133 2134uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2135{ 2136 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2137} 2138 2139status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2140{ 2141 if (!isValidSyncEvent(event)) { 2142 return BAD_VALUE; 2143 } 2144 2145 Mutex::Autolock _l(mLock); 2146 2147 for (size_t i = 0; i < mTracks.size(); ++i) { 2148 sp<Track> track = mTracks[i]; 2149 if (event->triggerSession() == track->sessionId()) { 2150 track->setSyncEvent(event); 2151 return NO_ERROR; 2152 } 2153 } 2154 2155 return NAME_NOT_FOUND; 2156} 2157 2158bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2159{ 2160 switch (event->type()) { 2161 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2162 return true; 2163 default: 2164 break; 2165 } 2166 return false; 2167} 2168 2169void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2170{ 2171 size_t count = tracksToRemove.size(); 2172 if (CC_UNLIKELY(count)) { 2173 for (size_t i = 0 ; i < count ; i++) { 2174 const sp<Track>& track = tracksToRemove.itemAt(i); 2175 if ((track->sharedBuffer() != 0) && 2176 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2177 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2178 } 2179 } 2180 } 2181 2182} 2183 2184// ---------------------------------------------------------------------------- 2185 2186AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2187 audio_io_handle_t id, uint32_t device, type_t type) 2188 : PlaybackThread(audioFlinger, output, id, device, type), 2189 // mAudioMixer below 2190 // mFastMixer below 2191 mFastMixerFutex(0) 2192 // mOutputSink below 2193 // mPipeSink below 2194 // mNormalSink below 2195{ 2196 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2197 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2198 "mFrameCount=%d, mNormalFrameCount=%d", 2199 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2200 mNormalFrameCount); 2201 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2202 2203 // FIXME - Current mixer implementation only supports stereo output 2204 if (mChannelCount == 1) { 2205 ALOGE("Invalid audio hardware channel count"); 2206 } 2207 2208 // create an NBAIO sink for the HAL output stream, and negotiate 2209 mOutputSink = new AudioStreamOutSink(output->stream); 2210 size_t numCounterOffers = 0; 2211 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2212 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2213 ALOG_ASSERT(index == 0); 2214 2215 // initialize fast mixer depending on configuration 2216 bool initFastMixer; 2217 switch (kUseFastMixer) { 2218 case FastMixer_Never: 2219 initFastMixer = false; 2220 break; 2221 case FastMixer_Always: 2222 initFastMixer = true; 2223 break; 2224 case FastMixer_Static: 2225 case FastMixer_Dynamic: 2226 initFastMixer = mFrameCount < mNormalFrameCount; 2227 break; 2228 } 2229 if (initFastMixer) { 2230 2231 // create a MonoPipe to connect our submix to FastMixer 2232 NBAIO_Format format = mOutputSink->format(); 2233 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2234 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2235 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2236 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2237 const NBAIO_Format offers[1] = {format}; 2238 size_t numCounterOffers = 0; 2239 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2240 ALOG_ASSERT(index == 0); 2241 monoPipe->setAvgFrames((mScreenState & 1) ? 2242 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2243 mPipeSink = monoPipe; 2244 2245#ifdef TEE_SINK_FRAMES 2246 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2247 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2248 numCounterOffers = 0; 2249 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2250 ALOG_ASSERT(index == 0); 2251 mTeeSink = teeSink; 2252 PipeReader *teeSource = new PipeReader(*teeSink); 2253 numCounterOffers = 0; 2254 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2255 ALOG_ASSERT(index == 0); 2256 mTeeSource = teeSource; 2257#endif 2258 2259 // create fast mixer and configure it initially with just one fast track for our submix 2260 mFastMixer = new FastMixer(); 2261 FastMixerStateQueue *sq = mFastMixer->sq(); 2262#ifdef STATE_QUEUE_DUMP 2263 sq->setObserverDump(&mStateQueueObserverDump); 2264 sq->setMutatorDump(&mStateQueueMutatorDump); 2265#endif 2266 FastMixerState *state = sq->begin(); 2267 FastTrack *fastTrack = &state->mFastTracks[0]; 2268 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2269 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2270 fastTrack->mVolumeProvider = NULL; 2271 fastTrack->mGeneration++; 2272 state->mFastTracksGen++; 2273 state->mTrackMask = 1; 2274 // fast mixer will use the HAL output sink 2275 state->mOutputSink = mOutputSink.get(); 2276 state->mOutputSinkGen++; 2277 state->mFrameCount = mFrameCount; 2278 state->mCommand = FastMixerState::COLD_IDLE; 2279 // already done in constructor initialization list 2280 //mFastMixerFutex = 0; 2281 state->mColdFutexAddr = &mFastMixerFutex; 2282 state->mColdGen++; 2283 state->mDumpState = &mFastMixerDumpState; 2284 state->mTeeSink = mTeeSink.get(); 2285 sq->end(); 2286 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2287 2288 // start the fast mixer 2289 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2290 pid_t tid = mFastMixer->getTid(); 2291 int err = requestPriority(getpid_cached, tid, 2); 2292 if (err != 0) { 2293 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2294 2, getpid_cached, tid, err); 2295 } 2296 2297#ifdef AUDIO_WATCHDOG 2298 // create and start the watchdog 2299 mAudioWatchdog = new AudioWatchdog(); 2300 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2301 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2302 tid = mAudioWatchdog->getTid(); 2303 err = requestPriority(getpid_cached, tid, 1); 2304 if (err != 0) { 2305 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2306 1, getpid_cached, tid, err); 2307 } 2308#endif 2309 2310 } else { 2311 mFastMixer = NULL; 2312 } 2313 2314 switch (kUseFastMixer) { 2315 case FastMixer_Never: 2316 case FastMixer_Dynamic: 2317 mNormalSink = mOutputSink; 2318 break; 2319 case FastMixer_Always: 2320 mNormalSink = mPipeSink; 2321 break; 2322 case FastMixer_Static: 2323 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2324 break; 2325 } 2326} 2327 2328AudioFlinger::MixerThread::~MixerThread() 2329{ 2330 if (mFastMixer != NULL) { 2331 FastMixerStateQueue *sq = mFastMixer->sq(); 2332 FastMixerState *state = sq->begin(); 2333 if (state->mCommand == FastMixerState::COLD_IDLE) { 2334 int32_t old = android_atomic_inc(&mFastMixerFutex); 2335 if (old == -1) { 2336 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2337 } 2338 } 2339 state->mCommand = FastMixerState::EXIT; 2340 sq->end(); 2341 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2342 mFastMixer->join(); 2343 // Though the fast mixer thread has exited, it's state queue is still valid. 2344 // We'll use that extract the final state which contains one remaining fast track 2345 // corresponding to our sub-mix. 2346 state = sq->begin(); 2347 ALOG_ASSERT(state->mTrackMask == 1); 2348 FastTrack *fastTrack = &state->mFastTracks[0]; 2349 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2350 delete fastTrack->mBufferProvider; 2351 sq->end(false /*didModify*/); 2352 delete mFastMixer; 2353 if (mAudioWatchdog != 0) { 2354 mAudioWatchdog->requestExit(); 2355 mAudioWatchdog->requestExitAndWait(); 2356 mAudioWatchdog.clear(); 2357 } 2358 } 2359 delete mAudioMixer; 2360} 2361 2362class CpuStats { 2363public: 2364 CpuStats(); 2365 void sample(const String8 &title); 2366#ifdef DEBUG_CPU_USAGE 2367private: 2368 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2369 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2370 2371 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2372 2373 int mCpuNum; // thread's current CPU number 2374 int mCpukHz; // frequency of thread's current CPU in kHz 2375#endif 2376}; 2377 2378CpuStats::CpuStats() 2379#ifdef DEBUG_CPU_USAGE 2380 : mCpuNum(-1), mCpukHz(-1) 2381#endif 2382{ 2383} 2384 2385void CpuStats::sample(const String8 &title) { 2386#ifdef DEBUG_CPU_USAGE 2387 // get current thread's delta CPU time in wall clock ns 2388 double wcNs; 2389 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2390 2391 // record sample for wall clock statistics 2392 if (valid) { 2393 mWcStats.sample(wcNs); 2394 } 2395 2396 // get the current CPU number 2397 int cpuNum = sched_getcpu(); 2398 2399 // get the current CPU frequency in kHz 2400 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2401 2402 // check if either CPU number or frequency changed 2403 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2404 mCpuNum = cpuNum; 2405 mCpukHz = cpukHz; 2406 // ignore sample for purposes of cycles 2407 valid = false; 2408 } 2409 2410 // if no change in CPU number or frequency, then record sample for cycle statistics 2411 if (valid && mCpukHz > 0) { 2412 double cycles = wcNs * cpukHz * 0.000001; 2413 mHzStats.sample(cycles); 2414 } 2415 2416 unsigned n = mWcStats.n(); 2417 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2418 if ((n & 127) == 1) { 2419 long long elapsed = mCpuUsage.elapsed(); 2420 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2421 double perLoop = elapsed / (double) n; 2422 double perLoop100 = perLoop * 0.01; 2423 double perLoop1k = perLoop * 0.001; 2424 double mean = mWcStats.mean(); 2425 double stddev = mWcStats.stddev(); 2426 double minimum = mWcStats.minimum(); 2427 double maximum = mWcStats.maximum(); 2428 double meanCycles = mHzStats.mean(); 2429 double stddevCycles = mHzStats.stddev(); 2430 double minCycles = mHzStats.minimum(); 2431 double maxCycles = mHzStats.maximum(); 2432 mCpuUsage.resetElapsed(); 2433 mWcStats.reset(); 2434 mHzStats.reset(); 2435 ALOGD("CPU usage for %s over past %.1f secs\n" 2436 " (%u mixer loops at %.1f mean ms per loop):\n" 2437 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2438 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2439 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2440 title.string(), 2441 elapsed * .000000001, n, perLoop * .000001, 2442 mean * .001, 2443 stddev * .001, 2444 minimum * .001, 2445 maximum * .001, 2446 mean / perLoop100, 2447 stddev / perLoop100, 2448 minimum / perLoop100, 2449 maximum / perLoop100, 2450 meanCycles / perLoop1k, 2451 stddevCycles / perLoop1k, 2452 minCycles / perLoop1k, 2453 maxCycles / perLoop1k); 2454 2455 } 2456 } 2457#endif 2458}; 2459 2460void AudioFlinger::PlaybackThread::checkSilentMode_l() 2461{ 2462 if (!mMasterMute) { 2463 char value[PROPERTY_VALUE_MAX]; 2464 if (property_get("ro.audio.silent", value, "0") > 0) { 2465 char *endptr; 2466 unsigned long ul = strtoul(value, &endptr, 0); 2467 if (*endptr == '\0' && ul != 0) { 2468 ALOGD("Silence is golden"); 2469 // The setprop command will not allow a property to be changed after 2470 // the first time it is set, so we don't have to worry about un-muting. 2471 setMasterMute_l(true); 2472 } 2473 } 2474 } 2475} 2476 2477bool AudioFlinger::PlaybackThread::threadLoop() 2478{ 2479 Vector< sp<Track> > tracksToRemove; 2480 2481 standbyTime = systemTime(); 2482 2483 // MIXER 2484 nsecs_t lastWarning = 0; 2485 2486 // DUPLICATING 2487 // FIXME could this be made local to while loop? 2488 writeFrames = 0; 2489 2490 cacheParameters_l(); 2491 sleepTime = idleSleepTime; 2492 2493if (mType == MIXER) { 2494 sleepTimeShift = 0; 2495} 2496 2497 CpuStats cpuStats; 2498 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2499 2500 acquireWakeLock(); 2501 2502 while (!exitPending()) 2503 { 2504 cpuStats.sample(myName); 2505 2506 Vector< sp<EffectChain> > effectChains; 2507 2508 processConfigEvents(); 2509 2510 { // scope for mLock 2511 2512 Mutex::Autolock _l(mLock); 2513 2514 if (checkForNewParameters_l()) { 2515 cacheParameters_l(); 2516 } 2517 2518 saveOutputTracks(); 2519 2520 // put audio hardware into standby after short delay 2521 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2522 mSuspended > 0)) { 2523 if (!mStandby) { 2524 2525 threadLoop_standby(); 2526 2527 mStandby = true; 2528 mBytesWritten = 0; 2529 } 2530 2531 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2532 // we're about to wait, flush the binder command buffer 2533 IPCThreadState::self()->flushCommands(); 2534 2535 clearOutputTracks(); 2536 2537 if (exitPending()) break; 2538 2539 releaseWakeLock_l(); 2540 // wait until we have something to do... 2541 ALOGV("%s going to sleep", myName.string()); 2542 mWaitWorkCV.wait(mLock); 2543 ALOGV("%s waking up", myName.string()); 2544 acquireWakeLock_l(); 2545 2546 mMixerStatus = MIXER_IDLE; 2547 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2548 2549 checkSilentMode_l(); 2550 2551 standbyTime = systemTime() + standbyDelay; 2552 sleepTime = idleSleepTime; 2553 if (mType == MIXER) { 2554 sleepTimeShift = 0; 2555 } 2556 2557 continue; 2558 } 2559 } 2560 2561 // mMixerStatusIgnoringFastTracks is also updated internally 2562 mMixerStatus = prepareTracks_l(&tracksToRemove); 2563 2564 // prevent any changes in effect chain list and in each effect chain 2565 // during mixing and effect process as the audio buffers could be deleted 2566 // or modified if an effect is created or deleted 2567 lockEffectChains_l(effectChains); 2568 } 2569 2570 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2571 threadLoop_mix(); 2572 } else { 2573 threadLoop_sleepTime(); 2574 } 2575 2576 if (mSuspended > 0) { 2577 sleepTime = suspendSleepTimeUs(); 2578 } 2579 2580 // only process effects if we're going to write 2581 if (sleepTime == 0) { 2582 for (size_t i = 0; i < effectChains.size(); i ++) { 2583 effectChains[i]->process_l(); 2584 } 2585 } 2586 2587 // enable changes in effect chain 2588 unlockEffectChains(effectChains); 2589 2590 // sleepTime == 0 means we must write to audio hardware 2591 if (sleepTime == 0) { 2592 2593 threadLoop_write(); 2594 2595if (mType == MIXER) { 2596 // write blocked detection 2597 nsecs_t now = systemTime(); 2598 nsecs_t delta = now - mLastWriteTime; 2599 if (!mStandby && delta > maxPeriod) { 2600 mNumDelayedWrites++; 2601 if ((now - lastWarning) > kWarningThrottleNs) { 2602#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2603 ScopedTrace st(ATRACE_TAG, "underrun"); 2604#endif 2605 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2606 ns2ms(delta), mNumDelayedWrites, this); 2607 lastWarning = now; 2608 } 2609 } 2610} 2611 2612 mStandby = false; 2613 } else { 2614 usleep(sleepTime); 2615 } 2616 2617 // Finally let go of removed track(s), without the lock held 2618 // since we can't guarantee the destructors won't acquire that 2619 // same lock. This will also mutate and push a new fast mixer state. 2620 threadLoop_removeTracks(tracksToRemove); 2621 tracksToRemove.clear(); 2622 2623 // FIXME I don't understand the need for this here; 2624 // it was in the original code but maybe the 2625 // assignment in saveOutputTracks() makes this unnecessary? 2626 clearOutputTracks(); 2627 2628 // Effect chains will be actually deleted here if they were removed from 2629 // mEffectChains list during mixing or effects processing 2630 effectChains.clear(); 2631 2632 // FIXME Note that the above .clear() is no longer necessary since effectChains 2633 // is now local to this block, but will keep it for now (at least until merge done). 2634 } 2635 2636if (mType == MIXER || mType == DIRECT) { 2637 // put output stream into standby mode 2638 if (!mStandby) { 2639 mOutput->stream->common.standby(&mOutput->stream->common); 2640 } 2641} 2642if (mType == DUPLICATING) { 2643 // for DuplicatingThread, standby mode is handled by the outputTracks 2644} 2645 2646 releaseWakeLock(); 2647 2648 ALOGV("Thread %p type %d exiting", this, mType); 2649 return false; 2650} 2651 2652void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2653{ 2654 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2655} 2656 2657void AudioFlinger::MixerThread::threadLoop_write() 2658{ 2659 // FIXME we should only do one push per cycle; confirm this is true 2660 // Start the fast mixer if it's not already running 2661 if (mFastMixer != NULL) { 2662 FastMixerStateQueue *sq = mFastMixer->sq(); 2663 FastMixerState *state = sq->begin(); 2664 if (state->mCommand != FastMixerState::MIX_WRITE && 2665 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2666 if (state->mCommand == FastMixerState::COLD_IDLE) { 2667 int32_t old = android_atomic_inc(&mFastMixerFutex); 2668 if (old == -1) { 2669 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2670 } 2671 if (mAudioWatchdog != 0) { 2672 mAudioWatchdog->resume(); 2673 } 2674 } 2675 state->mCommand = FastMixerState::MIX_WRITE; 2676 sq->end(); 2677 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2678 if (kUseFastMixer == FastMixer_Dynamic) { 2679 mNormalSink = mPipeSink; 2680 } 2681 } else { 2682 sq->end(false /*didModify*/); 2683 } 2684 } 2685 PlaybackThread::threadLoop_write(); 2686} 2687 2688// shared by MIXER and DIRECT, overridden by DUPLICATING 2689void AudioFlinger::PlaybackThread::threadLoop_write() 2690{ 2691 // FIXME rewrite to reduce number of system calls 2692 mLastWriteTime = systemTime(); 2693 mInWrite = true; 2694 int bytesWritten; 2695 2696 // If an NBAIO sink is present, use it to write the normal mixer's submix 2697 if (mNormalSink != 0) { 2698#define mBitShift 2 // FIXME 2699 size_t count = mixBufferSize >> mBitShift; 2700#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2701 Tracer::traceBegin(ATRACE_TAG, "write"); 2702#endif 2703 // update the setpoint when gScreenState changes 2704 uint32_t screenState = gScreenState; 2705 if (screenState != mScreenState) { 2706 mScreenState = screenState; 2707 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2708 if (pipe != NULL) { 2709 pipe->setAvgFrames((mScreenState & 1) ? 2710 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2711 } 2712 } 2713 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2714#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2715 Tracer::traceEnd(ATRACE_TAG); 2716#endif 2717 if (framesWritten > 0) { 2718 bytesWritten = framesWritten << mBitShift; 2719 } else { 2720 bytesWritten = framesWritten; 2721 } 2722 // otherwise use the HAL / AudioStreamOut directly 2723 } else { 2724 // Direct output thread. 2725 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2726 } 2727 2728 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2729 mNumWrites++; 2730 mInWrite = false; 2731} 2732 2733void AudioFlinger::MixerThread::threadLoop_standby() 2734{ 2735 // Idle the fast mixer if it's currently running 2736 if (mFastMixer != NULL) { 2737 FastMixerStateQueue *sq = mFastMixer->sq(); 2738 FastMixerState *state = sq->begin(); 2739 if (!(state->mCommand & FastMixerState::IDLE)) { 2740 state->mCommand = FastMixerState::COLD_IDLE; 2741 state->mColdFutexAddr = &mFastMixerFutex; 2742 state->mColdGen++; 2743 mFastMixerFutex = 0; 2744 sq->end(); 2745 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2746 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2747 if (kUseFastMixer == FastMixer_Dynamic) { 2748 mNormalSink = mOutputSink; 2749 } 2750 if (mAudioWatchdog != 0) { 2751 mAudioWatchdog->pause(); 2752 } 2753 } else { 2754 sq->end(false /*didModify*/); 2755 } 2756 } 2757 PlaybackThread::threadLoop_standby(); 2758} 2759 2760// shared by MIXER and DIRECT, overridden by DUPLICATING 2761void AudioFlinger::PlaybackThread::threadLoop_standby() 2762{ 2763 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2764 mOutput->stream->common.standby(&mOutput->stream->common); 2765} 2766 2767void AudioFlinger::MixerThread::threadLoop_mix() 2768{ 2769 // obtain the presentation timestamp of the next output buffer 2770 int64_t pts; 2771 status_t status = INVALID_OPERATION; 2772 2773 if (NULL != mOutput->stream->get_next_write_timestamp) { 2774 status = mOutput->stream->get_next_write_timestamp( 2775 mOutput->stream, &pts); 2776 } 2777 2778 if (status != NO_ERROR) { 2779 pts = AudioBufferProvider::kInvalidPTS; 2780 } 2781 2782 // mix buffers... 2783 mAudioMixer->process(pts); 2784 // increase sleep time progressively when application underrun condition clears. 2785 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2786 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2787 // such that we would underrun the audio HAL. 2788 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2789 sleepTimeShift--; 2790 } 2791 sleepTime = 0; 2792 standbyTime = systemTime() + standbyDelay; 2793 //TODO: delay standby when effects have a tail 2794} 2795 2796void AudioFlinger::MixerThread::threadLoop_sleepTime() 2797{ 2798 // If no tracks are ready, sleep once for the duration of an output 2799 // buffer size, then write 0s to the output 2800 if (sleepTime == 0) { 2801 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2802 sleepTime = activeSleepTime >> sleepTimeShift; 2803 if (sleepTime < kMinThreadSleepTimeUs) { 2804 sleepTime = kMinThreadSleepTimeUs; 2805 } 2806 // reduce sleep time in case of consecutive application underruns to avoid 2807 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2808 // duration we would end up writing less data than needed by the audio HAL if 2809 // the condition persists. 2810 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2811 sleepTimeShift++; 2812 } 2813 } else { 2814 sleepTime = idleSleepTime; 2815 } 2816 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2817 memset (mMixBuffer, 0, mixBufferSize); 2818 sleepTime = 0; 2819 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2820 } 2821 // TODO add standby time extension fct of effect tail 2822} 2823 2824// prepareTracks_l() must be called with ThreadBase::mLock held 2825AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2826 Vector< sp<Track> > *tracksToRemove) 2827{ 2828 2829 mixer_state mixerStatus = MIXER_IDLE; 2830 // find out which tracks need to be processed 2831 size_t count = mActiveTracks.size(); 2832 size_t mixedTracks = 0; 2833 size_t tracksWithEffect = 0; 2834 // counts only _active_ fast tracks 2835 size_t fastTracks = 0; 2836 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2837 2838 float masterVolume = mMasterVolume; 2839 bool masterMute = mMasterMute; 2840 2841 if (masterMute) { 2842 masterVolume = 0; 2843 } 2844 // Delegate master volume control to effect in output mix effect chain if needed 2845 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2846 if (chain != 0) { 2847 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2848 chain->setVolume_l(&v, &v); 2849 masterVolume = (float)((v + (1 << 23)) >> 24); 2850 chain.clear(); 2851 } 2852 2853 // prepare a new state to push 2854 FastMixerStateQueue *sq = NULL; 2855 FastMixerState *state = NULL; 2856 bool didModify = false; 2857 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2858 if (mFastMixer != NULL) { 2859 sq = mFastMixer->sq(); 2860 state = sq->begin(); 2861 } 2862 2863 for (size_t i=0 ; i<count ; i++) { 2864 sp<Track> t = mActiveTracks[i].promote(); 2865 if (t == 0) continue; 2866 2867 // this const just means the local variable doesn't change 2868 Track* const track = t.get(); 2869 2870 // process fast tracks 2871 if (track->isFastTrack()) { 2872 2873 // It's theoretically possible (though unlikely) for a fast track to be created 2874 // and then removed within the same normal mix cycle. This is not a problem, as 2875 // the track never becomes active so it's fast mixer slot is never touched. 2876 // The converse, of removing an (active) track and then creating a new track 2877 // at the identical fast mixer slot within the same normal mix cycle, 2878 // is impossible because the slot isn't marked available until the end of each cycle. 2879 int j = track->mFastIndex; 2880 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2881 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2882 FastTrack *fastTrack = &state->mFastTracks[j]; 2883 2884 // Determine whether the track is currently in underrun condition, 2885 // and whether it had a recent underrun. 2886 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2887 FastTrackUnderruns underruns = ftDump->mUnderruns; 2888 uint32_t recentFull = (underruns.mBitFields.mFull - 2889 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2890 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2891 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2892 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2893 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2894 uint32_t recentUnderruns = recentPartial + recentEmpty; 2895 track->mObservedUnderruns = underruns; 2896 // don't count underruns that occur while stopping or pausing 2897 // or stopped which can occur when flush() is called while active 2898 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2899 track->mUnderrunCount += recentUnderruns; 2900 } 2901 2902 // This is similar to the state machine for normal tracks, 2903 // with a few modifications for fast tracks. 2904 bool isActive = true; 2905 switch (track->mState) { 2906 case TrackBase::STOPPING_1: 2907 // track stays active in STOPPING_1 state until first underrun 2908 if (recentUnderruns > 0) { 2909 track->mState = TrackBase::STOPPING_2; 2910 } 2911 break; 2912 case TrackBase::PAUSING: 2913 // ramp down is not yet implemented 2914 track->setPaused(); 2915 break; 2916 case TrackBase::RESUMING: 2917 // ramp up is not yet implemented 2918 track->mState = TrackBase::ACTIVE; 2919 break; 2920 case TrackBase::ACTIVE: 2921 if (recentFull > 0 || recentPartial > 0) { 2922 // track has provided at least some frames recently: reset retry count 2923 track->mRetryCount = kMaxTrackRetries; 2924 } 2925 if (recentUnderruns == 0) { 2926 // no recent underruns: stay active 2927 break; 2928 } 2929 // there has recently been an underrun of some kind 2930 if (track->sharedBuffer() == 0) { 2931 // were any of the recent underruns "empty" (no frames available)? 2932 if (recentEmpty == 0) { 2933 // no, then ignore the partial underruns as they are allowed indefinitely 2934 break; 2935 } 2936 // there has recently been an "empty" underrun: decrement the retry counter 2937 if (--(track->mRetryCount) > 0) { 2938 break; 2939 } 2940 // indicate to client process that the track was disabled because of underrun; 2941 // it will then automatically call start() when data is available 2942 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2943 // remove from active list, but state remains ACTIVE [confusing but true] 2944 isActive = false; 2945 break; 2946 } 2947 // fall through 2948 case TrackBase::STOPPING_2: 2949 case TrackBase::PAUSED: 2950 case TrackBase::TERMINATED: 2951 case TrackBase::STOPPED: 2952 case TrackBase::FLUSHED: // flush() while active 2953 // Check for presentation complete if track is inactive 2954 // We have consumed all the buffers of this track. 2955 // This would be incomplete if we auto-paused on underrun 2956 { 2957 size_t audioHALFrames = 2958 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2959 size_t framesWritten = 2960 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2961 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2962 // track stays in active list until presentation is complete 2963 break; 2964 } 2965 } 2966 if (track->isStopping_2()) { 2967 track->mState = TrackBase::STOPPED; 2968 } 2969 if (track->isStopped()) { 2970 // Can't reset directly, as fast mixer is still polling this track 2971 // track->reset(); 2972 // So instead mark this track as needing to be reset after push with ack 2973 resetMask |= 1 << i; 2974 } 2975 isActive = false; 2976 break; 2977 case TrackBase::IDLE: 2978 default: 2979 LOG_FATAL("unexpected track state %d", track->mState); 2980 } 2981 2982 if (isActive) { 2983 // was it previously inactive? 2984 if (!(state->mTrackMask & (1 << j))) { 2985 ExtendedAudioBufferProvider *eabp = track; 2986 VolumeProvider *vp = track; 2987 fastTrack->mBufferProvider = eabp; 2988 fastTrack->mVolumeProvider = vp; 2989 fastTrack->mSampleRate = track->mSampleRate; 2990 fastTrack->mChannelMask = track->mChannelMask; 2991 fastTrack->mGeneration++; 2992 state->mTrackMask |= 1 << j; 2993 didModify = true; 2994 // no acknowledgement required for newly active tracks 2995 } 2996 // cache the combined master volume and stream type volume for fast mixer; this 2997 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2998 track->mCachedVolume = track->isMuted() ? 2999 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3000 ++fastTracks; 3001 } else { 3002 // was it previously active? 3003 if (state->mTrackMask & (1 << j)) { 3004 fastTrack->mBufferProvider = NULL; 3005 fastTrack->mGeneration++; 3006 state->mTrackMask &= ~(1 << j); 3007 didModify = true; 3008 // If any fast tracks were removed, we must wait for acknowledgement 3009 // because we're about to decrement the last sp<> on those tracks. 