AudioFlinger.cpp revision 1ea6d23396118a9cfe912b7b8a4e6f231e318ea2
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// ---------------------------------------------------------------------------- 165 166#ifdef ADD_BATTERY_DATA 167// To collect the amplifier usage 168static void addBatteryData(uint32_t params) { 169 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 170 if (service == NULL) { 171 // it already logged 172 return; 173 } 174 175 service->addBatteryData(params); 176} 177#endif 178 179static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 180{ 181 const hw_module_t *mod; 182 int rc; 183 184 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 185 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 186 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 187 if (rc) { 188 goto out; 189 } 190 rc = audio_hw_device_open(mod, dev); 191 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 197 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 198 rc = BAD_VALUE; 199 goto out; 200 } 201 return 0; 202 203out: 204 *dev = NULL; 205 return rc; 206} 207 208// ---------------------------------------------------------------------------- 209 210AudioFlinger::AudioFlinger() 211 : BnAudioFlinger(), 212 mPrimaryHardwareDev(NULL), 213 mHardwareStatus(AUDIO_HW_IDLE), 214 mMasterVolume(1.0f), 215 mMasterVolumeSW(1.0f), 216 mMasterVolumeSupportLvl(MVS_NONE), 217 mMasterMute(false), 218 mNextUniqueId(1), 219 mMode(AUDIO_MODE_INVALID), 220 mBtNrecIsOff(false) 221{ 222} 223 224void AudioFlinger::onFirstRef() 225{ 226 int rc = 0; 227 228 Mutex::Autolock _l(mLock); 229 230 /* TODO: move all this work into an Init() function */ 231 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 232 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 233 uint32_t int_val; 234 if (1 == sscanf(val_str, "%u", &int_val)) { 235 mStandbyTimeInNsecs = milliseconds(int_val); 236 ALOGI("Using %u mSec as standby time.", int_val); 237 } else { 238 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 239 ALOGI("Using default %u mSec as standby time.", 240 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 241 } 242 } 243 244 mMode = AUDIO_MODE_NORMAL; 245} 246 247AudioFlinger::~AudioFlinger() 248{ 249 250 while (!mRecordThreads.isEmpty()) { 251 // closeInput() will remove first entry from mRecordThreads 252 closeInput(mRecordThreads.keyAt(0)); 253 } 254 while (!mPlaybackThreads.isEmpty()) { 255 // closeOutput() will remove first entry from mPlaybackThreads 256 closeOutput(mPlaybackThreads.keyAt(0)); 257 } 258 259 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 260 // no mHardwareLock needed, as there are no other references to this 261 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 262 delete mAudioHwDevs.valueAt(i); 263 } 264} 265 266static const char * const audio_interfaces[] = { 267 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 268 AUDIO_HARDWARE_MODULE_ID_A2DP, 269 AUDIO_HARDWARE_MODULE_ID_USB, 270}; 271#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 272 273audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 274{ 275 // if module is 0, the request comes from an old policy manager and we should load 276 // well known modules 277 if (module == 0) { 278 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 279 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 280 loadHwModule_l(audio_interfaces[i]); 281 } 282 } else { 283 // check a match for the requested module handle 284 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 285 if (audioHwdevice != NULL) { 286 return audioHwdevice->hwDevice(); 287 } 288 } 289 // then try to find a module supporting the requested device. 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 292 if ((dev->get_supported_devices(dev) & devices) == devices) 293 return dev; 294 } 295 296 return NULL; 297} 298 299status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 305 result.append("Clients:\n"); 306 for (size_t i = 0; i < mClients.size(); ++i) { 307 sp<Client> client = mClients.valueAt(i).promote(); 308 if (client != 0) { 309 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 310 result.append(buffer); 311 } 312 } 313 314 result.append("Global session refs:\n"); 315 result.append(" session pid count\n"); 316 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 317 AudioSessionRef *r = mAudioSessionRefs[i]; 318 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 319 result.append(buffer); 320 } 321 write(fd, result.string(), result.size()); 322 return NO_ERROR; 323} 324 325 326status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 327{ 328 const size_t SIZE = 256; 329 char buffer[SIZE]; 330 String8 result; 331 hardware_call_state hardwareStatus = mHardwareStatus; 332 333 snprintf(buffer, SIZE, "Hardware status: %d\n" 334 "Standby Time mSec: %u\n", 335 hardwareStatus, 336 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 337 result.append(buffer); 338 write(fd, result.string(), result.size()); 339 return NO_ERROR; 340} 341 342status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353 return NO_ERROR; 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 audio_channel_mask_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 // check if an effect chain with the same session ID is present on another 473 // output thread and move it here. 474 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 475 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 476 if (mPlaybackThreads.keyAt(i) != output) { 477 uint32_t sessions = t->hasAudioSession(*sessionId); 478 if (sessions & PlaybackThread::EFFECT_SESSION) { 479 effectThread = t.get(); 480 break; 481 } 482 } 483 } 484 lSessionId = *sessionId; 485 } else { 486 // if no audio session id is provided, create one here 487 lSessionId = nextUniqueId(); 488 if (sessionId != NULL) { 489 *sessionId = lSessionId; 490 } 491 } 492 ALOGV("createTrack() lSessionId: %d", lSessionId); 493 494 track = thread->createTrack_l(client, streamType, sampleRate, format, 495 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 496 497 // move effect chain to this output thread if an effect on same session was waiting 498 // for a track to be created 499 if (lStatus == NO_ERROR && effectThread != NULL) { 500 Mutex::Autolock _dl(thread->mLock); 501 Mutex::Autolock _sl(effectThread->mLock); 502 moveEffectChain_l(lSessionId, effectThread, thread, true); 503 } 504 505 // Look for sync events awaiting for a session to be used. 506 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 507 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 508 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 509 if (lStatus == NO_ERROR) { 510 track->setSyncEvent(mPendingSyncEvents[i]); 511 } else { 512 mPendingSyncEvents[i]->cancel(); 513 } 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 RecordThread::RecordTrack *track = thread->track(); 872 if (track != NULL) { 873 audio_devices_t device = (audio_devices_t)( 874 thread->device() & AUDIO_DEVICE_IN_ALL); 875 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 876 thread->setEffectSuspended(FX_IID_AEC, 877 suspend, 878 track->sessionId()); 879 thread->setEffectSuspended(FX_IID_NS, 880 suspend, 881 track->sessionId()); 882 } 883 } 884 mBtNrecIsOff = btNrecIsOff; 885 } 886 } 887 String8 screenState; 888 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 889 bool isOff = screenState == "off"; 890 if (isOff != (gScreenState & 1)) { 891 gScreenState = ((gScreenState & ~1) + 2) | isOff; 892 } 893 } 894 return final_result; 895 } 896 897 // hold a strong ref on thread in case closeOutput() or closeInput() is called 898 // and the thread is exited once the lock is released 899 sp<ThreadBase> thread; 900 { 901 Mutex::Autolock _l(mLock); 902 thread = checkPlaybackThread_l(ioHandle); 903 if (thread == 0) { 904 thread = checkRecordThread_l(ioHandle); 905 } else if (thread == primaryPlaybackThread_l()) { 906 // indicate output device change to all input threads for pre processing 907 AudioParameter param = AudioParameter(keyValuePairs); 908 int value; 909 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 910 (value != 0)) { 911 for (size_t i = 0; i < mRecordThreads.size(); i++) { 912 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 913 } 914 } 915 } 916 } 917 if (thread != 0) { 918 return thread->setParameters(keyValuePairs); 919 } 920 return BAD_VALUE; 921} 922 923String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 924{ 925// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 926// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 927 928 Mutex::Autolock _l(mLock); 929 930 if (ioHandle == 0) { 931 String8 out_s8; 932 933 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 934 char *s; 935 { 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 938 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 939 s = dev->get_parameters(dev, keys.string()); 940 mHardwareStatus = AUDIO_HW_IDLE; 941 } 942 out_s8 += String8(s ? s : ""); 943 free(s); 944 } 945 return out_s8; 946 } 947 948 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 949 if (playbackThread != NULL) { 950 return playbackThread->getParameters(keys); 951 } 952 RecordThread *recordThread = checkRecordThread_l(ioHandle); 953 if (recordThread != NULL) { 954 return recordThread->getParameters(keys); 955 } 956 return String8(""); 957} 958 959size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 960 audio_channel_mask_t channelMask) const 961{ 962 status_t ret = initCheck(); 963 if (ret != NO_ERROR) { 964 return 0; 965 } 966 967 AutoMutex lock(mHardwareLock); 968 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 969 struct audio_config config = { 970 sample_rate: sampleRate, 971 channel_mask: channelMask, 972 format: format, 973 }; 974 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 975 mHardwareStatus = AUDIO_HW_IDLE; 976 return size; 977} 978 979unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 980{ 981 Mutex::Autolock _l(mLock); 982 983 RecordThread *recordThread = checkRecordThread_l(ioHandle); 984 if (recordThread != NULL) { 985 return recordThread->getInputFramesLost(); 986 } 987 return 0; 988} 989 990status_t AudioFlinger::setVoiceVolume(float value) 991{ 992 status_t ret = initCheck(); 993 if (ret != NO_ERROR) { 994 return ret; 995 } 996 997 // check calling permissions 998 if (!settingsAllowed()) { 999 return PERMISSION_DENIED; 1000 } 1001 1002 AutoMutex lock(mHardwareLock); 1003 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1004 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1005 mHardwareStatus = AUDIO_HW_IDLE; 1006 1007 return ret; 1008} 1009 1010status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1011 audio_io_handle_t output) const 1012{ 1013 status_t status; 1014 1015 Mutex::Autolock _l(mLock); 1016 1017 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1018 if (playbackThread != NULL) { 1019 return playbackThread->getRenderPosition(halFrames, dspFrames); 1020 } 1021 1022 return BAD_VALUE; 1023} 1024 1025void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1026{ 1027 1028 Mutex::Autolock _l(mLock); 1029 1030 pid_t pid = IPCThreadState::self()->getCallingPid(); 1031 if (mNotificationClients.indexOfKey(pid) < 0) { 1032 sp<NotificationClient> notificationClient = new NotificationClient(this, 1033 client, 1034 pid); 1035 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1036 1037 mNotificationClients.add(pid, notificationClient); 1038 1039 sp<IBinder> binder = client->asBinder(); 1040 binder->linkToDeath(notificationClient); 1041 1042 // the config change is always sent from playback or record threads to avoid deadlock 1043 // with AudioSystem::gLock 1044 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1045 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1046 } 1047 1048 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1049 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1050 } 1051 } 1052} 1053 1054void AudioFlinger::removeNotificationClient(pid_t pid) 1055{ 1056 Mutex::Autolock _l(mLock); 1057 1058 mNotificationClients.removeItem(pid); 1059 1060 ALOGV("%d died, releasing its sessions", pid); 1061 size_t num = mAudioSessionRefs.size(); 1062 bool removed = false; 1063 for (size_t i = 0; i< num; ) { 1064 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1065 ALOGV(" pid %d @ %d", ref->mPid, i); 1066 if (ref->mPid == pid) { 1067 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1068 mAudioSessionRefs.removeAt(i); 1069 delete ref; 1070 removed = true; 1071 num--; 1072 } else { 1073 i++; 1074 } 1075 } 1076 if (removed) { 1077 purgeStaleEffects_l(); 1078 } 1079} 1080 1081// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1082void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1083{ 1084 size_t size = mNotificationClients.size(); 1085 for (size_t i = 0; i < size; i++) { 1086 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1087 param2); 1088 } 1089} 1090 1091// removeClient_l() must be called with AudioFlinger::mLock held 1092void AudioFlinger::removeClient_l(pid_t pid) 1093{ 1094 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1095 mClients.removeItem(pid); 1096} 1097 1098// getEffectThread_l() must be called with AudioFlinger::mLock held 1099sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1100{ 1101 sp<PlaybackThread> thread; 1102 1103 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1104 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1105 ALOG_ASSERT(thread == 0); 1106 thread = mPlaybackThreads.valueAt(i); 1107 } 1108 } 1109 1110 return thread; 1111} 1112 1113// ---------------------------------------------------------------------------- 1114 1115AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1116 uint32_t device, type_t type) 1117 : Thread(false), 1118 mType(type), 1119 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1120 // mChannelMask 1121 mChannelCount(0), 1122 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1123 mParamStatus(NO_ERROR), 1124 mStandby(false), mDevice((audio_devices_t) device), mId(id), 1125 mDeathRecipient(new PMDeathRecipient(this)) 1126{ 1127} 1128 1129AudioFlinger::ThreadBase::~ThreadBase() 1130{ 1131 mParamCond.broadcast(); 1132 // do not lock the mutex in destructor 1133 releaseWakeLock_l(); 1134 if (mPowerManager != 0) { 1135 sp<IBinder> binder = mPowerManager->asBinder(); 1136 binder->unlinkToDeath(mDeathRecipient); 1137 } 1138} 1139 1140void AudioFlinger::ThreadBase::exit() 1141{ 1142 ALOGV("ThreadBase::exit"); 1143 { 1144 // This lock prevents the following race in thread (uniprocessor for illustration): 1145 // if (!exitPending()) { 1146 // // context switch from here to exit() 1147 // // exit() calls requestExit(), what exitPending() observes 1148 // // exit() calls signal(), which is dropped since no waiters 1149 // // context switch back from exit() to here 1150 // mWaitWorkCV.wait(...); 1151 // // now thread is hung 1152 // } 1153 AutoMutex lock(mLock); 1154 requestExit(); 1155 mWaitWorkCV.signal(); 1156 } 1157 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1158 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1159 requestExitAndWait(); 1160} 1161 1162status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1163{ 1164 status_t status; 1165 1166 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1167 Mutex::Autolock _l(mLock); 1168 1169 mNewParameters.add(keyValuePairs); 1170 mWaitWorkCV.signal(); 1171 // wait condition with timeout in case the thread loop has exited 1172 // before the request could be processed 1173 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1174 status = mParamStatus; 1175 mWaitWorkCV.signal(); 1176 } else { 1177 status = TIMED_OUT; 1178 } 1179 return status; 1180} 1181 1182void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1183{ 1184 Mutex::Autolock _l(mLock); 1185 sendConfigEvent_l(event, param); 1186} 1187 1188// sendConfigEvent_l() must be called with ThreadBase::mLock held 1189void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1190{ 1191 ConfigEvent configEvent; 1192 configEvent.mEvent = event; 1193 configEvent.mParam = param; 1194 mConfigEvents.add(configEvent); 1195 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1196 mWaitWorkCV.signal(); 1197} 1198 1199void AudioFlinger::ThreadBase::processConfigEvents() 1200{ 1201 mLock.lock(); 1202 while (!mConfigEvents.isEmpty()) { 1203 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1204 ConfigEvent configEvent = mConfigEvents[0]; 1205 mConfigEvents.removeAt(0); 1206 // release mLock before locking AudioFlinger mLock: lock order is always 1207 // AudioFlinger then ThreadBase to avoid cross deadlock 1208 mLock.unlock(); 1209 mAudioFlinger->mLock.lock(); 1210 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1211 mAudioFlinger->mLock.unlock(); 1212 mLock.lock(); 1213 } 1214 mLock.unlock(); 1215} 1216 1217status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1218{ 1219 const size_t SIZE = 256; 1220 char buffer[SIZE]; 1221 String8 result; 1222 1223 bool locked = tryLock(mLock); 1224 if (!locked) { 1225 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1226 write(fd, buffer, strlen(buffer)); 1227 } 1228 1229 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1246 result.append(buffer); 1247 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1248 result.append(buffer); 1249 1250 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1251 result.append(buffer); 1252 result.append(" Index Command"); 1253 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1254 snprintf(buffer, SIZE, "\n %02d ", i); 1255 result.append(buffer); 1256 result.append(mNewParameters[i]); 1257 } 1258 1259 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1260 result.append(buffer); 1261 snprintf(buffer, SIZE, " Index event param\n"); 1262 result.append(buffer); 1263 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1264 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1265 result.append(buffer); 1266 } 1267 result.append("\n"); 1268 1269 write(fd, result.string(), result.size()); 1270 1271 if (locked) { 1272 mLock.unlock(); 1273 } 1274 return NO_ERROR; 1275} 1276 1277status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1278{ 1279 const size_t SIZE = 256; 1280 char buffer[SIZE]; 1281 String8 result; 1282 1283 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1284 write(fd, buffer, strlen(buffer)); 1285 1286 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1287 sp<EffectChain> chain = mEffectChains[i]; 1288 if (chain != 0) { 1289 chain->dump(fd, args); 1290 } 1291 } 1292 return NO_ERROR; 1293} 1294 1295void AudioFlinger::ThreadBase::acquireWakeLock() 1296{ 1297 Mutex::Autolock _l(mLock); 1298 acquireWakeLock_l(); 1299} 1300 1301void AudioFlinger::ThreadBase::acquireWakeLock_l() 1302{ 1303 if (mPowerManager == 0) { 1304 // use checkService() to avoid blocking if power service is not up yet 1305 sp<IBinder> binder = 1306 defaultServiceManager()->checkService(String16("power")); 1307 if (binder == 0) { 1308 ALOGW("Thread %s cannot connect to the power manager service", mName); 1309 } else { 1310 mPowerManager = interface_cast<IPowerManager>(binder); 1311 binder->linkToDeath(mDeathRecipient); 1312 } 1313 } 1314 if (mPowerManager != 0) { 1315 sp<IBinder> binder = new BBinder(); 1316 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1317 binder, 1318 String16(mName)); 1319 if (status == NO_ERROR) { 1320 mWakeLockToken = binder; 1321 } 1322 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1323 } 1324} 1325 1326void AudioFlinger::ThreadBase::releaseWakeLock() 1327{ 1328 Mutex::Autolock _l(mLock); 1329 releaseWakeLock_l(); 1330} 1331 1332void AudioFlinger::ThreadBase::releaseWakeLock_l() 1333{ 1334 if (mWakeLockToken != 0) { 1335 ALOGV("releaseWakeLock_l() %s", mName); 1336 if (mPowerManager != 0) { 1337 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1338 } 1339 mWakeLockToken.clear(); 1340 } 1341} 1342 1343void AudioFlinger::ThreadBase::clearPowerManager() 1344{ 1345 Mutex::Autolock _l(mLock); 1346 releaseWakeLock_l(); 1347 mPowerManager.clear(); 1348} 1349 1350void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1351{ 1352 sp<ThreadBase> thread = mThread.promote(); 1353 if (thread != 0) { 1354 thread->clearPowerManager(); 1355 } 1356 ALOGW("power manager service died !!!"); 1357} 1358 1359void AudioFlinger::ThreadBase::setEffectSuspended( 1360 const effect_uuid_t *type, bool suspend, int sessionId) 1361{ 1362 Mutex::Autolock _l(mLock); 1363 setEffectSuspended_l(type, suspend, sessionId); 1364} 1365 1366void AudioFlinger::ThreadBase::setEffectSuspended_l( 1367 const effect_uuid_t *type, bool suspend, int sessionId) 1368{ 1369 sp<EffectChain> chain = getEffectChain_l(sessionId); 1370 if (chain != 0) { 1371 if (type != NULL) { 1372 chain->setEffectSuspended_l(type, suspend); 1373 } else { 1374 chain->setEffectSuspendedAll_l(suspend); 1375 } 1376 } 1377 1378 updateSuspendedSessions_l(type, suspend, sessionId); 1379} 1380 1381void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1382{ 1383 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1384 if (index < 0) { 1385 return; 1386 } 1387 1388 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1389 mSuspendedSessions.editValueAt(index); 1390 1391 for (size_t i = 0; i < sessionEffects.size(); i++) { 1392 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1393 for (int j = 0; j < desc->mRefCount; j++) { 1394 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1395 chain->setEffectSuspendedAll_l(true); 1396 } else { 1397 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1398 desc->mType.timeLow); 1399 chain->setEffectSuspended_l(&desc->mType, true); 1400 } 1401 } 1402 } 1403} 1404 1405void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1406 bool suspend, 1407 int sessionId) 1408{ 1409 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1410 1411 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1412 1413 if (suspend) { 1414 if (index >= 0) { 1415 sessionEffects = mSuspendedSessions.editValueAt(index); 1416 } else { 1417 mSuspendedSessions.add(sessionId, sessionEffects); 1418 } 1419 } else { 1420 if (index < 0) { 1421 return; 1422 } 1423 sessionEffects = mSuspendedSessions.editValueAt(index); 1424 } 1425 1426 1427 int key = EffectChain::kKeyForSuspendAll; 1428 if (type != NULL) { 1429 key = type->timeLow; 1430 } 1431 index = sessionEffects.indexOfKey(key); 1432 1433 sp<SuspendedSessionDesc> desc; 1434 if (suspend) { 1435 if (index >= 0) { 1436 desc = sessionEffects.valueAt(index); 1437 } else { 1438 desc = new SuspendedSessionDesc(); 1439 if (type != NULL) { 1440 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1441 } 1442 sessionEffects.add(key, desc); 1443 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1444 } 1445 desc->mRefCount++; 1446 } else { 1447 if (index < 0) { 1448 return; 1449 } 1450 desc = sessionEffects.valueAt(index); 1451 if (--desc->mRefCount == 0) { 1452 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1453 sessionEffects.removeItemsAt(index); 1454 if (sessionEffects.isEmpty()) { 1455 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1456 sessionId); 1457 mSuspendedSessions.removeItem(sessionId); 1458 } 1459 } 1460 } 1461 if (!sessionEffects.isEmpty()) { 1462 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1463 } 1464} 1465 1466void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1467 bool enabled, 1468 int sessionId) 1469{ 1470 Mutex::Autolock _l(mLock); 1471 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1472} 1473 1474void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1475 bool enabled, 1476 int sessionId) 1477{ 1478 if (mType != RECORD) { 1479 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1480 // another session. This gives the priority to well behaved effect control panels 1481 // and applications not using global effects. 1482 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1483 // global effects 1484 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1485 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1486 } 1487 } 1488 1489 sp<EffectChain> chain = getEffectChain_l(sessionId); 1490 if (chain != 0) { 1491 chain->checkSuspendOnEffectEnabled(effect, enabled); 1492 } 1493} 1494 1495// ---------------------------------------------------------------------------- 1496 1497AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1498 AudioStreamOut* output, 1499 audio_io_handle_t id, 1500 uint32_t device, 1501 type_t type) 1502 : ThreadBase(audioFlinger, id, device, type), 1503 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1504 // Assumes constructor is called by AudioFlinger with it's mLock held, 1505 // but it would be safer to explicitly pass initial masterMute as parameter 1506 mMasterMute(audioFlinger->masterMute_l()), 1507 // mStreamTypes[] initialized in constructor body 1508 mOutput(output), 1509 // Assumes constructor is called by AudioFlinger with it's mLock held, 1510 // but it would be safer to explicitly pass initial masterVolume as parameter 1511 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1512 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1513 mMixerStatus(MIXER_IDLE), 1514 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1515 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1516 mScreenState(gScreenState), 1517 // index 0 is reserved for normal mixer's submix 1518 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1519{ 1520 snprintf(mName, kNameLength, "AudioOut_%X", id); 1521 1522 readOutputParameters(); 1523 1524 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1525 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1526 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1527 stream = (audio_stream_type_t) (stream + 1)) { 1528 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1529 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1530 } 1531 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1532 // because mAudioFlinger doesn't have one to copy from 1533} 1534 1535AudioFlinger::PlaybackThread::~PlaybackThread() 1536{ 1537 delete [] mMixBuffer; 1538} 1539 1540status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1541{ 1542 dumpInternals(fd, args); 1543 dumpTracks(fd, args); 1544 dumpEffectChains(fd, args); 1545 return NO_ERROR; 1546} 1547 1548status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1549{ 1550 const size_t SIZE = 256; 1551 char buffer[SIZE]; 1552 String8 result; 1553 1554 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1555 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1556 const stream_type_t *st = &mStreamTypes[i]; 1557 if (i > 0) { 1558 result.appendFormat(", "); 1559 } 1560 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1561 if (st->mute) { 1562 result.append("M"); 1563 } 1564 } 1565 result.append("\n"); 1566 write(fd, result.string(), result.length()); 1567 result.clear(); 1568 1569 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1570 result.append(buffer); 1571 Track::appendDumpHeader(result); 1572 for (size_t i = 0; i < mTracks.size(); ++i) { 1573 sp<Track> track = mTracks[i]; 1574 if (track != 0) { 1575 track->dump(buffer, SIZE); 1576 result.append(buffer); 1577 } 1578 } 1579 1580 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1581 result.append(buffer); 1582 Track::appendDumpHeader(result); 1583 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1584 sp<Track> track = mActiveTracks[i].promote(); 1585 if (track != 0) { 1586 track->dump(buffer, SIZE); 1587 result.append(buffer); 1588 } 1589 } 1590 write(fd, result.string(), result.size()); 1591 1592 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1593 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1594 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1595 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1596 1597 return NO_ERROR; 1598} 1599 1600status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1601{ 1602 const size_t SIZE = 256; 1603 char buffer[SIZE]; 1604 String8 result; 1605 1606 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1607 result.append(buffer); 1608 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1609 result.append(buffer); 1610 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1611 result.append(buffer); 1612 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1613 result.append(buffer); 1614 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1615 result.append(buffer); 1616 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1617 result.append(buffer); 1618 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1619 result.append(buffer); 1620 write(fd, result.string(), result.size()); 1621 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1622 1623 dumpBase(fd, args); 1624 1625 return NO_ERROR; 1626} 1627 1628// Thread virtuals 1629status_t AudioFlinger::PlaybackThread::readyToRun() 1630{ 1631 status_t status = initCheck(); 1632 if (status == NO_ERROR) { 1633 ALOGI("AudioFlinger's thread %p ready to run", this); 1634 } else { 1635 ALOGE("No working audio driver found."); 1636 } 1637 return status; 1638} 1639 1640void AudioFlinger::PlaybackThread::onFirstRef() 1641{ 1642 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1643} 1644 1645// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1646sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1647 const sp<AudioFlinger::Client>& client, 1648 audio_stream_type_t streamType, 1649 uint32_t sampleRate, 1650 audio_format_t format, 1651 audio_channel_mask_t channelMask, 1652 int frameCount, 1653 const sp<IMemory>& sharedBuffer, 1654 int sessionId, 1655 IAudioFlinger::track_flags_t flags, 1656 pid_t tid, 1657 status_t *status) 1658{ 1659 sp<Track> track; 1660 status_t lStatus; 1661 1662 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1663 1664 // client expresses a preference for FAST, but we get the final say 1665 if (flags & IAudioFlinger::TRACK_FAST) { 1666 if ( 1667 // not timed 1668 (!isTimed) && 1669 // either of these use cases: 1670 ( 1671 // use case 1: shared buffer with any frame count 1672 ( 1673 (sharedBuffer != 0) 1674 ) || 1675 // use case 2: callback handler and frame count is default or at least as large as HAL 1676 ( 1677 (tid != -1) && 1678 ((frameCount == 0) || 1679 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1680 ) 1681 ) && 1682 // PCM data 1683 audio_is_linear_pcm(format) && 1684 // mono or stereo 1685 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1686 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1687#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1688 // hardware sample rate 1689 (sampleRate == mSampleRate) && 1690#endif 1691 // normal mixer has an associated fast mixer 1692 hasFastMixer() && 1693 // there are sufficient fast track slots available 1694 (mFastTrackAvailMask != 0) 1695 // FIXME test that MixerThread for this fast track has a capable output HAL 1696 // FIXME add a permission test also? 1697 ) { 1698 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1699 if (frameCount == 0) { 1700 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1701 } 1702 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1703 frameCount, mFrameCount); 1704 } else { 1705 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1706 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1707 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1708 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1709 audio_is_linear_pcm(format), 1710 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1711 flags &= ~IAudioFlinger::TRACK_FAST; 1712 // For compatibility with AudioTrack calculation, buffer depth is forced 1713 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1714 // This is probably too conservative, but legacy application code may depend on it. 1715 // If you change this calculation, also review the start threshold which is related. 1716 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1717 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1718 if (minBufCount < 2) { 1719 minBufCount = 2; 1720 } 1721 int minFrameCount = mNormalFrameCount * minBufCount; 1722 if (frameCount < minFrameCount) { 1723 frameCount = minFrameCount; 1724 } 1725 } 1726 } 1727 1728 if (mType == DIRECT) { 1729 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1730 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1731 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1732 "for output %p with format %d", 1733 sampleRate, format, channelMask, mOutput, mFormat); 1734 lStatus = BAD_VALUE; 1735 goto Exit; 1736 } 1737 } 1738 } else { 1739 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1740 if (sampleRate > mSampleRate*2) { 1741 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1742 lStatus = BAD_VALUE; 1743 goto Exit; 1744 } 1745 } 1746 1747 lStatus = initCheck(); 1748 if (lStatus != NO_ERROR) { 1749 ALOGE("Audio driver not initialized."); 1750 goto Exit; 1751 } 1752 1753 { // scope for mLock 1754 Mutex::Autolock _l(mLock); 1755 1756 // all tracks in same audio session must share the same routing strategy otherwise 1757 // conflicts will happen when tracks are moved from one output to another by audio policy 1758 // manager 1759 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1760 for (size_t i = 0; i < mTracks.size(); ++i) { 1761 sp<Track> t = mTracks[i]; 1762 if (t != 0 && !t->isOutputTrack()) { 1763 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1764 if (sessionId == t->sessionId() && strategy != actual) { 1765 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1766 strategy, actual); 1767 lStatus = BAD_VALUE; 1768 goto Exit; 1769 } 1770 } 1771 } 1772 1773 if (!isTimed) { 1774 track = new Track(this, client, streamType, sampleRate, format, 1775 channelMask, frameCount, sharedBuffer, sessionId, flags); 1776 } else { 1777 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1778 channelMask, frameCount, sharedBuffer, sessionId); 1779 } 1780 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1781 lStatus = NO_MEMORY; 1782 goto Exit; 1783 } 1784 mTracks.add(track); 1785 1786 sp<EffectChain> chain = getEffectChain_l(sessionId); 1787 if (chain != 0) { 1788 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1789 track->setMainBuffer(chain->inBuffer()); 1790 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1791 chain->incTrackCnt(); 1792 } 1793 } 1794 1795 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1796 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1797 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1798 // so ask activity manager to do this on our behalf 1799 int err = requestPriority(callingPid, tid, 1); 1800 if (err != 0) { 1801 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1802 1, callingPid, tid, err); 1803 } 1804 } 1805 1806 lStatus = NO_ERROR; 1807 1808Exit: 1809 if (status) { 1810 *status = lStatus; 1811 } 1812 return track; 1813} 1814 1815uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1816{ 1817 if (mFastMixer != NULL) { 1818 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1819 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1820 } 1821 return latency; 1822} 1823 1824uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1825{ 1826 return latency; 1827} 1828 1829uint32_t AudioFlinger::PlaybackThread::latency() const 1830{ 1831 Mutex::Autolock _l(mLock); 1832 return latency_l(); 1833} 1834uint32_t AudioFlinger::PlaybackThread::latency_l() const 1835{ 1836 if (initCheck() == NO_ERROR) { 1837 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1838 } else { 1839 return 0; 1840 } 1841} 1842 1843void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 mMasterVolume = value; 1847} 1848 1849void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 setMasterMute_l(muted); 1853} 1854 1855void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1856{ 1857 Mutex::Autolock _l(mLock); 1858 mStreamTypes[stream].volume = value; 1859} 1860 1861void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1862{ 1863 Mutex::Autolock _l(mLock); 1864 mStreamTypes[stream].mute = muted; 1865} 1866 1867float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1868{ 1869 Mutex::Autolock _l(mLock); 1870 return mStreamTypes[stream].volume; 1871} 1872 1873// addTrack_l() must be called with ThreadBase::mLock held 1874status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1875{ 1876 status_t status = ALREADY_EXISTS; 1877 1878 // set retry count for buffer fill 1879 track->mRetryCount = kMaxTrackStartupRetries; 1880 if (mActiveTracks.indexOf(track) < 0) { 1881 // the track is newly added, make sure it fills up all its 1882 // buffers before playing. This is to ensure the client will 1883 // effectively get the latency it requested. 