3010 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3011 } else { 3012 LOG_FATAL("fast track %d should have been active", j); 3013 } 3014 tracksToRemove->add(track); 3015 // Avoids a misleading display in dumpsys 3016 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3017 } 3018 continue; 3019 } 3020 3021 { // local variable scope to avoid goto warning 3022 3023 audio_track_cblk_t* cblk = track->cblk(); 3024 3025 // The first time a track is added we wait 3026 // for all its buffers to be filled before processing it 3027 int name = track->name(); 3028 // make sure that we have enough frames to mix one full buffer. 3029 // enforce this condition only once to enable draining the buffer in case the client 3030 // app does not call stop() and relies on underrun to stop: 3031 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3032 // during last round 3033 uint32_t minFrames = 1; 3034 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3035 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3036 if (t->sampleRate() == (int)mSampleRate) { 3037 minFrames = mNormalFrameCount; 3038 } else { 3039 // +1 for rounding and +1 for additional sample needed for interpolation 3040 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3041 // add frames already consumed but not yet released by the resampler 3042 // because cblk->framesReady() will include these frames 3043 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3044 // the minimum track buffer size is normally twice the number of frames necessary 3045 // to fill one buffer and the resampler should not leave more than one buffer worth 3046 // of unreleased frames after each pass, but just in case... 3047 ALOG_ASSERT(minFrames <= cblk->frameCount); 3048 } 3049 } 3050 if ((track->framesReady() >= minFrames) && track->isReady() && 3051 !track->isPaused() && !track->isTerminated()) 3052 { 3053 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3054 3055 mixedTracks++; 3056 3057 // track->mainBuffer() != mMixBuffer means there is an effect chain 3058 // connected to the track 3059 chain.clear(); 3060 if (track->mainBuffer() != mMixBuffer) { 3061 chain = getEffectChain_l(track->sessionId()); 3062 // Delegate volume control to effect in track effect chain if needed 3063 if (chain != 0) { 3064 tracksWithEffect++; 3065 } else { 3066 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3067 name, track->sessionId()); 3068 } 3069 } 3070 3071 3072 int param = AudioMixer::VOLUME; 3073 if (track->mFillingUpStatus == Track::FS_FILLED) { 3074 // no ramp for the first volume setting 3075 track->mFillingUpStatus = Track::FS_ACTIVE; 3076 if (track->mState == TrackBase::RESUMING) { 3077 track->mState = TrackBase::ACTIVE; 3078 param = AudioMixer::RAMP_VOLUME; 3079 } 3080 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3081 } else if (cblk->server != 0) { 3082 // If the track is stopped before the first frame was mixed, 3083 // do not apply ramp 3084 param = AudioMixer::RAMP_VOLUME; 3085 } 3086 3087 // compute volume for this track 3088 uint32_t vl, vr, va; 3089 if (track->isMuted() || track->isPausing() || 3090 mStreamTypes[track->streamType()].mute) { 3091 vl = vr = va = 0; 3092 if (track->isPausing()) { 3093 track->setPaused(); 3094 } 3095 } else { 3096 3097 // read original volumes with volume control 3098 float typeVolume = mStreamTypes[track->streamType()].volume; 3099 float v = masterVolume * typeVolume; 3100 uint32_t vlr = cblk->getVolumeLR(); 3101 vl = vlr & 0xFFFF; 3102 vr = vlr >> 16; 3103 // track volumes come from shared memory, so can't be trusted and must be clamped 3104 if (vl > MAX_GAIN_INT) { 3105 ALOGV("Track left volume out of range: %04X", vl); 3106 vl = MAX_GAIN_INT; 3107 } 3108 if (vr > MAX_GAIN_INT) { 3109 ALOGV("Track right volume out of range: %04X", vr); 3110 vr = MAX_GAIN_INT; 3111 } 3112 // now apply the master volume and stream type volume 3113 vl = (uint32_t)(v * vl) << 12; 3114 vr = (uint32_t)(v * vr) << 12; 3115 // assuming master volume and stream type volume each go up to 1.0, 3116 // vl and vr are now in 8.24 format 3117 3118 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3119 // send level comes from shared memory and so may be corrupt 3120 if (sendLevel > MAX_GAIN_INT) { 3121 ALOGV("Track send level out of range: %04X", sendLevel); 3122 sendLevel = MAX_GAIN_INT; 3123 } 3124 va = (uint32_t)(v * sendLevel); 3125 } 3126 // Delegate volume control to effect in track effect chain if needed 3127 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3128 // Do not ramp volume if volume is controlled by effect 3129 param = AudioMixer::VOLUME; 3130 track->mHasVolumeController = true; 3131 } else { 3132 // force no volume ramp when volume controller was just disabled or removed 3133 // from effect chain to avoid volume spike 3134 if (track->mHasVolumeController) { 3135 param = AudioMixer::VOLUME; 3136 } 3137 track->mHasVolumeController = false; 3138 } 3139 3140 // Convert volumes from 8.24 to 4.12 format 3141 // This additional clamping is needed in case chain->setVolume_l() overshot 3142 vl = (vl + (1 << 11)) >> 12; 3143 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3144 vr = (vr + (1 << 11)) >> 12; 3145 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3146 3147 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3148 3149 // XXX: these things DON'T need to be done each time 3150 mAudioMixer->setBufferProvider(name, track); 3151 mAudioMixer->enable(name); 3152 3153 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3154 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3155 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3156 mAudioMixer->setParameter( 3157 name, 3158 AudioMixer::TRACK, 3159 AudioMixer::FORMAT, (void *)track->format()); 3160 mAudioMixer->setParameter( 3161 name, 3162 AudioMixer::TRACK, 3163 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3164 mAudioMixer->setParameter( 3165 name, 3166 AudioMixer::RESAMPLE, 3167 AudioMixer::SAMPLE_RATE, 3168 (void *)(cblk->sampleRate)); 3169 mAudioMixer->setParameter( 3170 name, 3171 AudioMixer::TRACK, 3172 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3173 mAudioMixer->setParameter( 3174 name, 3175 AudioMixer::TRACK, 3176 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3177 3178 // reset retry count 3179 track->mRetryCount = kMaxTrackRetries; 3180 3181 // If one track is ready, set the mixer ready if: 3182 // - the mixer was not ready during previous round OR 3183 // - no other track is not ready 3184 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3185 mixerStatus != MIXER_TRACKS_ENABLED) { 3186 mixerStatus = MIXER_TRACKS_READY; 3187 } 3188 } else { 3189 // clear effect chain input buffer if an active track underruns to avoid sending 3190 // previous audio buffer again to effects 3191 chain = getEffectChain_l(track->sessionId()); 3192 if (chain != 0) { 3193 chain->clearInputBuffer(); 3194 } 3195 3196 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3197 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3198 track->isStopped() || track->isPaused()) { 3199 // We have consumed all the buffers of this track. 3200 // Remove it from the list of active tracks. 3201 // TODO: use actual buffer filling status instead of latency when available from 3202 // audio HAL 3203 size_t audioHALFrames = 3204 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3205 size_t framesWritten = 3206 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3207 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3208 if (track->isStopped()) { 3209 track->reset(); 3210 } 3211 tracksToRemove->add(track); 3212 } 3213 } else { 3214 track->mUnderrunCount++; 3215 // No buffers for this track. Give it a few chances to 3216 // fill a buffer, then remove it from active list. 3217 if (--(track->mRetryCount) <= 0) { 3218 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3219 tracksToRemove->add(track); 3220 // indicate to client process that the track was disabled because of underrun; 3221 // it will then automatically call start() when data is available 3222 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3223 // If one track is not ready, mark the mixer also not ready if: 3224 // - the mixer was ready during previous round OR 3225 // - no other track is ready 3226 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3227 mixerStatus != MIXER_TRACKS_READY) { 3228 mixerStatus = MIXER_TRACKS_ENABLED; 3229 } 3230 } 3231 mAudioMixer->disable(name); 3232 } 3233 3234 } // local variable scope to avoid goto warning 3235track_is_ready: ; 3236 3237 } 3238 3239 // Push the new FastMixer state if necessary 3240 bool pauseAudioWatchdog = false; 3241 if (didModify) { 3242 state->mFastTracksGen++; 3243 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3244 if (kUseFastMixer == FastMixer_Dynamic && 3245 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3246 state->mCommand = FastMixerState::COLD_IDLE; 3247 state->mColdFutexAddr = &mFastMixerFutex; 3248 state->mColdGen++; 3249 mFastMixerFutex = 0; 3250 if (kUseFastMixer == FastMixer_Dynamic) { 3251 mNormalSink = mOutputSink; 3252 } 3253 // If we go into cold idle, need to wait for acknowledgement 3254 // so that fast mixer stops doing I/O. 3255 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3256 pauseAudioWatchdog = true; 3257 } 3258 sq->end(); 3259 } 3260 if (sq != NULL) { 3261 sq->end(didModify); 3262 sq->push(block); 3263 } 3264 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3265 mAudioWatchdog->pause(); 3266 } 3267 3268 // Now perform the deferred reset on fast tracks that have stopped 3269 while (resetMask != 0) { 3270 size_t i = __builtin_ctz(resetMask); 3271 ALOG_ASSERT(i < count); 3272 resetMask &= ~(1 << i); 3273 sp<Track> t = mActiveTracks[i].promote(); 3274 if (t == 0) continue; 3275 Track* track = t.get(); 3276 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3277 track->reset(); 3278 } 3279 3280 // remove all the tracks that need to be... 3281 count = tracksToRemove->size(); 3282 if (CC_UNLIKELY(count)) { 3283 for (size_t i=0 ; i<count ; i++) { 3284 const sp<Track>& track = tracksToRemove->itemAt(i); 3285 mActiveTracks.remove(track); 3286 if (track->mainBuffer() != mMixBuffer) { 3287 chain = getEffectChain_l(track->sessionId()); 3288 if (chain != 0) { 3289 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3290 chain->decActiveTrackCnt(); 3291 } 3292 } 3293 if (track->isTerminated()) { 3294 removeTrack_l(track); 3295 } 3296 } 3297 } 3298 3299 // mix buffer must be cleared if all tracks are connected to an 3300 // effect chain as in this case the mixer will not write to 3301 // mix buffer and track effects will accumulate into it 3302 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3303 // FIXME as a performance optimization, should remember previous zero status 3304 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3305 } 3306 3307 // if any fast tracks, then status is ready 3308 mMixerStatusIgnoringFastTracks = mixerStatus; 3309 if (fastTracks > 0) { 3310 mixerStatus = MIXER_TRACKS_READY; 3311 } 3312 return mixerStatus; 3313} 3314 3315/* 3316The derived values that are cached: 3317 - mixBufferSize from frame count * frame size 3318 - activeSleepTime from activeSleepTimeUs() 3319 - idleSleepTime from idleSleepTimeUs() 3320 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3321 - maxPeriod from frame count and sample rate (MIXER only) 3322 3323The parameters that affect these derived values are: 3324 - frame count 3325 - frame size 3326 - sample rate 3327 - device type: A2DP or not 3328 - device latency 3329 - format: PCM or not 3330 - active sleep time 3331 - idle sleep time 3332*/ 3333 3334void AudioFlinger::PlaybackThread::cacheParameters_l() 3335{ 3336 mixBufferSize = mNormalFrameCount * mFrameSize; 3337 activeSleepTime = activeSleepTimeUs(); 3338 idleSleepTime = idleSleepTimeUs(); 3339} 3340 3341void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3342{ 3343 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3344 this, streamType, mTracks.size()); 3345 Mutex::Autolock _l(mLock); 3346 3347 size_t size = mTracks.size(); 3348 for (size_t i = 0; i < size; i++) { 3349 sp<Track> t = mTracks[i]; 3350 if (t->streamType() == streamType) { 3351 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3352 t->mCblk->cv.signal(); 3353 } 3354 } 3355} 3356 3357// getTrackName_l() must be called with ThreadBase::mLock held 3358int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3359{ 3360 return mAudioMixer->getTrackName(channelMask); 3361} 3362 3363// deleteTrackName_l() must be called with ThreadBase::mLock held 3364void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3365{ 3366 ALOGV("remove track (%d) and delete from mixer", name); 3367 mAudioMixer->deleteTrackName(name); 3368} 3369 3370// checkForNewParameters_l() must be called with ThreadBase::mLock held 3371bool AudioFlinger::MixerThread::checkForNewParameters_l() 3372{ 3373 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3374 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3375 bool reconfig = false; 3376 3377 while (!mNewParameters.isEmpty()) { 3378 3379 if (mFastMixer != NULL) { 3380 FastMixerStateQueue *sq = mFastMixer->sq(); 3381 FastMixerState *state = sq->begin(); 3382 if (!(state->mCommand & FastMixerState::IDLE)) { 3383 previousCommand = state->mCommand; 3384 state->mCommand = FastMixerState::HOT_IDLE; 3385 sq->end(); 3386 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3387 } else { 3388 sq->end(false /*didModify*/); 3389 } 3390 } 3391 3392 status_t status = NO_ERROR; 3393 String8 keyValuePair = mNewParameters[0]; 3394 AudioParameter param = AudioParameter(keyValuePair); 3395 int value; 3396 3397 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3398 reconfig = true; 3399 } 3400 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3401 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3402 status = BAD_VALUE; 3403 } else { 3404 reconfig = true; 3405 } 3406 } 3407 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3408 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3409 status = BAD_VALUE; 3410 } else { 3411 reconfig = true; 3412 } 3413 } 3414 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3415 // do not accept frame count changes if tracks are open as the track buffer 3416 // size depends on frame count and correct behavior would not be guaranteed 3417 // if frame count is changed after track creation 3418 if (!mTracks.isEmpty()) { 3419 status = INVALID_OPERATION; 3420 } else { 3421 reconfig = true; 3422 } 3423 } 3424 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3425#ifdef ADD_BATTERY_DATA 3426 // when changing the audio output device, call addBatteryData to notify 3427 // the change 3428 if ((int)mDevice != value) { 3429 uint32_t params = 0; 3430 // check whether speaker is on 3431 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3432 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3433 } 3434 3435 int deviceWithoutSpeaker 3436 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3437 // check if any other device (except speaker) is on 3438 if (value & deviceWithoutSpeaker ) { 3439 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3440 } 3441 3442 if (params != 0) { 3443 addBatteryData(params); 3444 } 3445 } 3446#endif 3447 3448 // forward device change to effects that have requested to be 3449 // aware of attached audio device. 3450 mDevice = (audio_devices_t) value; 3451 for (size_t i = 0; i < mEffectChains.size(); i++) { 3452 mEffectChains[i]->setDevice_l(mDevice); 3453 } 3454 } 3455 3456 if (status == NO_ERROR) { 3457 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3458 keyValuePair.string()); 3459 if (!mStandby && status == INVALID_OPERATION) { 3460 mOutput->stream->common.standby(&mOutput->stream->common); 3461 mStandby = true; 3462 mBytesWritten = 0; 3463 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3464 keyValuePair.string()); 3465 } 3466 if (status == NO_ERROR && reconfig) { 3467 delete mAudioMixer; 3468 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3469 mAudioMixer = NULL; 3470 readOutputParameters(); 3471 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3472 for (size_t i = 0; i < mTracks.size() ; i++) { 3473 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3474 if (name < 0) break; 3475 mTracks[i]->mName = name; 3476 // limit track sample rate to 2 x new output sample rate 3477 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3478 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3479 } 3480 } 3481 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3482 } 3483 } 3484 3485 mNewParameters.removeAt(0); 3486 3487 mParamStatus = status; 3488 mParamCond.signal(); 3489 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3490 // already timed out waiting for the status and will never signal the condition. 3491 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3492 } 3493 3494 if (!(previousCommand & FastMixerState::IDLE)) { 3495 ALOG_ASSERT(mFastMixer != NULL); 3496 FastMixerStateQueue *sq = mFastMixer->sq(); 3497 FastMixerState *state = sq->begin(); 3498 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3499 state->mCommand = previousCommand; 3500 sq->end(); 3501 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3502 } 3503 3504 return reconfig; 3505} 3506 3507status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3508{ 3509 const size_t SIZE = 256; 3510 char buffer[SIZE]; 3511 String8 result; 3512 3513 PlaybackThread::dumpInternals(fd, args); 3514 3515 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3516 result.append(buffer); 3517 write(fd, result.string(), result.size()); 3518 3519 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3520 FastMixerDumpState copy = mFastMixerDumpState; 3521 copy.dump(fd); 3522 3523#ifdef STATE_QUEUE_DUMP 3524 // Similar for state queue 3525 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3526 observerCopy.dump(fd); 3527 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3528 mutatorCopy.dump(fd); 3529#endif 3530 3531 // Write the tee output to a .wav file 3532 NBAIO_Source *teeSource = mTeeSource.get(); 3533 if (teeSource != NULL) { 3534 char teePath[64]; 3535 struct timeval tv; 3536 gettimeofday(&tv, NULL); 3537 struct tm tm; 3538 localtime_r(&tv.tv_sec, &tm); 3539 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3540 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3541 if (teeFd >= 0) { 3542 char wavHeader[44]; 3543 memcpy(wavHeader, 3544 "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", 3545 sizeof(wavHeader)); 3546 NBAIO_Format format = teeSource->format(); 3547 unsigned channelCount = Format_channelCount(format); 3548 ALOG_ASSERT(channelCount <= FCC_2); 3549 unsigned sampleRate = Format_sampleRate(format); 3550 wavHeader[22] = channelCount; // number of channels 3551 wavHeader[24] = sampleRate; // sample rate 3552 wavHeader[25] = sampleRate >> 8; 3553 wavHeader[32] = channelCount * 2; // block alignment 3554 write(teeFd, wavHeader, sizeof(wavHeader)); 3555 size_t total = 0; 3556 bool firstRead = true; 3557 for (;;) { 3558#define TEE_SINK_READ 1024 3559 short buffer[TEE_SINK_READ * FCC_2]; 3560 size_t count = TEE_SINK_READ; 3561 ssize_t actual = teeSource->read(buffer, count); 3562 bool wasFirstRead = firstRead; 3563 firstRead = false; 3564 if (actual <= 0) { 3565 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3566 continue; 3567 } 3568 break; 3569 } 3570 ALOG_ASSERT(actual <= (ssize_t)count); 3571 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3572 total += actual; 3573 } 3574 lseek(teeFd, (off_t) 4, SEEK_SET); 3575 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3576 write(teeFd, &temp, sizeof(temp)); 3577 lseek(teeFd, (off_t) 40, SEEK_SET); 3578 temp = total * channelCount * sizeof(short); 3579 write(teeFd, &temp, sizeof(temp)); 3580 close(teeFd); 3581 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3582 } else { 3583 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3584 } 3585 } 3586 3587 if (mAudioWatchdog != 0) { 3588 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3589 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3590 wdCopy.dump(fd); 3591 } 3592 3593 return NO_ERROR; 3594} 3595 3596uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3597{ 3598 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3599} 3600 3601uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3602{ 3603 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3604} 3605 3606void AudioFlinger::MixerThread::cacheParameters_l() 3607{ 3608 PlaybackThread::cacheParameters_l(); 3609 3610 // FIXME: Relaxed timing because of a certain device that can't meet latency 3611 // Should be reduced to 2x after the vendor fixes the driver issue 3612 // increase threshold again due to low power audio mode. The way this warning 3613 // threshold is calculated and its usefulness should be reconsidered anyway. 3614 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3615} 3616 3617// ---------------------------------------------------------------------------- 3618AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3619 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3620 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3621 // mLeftVolFloat, mRightVolFloat 3622{ 3623} 3624 3625AudioFlinger::DirectOutputThread::~DirectOutputThread() 3626{ 3627} 3628 3629AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3630 Vector< sp<Track> > *tracksToRemove 3631) 3632{ 3633 sp<Track> trackToRemove; 3634 3635 mixer_state mixerStatus = MIXER_IDLE; 3636 3637 // find out which tracks need to be processed 3638 if (mActiveTracks.size() != 0) { 3639 sp<Track> t = mActiveTracks[0].promote(); 3640 // The track died recently 3641 if (t == 0) return MIXER_IDLE; 3642 3643 Track* const track = t.get(); 3644 audio_track_cblk_t* cblk = track->cblk(); 3645 3646 // The first time a track is added we wait 3647 // for all its buffers to be filled before processing it 3648 uint32_t minFrames; 3649 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3650 minFrames = mNormalFrameCount; 3651 } else { 3652 minFrames = 1; 3653 } 3654 if ((track->framesReady() >= minFrames) && track->isReady() && 3655 !track->isPaused() && !track->isTerminated()) 3656 { 3657 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3658 3659 if (track->mFillingUpStatus == Track::FS_FILLED) { 3660 track->mFillingUpStatus = Track::FS_ACTIVE; 3661 mLeftVolFloat = mRightVolFloat = 0; 3662 if (track->mState == TrackBase::RESUMING) { 3663 track->mState = TrackBase::ACTIVE; 3664 } 3665 } 3666 3667 // compute volume for this track 3668 float left, right; 3669 if (track->isMuted() || mMasterMute || track->isPausing() || 3670 mStreamTypes[track->streamType()].mute) { 3671 left = right = 0; 3672 if (track->isPausing()) { 3673 track->setPaused(); 3674 } 3675 } else { 3676 float typeVolume = mStreamTypes[track->streamType()].volume; 3677 float v = mMasterVolume * typeVolume; 3678 uint32_t vlr = cblk->getVolumeLR(); 3679 float v_clamped = v * (vlr & 0xFFFF); 3680 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3681 left = v_clamped/MAX_GAIN; 3682 v_clamped = v * (vlr >> 16); 3683 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3684 right = v_clamped/MAX_GAIN; 3685 } 3686 3687 if (left != mLeftVolFloat || right != mRightVolFloat) { 3688 mLeftVolFloat = left; 3689 mRightVolFloat = right; 3690 3691 // Convert volumes from float to 8.24 3692 uint32_t vl = (uint32_t)(left * (1 << 24)); 3693 uint32_t vr = (uint32_t)(right * (1 << 24)); 3694 3695 // Delegate volume control to effect in track effect chain if needed 3696 // only one effect chain can be present on DirectOutputThread, so if 3697 // there is one, the track is connected to it 3698 if (!mEffectChains.isEmpty()) { 3699 // Do not ramp volume if volume is controlled by effect 3700 mEffectChains[0]->setVolume_l(&vl, &vr); 3701 left = (float)vl / (1 << 24); 3702 right = (float)vr / (1 << 24); 3703 } 3704 mOutput->stream->set_volume(mOutput->stream, left, right); 3705 } 3706 3707 // reset retry count 3708 track->mRetryCount = kMaxTrackRetriesDirect; 3709 mActiveTrack = t; 3710 mixerStatus = MIXER_TRACKS_READY; 3711 } else { 3712 // clear effect chain input buffer if an active track underruns to avoid sending 3713 // previous audio buffer again to effects 3714 if (!mEffectChains.isEmpty()) { 3715 mEffectChains[0]->clearInputBuffer(); 3716 } 3717 3718 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3719 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3720 track->isStopped() || track->isPaused()) { 3721 // We have consumed all the buffers of this track. 3722 // Remove it from the list of active tracks. 3723 // TODO: implement behavior for compressed audio 3724 size_t audioHALFrames = 3725 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3726 size_t framesWritten = 3727 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3728 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3729 if (track->isStopped()) { 3730 track->reset(); 3731 } 3732 trackToRemove = track; 3733 } 3734 } else { 3735 // No buffers for this track. Give it a few chances to 3736 // fill a buffer, then remove it from active list. 3737 if (--(track->mRetryCount) <= 0) { 3738 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3739 trackToRemove = track; 3740 } else { 3741 mixerStatus = MIXER_TRACKS_ENABLED; 3742 } 3743 } 3744 } 3745 } 3746 3747 // FIXME merge this with similar code for removing multiple tracks 3748 // remove all the tracks that need to be... 3749 if (CC_UNLIKELY(trackToRemove != 0)) { 3750 tracksToRemove->add(trackToRemove); 3751 mActiveTracks.remove(trackToRemove); 3752 if (!mEffectChains.isEmpty()) { 3753 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3754 trackToRemove->sessionId()); 3755 mEffectChains[0]->decActiveTrackCnt(); 3756 } 3757 if (trackToRemove->isTerminated()) { 3758 removeTrack_l(trackToRemove); 3759 } 3760 } 3761 3762 return mixerStatus; 3763} 3764 3765void AudioFlinger::DirectOutputThread::threadLoop_mix() 3766{ 3767 AudioBufferProvider::Buffer buffer; 3768 size_t frameCount = mFrameCount; 3769 int8_t *curBuf = (int8_t *)mMixBuffer; 3770 // output audio to hardware 3771 while (frameCount) { 3772 buffer.frameCount = frameCount; 3773 mActiveTrack->getNextBuffer(&buffer); 3774 if (CC_UNLIKELY(buffer.raw == NULL)) { 3775 memset(curBuf, 0, frameCount * mFrameSize); 3776 break; 3777 } 3778 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3779 frameCount -= buffer.frameCount; 3780 curBuf += buffer.frameCount * mFrameSize; 3781 mActiveTrack->releaseBuffer(&buffer); 3782 } 3783 sleepTime = 0; 3784 standbyTime = systemTime() + standbyDelay; 3785 mActiveTrack.clear(); 3786 3787} 3788 3789void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3790{ 3791 if (sleepTime == 0) { 3792 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3793 sleepTime = activeSleepTime; 3794 } else { 3795 sleepTime = idleSleepTime; 3796 } 3797 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3798 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3799 sleepTime = 0; 3800 } 3801} 3802 3803// getTrackName_l() must be called with ThreadBase::mLock held 3804int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3805{ 3806 return 0; 3807} 3808 3809// deleteTrackName_l() must be called with ThreadBase::mLock held 3810void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3811{ 3812} 3813 3814// checkForNewParameters_l() must be called with ThreadBase::mLock held 3815bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3816{ 3817 bool reconfig = false; 3818 3819 while (!mNewParameters.isEmpty()) { 3820 status_t status = NO_ERROR; 3821 String8 keyValuePair = mNewParameters[0]; 3822 AudioParameter param = AudioParameter(keyValuePair); 3823 int value; 3824 3825 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3826 // do not accept frame count changes if tracks are open as the track buffer 3827 // size depends on frame count and correct behavior would not be garantied 3828 // if frame count is changed after track creation 3829 if (!mTracks.isEmpty()) { 3830 status = INVALID_OPERATION; 3831 } else { 3832 reconfig = true; 3833 } 3834 } 3835 if (status == NO_ERROR) { 3836 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3837 keyValuePair.string()); 3838 if (!mStandby && status == INVALID_OPERATION) { 3839 mOutput->stream->common.standby(&mOutput->stream->common); 3840 mStandby = true; 3841 mBytesWritten = 0; 3842 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3843 keyValuePair.string()); 3844 } 3845 if (status == NO_ERROR && reconfig) { 3846 readOutputParameters(); 3847 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3848 } 3849 } 3850 3851 mNewParameters.removeAt(0); 3852 3853 mParamStatus = status; 3854 mParamCond.signal(); 3855 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3856 // already timed out waiting for the status and will never signal the condition. 3857 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3858 } 3859 return reconfig; 3860} 3861 3862uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3863{ 3864 uint32_t time; 3865 if (audio_is_linear_pcm(mFormat)) { 3866 time = PlaybackThread::activeSleepTimeUs(); 3867 } else { 3868 time = 10000; 3869 } 3870 return time; 3871} 3872 3873uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3874{ 3875 uint32_t time; 3876 if (audio_is_linear_pcm(mFormat)) { 3877 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3878 } else { 3879 time = 10000; 3880 } 3881 return time; 3882} 3883 3884uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3885{ 3886 uint32_t time; 3887 if (audio_is_linear_pcm(mFormat)) { 3888 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3889 } else { 3890 time = 10000; 3891 } 3892 return time; 3893} 3894 3895void AudioFlinger::DirectOutputThread::cacheParameters_l() 3896{ 3897 PlaybackThread::cacheParameters_l(); 3898 3899 // use shorter standby delay as on normal output to release 3900 // hardware resources as soon as possible 3901 standbyDelay = microseconds(activeSleepTime*2); 3902} 3903 3904// ---------------------------------------------------------------------------- 3905 3906AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3907 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3908 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3909 mWaitTimeMs(UINT_MAX) 3910{ 3911 addOutputTrack(mainThread); 3912} 3913 3914AudioFlinger::DuplicatingThread::~DuplicatingThread() 3915{ 3916 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3917 mOutputTracks[i]->destroy(); 3918 } 3919} 3920 3921void AudioFlinger::DuplicatingThread::threadLoop_mix() 3922{ 3923 // mix buffers... 3924 if (outputsReady(outputTracks)) { 3925 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3926 } else { 3927 memset(mMixBuffer, 0, mixBufferSize); 3928 } 3929 sleepTime = 0; 3930 writeFrames = mNormalFrameCount; 3931 standbyTime = systemTime() + standbyDelay; 3932} 3933 3934void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3935{ 3936 if (sleepTime == 0) { 3937 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3938 sleepTime = activeSleepTime; 3939 } else { 3940 sleepTime = idleSleepTime; 3941 } 3942 } else if (mBytesWritten != 0) { 3943 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3944 writeFrames = mNormalFrameCount; 3945 memset(mMixBuffer, 0, mixBufferSize); 3946 } else { 3947 // flush remaining overflow buffers in output tracks 3948 writeFrames = 0; 3949 } 3950 sleepTime = 0; 3951 } 3952} 3953 3954void AudioFlinger::DuplicatingThread::threadLoop_write() 3955{ 3956 for (size_t i = 0; i < outputTracks.size(); i++) { 3957 outputTracks[i]->write(mMixBuffer, writeFrames); 3958 } 3959 mBytesWritten += mixBufferSize; 3960} 3961 3962void AudioFlinger::DuplicatingThread::threadLoop_standby() 3963{ 3964 // DuplicatingThread implements standby by stopping all tracks 3965 for (size_t i = 0; i < outputTracks.size(); i++) { 3966 outputTracks[i]->stop(); 3967 } 3968} 3969 3970void AudioFlinger::DuplicatingThread::saveOutputTracks() 3971{ 3972 outputTracks = mOutputTracks; 3973} 3974 3975void AudioFlinger::DuplicatingThread::clearOutputTracks() 3976{ 3977 outputTracks.clear(); 3978} 3979 3980void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3981{ 3982 Mutex::Autolock _l(mLock); 3983 // FIXME explain this formula 3984 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3985 OutputTrack *outputTrack = new OutputTrack(thread, 3986 this, 3987 mSampleRate, 3988 mFormat, 3989 mChannelMask, 3990 frameCount); 3991 if (outputTrack->cblk() != NULL) { 3992 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3993 mOutputTracks.add(outputTrack); 3994 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3995 updateWaitTime_l(); 3996 } 3997} 3998 3999void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4000{ 4001 Mutex::Autolock _l(mLock); 4002 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4003 if (mOutputTracks[i]->thread() == thread) { 4004 mOutputTracks[i]->destroy(); 4005 mOutputTracks.removeAt(i); 4006 updateWaitTime_l(); 4007 return; 4008 } 4009 } 4010 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4011} 4012 4013// caller must hold mLock 4014void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4015{ 4016 mWaitTimeMs = UINT_MAX; 4017 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4018 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4019 if (strong != 0) { 4020 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4021 if (waitTimeMs < mWaitTimeMs) { 4022 mWaitTimeMs = waitTimeMs; 4023 } 4024 } 4025 } 4026} 4027 4028 4029bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4030{ 4031 for (size_t i = 0; i < outputTracks.size(); i++) { 4032 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4033 if (thread == 0) { 4034 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4035 return false; 4036 } 4037 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4038 // see note at standby() declaration 4039 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4040 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4041 return false; 4042 } 4043 } 4044 return true; 4045} 4046 4047uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4048{ 4049 return (mWaitTimeMs * 1000) / 2; 4050} 4051 4052void AudioFlinger::DuplicatingThread::cacheParameters_l() 4053{ 4054 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4055 updateWaitTime_l(); 4056 4057 MixerThread::cacheParameters_l(); 4058} 4059 4060// ---------------------------------------------------------------------------- 4061 4062// TrackBase constructor must be called with AudioFlinger::mLock held 4063AudioFlinger::ThreadBase::TrackBase::TrackBase( 4064 ThreadBase *thread, 4065 const sp<Client>& client, 4066 uint32_t sampleRate, 4067 audio_format_t format, 4068 uint32_t channelMask, 4069 int frameCount, 4070 const sp<IMemory>& sharedBuffer, 4071 int sessionId) 4072 : RefBase(), 4073 mThread(thread), 4074 mClient(client), 4075 mCblk(NULL), 4076 // mBuffer 4077 // mBufferEnd 4078 mFrameCount(0), 4079 mState(IDLE), 4080 mSampleRate(sampleRate), 4081 mFormat(format), 4082 mStepServerFailed(false), 4083 mSessionId(sessionId) 4084 // mChannelCount 4085 // mChannelMask 4086{ 4087 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4088 4089 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4090 size_t size = sizeof(audio_track_cblk_t); 4091 uint8_t channelCount = popcount(channelMask); 4092 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4093 if (sharedBuffer == 0) { 4094 size += bufferSize; 4095 } 4096 4097 if (client != NULL) { 4098 mCblkMemory = client->heap()->allocate(size); 4099 if (mCblkMemory != 0) { 4100 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4101 if (mCblk != NULL) { // construct the shared structure in-place. 4102 new(mCblk) audio_track_cblk_t(); 4103 // clear all buffers 4104 mCblk->frameCount = frameCount; 4105 mCblk->sampleRate = sampleRate; 4106// uncomment the following lines to quickly test 32-bit wraparound 4107// mCblk->user = 0xffff0000; 4108// mCblk->server = 0xffff0000; 4109// mCblk->userBase = 0xffff0000; 4110// mCblk->serverBase = 0xffff0000; 4111 mChannelCount = channelCount; 4112 mChannelMask = channelMask; 4113 if (sharedBuffer == 0) { 4114 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4115 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4116 // Force underrun condition to avoid false underrun callback until first data is 4117 // written to buffer (other flags are cleared) 4118 mCblk->flags = CBLK_UNDERRUN_ON; 4119 } else { 4120 mBuffer = sharedBuffer->pointer(); 4121 } 4122 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4123 } 4124 } else { 4125 ALOGE("not enough memory for AudioTrack size=%u", size); 4126 client->heap()->dump("AudioTrack"); 4127 return; 4128 } 4129 } else { 4130 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4131 // construct the shared structure in-place. 4132 new(mCblk) audio_track_cblk_t(); 4133 // clear all buffers 4134 mCblk->frameCount = frameCount; 4135 mCblk->sampleRate = sampleRate; 4136// uncomment the following lines to quickly test 32-bit wraparound 4137// mCblk->user = 0xffff0000; 4138// mCblk->server = 0xffff0000; 4139// mCblk->userBase = 0xffff0000; 4140// mCblk->serverBase = 0xffff0000; 4141 mChannelCount = channelCount; 4142 mChannelMask = channelMask; 4143 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4144 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4145 // Force underrun condition to avoid false underrun callback until first data is 4146 // written to buffer (other flags are cleared) 4147 mCblk->flags = CBLK_UNDERRUN_ON; 4148 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4149 } 4150} 4151 4152AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4153{ 4154 if (mCblk != NULL) { 4155 if (mClient == 0) { 4156 delete mCblk; 4157 } else { 4158 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4159 } 4160 } 4161 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4162 if (mClient != 0) { 4163 // Client destructor must run with AudioFlinger mutex locked 4164 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4165 // If the client's reference count drops to zero, the associated destructor 4166 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4167 // relying on the automatic clear() at end of scope. 4168 mClient.clear(); 4169 } 4170} 4171 4172// AudioBufferProvider interface 4173// getNextBuffer() = 0; 4174// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4175void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4176{ 4177 buffer->raw = NULL; 4178 mFrameCount = buffer->frameCount; 4179 // FIXME See note at getNextBuffer() 4180 (void) step(); // ignore return value of step() 4181 buffer->frameCount = 0; 4182} 4183 4184bool AudioFlinger::ThreadBase::TrackBase::step() { 4185 bool result; 4186 audio_track_cblk_t* cblk = this->cblk(); 4187 4188 result = cblk->stepServer(mFrameCount); 4189 if (!result) { 4190 ALOGV("stepServer failed acquiring cblk mutex"); 4191 mStepServerFailed = true; 4192 } 4193 return result; 4194} 4195 4196void AudioFlinger::ThreadBase::TrackBase::reset() { 4197 audio_track_cblk_t* cblk = this->cblk(); 4198 4199 cblk->user = 0; 4200 cblk->server = 0; 4201 cblk->userBase = 0; 4202 cblk->serverBase = 0; 4203 mStepServerFailed = false; 4204 ALOGV("TrackBase::reset"); 4205} 4206 4207int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4208 return (int)mCblk->sampleRate; 4209} 4210 4211void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4212 audio_track_cblk_t* cblk = this->cblk(); 4213 size_t frameSize = cblk->frameSize; 4214 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4215 int8_t *bufferEnd = bufferStart + frames * frameSize; 4216 4217 // Check validity of returned pointer in case the track control block would have been corrupted. 4218 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4219 "TrackBase::getBuffer buffer out of range:\n" 4220 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4221 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4222 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4223 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4224 4225 return bufferStart; 4226} 4227 4228status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4229{ 4230 mSyncEvents.add(event); 4231 return NO_ERROR; 4232} 4233 4234// ---------------------------------------------------------------------------- 4235 4236// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4237AudioFlinger::PlaybackThread::Track::Track( 4238 PlaybackThread *thread, 4239 const sp<Client>& client, 4240 audio_stream_type_t streamType, 4241 uint32_t sampleRate, 4242 audio_format_t format, 4243 uint32_t channelMask, 4244 int frameCount, 4245 const sp<IMemory>& sharedBuffer, 4246 int sessionId, 4247 IAudioFlinger::track_flags_t flags) 4248 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4249 mMute(false), 4250 mFillingUpStatus(FS_INVALID), 4251 // mRetryCount initialized later when needed 4252 mSharedBuffer(sharedBuffer), 4253 mStreamType(streamType), 4254 mName(-1), // see note below 4255 mMainBuffer(thread->mixBuffer()), 4256 mAuxBuffer(NULL), 4257 mAuxEffectId(0), mHasVolumeController(false), 4258 mPresentationCompleteFrames(0), 4259 mFlags(flags), 4260 mFastIndex(-1), 4261 mUnderrunCount(0), 4262 mCachedVolume(1.0) 4263{ 4264 if (mCblk != NULL) { 4265 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4266 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4267 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4268 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4269 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4270 mCblk->mName = mName; 4271 if (mName < 0) { 4272 ALOGE("no more track names available"); 4273 return; 4274 } 4275 // only allocate a fast track index if we were able to allocate a normal track name 4276 if (flags & IAudioFlinger::TRACK_FAST) { 4277 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4278 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4279 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4280 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4281 // FIXME This is too eager. We allocate a fast track index before the 4282 // fast track becomes active. Since fast tracks are a scarce resource, 4283 // this means we are potentially denying other more important fast tracks from 4284 // being created. It would be better to allocate the index dynamically. 4285 mFastIndex = i; 4286 mCblk->mName = i; 4287 // Read the initial underruns because this field is never cleared by the fast mixer 4288 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4289 thread->mFastTrackAvailMask &= ~(1 << i); 4290 } 4291 } 4292 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4293} 4294 4295AudioFlinger::PlaybackThread::Track::~Track() 4296{ 4297 ALOGV("PlaybackThread::Track destructor"); 4298 sp<ThreadBase> thread = mThread.promote(); 4299 if (thread != 0) { 4300 Mutex::Autolock _l(thread->mLock); 4301 mState = TERMINATED; 4302 } 4303} 4304 4305void AudioFlinger::PlaybackThread::Track::destroy() 4306{ 4307 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4308 // by removing it from mTracks vector, so there is a risk that this Tracks's 4309 // destructor is called. As the destructor needs to lock mLock, 4310 // we must acquire a strong reference on this Track before locking mLock 4311 // here so that the destructor is called only when exiting this function. 4312 // On the other hand, as long as Track::destroy() is only called by 4313 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4314 // this Track with its member mTrack. 4315 sp<Track> keep(this); 4316 { // scope for mLock 4317 sp<ThreadBase> thread = mThread.promote(); 4318 if (thread != 0) { 4319 if (!isOutputTrack()) { 4320 if (mState == ACTIVE || mState == RESUMING) { 4321 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4322 4323#ifdef ADD_BATTERY_DATA 4324 // to track the speaker usage 4325 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4326#endif 4327 } 4328 AudioSystem::releaseOutput(thread->id()); 4329 } 4330 Mutex::Autolock _l(thread->mLock); 4331 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4332 playbackThread->destroyTrack_l(this); 4333 } 4334 } 4335} 4336 4337/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4338{ 4339 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4340 " Server User Main buf Aux Buf Flags Underruns\n"); 4341} 4342 4343void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4344{ 4345 uint32_t vlr = mCblk->getVolumeLR(); 4346 if (isFastTrack()) { 4347 sprintf(buffer, " F %2d", mFastIndex); 4348 } else { 4349 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4350 } 4351 track_state state = mState; 4352 char stateChar; 4353 switch (state) { 4354 case IDLE: 4355 stateChar = 'I'; 4356 break; 4357 case TERMINATED: 4358 stateChar = 'T'; 4359 break; 4360 case STOPPING_1: 4361 stateChar = 's'; 4362 break; 4363 case STOPPING_2: 4364 stateChar = '5'; 4365 break; 4366 case STOPPED: 4367 stateChar = 'S'; 4368 break; 4369 case RESUMING: 4370 stateChar = 'R'; 4371 break; 4372 case ACTIVE: 4373 stateChar = 'A'; 4374 break; 4375 case PAUSING: 4376 stateChar = 'p'; 4377 break; 4378 case PAUSED: 4379 stateChar = 'P'; 4380 break; 4381 case FLUSHED: 4382 stateChar = 'F'; 4383 break; 4384 default: 4385 stateChar = '?'; 4386 break; 4387 } 4388 char nowInUnderrun; 4389 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4390 case UNDERRUN_FULL: 4391 nowInUnderrun = ' '; 4392 break; 4393 case UNDERRUN_PARTIAL: 4394 nowInUnderrun = '<'; 4395 break; 4396 case UNDERRUN_EMPTY: 4397 nowInUnderrun = '*'; 4398 break; 4399 default: 4400 nowInUnderrun = '?'; 4401 break; 4402 } 4403 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4404 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4405 (mClient == 0) ? getpid_cached : mClient->pid(), 4406 mStreamType, 4407 mFormat, 4408 mChannelMask, 4409 mSessionId, 4410 mFrameCount, 4411 mCblk->frameCount, 4412 stateChar, 4413 mMute, 4414 mFillingUpStatus, 4415 mCblk->sampleRate, 4416 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4417 20.0 * log10((vlr >> 16) / 4096.0), 4418 mCblk->server, 4419 mCblk->user, 4420 (int)mMainBuffer, 4421 (int)mAuxBuffer, 4422 mCblk->flags, 4423 mUnderrunCount, 4424 nowInUnderrun); 4425} 4426 4427// AudioBufferProvider interface 4428status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4429 AudioBufferProvider::Buffer* buffer, int64_t pts) 4430{ 4431 audio_track_cblk_t* cblk = this->cblk(); 4432 uint32_t framesReady; 4433 uint32_t framesReq = buffer->frameCount; 4434 4435 // Check if last stepServer failed, try to step now 4436 if (mStepServerFailed) { 4437 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4438 // Since the fast mixer is higher priority than client callback thread, 4439 // it does not result in priority inversion for client. 4440 // But a non-blocking solution would be preferable to avoid 4441 // fast mixer being unable to tryLock(), and 4442 // to avoid the extra context switches if the client wakes up, 4443 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4444 if (!step()) goto getNextBuffer_exit; 4445 ALOGV("stepServer recovered"); 4446 mStepServerFailed = false; 4447 } 4448 4449 // FIXME Same as above 4450 framesReady = cblk->framesReady(); 4451 4452 if (CC_LIKELY(framesReady)) { 4453 uint32_t s = cblk->server; 4454 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4455 4456 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4457 if (framesReq > framesReady) { 4458 framesReq = framesReady; 4459 } 4460 if (framesReq > bufferEnd - s) { 4461 framesReq = bufferEnd - s; 4462 } 4463 4464 buffer->raw = getBuffer(s, framesReq); 4465 buffer->frameCount = framesReq; 4466 return NO_ERROR; 4467 } 4468 4469getNextBuffer_exit: 4470 buffer->raw = NULL; 4471 buffer->frameCount = 0; 4472 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4473 return NOT_ENOUGH_DATA; 4474} 4475 4476// Note that framesReady() takes a mutex on the control block using tryLock(). 4477// This could result in priority inversion if framesReady() is called by the normal mixer, 4478// as the normal mixer thread runs at lower 4479// priority than the client's callback thread: there is a short window within framesReady() 4480// during which the normal mixer could be preempted, and the client callback would block. 4481// Another problem can occur if framesReady() is called by the fast mixer: 4482// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4483// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4484size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4485 return mCblk->framesReady(); 4486} 4487 4488// Don't call for fast tracks; the framesReady() could result in priority inversion 4489bool AudioFlinger::PlaybackThread::Track::isReady() const { 4490 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4491 4492 if (framesReady() >= mCblk->frameCount || 4493 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4494 mFillingUpStatus = FS_FILLED; 4495 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4496 return true; 4497 } 4498 return false; 4499} 4500 4501status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4502 int triggerSession) 4503{ 4504 status_t status = NO_ERROR; 4505 ALOGV("start(%d), calling pid %d session %d", 4506 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4507 4508 sp<ThreadBase> thread = mThread.promote(); 4509 if (thread != 0) { 4510 Mutex::Autolock _l(thread->mLock); 4511 track_state state = mState; 4512 // here the track could be either new, or restarted 4513 // in both cases "unstop" the track 4514 if (mState == PAUSED) { 4515 mState = TrackBase::RESUMING; 4516 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4517 } else { 4518 mState = TrackBase::ACTIVE; 4519 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4520 } 4521 4522 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4523 thread->mLock.unlock(); 4524 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4525 thread->mLock.lock(); 4526 4527#ifdef ADD_BATTERY_DATA 4528 // to track the speaker usage 4529 if (status == NO_ERROR) { 4530 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4531 } 4532#endif 4533 } 4534 if (status == NO_ERROR) { 4535 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4536 playbackThread->addTrack_l(this); 4537 } else { 4538 mState = state; 4539 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4540 } 4541 } else { 4542 status = BAD_VALUE; 4543 } 4544 return status; 4545} 4546 4547void AudioFlinger::PlaybackThread::Track::stop() 4548{ 4549 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4550 sp<ThreadBase> thread = mThread.promote(); 4551 if (thread != 0) { 4552 Mutex::Autolock _l(thread->mLock); 4553 track_state state = mState; 4554 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4555 // If the track is not active (PAUSED and buffers full), flush buffers 4556 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4557 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4558 reset(); 4559 mState = STOPPED; 4560 } else if (!isFastTrack()) { 4561 mState = STOPPED; 4562 } else { 4563 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4564 // and then to STOPPED and reset() when presentation is complete 4565 mState = STOPPING_1; 4566 } 4567 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4568 } 4569 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4570 thread->mLock.unlock(); 4571 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4572 thread->mLock.lock(); 4573 4574#ifdef ADD_BATTERY_DATA 4575 // to track the speaker usage 4576 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4577#endif 4578 } 4579 } 4580} 4581 4582void AudioFlinger::PlaybackThread::Track::pause() 4583{ 4584 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4585 sp<ThreadBase> thread = mThread.promote(); 4586 if (thread != 0) { 4587 Mutex::Autolock _l(thread->mLock); 4588 if (mState == ACTIVE || mState == RESUMING) { 4589 mState = PAUSING; 4590 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4591 if (!isOutputTrack()) { 4592 thread->mLock.unlock(); 4593 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4594 thread->mLock.lock(); 4595 4596#ifdef ADD_BATTERY_DATA 4597 // to track the speaker usage 4598 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4599#endif 4600 } 4601 } 4602 } 4603} 4604 4605void AudioFlinger::PlaybackThread::Track::flush() 4606{ 4607 ALOGV("flush(%d)", mName); 4608 sp<ThreadBase> thread = mThread.promote(); 4609 if (thread != 0) { 4610 Mutex::Autolock _l(thread->mLock); 4611 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4612 mState != PAUSING) { 4613 return; 4614 } 4615 // No point remaining in PAUSED state after a flush => go to 4616 // FLUSHED state 4617 mState = FLUSHED; 4618 // do not reset the track if it is still in the process of being stopped or paused. 4619 // this will be done by prepareTracks_l() when the track is stopped. 4620 // prepareTracks_l() will see mState == FLUSHED, then 4621 // remove from active track list, reset(), and trigger presentation complete 4622 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4623 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4624 reset(); 4625 } 4626 } 4627} 4628 4629void AudioFlinger::PlaybackThread::Track::reset() 4630{ 4631 // Do not reset twice to avoid discarding data written just after a flush and before 4632 // the audioflinger thread detects the track is stopped. 4633 if (!mResetDone) { 4634 TrackBase::reset(); 4635 // Force underrun condition to avoid false underrun callback until first data is 4636 // written to buffer 4637 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4638 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4639 mFillingUpStatus = FS_FILLING; 4640 mResetDone = true; 4641 if (mState == FLUSHED) { 4642 mState = IDLE; 4643 } 4644 } 4645} 4646 4647void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4648{ 4649 mMute = muted; 4650} 4651 4652status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4653{ 4654 status_t status = DEAD_OBJECT; 4655 sp<ThreadBase> thread = mThread.promote(); 4656 if (thread != 0) { 4657 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4658 sp<AudioFlinger> af = mClient->audioFlinger(); 4659 4660 Mutex::Autolock _l(af->mLock); 4661 4662 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4663 4664 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4665 Mutex::Autolock _dl(playbackThread->mLock); 4666 Mutex::Autolock _sl(srcThread->mLock); 4667 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4668 if (chain == 0) { 4669 return INVALID_OPERATION; 4670 } 4671 4672 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4673 if (effect == 0) { 4674 return INVALID_OPERATION; 4675 } 4676 srcThread->removeEffect_l(effect); 4677 playbackThread->addEffect_l(effect); 4678 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4679 if (effect->state() == EffectModule::ACTIVE || 4680 effect->state() == EffectModule::STOPPING) { 4681 effect->start(); 4682 } 4683 4684 sp<EffectChain> dstChain = effect->chain().promote(); 4685 if (dstChain == 0) { 4686 srcThread->addEffect_l(effect); 4687 return INVALID_OPERATION; 4688 } 4689 AudioSystem::unregisterEffect(effect->id()); 4690 AudioSystem::registerEffect(&effect->desc(), 4691 srcThread->id(), 4692 dstChain->strategy(), 4693 AUDIO_SESSION_OUTPUT_MIX, 4694 effect->id()); 4695 } 4696 status = playbackThread->attachAuxEffect(this, EffectId); 4697 } 4698 return status; 4699} 4700 4701void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4702{ 4703 mAuxEffectId = EffectId; 4704 mAuxBuffer = buffer; 4705} 4706 4707bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4708 size_t audioHalFrames) 4709{ 4710 // a track is considered presented when the total number of frames written to audio HAL 4711 // corresponds to the number of frames written when presentationComplete() is called for the 4712 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4713 if (mPresentationCompleteFrames == 0) { 4714 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4715 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4716 mPresentationCompleteFrames, audioHalFrames); 4717 } 4718 if (framesWritten >= mPresentationCompleteFrames) { 4719 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4720 mSessionId, framesWritten); 4721 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4722 return true; 4723 } 4724 return false; 4725} 4726 4727void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4728{ 4729 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4730 if (mSyncEvents[i]->type() == type) { 4731 mSyncEvents[i]->trigger(); 4732 mSyncEvents.removeAt(i); 4733 i--; 4734 } 4735 } 4736} 4737 4738// implement VolumeBufferProvider interface 4739 4740uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4741{ 4742 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4743 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4744 uint32_t vlr = mCblk->getVolumeLR(); 4745 uint32_t vl = vlr & 0xFFFF; 4746 uint32_t vr = vlr >> 16; 4747 // track volumes come from shared memory, so can't be trusted and must be clamped 4748 if (vl > MAX_GAIN_INT) { 4749 vl = MAX_GAIN_INT; 4750 } 4751 if (vr > MAX_GAIN_INT) { 4752 vr = MAX_GAIN_INT; 4753 } 4754 // now apply the cached master volume and stream type volume; 4755 // this is trusted but lacks any synchronization or barrier so may be stale 4756 float v = mCachedVolume; 4757 vl *= v; 4758 vr *= v; 4759 // re-combine into U4.16 4760 vlr = (vr << 16) | (vl & 0xFFFF); 4761 // FIXME look at mute, pause, and stop flags 4762 return vlr; 4763} 4764 4765status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4766{ 4767 if (mState == TERMINATED || mState == PAUSED || 4768 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4769 (mState == STOPPED)))) { 4770 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4771 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4772 event->cancel(); 4773 return INVALID_OPERATION; 4774 } 4775 TrackBase::setSyncEvent(event); 4776 return NO_ERROR; 4777} 4778 4779// timed audio tracks 4780 4781sp<AudioFlinger::PlaybackThread::TimedTrack> 4782AudioFlinger::PlaybackThread::TimedTrack::create( 4783 PlaybackThread *thread, 4784 const sp<Client>& client, 4785 audio_stream_type_t streamType, 4786 uint32_t sampleRate, 4787 audio_format_t format, 4788 uint32_t channelMask, 4789 int frameCount, 4790 const sp<IMemory>& sharedBuffer, 4791 int sessionId) { 4792 if (!client->reserveTimedTrack()) 4793 return 0; 4794 4795 return new TimedTrack( 4796 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4797 sharedBuffer, sessionId); 4798} 4799 4800AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4801 PlaybackThread *thread, 4802 const sp<Client>& client, 4803 audio_stream_type_t streamType, 4804 uint32_t sampleRate, 4805 audio_format_t format, 4806 uint32_t channelMask, 4807 int frameCount, 4808 const sp<IMemory>& sharedBuffer, 4809 int sessionId) 4810 : Track(thread, client, streamType, sampleRate, format, channelMask, 4811 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4812 mQueueHeadInFlight(false), 4813 mTrimQueueHeadOnRelease(false), 4814 mFramesPendingInQueue(0), 4815 mTimedSilenceBuffer(NULL), 4816 mTimedSilenceBufferSize(0), 4817 mTimedAudioOutputOnTime(false), 4818 mMediaTimeTransformValid(false) 4819{ 4820 LocalClock lc; 4821 mLocalTimeFreq = lc.getLocalFreq(); 4822 4823 mLocalTimeToSampleTransform.a_zero = 0; 4824 mLocalTimeToSampleTransform.b_zero = 0; 4825 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4826 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4827 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4828 &mLocalTimeToSampleTransform.a_to_b_denom); 4829 4830 mMediaTimeToSampleTransform.a_zero = 0; 4831 mMediaTimeToSampleTransform.b_zero = 0; 4832 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4833 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4834 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4835 &mMediaTimeToSampleTransform.a_to_b_denom); 4836} 4837 4838AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4839 mClient->releaseTimedTrack(); 4840 delete [] mTimedSilenceBuffer; 4841} 4842 4843status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4844 size_t size, sp<IMemory>* buffer) { 4845 4846 Mutex::Autolock _l(mTimedBufferQueueLock); 4847 4848 trimTimedBufferQueue_l(); 4849 4850 // lazily initialize the shared memory heap for timed buffers 4851 if (mTimedMemoryDealer == NULL) { 4852 const int kTimedBufferHeapSize = 512 << 10; 4853 4854 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4855 "AudioFlingerTimed"); 4856 if (mTimedMemoryDealer == NULL) 4857 return NO_MEMORY; 4858 } 4859 4860 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4861 if (newBuffer == NULL) { 4862 newBuffer = mTimedMemoryDealer->allocate(size); 4863 if (newBuffer == NULL) 4864 return NO_MEMORY; 4865 } 4866 4867 *buffer = newBuffer; 4868 return NO_ERROR; 4869} 4870 4871// caller must hold mTimedBufferQueueLock 4872void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4873 int64_t mediaTimeNow; 4874 { 4875 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4876 if (!mMediaTimeTransformValid) 4877 return; 4878 4879 int64_t targetTimeNow; 4880 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4881 ? mCCHelper.getCommonTime(&targetTimeNow) 4882 : mCCHelper.getLocalTime(&targetTimeNow); 4883 4884 if (OK != res) 4885 return; 4886 4887 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4888 &mediaTimeNow)) { 4889 return; 4890 } 4891 } 4892 4893 size_t trimEnd; 4894 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4895 int64_t bufEnd; 4896 4897 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4898 // We have a next buffer. Just use its PTS as the PTS of the frame 4899 // following the last frame in this buffer. If the stream is sparse 4900 // (ie, there are deliberate gaps left in the stream which should be 4901 // filled with silence by the TimedAudioTrack), then this can result 4902 // in one extra buffer being left un-trimmed when it could have 4903 // been. In general, this is not typical, and we would rather 4904 // optimized away the TS calculation below for the more common case 4905 // where PTSes are contiguous. 