1884 track->mFillingUpStatus = Track::FS_FILLING; 1885 track->mResetDone = false; 1886 track->mPresentationCompleteFrames = 0; 1887 mActiveTracks.add(track); 1888 if (track->mainBuffer() != mMixBuffer) { 1889 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1890 if (chain != 0) { 1891 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1892 chain->incActiveTrackCnt(); 1893 } 1894 } 1895 1896 status = NO_ERROR; 1897 } 1898 1899 ALOGV("mWaitWorkCV.broadcast"); 1900 mWaitWorkCV.broadcast(); 1901 1902 return status; 1903} 1904 1905// destroyTrack_l() must be called with ThreadBase::mLock held 1906void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1907{ 1908 track->mState = TrackBase::TERMINATED; 1909 // active tracks are removed by threadLoop() 1910 if (mActiveTracks.indexOf(track) < 0) { 1911 removeTrack_l(track); 1912 } 1913} 1914 1915void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1916{ 1917 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1918 mTracks.remove(track); 1919 deleteTrackName_l(track->name()); 1920 // redundant as track is about to be destroyed, for dumpsys only 1921 track->mName = -1; 1922 if (track->isFastTrack()) { 1923 int index = track->mFastIndex; 1924 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1925 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1926 mFastTrackAvailMask |= 1 << index; 1927 // redundant as track is about to be destroyed, for dumpsys only 1928 track->mFastIndex = -1; 1929 } 1930 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1931 if (chain != 0) { 1932 chain->decTrackCnt(); 1933 } 1934} 1935 1936String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1937{ 1938 String8 out_s8 = String8(""); 1939 char *s; 1940 1941 Mutex::Autolock _l(mLock); 1942 if (initCheck() != NO_ERROR) { 1943 return out_s8; 1944 } 1945 1946 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1947 out_s8 = String8(s); 1948 free(s); 1949 return out_s8; 1950} 1951 1952// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1953void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1954 AudioSystem::OutputDescriptor desc; 1955 void *param2 = NULL; 1956 1957 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1958 1959 switch (event) { 1960 case AudioSystem::OUTPUT_OPENED: 1961 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1962 desc.channels = mChannelMask; 1963 desc.samplingRate = mSampleRate; 1964 desc.format = mFormat; 1965 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1966 desc.latency = latency(); 1967 param2 = &desc; 1968 break; 1969 1970 case AudioSystem::STREAM_CONFIG_CHANGED: 1971 param2 = ¶m; 1972 case AudioSystem::OUTPUT_CLOSED: 1973 default: 1974 break; 1975 } 1976 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1977} 1978 1979void AudioFlinger::PlaybackThread::readOutputParameters() 1980{ 1981 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1982 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1983 mChannelCount = (uint16_t)popcount(mChannelMask); 1984 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1985 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1986 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1987 if (mFrameCount & 15) { 1988 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1989 mFrameCount); 1990 } 1991 1992 // Calculate size of normal mix buffer relative to the HAL output buffer size 1993 double multiplier = 1.0; 1994 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1995 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1996 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1997 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1998 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1999 maxNormalFrameCount = maxNormalFrameCount & ~15; 2000 if (maxNormalFrameCount < minNormalFrameCount) { 2001 maxNormalFrameCount = minNormalFrameCount; 2002 } 2003 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2004 if (multiplier <= 1.0) { 2005 multiplier = 1.0; 2006 } else if (multiplier <= 2.0) { 2007 if (2 * mFrameCount <= maxNormalFrameCount) { 2008 multiplier = 2.0; 2009 } else { 2010 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2011 } 2012 } else { 2013 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2014 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2015 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2016 // FIXME this rounding up should not be done if no HAL SRC 2017 uint32_t truncMult = (uint32_t) multiplier; 2018 if ((truncMult & 1)) { 2019 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2020 ++truncMult; 2021 } 2022 } 2023 multiplier = (double) truncMult; 2024 } 2025 } 2026 mNormalFrameCount = multiplier * mFrameCount; 2027 // round up to nearest 16 frames to satisfy AudioMixer 2028 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2029 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2030 2031 delete[] mMixBuffer; 2032 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2033 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2034 2035 // force reconfiguration of effect chains and engines to take new buffer size and audio 2036 // parameters into account 2037 // Note that mLock is not held when readOutputParameters() is called from the constructor 2038 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2039 // matter. 2040 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2041 Vector< sp<EffectChain> > effectChains = mEffectChains; 2042 for (size_t i = 0; i < effectChains.size(); i ++) { 2043 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2044 } 2045} 2046 2047 2048status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2049{ 2050 if (halFrames == NULL || dspFrames == NULL) { 2051 return BAD_VALUE; 2052 } 2053 Mutex::Autolock _l(mLock); 2054 if (initCheck() != NO_ERROR) { 2055 return INVALID_OPERATION; 2056 } 2057 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2058 2059 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2060} 2061 2062uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2063{ 2064 Mutex::Autolock _l(mLock); 2065 uint32_t result = 0; 2066 if (getEffectChain_l(sessionId) != 0) { 2067 result = EFFECT_SESSION; 2068 } 2069 2070 for (size_t i = 0; i < mTracks.size(); ++i) { 2071 sp<Track> track = mTracks[i]; 2072 if (sessionId == track->sessionId() && 2073 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2074 result |= TRACK_SESSION; 2075 break; 2076 } 2077 } 2078 2079 return result; 2080} 2081 2082uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2083{ 2084 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2085 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2086 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2087 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2088 } 2089 for (size_t i = 0; i < mTracks.size(); i++) { 2090 sp<Track> track = mTracks[i]; 2091 if (sessionId == track->sessionId() && 2092 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2093 return AudioSystem::getStrategyForStream(track->streamType()); 2094 } 2095 } 2096 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2097} 2098 2099 2100AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2101{ 2102 Mutex::Autolock _l(mLock); 2103 return mOutput; 2104} 2105 2106AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2107{ 2108 Mutex::Autolock _l(mLock); 2109 AudioStreamOut *output = mOutput; 2110 mOutput = NULL; 2111 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2112 // must push a NULL and wait for ack 2113 mOutputSink.clear(); 2114 mPipeSink.clear(); 2115 mNormalSink.clear(); 2116 return output; 2117} 2118 2119// this method must always be called either with ThreadBase mLock held or inside the thread loop 2120audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2121{ 2122 if (mOutput == NULL) { 2123 return NULL; 2124 } 2125 return &mOutput->stream->common; 2126} 2127 2128uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2129{ 2130 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2131} 2132 2133status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2134{ 2135 if (!isValidSyncEvent(event)) { 2136 return BAD_VALUE; 2137 } 2138 2139 Mutex::Autolock _l(mLock); 2140 2141 for (size_t i = 0; i < mTracks.size(); ++i) { 2142 sp<Track> track = mTracks[i]; 2143 if (event->triggerSession() == track->sessionId()) { 2144 track->setSyncEvent(event); 2145 return NO_ERROR; 2146 } 2147 } 2148 2149 return NAME_NOT_FOUND; 2150} 2151 2152bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2153{ 2154 switch (event->type()) { 2155 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2156 return true; 2157 default: 2158 break; 2159 } 2160 return false; 2161} 2162 2163void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2164{ 2165 size_t count = tracksToRemove.size(); 2166 if (CC_UNLIKELY(count)) { 2167 for (size_t i = 0 ; i < count ; i++) { 2168 const sp<Track>& track = tracksToRemove.itemAt(i); 2169 if ((track->sharedBuffer() != 0) && 2170 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2171 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2172 } 2173 } 2174 } 2175 2176} 2177 2178// ---------------------------------------------------------------------------- 2179 2180AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2181 audio_io_handle_t id, uint32_t device, type_t type) 2182 : PlaybackThread(audioFlinger, output, id, device, type), 2183 // mAudioMixer below 2184 // mFastMixer below 2185 mFastMixerFutex(0) 2186 // mOutputSink below 2187 // mPipeSink below 2188 // mNormalSink below 2189{ 2190 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2191 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2192 "mFrameCount=%d, mNormalFrameCount=%d", 2193 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2194 mNormalFrameCount); 2195 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2196 2197 // FIXME - Current mixer implementation only supports stereo output 2198 if (mChannelCount != FCC_2) { 2199 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2200 } 2201 2202 // create an NBAIO sink for the HAL output stream, and negotiate 2203 mOutputSink = new AudioStreamOutSink(output->stream); 2204 size_t numCounterOffers = 0; 2205 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2206 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2207 ALOG_ASSERT(index == 0); 2208 2209 // initialize fast mixer depending on configuration 2210 bool initFastMixer; 2211 switch (kUseFastMixer) { 2212 case FastMixer_Never: 2213 initFastMixer = false; 2214 break; 2215 case FastMixer_Always: 2216 initFastMixer = true; 2217 break; 2218 case FastMixer_Static: 2219 case FastMixer_Dynamic: 2220 initFastMixer = mFrameCount < mNormalFrameCount; 2221 break; 2222 } 2223 if (initFastMixer) { 2224 2225 // create a MonoPipe to connect our submix to FastMixer 2226 NBAIO_Format format = mOutputSink->format(); 2227 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2228 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2229 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2230 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2231 const NBAIO_Format offers[1] = {format}; 2232 size_t numCounterOffers = 0; 2233 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2234 ALOG_ASSERT(index == 0); 2235 monoPipe->setAvgFrames((mScreenState & 1) ? 2236 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2237 mPipeSink = monoPipe; 2238 2239#ifdef TEE_SINK_FRAMES 2240 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2241 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2242 numCounterOffers = 0; 2243 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2244 ALOG_ASSERT(index == 0); 2245 mTeeSink = teeSink; 2246 PipeReader *teeSource = new PipeReader(*teeSink); 2247 numCounterOffers = 0; 2248 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2249 ALOG_ASSERT(index == 0); 2250 mTeeSource = teeSource; 2251#endif 2252 2253 // create fast mixer and configure it initially with just one fast track for our submix 2254 mFastMixer = new FastMixer(); 2255 FastMixerStateQueue *sq = mFastMixer->sq(); 2256#ifdef STATE_QUEUE_DUMP 2257 sq->setObserverDump(&mStateQueueObserverDump); 2258 sq->setMutatorDump(&mStateQueueMutatorDump); 2259#endif 2260 FastMixerState *state = sq->begin(); 2261 FastTrack *fastTrack = &state->mFastTracks[0]; 2262 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2263 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2264 fastTrack->mVolumeProvider = NULL; 2265 fastTrack->mGeneration++; 2266 state->mFastTracksGen++; 2267 state->mTrackMask = 1; 2268 // fast mixer will use the HAL output sink 2269 state->mOutputSink = mOutputSink.get(); 2270 state->mOutputSinkGen++; 2271 state->mFrameCount = mFrameCount; 2272 state->mCommand = FastMixerState::COLD_IDLE; 2273 // already done in constructor initialization list 2274 //mFastMixerFutex = 0; 2275 state->mColdFutexAddr = &mFastMixerFutex; 2276 state->mColdGen++; 2277 state->mDumpState = &mFastMixerDumpState; 2278 state->mTeeSink = mTeeSink.get(); 2279 sq->end(); 2280 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2281 2282 // start the fast mixer 2283 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2284 pid_t tid = mFastMixer->getTid(); 2285 int err = requestPriority(getpid_cached, tid, 2); 2286 if (err != 0) { 2287 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2288 2, getpid_cached, tid, err); 2289 } 2290 2291#ifdef AUDIO_WATCHDOG 2292 // create and start the watchdog 2293 mAudioWatchdog = new AudioWatchdog(); 2294 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2295 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2296 tid = mAudioWatchdog->getTid(); 2297 err = requestPriority(getpid_cached, tid, 1); 2298 if (err != 0) { 2299 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2300 1, getpid_cached, tid, err); 2301 } 2302#endif 2303 2304 } else { 2305 mFastMixer = NULL; 2306 } 2307 2308 switch (kUseFastMixer) { 2309 case FastMixer_Never: 2310 case FastMixer_Dynamic: 2311 mNormalSink = mOutputSink; 2312 break; 2313 case FastMixer_Always: 2314 mNormalSink = mPipeSink; 2315 break; 2316 case FastMixer_Static: 2317 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2318 break; 2319 } 2320} 2321 2322AudioFlinger::MixerThread::~MixerThread() 2323{ 2324 if (mFastMixer != NULL) { 2325 FastMixerStateQueue *sq = mFastMixer->sq(); 2326 FastMixerState *state = sq->begin(); 2327 if (state->mCommand == FastMixerState::COLD_IDLE) { 2328 int32_t old = android_atomic_inc(&mFastMixerFutex); 2329 if (old == -1) { 2330 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2331 } 2332 } 2333 state->mCommand = FastMixerState::EXIT; 2334 sq->end(); 2335 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2336 mFastMixer->join(); 2337 // Though the fast mixer thread has exited, it's state queue is still valid. 2338 // We'll use that extract the final state which contains one remaining fast track 2339 // corresponding to our sub-mix. 2340 state = sq->begin(); 2341 ALOG_ASSERT(state->mTrackMask == 1); 2342 FastTrack *fastTrack = &state->mFastTracks[0]; 2343 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2344 delete fastTrack->mBufferProvider; 2345 sq->end(false /*didModify*/); 2346 delete mFastMixer; 2347 if (mAudioWatchdog != 0) { 2348 mAudioWatchdog->requestExit(); 2349 mAudioWatchdog->requestExitAndWait(); 2350 mAudioWatchdog.clear(); 2351 } 2352 } 2353 delete mAudioMixer; 2354} 2355 2356class CpuStats { 2357public: 2358 CpuStats(); 2359 void sample(const String8 &title); 2360#ifdef DEBUG_CPU_USAGE 2361private: 2362 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2363 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2364 2365 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2366 2367 int mCpuNum; // thread's current CPU number 2368 int mCpukHz; // frequency of thread's current CPU in kHz 2369#endif 2370}; 2371 2372CpuStats::CpuStats() 2373#ifdef DEBUG_CPU_USAGE 2374 : mCpuNum(-1), mCpukHz(-1) 2375#endif 2376{ 2377} 2378 2379void CpuStats::sample(const String8 &title) { 2380#ifdef DEBUG_CPU_USAGE 2381 // get current thread's delta CPU time in wall clock ns 2382 double wcNs; 2383 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2384 2385 // record sample for wall clock statistics 2386 if (valid) { 2387 mWcStats.sample(wcNs); 2388 } 2389 2390 // get the current CPU number 2391 int cpuNum = sched_getcpu(); 2392 2393 // get the current CPU frequency in kHz 2394 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2395 2396 // check if either CPU number or frequency changed 2397 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2398 mCpuNum = cpuNum; 2399 mCpukHz = cpukHz; 2400 // ignore sample for purposes of cycles 2401 valid = false; 2402 } 2403 2404 // if no change in CPU number or frequency, then record sample for cycle statistics 2405 if (valid && mCpukHz > 0) { 2406 double cycles = wcNs * cpukHz * 0.000001; 2407 mHzStats.sample(cycles); 2408 } 2409 2410 unsigned n = mWcStats.n(); 2411 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2412 if ((n & 127) == 1) { 2413 long long elapsed = mCpuUsage.elapsed(); 2414 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2415 double perLoop = elapsed / (double) n; 2416 double perLoop100 = perLoop * 0.01; 2417 double perLoop1k = perLoop * 0.001; 2418 double mean = mWcStats.mean(); 2419 double stddev = mWcStats.stddev(); 2420 double minimum = mWcStats.minimum(); 2421 double maximum = mWcStats.maximum(); 2422 double meanCycles = mHzStats.mean(); 2423 double stddevCycles = mHzStats.stddev(); 2424 double minCycles = mHzStats.minimum(); 2425 double maxCycles = mHzStats.maximum(); 2426 mCpuUsage.resetElapsed(); 2427 mWcStats.reset(); 2428 mHzStats.reset(); 2429 ALOGD("CPU usage for %s over past %.1f secs\n" 2430 " (%u mixer loops at %.1f mean ms per loop):\n" 2431 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2432 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2433 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2434 title.string(), 2435 elapsed * .000000001, n, perLoop * .000001, 2436 mean * .001, 2437 stddev * .001, 2438 minimum * .001, 2439 maximum * .001, 2440 mean / perLoop100, 2441 stddev / perLoop100, 2442 minimum / perLoop100, 2443 maximum / perLoop100, 2444 meanCycles / perLoop1k, 2445 stddevCycles / perLoop1k, 2446 minCycles / perLoop1k, 2447 maxCycles / perLoop1k); 2448 2449 } 2450 } 2451#endif 2452}; 2453 2454void AudioFlinger::PlaybackThread::checkSilentMode_l() 2455{ 2456 if (!mMasterMute) { 2457 char value[PROPERTY_VALUE_MAX]; 2458 if (property_get("ro.audio.silent", value, "0") > 0) { 2459 char *endptr; 2460 unsigned long ul = strtoul(value, &endptr, 0); 2461 if (*endptr == '\0' && ul != 0) { 2462 ALOGD("Silence is golden"); 2463 // The setprop command will not allow a property to be changed after 2464 // the first time it is set, so we don't have to worry about un-muting. 2465 setMasterMute_l(true); 2466 } 2467 } 2468 } 2469} 2470 2471bool AudioFlinger::PlaybackThread::threadLoop() 2472{ 2473 Vector< sp<Track> > tracksToRemove; 2474 2475 standbyTime = systemTime(); 2476 2477 // MIXER 2478 nsecs_t lastWarning = 0; 2479 2480 // DUPLICATING 2481 // FIXME could this be made local to while loop? 2482 writeFrames = 0; 2483 2484 cacheParameters_l(); 2485 sleepTime = idleSleepTime; 2486 2487 if (mType == MIXER) { 2488 sleepTimeShift = 0; 2489 } 2490 2491 CpuStats cpuStats; 2492 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2493 2494 acquireWakeLock(); 2495 2496 while (!exitPending()) 2497 { 2498 cpuStats.sample(myName); 2499 2500 Vector< sp<EffectChain> > effectChains; 2501 2502 processConfigEvents(); 2503 2504 { // scope for mLock 2505 2506 Mutex::Autolock _l(mLock); 2507 2508 if (checkForNewParameters_l()) { 2509 cacheParameters_l(); 2510 } 2511 2512 saveOutputTracks(); 2513 2514 // put audio hardware into standby after short delay 2515 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2516 isSuspended())) { 2517 if (!mStandby) { 2518 2519 threadLoop_standby(); 2520 2521 mStandby = true; 2522 mBytesWritten = 0; 2523 } 2524 2525 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2526 // we're about to wait, flush the binder command buffer 2527 IPCThreadState::self()->flushCommands(); 2528 2529 clearOutputTracks(); 2530 2531 if (exitPending()) break; 2532 2533 releaseWakeLock_l(); 2534 // wait until we have something to do... 2535 ALOGV("%s going to sleep", myName.string()); 2536 mWaitWorkCV.wait(mLock); 2537 ALOGV("%s waking up", myName.string()); 2538 acquireWakeLock_l(); 2539 2540 mMixerStatus = MIXER_IDLE; 2541 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2542 2543 checkSilentMode_l(); 2544 2545 standbyTime = systemTime() + standbyDelay; 2546 sleepTime = idleSleepTime; 2547 if (mType == MIXER) { 2548 sleepTimeShift = 0; 2549 } 2550 2551 continue; 2552 } 2553 } 2554 2555 // mMixerStatusIgnoringFastTracks is also updated internally 2556 mMixerStatus = prepareTracks_l(&tracksToRemove); 2557 2558 // prevent any changes in effect chain list and in each effect chain 2559 // during mixing and effect process as the audio buffers could be deleted 2560 // or modified if an effect is created or deleted 2561 lockEffectChains_l(effectChains); 2562 } 2563 2564 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2565 threadLoop_mix(); 2566 } else { 2567 threadLoop_sleepTime(); 2568 } 2569 2570 if (isSuspended()) { 2571 sleepTime = suspendSleepTimeUs(); 2572 } 2573 2574 // only process effects if we're going to write 2575 if (sleepTime == 0) { 2576 for (size_t i = 0; i < effectChains.size(); i ++) { 2577 effectChains[i]->process_l(); 2578 } 2579 } 2580 2581 // enable changes in effect chain 2582 unlockEffectChains(effectChains); 2583 2584 // sleepTime == 0 means we must write to audio hardware 2585 if (sleepTime == 0) { 2586 2587 threadLoop_write(); 2588 2589if (mType == MIXER) { 2590 // write blocked detection 2591 nsecs_t now = systemTime(); 2592 nsecs_t delta = now - mLastWriteTime; 2593 if (!mStandby && delta > maxPeriod) { 2594 mNumDelayedWrites++; 2595 if ((now - lastWarning) > kWarningThrottleNs) { 2596#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2597 ScopedTrace st(ATRACE_TAG, "underrun"); 2598#endif 2599 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2600 ns2ms(delta), mNumDelayedWrites, this); 2601 lastWarning = now; 2602 } 2603 } 2604} 2605 2606 mStandby = false; 2607 } else { 2608 usleep(sleepTime); 2609 } 2610 2611 // Finally let go of removed track(s), without the lock held 2612 // since we can't guarantee the destructors won't acquire that 2613 // same lock. This will also mutate and push a new fast mixer state. 2614 threadLoop_removeTracks(tracksToRemove); 2615 tracksToRemove.clear(); 2616 2617 // FIXME I don't understand the need for this here; 2618 // it was in the original code but maybe the 2619 // assignment in saveOutputTracks() makes this unnecessary? 2620 clearOutputTracks(); 2621 2622 // Effect chains will be actually deleted here if they were removed from 2623 // mEffectChains list during mixing or effects processing 2624 effectChains.clear(); 2625 2626 // FIXME Note that the above .clear() is no longer necessary since effectChains 2627 // is now local to this block, but will keep it for now (at least until merge done). 2628 } 2629 2630 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2631 if (mType == MIXER || mType == DIRECT) { 2632 // put output stream into standby mode 2633 if (!mStandby) { 2634 mOutput->stream->common.standby(&mOutput->stream->common); 2635 } 2636 } 2637 2638 releaseWakeLock(); 2639 2640 ALOGV("Thread %p type %d exiting", this, mType); 2641 return false; 2642} 2643 2644void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2645{ 2646 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2647} 2648 2649void AudioFlinger::MixerThread::threadLoop_write() 2650{ 2651 // FIXME we should only do one push per cycle; confirm this is true 2652 // Start the fast mixer if it's not already running 2653 if (mFastMixer != NULL) { 2654 FastMixerStateQueue *sq = mFastMixer->sq(); 2655 FastMixerState *state = sq->begin(); 2656 if (state->mCommand != FastMixerState::MIX_WRITE && 2657 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2658 if (state->mCommand == FastMixerState::COLD_IDLE) { 2659 int32_t old = android_atomic_inc(&mFastMixerFutex); 2660 if (old == -1) { 2661 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2662 } 2663 if (mAudioWatchdog != 0) { 2664 mAudioWatchdog->resume(); 2665 } 2666 } 2667 state->mCommand = FastMixerState::MIX_WRITE; 2668 sq->end(); 2669 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2670 if (kUseFastMixer == FastMixer_Dynamic) { 2671 mNormalSink = mPipeSink; 2672 } 2673 } else { 2674 sq->end(false /*didModify*/); 2675 } 2676 } 2677 PlaybackThread::threadLoop_write(); 2678} 2679 2680// shared by MIXER and DIRECT, overridden by DUPLICATING 2681void AudioFlinger::PlaybackThread::threadLoop_write() 2682{ 2683 // FIXME rewrite to reduce number of system calls 2684 mLastWriteTime = systemTime(); 2685 mInWrite = true; 2686 int bytesWritten; 2687 2688 // If an NBAIO sink is present, use it to write the normal mixer's submix 2689 if (mNormalSink != 0) { 2690#define mBitShift 2 // FIXME 2691 size_t count = mixBufferSize >> mBitShift; 2692#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2693 Tracer::traceBegin(ATRACE_TAG, "write"); 2694#endif 2695 // update the setpoint when gScreenState changes 2696 uint32_t screenState = gScreenState; 2697 if (screenState != mScreenState) { 2698 mScreenState = screenState; 2699 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2700 if (pipe != NULL) { 2701 pipe->setAvgFrames((mScreenState & 1) ? 2702 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2703 } 2704 } 2705 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2706#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2707 Tracer::traceEnd(ATRACE_TAG); 2708#endif 2709 if (framesWritten > 0) { 2710 bytesWritten = framesWritten << mBitShift; 2711 } else { 2712 bytesWritten = framesWritten; 2713 } 2714 // otherwise use the HAL / AudioStreamOut directly 2715 } else { 2716 // Direct output thread. 2717 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2718 } 2719 2720 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2721 mNumWrites++; 2722 mInWrite = false; 2723} 2724 2725void AudioFlinger::MixerThread::threadLoop_standby() 2726{ 2727 // Idle the fast mixer if it's currently running 2728 if (mFastMixer != NULL) { 2729 FastMixerStateQueue *sq = mFastMixer->sq(); 2730 FastMixerState *state = sq->begin(); 2731 if (!(state->mCommand & FastMixerState::IDLE)) { 2732 state->mCommand = FastMixerState::COLD_IDLE; 2733 state->mColdFutexAddr = &mFastMixerFutex; 2734 state->mColdGen++; 2735 mFastMixerFutex = 0; 2736 sq->end(); 2737 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2738 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2739 if (kUseFastMixer == FastMixer_Dynamic) { 2740 mNormalSink = mOutputSink; 2741 } 2742 if (mAudioWatchdog != 0) { 2743 mAudioWatchdog->pause(); 2744 } 2745 } else { 2746 sq->end(false /*didModify*/); 2747 } 2748 } 2749 PlaybackThread::threadLoop_standby(); 2750} 2751 2752// shared by MIXER and DIRECT, overridden by DUPLICATING 2753void AudioFlinger::PlaybackThread::threadLoop_standby() 2754{ 2755 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2756 mOutput->stream->common.standby(&mOutput->stream->common); 2757} 2758 2759void AudioFlinger::MixerThread::threadLoop_mix() 2760{ 2761 // obtain the presentation timestamp of the next output buffer 2762 int64_t pts; 2763 status_t status = INVALID_OPERATION; 2764 2765 if (NULL != mOutput->stream->get_next_write_timestamp) { 2766 status = mOutput->stream->get_next_write_timestamp( 2767 mOutput->stream, &pts); 2768 } 2769 2770 if (status != NO_ERROR) { 2771 pts = AudioBufferProvider::kInvalidPTS; 2772 } 2773 2774 // mix buffers... 2775 mAudioMixer->process(pts); 2776 // increase sleep time progressively when application underrun condition clears. 2777 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2778 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2779 // such that we would underrun the audio HAL. 2780 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2781 sleepTimeShift--; 2782 } 2783 sleepTime = 0; 2784 standbyTime = systemTime() + standbyDelay; 2785 //TODO: delay standby when effects have a tail 2786} 2787 2788void AudioFlinger::MixerThread::threadLoop_sleepTime() 2789{ 2790 // If no tracks are ready, sleep once for the duration of an output 2791 // buffer size, then write 0s to the output 2792 if (sleepTime == 0) { 2793 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2794 sleepTime = activeSleepTime >> sleepTimeShift; 2795 if (sleepTime < kMinThreadSleepTimeUs) { 2796 sleepTime = kMinThreadSleepTimeUs; 2797 } 2798 // reduce sleep time in case of consecutive application underruns to avoid 2799 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2800 // duration we would end up writing less data than needed by the audio HAL if 2801 // the condition persists. 2802 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2803 sleepTimeShift++; 2804 } 2805 } else { 2806 sleepTime = idleSleepTime; 2807 } 2808 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2809 memset (mMixBuffer, 0, mixBufferSize); 2810 sleepTime = 0; 2811 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2812 } 2813 // TODO add standby time extension fct of effect tail 2814} 2815 2816// prepareTracks_l() must be called with ThreadBase::mLock held 2817AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2818 Vector< sp<Track> > *tracksToRemove) 2819{ 2820 2821 mixer_state mixerStatus = MIXER_IDLE; 2822 // find out which tracks need to be processed 2823 size_t count = mActiveTracks.size(); 2824 size_t mixedTracks = 0; 2825 size_t tracksWithEffect = 0; 2826 // counts only _active_ fast tracks 2827 size_t fastTracks = 0; 2828 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2829 2830 float masterVolume = mMasterVolume; 2831 bool masterMute = mMasterMute; 2832 2833 if (masterMute) { 2834 masterVolume = 0; 2835 } 2836 // Delegate master volume control to effect in output mix effect chain if needed 2837 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2838 if (chain != 0) { 2839 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2840 chain->setVolume_l(&v, &v); 2841 masterVolume = (float)((v + (1 << 23)) >> 24); 2842 chain.clear(); 2843 } 2844 2845 // prepare a new state to push 2846 FastMixerStateQueue *sq = NULL; 2847 FastMixerState *state = NULL; 2848 bool didModify = false; 2849 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2850 if (mFastMixer != NULL) { 2851 sq = mFastMixer->sq(); 2852 state = sq->begin(); 2853 } 2854 2855 for (size_t i=0 ; i<count ; i++) { 2856 sp<Track> t = mActiveTracks[i].promote(); 2857 if (t == 0) continue; 2858 2859 // this const just means the local variable doesn't change 2860 Track* const track = t.get(); 2861 2862 // process fast tracks 2863 if (track->isFastTrack()) { 2864 2865 // It's theoretically possible (though unlikely) for a fast track to be created 2866 // and then removed within the same normal mix cycle. This is not a problem, as 2867 // the track never becomes active so it's fast mixer slot is never touched. 2868 // The converse, of removing an (active) track and then creating a new track 2869 // at the identical fast mixer slot within the same normal mix cycle, 2870 // is impossible because the slot isn't marked available until the end of each cycle. 2871 int j = track->mFastIndex; 2872 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2873 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2874 FastTrack *fastTrack = &state->mFastTracks[j]; 2875 2876 // Determine whether the track is currently in underrun condition, 2877 // and whether it had a recent underrun. 2878 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2879 FastTrackUnderruns underruns = ftDump->mUnderruns; 2880 uint32_t recentFull = (underruns.mBitFields.mFull - 2881 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2882 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2883 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2884 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2885 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2886 uint32_t recentUnderruns = recentPartial + recentEmpty; 2887 track->mObservedUnderruns = underruns; 2888 // don't count underruns that occur while stopping or pausing 2889 // or stopped which can occur when flush() is called while active 2890 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2891 track->mUnderrunCount += recentUnderruns; 2892 } 2893 2894 // This is similar to the state machine for normal tracks, 2895 // with a few modifications for fast tracks. 