4906 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4907 } else { 4908 // We have no next buffer. Compute the PTS of the frame following 4909 // the last frame in this buffer by computing the duration of of 4910 // this frame in media time units and adding it to the PTS of the 4911 // buffer. 4912 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4913 / mCblk->frameSize; 4914 4915 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4916 &bufEnd)) { 4917 ALOGE("Failed to convert frame count of %lld to media time" 4918 " duration" " (scale factor %d/%u) in %s", 4919 frameCount, 4920 mMediaTimeToSampleTransform.a_to_b_numer, 4921 mMediaTimeToSampleTransform.a_to_b_denom, 4922 __PRETTY_FUNCTION__); 4923 break; 4924 } 4925 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4926 } 4927 4928 if (bufEnd > mediaTimeNow) 4929 break; 4930 4931 // Is the buffer we want to use in the middle of a mix operation right 4932 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4933 // from the mixer which should be coming back shortly. 4934 if (!trimEnd && mQueueHeadInFlight) { 4935 mTrimQueueHeadOnRelease = true; 4936 } 4937 } 4938 4939 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4940 if (trimStart < trimEnd) { 4941 // Update the bookkeeping for framesReady() 4942 for (size_t i = trimStart; i < trimEnd; ++i) { 4943 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4944 } 4945 4946 // Now actually remove the buffers from the queue. 4947 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4948 } 4949} 4950 4951void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4952 const char* logTag) { 4953 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4954 "%s called (reason \"%s\"), but timed buffer queue has no" 4955 " elements to trim.", __FUNCTION__, logTag); 4956 4957 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4958 mTimedBufferQueue.removeAt(0); 4959} 4960 4961void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4962 const TimedBuffer& buf, 4963 const char* logTag) { 4964 uint32_t bufBytes = buf.buffer()->size(); 4965 uint32_t consumedAlready = buf.position(); 4966 4967 ALOG_ASSERT(consumedAlready <= bufBytes, 4968 "Bad bookkeeping while updating frames pending. Timed buffer is" 4969 " only %u bytes long, but claims to have consumed %u" 4970 " bytes. (update reason: \"%s\")", 4971 bufBytes, consumedAlready, logTag); 4972 4973 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4974 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4975 "Bad bookkeeping while updating frames pending. Should have at" 4976 " least %u queued frames, but we think we have only %u. (update" 4977 " reason: \"%s\")", 4978 bufFrames, mFramesPendingInQueue, logTag); 4979 4980 mFramesPendingInQueue -= bufFrames; 4981} 4982 4983status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4984 const sp<IMemory>& buffer, int64_t pts) { 4985 4986 { 4987 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4988 if (!mMediaTimeTransformValid) 4989 return INVALID_OPERATION; 4990 } 4991 4992 Mutex::Autolock _l(mTimedBufferQueueLock); 4993 4994 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4995 mFramesPendingInQueue += bufFrames; 4996 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4997 4998 return NO_ERROR; 4999} 5000 5001status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5002 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5003 5004 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5005 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5006 target); 5007 5008 if (!(target == TimedAudioTrack::LOCAL_TIME || 5009 target == TimedAudioTrack::COMMON_TIME)) { 5010 return BAD_VALUE; 5011 } 5012 5013 Mutex::Autolock lock(mMediaTimeTransformLock); 5014 mMediaTimeTransform = xform; 5015 mMediaTimeTransformTarget = target; 5016 mMediaTimeTransformValid = true; 5017 5018 return NO_ERROR; 5019} 5020 5021#define min(a, b) ((a) < (b) ? (a) : (b)) 5022 5023// implementation of getNextBuffer for tracks whose buffers have timestamps 5024status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5025 AudioBufferProvider::Buffer* buffer, int64_t pts) 5026{ 5027 if (pts == AudioBufferProvider::kInvalidPTS) { 5028 buffer->raw = NULL; 5029 buffer->frameCount = 0; 5030 mTimedAudioOutputOnTime = false; 5031 return INVALID_OPERATION; 5032 } 5033 5034 Mutex::Autolock _l(mTimedBufferQueueLock); 5035 5036 ALOG_ASSERT(!mQueueHeadInFlight, 5037 "getNextBuffer called without releaseBuffer!"); 5038 5039 while (true) { 5040 5041 // if we have no timed buffers, then fail 5042 if (mTimedBufferQueue.isEmpty()) { 5043 buffer->raw = NULL; 5044 buffer->frameCount = 0; 5045 return NOT_ENOUGH_DATA; 5046 } 5047 5048 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5049 5050 // calculate the PTS of the head of the timed buffer queue expressed in 5051 // local time 5052 int64_t headLocalPTS; 5053 { 5054 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5055 5056 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5057 5058 if (mMediaTimeTransform.a_to_b_denom == 0) { 5059 // the transform represents a pause, so yield silence 5060 timedYieldSilence_l(buffer->frameCount, buffer); 5061 return NO_ERROR; 5062 } 5063 5064 int64_t transformedPTS; 5065 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5066 &transformedPTS)) { 5067 // the transform failed. this shouldn't happen, but if it does 5068 // then just drop this buffer 5069 ALOGW("timedGetNextBuffer transform failed"); 5070 buffer->raw = NULL; 5071 buffer->frameCount = 0; 5072 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5073 return NO_ERROR; 5074 } 5075 5076 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5077 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5078 &headLocalPTS)) { 5079 buffer->raw = NULL; 5080 buffer->frameCount = 0; 5081 return INVALID_OPERATION; 5082 } 5083 } else { 5084 headLocalPTS = transformedPTS; 5085 } 5086 } 5087 5088 // adjust the head buffer's PTS to reflect the portion of the head buffer 5089 // that has already been consumed 5090 int64_t effectivePTS = headLocalPTS + 5091 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5092 5093 // Calculate the delta in samples between the head of the input buffer 5094 // queue and the start of the next output buffer that will be written. 5095 // If the transformation fails because of over or underflow, it means 5096 // that the sample's position in the output stream is so far out of 5097 // whack that it should just be dropped. 5098 int64_t sampleDelta; 5099 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5100 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5101 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5102 " mix"); 5103 continue; 5104 } 5105 if (!mLocalTimeToSampleTransform.doForwardTransform( 5106 (effectivePTS - pts) << 32, &sampleDelta)) { 5107 ALOGV("*** too late during sample rate transform: dropped buffer"); 5108 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5109 continue; 5110 } 5111 5112 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5113 " sampleDelta=[%d.%08x]", 5114 head.pts(), head.position(), pts, 5115 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5116 + (sampleDelta >> 32)), 5117 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5118 5119 // if the delta between the ideal placement for the next input sample and 5120 // the current output position is within this threshold, then we will 5121 // concatenate the next input samples to the previous output 5122 const int64_t kSampleContinuityThreshold = 5123 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5124 5125 // if this is the first buffer of audio that we're emitting from this track 5126 // then it should be almost exactly on time. 5127 const int64_t kSampleStartupThreshold = 1LL << 32; 5128 5129 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5130 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5131 // the next input is close enough to being on time, so concatenate it 5132 // with the last output 5133 timedYieldSamples_l(buffer); 5134 5135 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5136 head.position(), buffer->frameCount); 5137 return NO_ERROR; 5138 } 5139 5140 // Looks like our output is not on time. Reset our on timed status. 5141 // Next time we mix samples from our input queue, then should be within 5142 // the StartupThreshold. 5143 mTimedAudioOutputOnTime = false; 5144 if (sampleDelta > 0) { 5145 // the gap between the current output position and the proper start of 5146 // the next input sample is too big, so fill it with silence 5147 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5148 5149 timedYieldSilence_l(framesUntilNextInput, buffer); 5150 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5151 return NO_ERROR; 5152 } else { 5153 // the next input sample is late 5154 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5155 size_t onTimeSamplePosition = 5156 head.position() + lateFrames * mCblk->frameSize; 5157 5158 if (onTimeSamplePosition > head.buffer()->size()) { 5159 // all the remaining samples in the head are too late, so 5160 // drop it and move on 5161 ALOGV("*** too late: dropped buffer"); 5162 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5163 continue; 5164 } else { 5165 // skip over the late samples 5166 head.setPosition(onTimeSamplePosition); 5167 5168 // yield the available samples 5169 timedYieldSamples_l(buffer); 5170 5171 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5172 return NO_ERROR; 5173 } 5174 } 5175 } 5176} 5177 5178// Yield samples from the timed buffer queue head up to the given output 5179// buffer's capacity. 5180// 5181// Caller must hold mTimedBufferQueueLock 5182void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5183 AudioBufferProvider::Buffer* buffer) { 5184 5185 const TimedBuffer& head = mTimedBufferQueue[0]; 5186 5187 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5188 head.position()); 5189 5190 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5191 mCblk->frameSize); 5192 size_t framesRequested = buffer->frameCount; 5193 buffer->frameCount = min(framesLeftInHead, framesRequested); 5194 5195 mQueueHeadInFlight = true; 5196 mTimedAudioOutputOnTime = true; 5197} 5198 5199// Yield samples of silence up to the given output buffer's capacity 5200// 5201// Caller must hold mTimedBufferQueueLock 5202void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5203 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5204 5205 // lazily allocate a buffer filled with silence 5206 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5207 delete [] mTimedSilenceBuffer; 5208 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5209 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5210 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5211 } 5212 5213 buffer->raw = mTimedSilenceBuffer; 5214 size_t framesRequested = buffer->frameCount; 5215 buffer->frameCount = min(numFrames, framesRequested); 5216 5217 mTimedAudioOutputOnTime = false; 5218} 5219 5220// AudioBufferProvider interface 5221void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5222 AudioBufferProvider::Buffer* buffer) { 5223 5224 Mutex::Autolock _l(mTimedBufferQueueLock); 5225 5226 // If the buffer which was just released is part of the buffer at the head 5227 // of the queue, be sure to update the amt of the buffer which has been 5228 // consumed. If the buffer being returned is not part of the head of the 5229 // queue, its either because the buffer is part of the silence buffer, or 5230 // because the head of the timed queue was trimmed after the mixer called 5231 // getNextBuffer but before the mixer called releaseBuffer. 5232 if (buffer->raw == mTimedSilenceBuffer) { 5233 ALOG_ASSERT(!mQueueHeadInFlight, 5234 "Queue head in flight during release of silence buffer!"); 5235 goto done; 5236 } 5237 5238 ALOG_ASSERT(mQueueHeadInFlight, 5239 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5240 " head in flight."); 5241 5242 if (mTimedBufferQueue.size()) { 5243 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5244 5245 void* start = head.buffer()->pointer(); 5246 void* end = reinterpret_cast<void*>( 5247 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5248 + head.buffer()->size()); 5249 5250 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5251 "released buffer not within the head of the timed buffer" 5252 " queue; qHead = [%p, %p], released buffer = %p", 5253 start, end, buffer->raw); 5254 5255 head.setPosition(head.position() + 5256 (buffer->frameCount * mCblk->frameSize)); 5257 mQueueHeadInFlight = false; 5258 5259 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5260 "Bad bookkeeping during releaseBuffer! Should have at" 5261 " least %u queued frames, but we think we have only %u", 5262 buffer->frameCount, mFramesPendingInQueue); 5263 5264 mFramesPendingInQueue -= buffer->frameCount; 5265 5266 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5267 || mTrimQueueHeadOnRelease) { 5268 trimTimedBufferQueueHead_l("releaseBuffer"); 5269 mTrimQueueHeadOnRelease = false; 5270 } 5271 } else { 5272 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5273 " buffers in the timed buffer queue"); 5274 } 5275 5276done: 5277 buffer->raw = 0; 5278 buffer->frameCount = 0; 5279} 5280 5281size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5282 Mutex::Autolock _l(mTimedBufferQueueLock); 5283 return mFramesPendingInQueue; 5284} 5285 5286AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5287 : mPTS(0), mPosition(0) {} 5288 5289AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5290 const sp<IMemory>& buffer, int64_t pts) 5291 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5292 5293// ---------------------------------------------------------------------------- 5294 5295// RecordTrack constructor must be called with AudioFlinger::mLock held 5296AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5297 RecordThread *thread, 5298 const sp<Client>& client, 5299 uint32_t sampleRate, 5300 audio_format_t format, 5301 uint32_t channelMask, 5302 int frameCount, 5303 int sessionId) 5304 : TrackBase(thread, client, sampleRate, format, 5305 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5306 mOverflow(false) 5307{ 5308 if (mCblk != NULL) { 5309 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5310 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5311 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5312 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5313 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5314 } else { 5315 mCblk->frameSize = sizeof(int8_t); 5316 } 5317 } 5318} 5319 5320AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5321{ 5322 sp<ThreadBase> thread = mThread.promote(); 5323 if (thread != 0) { 5324 AudioSystem::releaseInput(thread->id()); 5325 } 5326} 5327 5328// AudioBufferProvider interface 5329status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5330{ 5331 audio_track_cblk_t* cblk = this->cblk(); 5332 uint32_t framesAvail; 5333 uint32_t framesReq = buffer->frameCount; 5334 5335 // Check if last stepServer failed, try to step now 5336 if (mStepServerFailed) { 5337 if (!step()) goto getNextBuffer_exit; 5338 ALOGV("stepServer recovered"); 5339 mStepServerFailed = false; 5340 } 5341 5342 framesAvail = cblk->framesAvailable_l(); 5343 5344 if (CC_LIKELY(framesAvail)) { 5345 uint32_t s = cblk->server; 5346 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5347 5348 if (framesReq > framesAvail) { 5349 framesReq = framesAvail; 5350 } 5351 if (framesReq > bufferEnd - s) { 5352 framesReq = bufferEnd - s; 5353 } 5354 5355 buffer->raw = getBuffer(s, framesReq); 5356 buffer->frameCount = framesReq; 5357 return NO_ERROR; 5358 } 5359 5360getNextBuffer_exit: 5361 buffer->raw = NULL; 5362 buffer->frameCount = 0; 5363 return NOT_ENOUGH_DATA; 5364} 5365 5366status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5367 int triggerSession) 5368{ 5369 sp<ThreadBase> thread = mThread.promote(); 5370 if (thread != 0) { 5371 RecordThread *recordThread = (RecordThread *)thread.get(); 5372 return recordThread->start(this, event, triggerSession); 5373 } else { 5374 return BAD_VALUE; 5375 } 5376} 5377 5378void AudioFlinger::RecordThread::RecordTrack::stop() 5379{ 5380 sp<ThreadBase> thread = mThread.promote(); 5381 if (thread != 0) { 5382 RecordThread *recordThread = (RecordThread *)thread.get(); 5383 recordThread->stop(this); 5384 TrackBase::reset(); 5385 // Force overrun condition to avoid false overrun callback until first data is 5386 // read from buffer 5387 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5388 } 5389} 5390 5391void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5392{ 5393 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5394 (mClient == 0) ? getpid_cached : mClient->pid(), 5395 mFormat, 5396 mChannelMask, 5397 mSessionId, 5398 mFrameCount, 5399 mState, 5400 mCblk->sampleRate, 5401 mCblk->server, 5402 mCblk->user); 5403} 5404 5405 5406// ---------------------------------------------------------------------------- 5407 5408AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5409 PlaybackThread *playbackThread, 5410 DuplicatingThread *sourceThread, 5411 uint32_t sampleRate, 5412 audio_format_t format, 5413 uint32_t channelMask, 5414 int frameCount) 5415 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5416 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5417 mActive(false), mSourceThread(sourceThread) 5418{ 5419 5420 if (mCblk != NULL) { 5421 mCblk->flags |= CBLK_DIRECTION_OUT; 5422 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5423 mOutBuffer.frameCount = 0; 5424 playbackThread->mTracks.add(this); 5425 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5426 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5427 mCblk, mBuffer, mCblk->buffers, 5428 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5429 } else { 5430 ALOGW("Error creating output track on thread %p", playbackThread); 5431 } 5432} 5433 5434AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5435{ 5436 clearBufferQueue(); 5437} 5438 5439status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5440 int triggerSession) 5441{ 5442 status_t status = Track::start(event, triggerSession); 5443 if (status != NO_ERROR) { 5444 return status; 5445 } 5446 5447 mActive = true; 5448 mRetryCount = 127; 5449 return status; 5450} 5451 5452void AudioFlinger::PlaybackThread::OutputTrack::stop() 5453{ 5454 Track::stop(); 5455 clearBufferQueue(); 5456 mOutBuffer.frameCount = 0; 5457 mActive = false; 5458} 5459 5460bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5461{ 5462 Buffer *pInBuffer; 5463 Buffer inBuffer; 5464 uint32_t channelCount = mChannelCount; 5465 bool outputBufferFull = false; 5466 inBuffer.frameCount = frames; 5467 inBuffer.i16 = data; 5468 5469 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5470 5471 if (!mActive && frames != 0) { 5472 start(); 5473 sp<ThreadBase> thread = mThread.promote(); 5474 if (thread != 0) { 5475 MixerThread *mixerThread = (MixerThread *)thread.get(); 5476 if (mCblk->frameCount > frames){ 5477 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5478 uint32_t startFrames = (mCblk->frameCount - frames); 5479 pInBuffer = new Buffer; 5480 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5481 pInBuffer->frameCount = startFrames; 5482 pInBuffer->i16 = pInBuffer->mBuffer; 5483 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5484 mBufferQueue.add(pInBuffer); 5485 } else { 5486 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5487 } 5488 } 5489 } 5490 } 5491 5492 while (waitTimeLeftMs) { 5493 // First write pending buffers, then new data 5494 if (mBufferQueue.size()) { 5495 pInBuffer = mBufferQueue.itemAt(0); 5496 } else { 5497 pInBuffer = &inBuffer; 5498 } 5499 5500 if (pInBuffer->frameCount == 0) { 5501 break; 5502 } 5503 5504 if (mOutBuffer.frameCount == 0) { 5505 mOutBuffer.frameCount = pInBuffer->frameCount; 5506 nsecs_t startTime = systemTime(); 5507 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5508 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5509 outputBufferFull = true; 5510 break; 5511 } 5512 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5513 if (waitTimeLeftMs >= waitTimeMs) { 5514 waitTimeLeftMs -= waitTimeMs; 5515 } else { 5516 waitTimeLeftMs = 0; 5517 } 5518 } 5519 5520 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5521 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5522 mCblk->stepUser(outFrames); 5523 pInBuffer->frameCount -= outFrames; 5524 pInBuffer->i16 += outFrames * channelCount; 5525 mOutBuffer.frameCount -= outFrames; 5526 mOutBuffer.i16 += outFrames * channelCount; 5527 5528 if (pInBuffer->frameCount == 0) { 5529 if (mBufferQueue.size()) { 5530 mBufferQueue.removeAt(0); 5531 delete [] pInBuffer->mBuffer; 5532 delete pInBuffer; 5533 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5534 } else { 5535 break; 5536 } 5537 } 5538 } 5539 5540 // If we could not write all frames, allocate a buffer and queue it for next time. 5541 if (inBuffer.frameCount) { 5542 sp<ThreadBase> thread = mThread.promote(); 5543 if (thread != 0 && !thread->standby()) { 5544 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5545 pInBuffer = new Buffer; 5546 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5547 pInBuffer->frameCount = inBuffer.frameCount; 5548 pInBuffer->i16 = pInBuffer->mBuffer; 5549 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5550 mBufferQueue.add(pInBuffer); 5551 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5552 } else { 5553 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5554 } 5555 } 5556 } 5557 5558 // Calling write() with a 0 length buffer, means that no more data will be written: 5559 // If no more buffers are pending, fill output track buffer to make sure it is started 5560 // by output mixer. 5561 if (frames == 0 && mBufferQueue.size() == 0) { 5562 if (mCblk->user < mCblk->frameCount) { 5563 frames = mCblk->frameCount - mCblk->user; 5564 pInBuffer = new Buffer; 5565 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5566 pInBuffer->frameCount = frames; 5567 pInBuffer->i16 = pInBuffer->mBuffer; 5568 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5569 mBufferQueue.add(pInBuffer); 5570 } else if (mActive) { 5571 stop(); 5572 } 5573 } 5574 5575 return outputBufferFull; 5576} 5577 5578status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5579{ 5580 int active; 5581 status_t result; 5582 audio_track_cblk_t* cblk = mCblk; 5583 uint32_t framesReq = buffer->frameCount; 5584 5585// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5586 buffer->frameCount = 0; 5587 5588 uint32_t framesAvail = cblk->framesAvailable(); 5589 5590 5591 if (framesAvail == 0) { 5592 Mutex::Autolock _l(cblk->lock); 5593 goto start_loop_here; 5594 while (framesAvail == 0) { 5595 active = mActive; 5596 if (CC_UNLIKELY(!active)) { 5597 ALOGV("Not active and NO_MORE_BUFFERS"); 5598 return NO_MORE_BUFFERS; 5599 } 5600 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5601 if (result != NO_ERROR) { 5602 return NO_MORE_BUFFERS; 5603 } 5604 // read the server count again 5605 start_loop_here: 5606 framesAvail = cblk->framesAvailable_l(); 5607 } 5608 } 5609 5610// if (framesAvail < framesReq) { 5611// return NO_MORE_BUFFERS; 5612// } 5613 5614 if (framesReq > framesAvail) { 5615 framesReq = framesAvail; 5616 } 5617 5618 uint32_t u = cblk->user; 5619 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5620 5621 if (framesReq > bufferEnd - u) { 5622 framesReq = bufferEnd - u; 5623 } 5624 5625 buffer->frameCount = framesReq; 5626 buffer->raw = (void *)cblk->buffer(u); 5627 return NO_ERROR; 5628} 5629 5630 5631void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5632{ 5633 size_t size = mBufferQueue.size(); 5634 5635 for (size_t i = 0; i < size; i++) { 5636 Buffer *pBuffer = mBufferQueue.itemAt(i); 5637 delete [] pBuffer->mBuffer; 5638 delete pBuffer; 5639 } 5640 mBufferQueue.clear(); 5641} 5642 5643// ---------------------------------------------------------------------------- 5644 5645AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5646 : RefBase(), 5647 mAudioFlinger(audioFlinger), 5648 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5649 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5650 mPid(pid), 5651 mTimedTrackCount(0) 5652{ 5653 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5654} 5655 5656// Client destructor must be called with AudioFlinger::mLock held 5657AudioFlinger::Client::~Client() 5658{ 5659 mAudioFlinger->removeClient_l(mPid); 5660} 5661 5662sp<MemoryDealer> AudioFlinger::Client::heap() const 5663{ 5664 return mMemoryDealer; 5665} 5666 5667// Reserve one of the limited slots for a timed audio track associated 5668// with this client 5669bool AudioFlinger::Client::reserveTimedTrack() 5670{ 5671 const int kMaxTimedTracksPerClient = 4; 5672 5673 Mutex::Autolock _l(mTimedTrackLock); 5674 5675 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5676 ALOGW("can not create timed track - pid %d has exceeded the limit", 5677 mPid); 5678 return false; 5679 } 5680 5681 mTimedTrackCount++; 5682 return true; 5683} 5684 5685// Release a slot for a timed audio track 5686void AudioFlinger::Client::releaseTimedTrack() 5687{ 5688 Mutex::Autolock _l(mTimedTrackLock); 5689 mTimedTrackCount--; 5690} 5691 5692// ---------------------------------------------------------------------------- 5693 5694AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5695 const sp<IAudioFlingerClient>& client, 5696 pid_t pid) 5697 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5698{ 5699} 5700 5701AudioFlinger::NotificationClient::~NotificationClient() 5702{ 5703} 5704 5705void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5706{ 5707 sp<NotificationClient> keep(this); 5708 mAudioFlinger->removeNotificationClient(mPid); 5709} 5710 5711// ---------------------------------------------------------------------------- 5712 5713AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5714 : BnAudioTrack(), 5715 mTrack(track) 5716{ 5717} 5718 5719AudioFlinger::TrackHandle::~TrackHandle() { 5720 // just stop the track on deletion, associated resources 5721 // will be freed from the main thread once all pending buffers have 5722 // been played. Unless it's not in the active track list, in which 5723 // case we free everything now... 5724 mTrack->destroy(); 5725} 5726 5727sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5728 return mTrack->getCblk(); 5729} 5730 5731status_t AudioFlinger::TrackHandle::start() { 5732 return mTrack->start(); 5733} 5734 5735void AudioFlinger::TrackHandle::stop() { 5736 mTrack->stop(); 5737} 5738 5739void AudioFlinger::TrackHandle::flush() { 5740 mTrack->flush(); 5741} 5742 5743void AudioFlinger::TrackHandle::mute(bool e) { 5744 mTrack->mute(e); 5745} 5746 5747void AudioFlinger::TrackHandle::pause() { 5748 mTrack->pause(); 5749} 5750 5751status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5752{ 5753 return mTrack->attachAuxEffect(EffectId); 5754} 5755 5756status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5757 sp<IMemory>* buffer) { 5758 if (!mTrack->isTimedTrack()) 5759 return INVALID_OPERATION; 5760 5761 PlaybackThread::TimedTrack* tt = 5762 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5763 return tt->allocateTimedBuffer(size, buffer); 5764} 5765 5766status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5767 int64_t pts) { 5768 if (!mTrack->isTimedTrack()) 5769 return INVALID_OPERATION; 5770 5771 PlaybackThread::TimedTrack* tt = 5772 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5773 return tt->queueTimedBuffer(buffer, pts); 5774} 5775 5776status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5777 const LinearTransform& xform, int target) { 5778 5779 if (!mTrack->isTimedTrack()) 5780 return INVALID_OPERATION; 5781 5782 PlaybackThread::TimedTrack* tt = 5783 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5784 return tt->setMediaTimeTransform( 5785 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5786} 5787 5788status_t AudioFlinger::TrackHandle::onTransact( 5789 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5790{ 5791 return BnAudioTrack::onTransact(code, data, reply, flags); 5792} 5793 5794// ---------------------------------------------------------------------------- 5795 5796sp<IAudioRecord> AudioFlinger::openRecord( 5797 pid_t pid, 5798 audio_io_handle_t input, 5799 uint32_t sampleRate, 5800 audio_format_t format, 5801 uint32_t channelMask, 5802 int frameCount, 5803 IAudioFlinger::track_flags_t flags, 5804 int *sessionId, 5805 status_t *status) 5806{ 5807 sp<RecordThread::RecordTrack> recordTrack; 5808 sp<RecordHandle> recordHandle; 5809 sp<Client> client; 5810 status_t lStatus; 5811 RecordThread *thread; 5812 size_t inFrameCount; 5813 int lSessionId; 5814 5815 // check calling permissions 5816 if (!recordingAllowed()) { 5817 lStatus = PERMISSION_DENIED; 5818 goto Exit; 5819 } 5820 5821 // add client to list 5822 { // scope for mLock 5823 Mutex::Autolock _l(mLock); 5824 thread = checkRecordThread_l(input); 5825 if (thread == NULL) { 5826 lStatus = BAD_VALUE; 5827 goto Exit; 5828 } 5829 5830 client = registerPid_l(pid); 5831 5832 // If no audio session id is provided, create one here 5833 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5834 lSessionId = *sessionId; 5835 } else { 5836 lSessionId = nextUniqueId(); 5837 if (sessionId != NULL) { 5838 *sessionId = lSessionId; 5839 } 5840 } 5841 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5842 recordTrack = thread->createRecordTrack_l(client, 5843 sampleRate, 5844 format, 5845 channelMask, 5846 frameCount, 5847 lSessionId, 5848 &lStatus); 5849 } 5850 if (lStatus != NO_ERROR) { 5851 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5852 // destructor is called by the TrackBase destructor with mLock held 5853 client.clear(); 5854 recordTrack.clear(); 5855 goto Exit; 5856 } 5857 5858 // return to handle to client 5859 recordHandle = new RecordHandle(recordTrack); 5860 lStatus = NO_ERROR; 5861 5862Exit: 5863 if (status) { 5864 *status = lStatus; 5865 } 5866 return recordHandle; 5867} 5868 5869// ---------------------------------------------------------------------------- 5870 5871AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5872 : BnAudioRecord(), 5873 mRecordTrack(recordTrack) 5874{ 5875} 5876 5877AudioFlinger::RecordHandle::~RecordHandle() { 5878 stop(); 5879} 5880 5881sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5882 return mRecordTrack->getCblk(); 5883} 5884 5885status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5886 ALOGV("RecordHandle::start()"); 5887 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5888} 5889 5890void AudioFlinger::RecordHandle::stop() { 5891 ALOGV("RecordHandle::stop()"); 5892 mRecordTrack->stop(); 5893} 5894 5895status_t AudioFlinger::RecordHandle::onTransact( 5896 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5897{ 5898 return BnAudioRecord::onTransact(code, data, reply, flags); 5899} 5900 5901// ---------------------------------------------------------------------------- 5902 5903AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5904 AudioStreamIn *input, 5905 uint32_t sampleRate, 5906 uint32_t channels, 5907 audio_io_handle_t id, 5908 uint32_t device) : 5909 ThreadBase(audioFlinger, id, device, RECORD), 5910 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5911 // mRsmpInIndex and mInputBytes set by readInputParameters() 5912 mReqChannelCount(popcount(channels)), 5913 mReqSampleRate(sampleRate) 5914 // mBytesRead is only meaningful while active, and so is cleared in start() 5915 // (but might be better to also clear here for dump?) 5916{ 5917 snprintf(mName, kNameLength, "AudioIn_%X", id); 5918 5919 readInputParameters(); 5920} 5921 5922 5923AudioFlinger::RecordThread::~RecordThread() 5924{ 5925 delete[] mRsmpInBuffer; 5926 delete mResampler; 5927 delete[] mRsmpOutBuffer; 5928} 5929 5930void AudioFlinger::RecordThread::onFirstRef() 5931{ 5932 run(mName, PRIORITY_URGENT_AUDIO); 5933} 5934 5935status_t AudioFlinger::RecordThread::readyToRun() 5936{ 5937 status_t status = initCheck(); 5938 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5939 return status; 5940} 5941 5942bool AudioFlinger::RecordThread::threadLoop() 5943{ 5944 AudioBufferProvider::Buffer buffer; 5945 sp<RecordTrack> activeTrack; 5946 Vector< sp<EffectChain> > effectChains; 5947 5948 nsecs_t lastWarning = 0; 5949 5950 acquireWakeLock(); 5951 5952 // start recording 5953 while (!exitPending()) { 5954 5955 processConfigEvents(); 5956 5957 { // scope for mLock 5958 Mutex::Autolock _l(mLock); 5959 checkForNewParameters_l(); 5960 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5961 if (!mStandby) { 5962 mInput->stream->common.standby(&mInput->stream->common); 5963 mStandby = true; 5964 } 5965 5966 if (exitPending()) break; 5967 5968 releaseWakeLock_l(); 5969 ALOGV("RecordThread: loop stopping"); 5970 // go to sleep 5971 mWaitWorkCV.wait(mLock); 5972 ALOGV("RecordThread: loop starting"); 5973 acquireWakeLock_l(); 5974 continue; 5975 } 5976 if (mActiveTrack != 0) { 5977 if (mActiveTrack->mState == TrackBase::PAUSING) { 5978 if (!mStandby) { 5979 mInput->stream->common.standby(&mInput->stream->common); 5980 mStandby = true; 5981 } 5982 mActiveTrack.clear(); 5983 mStartStopCond.broadcast(); 5984 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5985 if (mReqChannelCount != mActiveTrack->channelCount()) { 5986 mActiveTrack.clear(); 5987 mStartStopCond.broadcast(); 5988 } else if (mBytesRead != 0) { 5989 // record start succeeds only if first read from audio input 5990 // succeeds 5991 if (mBytesRead > 0) { 5992 mActiveTrack->mState = TrackBase::ACTIVE; 5993 } else { 5994 mActiveTrack.clear(); 5995 } 5996 mStartStopCond.broadcast(); 5997 } 5998 mStandby = false; 5999 } 6000 } 6001 lockEffectChains_l(effectChains); 6002 } 6003 6004 if (mActiveTrack != 0) { 6005 if (mActiveTrack->mState != TrackBase::ACTIVE && 6006 mActiveTrack->mState != TrackBase::RESUMING) { 6007 unlockEffectChains(effectChains); 6008 usleep(kRecordThreadSleepUs); 6009 continue; 6010 } 6011 for (size_t i = 0; i < effectChains.size(); i ++) { 6012 effectChains[i]->process_l(); 6013 } 6014 6015 buffer.frameCount = mFrameCount; 6016 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6017 size_t framesOut = buffer.frameCount; 6018 if (mResampler == NULL) { 6019 // no resampling 6020 while (framesOut) { 6021 size_t framesIn = mFrameCount - mRsmpInIndex; 6022 if (framesIn) { 6023 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6024 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6025 if (framesIn > framesOut) 6026 framesIn = framesOut; 6027 mRsmpInIndex += framesIn; 6028 framesOut -= framesIn; 6029 if ((int)mChannelCount == mReqChannelCount || 6030 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6031 memcpy(dst, src, framesIn * mFrameSize); 6032 } else { 6033 int16_t *src16 = (int16_t *)src; 6034 int16_t *dst16 = (int16_t *)dst; 6035 if (mChannelCount == 1) { 6036 while (framesIn--) { 6037 *dst16++ = *src16; 6038 *dst16++ = *src16++; 6039 } 6040 } else { 6041 while (framesIn--) { 6042 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6043 src16 += 2; 6044 } 6045 } 6046 } 6047 } 6048 if (framesOut && mFrameCount == mRsmpInIndex) { 6049 if (framesOut == mFrameCount && 6050 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6051 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6052 framesOut = 0; 6053 } else { 6054 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6055 mRsmpInIndex = 0; 6056 } 6057 if (mBytesRead < 0) { 6058 ALOGE("Error reading audio input"); 6059 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6060 // Force input into standby so that it tries to 6061 // recover at next read attempt 6062 mInput->stream->common.standby(&mInput->stream->common); 6063 usleep(kRecordThreadSleepUs); 6064 } 6065 mRsmpInIndex = mFrameCount; 6066 framesOut = 0; 6067 buffer.frameCount = 0; 6068 } 6069 } 6070 } 6071 } else { 6072 // resampling 6073 6074 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6075 // alter output frame count as if we were expecting stereo samples 6076 if (mChannelCount == 1 && mReqChannelCount == 1) { 6077 framesOut >>= 1; 6078 } 6079 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6080 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6081 // are 32 bit aligned which should be always true. 6082 if (mChannelCount == 2 && mReqChannelCount == 1) { 6083 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6084 // the resampler always outputs stereo samples: do post stereo to mono conversion 6085 int16_t *src = (int16_t *)mRsmpOutBuffer; 6086 int16_t *dst = buffer.i16; 6087 while (framesOut--) { 6088 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6089 src += 2; 6090 } 6091 } else { 6092 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6093 } 6094 6095 } 6096 if (mFramestoDrop == 0) { 6097 mActiveTrack->releaseBuffer(&buffer); 6098 } else { 6099 if (mFramestoDrop > 0) { 6100 mFramestoDrop -= buffer.frameCount; 6101 if (mFramestoDrop <= 0) { 6102 clearSyncStartEvent(); 6103 } 6104 } else { 6105 mFramestoDrop += buffer.frameCount; 6106 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6107 mSyncStartEvent->isCancelled()) { 6108 ALOGW("Synced record %s, session %d, trigger session %d", 6109 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6110 mActiveTrack->sessionId(), 6111 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6112 clearSyncStartEvent(); 6113 } 6114 } 6115 } 6116 mActiveTrack->overflow(); 6117 } 6118 // client isn't retrieving buffers fast enough 6119 else { 6120 if (!mActiveTrack->setOverflow()) { 6121 nsecs_t now = systemTime(); 6122 if ((now - lastWarning) > kWarningThrottleNs) { 6123 ALOGW("RecordThread: buffer overflow"); 6124 lastWarning = now; 6125 } 6126 } 6127 // Release the processor for a while before asking for a new buffer. 6128 // This will give the application more chance to read from the buffer and 6129 // clear the overflow. 6130 usleep(kRecordThreadSleepUs); 6131 } 6132 } 6133 // enable changes in effect chain 6134 unlockEffectChains(effectChains); 6135 effectChains.clear(); 6136 } 6137 6138 if (!mStandby) { 6139 mInput->stream->common.standby(&mInput->stream->common); 6140 } 6141 mActiveTrack.clear(); 6142 6143 mStartStopCond.broadcast(); 6144 6145 releaseWakeLock(); 6146 6147 ALOGV("RecordThread %p exiting", this); 6148 return false; 6149} 6150 6151 6152sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6153 const sp<AudioFlinger::Client>& client, 6154 uint32_t sampleRate, 6155 audio_format_t format, 6156 int channelMask, 6157 int frameCount, 6158 int sessionId, 6159 status_t *status) 6160{ 6161 sp<RecordTrack> track; 6162 status_t lStatus; 6163 6164 lStatus = initCheck(); 6165 if (lStatus != NO_ERROR) { 6166 ALOGE("Audio driver not initialized."); 6167 goto Exit; 6168 } 6169 6170 { // scope for mLock 6171 Mutex::Autolock _l(mLock); 6172 6173 track = new RecordTrack(this, client, sampleRate, 6174 format, channelMask, frameCount, sessionId); 6175 6176 if (track->getCblk() == 0) { 6177 lStatus = NO_MEMORY; 6178 goto Exit; 6179 } 6180 6181 mTrack = track.get(); 6182 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6183 bool suspend = audio_is_bluetooth_sco_device( 6184 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6185 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6186 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6187 } 6188 lStatus = NO_ERROR; 6189 6190Exit: 6191 if (status) { 6192 *status = lStatus; 6193 } 6194 return track; 6195} 6196 6197status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6198 AudioSystem::sync_event_t event, 6199 int triggerSession) 6200{ 6201 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6202 sp<ThreadBase> strongMe = this; 6203 status_t status = NO_ERROR; 6204 6205 if (event == AudioSystem::SYNC_EVENT_NONE) { 6206 clearSyncStartEvent(); 6207 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6208 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6209 triggerSession, 6210 recordTrack->sessionId(), 6211 syncStartEventCallback, 6212 this); 6213 // Sync event can be cancelled by the trigger session if the track is not in a 6214 // compatible state in which case we start record immediately 6215 if (mSyncStartEvent->isCancelled()) { 6216 clearSyncStartEvent(); 6217 } else { 6218 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6219 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6220 } 6221 } 6222 6223 { 6224 AutoMutex lock(mLock); 6225 if (mActiveTrack != 0) { 6226 if (recordTrack != mActiveTrack.get()) { 6227 status = -EBUSY; 6228 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6229 mActiveTrack->mState = TrackBase::ACTIVE; 6230 } 6231 return status; 6232 } 6233 6234 recordTrack->mState = TrackBase::IDLE; 6235 mActiveTrack = recordTrack; 6236 mLock.unlock(); 6237 status_t status = AudioSystem::startInput(mId); 6238 mLock.lock(); 6239 if (status != NO_ERROR) { 6240 mActiveTrack.clear(); 6241 clearSyncStartEvent(); 6242 return status; 6243 } 6244 mRsmpInIndex = mFrameCount; 6245 mBytesRead = 0; 6246 if (mResampler != NULL) { 6247 mResampler->reset(); 6248 } 6249 mActiveTrack->mState = TrackBase::RESUMING; 6250 // signal thread to start 6251 ALOGV("Signal record thread"); 6252 mWaitWorkCV.signal(); 6253 // do not wait for mStartStopCond if exiting 6254 if (exitPending()) { 6255 mActiveTrack.clear(); 6256 status = INVALID_OPERATION; 6257 goto startError; 6258 } 6259 mStartStopCond.wait(mLock); 6260 if (mActiveTrack == 0) { 6261 ALOGV("Record failed to start"); 6262 status = BAD_VALUE; 6263 goto startError; 6264 } 6265 ALOGV("Record started OK"); 6266 return status; 6267 } 6268startError: 6269 AudioSystem::stopInput(mId); 6270 clearSyncStartEvent(); 6271 return status; 6272} 6273 6274void AudioFlinger::RecordThread::clearSyncStartEvent() 6275{ 6276 if (mSyncStartEvent != 0) { 6277 mSyncStartEvent->cancel(); 6278 } 6279 mSyncStartEvent.clear(); 6280 mFramestoDrop = 0; 6281} 6282 6283void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6284{ 6285 sp<SyncEvent> strongEvent = event.promote(); 6286 6287 if (strongEvent != 0) { 6288 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6289 me->handleSyncStartEvent(strongEvent); 6290 } 6291} 6292 6293void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6294{ 6295 if (event == mSyncStartEvent) { 6296 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6297 // from audio HAL 6298 mFramestoDrop = mFrameCount * 2; 6299 } 6300} 6301 6302void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6303 ALOGV("RecordThread::stop"); 6304 sp<ThreadBase> strongMe = this; 6305 { 6306 AutoMutex lock(mLock); 6307 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6308 mActiveTrack->mState = TrackBase::PAUSING; 6309 // do not wait for mStartStopCond if exiting 6310 if (exitPending()) { 6311 return; 6312 } 6313 mStartStopCond.wait(mLock); 6314 // if we have been restarted, recordTrack == mActiveTrack.get() here 6315 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6316 mLock.unlock(); 6317 AudioSystem::stopInput(mId); 6318 mLock.lock(); 6319 ALOGV("Record stopped OK"); 6320 } 6321 } 6322 } 6323} 6324 6325bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6326{ 6327 return false; 6328} 6329 6330status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6331{ 6332 if (!isValidSyncEvent(event)) { 6333 return BAD_VALUE; 6334 } 6335 6336 Mutex::Autolock _l(mLock); 6337 6338 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6339 mTrack->setSyncEvent(event); 6340 return NO_ERROR; 6341 } 6342 return NAME_NOT_FOUND; 6343} 6344 6345status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6346{ 6347 const size_t SIZE = 256; 6348 char buffer[SIZE]; 6349 String8 result; 6350 6351 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6352 result.append(buffer); 6353 6354 if (mActiveTrack != 0) { 6355 result.append("Active Track:\n"); 6356 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6357 mActiveTrack->dump(buffer, SIZE); 6358 result.append(buffer); 6359 6360 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6361 result.append(buffer); 6362 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6363 result.append(buffer); 6364 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6365 result.append(buffer); 6366 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6367 result.append(buffer); 6368 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6369 result.append(buffer); 6370 6371 6372 } else { 6373 result.append("No record client\n"); 6374 } 6375 write(fd, result.string(), result.size()); 6376 6377 dumpBase(fd, args); 6378 dumpEffectChains(fd, args); 6379 6380 return NO_ERROR; 6381} 6382 6383// AudioBufferProvider interface 6384status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6385{ 6386 size_t framesReq = buffer->frameCount; 6387 size_t framesReady = mFrameCount - mRsmpInIndex; 6388 int channelCount; 6389 6390 if (framesReady == 0) { 6391 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6392 if (mBytesRead < 0) { 6393 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6394 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6395 // Force input into standby so that it tries to 6396 // recover at next read attempt 6397 mInput->stream->common.standby(&mInput->stream->common); 6398 usleep(kRecordThreadSleepUs); 6399 } 6400 buffer->raw = NULL; 6401 buffer->frameCount = 0; 6402 return NOT_ENOUGH_DATA; 6403 } 6404 mRsmpInIndex = 0; 6405 framesReady = mFrameCount; 6406 } 6407 6408 if (framesReq > framesReady) { 6409 framesReq = framesReady; 6410 } 6411 6412 if (mChannelCount == 1 && mReqChannelCount == 2) { 6413 channelCount = 1; 6414 } else { 6415 channelCount = 2; 6416 } 6417 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6418 buffer->frameCount = framesReq; 6419 return NO_ERROR; 6420} 6421 6422// AudioBufferProvider interface 6423void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6424{ 6425 mRsmpInIndex += buffer->frameCount; 6426 buffer->frameCount = 0; 6427} 6428 6429bool AudioFlinger::RecordThread::checkForNewParameters_l() 6430{ 6431 bool reconfig = false; 6432 6433 while (!mNewParameters.isEmpty()) { 6434 status_t status = NO_ERROR; 6435 String8 keyValuePair = mNewParameters[0]; 6436 AudioParameter param = AudioParameter(keyValuePair); 6437 int value; 6438 audio_format_t reqFormat = mFormat; 6439 int reqSamplingRate = mReqSampleRate; 6440 int reqChannelCount = mReqChannelCount; 6441 6442 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6443 reqSamplingRate = value; 6444 reconfig = true; 6445 } 6446 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6447 reqFormat = (audio_format_t) value; 6448 reconfig = true; 6449 } 6450 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6451 reqChannelCount = popcount(value); 6452 reconfig = true; 6453 } 6454 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6455 // do not accept frame count changes if tracks are open as the track buffer 6456 // size depends on frame count and correct behavior would not be guaranteed 6457 // if frame count is changed after track creation 6458 if (mActiveTrack != 0) { 6459 status = INVALID_OPERATION; 6460 } else { 6461 reconfig = true; 6462 } 6463 } 6464 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6465 // forward device change to effects that have requested to be 6466 // aware of attached audio device. 6467 for (size_t i = 0; i < mEffectChains.size(); i++) { 6468 mEffectChains[i]->setDevice_l(value); 6469 } 6470 // store input device and output device but do not forward output device to audio HAL. 6471 // Note that status is ignored by the caller for output device 6472 // (see AudioFlinger::setParameters() 6473 uint32_t /*audio_devices_t*/ newDevice = mDevice; 6474 if (value & AUDIO_DEVICE_OUT_ALL) { 6475 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6476 status = BAD_VALUE; 6477 } else { 6478 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6479 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6480 if (mTrack != NULL) { 6481 bool suspend = audio_is_bluetooth_sco_device( 6482 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6483 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6484 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6485 } 6486 } 6487 newDevice |= value; 6488 mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once 6489 } 6490 if (status == NO_ERROR) { 6491 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6492 if (status == INVALID_OPERATION) { 6493 mInput->stream->common.standby(&mInput->stream->common); 6494 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6495 keyValuePair.string()); 6496 } 6497 if (reconfig) { 6498 if (status == BAD_VALUE && 6499 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6500 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6501 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6502 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6503 (reqChannelCount <= FCC_2)) { 6504 status = NO_ERROR; 6505 } 6506 if (status == NO_ERROR) { 6507 readInputParameters(); 6508 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6509 } 6510 } 6511 } 6512 6513 mNewParameters.removeAt(0); 6514 6515 mParamStatus = status; 6516 mParamCond.signal(); 6517 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6518 // already timed out waiting for the status and will never signal the condition. 6519 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6520 } 6521 return reconfig; 6522} 6523 6524String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6525{ 6526 char *s; 6527 String8 out_s8 = String8(); 6528 6529 Mutex::Autolock _l(mLock); 6530 if (initCheck() != NO_ERROR) { 6531 return out_s8; 6532 } 6533 6534 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6535 out_s8 = String8(s); 6536 free(s); 6537 return out_s8; 6538} 6539 6540void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6541 AudioSystem::OutputDescriptor desc; 6542 void *param2 = NULL; 6543 6544 switch (event) { 6545 case AudioSystem::INPUT_OPENED: 6546 case AudioSystem::INPUT_CONFIG_CHANGED: 6547 desc.channels = mChannelMask; 6548 desc.samplingRate = mSampleRate; 6549 desc.format = mFormat; 6550 desc.frameCount = mFrameCount; 6551 desc.latency = 0; 6552 param2 = &desc; 6553 break; 6554 6555 case AudioSystem::INPUT_CLOSED: 6556 default: 6557 break; 6558 } 6559 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6560} 6561 6562void AudioFlinger::RecordThread::readInputParameters() 6563{ 6564 delete mRsmpInBuffer; 6565 // mRsmpInBuffer is always assigned a new[] below 6566 delete mRsmpOutBuffer; 6567 mRsmpOutBuffer = NULL; 6568 delete mResampler; 6569 mResampler = NULL; 6570 6571 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6572 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6573 mChannelCount = (uint16_t)popcount(mChannelMask); 6574 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6575 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6576 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6577 mFrameCount = mInputBytes / mFrameSize; 6578 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6579 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6580 6581 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6582 { 6583 int channelCount; 6584 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6585 // stereo to mono post process as the resampler always outputs stereo. 