2896 bool isActive = true; 2897 switch (track->mState) { 2898 case TrackBase::STOPPING_1: 2899 // track stays active in STOPPING_1 state until first underrun 2900 if (recentUnderruns > 0) { 2901 track->mState = TrackBase::STOPPING_2; 2902 } 2903 break; 2904 case TrackBase::PAUSING: 2905 // ramp down is not yet implemented 2906 track->setPaused(); 2907 break; 2908 case TrackBase::RESUMING: 2909 // ramp up is not yet implemented 2910 track->mState = TrackBase::ACTIVE; 2911 break; 2912 case TrackBase::ACTIVE: 2913 if (recentFull > 0 || recentPartial > 0) { 2914 // track has provided at least some frames recently: reset retry count 2915 track->mRetryCount = kMaxTrackRetries; 2916 } 2917 if (recentUnderruns == 0) { 2918 // no recent underruns: stay active 2919 break; 2920 } 2921 // there has recently been an underrun of some kind 2922 if (track->sharedBuffer() == 0) { 2923 // were any of the recent underruns "empty" (no frames available)? 2924 if (recentEmpty == 0) { 2925 // no, then ignore the partial underruns as they are allowed indefinitely 2926 break; 2927 } 2928 // there has recently been an "empty" underrun: decrement the retry counter 2929 if (--(track->mRetryCount) > 0) { 2930 break; 2931 } 2932 // indicate to client process that the track was disabled because of underrun; 2933 // it will then automatically call start() when data is available 2934 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2935 // remove from active list, but state remains ACTIVE [confusing but true] 2936 isActive = false; 2937 break; 2938 } 2939 // fall through 2940 case TrackBase::STOPPING_2: 2941 case TrackBase::PAUSED: 2942 case TrackBase::TERMINATED: 2943 case TrackBase::STOPPED: 2944 case TrackBase::FLUSHED: // flush() while active 2945 // Check for presentation complete if track is inactive 2946 // We have consumed all the buffers of this track. 2947 // This would be incomplete if we auto-paused on underrun 2948 { 2949 size_t audioHALFrames = 2950 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2951 size_t framesWritten = 2952 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2953 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2954 // track stays in active list until presentation is complete 2955 break; 2956 } 2957 } 2958 if (track->isStopping_2()) { 2959 track->mState = TrackBase::STOPPED; 2960 } 2961 if (track->isStopped()) { 2962 // Can't reset directly, as fast mixer is still polling this track 2963 // track->reset(); 2964 // So instead mark this track as needing to be reset after push with ack 2965 resetMask |= 1 << i; 2966 } 2967 isActive = false; 2968 break; 2969 case TrackBase::IDLE: 2970 default: 2971 LOG_FATAL("unexpected track state %d", track->mState); 2972 } 2973 2974 if (isActive) { 2975 // was it previously inactive? 2976 if (!(state->mTrackMask & (1 << j))) { 2977 ExtendedAudioBufferProvider *eabp = track; 2978 VolumeProvider *vp = track; 2979 fastTrack->mBufferProvider = eabp; 2980 fastTrack->mVolumeProvider = vp; 2981 fastTrack->mSampleRate = track->mSampleRate; 2982 fastTrack->mChannelMask = track->mChannelMask; 2983 fastTrack->mGeneration++; 2984 state->mTrackMask |= 1 << j; 2985 didModify = true; 2986 // no acknowledgement required for newly active tracks 2987 } 2988 // cache the combined master volume and stream type volume for fast mixer; this 2989 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2990 track->mCachedVolume = track->isMuted() ? 2991 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2992 ++fastTracks; 2993 } else { 2994 // was it previously active? 2995 if (state->mTrackMask & (1 << j)) { 2996 fastTrack->mBufferProvider = NULL; 2997 fastTrack->mGeneration++; 2998 state->mTrackMask &= ~(1 << j); 2999 didModify = true; 3000 // If any fast tracks were removed, we must wait for acknowledgement 3001 // because we're about to decrement the last sp<> on those tracks. 3002 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3003 } else { 3004 LOG_FATAL("fast track %d should have been active", j); 3005 } 3006 tracksToRemove->add(track); 3007 // Avoids a misleading display in dumpsys 3008 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3009 } 3010 continue; 3011 } 3012 3013 { // local variable scope to avoid goto warning 3014 3015 audio_track_cblk_t* cblk = track->cblk(); 3016 3017 // The first time a track is added we wait 3018 // for all its buffers to be filled before processing it 3019 int name = track->name(); 3020 // make sure that we have enough frames to mix one full buffer. 3021 // enforce this condition only once to enable draining the buffer in case the client 3022 // app does not call stop() and relies on underrun to stop: 3023 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3024 // during last round 3025 uint32_t minFrames = 1; 3026 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3027 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3028 if (t->sampleRate() == (int)mSampleRate) { 3029 minFrames = mNormalFrameCount; 3030 } else { 3031 // +1 for rounding and +1 for additional sample needed for interpolation 3032 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3033 // add frames already consumed but not yet released by the resampler 3034 // because cblk->framesReady() will include these frames 3035 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3036 // the minimum track buffer size is normally twice the number of frames necessary 3037 // to fill one buffer and the resampler should not leave more than one buffer worth 3038 // of unreleased frames after each pass, but just in case... 3039 ALOG_ASSERT(minFrames <= cblk->frameCount); 3040 } 3041 } 3042 if ((track->framesReady() >= minFrames) && track->isReady() && 3043 !track->isPaused() && !track->isTerminated()) 3044 { 3045 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3046 3047 mixedTracks++; 3048 3049 // track->mainBuffer() != mMixBuffer means there is an effect chain 3050 // connected to the track 3051 chain.clear(); 3052 if (track->mainBuffer() != mMixBuffer) { 3053 chain = getEffectChain_l(track->sessionId()); 3054 // Delegate volume control to effect in track effect chain if needed 3055 if (chain != 0) { 3056 tracksWithEffect++; 3057 } else { 3058 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3059 name, track->sessionId()); 3060 } 3061 } 3062 3063 3064 int param = AudioMixer::VOLUME; 3065 if (track->mFillingUpStatus == Track::FS_FILLED) { 3066 // no ramp for the first volume setting 3067 track->mFillingUpStatus = Track::FS_ACTIVE; 3068 if (track->mState == TrackBase::RESUMING) { 3069 track->mState = TrackBase::ACTIVE; 3070 param = AudioMixer::RAMP_VOLUME; 3071 } 3072 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3073 } else if (cblk->server != 0) { 3074 // If the track is stopped before the first frame was mixed, 3075 // do not apply ramp 3076 param = AudioMixer::RAMP_VOLUME; 3077 } 3078 3079 // compute volume for this track 3080 uint32_t vl, vr, va; 3081 if (track->isMuted() || track->isPausing() || 3082 mStreamTypes[track->streamType()].mute) { 3083 vl = vr = va = 0; 3084 if (track->isPausing()) { 3085 track->setPaused(); 3086 } 3087 } else { 3088 3089 // read original volumes with volume control 3090 float typeVolume = mStreamTypes[track->streamType()].volume; 3091 float v = masterVolume * typeVolume; 3092 uint32_t vlr = cblk->getVolumeLR(); 3093 vl = vlr & 0xFFFF; 3094 vr = vlr >> 16; 3095 // track volumes come from shared memory, so can't be trusted and must be clamped 3096 if (vl > MAX_GAIN_INT) { 3097 ALOGV("Track left volume out of range: %04X", vl); 3098 vl = MAX_GAIN_INT; 3099 } 3100 if (vr > MAX_GAIN_INT) { 3101 ALOGV("Track right volume out of range: %04X", vr); 3102 vr = MAX_GAIN_INT; 3103 } 3104 // now apply the master volume and stream type volume 3105 vl = (uint32_t)(v * vl) << 12; 3106 vr = (uint32_t)(v * vr) << 12; 3107 // assuming master volume and stream type volume each go up to 1.0, 3108 // vl and vr are now in 8.24 format 3109 3110 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3111 // send level comes from shared memory and so may be corrupt 3112 if (sendLevel > MAX_GAIN_INT) { 3113 ALOGV("Track send level out of range: %04X", sendLevel); 3114 sendLevel = MAX_GAIN_INT; 3115 } 3116 va = (uint32_t)(v * sendLevel); 3117 } 3118 // Delegate volume control to effect in track effect chain if needed 3119 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3120 // Do not ramp volume if volume is controlled by effect 3121 param = AudioMixer::VOLUME; 3122 track->mHasVolumeController = true; 3123 } else { 3124 // force no volume ramp when volume controller was just disabled or removed 3125 // from effect chain to avoid volume spike 3126 if (track->mHasVolumeController) { 3127 param = AudioMixer::VOLUME; 3128 } 3129 track->mHasVolumeController = false; 3130 } 3131 3132 // Convert volumes from 8.24 to 4.12 format 3133 // This additional clamping is needed in case chain->setVolume_l() overshot 3134 vl = (vl + (1 << 11)) >> 12; 3135 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3136 vr = (vr + (1 << 11)) >> 12; 3137 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3138 3139 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3140 3141 // XXX: these things DON'T need to be done each time 3142 mAudioMixer->setBufferProvider(name, track); 3143 mAudioMixer->enable(name); 3144 3145 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3146 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3147 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3148 mAudioMixer->setParameter( 3149 name, 3150 AudioMixer::TRACK, 3151 AudioMixer::FORMAT, (void *)track->format()); 3152 mAudioMixer->setParameter( 3153 name, 3154 AudioMixer::TRACK, 3155 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3156 mAudioMixer->setParameter( 3157 name, 3158 AudioMixer::RESAMPLE, 3159 AudioMixer::SAMPLE_RATE, 3160 (void *)(cblk->sampleRate)); 3161 mAudioMixer->setParameter( 3162 name, 3163 AudioMixer::TRACK, 3164 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3165 mAudioMixer->setParameter( 3166 name, 3167 AudioMixer::TRACK, 3168 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3169 3170 // reset retry count 3171 track->mRetryCount = kMaxTrackRetries; 3172 3173 // If one track is ready, set the mixer ready if: 3174 // - the mixer was not ready during previous round OR 3175 // - no other track is not ready 3176 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3177 mixerStatus != MIXER_TRACKS_ENABLED) { 3178 mixerStatus = MIXER_TRACKS_READY; 3179 } 3180 } else { 3181 // clear effect chain input buffer if an active track underruns to avoid sending 3182 // previous audio buffer again to effects 3183 chain = getEffectChain_l(track->sessionId()); 3184 if (chain != 0) { 3185 chain->clearInputBuffer(); 3186 } 3187 3188 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3189 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3190 track->isStopped() || track->isPaused()) { 3191 // We have consumed all the buffers of this track. 3192 // Remove it from the list of active tracks. 3193 // TODO: use actual buffer filling status instead of latency when available from 3194 // audio HAL 3195 size_t audioHALFrames = 3196 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3197 size_t framesWritten = 3198 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3199 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3200 if (track->isStopped()) { 3201 track->reset(); 3202 } 3203 tracksToRemove->add(track); 3204 } 3205 } else { 3206 track->mUnderrunCount++; 3207 // No buffers for this track. Give it a few chances to 3208 // fill a buffer, then remove it from active list. 3209 if (--(track->mRetryCount) <= 0) { 3210 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3211 tracksToRemove->add(track); 3212 // indicate to client process that the track was disabled because of underrun; 3213 // it will then automatically call start() when data is available 3214 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3215 // If one track is not ready, mark the mixer also not ready if: 3216 // - the mixer was ready during previous round OR 3217 // - no other track is ready 3218 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3219 mixerStatus != MIXER_TRACKS_READY) { 3220 mixerStatus = MIXER_TRACKS_ENABLED; 3221 } 3222 } 3223 mAudioMixer->disable(name); 3224 } 3225 3226 } // local variable scope to avoid goto warning 3227track_is_ready: ; 3228 3229 } 3230 3231 // Push the new FastMixer state if necessary 3232 bool pauseAudioWatchdog = false; 3233 if (didModify) { 3234 state->mFastTracksGen++; 3235 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3236 if (kUseFastMixer == FastMixer_Dynamic && 3237 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3238 state->mCommand = FastMixerState::COLD_IDLE; 3239 state->mColdFutexAddr = &mFastMixerFutex; 3240 state->mColdGen++; 3241 mFastMixerFutex = 0; 3242 if (kUseFastMixer == FastMixer_Dynamic) { 3243 mNormalSink = mOutputSink; 3244 } 3245 // If we go into cold idle, need to wait for acknowledgement 3246 // so that fast mixer stops doing I/O. 3247 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3248 pauseAudioWatchdog = true; 3249 } 3250 sq->end(); 3251 } 3252 if (sq != NULL) { 3253 sq->end(didModify); 3254 sq->push(block); 3255 } 3256 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3257 mAudioWatchdog->pause(); 3258 } 3259 3260 // Now perform the deferred reset on fast tracks that have stopped 3261 while (resetMask != 0) { 3262 size_t i = __builtin_ctz(resetMask); 3263 ALOG_ASSERT(i < count); 3264 resetMask &= ~(1 << i); 3265 sp<Track> t = mActiveTracks[i].promote(); 3266 if (t == 0) continue; 3267 Track* track = t.get(); 3268 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3269 track->reset(); 3270 } 3271 3272 // remove all the tracks that need to be... 3273 count = tracksToRemove->size(); 3274 if (CC_UNLIKELY(count)) { 3275 for (size_t i=0 ; i<count ; i++) { 3276 const sp<Track>& track = tracksToRemove->itemAt(i); 3277 mActiveTracks.remove(track); 3278 if (track->mainBuffer() != mMixBuffer) { 3279 chain = getEffectChain_l(track->sessionId()); 3280 if (chain != 0) { 3281 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3282 chain->decActiveTrackCnt(); 3283 } 3284 } 3285 if (track->isTerminated()) { 3286 removeTrack_l(track); 3287 } 3288 } 3289 } 3290 3291 // mix buffer must be cleared if all tracks are connected to an 3292 // effect chain as in this case the mixer will not write to 3293 // mix buffer and track effects will accumulate into it 3294 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3295 // FIXME as a performance optimization, should remember previous zero status 3296 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3297 } 3298 3299 // if any fast tracks, then status is ready 3300 mMixerStatusIgnoringFastTracks = mixerStatus; 3301 if (fastTracks > 0) { 3302 mixerStatus = MIXER_TRACKS_READY; 3303 } 3304 return mixerStatus; 3305} 3306 3307/* 3308The derived values that are cached: 3309 - mixBufferSize from frame count * frame size 3310 - activeSleepTime from activeSleepTimeUs() 3311 - idleSleepTime from idleSleepTimeUs() 3312 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3313 - maxPeriod from frame count and sample rate (MIXER only) 3314 3315The parameters that affect these derived values are: 3316 - frame count 3317 - frame size 3318 - sample rate 3319 - device type: A2DP or not 3320 - device latency 3321 - format: PCM or not 3322 - active sleep time 3323 - idle sleep time 3324*/ 3325 3326void AudioFlinger::PlaybackThread::cacheParameters_l() 3327{ 3328 mixBufferSize = mNormalFrameCount * mFrameSize; 3329 activeSleepTime = activeSleepTimeUs(); 3330 idleSleepTime = idleSleepTimeUs(); 3331} 3332 3333void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3334{ 3335 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3336 this, streamType, mTracks.size()); 3337 Mutex::Autolock _l(mLock); 3338 3339 size_t size = mTracks.size(); 3340 for (size_t i = 0; i < size; i++) { 3341 sp<Track> t = mTracks[i]; 3342 if (t->streamType() == streamType) { 3343 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3344 t->mCblk->cv.signal(); 3345 } 3346 } 3347} 3348 3349// getTrackName_l() must be called with ThreadBase::mLock held 3350int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3351{ 3352 return mAudioMixer->getTrackName(channelMask); 3353} 3354 3355// deleteTrackName_l() must be called with ThreadBase::mLock held 3356void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3357{ 3358 ALOGV("remove track (%d) and delete from mixer", name); 3359 mAudioMixer->deleteTrackName(name); 3360} 3361 3362// checkForNewParameters_l() must be called with ThreadBase::mLock held 3363bool AudioFlinger::MixerThread::checkForNewParameters_l() 3364{ 3365 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3366 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3367 bool reconfig = false; 3368 3369 while (!mNewParameters.isEmpty()) { 3370 3371 if (mFastMixer != NULL) { 3372 FastMixerStateQueue *sq = mFastMixer->sq(); 3373 FastMixerState *state = sq->begin(); 3374 if (!(state->mCommand & FastMixerState::IDLE)) { 3375 previousCommand = state->mCommand; 3376 state->mCommand = FastMixerState::HOT_IDLE; 3377 sq->end(); 3378 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3379 } else { 3380 sq->end(false /*didModify*/); 3381 } 3382 } 3383 3384 status_t status = NO_ERROR; 3385 String8 keyValuePair = mNewParameters[0]; 3386 AudioParameter param = AudioParameter(keyValuePair); 3387 int value; 3388 3389 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3390 reconfig = true; 3391 } 3392 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3393 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3394 status = BAD_VALUE; 3395 } else { 3396 reconfig = true; 3397 } 3398 } 3399 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3400 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3401 status = BAD_VALUE; 3402 } else { 3403 reconfig = true; 3404 } 3405 } 3406 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3407 // do not accept frame count changes if tracks are open as the track buffer 3408 // size depends on frame count and correct behavior would not be guaranteed 3409 // if frame count is changed after track creation 3410 if (!mTracks.isEmpty()) { 3411 status = INVALID_OPERATION; 3412 } else { 3413 reconfig = true; 3414 } 3415 } 3416 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3417#ifdef ADD_BATTERY_DATA 3418 // when changing the audio output device, call addBatteryData to notify 3419 // the change 3420 if ((int)mDevice != value) { 3421 uint32_t params = 0; 3422 // check whether speaker is on 3423 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3424 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3425 } 3426 3427 int deviceWithoutSpeaker 3428 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3429 // check if any other device (except speaker) is on 3430 if (value & deviceWithoutSpeaker ) { 3431 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3432 } 3433 3434 if (params != 0) { 3435 addBatteryData(params); 3436 } 3437 } 3438#endif 3439 3440 // forward device change to effects that have requested to be 3441 // aware of attached audio device. 3442 mDevice = (audio_devices_t) value; 3443 for (size_t i = 0; i < mEffectChains.size(); i++) { 3444 mEffectChains[i]->setDevice_l(mDevice); 3445 } 3446 } 3447 3448 if (status == NO_ERROR) { 3449 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3450 keyValuePair.string()); 3451 if (!mStandby && status == INVALID_OPERATION) { 3452 mOutput->stream->common.standby(&mOutput->stream->common); 3453 mStandby = true; 3454 mBytesWritten = 0; 3455 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3456 keyValuePair.string()); 3457 } 3458 if (status == NO_ERROR && reconfig) { 3459 delete mAudioMixer; 3460 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3461 mAudioMixer = NULL; 3462 readOutputParameters(); 3463 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3464 for (size_t i = 0; i < mTracks.size() ; i++) { 3465 int name = getTrackName_l(mTracks[i]->mChannelMask); 3466 if (name < 0) break; 3467 mTracks[i]->mName = name; 3468 // limit track sample rate to 2 x new output sample rate 3469 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3470 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3471 } 3472 } 3473 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3474 } 3475 } 3476 3477 mNewParameters.removeAt(0); 3478 3479 mParamStatus = status; 3480 mParamCond.signal(); 3481 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3482 // already timed out waiting for the status and will never signal the condition. 3483 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3484 } 3485 3486 if (!(previousCommand & FastMixerState::IDLE)) { 3487 ALOG_ASSERT(mFastMixer != NULL); 3488 FastMixerStateQueue *sq = mFastMixer->sq(); 3489 FastMixerState *state = sq->begin(); 3490 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3491 state->mCommand = previousCommand; 3492 sq->end(); 3493 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3494 } 3495 3496 return reconfig; 3497} 3498 3499status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3500{ 3501 const size_t SIZE = 256; 3502 char buffer[SIZE]; 3503 String8 result; 3504 3505 PlaybackThread::dumpInternals(fd, args); 3506 3507 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3508 result.append(buffer); 3509 write(fd, result.string(), result.size()); 3510 3511 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3512 FastMixerDumpState copy = mFastMixerDumpState; 3513 copy.dump(fd); 3514 3515#ifdef STATE_QUEUE_DUMP 3516 // Similar for state queue 3517 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3518 observerCopy.dump(fd); 3519 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3520 mutatorCopy.dump(fd); 3521#endif 3522 3523 // Write the tee output to a .wav file 3524 NBAIO_Source *teeSource = mTeeSource.get(); 3525 if (teeSource != NULL) { 3526 char teePath[64]; 3527 struct timeval tv; 3528 gettimeofday(&tv, NULL); 3529 struct tm tm; 3530 localtime_r(&tv.tv_sec, &tm); 3531 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3532 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3533 if (teeFd >= 0) { 3534 char wavHeader[44]; 3535 memcpy(wavHeader, 3536 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3537 sizeof(wavHeader)); 3538 NBAIO_Format format = teeSource->format(); 3539 unsigned channelCount = Format_channelCount(format); 3540 ALOG_ASSERT(channelCount <= FCC_2); 3541 unsigned sampleRate = Format_sampleRate(format); 3542 wavHeader[22] = channelCount; // number of channels 3543 wavHeader[24] = sampleRate; // sample rate 3544 wavHeader[25] = sampleRate >> 8; 3545 wavHeader[32] = channelCount * 2; // block alignment 3546 write(teeFd, wavHeader, sizeof(wavHeader)); 3547 size_t total = 0; 3548 bool firstRead = true; 3549 for (;;) { 3550#define TEE_SINK_READ 1024 3551 short buffer[TEE_SINK_READ * FCC_2]; 3552 size_t count = TEE_SINK_READ; 3553 ssize_t actual = teeSource->read(buffer, count); 3554 bool wasFirstRead = firstRead; 3555 firstRead = false; 3556 if (actual <= 0) { 3557 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3558 continue; 3559 } 3560 break; 3561 } 3562 ALOG_ASSERT(actual <= (ssize_t)count); 3563 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3564 total += actual; 3565 } 3566 lseek(teeFd, (off_t) 4, SEEK_SET); 3567 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3568 write(teeFd, &temp, sizeof(temp)); 3569 lseek(teeFd, (off_t) 40, SEEK_SET); 3570 temp = total * channelCount * sizeof(short); 3571 write(teeFd, &temp, sizeof(temp)); 3572 close(teeFd); 3573 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3574 } else { 3575 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3576 } 3577 } 3578 3579 if (mAudioWatchdog != 0) { 3580 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3581 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3582 wdCopy.dump(fd); 3583 } 3584 3585 return NO_ERROR; 3586} 3587 3588uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3589{ 3590 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3591} 3592 3593uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3594{ 3595 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3596} 3597 3598void AudioFlinger::MixerThread::cacheParameters_l() 3599{ 3600 PlaybackThread::cacheParameters_l(); 3601 3602 // FIXME: Relaxed timing because of a certain device that can't meet latency 3603 // Should be reduced to 2x after the vendor fixes the driver issue 3604 // increase threshold again due to low power audio mode. The way this warning 3605 // threshold is calculated and its usefulness should be reconsidered anyway. 3606 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3607} 3608 3609// ---------------------------------------------------------------------------- 3610AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3611 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3612 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3613 // mLeftVolFloat, mRightVolFloat 3614{ 3615} 3616 3617AudioFlinger::DirectOutputThread::~DirectOutputThread() 3618{ 3619} 3620 3621AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3622 Vector< sp<Track> > *tracksToRemove 3623) 3624{ 3625 sp<Track> trackToRemove; 3626 3627 mixer_state mixerStatus = MIXER_IDLE; 3628 3629 // find out which tracks need to be processed 3630 if (mActiveTracks.size() != 0) { 3631 sp<Track> t = mActiveTracks[0].promote(); 3632 // The track died recently 3633 if (t == 0) return MIXER_IDLE; 3634 3635 Track* const track = t.get(); 3636 audio_track_cblk_t* cblk = track->cblk(); 3637 3638 // The first time a track is added we wait 3639 // for all its buffers to be filled before processing it 3640 uint32_t minFrames; 3641 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3642 minFrames = mNormalFrameCount; 3643 } else { 3644 minFrames = 1; 3645 } 3646 if ((track->framesReady() >= minFrames) && track->isReady() && 3647 !track->isPaused() && !track->isTerminated()) 3648 { 3649 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3650 3651 if (track->mFillingUpStatus == Track::FS_FILLED) { 3652 track->mFillingUpStatus = Track::FS_ACTIVE; 3653 mLeftVolFloat = mRightVolFloat = 0; 3654 if (track->mState == TrackBase::RESUMING) { 3655 track->mState = TrackBase::ACTIVE; 3656 } 3657 } 3658 3659 // compute volume for this track 3660 float left, right; 3661 if (track->isMuted() || mMasterMute || track->isPausing() || 3662 mStreamTypes[track->streamType()].mute) { 3663 left = right = 0; 3664 if (track->isPausing()) { 3665 track->setPaused(); 3666 } 3667 } else { 3668 float typeVolume = mStreamTypes[track->streamType()].volume; 3669 float v = mMasterVolume * typeVolume; 3670 uint32_t vlr = cblk->getVolumeLR(); 3671 float v_clamped = v * (vlr & 0xFFFF); 3672 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3673 left = v_clamped/MAX_GAIN; 3674 v_clamped = v * (vlr >> 16); 3675 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3676 right = v_clamped/MAX_GAIN; 3677 } 3678 3679 if (left != mLeftVolFloat || right != mRightVolFloat) { 3680 mLeftVolFloat = left; 3681 mRightVolFloat = right; 3682 3683 // Convert volumes from float to 8.24 3684 uint32_t vl = (uint32_t)(left * (1 << 24)); 3685 uint32_t vr = (uint32_t)(right * (1 << 24)); 3686 3687 // Delegate volume control to effect in track effect chain if needed 3688 // only one effect chain can be present on DirectOutputThread, so if 3689 // there is one, the track is connected to it 3690 if (!mEffectChains.isEmpty()) { 3691 // Do not ramp volume if volume is controlled by effect 3692 mEffectChains[0]->setVolume_l(&vl, &vr); 3693 left = (float)vl / (1 << 24); 3694 right = (float)vr / (1 << 24); 3695 } 3696 mOutput->stream->set_volume(mOutput->stream, left, right); 3697 } 3698 3699 // reset retry count 3700 track->mRetryCount = kMaxTrackRetriesDirect; 3701 mActiveTrack = t; 3702 mixerStatus = MIXER_TRACKS_READY; 3703 } else { 3704 // clear effect chain input buffer if an active track underruns to avoid sending 3705 // previous audio buffer again to effects 3706 if (!mEffectChains.isEmpty()) { 3707 mEffectChains[0]->clearInputBuffer(); 3708 } 3709 3710 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3711 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3712 track->isStopped() || track->isPaused()) { 3713 // We have consumed all the buffers of this track. 3714 // Remove it from the list of active tracks. 3715 // TODO: implement behavior for compressed audio 3716 size_t audioHALFrames = 3717 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3718 size_t framesWritten = 3719 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3720 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3721 if (track->isStopped()) { 3722 track->reset(); 3723 } 3724 trackToRemove = track; 3725 } 3726 } else { 3727 // No buffers for this track. Give it a few chances to 3728 // fill a buffer, then remove it from active list. 3729 if (--(track->mRetryCount) <= 0) { 3730 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3731 trackToRemove = track; 3732 } else { 3733 mixerStatus = MIXER_TRACKS_ENABLED; 3734 } 3735 } 3736 } 3737 } 3738 3739 // FIXME merge this with similar code for removing multiple tracks 3740 // remove all the tracks that need to be... 3741 if (CC_UNLIKELY(trackToRemove != 0)) { 3742 tracksToRemove->add(trackToRemove); 3743 mActiveTracks.remove(trackToRemove); 3744 if (!mEffectChains.isEmpty()) { 3745 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3746 trackToRemove->sessionId()); 3747 mEffectChains[0]->decActiveTrackCnt(); 3748 } 3749 if (trackToRemove->isTerminated()) { 3750 removeTrack_l(trackToRemove); 3751 } 3752 } 3753 3754 return mixerStatus; 3755} 3756 3757void AudioFlinger::DirectOutputThread::threadLoop_mix() 3758{ 3759 AudioBufferProvider::Buffer buffer; 3760 size_t frameCount = mFrameCount; 3761 int8_t *curBuf = (int8_t *)mMixBuffer; 3762 // output audio to hardware 3763 while (frameCount) { 3764 buffer.frameCount = frameCount; 3765 mActiveTrack->getNextBuffer(&buffer); 3766 if (CC_UNLIKELY(buffer.raw == NULL)) { 3767 memset(curBuf, 0, frameCount * mFrameSize); 3768 break; 3769 } 3770 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3771 frameCount -= buffer.frameCount; 3772 curBuf += buffer.frameCount * mFrameSize; 3773 mActiveTrack->releaseBuffer(&buffer); 3774 } 3775 sleepTime = 0; 3776 standbyTime = systemTime() + standbyDelay; 3777 mActiveTrack.clear(); 3778 3779} 3780 3781void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3782{ 3783 if (sleepTime == 0) { 3784 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3785 sleepTime = activeSleepTime; 3786 } else { 3787 sleepTime = idleSleepTime; 3788 } 3789 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3790 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3791 sleepTime = 0; 3792 } 3793} 3794 3795// getTrackName_l() must be called with ThreadBase::mLock held 3796int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3797{ 3798 return 0; 3799} 3800 3801// deleteTrackName_l() must be called with ThreadBase::mLock held 3802void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3803{ 3804} 3805 3806// checkForNewParameters_l() must be called with ThreadBase::mLock held 3807bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3808{ 3809 bool reconfig = false; 3810 3811 while (!mNewParameters.isEmpty()) { 3812 status_t status = NO_ERROR; 3813 String8 keyValuePair = mNewParameters[0]; 3814 AudioParameter param = AudioParameter(keyValuePair); 3815 int value; 3816 3817 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3818 // do not accept frame count changes if tracks are open as the track buffer 3819 // size depends on frame count and correct behavior would not be garantied 3820 // if frame count is changed after track creation 3821 if (!mTracks.isEmpty()) { 3822 status = INVALID_OPERATION; 3823 } else { 3824 reconfig = true; 3825 } 3826 } 3827 if (status == NO_ERROR) { 3828 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3829 keyValuePair.string()); 3830 if (!mStandby && status == INVALID_OPERATION) { 3831 mOutput->stream->common.standby(&mOutput->stream->common); 3832 mStandby = true; 3833 mBytesWritten = 0; 3834 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3835 keyValuePair.string()); 3836 } 3837 if (status == NO_ERROR && reconfig) { 3838 readOutputParameters(); 3839 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3840 } 3841 } 3842 3843 mNewParameters.removeAt(0); 3844 3845 mParamStatus = status; 3846 mParamCond.signal(); 3847 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3848 // already timed out waiting for the status and will never signal the condition. 3849 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3850 } 3851 return reconfig; 3852} 3853 3854uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3855{ 3856 uint32_t time; 3857 if (audio_is_linear_pcm(mFormat)) { 3858 time = PlaybackThread::activeSleepTimeUs(); 3859 } else { 3860 time = 10000; 3861 } 3862 return time; 3863} 3864 3865uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3866{ 3867 uint32_t time; 3868 if (audio_is_linear_pcm(mFormat)) { 3869 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3870 } else { 3871 time = 10000; 3872 } 3873 return time; 3874} 3875 3876uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3877{ 3878 uint32_t time; 3879 if (audio_is_linear_pcm(mFormat)) { 3880 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3881 } else { 3882 time = 10000; 3883 } 3884 return time; 3885} 3886 3887void AudioFlinger::DirectOutputThread::cacheParameters_l() 3888{ 3889 PlaybackThread::cacheParameters_l(); 3890 3891 // use shorter standby delay as on normal output to release 3892 // hardware resources as soon as possible 3893 standbyDelay = microseconds(activeSleepTime*2); 3894} 3895 3896// ---------------------------------------------------------------------------- 3897 3898AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3899 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3900 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3901 mWaitTimeMs(UINT_MAX) 3902{ 3903 addOutputTrack(mainThread); 3904} 3905 3906AudioFlinger::DuplicatingThread::~DuplicatingThread() 3907{ 3908 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3909 mOutputTracks[i]->destroy(); 3910 } 3911} 3912 3913void AudioFlinger::DuplicatingThread::threadLoop_mix() 3914{ 3915 // mix buffers... 