6586 if (mChannelCount == 1 && mReqChannelCount == 2) { 6587 channelCount = 1; 6588 } else { 6589 channelCount = 2; 6590 } 6591 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6592 mResampler->setSampleRate(mSampleRate); 6593 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6594 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6595 6596 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6597 if (mChannelCount == 1 && mReqChannelCount == 1) { 6598 mFrameCount >>= 1; 6599 } 6600 6601 } 6602 mRsmpInIndex = mFrameCount; 6603} 6604 6605unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6606{ 6607 Mutex::Autolock _l(mLock); 6608 if (initCheck() != NO_ERROR) { 6609 return 0; 6610 } 6611 6612 return mInput->stream->get_input_frames_lost(mInput->stream); 6613} 6614 6615uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6616{ 6617 Mutex::Autolock _l(mLock); 6618 uint32_t result = 0; 6619 if (getEffectChain_l(sessionId) != 0) { 6620 result = EFFECT_SESSION; 6621 } 6622 6623 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6624 result |= TRACK_SESSION; 6625 } 6626 6627 return result; 6628} 6629 6630AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6631{ 6632 Mutex::Autolock _l(mLock); 6633 return mTrack; 6634} 6635 6636AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6637{ 6638 Mutex::Autolock _l(mLock); 6639 return mInput; 6640} 6641 6642AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6643{ 6644 Mutex::Autolock _l(mLock); 6645 AudioStreamIn *input = mInput; 6646 mInput = NULL; 6647 return input; 6648} 6649 6650// this method must always be called either with ThreadBase mLock held or inside the thread loop 6651audio_stream_t* AudioFlinger::RecordThread::stream() const 6652{ 6653 if (mInput == NULL) { 6654 return NULL; 6655 } 6656 return &mInput->stream->common; 6657} 6658 6659 6660// ---------------------------------------------------------------------------- 6661 6662audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6663{ 6664 if (!settingsAllowed()) { 6665 return 0; 6666 } 6667 Mutex::Autolock _l(mLock); 6668 return loadHwModule_l(name); 6669} 6670 6671// loadHwModule_l() must be called with AudioFlinger::mLock held 6672audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6673{ 6674 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6675 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6676 ALOGW("loadHwModule() module %s already loaded", name); 6677 return mAudioHwDevs.keyAt(i); 6678 } 6679 } 6680 6681 audio_hw_device_t *dev; 6682 6683 int rc = load_audio_interface(name, &dev); 6684 if (rc) { 6685 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6686 return 0; 6687 } 6688 6689 mHardwareStatus = AUDIO_HW_INIT; 6690 rc = dev->init_check(dev); 6691 mHardwareStatus = AUDIO_HW_IDLE; 6692 if (rc) { 6693 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6694 return 0; 6695 } 6696 6697 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6698 (NULL != dev->set_master_volume)) { 6699 AutoMutex lock(mHardwareLock); 6700 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6701 dev->set_master_volume(dev, mMasterVolume); 6702 mHardwareStatus = AUDIO_HW_IDLE; 6703 } 6704 6705 audio_module_handle_t handle = nextUniqueId(); 6706 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6707 6708 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6709 name, dev->common.module->name, dev->common.module->id, handle); 6710 6711 return handle; 6712 6713} 6714 6715audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6716 audio_devices_t *pDevices, 6717 uint32_t *pSamplingRate, 6718 audio_format_t *pFormat, 6719 audio_channel_mask_t *pChannelMask, 6720 uint32_t *pLatencyMs, 6721 audio_output_flags_t flags) 6722{ 6723 status_t status; 6724 PlaybackThread *thread = NULL; 6725 struct audio_config config = { 6726 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6727 channel_mask: pChannelMask ? *pChannelMask : 0, 6728 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6729 }; 6730 audio_stream_out_t *outStream = NULL; 6731 audio_hw_device_t *outHwDev; 6732 6733 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6734 module, 6735 (pDevices != NULL) ? (int)*pDevices : 0, 6736 config.sample_rate, 6737 config.format, 6738 config.channel_mask, 6739 flags); 6740 6741 if (pDevices == NULL || *pDevices == 0) { 6742 return 0; 6743 } 6744 6745 Mutex::Autolock _l(mLock); 6746 6747 outHwDev = findSuitableHwDev_l(module, *pDevices); 6748 if (outHwDev == NULL) 6749 return 0; 6750 6751 audio_io_handle_t id = nextUniqueId(); 6752 6753 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6754 6755 status = outHwDev->open_output_stream(outHwDev, 6756 id, 6757 *pDevices, 6758 (audio_output_flags_t)flags, 6759 &config, 6760 &outStream); 6761 6762 mHardwareStatus = AUDIO_HW_IDLE; 6763 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6764 outStream, 6765 config.sample_rate, 6766 config.format, 6767 config.channel_mask, 6768 status); 6769 6770 if (status == NO_ERROR && outStream != NULL) { 6771 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6772 6773 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6774 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6775 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6776 thread = new DirectOutputThread(this, output, id, *pDevices); 6777 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6778 } else { 6779 thread = new MixerThread(this, output, id, *pDevices); 6780 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6781 } 6782 mPlaybackThreads.add(id, thread); 6783 6784 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6785 if (pFormat != NULL) *pFormat = config.format; 6786 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6787 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6788 6789 // notify client processes of the new output creation 6790 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6791 6792 // the first primary output opened designates the primary hw device 6793 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6794 ALOGI("Using module %d has the primary audio interface", module); 6795 mPrimaryHardwareDev = outHwDev; 6796 6797 AutoMutex lock(mHardwareLock); 6798 mHardwareStatus = AUDIO_HW_SET_MODE; 6799 outHwDev->set_mode(outHwDev, mMode); 6800 6801 // Determine the level of master volume support the primary audio HAL has, 6802 // and set the initial master volume at the same time. 6803 float initialVolume = 1.0; 6804 mMasterVolumeSupportLvl = MVS_NONE; 6805 6806 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6807 if ((NULL != outHwDev->get_master_volume) && 6808 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6809 mMasterVolumeSupportLvl = MVS_FULL; 6810 } else { 6811 mMasterVolumeSupportLvl = MVS_SETONLY; 6812 initialVolume = 1.0; 6813 } 6814 6815 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6816 if ((NULL == outHwDev->set_master_volume) || 6817 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6818 mMasterVolumeSupportLvl = MVS_NONE; 6819 } 6820 // now that we have a primary device, initialize master volume on other devices 6821 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6822 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6823 6824 if ((dev != mPrimaryHardwareDev) && 6825 (NULL != dev->set_master_volume)) { 6826 dev->set_master_volume(dev, initialVolume); 6827 } 6828 } 6829 mHardwareStatus = AUDIO_HW_IDLE; 6830 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6831 ? initialVolume 6832 : 1.0; 6833 mMasterVolume = initialVolume; 6834 } 6835 return id; 6836 } 6837 6838 return 0; 6839} 6840 6841audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6842 audio_io_handle_t output2) 6843{ 6844 Mutex::Autolock _l(mLock); 6845 MixerThread *thread1 = checkMixerThread_l(output1); 6846 MixerThread *thread2 = checkMixerThread_l(output2); 6847 6848 if (thread1 == NULL || thread2 == NULL) { 6849 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6850 return 0; 6851 } 6852 6853 audio_io_handle_t id = nextUniqueId(); 6854 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6855 thread->addOutputTrack(thread2); 6856 mPlaybackThreads.add(id, thread); 6857 // notify client processes of the new output creation 6858 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6859 return id; 6860} 6861 6862status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6863{ 6864 // keep strong reference on the playback thread so that 6865 // it is not destroyed while exit() is executed 6866 sp<PlaybackThread> thread; 6867 { 6868 Mutex::Autolock _l(mLock); 6869 thread = checkPlaybackThread_l(output); 6870 if (thread == NULL) { 6871 return BAD_VALUE; 6872 } 6873 6874 ALOGV("closeOutput() %d", output); 6875 6876 if (thread->type() == ThreadBase::MIXER) { 6877 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6878 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6879 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6880 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6881 } 6882 } 6883 } 6884 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6885 mPlaybackThreads.removeItem(output); 6886 } 6887 thread->exit(); 6888 // The thread entity (active unit of execution) is no longer running here, 6889 // but the ThreadBase container still exists. 6890 6891 if (thread->type() != ThreadBase::DUPLICATING) { 6892 AudioStreamOut *out = thread->clearOutput(); 6893 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6894 // from now on thread->mOutput is NULL 6895 out->hwDev->close_output_stream(out->hwDev, out->stream); 6896 delete out; 6897 } 6898 return NO_ERROR; 6899} 6900 6901status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6902{ 6903 Mutex::Autolock _l(mLock); 6904 PlaybackThread *thread = checkPlaybackThread_l(output); 6905 6906 if (thread == NULL) { 6907 return BAD_VALUE; 6908 } 6909 6910 ALOGV("suspendOutput() %d", output); 6911 thread->suspend(); 6912 6913 return NO_ERROR; 6914} 6915 6916status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6917{ 6918 Mutex::Autolock _l(mLock); 6919 PlaybackThread *thread = checkPlaybackThread_l(output); 6920 6921 if (thread == NULL) { 6922 return BAD_VALUE; 6923 } 6924 6925 ALOGV("restoreOutput() %d", output); 6926 6927 thread->restore(); 6928 6929 return NO_ERROR; 6930} 6931 6932audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6933 audio_devices_t *pDevices, 6934 uint32_t *pSamplingRate, 6935 audio_format_t *pFormat, 6936 uint32_t *pChannelMask) 6937{ 6938 status_t status; 6939 RecordThread *thread = NULL; 6940 struct audio_config config = { 6941 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6942 channel_mask: pChannelMask ? *pChannelMask : 0, 6943 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6944 }; 6945 uint32_t reqSamplingRate = config.sample_rate; 6946 audio_format_t reqFormat = config.format; 6947 audio_channel_mask_t reqChannels = config.channel_mask; 6948 audio_stream_in_t *inStream = NULL; 6949 audio_hw_device_t *inHwDev; 6950 6951 if (pDevices == NULL || *pDevices == 0) { 6952 return 0; 6953 } 6954 6955 Mutex::Autolock _l(mLock); 6956 6957 inHwDev = findSuitableHwDev_l(module, *pDevices); 6958 if (inHwDev == NULL) 6959 return 0; 6960 6961 audio_io_handle_t id = nextUniqueId(); 6962 6963 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6964 &inStream); 6965 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6966 inStream, 6967 config.sample_rate, 6968 config.format, 6969 config.channel_mask, 6970 status); 6971 6972 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6973 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6974 // or stereo to mono conversions on 16 bit PCM inputs. 6975 if (status == BAD_VALUE && 6976 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6977 (config.sample_rate <= 2 * reqSamplingRate) && 6978 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6979 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6980 inStream = NULL; 6981 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6982 } 6983 6984 if (status == NO_ERROR && inStream != NULL) { 6985 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6986 6987 // Start record thread 6988 // RecorThread require both input and output device indication to forward to audio 6989 // pre processing modules 6990 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6991 thread = new RecordThread(this, 6992 input, 6993 reqSamplingRate, 6994 reqChannels, 6995 id, 6996 device); 6997 mRecordThreads.add(id, thread); 6998 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6999 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7000 if (pFormat != NULL) *pFormat = config.format; 7001 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7002 7003 input->stream->common.standby(&input->stream->common); 7004 7005 // notify client processes of the new input creation 7006 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7007 return id; 7008 } 7009 7010 return 0; 7011} 7012 7013status_t AudioFlinger::closeInput(audio_io_handle_t input) 7014{ 7015 // keep strong reference on the record thread so that 7016 // it is not destroyed while exit() is executed 7017 sp<RecordThread> thread; 7018 { 7019 Mutex::Autolock _l(mLock); 7020 thread = checkRecordThread_l(input); 7021 if (thread == 0) { 7022 return BAD_VALUE; 7023 } 7024 7025 ALOGV("closeInput() %d", input); 7026 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7027 mRecordThreads.removeItem(input); 7028 } 7029 thread->exit(); 7030 // The thread entity (active unit of execution) is no longer running here, 7031 // but the ThreadBase container still exists. 7032 7033 AudioStreamIn *in = thread->clearInput(); 7034 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7035 // from now on thread->mInput is NULL 7036 in->hwDev->close_input_stream(in->hwDev, in->stream); 7037 delete in; 7038 7039 return NO_ERROR; 7040} 7041 7042status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7043{ 7044 Mutex::Autolock _l(mLock); 7045 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7046 7047 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7048 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7049 thread->invalidateTracks(stream); 7050 } 7051 7052 return NO_ERROR; 7053} 7054 7055 7056int AudioFlinger::newAudioSessionId() 7057{ 7058 return nextUniqueId(); 7059} 7060 7061void AudioFlinger::acquireAudioSessionId(int audioSession) 7062{ 7063 Mutex::Autolock _l(mLock); 7064 pid_t caller = IPCThreadState::self()->getCallingPid(); 7065 ALOGV("acquiring %d from %d", audioSession, caller); 7066 size_t num = mAudioSessionRefs.size(); 7067 for (size_t i = 0; i< num; i++) { 7068 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7069 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7070 ref->mCnt++; 7071 ALOGV(" incremented refcount to %d", ref->mCnt); 7072 return; 7073 } 7074 } 7075 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7076 ALOGV(" added new entry for %d", audioSession); 7077} 7078 7079void AudioFlinger::releaseAudioSessionId(int audioSession) 7080{ 7081 Mutex::Autolock _l(mLock); 7082 pid_t caller = IPCThreadState::self()->getCallingPid(); 7083 ALOGV("releasing %d from %d", audioSession, caller); 7084 size_t num = mAudioSessionRefs.size(); 7085 for (size_t i = 0; i< num; i++) { 7086 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7087 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7088 ref->mCnt--; 7089 ALOGV(" decremented refcount to %d", ref->mCnt); 7090 if (ref->mCnt == 0) { 7091 mAudioSessionRefs.removeAt(i); 7092 delete ref; 7093 purgeStaleEffects_l(); 7094 } 7095 return; 7096 } 7097 } 7098 ALOGW("session id %d not found for pid %d", audioSession, caller); 7099} 7100 7101void AudioFlinger::purgeStaleEffects_l() { 7102 7103 ALOGV("purging stale effects"); 7104 7105 Vector< sp<EffectChain> > chains; 7106 7107 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7108 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7109 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7110 sp<EffectChain> ec = t->mEffectChains[j]; 7111 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7112 chains.push(ec); 7113 } 7114 } 7115 } 7116 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7117 sp<RecordThread> t = mRecordThreads.valueAt(i); 7118 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7119 sp<EffectChain> ec = t->mEffectChains[j]; 7120 chains.push(ec); 7121 } 7122 } 7123 7124 for (size_t i = 0; i < chains.size(); i++) { 7125 sp<EffectChain> ec = chains[i]; 7126 int sessionid = ec->sessionId(); 7127 sp<ThreadBase> t = ec->mThread.promote(); 7128 if (t == 0) { 7129 continue; 7130 } 7131 size_t numsessionrefs = mAudioSessionRefs.size(); 7132 bool found = false; 7133 for (size_t k = 0; k < numsessionrefs; k++) { 7134 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7135 if (ref->mSessionid == sessionid) { 7136 ALOGV(" session %d still exists for %d with %d refs", 7137 sessionid, ref->mPid, ref->mCnt); 7138 found = true; 7139 break; 7140 } 7141 } 7142 if (!found) { 7143 Mutex::Autolock _l (t->mLock); 7144 // remove all effects from the chain 7145 while (ec->mEffects.size()) { 7146 sp<EffectModule> effect = ec->mEffects[0]; 7147 effect->unPin(); 7148 t->removeEffect_l(effect); 7149 if (effect->purgeHandles()) { 7150 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7151 } 7152 AudioSystem::unregisterEffect(effect->id()); 7153 } 7154 } 7155 } 7156 return; 7157} 7158 7159// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7160AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7161{ 7162 return mPlaybackThreads.valueFor(output).get(); 7163} 7164 7165// checkMixerThread_l() must be called with AudioFlinger::mLock held 7166AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7167{ 7168 PlaybackThread *thread = checkPlaybackThread_l(output); 7169 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7170} 7171 7172// checkRecordThread_l() must be called with AudioFlinger::mLock held 7173AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7174{ 7175 return mRecordThreads.valueFor(input).get(); 7176} 7177 7178uint32_t AudioFlinger::nextUniqueId() 7179{ 7180 return android_atomic_inc(&mNextUniqueId); 7181} 7182 7183AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7184{ 7185 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7186 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7187 AudioStreamOut *output = thread->getOutput(); 7188 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7189 return thread; 7190 } 7191 } 7192 return NULL; 7193} 7194 7195uint32_t AudioFlinger::primaryOutputDevice_l() const 7196{ 7197 PlaybackThread *thread = primaryPlaybackThread_l(); 7198 7199 if (thread == NULL) { 7200 return 0; 7201 } 7202 7203 return thread->device(); 7204} 7205 7206sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7207 int triggerSession, 7208 int listenerSession, 7209 sync_event_callback_t callBack, 7210 void *cookie) 7211{ 7212 Mutex::Autolock _l(mLock); 7213 7214 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7215 status_t playStatus = NAME_NOT_FOUND; 7216 status_t recStatus = NAME_NOT_FOUND; 7217 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7218 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7219 if (playStatus == NO_ERROR) { 7220 return event; 7221 } 7222 } 7223 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7224 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7225 if (recStatus == NO_ERROR) { 7226 return event; 7227 } 7228 } 7229 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7230 mPendingSyncEvents.add(event); 7231 } else { 7232 ALOGV("createSyncEvent() invalid event %d", event->type()); 7233 event.clear(); 7234 } 7235 return event; 7236} 7237 7238// ---------------------------------------------------------------------------- 7239// Effect management 7240// ---------------------------------------------------------------------------- 7241 7242 7243status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7244{ 7245 Mutex::Autolock _l(mLock); 7246 return EffectQueryNumberEffects(numEffects); 7247} 7248 7249status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7250{ 7251 Mutex::Autolock _l(mLock); 7252 return EffectQueryEffect(index, descriptor); 7253} 7254 7255status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7256 effect_descriptor_t *descriptor) const 7257{ 7258 Mutex::Autolock _l(mLock); 7259 return EffectGetDescriptor(pUuid, descriptor); 7260} 7261 7262 7263sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7264 effect_descriptor_t *pDesc, 7265 const sp<IEffectClient>& effectClient, 7266 int32_t priority, 7267 audio_io_handle_t io, 7268 int sessionId, 7269 status_t *status, 7270 int *id, 7271 int *enabled) 7272{ 7273 status_t lStatus = NO_ERROR; 7274 sp<EffectHandle> handle; 7275 effect_descriptor_t desc; 7276 7277 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7278 pid, effectClient.get(), priority, sessionId, io); 7279 7280 if (pDesc == NULL) { 7281 lStatus = BAD_VALUE; 7282 goto Exit; 7283 } 7284 7285 // check audio settings permission for global effects 7286 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7287 lStatus = PERMISSION_DENIED; 7288 goto Exit; 7289 } 7290 7291 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7292 // that can only be created by audio policy manager (running in same process) 7293 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7294 lStatus = PERMISSION_DENIED; 7295 goto Exit; 7296 } 7297 7298 if (io == 0) { 7299 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7300 // output must be specified by AudioPolicyManager when using session 7301 // AUDIO_SESSION_OUTPUT_STAGE 7302 lStatus = BAD_VALUE; 7303 goto Exit; 7304 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7305 // if the output returned by getOutputForEffect() is removed before we lock the 7306 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7307 // and we will exit safely 7308 io = AudioSystem::getOutputForEffect(&desc); 7309 } 7310 } 7311 7312 { 7313 Mutex::Autolock _l(mLock); 7314 7315 7316 if (!EffectIsNullUuid(&pDesc->uuid)) { 7317 // if uuid is specified, request effect descriptor 7318 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7319 if (lStatus < 0) { 7320 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7321 goto Exit; 7322 } 7323 } else { 7324 // if uuid is not specified, look for an available implementation 7325 // of the required type in effect factory 7326 if (EffectIsNullUuid(&pDesc->type)) { 7327 ALOGW("createEffect() no effect type"); 7328 lStatus = BAD_VALUE; 7329 goto Exit; 7330 } 7331 uint32_t numEffects = 0; 7332 effect_descriptor_t d; 7333 d.flags = 0; // prevent compiler warning 7334 bool found = false; 7335 7336 lStatus = EffectQueryNumberEffects(&numEffects); 7337 if (lStatus < 0) { 7338 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7339 goto Exit; 7340 } 7341 for (uint32_t i = 0; i < numEffects; i++) { 7342 lStatus = EffectQueryEffect(i, &desc); 7343 if (lStatus < 0) { 7344 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7345 continue; 7346 } 7347 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7348 // If matching type found save effect descriptor. If the session is 7349 // 0 and the effect is not auxiliary, continue enumeration in case 7350 // an auxiliary version of this effect type is available 7351 found = true; 7352 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7353 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7354 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7355 break; 7356 } 7357 } 7358 } 7359 if (!found) { 7360 lStatus = BAD_VALUE; 7361 ALOGW("createEffect() effect not found"); 7362 goto Exit; 7363 } 7364 // For same effect type, chose auxiliary version over insert version if 7365 // connect to output mix (Compliance to OpenSL ES) 7366 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7367 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7368 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7369 } 7370 } 7371 7372 // Do not allow auxiliary effects on a session different from 0 (output mix) 7373 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7374 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7375 lStatus = INVALID_OPERATION; 7376 goto Exit; 7377 } 7378 7379 // check recording permission for visualizer 7380 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7381 !recordingAllowed()) { 7382 lStatus = PERMISSION_DENIED; 7383 goto Exit; 7384 } 7385 7386 // return effect descriptor 7387 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7388 7389 // If output is not specified try to find a matching audio session ID in one of the 7390 // output threads. 7391 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7392 // because of code checking output when entering the function. 7393 // Note: io is never 0 when creating an effect on an input 7394 if (io == 0) { 7395 // look for the thread where the specified audio session is present 7396 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7397 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7398 io = mPlaybackThreads.keyAt(i); 7399 break; 7400 } 7401 } 7402 if (io == 0) { 7403 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7404 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7405 io = mRecordThreads.keyAt(i); 7406 break; 7407 } 7408 } 7409 } 7410 // If no output thread contains the requested session ID, default to 7411 // first output. The effect chain will be moved to the correct output 7412 // thread when a track with the same session ID is created 7413 if (io == 0 && mPlaybackThreads.size()) { 7414 io = mPlaybackThreads.keyAt(0); 7415 } 7416 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7417 } 7418 ThreadBase *thread = checkRecordThread_l(io); 7419 if (thread == NULL) { 7420 thread = checkPlaybackThread_l(io); 7421 if (thread == NULL) { 7422 ALOGE("createEffect() unknown output thread"); 7423 lStatus = BAD_VALUE; 7424 goto Exit; 7425 } 7426 } 7427 7428 sp<Client> client = registerPid_l(pid); 7429 7430 // create effect on selected output thread 7431 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7432 &desc, enabled, &lStatus); 7433 if (handle != 0 && id != NULL) { 7434 *id = handle->id(); 7435 } 7436 } 7437 7438Exit: 7439 if (status != NULL) { 7440 *status = lStatus; 7441 } 7442 return handle; 7443} 7444 7445status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7446 audio_io_handle_t dstOutput) 7447{ 7448 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7449 sessionId, srcOutput, dstOutput); 7450 Mutex::Autolock _l(mLock); 7451 if (srcOutput == dstOutput) { 7452 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7453 return NO_ERROR; 7454 } 7455 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7456 if (srcThread == NULL) { 7457 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7458 return BAD_VALUE; 7459 } 7460 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7461 if (dstThread == NULL) { 7462 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7463 return BAD_VALUE; 7464 } 7465 7466 Mutex::Autolock _dl(dstThread->mLock); 7467 Mutex::Autolock _sl(srcThread->mLock); 7468 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7469 7470 return NO_ERROR; 7471} 7472 7473// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7474status_t AudioFlinger::moveEffectChain_l(int sessionId, 7475 AudioFlinger::PlaybackThread *srcThread, 7476 AudioFlinger::PlaybackThread *dstThread, 7477 bool reRegister) 7478{ 7479 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7480 sessionId, srcThread, dstThread); 7481 7482 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7483 if (chain == 0) { 7484 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7485 sessionId, srcThread); 7486 return INVALID_OPERATION; 7487 } 7488 7489 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7490 // so that a new chain is created with correct parameters when first effect is added. This is 7491 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7492 // removed. 7493 srcThread->removeEffectChain_l(chain); 7494 7495 // transfer all effects one by one so that new effect chain is created on new thread with 7496 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7497 audio_io_handle_t dstOutput = dstThread->id(); 7498 sp<EffectChain> dstChain; 7499 uint32_t strategy = 0; // prevent compiler warning 7500 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7501 while (effect != 0) { 7502 srcThread->removeEffect_l(effect); 7503 dstThread->addEffect_l(effect); 7504 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7505 if (effect->state() == EffectModule::ACTIVE || 7506 effect->state() == EffectModule::STOPPING) { 7507 effect->start(); 7508 } 7509 // if the move request is not received from audio policy manager, the effect must be 7510 // re-registered with the new strategy and output 7511 if (dstChain == 0) { 7512 dstChain = effect->chain().promote(); 7513 if (dstChain == 0) { 7514 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7515 srcThread->addEffect_l(effect); 7516 return NO_INIT; 7517 } 7518 strategy = dstChain->strategy(); 7519 } 7520 if (reRegister) { 7521 AudioSystem::unregisterEffect(effect->id()); 7522 AudioSystem::registerEffect(&effect->desc(), 7523 dstOutput, 7524 strategy, 7525 sessionId, 7526 effect->id()); 7527 } 7528 effect = chain->getEffectFromId_l(0); 7529 } 7530 7531 return NO_ERROR; 7532} 7533 7534 7535// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7536sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7537 const sp<AudioFlinger::Client>& client, 7538 const sp<IEffectClient>& effectClient, 7539 int32_t priority, 7540 int sessionId, 7541 effect_descriptor_t *desc, 7542 int *enabled, 7543 status_t *status 7544 ) 7545{ 7546 sp<EffectModule> effect; 7547 sp<EffectHandle> handle; 7548 status_t lStatus; 7549 sp<EffectChain> chain; 7550 bool chainCreated = false; 7551 bool effectCreated = false; 7552 bool effectRegistered = false; 7553 7554 lStatus = initCheck(); 7555 if (lStatus != NO_ERROR) { 7556 ALOGW("createEffect_l() Audio driver not initialized."); 7557 goto Exit; 7558 } 7559 7560 // Do not allow effects with session ID 0 on direct output or duplicating threads 7561 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7562 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7563 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7564 desc->name, sessionId); 7565 lStatus = BAD_VALUE; 7566 goto Exit; 7567 } 7568 // Only Pre processor effects are allowed on input threads and only on input threads 7569 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7570 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7571 desc->name, desc->flags, mType); 7572 lStatus = BAD_VALUE; 7573 goto Exit; 7574 } 7575 7576 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7577 7578 { // scope for mLock 7579 Mutex::Autolock _l(mLock); 7580 7581 // check for existing effect chain with the requested audio session 7582 chain = getEffectChain_l(sessionId); 7583 if (chain == 0) { 7584 // create a new chain for this session 7585 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7586 chain = new EffectChain(this, sessionId); 7587 addEffectChain_l(chain); 7588 chain->setStrategy(getStrategyForSession_l(sessionId)); 7589 chainCreated = true; 7590 } else { 7591 effect = chain->getEffectFromDesc_l(desc); 7592 } 7593 7594 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7595 7596 if (effect == 0) { 7597 int id = mAudioFlinger->nextUniqueId(); 7598 // Check CPU and memory usage 7599 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7600 if (lStatus != NO_ERROR) { 7601 goto Exit; 7602 } 7603 effectRegistered = true; 7604 // create a new effect module if none present in the chain 7605 effect = new EffectModule(this, chain, desc, id, sessionId); 7606 lStatus = effect->status(); 7607 if (lStatus != NO_ERROR) { 7608 goto Exit; 7609 } 7610 lStatus = chain->addEffect_l(effect); 7611 if (lStatus != NO_ERROR) { 7612 goto Exit; 7613 } 7614 effectCreated = true; 7615 7616 effect->setDevice(mDevice); 7617 effect->setMode(mAudioFlinger->getMode()); 7618 } 7619 // create effect handle and connect it to effect module 7620 handle = new EffectHandle(effect, client, effectClient, priority); 7621 lStatus = effect->addHandle(handle.get()); 7622 if (enabled != NULL) { 7623 *enabled = (int)effect->isEnabled(); 7624 } 7625 } 7626 7627Exit: 7628 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7629 Mutex::Autolock _l(mLock); 7630 if (effectCreated) { 7631 chain->removeEffect_l(effect); 7632 } 7633 if (effectRegistered) { 7634 AudioSystem::unregisterEffect(effect->id()); 7635 } 7636 if (chainCreated) { 7637 removeEffectChain_l(chain); 7638 } 7639 handle.clear(); 7640 } 7641 7642 if (status != NULL) { 7643 *status = lStatus; 7644 } 7645 return handle; 7646} 7647 7648sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7649{ 7650 Mutex::Autolock _l(mLock); 7651 return getEffect_l(sessionId, effectId); 7652} 7653 7654sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7655{ 7656 sp<EffectChain> chain = getEffectChain_l(sessionId); 7657 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7658} 7659 7660// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7661// PlaybackThread::mLock held 7662status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7663{ 7664 // check for existing effect chain with the requested audio session 7665 int sessionId = effect->sessionId(); 7666 sp<EffectChain> chain = getEffectChain_l(sessionId); 7667 bool chainCreated = false; 7668 7669 if (chain == 0) { 7670 // create a new chain for this session 7671 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7672 chain = new EffectChain(this, sessionId); 7673 addEffectChain_l(chain); 7674 chain->setStrategy(getStrategyForSession_l(sessionId)); 7675 chainCreated = true; 7676 } 7677 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7678 7679 if (chain->getEffectFromId_l(effect->id()) != 0) { 7680 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7681 this, effect->desc().name, chain.get()); 7682 return BAD_VALUE; 7683 } 7684 7685 status_t status = chain->addEffect_l(effect); 7686 if (status != NO_ERROR) { 7687 if (chainCreated) { 7688 removeEffectChain_l(chain); 7689 } 7690 return status; 7691 } 7692 7693 effect->setDevice(mDevice); 7694 effect->setMode(mAudioFlinger->getMode()); 7695 return NO_ERROR; 7696} 7697 7698void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7699 7700 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7701 effect_descriptor_t desc = effect->desc(); 7702 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7703 detachAuxEffect_l(effect->id()); 7704 } 7705 7706 sp<EffectChain> chain = effect->chain().promote(); 7707 if (chain != 0) { 7708 // remove effect chain if removing last effect 7709 if (chain->removeEffect_l(effect) == 0) { 7710 removeEffectChain_l(chain); 7711 } 7712 } else { 7713 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7714 } 7715} 7716 7717void AudioFlinger::ThreadBase::lockEffectChains_l( 7718 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7719{ 7720 effectChains = mEffectChains; 7721 for (size_t i = 0; i < mEffectChains.size(); i++) { 7722 mEffectChains[i]->lock(); 7723 } 7724} 7725 7726void AudioFlinger::ThreadBase::unlockEffectChains( 7727 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7728{ 7729 for (size_t i = 0; i < effectChains.size(); i++) { 7730 effectChains[i]->unlock(); 7731 } 7732} 7733 7734sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7735{ 7736 Mutex::Autolock _l(mLock); 7737 return getEffectChain_l(sessionId); 7738} 7739 7740sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7741{ 7742 size_t size = mEffectChains.size(); 7743 for (size_t i = 0; i < size; i++) { 7744 if (mEffectChains[i]->sessionId() == sessionId) { 7745 return mEffectChains[i]; 7746 } 7747 } 7748 return 0; 7749} 7750 7751void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7752{ 7753 Mutex::Autolock _l(mLock); 7754 size_t size = mEffectChains.size(); 7755 for (size_t i = 0; i < size; i++) { 7756 mEffectChains[i]->setMode_l(mode); 7757 } 7758} 7759 7760void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7761 EffectHandle *handle, 7762 bool unpinIfLast) { 7763 7764 Mutex::Autolock _l(mLock); 7765 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7766 // delete the effect module if removing last handle on it 7767 if (effect->removeHandle(handle) == 0) { 7768 if (!effect->isPinned() || unpinIfLast) { 7769 removeEffect_l(effect); 7770 AudioSystem::unregisterEffect(effect->id()); 7771 } 7772 } 7773} 7774 7775status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7776{ 7777 int session = chain->sessionId(); 7778 int16_t *buffer = mMixBuffer; 7779 bool ownsBuffer = false; 7780 7781 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7782 if (session > 0) { 7783 // Only one effect chain can be present in direct output thread and it uses 7784 // the mix buffer as input 7785 if (mType != DIRECT) { 7786 size_t numSamples = mNormalFrameCount * mChannelCount; 7787 buffer = new int16_t[numSamples]; 7788 memset(buffer, 0, numSamples * sizeof(int16_t)); 7789 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7790 ownsBuffer = true; 7791 } 7792 7793 // Attach all tracks with same session ID to this chain. 7794 for (size_t i = 0; i < mTracks.size(); ++i) { 7795 sp<Track> track = mTracks[i]; 7796 if (session == track->sessionId()) { 7797 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7798 track->setMainBuffer(buffer); 7799 chain->incTrackCnt(); 7800 } 7801 } 7802 7803 // indicate all active tracks in the chain 7804 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7805 sp<Track> track = mActiveTracks[i].promote(); 7806 if (track == 0) continue; 7807 if (session == track->sessionId()) { 7808 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7809 chain->incActiveTrackCnt(); 7810 } 7811 } 7812 } 7813 7814 chain->setInBuffer(buffer, ownsBuffer); 7815 chain->setOutBuffer(mMixBuffer); 7816 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7817 // chains list in order to be processed last as it contains output stage effects 7818 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7819 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7820 // after track specific effects and before output stage 7821 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7822 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7823 // Effect chain for other sessions are inserted at beginning of effect 7824 // chains list to be processed before output mix effects. Relative order between other 7825 // sessions is not important 7826 size_t size = mEffectChains.size(); 7827 size_t i = 0; 7828 for (i = 0; i < size; i++) { 7829 if (mEffectChains[i]->sessionId() < session) break; 7830 } 7831 mEffectChains.insertAt(chain, i); 7832 checkSuspendOnAddEffectChain_l(chain); 7833 7834 return NO_ERROR; 7835} 7836 7837size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7838{ 7839 int session = chain->sessionId(); 7840 7841 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7842 7843 for (size_t i = 0; i < mEffectChains.size(); i++) { 7844 if (chain == mEffectChains[i]) { 7845 mEffectChains.removeAt(i); 7846 // detach all active tracks from the chain 7847 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7848 sp<Track> track = mActiveTracks[i].promote(); 7849 if (track == 0) continue; 7850 if (session == track->sessionId()) { 7851 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7852 chain.get(), session); 7853 chain->decActiveTrackCnt(); 7854 } 7855 } 7856 7857 // detach all tracks with same session ID from this chain 7858 for (size_t i = 0; i < mTracks.size(); ++i) { 7859 sp<Track> track = mTracks[i]; 7860 if (session == track->sessionId()) { 7861 track->setMainBuffer(mMixBuffer); 7862 chain->decTrackCnt(); 7863 } 7864 } 7865 break; 7866 } 7867 } 7868 return mEffectChains.size(); 7869} 7870 7871status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7872 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7873{ 7874 Mutex::Autolock _l(mLock); 7875 return attachAuxEffect_l(track, EffectId); 7876} 7877 7878status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7879 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7880{ 7881 status_t status = NO_ERROR; 7882 7883 if (EffectId == 0) { 7884 track->setAuxBuffer(0, NULL); 7885 } else { 7886 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7887 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7888 if (effect != 0) { 7889 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7890 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7891 } else { 7892 status = INVALID_OPERATION; 7893 } 7894 } else { 7895 status = BAD_VALUE; 7896 } 7897 } 7898 return status; 7899} 7900 7901void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7902{ 7903 for (size_t i = 0; i < mTracks.size(); ++i) { 7904 sp<Track> track = mTracks[i]; 7905 if (track->auxEffectId() == effectId) { 7906 attachAuxEffect_l(track, 0); 7907 } 7908 } 7909} 7910 7911status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7912{ 7913 // only one chain per input thread 7914 if (mEffectChains.size() != 0) { 7915 return INVALID_OPERATION; 7916 } 7917 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7918 7919 chain->setInBuffer(NULL); 7920 chain->setOutBuffer(NULL); 7921 7922 checkSuspendOnAddEffectChain_l(chain); 7923 7924 mEffectChains.add(chain); 7925 7926 return NO_ERROR; 7927} 7928 7929size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7930{ 7931 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7932 ALOGW_IF(mEffectChains.size() != 1, 7933 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7934 chain.get(), mEffectChains.size(), this); 7935 if (mEffectChains.size() == 1) { 7936 mEffectChains.removeAt(0); 7937 } 7938 return 0; 7939} 7940 7941// ---------------------------------------------------------------------------- 7942// EffectModule implementation 7943// ---------------------------------------------------------------------------- 7944 7945#undef LOG_TAG 7946#define LOG_TAG "AudioFlinger::EffectModule" 7947 7948AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7949 const wp<AudioFlinger::EffectChain>& chain, 7950 effect_descriptor_t *desc, 7951 int id, 7952 int sessionId) 7953 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7954 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7955 // mDescriptor is set below 7956 // mConfig is set by configure() and not used before then 7957 mEffectInterface(NULL), 7958 mStatus(NO_INIT), mState(IDLE), 7959 // mMaxDisableWaitCnt is set by configure() and not used before then 7960 // mDisableWaitCnt is set by process() and updateState() and not used before then 7961 mSuspended(false) 7962{ 7963 ALOGV("Constructor %p", this); 7964 int lStatus; 7965 if (thread == NULL) { 7966 return; 7967 } 7968 7969 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7970 7971 // create effect engine from effect factory 7972 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7973 7974 if (mStatus != NO_ERROR) { 7975 return; 7976 } 7977 lStatus = init(); 7978 if (lStatus < 0) { 7979 mStatus = lStatus; 7980 goto Error; 7981 } 7982 7983 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7984 return; 7985Error: 7986 EffectRelease(mEffectInterface); 7987 mEffectInterface = NULL; 7988 ALOGV("Constructor Error %d", mStatus); 7989} 7990 7991AudioFlinger::EffectModule::~EffectModule() 7992{ 7993 ALOGV("Destructor %p", this); 7994 if (mEffectInterface != NULL) { 7995 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7996 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7997 sp<ThreadBase> thread = mThread.promote(); 7998 if (thread != 0) { 7999 audio_stream_t *stream = thread->stream(); 8000 if (stream != NULL) { 8001 stream->remove_audio_effect(stream, mEffectInterface); 8002 } 8003 } 8004 } 8005 // release effect engine 8006 EffectRelease(mEffectInterface); 8007 } 8008} 8009 8010status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8011{ 8012 status_t status; 8013 8014 Mutex::Autolock _l(mLock); 8015 int priority = handle->priority(); 8016 size_t size = mHandles.size(); 8017 EffectHandle *controlHandle = NULL; 8018 size_t i; 8019 for (i = 0; i < size; i++) { 8020 EffectHandle *h = mHandles[i]; 8021 if (h == NULL || h->destroyed_l()) continue; 8022 // first non destroyed handle is considered in control 8023 if (controlHandle == NULL) 8024 controlHandle = h; 8025 if (h->priority() <= priority) break; 8026 } 8027 // if inserted in first place, move effect control from previous owner to this handle 8028 if (i == 0) { 8029 bool enabled = false; 8030 if (controlHandle != NULL) { 8031 enabled = controlHandle->enabled(); 8032 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8033 } 8034 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8035 status = NO_ERROR; 8036 } else { 8037 status = ALREADY_EXISTS; 8038 } 8039 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8040 mHandles.insertAt(handle, i); 8041 return status; 8042} 8043 8044size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8045{ 8046 Mutex::Autolock _l(mLock); 8047 size_t size = mHandles.size(); 8048 size_t i; 8049 for (i = 0; i < size; i++) { 8050 if (mHandles[i] == handle) break; 8051 } 8052 if (i == size) { 8053 return size; 8054 } 8055 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8056 8057 mHandles.removeAt(i); 8058 // if removed from first place, move effect control from this handle to next in line 8059 if (i == 0) { 8060 EffectHandle *h = controlHandle_l(); 8061 if (h != NULL) { 8062 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8063 } 8064 } 8065 8066 // Prevent calls to process() and other functions on effect interface from now on. 8067 // The effect engine will be released by the destructor when the last strong reference on 8068 // this object is released which can happen after next process is called. 8069 if (mHandles.size() == 0 && !mPinned) { 8070 mState = DESTROYED; 8071 } 8072 8073 return size; 8074} 8075 8076// must be called with EffectModule::mLock held 8077AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8078{ 8079 // the first valid handle in the list has control over the module 8080 for (size_t i = 0; i < mHandles.size(); i++) { 8081 EffectHandle *h = mHandles[i]; 8082 if (h != NULL && !h->destroyed_l()) { 8083 return h; 8084 } 8085 } 8086 8087 return NULL; 8088} 8089 8090size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8091{ 8092 ALOGV("disconnect() %p handle %p", this, handle); 8093 // keep a strong reference on this EffectModule to avoid calling the 8094 // destructor before we exit 8095 sp<EffectModule> keep(this); 8096 { 8097 sp<ThreadBase> thread = mThread.promote(); 8098 if (thread != 0) { 8099 thread->disconnectEffect(keep, handle, unpinIfLast); 8100 } 8101 } 8102 return mHandles.size(); 8103} 8104 8105void AudioFlinger::EffectModule::updateState() { 8106 Mutex::Autolock _l(mLock); 8107 8108 switch (mState) { 8109 case RESTART: 8110 reset_l(); 8111 // FALL THROUGH 8112 8113 case STARTING: 8114 // clear auxiliary effect input buffer for next accumulation 8115 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8116 memset(mConfig.inputCfg.buffer.raw, 8117 0, 8118 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8119 } 8120 start_l(); 8121 mState = ACTIVE; 8122 break; 8123 case STOPPING: 8124 stop_l(); 8125 mDisableWaitCnt = mMaxDisableWaitCnt; 8126 mState = STOPPED; 8127 break; 8128 case STOPPED: 8129 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8130 // turn off sequence. 8131 if (--mDisableWaitCnt == 0) { 8132 reset_l(); 8133 mState = IDLE; 8134 } 8135 break; 8136 default: //IDLE , ACTIVE, DESTROYED 8137 break; 8138 } 8139} 8140 8141void AudioFlinger::EffectModule::process() 8142{ 8143 Mutex::Autolock _l(mLock); 8144 8145 if (mState == DESTROYED || mEffectInterface == NULL || 8146 mConfig.inputCfg.buffer.raw == NULL || 8147 mConfig.outputCfg.buffer.raw == NULL) { 8148 return; 8149 } 8150 8151 if (isProcessEnabled()) { 8152 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8153 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8154 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8155 mConfig.inputCfg.buffer.s32, 8156 mConfig.inputCfg.buffer.frameCount/2); 8157 } 8158 8159 // do the actual processing in the effect engine 8160 int ret = (*mEffectInterface)->process(mEffectInterface, 8161 &mConfig.inputCfg.buffer, 8162 &mConfig.outputCfg.buffer); 8163 8164 // force transition to IDLE state when engine is ready 8165 if (mState == STOPPED && ret == -ENODATA) { 8166 mDisableWaitCnt = 1; 8167 } 8168 8169 // clear auxiliary effect input buffer for next accumulation 8170 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8171 memset(mConfig.inputCfg.buffer.raw, 0, 8172 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8173 } 8174 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8175 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8176 // If an insert effect is idle and input buffer is different from output buffer, 8177 // accumulate input onto output 8178 sp<EffectChain> chain = mChain.promote(); 8179 if (chain != 0 && chain->activeTrackCnt() != 0) { 8180 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8181 int16_t *in = mConfig.inputCfg.buffer.s16; 8182 int16_t *out = mConfig.outputCfg.buffer.s16; 8183 for (size_t i = 0; i < frameCnt; i++) { 8184 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8185 } 8186 } 8187 } 8188} 8189 8190void AudioFlinger::EffectModule::reset_l() 8191{ 8192 if (mEffectInterface == NULL) { 8193 return; 8194 } 8195 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8196} 8197 8198status_t AudioFlinger::EffectModule::configure() 8199{ 8200 uint32_t channels; 8201 if (mEffectInterface == NULL) { 8202 return NO_INIT; 8203 } 8204 8205 sp<ThreadBase> thread = mThread.promote(); 8206 if (thread == 0) { 8207 return DEAD_OBJECT; 8208 } 8209 8210 // TODO: handle configuration of effects replacing track process 8211 if (thread->channelCount() == 1) { 8212 channels = AUDIO_CHANNEL_OUT_MONO; 8213 } else { 8214 channels = AUDIO_CHANNEL_OUT_STEREO; 8215 } 8216 8217 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8218 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8219 } else { 8220 mConfig.inputCfg.channels = channels; 8221 } 8222 mConfig.outputCfg.channels = channels; 8223 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8224 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8225 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8226 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8227 mConfig.inputCfg.bufferProvider.cookie = NULL; 8228 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8229 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8230 mConfig.outputCfg.bufferProvider.cookie = NULL; 8231 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8232 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8233 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8234 // Insert effect: 8235 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8236 // always overwrites output buffer: input buffer == output buffer 8237 // - in other sessions: 8238 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8239 // other effect: overwrites output buffer: input buffer == output buffer 8240 // Auxiliary effect: 8241 // accumulates in output buffer: input buffer != output buffer 8242 // Therefore: accumulate <=> input buffer != output buffer 8243 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8244 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8245 } else { 8246 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8247 } 8248 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8249 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8250 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8251 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8252 8253 ALOGV("configure() %p thread %p buffer %p framecount %d", 8254 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8255 8256 status_t cmdStatus; 8257 uint32_t size = sizeof(int); 8258 status_t status = (*mEffectInterface)->command(mEffectInterface, 8259 EFFECT_CMD_SET_CONFIG, 8260 sizeof(effect_config_t), 8261 &mConfig, 8262 &size, 8263 &cmdStatus); 8264 if (status == 0) { 8265 status = cmdStatus; 8266 } 8267 8268 if (status == 0 && 8269 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8270 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8271 effect_param_t *p = (effect_param_t *)buf32; 8272 8273 p->psize = sizeof(uint32_t); 8274 p->vsize = sizeof(uint32_t); 8275 size = sizeof(int); 8276 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8277 8278 uint32_t latency = 0; 8279 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8280 if (pbt != NULL) { 8281 latency = pbt->latency_l(); 8282 } 8283 8284 *((int32_t *)p->data + 1)= latency; 8285 (*mEffectInterface)->command(mEffectInterface, 8286 EFFECT_CMD_SET_PARAM, 8287 sizeof(effect_param_t) + 8, 8288 &buf32, 8289 &size, 8290 &cmdStatus); 8291 } 8292 8293 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8294 (1000 * mConfig.outputCfg.buffer.frameCount); 8295 8296 return status; 8297} 8298 8299status_t AudioFlinger::EffectModule::init() 8300{ 8301 Mutex::Autolock _l(mLock); 8302 if (mEffectInterface == NULL) { 8303 return NO_INIT; 8304 } 8305 status_t cmdStatus; 8306 uint32_t size = sizeof(status_t); 8307 status_t status = (*mEffectInterface)->command(mEffectInterface, 8308 EFFECT_CMD_INIT, 8309 0, 8310 NULL, 8311 &size, 8312 &cmdStatus); 8313 if (status == 0) { 8314 status = cmdStatus; 8315 } 8316 return status; 8317} 8318 8319status_t AudioFlinger::EffectModule::start() 8320{ 8321 Mutex::Autolock _l(mLock); 8322 return start_l(); 8323} 8324 8325status_t AudioFlinger::EffectModule::start_l() 8326{ 8327 if (mEffectInterface == NULL) { 8328 return NO_INIT; 8329 } 8330 status_t cmdStatus; 8331 uint32_t size = sizeof(status_t); 8332 status_t status = (*mEffectInterface)->command(mEffectInterface, 8333 EFFECT_CMD_ENABLE, 8334 0, 8335 NULL, 8336 &size, 8337 &cmdStatus); 8338 if (status == 0) { 8339 status = cmdStatus; 8340 } 8341 if (status == 0 && 8342 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8343 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8344 sp<ThreadBase> thread = mThread.promote(); 8345 if (thread != 0) { 8346 audio_stream_t *stream = thread->stream(); 8347 if (stream != NULL) { 8348 stream->add_audio_effect(stream, mEffectInterface); 8349 } 8350 } 8351 } 8352 return status; 8353} 8354 8355status_t AudioFlinger::EffectModule::stop() 8356{ 8357 Mutex::Autolock _l(mLock); 8358 return stop_l(); 8359} 8360 8361status_t AudioFlinger::EffectModule::stop_l() 8362{ 8363 if (mEffectInterface == NULL) { 8364 return NO_INIT; 8365 } 8366 status_t cmdStatus; 8367 uint32_t size = sizeof(status_t); 8368 status_t status = (*mEffectInterface)->command(mEffectInterface, 8369 EFFECT_CMD_DISABLE, 8370 0, 8371 NULL, 8372 &size, 8373 &cmdStatus); 8374 if (status == 0) { 8375 status = cmdStatus; 8376 } 8377 if (status == 0 && 8378 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8379 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8380 sp<ThreadBase> thread = mThread.promote(); 8381 if (thread != 0) { 8382 audio_stream_t *stream = thread->stream(); 8383 if (stream != NULL) { 8384 stream->remove_audio_effect(stream, mEffectInterface); 8385 } 8386 } 8387 } 8388 return status; 8389} 8390 8391status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8392 uint32_t cmdSize, 8393 void *pCmdData, 8394 uint32_t *replySize, 8395 void *pReplyData) 8396{ 8397 Mutex::Autolock _l(mLock); 8398// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8399 8400 if (mState == DESTROYED || mEffectInterface == NULL) { 8401 return NO_INIT; 8402 } 8403 status_t status = (*mEffectInterface)->command(mEffectInterface, 8404 cmdCode, 8405 cmdSize, 8406 pCmdData, 8407 replySize, 8408 pReplyData); 8409 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8410 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8411 for (size_t i = 1; i < mHandles.size(); i++) { 8412 EffectHandle *h = mHandles[i]; 8413 if (h != NULL && !h->destroyed_l()) { 8414 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8415 } 8416 } 8417 } 8418 return status; 8419} 8420 8421status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8422{ 8423 Mutex::Autolock _l(mLock); 8424 return setEnabled_l(enabled); 8425} 8426 8427// must be called with EffectModule::mLock held 8428status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8429{ 8430 8431 ALOGV("setEnabled %p enabled %d", this, enabled); 8432 8433 if (enabled != isEnabled()) { 8434 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8435 if (enabled && status != NO_ERROR) { 8436 return status; 8437 } 8438 8439 switch (mState) { 8440 // going from disabled to enabled 8441 case IDLE: 8442 mState = STARTING; 8443 break; 8444 case STOPPED: 8445 mState = RESTART; 8446 break; 8447 case STOPPING: 8448 mState = ACTIVE; 8449 break; 8450 8451 // going from enabled to disabled 8452 case RESTART: 8453 mState = STOPPED; 8454 break; 8455 case STARTING: 8456 mState = IDLE; 8457 break; 8458 case ACTIVE: 8459 mState = STOPPING; 8460 break; 8461 case DESTROYED: 8462 return NO_ERROR; // simply ignore as we are being destroyed 8463 } 8464 for (size_t i = 1; i < mHandles.size(); i++) { 8465 EffectHandle *h = mHandles[i]; 8466 if (h != NULL && !h->destroyed_l()) { 8467 h->setEnabled(enabled); 8468 } 8469 } 8470 } 8471 return NO_ERROR; 8472} 8473 8474bool AudioFlinger::EffectModule::isEnabled() const 8475{ 8476 switch (mState) { 8477 case RESTART: 8478 case STARTING: 8479 case ACTIVE: 8480 return true; 8481 case IDLE: 8482 case STOPPING: 8483 case STOPPED: 8484 case DESTROYED: 8485 default: 8486 return false; 8487 } 8488} 8489 8490bool AudioFlinger::EffectModule::isProcessEnabled() const 8491{ 8492 switch (mState) { 8493 case RESTART: 8494 case ACTIVE: 8495 case STOPPING: 8496 case STOPPED: 8497 return true; 8498 case IDLE: 8499 case STARTING: 8500 case DESTROYED: 8501 default: 8502 return false; 8503 } 8504} 8505 8506status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8507{ 8508 Mutex::Autolock _l(mLock); 8509 status_t status = NO_ERROR; 8510 8511 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8512 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8513 if (isProcessEnabled() && 8514 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8515 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8516 status_t cmdStatus; 8517 uint32_t volume[2]; 8518 uint32_t *pVolume = NULL; 8519 uint32_t size = sizeof(volume); 8520 volume[0] = *left; 8521 volume[1] = *right; 8522 if (controller) { 8523 pVolume = volume; 8524 } 8525 status = (*mEffectInterface)->command(mEffectInterface, 8526 EFFECT_CMD_SET_VOLUME, 8527 size, 8528 volume, 8529 &size, 8530 pVolume); 8531 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8532 *left = volume[0]; 8533 *right = volume[1]; 8534 } 8535 } 8536 return status; 8537} 8538 8539status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8540{ 8541 Mutex::Autolock _l(mLock); 8542 status_t status = NO_ERROR; 8543 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8544 // audio pre processing modules on RecordThread can receive both output and 8545 // input device indication in the same call 8546 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8547 if (dev) { 8548 status_t cmdStatus; 8549 uint32_t size = sizeof(status_t); 8550 8551 status = (*mEffectInterface)->command(mEffectInterface, 8552 EFFECT_CMD_SET_DEVICE, 8553 sizeof(uint32_t), 8554 &dev, 8555 &size, 8556 &cmdStatus); 8557 if (status == NO_ERROR) { 8558 status = cmdStatus; 8559 } 8560 } 8561 dev = device & AUDIO_DEVICE_IN_ALL; 8562 if (dev) { 8563 status_t cmdStatus; 8564 uint32_t size = sizeof(status_t); 8565 8566 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8567 EFFECT_CMD_SET_INPUT_DEVICE, 8568 sizeof(uint32_t), 8569 &dev, 8570 &size, 8571 &cmdStatus); 8572 if (status2 == NO_ERROR) { 8573 status2 = cmdStatus; 8574 } 8575 if (status == NO_ERROR) { 8576 status = status2; 8577 } 8578 } 8579 } 8580 return status; 8581} 8582 8583status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8584{ 8585 Mutex::Autolock _l(mLock); 8586 status_t status = NO_ERROR; 8587 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8588 status_t cmdStatus; 8589 uint32_t size = sizeof(status_t); 8590 status = (*mEffectInterface)->command(mEffectInterface, 8591 EFFECT_CMD_SET_AUDIO_MODE, 8592 sizeof(audio_mode_t), 8593 &mode, 8594 &size, 8595 &cmdStatus); 8596 if (status == NO_ERROR) { 8597 status = cmdStatus; 8598 } 8599 } 8600 return status; 8601} 8602 8603void AudioFlinger::EffectModule::setSuspended(bool suspended) 8604{ 8605 Mutex::Autolock _l(mLock); 8606 mSuspended = suspended; 8607} 8608 8609bool AudioFlinger::EffectModule::suspended() const 8610{ 8611 Mutex::Autolock _l(mLock); 8612 