3916 if (outputsReady(outputTracks)) { 3917 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3918 } else { 3919 memset(mMixBuffer, 0, mixBufferSize); 3920 } 3921 sleepTime = 0; 3922 writeFrames = mNormalFrameCount; 3923 standbyTime = systemTime() + standbyDelay; 3924} 3925 3926void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3927{ 3928 if (sleepTime == 0) { 3929 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3930 sleepTime = activeSleepTime; 3931 } else { 3932 sleepTime = idleSleepTime; 3933 } 3934 } else if (mBytesWritten != 0) { 3935 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3936 writeFrames = mNormalFrameCount; 3937 memset(mMixBuffer, 0, mixBufferSize); 3938 } else { 3939 // flush remaining overflow buffers in output tracks 3940 writeFrames = 0; 3941 } 3942 sleepTime = 0; 3943 } 3944} 3945 3946void AudioFlinger::DuplicatingThread::threadLoop_write() 3947{ 3948 for (size_t i = 0; i < outputTracks.size(); i++) { 3949 outputTracks[i]->write(mMixBuffer, writeFrames); 3950 } 3951 mBytesWritten += mixBufferSize; 3952} 3953 3954void AudioFlinger::DuplicatingThread::threadLoop_standby() 3955{ 3956 // DuplicatingThread implements standby by stopping all tracks 3957 for (size_t i = 0; i < outputTracks.size(); i++) { 3958 outputTracks[i]->stop(); 3959 } 3960} 3961 3962void AudioFlinger::DuplicatingThread::saveOutputTracks() 3963{ 3964 outputTracks = mOutputTracks; 3965} 3966 3967void AudioFlinger::DuplicatingThread::clearOutputTracks() 3968{ 3969 outputTracks.clear(); 3970} 3971 3972void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3973{ 3974 Mutex::Autolock _l(mLock); 3975 // FIXME explain this formula 3976 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3977 OutputTrack *outputTrack = new OutputTrack(thread, 3978 this, 3979 mSampleRate, 3980 mFormat, 3981 mChannelMask, 3982 frameCount); 3983 if (outputTrack->cblk() != NULL) { 3984 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3985 mOutputTracks.add(outputTrack); 3986 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3987 updateWaitTime_l(); 3988 } 3989} 3990 3991void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3992{ 3993 Mutex::Autolock _l(mLock); 3994 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3995 if (mOutputTracks[i]->thread() == thread) { 3996 mOutputTracks[i]->destroy(); 3997 mOutputTracks.removeAt(i); 3998 updateWaitTime_l(); 3999 return; 4000 } 4001 } 4002 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4003} 4004 4005// caller must hold mLock 4006void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4007{ 4008 mWaitTimeMs = UINT_MAX; 4009 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4010 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4011 if (strong != 0) { 4012 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4013 if (waitTimeMs < mWaitTimeMs) { 4014 mWaitTimeMs = waitTimeMs; 4015 } 4016 } 4017 } 4018} 4019 4020 4021bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4022{ 4023 for (size_t i = 0; i < outputTracks.size(); i++) { 4024 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4025 if (thread == 0) { 4026 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4027 return false; 4028 } 4029 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4030 // see note at standby() declaration 4031 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4032 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4033 return false; 4034 } 4035 } 4036 return true; 4037} 4038 4039uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4040{ 4041 return (mWaitTimeMs * 1000) / 2; 4042} 4043 4044void AudioFlinger::DuplicatingThread::cacheParameters_l() 4045{ 4046 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4047 updateWaitTime_l(); 4048 4049 MixerThread::cacheParameters_l(); 4050} 4051 4052// ---------------------------------------------------------------------------- 4053 4054// TrackBase constructor must be called with AudioFlinger::mLock held 4055AudioFlinger::ThreadBase::TrackBase::TrackBase( 4056 ThreadBase *thread, 4057 const sp<Client>& client, 4058 uint32_t sampleRate, 4059 audio_format_t format, 4060 audio_channel_mask_t channelMask, 4061 int frameCount, 4062 const sp<IMemory>& sharedBuffer, 4063 int sessionId) 4064 : RefBase(), 4065 mThread(thread), 4066 mClient(client), 4067 mCblk(NULL), 4068 // mBuffer 4069 // mBufferEnd 4070 mFrameCount(0), 4071 mState(IDLE), 4072 mSampleRate(sampleRate), 4073 mFormat(format), 4074 mStepServerFailed(false), 4075 mSessionId(sessionId) 4076 // mChannelCount 4077 // mChannelMask 4078{ 4079 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4080 4081 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4082 size_t size = sizeof(audio_track_cblk_t); 4083 uint8_t channelCount = popcount(channelMask); 4084 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4085 if (sharedBuffer == 0) { 4086 size += bufferSize; 4087 } 4088 4089 if (client != NULL) { 4090 mCblkMemory = client->heap()->allocate(size); 4091 if (mCblkMemory != 0) { 4092 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4093 if (mCblk != NULL) { // construct the shared structure in-place. 4094 new(mCblk) audio_track_cblk_t(); 4095 // clear all buffers 4096 mCblk->frameCount = frameCount; 4097 mCblk->sampleRate = sampleRate; 4098// uncomment the following lines to quickly test 32-bit wraparound 4099// mCblk->user = 0xffff0000; 4100// mCblk->server = 0xffff0000; 4101// mCblk->userBase = 0xffff0000; 4102// mCblk->serverBase = 0xffff0000; 4103 mChannelCount = channelCount; 4104 mChannelMask = channelMask; 4105 if (sharedBuffer == 0) { 4106 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4107 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4108 // Force underrun condition to avoid false underrun callback until first data is 4109 // written to buffer (other flags are cleared) 4110 mCblk->flags = CBLK_UNDERRUN_ON; 4111 } else { 4112 mBuffer = sharedBuffer->pointer(); 4113 } 4114 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4115 } 4116 } else { 4117 ALOGE("not enough memory for AudioTrack size=%u", size); 4118 client->heap()->dump("AudioTrack"); 4119 return; 4120 } 4121 } else { 4122 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4123 // construct the shared structure in-place. 4124 new(mCblk) audio_track_cblk_t(); 4125 // clear all buffers 4126 mCblk->frameCount = frameCount; 4127 mCblk->sampleRate = sampleRate; 4128// uncomment the following lines to quickly test 32-bit wraparound 4129// mCblk->user = 0xffff0000; 4130// mCblk->server = 0xffff0000; 4131// mCblk->userBase = 0xffff0000; 4132// mCblk->serverBase = 0xffff0000; 4133 mChannelCount = channelCount; 4134 mChannelMask = channelMask; 4135 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4136 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4137 // Force underrun condition to avoid false underrun callback until first data is 4138 // written to buffer (other flags are cleared) 4139 mCblk->flags = CBLK_UNDERRUN_ON; 4140 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4141 } 4142} 4143 4144AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4145{ 4146 if (mCblk != NULL) { 4147 if (mClient == 0) { 4148 delete mCblk; 4149 } else { 4150 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4151 } 4152 } 4153 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4154 if (mClient != 0) { 4155 // Client destructor must run with AudioFlinger mutex locked 4156 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4157 // If the client's reference count drops to zero, the associated destructor 4158 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4159 // relying on the automatic clear() at end of scope. 4160 mClient.clear(); 4161 } 4162} 4163 4164// AudioBufferProvider interface 4165// getNextBuffer() = 0; 4166// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4167void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4168{ 4169 buffer->raw = NULL; 4170 mFrameCount = buffer->frameCount; 4171 // FIXME See note at getNextBuffer() 4172 (void) step(); // ignore return value of step() 4173 buffer->frameCount = 0; 4174} 4175 4176bool AudioFlinger::ThreadBase::TrackBase::step() { 4177 bool result; 4178 audio_track_cblk_t* cblk = this->cblk(); 4179 4180 result = cblk->stepServer(mFrameCount); 4181 if (!result) { 4182 ALOGV("stepServer failed acquiring cblk mutex"); 4183 mStepServerFailed = true; 4184 } 4185 return result; 4186} 4187 4188void AudioFlinger::ThreadBase::TrackBase::reset() { 4189 audio_track_cblk_t* cblk = this->cblk(); 4190 4191 cblk->user = 0; 4192 cblk->server = 0; 4193 cblk->userBase = 0; 4194 cblk->serverBase = 0; 4195 mStepServerFailed = false; 4196 ALOGV("TrackBase::reset"); 4197} 4198 4199int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4200 return (int)mCblk->sampleRate; 4201} 4202 4203void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4204 audio_track_cblk_t* cblk = this->cblk(); 4205 size_t frameSize = cblk->frameSize; 4206 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4207 int8_t *bufferEnd = bufferStart + frames * frameSize; 4208 4209 // Check validity of returned pointer in case the track control block would have been corrupted. 4210 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4211 "TrackBase::getBuffer buffer out of range:\n" 4212 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4213 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4214 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4215 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4216 4217 return bufferStart; 4218} 4219 4220status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4221{ 4222 mSyncEvents.add(event); 4223 return NO_ERROR; 4224} 4225 4226// ---------------------------------------------------------------------------- 4227 4228// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4229AudioFlinger::PlaybackThread::Track::Track( 4230 PlaybackThread *thread, 4231 const sp<Client>& client, 4232 audio_stream_type_t streamType, 4233 uint32_t sampleRate, 4234 audio_format_t format, 4235 audio_channel_mask_t channelMask, 4236 int frameCount, 4237 const sp<IMemory>& sharedBuffer, 4238 int sessionId, 4239 IAudioFlinger::track_flags_t flags) 4240 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4241 mMute(false), 4242 mFillingUpStatus(FS_INVALID), 4243 // mRetryCount initialized later when needed 4244 mSharedBuffer(sharedBuffer), 4245 mStreamType(streamType), 4246 mName(-1), // see note below 4247 mMainBuffer(thread->mixBuffer()), 4248 mAuxBuffer(NULL), 4249 mAuxEffectId(0), mHasVolumeController(false), 4250 mPresentationCompleteFrames(0), 4251 mFlags(flags), 4252 mFastIndex(-1), 4253 mUnderrunCount(0), 4254 mCachedVolume(1.0) 4255{ 4256 if (mCblk != NULL) { 4257 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4258 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4259 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4260 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4261 mName = thread->getTrackName_l(channelMask); 4262 mCblk->mName = mName; 4263 if (mName < 0) { 4264 ALOGE("no more track names available"); 4265 return; 4266 } 4267 // only allocate a fast track index if we were able to allocate a normal track name 4268 if (flags & IAudioFlinger::TRACK_FAST) { 4269 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4270 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4271 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4272 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4273 // FIXME This is too eager. We allocate a fast track index before the 4274 // fast track becomes active. Since fast tracks are a scarce resource, 4275 // this means we are potentially denying other more important fast tracks from 4276 // being created. It would be better to allocate the index dynamically. 4277 mFastIndex = i; 4278 mCblk->mName = i; 4279 // Read the initial underruns because this field is never cleared by the fast mixer 4280 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4281 thread->mFastTrackAvailMask &= ~(1 << i); 4282 } 4283 } 4284 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4285} 4286 4287AudioFlinger::PlaybackThread::Track::~Track() 4288{ 4289 ALOGV("PlaybackThread::Track destructor"); 4290 sp<ThreadBase> thread = mThread.promote(); 4291 if (thread != 0) { 4292 Mutex::Autolock _l(thread->mLock); 4293 mState = TERMINATED; 4294 } 4295} 4296 4297void AudioFlinger::PlaybackThread::Track::destroy() 4298{ 4299 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4300 // by removing it from mTracks vector, so there is a risk that this Tracks's 4301 // destructor is called. As the destructor needs to lock mLock, 4302 // we must acquire a strong reference on this Track before locking mLock 4303 // here so that the destructor is called only when exiting this function. 4304 // On the other hand, as long as Track::destroy() is only called by 4305 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4306 // this Track with its member mTrack. 4307 sp<Track> keep(this); 4308 { // scope for mLock 4309 sp<ThreadBase> thread = mThread.promote(); 4310 if (thread != 0) { 4311 if (!isOutputTrack()) { 4312 if (mState == ACTIVE || mState == RESUMING) { 4313 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4314 4315#ifdef ADD_BATTERY_DATA 4316 // to track the speaker usage 4317 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4318#endif 4319 } 4320 AudioSystem::releaseOutput(thread->id()); 4321 } 4322 Mutex::Autolock _l(thread->mLock); 4323 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4324 playbackThread->destroyTrack_l(this); 4325 } 4326 } 4327} 4328 4329/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4330{ 4331 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4332 " Server User Main buf Aux Buf Flags Underruns\n"); 4333} 4334 4335void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4336{ 4337 uint32_t vlr = mCblk->getVolumeLR(); 4338 if (isFastTrack()) { 4339 sprintf(buffer, " F %2d", mFastIndex); 4340 } else { 4341 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4342 } 4343 track_state state = mState; 4344 char stateChar; 4345 switch (state) { 4346 case IDLE: 4347 stateChar = 'I'; 4348 break; 4349 case TERMINATED: 4350 stateChar = 'T'; 4351 break; 4352 case STOPPING_1: 4353 stateChar = 's'; 4354 break; 4355 case STOPPING_2: 4356 stateChar = '5'; 4357 break; 4358 case STOPPED: 4359 stateChar = 'S'; 4360 break; 4361 case RESUMING: 4362 stateChar = 'R'; 4363 break; 4364 case ACTIVE: 4365 stateChar = 'A'; 4366 break; 4367 case PAUSING: 4368 stateChar = 'p'; 4369 break; 4370 case PAUSED: 4371 stateChar = 'P'; 4372 break; 4373 case FLUSHED: 4374 stateChar = 'F'; 4375 break; 4376 default: 4377 stateChar = '?'; 4378 break; 4379 } 4380 char nowInUnderrun; 4381 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4382 case UNDERRUN_FULL: 4383 nowInUnderrun = ' '; 4384 break; 4385 case UNDERRUN_PARTIAL: 4386 nowInUnderrun = '<'; 4387 break; 4388 case UNDERRUN_EMPTY: 4389 nowInUnderrun = '*'; 4390 break; 4391 default: 4392 nowInUnderrun = '?'; 4393 break; 4394 } 4395 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4396 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4397 (mClient == 0) ? getpid_cached : mClient->pid(), 4398 mStreamType, 4399 mFormat, 4400 mChannelMask, 4401 mSessionId, 4402 mFrameCount, 4403 mCblk->frameCount, 4404 stateChar, 4405 mMute, 4406 mFillingUpStatus, 4407 mCblk->sampleRate, 4408 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4409 20.0 * log10((vlr >> 16) / 4096.0), 4410 mCblk->server, 4411 mCblk->user, 4412 (int)mMainBuffer, 4413 (int)mAuxBuffer, 4414 mCblk->flags, 4415 mUnderrunCount, 4416 nowInUnderrun); 4417} 4418 4419// AudioBufferProvider interface 4420status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4421 AudioBufferProvider::Buffer* buffer, int64_t pts) 4422{ 4423 audio_track_cblk_t* cblk = this->cblk(); 4424 uint32_t framesReady; 4425 uint32_t framesReq = buffer->frameCount; 4426 4427 // Check if last stepServer failed, try to step now 4428 if (mStepServerFailed) { 4429 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4430 // Since the fast mixer is higher priority than client callback thread, 4431 // it does not result in priority inversion for client. 4432 // But a non-blocking solution would be preferable to avoid 4433 // fast mixer being unable to tryLock(), and 4434 // to avoid the extra context switches if the client wakes up, 4435 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4436 if (!step()) goto getNextBuffer_exit; 4437 ALOGV("stepServer recovered"); 4438 mStepServerFailed = false; 4439 } 4440 4441 // FIXME Same as above 4442 framesReady = cblk->framesReady(); 4443 4444 if (CC_LIKELY(framesReady)) { 4445 uint32_t s = cblk->server; 4446 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4447 4448 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4449 if (framesReq > framesReady) { 4450 framesReq = framesReady; 4451 } 4452 if (framesReq > bufferEnd - s) { 4453 framesReq = bufferEnd - s; 4454 } 4455 4456 buffer->raw = getBuffer(s, framesReq); 4457 buffer->frameCount = framesReq; 4458 return NO_ERROR; 4459 } 4460 4461getNextBuffer_exit: 4462 buffer->raw = NULL; 4463 buffer->frameCount = 0; 4464 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4465 return NOT_ENOUGH_DATA; 4466} 4467 4468// Note that framesReady() takes a mutex on the control block using tryLock(). 4469// This could result in priority inversion if framesReady() is called by the normal mixer, 4470// as the normal mixer thread runs at lower 4471// priority than the client's callback thread: there is a short window within framesReady() 4472// during which the normal mixer could be preempted, and the client callback would block. 4473// Another problem can occur if framesReady() is called by the fast mixer: 4474// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4475// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4476size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4477 return mCblk->framesReady(); 4478} 4479 4480// Don't call for fast tracks; the framesReady() could result in priority inversion 4481bool AudioFlinger::PlaybackThread::Track::isReady() const { 4482 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4483 4484 if (framesReady() >= mCblk->frameCount || 4485 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4486 mFillingUpStatus = FS_FILLED; 4487 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4488 return true; 4489 } 4490 return false; 4491} 4492 4493status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4494 int triggerSession) 4495{ 4496 status_t status = NO_ERROR; 4497 ALOGV("start(%d), calling pid %d session %d", 4498 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4499 4500 sp<ThreadBase> thread = mThread.promote(); 4501 if (thread != 0) { 4502 Mutex::Autolock _l(thread->mLock); 4503 track_state state = mState; 4504 // here the track could be either new, or restarted 4505 // in both cases "unstop" the track 4506 if (mState == PAUSED) { 4507 mState = TrackBase::RESUMING; 4508 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4509 } else { 4510 mState = TrackBase::ACTIVE; 4511 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4512 } 4513 4514 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4515 thread->mLock.unlock(); 4516 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4517 thread->mLock.lock(); 4518 4519#ifdef ADD_BATTERY_DATA 4520 // to track the speaker usage 4521 if (status == NO_ERROR) { 4522 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4523 } 4524#endif 4525 } 4526 if (status == NO_ERROR) { 4527 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4528 playbackThread->addTrack_l(this); 4529 } else { 4530 mState = state; 4531 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4532 } 4533 } else { 4534 status = BAD_VALUE; 4535 } 4536 return status; 4537} 4538 4539void AudioFlinger::PlaybackThread::Track::stop() 4540{ 4541 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4542 sp<ThreadBase> thread = mThread.promote(); 4543 if (thread != 0) { 4544 Mutex::Autolock _l(thread->mLock); 4545 track_state state = mState; 4546 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4547 // If the track is not active (PAUSED and buffers full), flush buffers 4548 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4549 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4550 reset(); 4551 mState = STOPPED; 4552 } else if (!isFastTrack()) { 4553 mState = STOPPED; 4554 } else { 4555 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4556 // and then to STOPPED and reset() when presentation is complete 4557 mState = STOPPING_1; 4558 } 4559 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4560 } 4561 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4562 thread->mLock.unlock(); 4563 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4564 thread->mLock.lock(); 4565 4566#ifdef ADD_BATTERY_DATA 4567 // to track the speaker usage 4568 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4569#endif 4570 } 4571 } 4572} 4573 4574void AudioFlinger::PlaybackThread::Track::pause() 4575{ 4576 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4577 sp<ThreadBase> thread = mThread.promote(); 4578 if (thread != 0) { 4579 Mutex::Autolock _l(thread->mLock); 4580 if (mState == ACTIVE || mState == RESUMING) { 4581 mState = PAUSING; 4582 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4583 if (!isOutputTrack()) { 4584 thread->mLock.unlock(); 4585 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4586 thread->mLock.lock(); 4587 4588#ifdef ADD_BATTERY_DATA 4589 // to track the speaker usage 4590 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4591#endif 4592 } 4593 } 4594 } 4595} 4596 4597void AudioFlinger::PlaybackThread::Track::flush() 4598{ 4599 ALOGV("flush(%d)", mName); 4600 sp<ThreadBase> thread = mThread.promote(); 4601 if (thread != 0) { 4602 Mutex::Autolock _l(thread->mLock); 4603 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4604 mState != PAUSING) { 4605 return; 4606 } 4607 // No point remaining in PAUSED state after a flush => go to 4608 // FLUSHED state 4609 mState = FLUSHED; 4610 // do not reset the track if it is still in the process of being stopped or paused. 4611 // this will be done by prepareTracks_l() when the track is stopped. 4612 // prepareTracks_l() will see mState == FLUSHED, then 4613 // remove from active track list, reset(), and trigger presentation complete 4614 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4615 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4616 reset(); 4617 } 4618 } 4619} 4620 4621void AudioFlinger::PlaybackThread::Track::reset() 4622{ 4623 // Do not reset twice to avoid discarding data written just after a flush and before 4624 // the audioflinger thread detects the track is stopped. 4625 if (!mResetDone) { 4626 TrackBase::reset(); 4627 // Force underrun condition to avoid false underrun callback until first data is 4628 // written to buffer 4629 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4630 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4631 mFillingUpStatus = FS_FILLING; 4632 mResetDone = true; 4633 if (mState == FLUSHED) { 4634 mState = IDLE; 4635 } 4636 } 4637} 4638 4639void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4640{ 4641 mMute = muted; 4642} 4643 4644status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4645{ 4646 status_t status = DEAD_OBJECT; 4647 sp<ThreadBase> thread = mThread.promote(); 4648 if (thread != 0) { 4649 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4650 sp<AudioFlinger> af = mClient->audioFlinger(); 4651 4652 Mutex::Autolock _l(af->mLock); 4653 4654 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4655 4656 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4657 Mutex::Autolock _dl(playbackThread->mLock); 4658 Mutex::Autolock _sl(srcThread->mLock); 4659 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4660 if (chain == 0) { 4661 return INVALID_OPERATION; 4662 } 4663 4664 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4665 if (effect == 0) { 4666 return INVALID_OPERATION; 4667 } 4668 srcThread->removeEffect_l(effect); 4669 playbackThread->addEffect_l(effect); 4670 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4671 if (effect->state() == EffectModule::ACTIVE || 4672 effect->state() == EffectModule::STOPPING) { 4673 effect->start(); 4674 } 4675 4676 sp<EffectChain> dstChain = effect->chain().promote(); 4677 if (dstChain == 0) { 4678 srcThread->addEffect_l(effect); 4679 return INVALID_OPERATION; 4680 } 4681 AudioSystem::unregisterEffect(effect->id()); 4682 AudioSystem::registerEffect(&effect->desc(), 4683 srcThread->id(), 4684 dstChain->strategy(), 4685 AUDIO_SESSION_OUTPUT_MIX, 4686 effect->id()); 4687 } 4688 status = playbackThread->attachAuxEffect(this, EffectId); 4689 } 4690 return status; 4691} 4692 4693void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4694{ 4695 mAuxEffectId = EffectId; 4696 mAuxBuffer = buffer; 4697} 4698 4699bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4700 size_t audioHalFrames) 4701{ 4702 // a track is considered presented when the total number of frames written to audio HAL 4703 // corresponds to the number of frames written when presentationComplete() is called for the 4704 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4705 if (mPresentationCompleteFrames == 0) { 4706 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4707 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4708 mPresentationCompleteFrames, audioHalFrames); 4709 } 4710 if (framesWritten >= mPresentationCompleteFrames) { 4711 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4712 mSessionId, framesWritten); 4713 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4714 return true; 4715 } 4716 return false; 4717} 4718 4719void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4720{ 4721 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4722 if (mSyncEvents[i]->type() == type) { 4723 mSyncEvents[i]->trigger(); 4724 mSyncEvents.removeAt(i); 4725 i--; 4726 } 4727 } 4728} 4729 4730// implement VolumeBufferProvider interface 4731 4732uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4733{ 4734 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4735 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4736 uint32_t vlr = mCblk->getVolumeLR(); 4737 uint32_t vl = vlr & 0xFFFF; 4738 uint32_t vr = vlr >> 16; 4739 // track volumes come from shared memory, so can't be trusted and must be clamped 4740 if (vl > MAX_GAIN_INT) { 4741 vl = MAX_GAIN_INT; 4742 } 4743 if (vr > MAX_GAIN_INT) { 4744 vr = MAX_GAIN_INT; 4745 } 4746 // now apply the cached master volume and stream type volume; 4747 // this is trusted but lacks any synchronization or barrier so may be stale 4748 float v = mCachedVolume; 4749 vl *= v; 4750 vr *= v; 4751 // re-combine into U4.16 4752 vlr = (vr << 16) | (vl & 0xFFFF); 4753 // FIXME look at mute, pause, and stop flags 4754 return vlr; 4755} 4756 4757status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4758{ 4759 if (mState == TERMINATED || mState == PAUSED || 4760 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4761 (mState == STOPPED)))) { 4762 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4763 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4764 event->cancel(); 4765 return INVALID_OPERATION; 4766 } 4767 TrackBase::setSyncEvent(event); 4768 return NO_ERROR; 4769} 4770 4771// timed audio tracks 4772 4773sp<AudioFlinger::PlaybackThread::TimedTrack> 4774AudioFlinger::PlaybackThread::TimedTrack::create( 4775 PlaybackThread *thread, 4776 const sp<Client>& client, 4777 audio_stream_type_t streamType, 4778 uint32_t sampleRate, 4779 audio_format_t format, 4780 audio_channel_mask_t channelMask, 4781 int frameCount, 4782 const sp<IMemory>& sharedBuffer, 4783 int sessionId) { 4784 if (!client->reserveTimedTrack()) 4785 return 0; 4786 4787 return new TimedTrack( 4788 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4789 sharedBuffer, sessionId); 4790} 4791 4792AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4793 PlaybackThread *thread, 4794 const sp<Client>& client, 4795 audio_stream_type_t streamType, 4796 uint32_t sampleRate, 4797 audio_format_t format, 4798 audio_channel_mask_t channelMask, 4799 int frameCount, 4800 const sp<IMemory>& sharedBuffer, 4801 int sessionId) 4802 : Track(thread, client, streamType, sampleRate, format, channelMask, 4803 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4804 mQueueHeadInFlight(false), 4805 mTrimQueueHeadOnRelease(false), 4806 mFramesPendingInQueue(0), 4807 mTimedSilenceBuffer(NULL), 4808 mTimedSilenceBufferSize(0), 4809 mTimedAudioOutputOnTime(false), 4810 mMediaTimeTransformValid(false) 4811{ 4812 LocalClock lc; 4813 mLocalTimeFreq = lc.getLocalFreq(); 4814 4815 mLocalTimeToSampleTransform.a_zero = 0; 4816 mLocalTimeToSampleTransform.b_zero = 0; 4817 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4818 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4819 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4820 &mLocalTimeToSampleTransform.a_to_b_denom); 4821 4822 mMediaTimeToSampleTransform.a_zero = 0; 4823 mMediaTimeToSampleTransform.b_zero = 0; 4824 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4825 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4826 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4827 &mMediaTimeToSampleTransform.a_to_b_denom); 4828} 4829 4830AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4831 mClient->releaseTimedTrack(); 4832 delete [] mTimedSilenceBuffer; 4833} 4834 4835status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4836 size_t size, sp<IMemory>* buffer) { 4837 4838 Mutex::Autolock _l(mTimedBufferQueueLock); 4839 4840 trimTimedBufferQueue_l(); 4841 4842 // lazily initialize the shared memory heap for timed buffers 4843 if (mTimedMemoryDealer == NULL) { 4844 const int kTimedBufferHeapSize = 512 << 10; 4845 4846 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4847 "AudioFlingerTimed"); 4848 if (mTimedMemoryDealer == NULL) 4849 return NO_MEMORY; 4850 } 4851 4852 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4853 if (newBuffer == NULL) { 4854 newBuffer = mTimedMemoryDealer->allocate(size); 4855 if (newBuffer == NULL) 4856 return NO_MEMORY; 4857 } 4858 4859 *buffer = newBuffer; 4860 return NO_ERROR; 4861} 4862 4863// caller must hold mTimedBufferQueueLock 4864void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4865 int64_t mediaTimeNow; 4866 { 4867 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4868 if (!mMediaTimeTransformValid) 4869 return; 4870 4871 int64_t targetTimeNow; 4872 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4873 ? mCCHelper.getCommonTime(&targetTimeNow) 4874 : mCCHelper.getLocalTime(&targetTimeNow); 4875 4876 if (OK != res) 4877 return; 4878 4879 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4880 &mediaTimeNow)) { 4881 return; 4882 } 4883 } 4884 4885 size_t trimEnd; 4886 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4887 int64_t bufEnd; 4888 4889 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4890 // We have a next buffer. Just use its PTS as the PTS of the frame 4891 // following the last frame in this buffer. If the stream is sparse 4892 // (ie, there are deliberate gaps left in the stream which should be 4893 // filled with silence by the TimedAudioTrack), then this can result 4894 // in one extra buffer being left un-trimmed when it could have 4895 // been. In general, this is not typical, and we would rather 4896 // optimized away the TS calculation below for the more common case 4897 // where PTSes are contiguous. 4898 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4899 } else { 4900 // We have no next buffer. Compute the PTS of the frame following 4901 // the last frame in this buffer by computing the duration of of 4902 // this frame in media time units and adding it to the PTS of the 4903 // buffer. 4904 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4905 / mCblk->frameSize; 4906 4907 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4908 &bufEnd)) { 4909 ALOGE("Failed to convert frame count of %lld to media time" 4910 " duration" " (scale factor %d/%u) in %s", 4911 frameCount, 4912 mMediaTimeToSampleTransform.a_to_b_numer, 4913 mMediaTimeToSampleTransform.a_to_b_denom, 4914 __PRETTY_FUNCTION__); 4915 break; 4916 } 4917 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4918 } 4919 4920 if (bufEnd > mediaTimeNow) 4921 break; 4922 4923 // Is the buffer we want to use in the middle of a mix operation right 4924 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4925 // from the mixer which should be coming back shortly. 4926 if (!trimEnd && mQueueHeadInFlight) { 4927 mTrimQueueHeadOnRelease = true; 4928 } 4929 } 4930 4931 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4932 if (trimStart < trimEnd) { 4933 // Update the bookkeeping for framesReady() 4934 for (size_t i = trimStart; i < trimEnd; ++i) { 4935 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4936 } 4937 4938 // Now actually remove the buffers from the queue. 4939 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4940 } 4941} 4942 4943void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4944 const char* logTag) { 4945 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4946 "%s called (reason \"%s\"), but timed buffer queue has no" 4947 " elements to trim.", __FUNCTION__, logTag); 4948 4949 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4950 mTimedBufferQueue.removeAt(0); 4951} 4952 4953void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4954 const TimedBuffer& buf, 4955 const char* logTag) { 4956 uint32_t bufBytes = buf.buffer()->size(); 4957 uint32_t consumedAlready = buf.position(); 4958 4959 ALOG_ASSERT(consumedAlready <= bufBytes, 4960 "Bad bookkeeping while updating frames pending. Timed buffer is" 4961 " only %u bytes long, but claims to have consumed %u" 4962 " bytes. (update reason: \"%s\")", 4963 bufBytes, consumedAlready, logTag); 4964 4965 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4966 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4967 "Bad bookkeeping while updating frames pending. Should have at" 4968 " least %u queued frames, but we think we have only %u. (update" 4969 " reason: \"%s\")", 4970 bufFrames, mFramesPendingInQueue, logTag); 4971 4972 mFramesPendingInQueue -= bufFrames; 4973} 4974 4975status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4976 const sp<IMemory>& buffer, int64_t pts) { 4977 4978 { 4979 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4980 if (!mMediaTimeTransformValid) 4981 return INVALID_OPERATION; 4982 } 4983 4984 Mutex::Autolock _l(mTimedBufferQueueLock); 4985 4986 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4987 mFramesPendingInQueue += bufFrames; 4988 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4989 4990 return NO_ERROR; 4991} 4992 4993status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4994 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4995 4996 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4997 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4998 target); 4999 5000 if (!