return mSuspended; 8613} 8614 8615bool AudioFlinger::EffectModule::purgeHandles() 8616{ 8617 bool enabled = false; 8618 Mutex::Autolock _l(mLock); 8619 for (size_t i = 0; i < mHandles.size(); i++) { 8620 EffectHandle *handle = mHandles[i]; 8621 if (handle != NULL && !handle->destroyed_l()) { 8622 handle->effect().clear(); 8623 if (handle->hasControl()) { 8624 enabled = handle->enabled(); 8625 } 8626 } 8627 } 8628 return enabled; 8629} 8630 8631status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8632{ 8633 const size_t SIZE = 256; 8634 char buffer[SIZE]; 8635 String8 result; 8636 8637 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8638 result.append(buffer); 8639 8640 bool locked = tryLock(mLock); 8641 // failed to lock - AudioFlinger is probably deadlocked 8642 if (!locked) { 8643 result.append("\t\tCould not lock Fx mutex:\n"); 8644 } 8645 8646 result.append("\t\tSession Status State Engine:\n"); 8647 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8648 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8649 result.append(buffer); 8650 8651 result.append("\t\tDescriptor:\n"); 8652 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8653 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8654 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8655 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8656 result.append(buffer); 8657 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8658 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8659 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8660 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8661 result.append(buffer); 8662 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8663 mDescriptor.apiVersion, 8664 mDescriptor.flags); 8665 result.append(buffer); 8666 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8667 mDescriptor.name); 8668 result.append(buffer); 8669 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8670 mDescriptor.implementor); 8671 result.append(buffer); 8672 8673 result.append("\t\t- Input configuration:\n"); 8674 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8675 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8676 (uint32_t)mConfig.inputCfg.buffer.raw, 8677 mConfig.inputCfg.buffer.frameCount, 8678 mConfig.inputCfg.samplingRate, 8679 mConfig.inputCfg.channels, 8680 mConfig.inputCfg.format); 8681 result.append(buffer); 8682 8683 result.append("\t\t- Output configuration:\n"); 8684 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8685 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8686 (uint32_t)mConfig.outputCfg.buffer.raw, 8687 mConfig.outputCfg.buffer.frameCount, 8688 mConfig.outputCfg.samplingRate, 8689 mConfig.outputCfg.channels, 8690 mConfig.outputCfg.format); 8691 result.append(buffer); 8692 8693 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8694 result.append(buffer); 8695 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8696 for (size_t i = 0; i < mHandles.size(); ++i) { 8697 EffectHandle *handle = mHandles[i]; 8698 if (handle != NULL && !handle->destroyed_l()) { 8699 handle->dump(buffer, SIZE); 8700 result.append(buffer); 8701 } 8702 } 8703 8704 result.append("\n"); 8705 8706 write(fd, result.string(), result.length()); 8707 8708 if (locked) { 8709 mLock.unlock(); 8710 } 8711 8712 return NO_ERROR; 8713} 8714 8715// ---------------------------------------------------------------------------- 8716// EffectHandle implementation 8717// ---------------------------------------------------------------------------- 8718 8719#undef LOG_TAG 8720#define LOG_TAG "AudioFlinger::EffectHandle" 8721 8722AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8723 const sp<AudioFlinger::Client>& client, 8724 const sp<IEffectClient>& effectClient, 8725 int32_t priority) 8726 : BnEffect(), 8727 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8728 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8729{ 8730 ALOGV("constructor %p", this); 8731 8732 if (client == 0) { 8733 return; 8734 } 8735 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8736 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8737 if (mCblkMemory != 0) { 8738 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8739 8740 if (mCblk != NULL) { 8741 new(mCblk) effect_param_cblk_t(); 8742 mBuffer = (uint8_t *)mCblk + bufOffset; 8743 } 8744 } else { 8745 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8746 return; 8747 } 8748} 8749 8750AudioFlinger::EffectHandle::~EffectHandle() 8751{ 8752 ALOGV("Destructor %p", this); 8753 8754 if (mEffect == 0) { 8755 mDestroyed = true; 8756 return; 8757 } 8758 mEffect->lock(); 8759 mDestroyed = true; 8760 mEffect->unlock(); 8761 disconnect(false); 8762} 8763 8764status_t AudioFlinger::EffectHandle::enable() 8765{ 8766 ALOGV("enable %p", this); 8767 if (!mHasControl) return INVALID_OPERATION; 8768 if (mEffect == 0) return DEAD_OBJECT; 8769 8770 if (mEnabled) { 8771 return NO_ERROR; 8772 } 8773 8774 mEnabled = true; 8775 8776 sp<ThreadBase> thread = mEffect->thread().promote(); 8777 if (thread != 0) { 8778 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8779 } 8780 8781 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8782 if (mEffect->suspended()) { 8783 return NO_ERROR; 8784 } 8785 8786 status_t status = mEffect->setEnabled(true); 8787 if (status != NO_ERROR) { 8788 if (thread != 0) { 8789 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8790 } 8791 mEnabled = false; 8792 } 8793 return status; 8794} 8795 8796status_t AudioFlinger::EffectHandle::disable() 8797{ 8798 ALOGV("disable %p", this); 8799 if (!mHasControl) return INVALID_OPERATION; 8800 if (mEffect == 0) return DEAD_OBJECT; 8801 8802 if (!mEnabled) { 8803 return NO_ERROR; 8804 } 8805 mEnabled = false; 8806 8807 if (mEffect->suspended()) { 8808 return NO_ERROR; 8809 } 8810 8811 status_t status = mEffect->setEnabled(false); 8812 8813 sp<ThreadBase> thread = mEffect->thread().promote(); 8814 if (thread != 0) { 8815 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8816 } 8817 8818 return status; 8819} 8820 8821void AudioFlinger::EffectHandle::disconnect() 8822{ 8823 disconnect(true); 8824} 8825 8826void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8827{ 8828 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8829 if (mEffect == 0) { 8830 return; 8831 } 8832 // restore suspended effects if the disconnected handle was enabled and the last one. 8833 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8834 sp<ThreadBase> thread = mEffect->thread().promote(); 8835 if (thread != 0) { 8836 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8837 } 8838 } 8839 8840 // release sp on module => module destructor can be called now 8841 mEffect.clear(); 8842 if (mClient != 0) { 8843 if (mCblk != NULL) { 8844 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8845 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8846 } 8847 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8848 // Client destructor must run with AudioFlinger mutex locked 8849 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8850 mClient.clear(); 8851 } 8852} 8853 8854status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8855 uint32_t cmdSize, 8856 void *pCmdData, 8857 uint32_t *replySize, 8858 void *pReplyData) 8859{ 8860// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8861// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8862 8863 // only get parameter command is permitted for applications not controlling the effect 8864 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8865 return INVALID_OPERATION; 8866 } 8867 if (mEffect == 0) return DEAD_OBJECT; 8868 if (mClient == 0) return INVALID_OPERATION; 8869 8870 // handle commands that are not forwarded transparently to effect engine 8871 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8872 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8873 // no risk to block the whole media server process or mixer threads is we are stuck here 8874 Mutex::Autolock _l(mCblk->lock); 8875 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8876 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8877 mCblk->serverIndex = 0; 8878 mCblk->clientIndex = 0; 8879 return BAD_VALUE; 8880 } 8881 status_t status = NO_ERROR; 8882 while (mCblk->serverIndex < mCblk->clientIndex) { 8883 int reply; 8884 uint32_t rsize = sizeof(int); 8885 int *p = (int *)(mBuffer + mCblk->serverIndex); 8886 int size = *p++; 8887 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8888 ALOGW("command(): invalid parameter block size"); 8889 break; 8890 } 8891 effect_param_t *param = (effect_param_t *)p; 8892 if (param->psize == 0 || param->vsize == 0) { 8893 ALOGW("command(): null parameter or value size"); 8894 mCblk->serverIndex += size; 8895 continue; 8896 } 8897 uint32_t psize = sizeof(effect_param_t) + 8898 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8899 param->vsize; 8900 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8901 psize, 8902 p, 8903 &rsize, 8904 &reply); 8905 // stop at first error encountered 8906 if (ret != NO_ERROR) { 8907 status = ret; 8908 *(int *)pReplyData = reply; 8909 break; 8910 } else if (reply != NO_ERROR) { 8911 *(int *)pReplyData = reply; 8912 break; 8913 } 8914 mCblk->serverIndex += size; 8915 } 8916 mCblk->serverIndex = 0; 8917 mCblk->clientIndex = 0; 8918 return status; 8919 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8920 *(int *)pReplyData = NO_ERROR; 8921 return enable(); 8922 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8923 *(int *)pReplyData = NO_ERROR; 8924 return disable(); 8925 } 8926 8927 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8928} 8929 8930void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8931{ 8932 ALOGV("setControl %p control %d", this, hasControl); 8933 8934 mHasControl = hasControl; 8935 mEnabled = enabled; 8936 8937 if (signal && mEffectClient != 0) { 8938 mEffectClient->controlStatusChanged(hasControl); 8939 } 8940} 8941 8942void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8943 uint32_t cmdSize, 8944 void *pCmdData, 8945 uint32_t replySize, 8946 void *pReplyData) 8947{ 8948 if (mEffectClient != 0) { 8949 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8950 } 8951} 8952 8953 8954 8955void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8956{ 8957 if (mEffectClient != 0) { 8958 mEffectClient->enableStatusChanged(enabled); 8959 } 8960} 8961 8962status_t AudioFlinger::EffectHandle::onTransact( 8963 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8964{ 8965 return BnEffect::onTransact(code, data, reply, flags); 8966} 8967 8968 8969void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8970{ 8971 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8972 8973 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8974 (mClient == 0) ? getpid_cached : mClient->pid(), 8975 mPriority, 8976 mHasControl, 8977 !locked, 8978 mCblk ? mCblk->clientIndex : 0, 8979 mCblk ? mCblk->serverIndex : 0 8980 ); 8981 8982 if (locked) { 8983 mCblk->lock.unlock(); 8984 } 8985} 8986 8987#undef LOG_TAG 8988#define LOG_TAG "AudioFlinger::EffectChain" 8989 8990AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8991 int sessionId) 8992 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8993 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8994 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8995{ 8996 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8997 if (thread == NULL) { 8998 return; 8999 } 9000 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9001 thread->frameCount(); 9002} 9003 9004AudioFlinger::EffectChain::~EffectChain() 9005{ 9006 if (mOwnInBuffer) { 9007 delete mInBuffer; 9008 } 9009 9010} 9011 9012// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9013sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9014{ 9015 size_t size = mEffects.size(); 9016 9017 for (size_t i = 0; i < size; i++) { 9018 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9019 return mEffects[i]; 9020 } 9021 } 9022 return 0; 9023} 9024 9025// getEffectFromId_l() must be called with ThreadBase::mLock held 9026sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9027{ 9028 size_t size = mEffects.size(); 9029 9030 for (size_t i = 0; i < size; i++) { 9031 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9032 if (id == 0 || mEffects[i]->id() == id) { 9033 return mEffects[i]; 9034 } 9035 } 9036 return 0; 9037} 9038 9039// getEffectFromType_l() must be called with ThreadBase::mLock held 9040sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9041 const effect_uuid_t *type) 9042{ 9043 size_t size = mEffects.size(); 9044 9045 for (size_t i = 0; i < size; i++) { 9046 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9047 return mEffects[i]; 9048 } 9049 } 9050 return 0; 9051} 9052 9053void AudioFlinger::EffectChain::clearInputBuffer() 9054{ 9055 Mutex::Autolock _l(mLock); 9056 sp<ThreadBase> thread = mThread.promote(); 9057 if (thread == 0) { 9058 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9059 return; 9060 } 9061 clearInputBuffer_l(thread); 9062} 9063 9064// Must be called with EffectChain::mLock locked 9065void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9066{ 9067 size_t numSamples = thread->frameCount() * thread->channelCount(); 9068 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9069 9070} 9071 9072// Must be called with EffectChain::mLock locked 9073void AudioFlinger::EffectChain::process_l() 9074{ 9075 sp<ThreadBase> thread = mThread.promote(); 9076 if (thread == 0) { 9077 ALOGW("process_l(): cannot promote mixer thread"); 9078 return; 9079 } 9080 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9081 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9082 // always process effects unless no more tracks are on the session and the effect tail 9083 // has been rendered 9084 bool doProcess = true; 9085 if (!isGlobalSession) { 9086 bool tracksOnSession = (trackCnt() != 0); 9087 9088 if (!tracksOnSession && mTailBufferCount == 0) { 9089 doProcess = false; 9090 } 9091 9092 if (activeTrackCnt() == 0) { 9093 // if no track is active and the effect tail has not been rendered, 9094 // the input buffer must be cleared here as the mixer process will not do it 9095 if (tracksOnSession || mTailBufferCount > 0) { 9096 clearInputBuffer_l(thread); 9097 if (mTailBufferCount > 0) { 9098 mTailBufferCount--; 9099 } 9100 } 9101 } 9102 } 9103 9104 size_t size = mEffects.size(); 9105 if (doProcess) { 9106 for (size_t i = 0; i < size; i++) { 9107 mEffects[i]->process(); 9108 } 9109 } 9110 for (size_t i = 0; i < size; i++) { 9111 mEffects[i]->updateState(); 9112 } 9113} 9114 9115// addEffect_l() must be called with PlaybackThread::mLock held 9116status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9117{ 9118 effect_descriptor_t desc = effect->desc(); 9119 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9120 9121 Mutex::Autolock _l(mLock); 9122 effect->setChain(this); 9123 sp<ThreadBase> thread = mThread.promote(); 9124 if (thread == 0) { 9125 return NO_INIT; 9126 } 9127 effect->setThread(thread); 9128 9129 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9130 // Auxiliary effects are inserted at the beginning of mEffects vector as 9131 // they are processed first and accumulated in chain input buffer 9132 mEffects.insertAt(effect, 0); 9133 9134 // the input buffer for auxiliary effect contains mono samples in 9135 // 32 bit format. This is to avoid saturation in AudoMixer 9136 // accumulation stage. Saturation is done in EffectModule::process() before 9137 // calling the process in effect engine 9138 size_t numSamples = thread->frameCount(); 9139 int32_t *buffer = new int32_t[numSamples]; 9140 memset(buffer, 0, numSamples * sizeof(int32_t)); 9141 effect->setInBuffer((int16_t *)buffer); 9142 // auxiliary effects output samples to chain input buffer for further processing 9143 // by insert effects 9144 effect->setOutBuffer(mInBuffer); 9145 } else { 9146 // Insert effects are inserted at the end of mEffects vector as they are processed 9147 // after track and auxiliary effects. 9148 // Insert effect order as a function of indicated preference: 9149 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9150 // another effect is present 9151 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9152 // last effect claiming first position 9153 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9154 // first effect claiming last position 9155 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9156 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9157 // already present 9158 9159 size_t size = mEffects.size(); 9160 size_t idx_insert = size; 9161 ssize_t idx_insert_first = -1; 9162 ssize_t idx_insert_last = -1; 9163 9164 for (size_t i = 0; i < size; i++) { 9165 effect_descriptor_t d = mEffects[i]->desc(); 9166 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9167 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9168 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9169 // check invalid effect chaining combinations 9170 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9171 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9172 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9173 return INVALID_OPERATION; 9174 } 9175 // remember position of first insert effect and by default 9176 // select this as insert position for new effect 9177 if (idx_insert == size) { 9178 idx_insert = i; 9179 } 9180 // remember position of last insert effect claiming 9181 // first position 9182 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9183 idx_insert_first = i; 9184 } 9185 // remember position of first insert effect claiming 9186 // last position 9187 if (iPref == EFFECT_FLAG_INSERT_LAST && 9188 idx_insert_last == -1) { 9189 idx_insert_last = i; 9190 } 9191 } 9192 } 9193 9194 // modify idx_insert from first position if needed 9195 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9196 if (idx_insert_last != -1) { 9197 idx_insert = idx_insert_last; 9198 } else { 9199 idx_insert = size; 9200 } 9201 } else { 9202 if (idx_insert_first != -1) { 9203 idx_insert = idx_insert_first + 1; 9204 } 9205 } 9206 9207 // always read samples from chain input buffer 9208 effect->setInBuffer(mInBuffer); 9209 9210 // if last effect in the chain, output samples to chain 9211 // output buffer, otherwise to chain input buffer 9212 if (idx_insert == size) { 9213 if (idx_insert != 0) { 9214 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9215 mEffects[idx_insert-1]->configure(); 9216 } 9217 effect->setOutBuffer(mOutBuffer); 9218 } else { 9219 effect->setOutBuffer(mInBuffer); 9220 } 9221 mEffects.insertAt(effect, idx_insert); 9222 9223 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9224 } 9225 effect->configure(); 9226 return NO_ERROR; 9227} 9228 9229// removeEffect_l() must be called with PlaybackThread::mLock held 9230size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9231{ 9232 Mutex::Autolock _l(mLock); 9233 size_t size = mEffects.size(); 9234 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9235 9236 for (size_t i = 0; i < size; i++) { 9237 if (effect == mEffects[i]) { 9238 // calling stop here will remove pre-processing effect from the audio HAL. 9239 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9240 // the middle of a read from audio HAL 9241 if (mEffects[i]->state() == EffectModule::ACTIVE || 9242 mEffects[i]->state() == EffectModule::STOPPING) { 9243 mEffects[i]->stop(); 9244 } 9245 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9246 delete[] effect->inBuffer(); 9247 } else { 9248 if (i == size - 1 && i != 0) { 9249 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9250 mEffects[i - 1]->configure(); 9251 } 9252 } 9253 mEffects.removeAt(i); 9254 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9255 break; 9256 } 9257 } 9258 9259 return mEffects.size(); 9260} 9261 9262// setDevice_l() must be called with PlaybackThread::mLock held 9263void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9264{ 9265 size_t size = mEffects.size(); 9266 for (size_t i = 0; i < size; i++) { 9267 mEffects[i]->setDevice(device); 9268 } 9269} 9270 9271// setMode_l() must be called with PlaybackThread::mLock held 9272void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9273{ 9274 size_t size = mEffects.size(); 9275 for (size_t i = 0; i < size; i++) { 9276 mEffects[i]->setMode(mode); 9277 } 9278} 9279 9280// setVolume_l() must be called with PlaybackThread::mLock held 9281bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9282{ 9283 uint32_t newLeft = *left; 9284 uint32_t newRight = *right; 9285 bool hasControl = false; 9286 int ctrlIdx = -1; 9287 size_t size = mEffects.size(); 9288 9289 // first update volume controller 9290 for (size_t i = size; i > 0; i--) { 9291 if (mEffects[i - 1]->isProcessEnabled() && 9292 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9293 ctrlIdx = i - 1; 9294 hasControl = true; 9295 break; 9296 } 9297 } 9298 9299 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9300 if (hasControl) { 9301 *left = mNewLeftVolume; 9302 *right = mNewRightVolume; 9303 } 9304 return hasControl; 9305 } 9306 9307 mVolumeCtrlIdx = ctrlIdx; 9308 mLeftVolume = newLeft; 9309 mRightVolume = newRight; 9310 9311 // second get volume update from volume controller 9312 if (ctrlIdx >= 0) { 9313 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9314 mNewLeftVolume = newLeft; 9315 mNewRightVolume = newRight; 9316 } 9317 // then indicate volume to all other effects in chain. 9318 // Pass altered volume to effects before volume controller 9319 // and requested volume to effects after controller 9320 uint32_t lVol = newLeft; 9321 uint32_t rVol = newRight; 9322 9323 for (size_t i = 0; i < size; i++) { 9324 if ((int)i == ctrlIdx) continue; 9325 // this also works for ctrlIdx == -1 when there is no volume controller 9326 if ((int)i > ctrlIdx) { 9327 lVol = *left; 9328 rVol = *right; 9329 } 9330 mEffects[i]->setVolume(&lVol, &rVol, false); 9331 } 9332 *left = newLeft; 9333 *right = newRight; 9334 9335 return hasControl; 9336} 9337 9338status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9339{ 9340 const size_t SIZE = 256; 9341 char buffer[SIZE]; 9342 String8 result; 9343 9344 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9345 result.append(buffer); 9346 9347 bool locked = tryLock(mLock); 9348 // failed to lock - AudioFlinger is probably deadlocked 9349 if (!locked) { 9350 result.append("\tCould not lock mutex:\n"); 9351 } 9352 9353 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9354 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9355 mEffects.size(), 9356 (uint32_t)mInBuffer, 9357 (uint32_t)mOutBuffer, 9358 mActiveTrackCnt); 9359 result.append(buffer); 9360 write(fd, result.string(), result.size()); 9361 9362 for (size_t i = 0; i < mEffects.size(); ++i) { 9363 sp<EffectModule> effect = mEffects[i]; 9364 if (effect != 0) { 9365 effect->dump(fd, args); 9366 } 9367 } 9368 9369 if (locked) { 9370 mLock.unlock(); 9371 } 9372 9373 return NO_ERROR; 9374} 9375 9376// must be called with ThreadBase::mLock held 9377void AudioFlinger::EffectChain::setEffectSuspended_l( 9378 const effect_uuid_t *type, bool suspend) 9379{ 9380 sp<SuspendedEffectDesc> desc; 9381 // use effect type UUID timelow as key as there is no real risk of identical 9382 // timeLow fields among effect type UUIDs. 9383 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9384 if (suspend) { 9385 if (index >= 0) { 9386 desc = mSuspendedEffects.valueAt(index); 9387 } else { 9388 desc = new SuspendedEffectDesc(); 9389 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9390 mSuspendedEffects.add(type->timeLow, desc); 9391 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9392 } 9393 if (desc->mRefCount++ == 0) { 9394 sp<EffectModule> effect = getEffectIfEnabled(type); 9395 if (effect != 0) { 9396 desc->mEffect = effect; 9397 effect->setSuspended(true); 9398 effect->setEnabled(false); 9399 } 9400 } 9401 } else { 9402 if (index < 0) { 9403 return; 9404 } 9405 desc = mSuspendedEffects.valueAt(index); 9406 if (desc->mRefCount <= 0) { 9407 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9408 desc->mRefCount = 1; 9409 } 9410 if (--desc->mRefCount == 0) { 9411 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9412 if (desc->mEffect != 0) { 9413 sp<EffectModule> effect = desc->mEffect.promote(); 9414 if (effect != 0) { 9415 effect->setSuspended(false); 9416 effect->lock(); 9417 EffectHandle *handle = effect->controlHandle_l(); 9418 if (handle != NULL && !handle->destroyed_l()) { 9419 effect->setEnabled_l(handle->enabled()); 9420 } 9421 effect->unlock(); 9422 } 9423 desc->mEffect.clear(); 9424 } 9425 mSuspendedEffects.removeItemsAt(index); 9426 } 9427 } 9428} 9429 9430// must be called with ThreadBase::mLock held 9431void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9432{ 9433 sp<SuspendedEffectDesc> desc; 9434 9435 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9436 if (suspend) { 9437 if (index >= 0) { 9438 desc = mSuspendedEffects.valueAt(index); 9439 } else { 9440 desc = new SuspendedEffectDesc(); 9441 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9442 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9443 } 9444 if (desc->mRefCount++ == 0) { 9445 Vector< sp<EffectModule> > effects; 9446 getSuspendEligibleEffects(effects); 9447 for (size_t i = 0; i < effects.size(); i++) { 9448 setEffectSuspended_l(&effects[i]->desc().type, true); 9449 } 9450 } 9451 } else { 9452 if (index < 0) { 9453 return; 9454 } 9455 desc = mSuspendedEffects.valueAt(index); 9456 if (desc->mRefCount <= 0) { 9457 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9458 desc->mRefCount = 1; 9459 } 9460 if (--desc->mRefCount == 0) { 9461 Vector<const effect_uuid_t *> types; 9462 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9463 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9464 continue; 9465 } 9466 types.add(&mSuspendedEffects.valueAt(i)->mType); 9467 } 9468 for (size_t i = 0; i < types.size(); i++) { 9469 setEffectSuspended_l(types[i], false); 9470 } 9471 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9472 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9473 } 9474 } 9475} 9476 9477 9478// The volume effect is used for automated tests only 9479#ifndef OPENSL_ES_H_ 9480static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9481 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9482const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9483#endif //OPENSL_ES_H_ 9484 9485bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9486{ 9487 // auxiliary effects and visualizer are never suspended on output mix 9488 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9489 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9490 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9491 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9492 return false; 9493 } 9494 return true; 9495} 9496 9497void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9498{ 9499 effects.clear(); 9500 for (size_t i = 0; i < mEffects.size(); i++) { 9501 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9502 effects.add(mEffects[i]); 9503 } 9504 } 9505} 9506 9507sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9508 const effect_uuid_t *type) 9509{ 9510 sp<EffectModule> effect = getEffectFromType_l(type); 9511 return effect != 0 && effect->isEnabled() ? effect : 0; 9512} 9513 9514void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9515 bool enabled) 9516{ 9517 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9518 if (enabled) { 9519 if (index < 0) { 9520 // if the effect is not suspend check if all effects are suspended 9521 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9522 if (index < 0) { 9523 return; 9524 } 9525 if (!isEffectEligibleForSuspend(effect->desc())) { 9526 return; 9527 } 9528 setEffectSuspended_l(&effect->desc().type, enabled); 9529 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9530 if (index < 0) { 9531 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9532 return; 9533 } 9534 } 9535 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9536 effect->desc().type.timeLow); 9537 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9538 // if effect is requested to suspended but was not yet enabled, supend it now. 9539 if (desc->mEffect == 0) { 9540 desc->mEffect = effect; 9541 effect->setEnabled(false); 9542 effect->setSuspended(true); 9543 } 9544 } else { 9545 if (index < 0) { 9546 return; 9547 } 9548 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9549 effect->desc().type.timeLow); 9550 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9551 desc->mEffect.clear(); 9552 effect->setSuspended(false); 9553 } 9554} 9555 9556#undef LOG_TAG 9557#define LOG_TAG "AudioFlinger" 9558 9559// ---------------------------------------------------------------------------- 9560 9561status_t AudioFlinger::onTransact( 9562 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9563{ 9564 return BnAudioFlinger::onTransact(code, data, reply, flags); 9565} 9566 9567}; // namespace android 9568