(target == TimedAudioTrack::LOCAL_TIME || 5001 target == TimedAudioTrack::COMMON_TIME)) { 5002 return BAD_VALUE; 5003 } 5004 5005 Mutex::Autolock lock(mMediaTimeTransformLock); 5006 mMediaTimeTransform = xform; 5007 mMediaTimeTransformTarget = target; 5008 mMediaTimeTransformValid = true; 5009 5010 return NO_ERROR; 5011} 5012 5013#define min(a, b) ((a) < (b) ? (a) : (b)) 5014 5015// implementation of getNextBuffer for tracks whose buffers have timestamps 5016status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5017 AudioBufferProvider::Buffer* buffer, int64_t pts) 5018{ 5019 if (pts == AudioBufferProvider::kInvalidPTS) { 5020 buffer->raw = NULL; 5021 buffer->frameCount = 0; 5022 mTimedAudioOutputOnTime = false; 5023 return INVALID_OPERATION; 5024 } 5025 5026 Mutex::Autolock _l(mTimedBufferQueueLock); 5027 5028 ALOG_ASSERT(!mQueueHeadInFlight, 5029 "getNextBuffer called without releaseBuffer!"); 5030 5031 while (true) { 5032 5033 // if we have no timed buffers, then fail 5034 if (mTimedBufferQueue.isEmpty()) { 5035 buffer->raw = NULL; 5036 buffer->frameCount = 0; 5037 return NOT_ENOUGH_DATA; 5038 } 5039 5040 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5041 5042 // calculate the PTS of the head of the timed buffer queue expressed in 5043 // local time 5044 int64_t headLocalPTS; 5045 { 5046 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5047 5048 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5049 5050 if (mMediaTimeTransform.a_to_b_denom == 0) { 5051 // the transform represents a pause, so yield silence 5052 timedYieldSilence_l(buffer->frameCount, buffer); 5053 return NO_ERROR; 5054 } 5055 5056 int64_t transformedPTS; 5057 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5058 &transformedPTS)) { 5059 // the transform failed. this shouldn't happen, but if it does 5060 // then just drop this buffer 5061 ALOGW("timedGetNextBuffer transform failed"); 5062 buffer->raw = NULL; 5063 buffer->frameCount = 0; 5064 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5065 return NO_ERROR; 5066 } 5067 5068 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5069 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5070 &headLocalPTS)) { 5071 buffer->raw = NULL; 5072 buffer->frameCount = 0; 5073 return INVALID_OPERATION; 5074 } 5075 } else { 5076 headLocalPTS = transformedPTS; 5077 } 5078 } 5079 5080 // adjust the head buffer's PTS to reflect the portion of the head buffer 5081 // that has already been consumed 5082 int64_t effectivePTS = headLocalPTS + 5083 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5084 5085 // Calculate the delta in samples between the head of the input buffer 5086 // queue and the start of the next output buffer that will be written. 5087 // If the transformation fails because of over or underflow, it means 5088 // that the sample's position in the output stream is so far out of 5089 // whack that it should just be dropped. 5090 int64_t sampleDelta; 5091 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5092 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5093 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5094 " mix"); 5095 continue; 5096 } 5097 if (!mLocalTimeToSampleTransform.doForwardTransform( 5098 (effectivePTS - pts) << 32, &sampleDelta)) { 5099 ALOGV("*** too late during sample rate transform: dropped buffer"); 5100 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5101 continue; 5102 } 5103 5104 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5105 " sampleDelta=[%d.%08x]", 5106 head.pts(), head.position(), pts, 5107 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5108 + (sampleDelta >> 32)), 5109 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5110 5111 // if the delta between the ideal placement for the next input sample and 5112 // the current output position is within this threshold, then we will 5113 // concatenate the next input samples to the previous output 5114 const int64_t kSampleContinuityThreshold = 5115 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5116 5117 // if this is the first buffer of audio that we're emitting from this track 5118 // then it should be almost exactly on time. 5119 const int64_t kSampleStartupThreshold = 1LL << 32; 5120 5121 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5122 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5123 // the next input is close enough to being on time, so concatenate it 5124 // with the last output 5125 timedYieldSamples_l(buffer); 5126 5127 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5128 head.position(), buffer->frameCount); 5129 return NO_ERROR; 5130 } 5131 5132 // Looks like our output is not on time. Reset our on timed status. 5133 // Next time we mix samples from our input queue, then should be within 5134 // the StartupThreshold. 5135 mTimedAudioOutputOnTime = false; 5136 if (sampleDelta > 0) { 5137 // the gap between the current output position and the proper start of 5138 // the next input sample is too big, so fill it with silence 5139 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5140 5141 timedYieldSilence_l(framesUntilNextInput, buffer); 5142 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5143 return NO_ERROR; 5144 } else { 5145 // the next input sample is late 5146 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5147 size_t onTimeSamplePosition = 5148 head.position() + lateFrames * mCblk->frameSize; 5149 5150 if (onTimeSamplePosition > head.buffer()->size()) { 5151 // all the remaining samples in the head are too late, so 5152 // drop it and move on 5153 ALOGV("*** too late: dropped buffer"); 5154 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5155 continue; 5156 } else { 5157 // skip over the late samples 5158 head.setPosition(onTimeSamplePosition); 5159 5160 // yield the available samples 5161 timedYieldSamples_l(buffer); 5162 5163 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5164 return NO_ERROR; 5165 } 5166 } 5167 } 5168} 5169 5170// Yield samples from the timed buffer queue head up to the given output 5171// buffer's capacity. 5172// 5173// Caller must hold mTimedBufferQueueLock 5174void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5175 AudioBufferProvider::Buffer* buffer) { 5176 5177 const TimedBuffer& head = mTimedBufferQueue[0]; 5178 5179 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5180 head.position()); 5181 5182 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5183 mCblk->frameSize); 5184 size_t framesRequested = buffer->frameCount; 5185 buffer->frameCount = min(framesLeftInHead, framesRequested); 5186 5187 mQueueHeadInFlight = true; 5188 mTimedAudioOutputOnTime = true; 5189} 5190 5191// Yield samples of silence up to the given output buffer's capacity 5192// 5193// Caller must hold mTimedBufferQueueLock 5194void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5195 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5196 5197 // lazily allocate a buffer filled with silence 5198 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5199 delete [] mTimedSilenceBuffer; 5200 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5201 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5202 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5203 } 5204 5205 buffer->raw = mTimedSilenceBuffer; 5206 size_t framesRequested = buffer->frameCount; 5207 buffer->frameCount = min(numFrames, framesRequested); 5208 5209 mTimedAudioOutputOnTime = false; 5210} 5211 5212// AudioBufferProvider interface 5213void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5214 AudioBufferProvider::Buffer* buffer) { 5215 5216 Mutex::Autolock _l(mTimedBufferQueueLock); 5217 5218 // If the buffer which was just released is part of the buffer at the head 5219 // of the queue, be sure to update the amt of the buffer which has been 5220 // consumed. If the buffer being returned is not part of the head of the 5221 // queue, its either because the buffer is part of the silence buffer, or 5222 // because the head of the timed queue was trimmed after the mixer called 5223 // getNextBuffer but before the mixer called releaseBuffer. 5224 if (buffer->raw == mTimedSilenceBuffer) { 5225 ALOG_ASSERT(!mQueueHeadInFlight, 5226 "Queue head in flight during release of silence buffer!"); 5227 goto done; 5228 } 5229 5230 ALOG_ASSERT(mQueueHeadInFlight, 5231 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5232 " head in flight."); 5233 5234 if (mTimedBufferQueue.size()) { 5235 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5236 5237 void* start = head.buffer()->pointer(); 5238 void* end = reinterpret_cast<void*>( 5239 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5240 + head.buffer()->size()); 5241 5242 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5243 "released buffer not within the head of the timed buffer" 5244 " queue; qHead = [%p, %p], released buffer = %p", 5245 start, end, buffer->raw); 5246 5247 head.setPosition(head.position() + 5248 (buffer->frameCount * mCblk->frameSize)); 5249 mQueueHeadInFlight = false; 5250 5251 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5252 "Bad bookkeeping during releaseBuffer! Should have at" 5253 " least %u queued frames, but we think we have only %u", 5254 buffer->frameCount, mFramesPendingInQueue); 5255 5256 mFramesPendingInQueue -= buffer->frameCount; 5257 5258 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5259 || mTrimQueueHeadOnRelease) { 5260 trimTimedBufferQueueHead_l("releaseBuffer"); 5261 mTrimQueueHeadOnRelease = false; 5262 } 5263 } else { 5264 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5265 " buffers in the timed buffer queue"); 5266 } 5267 5268done: 5269 buffer->raw = 0; 5270 buffer->frameCount = 0; 5271} 5272 5273size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5274 Mutex::Autolock _l(mTimedBufferQueueLock); 5275 return mFramesPendingInQueue; 5276} 5277 5278AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5279 : mPTS(0), mPosition(0) {} 5280 5281AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5282 const sp<IMemory>& buffer, int64_t pts) 5283 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5284 5285// ---------------------------------------------------------------------------- 5286 5287// RecordTrack constructor must be called with AudioFlinger::mLock held 5288AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5289 RecordThread *thread, 5290 const sp<Client>& client, 5291 uint32_t sampleRate, 5292 audio_format_t format, 5293 audio_channel_mask_t channelMask, 5294 int frameCount, 5295 int sessionId) 5296 : TrackBase(thread, client, sampleRate, format, 5297 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5298 mOverflow(false) 5299{ 5300 if (mCblk != NULL) { 5301 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5302 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5303 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5304 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5305 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5306 } else { 5307 mCblk->frameSize = sizeof(int8_t); 5308 } 5309 } 5310} 5311 5312AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5313{ 5314 sp<ThreadBase> thread = mThread.promote(); 5315 if (thread != 0) { 5316 AudioSystem::releaseInput(thread->id()); 5317 } 5318} 5319 5320// AudioBufferProvider interface 5321status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5322{ 5323 audio_track_cblk_t* cblk = this->cblk(); 5324 uint32_t framesAvail; 5325 uint32_t framesReq = buffer->frameCount; 5326 5327 // Check if last stepServer failed, try to step now 5328 if (mStepServerFailed) { 5329 if (!step()) goto getNextBuffer_exit; 5330 ALOGV("stepServer recovered"); 5331 mStepServerFailed = false; 5332 } 5333 5334 framesAvail = cblk->framesAvailable_l(); 5335 5336 if (CC_LIKELY(framesAvail)) { 5337 uint32_t s = cblk->server; 5338 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5339 5340 if (framesReq > framesAvail) { 5341 framesReq = framesAvail; 5342 } 5343 if (framesReq > bufferEnd - s) { 5344 framesReq = bufferEnd - s; 5345 } 5346 5347 buffer->raw = getBuffer(s, framesReq); 5348 buffer->frameCount = framesReq; 5349 return NO_ERROR; 5350 } 5351 5352getNextBuffer_exit: 5353 buffer->raw = NULL; 5354 buffer->frameCount = 0; 5355 return NOT_ENOUGH_DATA; 5356} 5357 5358status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5359 int triggerSession) 5360{ 5361 sp<ThreadBase> thread = mThread.promote(); 5362 if (thread != 0) { 5363 RecordThread *recordThread = (RecordThread *)thread.get(); 5364 return recordThread->start(this, event, triggerSession); 5365 } else { 5366 return BAD_VALUE; 5367 } 5368} 5369 5370void AudioFlinger::RecordThread::RecordTrack::stop() 5371{ 5372 sp<ThreadBase> thread = mThread.promote(); 5373 if (thread != 0) { 5374 RecordThread *recordThread = (RecordThread *)thread.get(); 5375 recordThread->stop(this); 5376 TrackBase::reset(); 5377 // Force overrun condition to avoid false overrun callback until first data is 5378 // read from buffer 5379 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5380 } 5381} 5382 5383void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5384{ 5385 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5386 (mClient == 0) ? getpid_cached : mClient->pid(), 5387 mFormat, 5388 mChannelMask, 5389 mSessionId, 5390 mFrameCount, 5391 mState, 5392 mCblk->sampleRate, 5393 mCblk->server, 5394 mCblk->user); 5395} 5396 5397 5398// ---------------------------------------------------------------------------- 5399 5400AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5401 PlaybackThread *playbackThread, 5402 DuplicatingThread *sourceThread, 5403 uint32_t sampleRate, 5404 audio_format_t format, 5405 audio_channel_mask_t channelMask, 5406 int frameCount) 5407 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5408 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5409 mActive(false), mSourceThread(sourceThread) 5410{ 5411 5412 if (mCblk != NULL) { 5413 mCblk->flags |= CBLK_DIRECTION_OUT; 5414 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5415 mOutBuffer.frameCount = 0; 5416 playbackThread->mTracks.add(this); 5417 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5418 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5419 mCblk, mBuffer, mCblk->buffers, 5420 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5421 } else { 5422 ALOGW("Error creating output track on thread %p", playbackThread); 5423 } 5424} 5425 5426AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5427{ 5428 clearBufferQueue(); 5429} 5430 5431status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5432 int triggerSession) 5433{ 5434 status_t status = Track::start(event, triggerSession); 5435 if (status != NO_ERROR) { 5436 return status; 5437 } 5438 5439 mActive = true; 5440 mRetryCount = 127; 5441 return status; 5442} 5443 5444void AudioFlinger::PlaybackThread::OutputTrack::stop() 5445{ 5446 Track::stop(); 5447 clearBufferQueue(); 5448 mOutBuffer.frameCount = 0; 5449 mActive = false; 5450} 5451 5452bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5453{ 5454 Buffer *pInBuffer; 5455 Buffer inBuffer; 5456 uint32_t channelCount = mChannelCount; 5457 bool outputBufferFull = false; 5458 inBuffer.frameCount = frames; 5459 inBuffer.i16 = data; 5460 5461 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5462 5463 if (!mActive && frames != 0) { 5464 start(); 5465 sp<ThreadBase> thread = mThread.promote(); 5466 if (thread != 0) { 5467 MixerThread *mixerThread = (MixerThread *)thread.get(); 5468 if (mCblk->frameCount > frames){ 5469 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5470 uint32_t startFrames = (mCblk->frameCount - frames); 5471 pInBuffer = new Buffer; 5472 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5473 pInBuffer->frameCount = startFrames; 5474 pInBuffer->i16 = pInBuffer->mBuffer; 5475 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5476 mBufferQueue.add(pInBuffer); 5477 } else { 5478 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5479 } 5480 } 5481 } 5482 } 5483 5484 while (waitTimeLeftMs) { 5485 // First write pending buffers, then new data 5486 if (mBufferQueue.size()) { 5487 pInBuffer = mBufferQueue.itemAt(0); 5488 } else { 5489 pInBuffer = &inBuffer; 5490 } 5491 5492 if (pInBuffer->frameCount == 0) { 5493 break; 5494 } 5495 5496 if (mOutBuffer.frameCount == 0) { 5497 mOutBuffer.frameCount = pInBuffer->frameCount; 5498 nsecs_t startTime = systemTime(); 5499 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5500 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5501 outputBufferFull = true; 5502 break; 5503 } 5504 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5505 if (waitTimeLeftMs >= waitTimeMs) { 5506 waitTimeLeftMs -= waitTimeMs; 5507 } else { 5508 waitTimeLeftMs = 0; 5509 } 5510 } 5511 5512 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5513 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5514 mCblk->stepUser(outFrames); 5515 pInBuffer->frameCount -= outFrames; 5516 pInBuffer->i16 += outFrames * channelCount; 5517 mOutBuffer.frameCount -= outFrames; 5518 mOutBuffer.i16 += outFrames * channelCount; 5519 5520 if (pInBuffer->frameCount == 0) { 5521 if (mBufferQueue.size()) { 5522 mBufferQueue.removeAt(0); 5523 delete [] pInBuffer->mBuffer; 5524 delete pInBuffer; 5525 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5526 } else { 5527 break; 5528 } 5529 } 5530 } 5531 5532 // If we could not write all frames, allocate a buffer and queue it for next time. 5533 if (inBuffer.frameCount) { 5534 sp<ThreadBase> thread = mThread.promote(); 5535 if (thread != 0 && !thread->standby()) { 5536 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5537 pInBuffer = new Buffer; 5538 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5539 pInBuffer->frameCount = inBuffer.frameCount; 5540 pInBuffer->i16 = pInBuffer->mBuffer; 5541 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5542 mBufferQueue.add(pInBuffer); 5543 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5544 } else { 5545 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5546 } 5547 } 5548 } 5549 5550 // Calling write() with a 0 length buffer, means that no more data will be written: 5551 // If no more buffers are pending, fill output track buffer to make sure it is started 5552 // by output mixer. 5553 if (frames == 0 && mBufferQueue.size() == 0) { 5554 if (mCblk->user < mCblk->frameCount) { 5555 frames = mCblk->frameCount - mCblk->user; 5556 pInBuffer = new Buffer; 5557 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5558 pInBuffer->frameCount = frames; 5559 pInBuffer->i16 = pInBuffer->mBuffer; 5560 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5561 mBufferQueue.add(pInBuffer); 5562 } else if (mActive) { 5563 stop(); 5564 } 5565 } 5566 5567 return outputBufferFull; 5568} 5569 5570status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5571{ 5572 int active; 5573 status_t result; 5574 audio_track_cblk_t* cblk = mCblk; 5575 uint32_t framesReq = buffer->frameCount; 5576 5577// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5578 buffer->frameCount = 0; 5579 5580 uint32_t framesAvail = cblk->framesAvailable(); 5581 5582 5583 if (framesAvail == 0) { 5584 Mutex::Autolock _l(cblk->lock); 5585 goto start_loop_here; 5586 while (framesAvail == 0) { 5587 active = mActive; 5588 if (CC_UNLIKELY(!active)) { 5589 ALOGV("Not active and NO_MORE_BUFFERS"); 5590 return NO_MORE_BUFFERS; 5591 } 5592 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5593 if (result != NO_ERROR) { 5594 return NO_MORE_BUFFERS; 5595 } 5596 // read the server count again 5597 start_loop_here: 5598 framesAvail = cblk->framesAvailable_l(); 5599 } 5600 } 5601 5602// if (framesAvail < framesReq) { 5603// return NO_MORE_BUFFERS; 5604// } 5605 5606 if (framesReq > framesAvail) { 5607 framesReq = framesAvail; 5608 } 5609 5610 uint32_t u = cblk->user; 5611 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5612 5613 if (framesReq > bufferEnd - u) { 5614 framesReq = bufferEnd - u; 5615 } 5616 5617 buffer->frameCount = framesReq; 5618 buffer->raw = (void *)cblk->buffer(u); 5619 return NO_ERROR; 5620} 5621 5622 5623void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5624{ 5625 size_t size = mBufferQueue.size(); 5626 5627 for (size_t i = 0; i < size; i++) { 5628 Buffer *pBuffer = mBufferQueue.itemAt(i); 5629 delete [] pBuffer->mBuffer; 5630 delete pBuffer; 5631 } 5632 mBufferQueue.clear(); 5633} 5634 5635// ---------------------------------------------------------------------------- 5636 5637AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5638 : RefBase(), 5639 mAudioFlinger(audioFlinger), 5640 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5641 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5642 mPid(pid), 5643 mTimedTrackCount(0) 5644{ 5645 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5646} 5647 5648// Client destructor must be called with AudioFlinger::mLock held 5649AudioFlinger::Client::~Client() 5650{ 5651 mAudioFlinger->removeClient_l(mPid); 5652} 5653 5654sp<MemoryDealer> AudioFlinger::Client::heap() const 5655{ 5656 return mMemoryDealer; 5657} 5658 5659// Reserve one of the limited slots for a timed audio track associated 5660// with this client 5661bool AudioFlinger::Client::reserveTimedTrack() 5662{ 5663 const int kMaxTimedTracksPerClient = 4; 5664 5665 Mutex::Autolock _l(mTimedTrackLock); 5666 5667 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5668 ALOGW("can not create timed track - pid %d has exceeded the limit", 5669 mPid); 5670 return false; 5671 } 5672 5673 mTimedTrackCount++; 5674 return true; 5675} 5676 5677// Release a slot for a timed audio track 5678void AudioFlinger::Client::releaseTimedTrack() 5679{ 5680 Mutex::Autolock _l(mTimedTrackLock); 5681 mTimedTrackCount--; 5682} 5683 5684// ---------------------------------------------------------------------------- 5685 5686AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5687 const sp<IAudioFlingerClient>& client, 5688 pid_t pid) 5689 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5690{ 5691} 5692 5693AudioFlinger::NotificationClient::~NotificationClient() 5694{ 5695} 5696 5697void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5698{ 5699 sp<NotificationClient> keep(this); 5700 mAudioFlinger->removeNotificationClient(mPid); 5701} 5702 5703// ---------------------------------------------------------------------------- 5704 5705AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5706 : BnAudioTrack(), 5707 mTrack(track) 5708{ 5709} 5710 5711AudioFlinger::TrackHandle::~TrackHandle() { 5712 // just stop the track on deletion, associated resources 5713 // will be freed from the main thread once all pending buffers have 5714 // been played. Unless it's not in the active track list, in which 5715 // case we free everything now... 5716 mTrack->destroy(); 5717} 5718 5719sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5720 return mTrack->getCblk(); 5721} 5722 5723status_t AudioFlinger::TrackHandle::start() { 5724 return mTrack->start(); 5725} 5726 5727void AudioFlinger::TrackHandle::stop() { 5728 mTrack->stop(); 5729} 5730 5731void AudioFlinger::TrackHandle::flush() { 5732 mTrack->flush(); 5733} 5734 5735void AudioFlinger::TrackHandle::mute(bool e) { 5736 mTrack->mute(e); 5737} 5738 5739void AudioFlinger::TrackHandle::pause() { 5740 mTrack->pause(); 5741} 5742 5743status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5744{ 5745 return mTrack->attachAuxEffect(EffectId); 5746} 5747 5748status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5749 sp<IMemory>* buffer) { 5750 if (!mTrack->isTimedTrack()) 5751 return INVALID_OPERATION; 5752 5753 PlaybackThread::TimedTrack* tt = 5754 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5755 return tt->allocateTimedBuffer(size, buffer); 5756} 5757 5758status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5759 int64_t pts) { 5760 if (!mTrack->isTimedTrack()) 5761 return INVALID_OPERATION; 5762 5763 PlaybackThread::TimedTrack* tt = 5764 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5765 return tt->queueTimedBuffer(buffer, pts); 5766} 5767 5768status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5769 const LinearTransform& xform, int target) { 5770 5771 if (!mTrack->isTimedTrack()) 5772 return INVALID_OPERATION; 5773 5774 PlaybackThread::TimedTrack* tt = 5775 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5776 return tt->setMediaTimeTransform( 5777 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5778} 5779 5780status_t AudioFlinger::TrackHandle::onTransact( 5781 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5782{ 5783 return BnAudioTrack::onTransact(code, data, reply, flags); 5784} 5785 5786// ---------------------------------------------------------------------------- 5787 5788sp<IAudioRecord> AudioFlinger::openRecord( 5789 pid_t pid, 5790 audio_io_handle_t input, 5791 uint32_t sampleRate, 5792 audio_format_t format, 5793 audio_channel_mask_t channelMask, 5794 int frameCount, 5795 IAudioFlinger::track_flags_t flags, 5796 pid_t tid, 5797 int *sessionId, 5798 status_t *status) 5799{ 5800 sp<RecordThread::RecordTrack> recordTrack; 5801 sp<RecordHandle> recordHandle; 5802 sp<Client> client; 5803 status_t lStatus; 5804 RecordThread *thread; 5805 size_t inFrameCount; 5806 int lSessionId; 5807 5808 // check calling permissions 5809 if (!recordingAllowed()) { 5810 lStatus = PERMISSION_DENIED; 5811 goto Exit; 5812 } 5813 5814 // add client to list 5815 { // scope for mLock 5816 Mutex::Autolock _l(mLock); 5817 thread = checkRecordThread_l(input); 5818 if (thread == NULL) { 5819 lStatus = BAD_VALUE; 5820 goto Exit; 5821 } 5822 5823 client = registerPid_l(pid); 5824 5825 // If no audio session id is provided, create one here 5826 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5827 lSessionId = *sessionId; 5828 } else { 5829 lSessionId = nextUniqueId(); 5830 if (sessionId != NULL) { 5831 *sessionId = lSessionId; 5832 } 5833 } 5834 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5835 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5836 frameCount, lSessionId, flags, tid, &lStatus); 5837 } 5838 if (lStatus != NO_ERROR) { 5839 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5840 // destructor is called by the TrackBase destructor with mLock held 5841 client.clear(); 5842 recordTrack.clear(); 5843 goto Exit; 5844 } 5845 5846 // return to handle to client 5847 recordHandle = new RecordHandle(recordTrack); 5848 lStatus = NO_ERROR; 5849 5850Exit: 5851 if (status) { 5852 *status = lStatus; 5853 } 5854 return recordHandle; 5855} 5856 5857// ---------------------------------------------------------------------------- 5858 5859AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5860 : BnAudioRecord(), 5861 mRecordTrack(recordTrack) 5862{ 5863} 5864 5865AudioFlinger::RecordHandle::~RecordHandle() { 5866 stop(); 5867} 5868 5869sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5870 return mRecordTrack->getCblk(); 5871} 5872 5873status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5874 ALOGV("RecordHandle::start()"); 5875 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5876} 5877 5878void AudioFlinger::RecordHandle::stop() { 5879 ALOGV("RecordHandle::stop()"); 5880 mRecordTrack->stop(); 5881} 5882 5883status_t AudioFlinger::RecordHandle::onTransact( 5884 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5885{ 5886 return BnAudioRecord::onTransact(code, data, reply, flags); 5887} 5888 5889// ---------------------------------------------------------------------------- 5890 5891AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5892 AudioStreamIn *input, 5893 uint32_t sampleRate, 5894 audio_channel_mask_t channelMask, 5895 audio_io_handle_t id, 5896 uint32_t device) : 5897 ThreadBase(audioFlinger, id, device, RECORD), 5898 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5899 // mRsmpInIndex and mInputBytes set by readInputParameters() 5900 mReqChannelCount(popcount(channelMask)), 5901 mReqSampleRate(sampleRate) 5902 // mBytesRead is only meaningful while active, and so is cleared in start() 5903 // (but might be better to also clear here for dump?) 5904{ 5905 snprintf(mName, kNameLength, "AudioIn_%X", id); 5906 5907 readInputParameters(); 5908} 5909 5910 5911AudioFlinger::RecordThread::~RecordThread() 5912{ 5913 delete[] mRsmpInBuffer; 5914 delete mResampler; 5915 delete[] mRsmpOutBuffer; 5916} 5917 5918void AudioFlinger::RecordThread::onFirstRef() 5919{ 5920 run(mName, PRIORITY_URGENT_AUDIO); 5921} 5922 5923status_t AudioFlinger::RecordThread::readyToRun() 5924{ 5925 status_t status = initCheck(); 5926 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5927 return status; 5928} 5929 5930bool AudioFlinger::RecordThread::threadLoop() 5931{ 5932 AudioBufferProvider::Buffer buffer; 5933 sp<RecordTrack> activeTrack; 5934 Vector< sp<EffectChain> > effectChains; 5935 5936 nsecs_t lastWarning = 0; 5937 5938 acquireWakeLock(); 5939 5940 // start recording 5941 while (!exitPending()) { 5942 5943 processConfigEvents(); 5944 5945 { // scope for mLock 5946 Mutex::Autolock _l(mLock); 5947 checkForNewParameters_l(); 5948 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5949 if (!mStandby) { 5950 mInput->stream->common.standby(&mInput->stream->common); 5951 mStandby = true; 5952 } 5953 5954 if (exitPending()) break; 5955 5956 releaseWakeLock_l(); 5957 ALOGV("RecordThread: loop stopping"); 5958 // go to sleep 5959 mWaitWorkCV.wait(mLock); 5960 ALOGV("RecordThread: loop starting"); 5961 acquireWakeLock_l(); 5962 continue; 5963 } 5964 if (mActiveTrack != 0) { 5965 if (mActiveTrack->mState == TrackBase::PAUSING) { 5966 if (!mStandby) { 5967 mInput->stream->common.standby(&mInput->stream->common); 5968 mStandby = true; 5969 } 5970 mActiveTrack.clear(); 5971 mStartStopCond.broadcast(); 5972 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5973 if (mReqChannelCount != mActiveTrack->channelCount()) { 5974 mActiveTrack.clear(); 5975 mStartStopCond.broadcast(); 5976 } else if (mBytesRead != 0) { 5977 // record start succeeds only if first read from audio input 5978 // succeeds 5979 if (mBytesRead > 0) { 5980 mActiveTrack->mState = TrackBase::ACTIVE; 5981 } else { 5982 mActiveTrack.clear(); 5983 } 5984 mStartStopCond.broadcast(); 5985 } 5986 mStandby = false; 5987 } 5988 } 5989 lockEffectChains_l(effectChains); 5990 } 5991 5992 if (mActiveTrack != 0) { 5993 if (mActiveTrack->mState != TrackBase::ACTIVE && 5994 mActiveTrack->mState != TrackBase::RESUMING) { 5995 unlockEffectChains(effectChains); 5996 usleep(kRecordThreadSleepUs); 5997 continue; 5998 } 5999 for (size_t i = 0; i < effectChains.size(); i ++) { 6000 effectChains[i]->process_l(); 6001 } 6002 6003 buffer.frameCount = mFrameCount; 6004 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6005 size_t framesOut = buffer.frameCount; 6006 if (mResampler == NULL) { 6007 // no resampling 6008 while (framesOut) { 6009 size_t framesIn = mFrameCount - mRsmpInIndex; 6010 if (framesIn) { 6011 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6012 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6013 if (framesIn > framesOut) 6014 framesIn = framesOut; 6015 mRsmpInIndex += framesIn; 6016 framesOut -= framesIn; 6017 if ((int)mChannelCount == mReqChannelCount || 6018 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6019 memcpy(dst, src, framesIn * mFrameSize); 6020 } else { 6021 int16_t *src16 = (int16_t *)src; 6022 int16_t *dst16 = (int16_t *)dst; 6023 if (mChannelCount == 1) { 6024 while (framesIn--) { 6025 *dst16++ = *src16; 6026 *dst16++ = *src16++; 6027 } 6028 } else { 6029 while (framesIn--) { 6030 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6031 src16 += 2; 6032 } 6033 } 6034 } 6035 } 6036 if (framesOut && mFrameCount == mRsmpInIndex) { 6037 if (framesOut == mFrameCount && 6038 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6039 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6040 framesOut = 0; 6041 } else { 6042 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6043 mRsmpInIndex = 0; 6044 } 6045 if (mBytesRead < 0) { 6046 ALOGE("Error reading audio input"); 6047 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6048 // Force input into standby so that it tries to 6049 // recover at next read attempt 6050 mInput->stream->common.standby(&mInput->stream->common); 6051 usleep(kRecordThreadSleepUs); 6052 } 6053 mRsmpInIndex = mFrameCount; 6054 framesOut = 0; 6055 buffer.frameCount = 0; 6056 } 6057 } 6058 } 6059 } else { 6060 // resampling 6061 6062 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6063 // alter output frame count as if we were expecting stereo samples 6064 if (mChannelCount == 1 && mReqChannelCount == 1) { 6065 framesOut >>= 1; 6066 } 6067 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6068 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6069 // are 32 bit aligned which should be always true. 6070 if (mChannelCount == 2 && mReqChannelCount == 1) { 6071 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6072 // the resampler always outputs stereo samples: do post stereo to mono conversion 6073 int16_t *src = (int16_t *)mRsmpOutBuffer; 6074 int16_t *dst = buffer.i16; 6075 while (framesOut--) { 6076 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6077 src += 2; 6078 } 6079 } else { 6080 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6081 } 6082 6083 } 6084 if (mFramestoDrop == 0) { 6085 mActiveTrack->releaseBuffer(&buffer); 6086 } else { 6087 if (mFramestoDrop > 0) { 6088 mFramestoDrop -= buffer.frameCount; 6089 if (mFramestoDrop <= 0) { 6090 clearSyncStartEvent(); 6091 } 6092 } else { 6093 mFramestoDrop += buffer.frameCount; 6094 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6095 mSyncStartEvent->isCancelled()) { 6096 ALOGW("Synced record %s, session %d, trigger session %d", 6097 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6098 mActiveTrack->sessionId(), 6099 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6100 clearSyncStartEvent(); 6101 } 6102 } 6103 } 6104 mActiveTrack->clearOverflow(); 6105 } 6106 // client isn't retrieving buffers fast enough 6107 else { 6108 if (!mActiveTrack->setOverflow()) { 6109 nsecs_t now = systemTime(); 6110 if ((now - lastWarning) > kWarningThrottleNs) { 6111 ALOGW("RecordThread: buffer overflow"); 6112 lastWarning = now; 6113 } 6114 } 6115 // Release the processor for a while before asking for a new buffer. 6116 // This will give the application more chance to read from the buffer and 6117 // clear the overflow. 6118 usleep(kRecordThreadSleepUs); 6119 } 6120 } 6121 // enable changes in effect chain 6122 unlockEffectChains(effectChains); 6123 effectChains.clear(); 6124 } 6125 6126 if (!mStandby) { 6127 mInput->stream->common.standby(&mInput->stream->common); 6128 } 6129 mActiveTrack.clear(); 6130 6131 mStartStopCond.broadcast(); 6132 6133 releaseWakeLock(); 6134 6135 ALOGV("RecordThread %p exiting", this); 6136 return false; 6137} 6138 6139 6140sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6141 const sp<AudioFlinger::Client>& client, 6142 uint32_t sampleRate, 6143 audio_format_t format, 6144 audio_channel_mask_t channelMask, 6145 int frameCount, 6146 int sessionId, 6147 IAudioFlinger::track_flags_t flags, 6148 pid_t tid, 6149 status_t *status) 6150{ 6151 sp<RecordTrack> track; 6152 status_t lStatus; 6153 6154 lStatus = initCheck(); 6155 if (lStatus != NO_ERROR) { 6156 ALOGE("Audio driver not initialized."); 6157 goto Exit; 6158 } 6159 6160 // FIXME use flags and tid similar to createTrack_l() 6161 6162 { // scope for mLock 6163 Mutex::Autolock _l(mLock); 6164 6165 track = new RecordTrack(this, client, sampleRate, 6166 format, channelMask, frameCount, sessionId); 6167 6168 if (track->getCblk() == 0) { 6169 lStatus = NO_MEMORY; 6170 goto Exit; 6171 } 6172 6173 mTrack = track.get(); 6174 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6175 bool suspend = audio_is_bluetooth_sco_device( 6176 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6177 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6178 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6179 } 6180 lStatus = NO_ERROR; 6181 6182Exit: 6183 if (status) { 6184 *status = lStatus; 6185 } 6186 return track; 6187} 6188 6189status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6190 AudioSystem::sync_event_t event, 6191 int triggerSession) 6192{ 6193 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6194 sp<ThreadBase> strongMe = this; 6195 status_t status = NO_ERROR; 6196 6197 if (event == AudioSystem::SYNC_EVENT_NONE) { 6198 clearSyncStartEvent(); 6199 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6200 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6201 triggerSession, 6202 recordTrack->sessionId(), 6203 syncStartEventCallback, 6204 this); 6205 // Sync event can be cancelled by the trigger session if the track is not in a 6206 // compatible state in which case we start record immediately 6207 if (mSyncStartEvent->isCancelled()) { 6208 clearSyncStartEvent(); 6209 } else { 6210 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6211 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6212 } 6213 } 6214 6215 { 6216 AutoMutex lock(mLock); 6217 if (mActiveTrack != 0) { 6218 if (recordTrack != mActiveTrack.get()) { 6219 status = -EBUSY; 6220 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6221 mActiveTrack->mState = TrackBase::ACTIVE; 6222 } 6223 return status; 6224 } 6225 6226 recordTrack->mState = TrackBase::IDLE; 6227 mActiveTrack = recordTrack; 6228 mLock.unlock(); 6229 status_t status = AudioSystem::startInput(mId); 6230 mLock.lock(); 6231 if (status != NO_ERROR) { 6232 mActiveTrack.clear(); 6233 clearSyncStartEvent(); 6234 return status; 6235 } 6236 mRsmpInIndex = mFrameCount; 6237 mBytesRead = 0; 6238 if (mResampler != NULL) { 6239 mResampler->reset(); 6240 } 6241 mActiveTrack->mState = TrackBase::RESUMING; 6242 // signal thread to start 6243 ALOGV("Signal record thread"); 6244 mWaitWorkCV.signal(); 6245 // do not wait for mStartStopCond if exiting 6246 if (exitPending()) { 6247 mActiveTrack.clear(); 6248 status = INVALID_OPERATION; 6249 goto startError; 6250 } 6251 mStartStopCond.wait(mLock); 6252 if (mActiveTrack == 0) { 6253 ALOGV("Record failed to start"); 6254 status = BAD_VALUE; 6255 goto startError; 6256 } 6257 ALOGV("Record started OK"); 6258 return status; 6259 } 6260startError: 6261 AudioSystem::stopInput(mId); 6262 clearSyncStartEvent(); 6263 return status; 6264} 6265 6266void AudioFlinger::RecordThread::clearSyncStartEvent() 6267{ 6268 if (mSyncStartEvent != 0) { 6269 mSyncStartEvent->cancel(); 6270 } 6271 mSyncStartEvent.clear(); 6272 mFramestoDrop = 0; 6273} 6274 6275void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6276{ 6277 sp<SyncEvent> strongEvent = event.promote(); 6278 6279 if (strongEvent != 0) { 6280 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6281 me->handleSyncStartEvent(strongEvent); 6282 } 6283} 6284 6285void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6286{ 6287 if (event == mSyncStartEvent) { 6288 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6289 // from audio HAL 6290 mFramestoDrop = mFrameCount * 2; 6291 } 6292} 6293 6294void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6295 ALOGV("RecordThread::stop"); 6296 sp<ThreadBase> strongMe = this; 6297 { 6298 AutoMutex lock(mLock); 6299 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6300 mActiveTrack->mState = TrackBase::PAUSING; 6301 // do not wait for mStartStopCond if exiting 6302 if (exitPending()) { 6303 return; 6304 } 6305 mStartStopCond.wait(mLock); 6306 // if we have been restarted, recordTrack == mActiveTrack.get() here 6307 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6308 mLock.unlock(); 6309 AudioSystem::stopInput(mId); 6310 mLock.lock(); 6311 ALOGV("Record stopped OK"); 6312 } 6313 } 6314 } 6315} 6316 6317bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6318{ 6319 return false; 6320} 6321 6322status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6323{ 6324 if (!isValidSyncEvent(event)) { 6325 return BAD_VALUE; 6326 } 6327 6328 Mutex::Autolock _l(mLock); 6329 6330 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6331 mTrack->setSyncEvent(event); 6332 return NO_ERROR; 6333 } 6334 return NAME_NOT_FOUND; 6335} 6336 6337status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6338{ 6339 const size_t SIZE = 256; 6340 char buffer[SIZE]; 6341 String8 result; 6342 6343 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6344 result.append(buffer); 6345 6346 if (mActiveTrack != 0) { 6347 result.append("Active Track:\n"); 6348 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6349 mActiveTrack->dump(buffer, SIZE); 6350 result.append(buffer); 6351 6352 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6353 result.append(buffer); 6354 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6355 result.append(buffer); 6356 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6357 result.append(buffer); 6358 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6359 result.append(buffer); 6360 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6361 result.append(buffer); 6362 6363 6364 } else { 6365 result.append("No record client\n"); 6366 } 6367 write(fd, result.string(), result.size()); 6368 6369 dumpBase(fd, args); 6370 dumpEffectChains(fd, args); 6371 6372 return NO_ERROR; 6373} 6374 6375// AudioBufferProvider interface 6376status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6377{ 6378 size_t framesReq = buffer->frameCount; 6379 size_t framesReady = mFrameCount - mRsmpInIndex; 6380 int channelCount; 6381 6382 if (framesReady == 0) { 6383 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6384 if (mBytesRead < 0) { 6385 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6386 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6387 // Force input into standby so that it tries to 6388 // recover at next read attempt 6389 mInput->stream->common.standby(&mInput->stream->common); 6390 usleep(kRecordThreadSleepUs); 6391 } 6392 buffer->raw = NULL; 6393 buffer->frameCount = 0; 6394 return NOT_ENOUGH_DATA; 6395 } 6396 mRsmpInIndex = 0; 6397 framesReady = mFrameCount; 6398 } 6399 6400 if (framesReq > framesReady) { 6401 framesReq = framesReady; 6402 } 6403 6404 if (mChannelCount == 1 && mReqChannelCount == 2) { 6405 channelCount = 1; 6406 } else { 6407 channelCount = 2; 6408 } 6409 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6410 buffer->frameCount = framesReq; 6411 return NO_ERROR; 6412} 6413 6414// AudioBufferProvider interface 6415void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6416{ 6417 mRsmpInIndex += buffer->frameCount; 6418 buffer->frameCount = 0; 6419} 6420 6421bool AudioFlinger::RecordThread::checkForNewParameters_l() 6422{ 6423 bool reconfig = false; 6424 6425 while (!mNewParameters.isEmpty()) { 6426 status_t status = NO_ERROR; 6427 String8 keyValuePair = mNewParameters[0]; 6428 AudioParameter param = AudioParameter(keyValuePair); 6429 int value; 6430 audio_format_t reqFormat = mFormat; 6431 int reqSamplingRate = mReqSampleRate; 6432 int reqChannelCount = mReqChannelCount; 6433 6434 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6435 reqSamplingRate = value; 6436 reconfig = true; 6437 } 6438 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6439 reqFormat = (audio_format_t) value; 6440 reconfig = true; 6441 } 6442 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6443 reqChannelCount = popcount(value); 6444 reconfig = true; 6445 } 6446 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6447 // do not accept frame count changes if tracks are open as the track buffer 6448 // size depends on frame count and correct behavior would not be guaranteed 6449 // if frame count is changed after track creation 6450 if (mActiveTrack != 0) { 6451 status = INVALID_OPERATION; 6452 } else { 6453 reconfig = true; 6454 } 6455 } 6456 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6457 // forward device change to effects that have requested to be 6458 // aware of attached audio device. 6459 for (size_t i = 0; i < mEffectChains.size(); i++) { 6460 mEffectChains[i]->setDevice_l(value); 6461 } 6462 // store input device and output device but do not forward output device to audio HAL. 6463 // Note that status is ignored by the caller for output device 6464 // (see AudioFlinger::setParameters() 6465 uint32_t /*audio_devices_t*/ newDevice = mDevice; 6466 if (value & AUDIO_DEVICE_OUT_ALL) { 6467 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6468 status = BAD_VALUE; 6469 } else { 6470 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6471 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6472 if (mTrack != NULL) { 6473 bool suspend = audio_is_bluetooth_sco_device( 6474 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6475 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6476 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6477 } 6478 } 6479 newDevice |= value; 6480 mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once 6481 } 6482 if (status == NO_ERROR) { 6483 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6484 if (status == INVALID_OPERATION) { 6485 mInput->stream->common.standby(&mInput->stream->common); 6486 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6487 keyValuePair.string()); 6488 } 6489 if (reconfig) { 6490 if (status == BAD_VALUE && 6491 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6492 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6493 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6494 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6495 (reqChannelCount <= FCC_2)) { 6496 status = NO_ERROR; 6497 } 6498 if (status == NO_ERROR) { 6499 readInputParameters(); 6500 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6501 } 6502 } 6503 } 6504 6505 mNewParameters.removeAt(0); 6506 6507 mParamStatus = status; 6508 mParamCond.signal(); 6509 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6510 // already timed out waiting for the status and will never signal the condition. 6511 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6512 } 6513 return reconfig; 6514} 6515 6516String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6517{ 6518 char *s; 6519 String8 out_s8 = String8(); 6520 6521 Mutex::Autolock _l(mLock); 6522 if (initCheck() != NO_ERROR) { 6523 return out_s8; 6524 } 6525 6526 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6527 out_s8 = String8(s); 6528 free(s); 6529 return out_s8; 6530} 6531 6532void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6533 AudioSystem::OutputDescriptor desc; 6534 void *param2 = NULL; 6535 6536 switch (event) { 6537 case AudioSystem::INPUT_OPENED: 6538 case AudioSystem::INPUT_CONFIG_CHANGED: 6539 desc.channels = mChannelMask; 6540 desc.samplingRate = mSampleRate; 6541 desc.format = mFormat; 6542 desc.frameCount = mFrameCount; 6543 desc.latency = 0; 6544 param2 = &desc; 6545 break; 6546 6547 case AudioSystem::INPUT_CLOSED: 6548 default: 6549 break; 6550 } 6551 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6552} 6553 6554void AudioFlinger::RecordThread::readInputParameters() 6555{ 6556 delete mRsmpInBuffer; 6557 // mRsmpInBuffer is always assigned a new[] below 6558 delete mRsmpOutBuffer; 6559 mRsmpOutBuffer = NULL; 6560 delete mResampler; 6561 mResampler = NULL; 6562 6563 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6564 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6565 mChannelCount = (uint16_t)popcount(mChannelMask); 6566 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6567 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6568 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6569 mFrameCount = mInputBytes / mFrameSize; 6570 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6571 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6572 6573 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6574 { 6575 int channelCount; 6576 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6577 // stereo to mono post process as the resampler always outputs stereo. 6578 if (mChannelCount == 1 && mReqChannelCount == 2) { 6579 channelCount = 1; 6580 } else { 6581 channelCount = 2; 6582 } 6583 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6584 mResampler->setSampleRate(mSampleRate); 6585 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6586 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6587 6588 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6589 if (mChannelCount == 1 && mReqChannelCount == 1) { 6590 mFrameCount >>= 1; 6591 } 6592 6593 } 6594 mRsmpInIndex = mFrameCount; 6595} 6596 6597unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6598{ 6599 Mutex::Autolock _l(mLock); 6600 if (initCheck() != NO_ERROR) { 6601 return 0; 6602 } 6603 6604 return mInput->stream->get_input_frames_lost(mInput->stream); 6605} 6606 6607uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6608{ 6609 Mutex::Autolock _l(mLock); 6610 uint32_t result = 0; 6611 if (getEffectChain_l(sessionId) != 0) { 6612 result = EFFECT_SESSION; 6613 } 6614 6615 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6616 result |= TRACK_SESSION; 6617 } 6618 6619 return result; 6620} 6621 6622AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6623{ 6624 Mutex::Autolock _l(mLock); 6625 return mTrack; 6626} 6627 6628AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6629{ 6630 Mutex::Autolock _l(mLock); 6631 AudioStreamIn *input = mInput; 6632 mInput = NULL; 6633 return input; 6634} 6635 6636// this method must always be called either with ThreadBase mLock held or inside the thread loop 6637audio_stream_t* AudioFlinger::RecordThread::stream() const 6638{ 6639 if (mInput == NULL) { 6640 return NULL; 6641 } 6642 return &mInput->stream->common; 6643} 6644 6645 6646// ---------------------------------------------------------------------------- 6647 6648audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6649{ 6650 if (!settingsAllowed()) { 6651 return 0; 6652 } 6653 Mutex::Autolock _l(mLock); 6654 return loadHwModule_l(name); 6655} 6656 6657// loadHwModule_l() must be called with AudioFlinger::mLock held 6658audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6659{ 6660 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6661 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6662 ALOGW("loadHwModule() module %s already loaded", name); 6663 return mAudioHwDevs.keyAt(i); 6664 } 6665 } 6666 6667 audio_hw_device_t *dev; 6668 6669 int rc = load_audio_interface(name, &dev); 6670 if (rc) { 6671 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6672 return 0; 6673 } 6674 6675 mHardwareStatus = AUDIO_HW_INIT; 6676 rc = dev->init_check(dev); 6677 mHardwareStatus = AUDIO_HW_IDLE; 6678 if (rc) { 6679 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6680 return 0; 6681 } 6682 6683 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6684 (NULL != dev->set_master_volume)) { 6685 AutoMutex lock(mHardwareLock); 6686 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6687 dev->set_master_volume(dev, mMasterVolume); 6688 mHardwareStatus = AUDIO_HW_IDLE; 6689 } 6690 6691 audio_module_handle_t handle = nextUniqueId(); 6692 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6693 6694 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6695 name, dev->common.module->name, dev->common.module->id, handle); 6696 6697 return handle; 6698 6699} 6700 6701audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6702 audio_devices_t *pDevices, 6703 uint32_t *pSamplingRate, 6704 audio_format_t *pFormat, 6705 audio_channel_mask_t *pChannelMask, 6706 uint32_t *pLatencyMs, 6707 audio_output_flags_t flags) 6708{ 6709 status_t status; 6710 PlaybackThread *thread = NULL; 6711 struct audio_config config = { 6712 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6713 channel_mask: pChannelMask ? *pChannelMask : 0, 6714 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6715 }; 6716 audio_stream_out_t *outStream = NULL; 6717 audio_hw_device_t *outHwDev; 6718 6719 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6720 module, 6721 (pDevices != NULL) ? (int)*pDevices : 0, 6722 config.sample_rate, 6723 config.format, 6724 config.channel_mask, 6725 flags); 6726 6727 if (pDevices == NULL || *pDevices == 0) { 6728 return 0; 6729 } 6730 6731 Mutex::Autolock _l(mLock); 6732 6733 outHwDev = findSuitableHwDev_l(module, *pDevices); 6734 if (outHwDev == NULL) 6735 return 0; 6736 6737 audio_io_handle_t id = nextUniqueId(); 6738 6739 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6740 6741 status = outHwDev->open_output_stream(outHwDev, 6742 id, 6743 *pDevices, 6744 (audio_output_flags_t)flags, 6745 &config, 6746 &outStream); 6747 6748 mHardwareStatus = AUDIO_HW_IDLE; 6749 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6750 outStream, 6751 config.sample_rate, 6752 config.format, 6753 config.channel_mask, 6754 status); 6755 6756 if (status == NO_ERROR && outStream != NULL) { 6757 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6758 6759 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6760 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6761 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6762 thread = new DirectOutputThread(this, output, id, *pDevices); 6763 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6764 } else { 6765 thread = new MixerThread(this, output, id, *pDevices); 6766 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6767 } 6768 mPlaybackThreads.add(id, thread); 6769 6770 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6771 if (pFormat != NULL) *pFormat = config.format; 6772 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6773 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6774 6775 // notify client processes of the new output creation 6776 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6777 6778 // the first primary output opened designates the primary hw device 6779 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6780 ALOGI("Using module %d has the primary audio interface", module); 6781 mPrimaryHardwareDev = outHwDev; 6782 6783 AutoMutex lock(mHardwareLock); 6784 mHardwareStatus = AUDIO_HW_SET_MODE; 6785 outHwDev->set_mode(outHwDev, mMode); 6786 6787 // Determine the level of master volume support the primary audio HAL has, 6788 // and set the initial master volume at the same time. 6789 float initialVolume = 1.0; 6790 mMasterVolumeSupportLvl = MVS_NONE; 6791 6792 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6793 if ((NULL != outHwDev->get_master_volume) && 6794 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6795 mMasterVolumeSupportLvl = MVS_FULL; 6796 } else { 6797 mMasterVolumeSupportLvl = MVS_SETONLY; 6798 initialVolume = 1.0; 6799 } 6800 6801 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6802 if ((NULL == outHwDev->set_master_volume) || 6803 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6804 mMasterVolumeSupportLvl = MVS_NONE; 6805 } 6806 // now that we have a primary device, initialize master volume on other devices 6807 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6808 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6809 6810 if ((dev != mPrimaryHardwareDev) && 6811 (NULL != dev->set_master_volume)) { 6812 dev->set_master_volume(dev, initialVolume); 6813 } 6814 } 6815 mHardwareStatus = AUDIO_HW_IDLE; 6816 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6817 ? initialVolume 6818 : 1.0; 6819 mMasterVolume = initialVolume; 6820 } 6821 return id; 6822 } 6823 6824 return 0; 6825} 6826 6827audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6828 audio_io_handle_t output2) 6829{ 6830 Mutex::Autolock _l(mLock); 6831 MixerThread *thread1 = checkMixerThread_l(output1); 6832 MixerThread *thread2 = checkMixerThread_l(output2); 6833 6834 if (thread1 == NULL || thread2 == NULL) { 6835 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6836 return 0; 6837 } 6838 6839 audio_io_handle_t id = nextUniqueId(); 6840 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6841 thread->addOutputTrack(thread2); 6842 mPlaybackThreads.add(id, thread); 6843 // notify client processes of the new output creation 6844 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6845 return id; 6846} 6847 6848status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6849{ 6850 // keep strong reference on the playback thread so that 6851 // it is not destroyed while exit() is executed 6852 sp<PlaybackThread> thread; 6853 { 6854 Mutex::Autolock _l(mLock); 6855 thread = checkPlaybackThread_l(output); 6856 if (thread == NULL) { 6857 return BAD_VALUE; 6858 } 6859 6860 ALOGV("closeOutput() %d", output); 6861 6862 if (thread->type() == ThreadBase::MIXER) { 6863 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6864 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6865 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6866 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6867 } 6868 } 6869 } 6870 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6871 mPlaybackThreads.removeItem(output); 6872 } 6873 thread->exit(); 6874 // The thread entity (active unit of execution) is no longer running here, 6875 // but the ThreadBase container still exists. 6876 6877 if (thread->type() != ThreadBase::DUPLICATING) { 6878 AudioStreamOut *out = thread->clearOutput(); 6879 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6880 // from now on thread->mOutput is NULL 6881 out->hwDev->close_output_stream(out->hwDev, out->stream); 6882 delete out; 6883 } 6884 return NO_ERROR; 6885} 6886 6887status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6888{ 6889 Mutex::Autolock _l(mLock); 6890 PlaybackThread *thread = checkPlaybackThread_l(output); 6891 6892 if (thread == NULL) { 6893 return BAD_VALUE; 6894 } 6895 6896 ALOGV("suspendOutput() %d", output); 6897 thread->suspend(); 6898 6899 return NO_ERROR; 6900} 6901 6902status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6903{ 6904 Mutex::Autolock _l(mLock); 6905 PlaybackThread *thread = checkPlaybackThread_l(output); 6906 6907 if (thread == NULL) { 6908 return BAD_VALUE; 6909 } 6910 6911 ALOGV("restoreOutput() %d", output); 6912 6913 thread->restore(); 6914 6915 return NO_ERROR; 6916} 6917 6918audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6919 audio_devices_t *pDevices, 6920 uint32_t *pSamplingRate, 6921 audio_format_t *pFormat, 6922 audio_channel_mask_t *pChannelMask) 6923{ 6924 status_t status; 6925 RecordThread *thread = NULL; 6926 struct audio_config config = { 6927 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6928 channel_mask: pChannelMask ? *pChannelMask : 0, 6929 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6930 }; 6931 uint32_t reqSamplingRate = config.sample_rate; 6932 audio_format_t reqFormat = config.format; 6933 audio_channel_mask_t reqChannels = config.channel_mask; 6934 audio_stream_in_t *inStream = NULL; 6935 audio_hw_device_t *inHwDev; 6936 6937 if (pDevices == NULL || *pDevices == 0) { 6938 return 0; 6939 } 6940 6941 Mutex::Autolock _l(mLock); 6942 6943 inHwDev = findSuitableHwDev_l(module, *pDevices); 6944 if (inHwDev == NULL) 6945 return 0; 6946 6947 audio_io_handle_t id = nextUniqueId(); 6948 6949 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6950 &inStream); 6951 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6952 inStream, 6953 config.sample_rate, 6954 config.format, 6955 config.channel_mask, 6956 status); 6957 6958 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6959 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6960 // or stereo to mono conversions on 16 bit PCM inputs. 6961 if (status == BAD_VALUE && 6962 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6963 (config.sample_rate <= 2 * reqSamplingRate) && 6964 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6965 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 6966 inStream = NULL; 6967 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6968 } 6969 6970 if (status == NO_ERROR && inStream != NULL) { 6971 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6972 6973 // Start record thread 6974 // RecorThread require both input and output device indication to forward to audio 6975 // pre processing modules 6976 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6977 thread = new RecordThread(this, 6978 input, 6979 reqSamplingRate, 6980 reqChannels, 6981 id, 6982 device); 6983 mRecordThreads.add(id, thread); 6984 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6985 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6986 if (pFormat != NULL) *pFormat = config.format; 6987 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6988 6989 input->stream->common.standby(&input->stream->common); 6990 6991 // notify client processes of the new input creation 6992 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6993 return id; 6994 } 6995 6996 return 0; 6997} 6998 6999status_t AudioFlinger::closeInput(audio_io_handle_t input) 7000{ 7001 // keep strong reference on the record thread so that 7002 // it is not destroyed while exit() is executed 7003 sp<RecordThread> thread; 7004 { 7005 Mutex::Autolock _l(mLock); 7006 thread = checkRecordThread_l(input); 7007 if (thread == 0) { 7008 return BAD_VALUE; 7009 } 7010 7011 ALOGV("closeInput() %d", input); 7012 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7013 mRecordThreads.removeItem(input); 7014 } 7015 thread->exit(); 7016 // The thread entity (active unit of execution) is no longer running here, 7017 // but the ThreadBase container still exists. 7018 7019 AudioStreamIn *in = thread->clearInput(); 7020 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7021 // from now on thread->mInput is NULL 7022 in->hwDev->close_input_stream(in->hwDev, in->stream); 7023 delete in; 7024 7025 return NO_ERROR; 7026} 7027 7028status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7029{ 7030 Mutex::Autolock _l(mLock); 7031 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7032 7033 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7034 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7035 thread->invalidateTracks(stream); 7036 } 7037 7038 return NO_ERROR; 7039} 7040 7041 7042int AudioFlinger::newAudioSessionId() 7043{ 7044 return nextUniqueId(); 7045} 7046 7047void AudioFlinger::acquireAudioSessionId(int audioSession) 7048{ 7049 Mutex::Autolock _l(mLock); 7050 pid_t caller = IPCThreadState::self()->getCallingPid(); 7051 ALOGV("acquiring %d from %d", audioSession, caller); 7052 size_t num = mAudioSessionRefs.size(); 7053 for (size_t i = 0; i< num; i++) { 7054 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7055 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7056 ref->mCnt++; 7057 ALOGV(" incremented refcount to %d", ref->mCnt); 7058 return; 7059 } 7060 } 7061 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7062 ALOGV(" added new entry for %d", audioSession); 7063} 7064 7065void AudioFlinger::releaseAudioSessionId(int audioSession) 7066{ 7067 Mutex::Autolock _l(mLock); 7068 pid_t caller = IPCThreadState::self()->getCallingPid(); 7069 ALOGV("releasing %d from %d", audioSession, caller); 7070 size_t num = mAudioSessionRefs.size(); 7071 for (size_t i = 0; i< num; i++) { 7072 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7073 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7074 ref->mCnt--; 7075 ALOGV(" decremented refcount to %d", ref->mCnt); 7076 if (ref->mCnt == 0) { 7077 mAudioSessionRefs.removeAt(i); 7078 delete ref; 7079 purgeStaleEffects_l(); 7080 } 7081 return; 7082 } 7083 } 7084 ALOGW("session id %d not found for pid %d", audioSession, caller); 7085} 7086 7087void AudioFlinger::purgeStaleEffects_l() { 7088 7089 ALOGV("purging stale effects"); 7090 7091 Vector< sp<EffectChain> > chains; 7092 7093 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7094 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7095 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7096 sp<EffectChain> ec = t->mEffectChains[j]; 7097 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7098 chains.push(ec); 7099 } 7100 } 7101 } 7102 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7103 sp<RecordThread> t = mRecordThreads.valueAt(i); 7104 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7105 sp<EffectChain> ec = t->mEffectChains[j]; 7106 chains.push(ec); 7107 } 7108 } 7109 7110 for (size_t i = 0; i < chains.size(); i++) { 7111 sp<EffectChain> ec = chains[i]; 7112 int sessionid = ec->sessionId(); 7113 sp<ThreadBase> t = ec->mThread.promote(); 7114 if (t == 0) { 7115 continue; 7116 } 7117 size_t numsessionrefs = mAudioSessionRefs.size(); 7118 bool found = false; 7119 for (size_t k = 0; k < numsessionrefs; k++) { 7120 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7121 if (ref->mSessionid == sessionid) { 7122 ALOGV(" session %d still exists for %d with %d refs", 7123 sessionid, ref->mPid, ref->mCnt); 7124 found = true; 7125 break; 7126 } 7127 } 7128 if (!found) { 7129 Mutex::Autolock _l (t->mLock); 7130 // remove all effects from the chain 7131 while (ec->mEffects.size()) { 7132 sp<EffectModule> effect = ec->mEffects[0]; 7133 effect->unPin(); 7134 t->removeEffect_l(effect); 7135 if (effect->purgeHandles()) { 7136 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7137 } 7138 AudioSystem::unregisterEffect(effect->id()); 7139 } 7140 } 7141 } 7142 return; 7143} 7144 7145// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7146AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7147{ 7148 return mPlaybackThreads.valueFor(output).get(); 7149} 7150 7151// checkMixerThread_l() must be called with AudioFlinger::mLock held 7152AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7153{ 7154 PlaybackThread *thread = checkPlaybackThread_l(output); 7155 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7156} 7157 7158// checkRecordThread_l() must be called with AudioFlinger::mLock held 7159AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7160{ 7161 return mRecordThreads.valueFor(input).get(); 7162} 7163 7164uint32_t AudioFlinger::nextUniqueId() 7165{ 7166 return android_atomic_inc(&mNextUniqueId); 7167} 7168 7169AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7170{ 7171 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7172 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7173 AudioStreamOut *output = thread->getOutput(); 7174 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7175 return thread; 7176 } 7177 } 7178 return NULL; 7179} 7180 7181uint32_t AudioFlinger::primaryOutputDevice_l() const 7182{ 7183 PlaybackThread *thread = primaryPlaybackThread_l(); 7184 7185 if (thread == NULL) { 7186 return 0; 7187 } 7188 7189 return thread->device(); 7190} 7191 7192sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7193 int triggerSession, 7194 int listenerSession, 7195 sync_event_callback_t callBack, 7196 void *cookie) 7197{ 7198 Mutex::Autolock _l(mLock); 7199 7200 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7201 status_t playStatus = NAME_NOT_FOUND; 7202 status_t recStatus = NAME_NOT_FOUND; 7203 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7204 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7205 if (playStatus == NO_ERROR) { 7206 return event; 7207 } 7208 } 7209 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7210 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7211 if (recStatus == NO_ERROR) { 7212 return event; 7213 } 7214 } 7215 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7216 mPendingSyncEvents.add(event); 7217 } else { 7218 ALOGV("createSyncEvent() invalid event %d", event->type()); 7219 event.clear(); 7220 } 7221 return event; 7222} 7223 7224// ---------------------------------------------------------------------------- 7225// Effect management 7226// ---------------------------------------------------------------------------- 7227 7228 7229status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7230{ 7231 Mutex::Autolock _l(mLock); 7232 return EffectQueryNumberEffects(numEffects); 7233} 7234 7235status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7236{ 7237 Mutex::Autolock _l(mLock); 7238 return EffectQueryEffect(index, descriptor); 7239} 7240 7241status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7242 effect_descriptor_t *descriptor) const 7243{ 7244 Mutex::Autolock _l(mLock); 7245 return EffectGetDescriptor(pUuid, descriptor); 7246} 7247 7248 7249sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7250 effect_descriptor_t *pDesc, 7251 const sp<IEffectClient>& effectClient, 7252 int32_t priority, 7253 audio_io_handle_t io, 7254 int sessionId, 7255 status_t *status, 7256 int *id, 7257 int *enabled) 7258{ 7259 status_t lStatus = NO_ERROR; 7260 sp<EffectHandle> handle; 7261 effect_descriptor_t desc; 7262 7263 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7264 pid, effectClient.get(), priority, sessionId, io); 7265 7266 if (pDesc == NULL) { 7267 lStatus = BAD_VALUE; 7268 goto Exit; 7269 } 7270 7271 // check audio settings permission for global effects 7272 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7273 lStatus = PERMISSION_DENIED; 7274 goto Exit; 7275 } 7276 7277 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7278 // that can only be created by audio policy manager (running in same process) 7279 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7280 lStatus = PERMISSION_DENIED; 7281 goto Exit; 7282 } 7283 7284 if (io == 0) { 7285 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7286 // output must be specified by AudioPolicyManager when using session 7287 // AUDIO_SESSION_OUTPUT_STAGE 7288 lStatus = BAD_VALUE; 7289 goto Exit; 7290 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7291 // if the output returned by getOutputForEffect() is removed before we lock the 7292 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7293 // and we will exit safely 7294 io = AudioSystem::getOutputForEffect(&desc); 7295 } 7296 } 7297 7298 { 7299 Mutex::Autolock _l(mLock); 7300 7301 7302 if (!EffectIsNullUuid(&pDesc->uuid)) { 7303 // if uuid is specified, request effect descriptor 7304 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7305 if (lStatus < 0) { 7306 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7307 goto Exit; 7308 } 7309 } else { 7310 // if uuid is not specified, look for an available implementation 7311 // of the required type in effect factory 7312 if (EffectIsNullUuid(&pDesc->type)) { 7313 ALOGW("createEffect() no effect type"); 7314 lStatus = BAD_VALUE; 7315 goto Exit; 7316 } 7317 uint32_t numEffects = 0; 7318 effect_descriptor_t d; 7319 d.flags = 0; // prevent compiler warning 7320 bool found = false; 7321 7322 lStatus = EffectQueryNumberEffects(&numEffects); 7323 if (lStatus < 0) { 7324 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7325 goto Exit; 7326 } 7327 for (uint32_t i = 0; i < numEffects; i++) { 7328 lStatus = EffectQueryEffect(i, &desc); 7329 if (lStatus < 0) { 7330 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7331 continue; 7332 } 7333 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7334 // If matching type found save effect descriptor. If the session is 7335 // 0 and the effect is not auxiliary, continue enumeration in case 7336 // an auxiliary version of this effect type is available 7337 found = true; 7338 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7339 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7340 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7341 break; 7342 } 7343 } 7344 } 7345 if (!found) { 7346 lStatus = BAD_VALUE; 7347 ALOGW("createEffect() effect not found"); 7348 goto Exit; 7349 } 7350 // For same effect type, chose auxiliary version over insert version if 7351 // connect to output mix (Compliance to OpenSL ES) 7352 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7353 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7354 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7355 } 7356 } 7357 7358 // Do not allow auxiliary effects on a session different from 0 (output mix) 7359 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7360 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7361 lStatus = INVALID_OPERATION; 7362 goto Exit; 7363 } 7364 7365 // check recording permission for visualizer 7366 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7367 !recordingAllowed()) { 7368 lStatus = PERMISSION_DENIED; 7369 goto Exit; 7370 } 7371 7372 // return effect descriptor 7373 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7374 7375 // If output is not specified try to find a matching audio session ID in one of the 7376 // output threads. 7377 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7378 // because of code checking output when entering the function. 7379 // Note: io is never 0 when creating an effect on an input 7380 if (io == 0) { 7381 // look for the thread where the specified audio session is present 7382 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7383 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7384 io = mPlaybackThreads.keyAt(i); 7385 break; 7386 } 7387 } 7388 if (io == 0) { 7389 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7390 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7391 io = mRecordThreads.keyAt(i); 7392 break; 7393 } 7394 } 7395 } 7396 // If no output thread contains the requested session ID, default to 7397 // first output. The effect chain will be moved to the correct output 7398 // thread when a track with the same session ID is created 7399 if (io == 0 && mPlaybackThreads.size()) { 7400 io = mPlaybackThreads.keyAt(0); 7401 } 7402 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7403 } 7404 ThreadBase *thread = checkRecordThread_l(io); 7405 if (thread == NULL) { 7406 thread = checkPlaybackThread_l(io); 7407 if (thread == NULL) { 7408 ALOGE("createEffect() unknown output thread"); 7409 lStatus = BAD_VALUE; 7410 goto Exit; 7411 } 7412 } 7413 7414 sp<Client> client = registerPid_l(pid); 7415 7416 // create effect on selected output thread 7417 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7418 &desc, enabled, &lStatus); 7419 if (handle != 0 && id != NULL) { 7420 *id = handle->id(); 7421 } 7422 } 7423 7424Exit: 7425 if (status != NULL) { 7426 *status = lStatus; 7427 } 7428 return handle; 7429} 7430 7431status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7432 audio_io_handle_t dstOutput) 7433{ 7434 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7435 sessionId, srcOutput, dstOutput); 7436 Mutex::Autolock _l(mLock); 7437 if (srcOutput == dstOutput) { 7438 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7439 return NO_ERROR; 7440 } 7441 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7442 if (srcThread == NULL) { 7443 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7444 return BAD_VALUE; 7445 } 7446 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7447 if (dstThread == NULL) { 7448 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7449 return BAD_VALUE; 7450 } 7451 7452 Mutex::Autolock _dl(dstThread->mLock); 7453 Mutex::Autolock _sl(srcThread->mLock); 7454 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7455 7456 return NO_ERROR; 7457} 7458 7459// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7460status_t AudioFlinger::moveEffectChain_l(int sessionId, 7461 AudioFlinger::PlaybackThread *srcThread, 7462 AudioFlinger::PlaybackThread *dstThread, 7463 bool reRegister) 7464{ 7465 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7466 sessionId, srcThread, dstThread); 7467 7468 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7469 if (chain == 0) { 7470 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7471 sessionId, srcThread); 7472 return INVALID_OPERATION; 7473 } 7474 7475 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7476 // so that a new chain is created with correct parameters when first effect is added. This is 7477 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7478 // removed. 7479 srcThread->removeEffectChain_l(chain); 7480 7481 // transfer all effects one by one so that new effect chain is created on new thread with 7482 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7483 audio_io_handle_t dstOutput = dstThread->id(); 7484 sp<EffectChain> dstChain; 7485 uint32_t strategy = 0; // prevent compiler warning 7486 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7487 while (effect != 0) { 7488 srcThread->removeEffect_l(effect); 7489 dstThread->addEffect_l(effect); 7490 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7491 if (effect->state() == EffectModule::ACTIVE || 7492 effect->state() == EffectModule::STOPPING) { 7493 effect->start(); 7494 } 7495 // if the move request is not received from audio policy manager, the effect must be 7496 // re-registered with the new strategy and output 7497 if (dstChain == 0) { 7498 dstChain = effect->chain().promote(); 7499 if (dstChain == 0) { 7500 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7501 srcThread->addEffect_l(effect); 7502 return NO_INIT; 7503 } 7504 strategy = dstChain->strategy(); 7505 } 7506 if (reRegister) { 7507 AudioSystem::unregisterEffect(effect->id()); 7508 AudioSystem::registerEffect(&effect->desc(), 7509 dstOutput, 7510 strategy, 7511 sessionId, 7512 effect->id()); 7513 } 7514 effect = chain->getEffectFromId_l(0); 7515 } 7516 7517 return NO_ERROR; 7518} 7519 7520 7521// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7522sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7523 const sp<AudioFlinger::Client>& client, 7524 const sp<IEffectClient>& effectClient, 7525 int32_t priority, 7526 int sessionId, 7527 effect_descriptor_t *desc, 7528 int *enabled, 7529 status_t *status 7530 ) 7531{ 7532 sp<EffectModule> effect; 7533 sp<EffectHandle> handle; 7534 status_t lStatus; 7535 sp<EffectChain> chain; 7536 bool chainCreated = false; 7537 bool effectCreated = false; 7538 bool effectRegistered = false; 7539 7540 lStatus = initCheck(); 7541 if (lStatus != NO_ERROR) { 7542 ALOGW("createEffect_l() Audio driver not initialized."); 7543 goto Exit; 7544 } 7545 7546 // Do not allow effects with session ID 0 on direct output or duplicating threads 7547 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7548 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7549 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7550 desc->name, sessionId); 7551 lStatus = BAD_VALUE; 7552 goto Exit; 7553 } 7554 // Only Pre processor effects are allowed on input threads and only on input threads 7555 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7556 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7557 desc->name, desc->flags, mType); 7558 lStatus = BAD_VALUE; 7559 goto Exit; 7560 } 7561 7562 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7563 7564 { // scope for mLock 7565 Mutex::Autolock _l(mLock); 7566 7567 // check for existing effect chain with the requested audio session 7568 chain = getEffectChain_l(sessionId); 7569 if (chain == 0) { 7570 // create a new chain for this session 7571 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7572 chain = new EffectChain(this, sessionId); 7573 addEffectChain_l(chain); 7574 chain->setStrategy(getStrategyForSession_l(sessionId)); 7575 chainCreated = true; 7576 } else { 7577 effect = chain->getEffectFromDesc_l(desc); 7578 } 7579 7580 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7581 7582 if (effect == 0) { 7583 int id = mAudioFlinger->nextUniqueId(); 7584 // Check CPU and memory usage 7585 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7586 if (lStatus != NO_ERROR) { 7587 goto Exit; 7588 } 7589 effectRegistered = true; 7590 // create a new effect module if none present in the chain 7591 effect = new EffectModule(this, chain, desc, id, sessionId); 7592 lStatus = effect->status(); 7593 if (lStatus != NO_ERROR) { 7594 goto Exit; 7595 } 7596 lStatus = chain->addEffect_l(effect); 7597 if (lStatus != NO_ERROR) { 7598 goto Exit; 7599 } 7600 effectCreated = true; 7601 7602 effect->setDevice(mDevice); 7603 effect->setMode(mAudioFlinger->getMode()); 7604 } 7605 // create effect handle and connect it to effect module 7606 handle = new EffectHandle(effect, client, effectClient, priority); 7607 lStatus = effect->addHandle(handle.get()); 7608 if (enabled != NULL) { 7609 *enabled = (int)effect->isEnabled(); 7610 } 7611 } 7612 7613Exit: 7614 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7615 Mutex::Autolock _l(mLock); 7616 if (effectCreated) { 7617 chain->removeEffect_l(effect); 7618 } 7619 if (effectRegistered) { 7620 AudioSystem::unregisterEffect(effect->id()); 7621 } 7622 if (chainCreated) { 7623 removeEffectChain_l(chain); 7624 } 7625 handle.clear(); 7626 } 7627 7628 if (status != NULL) { 7629 *status = lStatus; 7630 } 7631 return handle; 7632} 7633 7634sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7635{ 7636 Mutex::Autolock _l(mLock); 7637 return getEffect_l(sessionId, effectId); 7638} 7639 7640sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7641{ 7642 sp<EffectChain> chain = getEffectChain_l(sessionId); 7643 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7644} 7645 7646// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7647// PlaybackThread::mLock held 7648status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7649{ 7650 // check for existing effect chain with the requested audio session 7651 int sessionId = effect->sessionId(); 7652 sp<EffectChain> chain = getEffectChain_l(sessionId); 7653 bool chainCreated = false; 7654 7655 if (chain == 0) { 7656 // create a new chain for this session 7657 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7658 chain = new EffectChain(this, sessionId); 7659 addEffectChain_l(chain); 7660 chain->setStrategy(getStrategyForSession_l(sessionId)); 7661 chainCreated = true; 7662 } 7663 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7664 7665 if (chain->getEffectFromId_l(effect->id()) != 0) { 7666 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7667 this, effect->desc().name, chain.get()); 7668 return BAD_VALUE; 7669 } 7670 7671 status_t status = chain->addEffect_l(effect); 7672 if (status != NO_ERROR) { 7673 if (chainCreated) { 7674 removeEffectChain_l(chain); 7675 } 7676 return status; 7677 } 7678 7679 effect->setDevice(mDevice); 7680 effect->setMode(mAudioFlinger->getMode()); 7681 return NO_ERROR; 7682} 7683 7684void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7685 7686 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7687 effect_descriptor_t desc = effect->desc(); 7688 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7689 detachAuxEffect_l(effect->id()); 7690 } 7691 7692 sp<EffectChain> chain = effect->chain().promote(); 7693 if (chain != 0) { 7694 // remove effect chain if removing last effect 7695 if (chain->removeEffect_l(effect) == 0) { 7696 removeEffectChain_l(chain); 7697 } 7698 } else { 7699 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7700 } 7701} 7702 7703void AudioFlinger::ThreadBase::lockEffectChains_l( 7704 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7705{ 7706 effectChains = mEffectChains; 7707 for (size_t i = 0; i < mEffectChains.size(); i++) { 7708 mEffectChains[i]->lock(); 7709 } 7710} 7711 7712void AudioFlinger::ThreadBase::unlockEffectChains( 7713 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7714{ 7715 for (size_t i = 0; i < effectChains.size(); i++) { 7716 effectChains[i]->unlock(); 7717 } 7718} 7719 7720sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7721{ 7722 Mutex::Autolock _l(mLock); 7723 return getEffectChain_l(sessionId); 7724} 7725 7726sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7727{ 7728 size_t size = mEffectChains.size(); 7729 for (size_t i = 0; i < size; i++) { 7730 if (mEffectChains[i]->sessionId() == sessionId) { 7731 return mEffectChains[i]; 7732 } 7733 } 7734 return 0; 7735} 7736 7737void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7738{ 7739 Mutex::Autolock _l(mLock); 7740 size_t size = mEffectChains.size(); 7741 for (size_t i = 0; i < size; i++) { 7742 mEffectChains[i]->setMode_l(mode); 7743 } 7744} 7745 7746void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7747 EffectHandle *handle, 7748 bool unpinIfLast) { 7749 7750 Mutex::Autolock _l(mLock); 7751 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7752 // delete the effect module if removing last handle on it 7753 if (effect->removeHandle(handle) == 0) { 7754 if (!effect->isPinned() || unpinIfLast) { 7755 removeEffect_l(effect); 7756 AudioSystem::unregisterEffect(effect->id()); 7757 } 7758 } 7759} 7760 7761status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7762{ 7763 int session = chain->sessionId(); 7764 int16_t *buffer = mMixBuffer; 7765 bool ownsBuffer = false; 7766 7767 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7768 if (session > 0) { 7769 // Only one effect chain can be present in direct output thread and it uses 7770 // the mix buffer as input 7771 if (mType != DIRECT) { 7772 size_t numSamples = mNormalFrameCount * mChannelCount; 7773 buffer = new int16_t[numSamples]; 7774 memset(buffer, 0, numSamples * sizeof(int16_t)); 7775 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7776 ownsBuffer = true; 7777 } 7778 7779 // Attach all tracks with same session ID to this chain. 7780 for (size_t i = 0; i < mTracks.size(); ++i) { 7781 sp<Track> track = mTracks[i]; 7782 if (session == track->sessionId()) { 7783 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7784 track->setMainBuffer(buffer); 7785 chain->incTrackCnt(); 7786 } 7787 } 7788 7789 // indicate all active tracks in the chain 7790 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7791 sp<Track> track = mActiveTracks[i].promote(); 7792 if (track == 0) continue; 7793 if (session == track->sessionId()) { 7794 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7795 chain->incActiveTrackCnt(); 7796 } 7797 } 7798 } 7799 7800 chain->setInBuffer(buffer, ownsBuffer); 7801 chain->setOutBuffer(mMixBuffer); 7802 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7803 // chains list in order to be processed last as it contains output stage effects 7804 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7805 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7806 // after track specific effects and before output stage 7807 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7808 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7809 // Effect chain for other sessions are inserted at beginning of effect 7810 // chains list to be processed before output mix effects. Relative order between other 7811 // sessions is not important 7812 size_t size = mEffectChains.size(); 7813 size_t i = 0; 7814 for (i = 0; i < size; i++) { 7815 if (mEffectChains[i]->sessionId() < session) break; 7816 } 7817 mEffectChains.insertAt(chain, i); 7818 checkSuspendOnAddEffectChain_l(chain); 7819 7820 return NO_ERROR; 7821} 7822 7823size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7824{ 7825 int session = chain->sessionId(); 7826 7827 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7828 7829 for (size_t i = 0; i < mEffectChains.size(); i++) { 7830 if (chain == mEffectChains[i]) { 7831 mEffectChains.removeAt(i); 7832 // detach all active tracks from the chain 7833 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7834 sp<Track> track = mActiveTracks[i].promote(); 7835 if (track == 0) continue; 7836 if (session == track->sessionId()) { 7837 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7838 chain.get(), session); 7839 chain->decActiveTrackCnt(); 7840 } 7841 } 7842 7843 // detach all tracks with same session ID from this chain 7844 for (size_t i = 0; i < mTracks.size(); ++i) { 7845 sp<Track> track = mTracks[i]; 7846 if (session == track->sessionId()) { 7847 track->setMainBuffer(mMixBuffer); 7848 chain->decTrackCnt(); 7849 } 7850 } 7851 break; 7852 } 7853 } 7854 return mEffectChains.size(); 7855} 7856 7857status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7858 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7859{ 7860 Mutex::Autolock _l(mLock); 7861 return attachAuxEffect_l(track, EffectId); 7862} 7863 7864status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7865 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7866{ 7867 status_t status = NO_ERROR; 7868 7869 if (EffectId == 0) { 7870 track->setAuxBuffer(0, NULL); 7871 } else { 7872 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7873 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7874 if (effect != 0) { 7875 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7876 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7877 } else { 7878 status = INVALID_OPERATION; 7879 } 7880 } else { 7881 status = BAD_VALUE; 7882 } 7883 } 7884 return status; 7885} 7886 7887void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7888{ 7889 for (size_t i = 0; i < mTracks.size(); ++i) { 7890 sp<Track> track = mTracks[i]; 7891 if (track->auxEffectId() == effectId) { 7892 attachAuxEffect_l(track, 0); 7893 } 7894 } 7895} 7896 7897status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7898{ 7899 // only one chain per input thread 7900 if (mEffectChains.size() != 0) { 7901 return INVALID_OPERATION; 7902 } 7903 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7904 7905 chain->setInBuffer(NULL); 7906 chain->setOutBuffer(NULL); 7907 7908 checkSuspendOnAddEffectChain_l(chain); 7909 7910 mEffectChains.add(chain); 7911 7912 return NO_ERROR; 7913} 7914 7915size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7916{ 7917 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7918 ALOGW_IF(mEffectChains.size() != 1, 7919 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7920 chain.get(), mEffectChains.size(), this); 7921 if (mEffectChains.size() == 1) { 7922 mEffectChains.removeAt(0); 7923 } 7924 return 0; 7925} 7926 7927// ---------------------------------------------------------------------------- 7928// EffectModule implementation 7929// ---------------------------------------------------------------------------- 7930 7931#undef LOG_TAG 7932#define LOG_TAG "AudioFlinger::EffectModule" 7933 7934AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7935 const wp<AudioFlinger::EffectChain>& chain, 7936 effect_descriptor_t *desc, 7937 int id, 7938 int sessionId) 7939 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7940 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7941 // mDescriptor is set below 7942 // mConfig is set by configure() and not used before then 7943 mEffectInterface(NULL), 7944 mStatus(NO_INIT), mState(IDLE), 7945 // mMaxDisableWaitCnt is set by configure() and not used before then 7946 // mDisableWaitCnt is set by process() and updateState() and not used before then 7947 mSuspended(false) 7948{ 7949 ALOGV("Constructor %p", this); 7950 int lStatus; 7951 if (thread == NULL) { 7952 return; 7953 } 7954 7955 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7956 7957 // create effect engine from effect factory 7958 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7959 7960 if (mStatus != NO_ERROR) { 7961 return; 7962 } 7963 lStatus = init(); 7964 if (lStatus < 0) { 7965 mStatus = lStatus; 7966 goto Error; 7967 } 7968 7969 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7970 return; 7971Error: 7972 EffectRelease(mEffectInterface); 7973 mEffectInterface = NULL; 7974 ALOGV("Constructor Error %d", mStatus); 7975} 7976 7977AudioFlinger::EffectModule::~EffectModule() 7978{ 7979 ALOGV("Destructor %p", this); 7980 if (mEffectInterface != NULL) { 7981 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7982 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7983 sp<ThreadBase> thread = mThread.promote(); 7984 if (thread != 0) { 7985 audio_stream_t *stream = thread->stream(); 7986 if (stream != NULL) { 7987 stream->remove_audio_effect(stream, mEffectInterface); 7988 } 7989 } 7990 } 7991 // release effect engine 7992 EffectRelease(mEffectInterface); 7993 } 7994} 7995 7996status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 7997{ 7998 status_t status; 7999 8000 Mutex::Autolock _l(mLock); 8001 int priority = handle->priority(); 8002 size_t size = mHandles.size(); 8003 EffectHandle *controlHandle = NULL; 8004 size_t i; 8005 for (i = 0; i < size; i++) { 8006 EffectHandle *h = mHandles[i]; 8007 if (h == NULL || h->destroyed_l()) continue; 8008 // first non destroyed handle is considered in control 8009 if (controlHandle == NULL) 8010 controlHandle = h; 8011 if (h->priority() <= priority) break; 8012 } 8013 // if inserted in first place, move effect control from previous owner to this handle 8014 if (i == 0) { 8015 bool enabled = false; 8016 if (controlHandle != NULL) { 8017 enabled = controlHandle->enabled(); 8018 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8019 } 8020 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8021 status = NO_ERROR; 8022 } else { 8023 status = ALREADY_EXISTS; 8024 } 8025 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8026 mHandles.insertAt(handle, i); 8027 return status; 8028} 8029 8030size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8031{ 8032 Mutex::Autolock _l(mLock); 8033 size_t size = mHandles.size(); 8034 size_t i; 8035 for (i = 0; i < size; i++) { 8036 if (mHandles[i] == handle) break; 8037 } 8038 if (i == size) { 8039 return size; 8040 } 8041 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8042 8043 mHandles.removeAt(i); 8044 // if removed from first place, move effect control from this handle to next in line 8045 if (i == 0) { 8046 EffectHandle *h = controlHandle_l(); 8047 if (h != NULL) { 8048 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8049 } 8050 } 8051 8052 // Prevent calls to process() and other functions on effect interface from now on. 8053 // The effect engine will be released by the destructor when the last strong reference on 8054 // this object is released which can happen after next process is called. 8055 if (mHandles.size() == 0 && !mPinned) { 8056 mState = DESTROYED; 8057 } 8058 8059 return size; 8060} 8061 8062// must be called with EffectModule::mLock held 8063AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8064{ 8065 // the first valid handle in the list has control over the module 8066 for (size_t i = 0; i < mHandles.size(); i++) { 8067 EffectHandle *h = mHandles[i]; 8068 if (h != NULL && !h->destroyed_l()) { 8069 return h; 8070 } 8071 } 8072 8073 return NULL; 8074} 8075 8076size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8077{ 8078 ALOGV("disconnect() %p handle %p", this, handle); 8079 // keep a strong reference on this EffectModule to avoid calling the 8080 // destructor before we exit 8081 sp<EffectModule> keep(this); 8082 { 8083 sp<ThreadBase> thread = mThread.promote(); 8084 if (thread != 0) { 8085 thread->disconnectEffect(keep, handle, unpinIfLast); 8086 } 8087 } 8088 return mHandles.size(); 8089} 8090 8091void AudioFlinger::EffectModule::updateState() { 8092 Mutex::Autolock _l(mLock); 8093 8094 switch (mState) { 8095 case RESTART: 8096 reset_l(); 8097 // FALL THROUGH 8098 8099 case STARTING: 8100 // clear auxiliary effect input buffer for next accumulation 8101 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8102 memset(mConfig.inputCfg.buffer.raw, 8103 0, 8104 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8105 } 8106 start_l(); 8107 mState = ACTIVE; 8108 break; 8109 case STOPPING: 8110 stop_l(); 8111 mDisableWaitCnt = mMaxDisableWaitCnt; 8112 mState = STOPPED; 8113 break; 8114 case STOPPED: 8115 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8116 // turn off sequence. 8117 if (--mDisableWaitCnt == 0) { 8118 reset_l(); 8119 mState = IDLE; 8120 } 8121 break; 8122 default: //IDLE , ACTIVE, DESTROYED 8123 break; 8124 } 8125} 8126 8127void AudioFlinger::EffectModule::process() 8128{ 8129 Mutex::Autolock _l(mLock); 8130 8131 if (mState == DESTROYED || mEffectInterface == NULL || 8132 mConfig.inputCfg.buffer.raw == NULL || 8133 mConfig.outputCfg.buffer.raw == NULL) { 8134 return; 8135 } 8136 8137 if (isProcessEnabled()) { 8138 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8139 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8140 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8141 mConfig.inputCfg.buffer.s32, 8142 mConfig.inputCfg.buffer.frameCount/2); 8143 } 8144 8145 // do the actual processing in the effect engine 8146 int ret = (*mEffectInterface)->process(mEffectInterface, 8147 &mConfig.inputCfg.buffer, 8148 &mConfig.outputCfg.buffer); 8149 8150 // force transition to IDLE state when engine is ready 8151 if (mState == STOPPED && ret == -ENODATA) { 8152 mDisableWaitCnt = 1; 8153 } 8154 8155 // clear auxiliary effect input buffer for next accumulation 8156 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8157 memset(mConfig.inputCfg.buffer.raw, 0, 8158 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8159 } 8160 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8161 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8162 // If an insert effect is idle and input buffer is different from output buffer, 8163 // accumulate input onto output 8164 sp<EffectChain> chain = mChain.promote(); 8165 if (chain != 0 && chain->activeTrackCnt() != 0) { 8166 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8167 int16_t *in = mConfig.inputCfg.buffer.s16; 8168 int16_t *out = mConfig.outputCfg.buffer.s16; 8169 for (size_t i = 0; i < frameCnt; i++) { 8170 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8171 } 8172 } 8173 } 8174} 8175 8176void AudioFlinger::EffectModule::reset_l() 8177{ 8178 if (mEffectInterface == NULL) { 8179 return; 8180 } 8181 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8182} 8183 8184status_t AudioFlinger::EffectModule::configure() 8185{ 8186 if (mEffectInterface == NULL) { 8187 return NO_INIT; 8188 } 8189 8190 sp<ThreadBase> thread = mThread.promote(); 8191 if (thread == 0) { 8192 return DEAD_OBJECT; 8193 } 8194 8195 // TODO: handle configuration of effects replacing track process 8196 audio_channel_mask_t channelMask = thread->channelMask(); 8197 8198 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8199 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8200 } else { 8201 mConfig.inputCfg.channels = channelMask; 8202 } 8203 mConfig.outputCfg.channels = channelMask; 8204 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8205 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8206 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8207 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8208 mConfig.inputCfg.bufferProvider.cookie = NULL; 8209 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8210 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8211 mConfig.outputCfg.bufferProvider.cookie = NULL; 8212 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8213 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8214 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8215 // Insert effect: 8216 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8217 // always overwrites output buffer: input buffer == output buffer 8218 // - in other sessions: 8219 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8220 // other effect: overwrites output buffer: input buffer == output buffer 8221 // Auxiliary effect: 8222 // accumulates in output buffer: input buffer != output buffer 8223 // Therefore: accumulate <=> input buffer != output buffer 8224 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8225 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8226 } else { 8227 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8228 } 8229 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8230 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8231 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8232 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8233 8234 ALOGV("configure() %p thread %p buffer %p framecount %d", 8235 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8236 8237 status_t cmdStatus; 8238 uint32_t size = sizeof(int); 8239 status_t status = (*mEffectInterface)->command(mEffectInterface, 8240 EFFECT_CMD_SET_CONFIG, 8241 sizeof(effect_config_t), 8242 &mConfig, 8243 &size, 8244 &cmdStatus); 8245 if (status == 0) { 8246 status = cmdStatus; 8247 } 8248 8249 if (status == 0 && 8250 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8251 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8252 effect_param_t *p = (effect_param_t *)buf32; 8253 8254 p->psize = sizeof(uint32_t); 8255 p->vsize = sizeof(uint32_t); 8256 size = sizeof(int); 8257 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8258 8259 uint32_t latency = 0; 8260 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8261 if (pbt != NULL) { 8262 latency = pbt->latency_l(); 8263 } 8264 8265 *((int32_t *)p->data + 1)= latency; 8266 (*mEffectInterface)->command(mEffectInterface, 8267 EFFECT_CMD_SET_PARAM, 8268 sizeof(effect_param_t) + 8, 8269 &buf32, 8270 &size, 8271 &cmdStatus); 8272 } 8273 8274 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8275 (1000 * mConfig.outputCfg.buffer.frameCount); 8276 8277 return status; 8278} 8279 8280status_t AudioFlinger::EffectModule::init() 8281{ 8282 Mutex::Autolock _l(mLock); 8283 if (mEffectInterface == NULL) { 8284 return NO_INIT; 8285 } 8286 status_t cmdStatus; 8287 uint32_t size = sizeof(status_t); 8288 status_t status = (*mEffectInterface)->command(mEffectInterface, 8289 EFFECT_CMD_INIT, 8290 0, 8291 NULL, 8292 &size, 8293 &cmdStatus); 8294 if (status == 0) { 8295 status = cmdStatus; 8296 } 8297 return status; 8298} 8299 8300status_t AudioFlinger::EffectModule::start() 8301{ 8302 Mutex::Autolock _l(mLock); 8303 return start_l(); 8304} 8305 8306status_t AudioFlinger::EffectModule::start_l() 8307{ 8308 if (mEffectInterface == NULL) { 8309 return NO_INIT; 8310 } 8311 status_t cmdStatus; 8312 uint32_t size = sizeof(status_t); 8313 status_t status = (*mEffectInterface)->command(mEffectInterface, 8314 EFFECT_CMD_ENABLE, 8315 0, 8316 NULL, 8317 &size, 8318 &cmdStatus); 8319 if (status == 0) { 8320 status = cmdStatus; 8321 } 8322 if (status == 0 && 8323 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8324 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8325 sp<ThreadBase> thread = mThread.promote(); 8326 if (thread != 0) { 8327 audio_stream_t *stream = thread->stream(); 8328 if (stream != NULL) { 8329 stream->add_audio_effect(stream, mEffectInterface); 8330 } 8331 } 8332 } 8333 return status; 8334} 8335 8336status_t AudioFlinger::EffectModule::stop() 8337{ 8338 Mutex::Autolock _l(mLock); 8339 return stop_l(); 8340} 8341 8342status_t AudioFlinger::EffectModule::stop_l() 8343{ 8344 if (mEffectInterface == NULL) { 8345 return NO_INIT; 8346 } 8347 status_t cmdStatus; 8348 uint32_t size = sizeof(status_t); 8349 status_t status = (*mEffectInterface)->command(mEffectInterface, 8350 EFFECT_CMD_DISABLE, 8351 0, 8352 NULL, 8353 &size, 8354 &cmdStatus); 8355 if (status == 0) { 8356 status = cmdStatus; 8357 } 8358 if (status == 0 && 8359 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8360 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8361 sp<ThreadBase> thread = mThread.promote(); 8362 if (thread != 0) { 8363 audio_stream_t *stream = thread->stream(); 8364 if (stream != NULL) { 8365 stream->remove_audio_effect(stream, mEffectInterface); 8366 } 8367 } 8368 } 8369 return status; 8370} 8371 8372status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8373 uint32_t cmdSize, 8374 void *pCmdData, 8375 uint32_t *replySize, 8376 void *pReplyData) 8377{ 8378 Mutex::Autolock _l(mLock); 8379// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8380 8381 if (mState == DESTROYED || mEffectInterface == NULL) { 8382 return NO_INIT; 8383 } 8384 status_t status = (*mEffectInterface)->command(mEffectInterface, 8385 cmdCode, 8386 cmdSize, 8387 pCmdData, 8388 replySize, 8389 pReplyData); 8390 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8391 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8392 for (size_t i = 1; i < mHandles.size(); i++) { 8393 EffectHandle *h = mHandles[i]; 8394 if (h != NULL && !h->destroyed_l()) { 8395 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8396 } 8397 } 8398 } 8399 return status; 8400} 8401 8402status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8403{ 8404 Mutex::Autolock _l(mLock); 8405 return setEnabled_l(enabled); 8406} 8407 8408// must be called with EffectModule::mLock held 8409status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8410{ 8411 8412 ALOGV("setEnabled %p enabled %d", this, enabled); 8413 8414 if (enabled != isEnabled()) { 8415 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8416 if (enabled && status != NO_ERROR) { 8417 return status; 8418 } 8419 8420 switch (mState) { 8421 // going from disabled to enabled 8422 case IDLE: 8423 mState = STARTING; 8424 break; 8425 case STOPPED: 8426 mState = RESTART; 8427 break; 8428 case STOPPING: 8429 mState = ACTIVE; 8430 break; 8431 8432 // going from enabled to disabled 8433 case RESTART: 8434 mState = STOPPED; 8435 break; 8436 case STARTING: 8437 mState = IDLE; 8438 break; 8439 case ACTIVE: 8440 mState = STOPPING; 8441 break; 8442 case DESTROYED: 8443 return NO_ERROR; // simply ignore as we are being destroyed 8444 } 8445 for (size_t i = 1; i < mHandles.size(); i++) { 8446 EffectHandle *h = mHandles[i]; 8447 if (h != NULL && !h->destroyed_l()) { 8448 h->setEnabled(enabled); 8449 } 8450 } 8451 } 8452 return NO_ERROR; 8453} 8454 8455bool AudioFlinger::EffectModule::isEnabled() const 8456{ 8457 switch (mState) { 8458 case RESTART: 8459 case STARTING: 8460 case ACTIVE: 8461 return true; 8462 case IDLE: 8463 case STOPPING: 8464 case STOPPED: 8465 case DESTROYED: 8466 default: 8467 return false; 8468 } 8469} 8470 8471bool AudioFlinger::EffectModule::isProcessEnabled() const 8472{ 8473 switch (mState) { 8474 case RESTART: 8475 case ACTIVE: 8476 case STOPPING: 8477 case STOPPED: 8478 return true; 8479 case IDLE: 8480 case STARTING: 8481 case DESTROYED: 8482 default: 8483 return false; 8484 } 8485} 8486 8487status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8488{ 8489 Mutex::Autolock _l(mLock); 8490 status_t status = NO_ERROR; 8491 8492 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8493 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8494 if (isProcessEnabled() && 8495 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8496 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8497 status_t cmdStatus; 8498 uint32_t volume[2]; 8499 uint32_t *pVolume = NULL; 8500 uint32_t size = sizeof(volume); 8501 volume[0] = *left; 8502 volume[1] = *right; 8503 if (controller) { 8504 pVolume = volume; 8505 } 8506 status = (*mEffectInterface)->command(mEffectInterface, 8507 EFFECT_CMD_SET_VOLUME, 8508 size, 8509 volume, 8510 &size, 8511 pVolume); 8512 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8513 *left = volume[0]; 8514 *right = volume[1]; 8515 } 8516 } 8517 return status; 8518} 8519 8520status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8521{ 8522 Mutex::Autolock _l(mLock); 8523 status_t status = NO_ERROR; 8524 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8525 // audio pre processing modules on RecordThread can receive both output and 8526 // input device indication in the same call 8527 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8528 if (dev) { 8529 status_t cmdStatus; 8530 uint32_t size = sizeof(status_t); 8531 8532 status = (*mEffectInterface)->command(mEffectInterface, 8533 EFFECT_CMD_SET_DEVICE, 8534 sizeof(uint32_t), 8535 &dev, 8536 &size, 8537 &cmdStatus); 8538 if (status == NO_ERROR) { 8539 status = cmdStatus; 8540 } 8541 } 8542 dev = device & AUDIO_DEVICE_IN_ALL; 8543 if (dev) { 8544 status_t cmdStatus; 8545 uint32_t size = sizeof(status_t); 8546 8547 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8548 EFFECT_CMD_SET_INPUT_DEVICE, 8549 sizeof(uint32_t), 8550 &dev, 8551 &size, 8552 &cmdStatus); 8553 if (status2 == NO_ERROR) { 8554 status2 = cmdStatus; 8555 } 8556 if (status == NO_ERROR) { 8557 status = status2; 8558 } 8559 } 8560 } 8561 return status; 8562} 8563 8564status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8565{ 8566 Mutex::Autolock _l(mLock); 8567 status_t status = NO_ERROR; 8568 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8569 status_t cmdStatus; 8570 uint32_t size = sizeof(status_t); 8571 status = (*mEffectInterface)->command(mEffectInterface, 8572 EFFECT_CMD_SET_AUDIO_MODE, 8573 sizeof(audio_mode_t), 8574 &mode, 8575 &size, 8576 &cmdStatus); 8577 if (status == NO_ERROR) { 8578 status = cmdStatus; 8579 } 8580 } 8581 return status; 8582} 8583 8584void AudioFlinger::EffectModule::setSuspended(bool suspended) 8585{ 8586 Mutex::Autolock _l(mLock); 8587 mSuspended = suspended; 8588} 8589 8590bool AudioFlinger::EffectModule::suspended() const 8591{ 8592 Mutex::Autolock _l(mLock); 8593 return mSuspended; 8594} 8595 8596bool AudioFlinger::EffectModule::purgeHandles() 8597{ 8598 bool enabled = false; 8599 Mutex::Autolock _l(mLock); 8600 for (size_t i = 0; i < mHandles.size(); i++) { 8601 EffectHandle *handle = mHandles[i]; 8602 if (handle != NULL && !handle->destroyed_l()) { 8603 handle->effect().clear(); 8604 if (handle->hasControl()) { 8605 enabled = handle->enabled(); 8606 } 8607 } 8608 } 8609 return enabled; 8610} 8611 8612status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8613{ 8614 const size_t SIZE = 256; 8615 char buffer[SIZE]; 8616 String8 result; 8617 8618 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8619 result.append(buffer); 8620 8621 bool locked = tryLock(mLock); 8622 // failed to lock - AudioFlinger is probably deadlocked 8623 if (!locked) { 8624 result.append("\t\tCould not lock Fx mutex:\n"); 8625 } 8626 8627 result.append("\t\tSession Status State Engine:\n"); 8628 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8629 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8630 result.append(buffer); 8631 8632 result.append("\t\tDescriptor:\n"); 8633 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8634 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8635 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8636 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8637 result.append(buffer); 8638 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8639 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8640 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8641 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8642 result.append(buffer); 8643 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8644 mDescriptor.apiVersion, 8645 mDescriptor.flags); 8646 result.append(buffer); 8647 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8648 mDescriptor.name); 8649 result.append(buffer); 8650 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8651 mDescriptor.implementor); 8652 result.append(buffer); 8653 8654 result.append("\t\t- Input configuration:\n"); 8655 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8656 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8657 (uint32_t)mConfig.inputCfg.buffer.raw, 8658 mConfig.inputCfg.buffer.frameCount, 8659 mConfig.inputCfg.samplingRate, 8660 mConfig.inputCfg.channels, 8661 mConfig.inputCfg.format); 8662 result.append(buffer); 8663 8664 result.append("\t\t- Output configuration:\n"); 8665 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8666 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8667 (uint32_t)mConfig.outputCfg.buffer.raw, 8668 mConfig.outputCfg.buffer.frameCount, 8669 mConfig.outputCfg.samplingRate, 8670 mConfig.outputCfg.channels, 8671 mConfig.outputCfg.format); 8672 result.append(buffer); 8673 8674 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8675 result.append(buffer); 8676 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8677 for (size_t i = 0; i < mHandles.size(); ++i) { 8678 EffectHandle *handle = mHandles[i]; 8679 if (handle != NULL && !handle->destroyed_l()) { 8680 handle->dump(buffer, SIZE); 8681 result.append(buffer); 8682 } 8683 } 8684 8685 result.append("\n"); 8686 8687 write(fd, result.string(), result.length()); 8688 8689 if (locked) { 8690 mLock.unlock(); 8691 } 8692 8693 return NO_ERROR; 8694} 8695 8696// ---------------------------------------------------------------------------- 8697// EffectHandle implementation 8698// ---------------------------------------------------------------------------- 8699 8700#undef LOG_TAG 8701#define LOG_TAG "AudioFlinger::EffectHandle" 8702 8703AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8704 const sp<AudioFlinger::Client>& client, 8705 const sp<IEffectClient>& effectClient, 8706 int32_t priority) 8707 : BnEffect(), 8708 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8709 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8710{ 8711 ALOGV("constructor %p", this); 8712 8713 if (client == 0) { 8714 return; 8715 } 8716 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8717 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8718 if (mCblkMemory != 0) { 8719 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8720 8721 if (mCblk != NULL) { 8722 new(mCblk) effect_param_cblk_t(); 8723 mBuffer = (uint8_t *)mCblk + bufOffset; 8724 } 8725 } else { 8726 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8727 return; 8728 } 8729} 8730 8731AudioFlinger::EffectHandle::~EffectHandle() 8732{ 8733 ALOGV("Destructor %p", this); 8734 8735 if (mEffect == 0) { 8736 mDestroyed = true; 8737 return; 8738 } 8739 mEffect->lock(); 8740 mDestroyed = true; 8741 mEffect->unlock(); 8742 disconnect(false); 8743} 8744 8745status_t AudioFlinger::EffectHandle::enable() 8746{ 8747 ALOGV("enable %p", this); 8748 if (!mHasControl) return INVALID_OPERATION; 8749 if (mEffect == 0) return DEAD_OBJECT; 8750 8751 if (mEnabled) { 8752 return NO_ERROR; 8753 } 8754 8755 mEnabled = true; 8756 8757 sp<ThreadBase> thread = mEffect->thread().promote(); 8758 if (thread != 0) { 8759 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8760 } 8761 8762 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8763 if (mEffect->suspended()) { 8764 return NO_ERROR; 8765 } 8766 8767 status_t status = mEffect->setEnabled(true); 8768 if (status != NO_ERROR) { 8769 if (thread != 0) { 8770 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8771 } 8772 mEnabled = false; 8773 } 8774 return status; 8775} 8776 8777status_t AudioFlinger::EffectHandle::disable() 8778{ 8779 ALOGV("disable %p", this); 8780 if (!mHasControl) return INVALID_OPERATION; 8781 if (mEffect == 0) return DEAD_OBJECT; 8782 8783 if (!mEnabled) { 8784 return NO_ERROR; 8785 } 8786 mEnabled = false; 8787 8788 if (mEffect->suspended()) { 8789 return NO_ERROR; 8790 } 8791 8792 status_t status = mEffect->setEnabled(false); 8793 8794 sp<ThreadBase> thread = mEffect->thread().promote(); 8795 if (thread != 0) { 8796 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8797 } 8798 8799 return status; 8800} 8801 8802void AudioFlinger::EffectHandle::disconnect() 8803{ 8804 disconnect(true); 8805} 8806 8807void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8808{ 8809 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8810 if (mEffect == 0) { 8811 return; 8812 } 8813 // restore suspended effects if the disconnected handle was enabled and the last one. 8814 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8815 sp<ThreadBase> thread = mEffect->thread().promote(); 8816 if (thread != 0) { 8817 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8818 } 8819 } 8820 8821 // release sp on module => module destructor can be called now 8822 mEffect.clear(); 8823 if (mClient != 0) { 8824 if (mCblk != NULL) { 8825 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8826 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8827 } 8828 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8829 // Client destructor must run with AudioFlinger mutex locked 8830 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8831 mClient.clear(); 8832 } 8833} 8834 8835status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8836 uint32_t cmdSize, 8837 void *pCmdData, 8838 uint32_t *replySize, 8839 void *pReplyData) 8840{ 8841// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8842// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8843 8844 // only get parameter command is permitted for applications not controlling the effect 8845 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8846 return INVALID_OPERATION; 8847 } 8848 if (mEffect == 0) return DEAD_OBJECT; 8849 if (mClient == 0) return INVALID_OPERATION; 8850 8851 // handle commands that are not forwarded transparently to effect engine 8852 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8853 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8854 // no risk to block the whole media server process or mixer threads is we are stuck here 8855 Mutex::Autolock _l(mCblk->lock); 8856 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8857 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8858 mCblk->serverIndex = 0; 8859 mCblk->clientIndex = 0; 8860 return BAD_VALUE; 8861 } 8862 status_t status = NO_ERROR; 8863 while (mCblk->serverIndex < mCblk->clientIndex) { 8864 int reply; 8865 uint32_t rsize = sizeof(int); 8866 int *p = (int *)(mBuffer + mCblk->serverIndex); 8867 int size = *p++; 8868 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8869 ALOGW("command(): invalid parameter block size"); 8870 break; 8871 } 8872 effect_param_t *param = (effect_param_t *)p; 8873 if (param->psize == 0 || param->vsize == 0) { 8874 ALOGW("command(): null parameter or value size"); 8875 mCblk->serverIndex += size; 8876 continue; 8877 } 8878 uint32_t psize = sizeof(effect_param_t) + 8879 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8880 param->vsize; 8881 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8882 psize, 8883 p, 8884 &rsize, 8885 &reply); 8886 // stop at first error encountered 8887 if (ret != NO_ERROR) { 8888 status = ret; 8889 *(int *)pReplyData = reply; 8890 break; 8891 } else if (reply != NO_ERROR) { 8892 *(int *)pReplyData = reply; 8893 break; 8894 } 8895 mCblk->serverIndex += size; 8896 } 8897 mCblk->serverIndex = 0; 8898 mCblk->clientIndex = 0; 8899 return status; 8900 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8901 *(int *)pReplyData = NO_ERROR; 8902 return enable(); 8903 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8904 *(int *)pReplyData = NO_ERROR; 8905 return disable(); 8906 } 8907 8908 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8909} 8910 8911void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8912{ 8913 ALOGV("setControl %p control %d", this, hasControl); 8914 8915 mHasControl = hasControl; 8916 mEnabled = enabled; 8917 8918 if (signal && mEffectClient != 0) { 8919 mEffectClient->controlStatusChanged(hasControl); 8920 } 8921} 8922 8923void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8924 uint32_t cmdSize, 8925 void *pCmdData, 8926 uint32_t replySize, 8927 void *pReplyData) 8928{ 8929 if (mEffectClient != 0) { 8930 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8931 } 8932} 8933 8934 8935 8936void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8937{ 8938 if (mEffectClient != 0) { 8939 mEffectClient->enableStatusChanged(enabled); 8940 } 8941} 8942 8943status_t AudioFlinger::EffectHandle::onTransact( 8944 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8945{ 8946 return BnEffect::onTransact(code, data, reply, flags); 8947} 8948 8949 8950void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8951{ 8952 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8953 8954 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8955 (mClient == 0) ? getpid_cached : mClient->pid(), 8956 mPriority, 8957 mHasControl, 8958 !locked, 8959 mCblk ? mCblk->clientIndex : 0, 8960 mCblk ? mCblk->serverIndex : 0 8961 ); 8962 8963 if (locked) { 8964 mCblk->lock.unlock(); 8965 } 8966} 8967 8968#undef LOG_TAG 8969#define LOG_TAG "AudioFlinger::EffectChain" 8970 8971AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8972 int sessionId) 8973 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8974 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8975 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8976{ 8977 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8978 if (thread == NULL) { 8979 return; 8980 } 8981 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8982 thread->frameCount(); 8983} 8984 8985AudioFlinger::EffectChain::~EffectChain() 8986{ 8987 if (mOwnInBuffer) { 8988 delete mInBuffer; 8989 } 8990 8991} 8992 8993// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8994sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8995{ 8996 size_t size = mEffects.size(); 8997 8998 for (size_t i = 0; i < size; i++) { 8999 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9000 return mEffects[i]; 9001 } 9002 } 9003 return 0; 9004} 9005 9006// getEffectFromId_l() must be called with ThreadBase::mLock held 9007sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9008{ 9009 size_t size = mEffects.size(); 9010 9011 for (size_t i = 0; i < size; i++) { 9012 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9013 if (id == 0 || mEffects[i]->id() == id) { 9014 return mEffects[i]; 9015 } 9016 } 9017 return 0; 9018} 9019 9020// getEffectFromType_l() must be called with ThreadBase::mLock held 9021sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9022 const effect_uuid_t *type) 9023{ 9024 size_t size = mEffects.size(); 9025 9026 for (size_t i = 0; i < size; i++) { 9027 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9028 return mEffects[i]; 9029 } 9030 } 9031 return 0; 9032} 9033 9034void AudioFlinger::EffectChain::clearInputBuffer() 9035{ 9036 Mutex::Autolock _l(mLock); 9037 sp<ThreadBase> thread = mThread.promote(); 9038 if (thread == 0) { 9039 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9040 return; 9041 } 9042 clearInputBuffer_l(thread); 9043} 9044 9045// Must be called with EffectChain::mLock locked 9046void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9047{ 9048 size_t numSamples = thread->frameCount() * thread->channelCount(); 9049 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9050 9051} 9052 9053// Must be called with EffectChain::mLock locked 9054void AudioFlinger::EffectChain::process_l() 9055{ 9056 sp<ThreadBase> thread = mThread.promote(); 9057 if (thread == 0) { 9058 ALOGW("process_l(): cannot promote mixer thread"); 9059 return; 9060 } 9061 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9062 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9063 // always process effects unless no more tracks are on the session and the effect tail 9064 // has been rendered 9065 bool doProcess = true; 9066 if (!isGlobalSession) { 9067 bool tracksOnSession = (trackCnt() != 0); 9068 9069 if (!tracksOnSession && mTailBufferCount == 0) { 9070 doProcess = false; 9071 } 9072 9073 if (activeTrackCnt() == 0) { 9074 // if no track is active and the effect tail has not been rendered, 9075 // the input buffer must be cleared here as the mixer process will not do it 9076 if (tracksOnSession || mTailBufferCount > 0) { 9077 clearInputBuffer_l(thread); 9078 if (mTailBufferCount > 0) { 9079 mTailBufferCount--; 9080 } 9081 } 9082 } 9083 } 9084 9085 size_t size = mEffects.size(); 9086 if (doProcess) { 9087 for (size_t i = 0; i < size; i++) { 9088 mEffects[i]->process(); 9089 } 9090 } 9091 for (size_t i = 0; i < size; i++) { 9092 mEffects[i]->updateState(); 9093 } 9094} 9095 9096// addEffect_l() must be called with PlaybackThread::mLock held 9097status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9098{ 9099 effect_descriptor_t desc = effect->desc(); 9100 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9101 9102 Mutex::Autolock _l(mLock); 9103 effect->setChain(this); 9104 sp<ThreadBase> thread = mThread.promote(); 9105 if (thread == 0) { 9106 return NO_INIT; 9107 } 9108 effect->setThread(thread); 9109 9110 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9111 // Auxiliary effects are inserted at the beginning of mEffects vector as 9112 // they are processed first and accumulated in chain input buffer 9113 mEffects.insertAt(effect, 0); 9114 9115 // the input buffer for auxiliary effect contains mono samples in 9116 // 32 bit format. This is to avoid saturation in AudoMixer 9117 // accumulation stage. Saturation is done in EffectModule::process() before 9118 // calling the process in effect engine 9119 size_t numSamples = thread->frameCount(); 9120 int32_t *buffer = new int32_t[numSamples]; 9121 memset(buffer, 0, numSamples * sizeof(int32_t)); 9122 effect->setInBuffer((int16_t *)buffer); 9123 // auxiliary effects output samples to chain input buffer for further processing 9124 // by insert effects 9125 effect->setOutBuffer(mInBuffer); 9126 } else { 9127 // Insert effects are inserted at the end of mEffects vector as they are processed 9128 // after track and auxiliary effects. 9129 // Insert effect order as a function of indicated preference: 9130 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9131 // another effect is present 9132 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9133 // last effect claiming first position 9134 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9135 // first effect claiming last position 9136 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9137 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9138 // already present 9139 9140 size_t size = mEffects.size(); 9141 size_t idx_insert = size; 9142 ssize_t idx_insert_first = -1; 9143 ssize_t idx_insert_last = -1; 9144 9145 for (size_t i = 0; i < size; i++) { 9146 effect_descriptor_t d = mEffects[i]->desc(); 9147 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9148 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9149 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9150 // check invalid effect chaining combinations 9151 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9152 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9153 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9154 return INVALID_OPERATION; 9155 } 9156 // remember position of first insert effect and by default 9157 // select this as insert position for new effect 9158 if (idx_insert == size) { 9159 idx_insert = i; 9160 } 9161 // remember position of last insert effect claiming 9162 // first position 9163 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9164 idx_insert_first = i; 9165 } 9166 // remember position of first insert effect claiming 9167 // last position 9168 if (iPref == EFFECT_FLAG_INSERT_LAST && 9169 idx_insert_last == -1) { 9170 idx_insert_last = i; 9171 } 9172 } 9173 } 9174 9175 // modify idx_insert from first position if needed 9176 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9177 if (idx_insert_last != -1) { 9178 idx_insert = idx_insert_last; 9179 } else { 9180 idx_insert = size; 9181 } 9182 } else { 9183 if (idx_insert_first != -1) { 9184 idx_insert = idx_insert_first + 1; 9185 } 9186 } 9187 9188 // always read samples from chain input buffer 9189 effect->setInBuffer(mInBuffer); 9190 9191 // if last effect in the chain, output samples to chain 9192 // output buffer, otherwise to chain input buffer 9193 if (idx_insert == size) { 9194 if (idx_insert != 0) { 9195 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9196 mEffects[idx_insert-1]->configure(); 9197 } 9198 effect->setOutBuffer(mOutBuffer); 9199 } else { 9200 effect->setOutBuffer(mInBuffer); 9201 } 9202 mEffects.insertAt(effect, idx_insert); 9203 9204 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9205 } 9206 effect->configure(); 9207 return NO_ERROR; 9208} 9209 9210// removeEffect_l() must be called with PlaybackThread::mLock held 9211size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9212{ 9213 Mutex::Autolock _l(mLock); 9214 size_t size = mEffects.size(); 9215 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9216 9217 for (size_t i = 0; i < size; i++) { 9218 if (effect == mEffects[i]) { 9219 // calling stop here will remove pre-processing effect from the audio HAL. 9220 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9221 // the middle of a read from audio HAL 9222 if (mEffects[i]->state() == EffectModule::ACTIVE || 9223 mEffects[i]->state() == EffectModule::STOPPING) { 9224 mEffects[i]->stop(); 9225 } 9226 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9227 delete[] effect->inBuffer(); 9228 } else { 9229 if (i == size - 1 && i != 0) { 9230 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9231 mEffects[i - 1]->configure(); 9232 } 9233 } 9234 mEffects.removeAt(i); 9235 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9236 break; 9237 } 9238 } 9239 9240 return mEffects.size(); 9241} 9242 9243// setDevice_l() must be called with PlaybackThread::mLock held 9244void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9245{ 9246 size_t size = mEffects.size(); 9247 for (size_t i = 0; i < size; i++) { 9248 mEffects[i]->setDevice(device); 9249 } 9250} 9251 9252// setMode_l() must be called with PlaybackThread::mLock held 9253void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9254{ 9255 size_t size = mEffects.size(); 9256 for (size_t i = 0; i < size; i++) { 9257 mEffects[i]->setMode(mode); 9258 } 9259} 9260 9261// setVolume_l() must be called with PlaybackThread::mLock held 9262bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9263{ 9264 uint32_t newLeft = *left; 9265 uint32_t newRight = *right; 9266 bool hasControl = false; 9267 int ctrlIdx = -1; 9268 size_t size = mEffects.size(); 9269 9270 // first update volume controller 9271 for (size_t i = size; i > 0; i--) { 9272 if (mEffects[i - 1]->isProcessEnabled() && 9273 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9274 ctrlIdx = i - 1; 9275 hasControl = true; 9276 break; 9277 } 9278 } 9279 9280 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9281 if (hasControl) { 9282 *left = mNewLeftVolume; 9283 *right = mNewRightVolume; 9284 } 9285 return hasControl; 9286 } 9287 9288 mVolumeCtrlIdx = ctrlIdx; 9289 mLeftVolume = newLeft; 9290 mRightVolume = newRight; 9291 9292 // second get volume update from volume controller 9293 if (ctrlIdx >= 0) { 9294 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9295 mNewLeftVolume = newLeft; 9296 mNewRightVolume = newRight; 9297 } 9298 // then indicate volume to all other effects in chain. 9299 // Pass altered volume to effects before volume controller 9300 // and requested volume to effects after controller 9301 uint32_t lVol = newLeft; 9302 uint32_t rVol = newRight; 9303 9304 for (size_t i = 0; i < size; i++) { 9305 if ((int)i == ctrlIdx) continue; 9306 // this also works for ctrlIdx == -1 when there is no volume controller 9307 if ((int)i > ctrlIdx) { 9308 lVol = *left; 9309 rVol = *right; 9310 } 9311 mEffects[i]->setVolume(&lVol, &rVol, false); 9312 } 9313 *left = newLeft; 9314 *right = newRight; 9315 9316 return hasControl; 9317} 9318 9319status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9320{ 9321 const size_t SIZE = 256; 9322 char buffer[SIZE]; 9323 String8 result; 9324 9325 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9326 result.append(buffer); 9327 9328 bool locked = tryLock(mLock); 9329 // failed to lock - AudioFlinger is probably deadlocked 9330 if (!locked) { 9331 result.append("\tCould not lock mutex:\n"); 9332 } 9333 9334 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9335 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9336 mEffects.size(), 9337 (uint32_t)mInBuffer, 9338 (uint32_t)mOutBuffer, 9339 mActiveTrackCnt); 9340 result.append(buffer); 9341 write(fd, result.string(), result.size()); 9342 9343 for (size_t i = 0; i < mEffects.size(); ++i) { 9344 sp<EffectModule> effect = mEffects[i]; 9345 if (effect != 0) { 9346 effect->dump(fd, args); 9347 } 9348 } 9349 9350 if (locked) { 9351 mLock.unlock(); 9352 } 9353 9354 return NO_ERROR; 9355} 9356 9357// must be called with ThreadBase::mLock held 9358void AudioFlinger::EffectChain::setEffectSuspended_l( 9359 const effect_uuid_t *type, bool suspend) 9360{ 9361 sp<SuspendedEffectDesc> desc; 9362 // use effect type UUID timelow as key as there is no real risk of identical 9363 // timeLow fields among effect type UUIDs. 9364 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9365 if (suspend) { 9366 if (index >= 0) { 9367 desc = mSuspendedEffects.valueAt(index); 9368 } else { 9369 desc = new SuspendedEffectDesc(); 9370 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9371 mSuspendedEffects.add(type->timeLow, desc); 9372 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9373 } 9374 if (desc->mRefCount++ == 0) { 9375 sp<EffectModule> effect = getEffectIfEnabled(type); 9376 if (effect != 0) { 9377 desc->mEffect = effect; 9378 effect->setSuspended(true); 9379 effect->setEnabled(false); 9380 } 9381 } 9382 } else { 9383 if (index < 0) { 9384 return; 9385 } 9386 desc = mSuspendedEffects.valueAt(index); 9387 if (desc->mRefCount <= 0) { 9388 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9389 desc->mRefCount = 1; 9390 } 9391 if (--desc->mRefCount == 0) { 9392 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9393 if (desc->mEffect != 0) { 9394 sp<EffectModule> effect = desc->mEffect.promote(); 9395 if (effect != 0) { 9396 effect->setSuspended(false); 9397 effect->lock(); 9398 EffectHandle *handle = effect->controlHandle_l(); 9399 if (handle != NULL && !handle->destroyed_l()) { 9400 effect->setEnabled_l(handle->enabled()); 9401 } 9402 effect->unlock(); 9403 } 9404 desc->mEffect.clear(); 9405 } 9406 mSuspendedEffects.removeItemsAt(index); 9407 } 9408 } 9409} 9410 9411// must be called with ThreadBase::mLock held 9412void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9413{ 9414 sp<SuspendedEffectDesc> desc; 9415 9416 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9417 if (suspend) { 9418 if (index >= 0) { 9419 desc = mSuspendedEffects.valueAt(index); 9420 } else { 9421 desc = new SuspendedEffectDesc(); 9422 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9423 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9424 } 9425 if (desc->mRefCount++ == 0) { 9426 Vector< sp<EffectModule> > effects; 9427 getSuspendEligibleEffects(effects); 9428 for (size_t i = 0; i < effects.size(); i++) { 9429 setEffectSuspended_l(&effects[i]->desc().type, true); 9430 } 9431 } 9432 } else { 9433 if (index < 0) { 9434 return; 9435 } 9436 desc = mSuspendedEffects.valueAt(index); 9437 if (desc->mRefCount <= 0) { 9438 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9439 desc->mRefCount = 1; 9440 } 9441 if (--desc->mRefCount == 0) { 9442 Vector<const effect_uuid_t *> types; 9443 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9444 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9445 continue; 9446 } 9447 types.add(&mSuspendedEffects.valueAt(i)->mType); 9448 } 9449 for (size_t i = 0; i < types.size(); i++) { 9450 setEffectSuspended_l(types[i], false); 9451 } 9452 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9453 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9454 } 9455 } 9456} 9457 9458 9459// The volume effect is used for automated tests only 9460#ifndef OPENSL_ES_H_ 9461static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9462 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9463const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9464#endif //OPENSL_ES_H_ 9465 9466bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9467{ 9468 // auxiliary effects and visualizer are never suspended on output mix 9469 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9470 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9471 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9472 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9473 return false; 9474 } 9475 return true; 9476} 9477 9478void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9479{ 9480 effects.clear(); 9481 for (size_t i = 0; i < mEffects.size(); i++) { 9482 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9483 effects.add(mEffects[i]); 9484 } 9485 } 9486} 9487 9488sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9489 const effect_uuid_t *type) 9490{ 9491 sp<EffectModule> effect = getEffectFromType_l(type); 9492 return effect != 0 && effect->isEnabled() ? effect : 0; 9493} 9494 9495void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9496 bool enabled) 9497{ 9498 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9499 if (enabled) { 9500 if (index < 0) { 9501 // if the effect is not suspend check if all effects are suspended 9502 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9503 if (index < 0) { 9504 return; 9505 } 9506 if (!isEffectEligibleForSuspend(effect->desc())) { 9507 return; 9508 } 9509 setEffectSuspended_l(&effect->desc().type, enabled); 9510 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9511 if (index < 0) { 9512 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9513 return; 9514 } 9515 } 9516 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9517 effect->desc().type.timeLow); 9518 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9519 // if effect is requested to suspended but was not yet enabled, supend it now. 9520 if (desc->mEffect == 0) { 9521 desc->mEffect = effect; 9522 effect->setEnabled(false); 9523 effect->setSuspended(true); 9524 } 9525 } else { 9526 if (index < 0) { 9527 return; 9528 } 9529 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9530 effect->desc().type.timeLow); 9531 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9532 desc->mEffect.clear(); 9533 effect->setSuspended(false); 9534 } 9535} 9536 9537#undef LOG_TAG 9538#define LOG_TAG "AudioFlinger" 9539 9540// ---------------------------------------------------------------------------- 9541 9542status_t AudioFlinger::onTransact( 9543 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9544{ 9545 return BnAudioFlinger::onTransact(code, data, reply, flags); 9546} 9547 9548}; // namespace android 9549