AudioFlinger.cpp revision 60a839204713e0f8258d082af83262b1eb33a6c3
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 <media/nbaio/AudioStreamOutSink.h> 80#include <media/nbaio/MonoPipe.h> 81#include <media/nbaio/MonoPipeReader.h> 82#include <media/nbaio/Pipe.h> 83#include <media/nbaio/PipeReader.h> 84#include <media/nbaio/SourceAudioBufferProvider.h> 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 169// for the track. The client then sub-divides this into smaller buffers for its use. 170// Currently the client uses double-buffering by default, but doesn't tell us about that. 171// So for now we just assume that client is double-buffered. 172// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 173// N-buffering, so AudioFlinger could allocate the right amount of memory. 174// See the client's minBufCount and mNotificationFramesAct calculations for details. 175static const int kFastTrackMultiplier = 2; 176 177// ---------------------------------------------------------------------------- 178 179#ifdef ADD_BATTERY_DATA 180// To collect the amplifier usage 181static void addBatteryData(uint32_t params) { 182 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 183 if (service == NULL) { 184 // it already logged 185 return; 186 } 187 188 service->addBatteryData(params); 189} 190#endif 191 192static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 193{ 194 const hw_module_t *mod; 195 int rc; 196 197 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 198 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 199 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 200 if (rc) { 201 goto out; 202 } 203 rc = audio_hw_device_open(mod, dev); 204 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 205 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 206 if (rc) { 207 goto out; 208 } 209 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 210 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 211 rc = BAD_VALUE; 212 goto out; 213 } 214 return 0; 215 216out: 217 *dev = NULL; 218 return rc; 219} 220 221// ---------------------------------------------------------------------------- 222 223AudioFlinger::AudioFlinger() 224 : BnAudioFlinger(), 225 mPrimaryHardwareDev(NULL), 226 mHardwareStatus(AUDIO_HW_IDLE), 227 mMasterVolume(1.0f), 228 mMasterMute(false), 229 mNextUniqueId(1), 230 mMode(AUDIO_MODE_INVALID), 231 mBtNrecIsOff(false) 232{ 233} 234 235void AudioFlinger::onFirstRef() 236{ 237 int rc = 0; 238 239 Mutex::Autolock _l(mLock); 240 241 /* TODO: move all this work into an Init() function */ 242 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 243 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 244 uint32_t int_val; 245 if (1 == sscanf(val_str, "%u", &int_val)) { 246 mStandbyTimeInNsecs = milliseconds(int_val); 247 ALOGI("Using %u mSec as standby time.", int_val); 248 } else { 249 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 250 ALOGI("Using default %u mSec as standby time.", 251 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 252 } 253 } 254 255 mMode = AUDIO_MODE_NORMAL; 256} 257 258AudioFlinger::~AudioFlinger() 259{ 260 while (!mRecordThreads.isEmpty()) { 261 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 262 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 263 } 264 while (!mPlaybackThreads.isEmpty()) { 265 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 266 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 267 } 268 269 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 270 // no mHardwareLock needed, as there are no other references to this 271 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 272 delete mAudioHwDevs.valueAt(i); 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367static bool tryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = tryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = tryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 dumpClients(fd, args); 403 dumpInternals(fd, args); 404 405 // dump playback threads 406 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 407 mPlaybackThreads.valueAt(i)->dump(fd, args); 408 } 409 410 // dump record threads 411 for (size_t i = 0; i < mRecordThreads.size(); i++) { 412 mRecordThreads.valueAt(i)->dump(fd, args); 413 } 414 415 // dump all hardware devs 416 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 417 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 418 dev->dump(dev, fd); 419 } 420 421 // dump the serially shared record tee sink 422 if (mRecordTeeSource != 0) { 423 dumpTee(fd, mRecordTeeSource); 424 } 425 426 if (locked) mLock.unlock(); 427 } 428 return NO_ERROR; 429} 430 431sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 432{ 433 // If pid is already in the mClients wp<> map, then use that entry 434 // (for which promote() is always != 0), otherwise create a new entry and Client. 435 sp<Client> client = mClients.valueFor(pid).promote(); 436 if (client == 0) { 437 client = new Client(this, pid); 438 mClients.add(pid, client); 439 } 440 441 return client; 442} 443 444// IAudioFlinger interface 445 446 447sp<IAudioTrack> AudioFlinger::createTrack( 448 pid_t pid, 449 audio_stream_type_t streamType, 450 uint32_t sampleRate, 451 audio_format_t format, 452 audio_channel_mask_t channelMask, 453 int frameCount, 454 IAudioFlinger::track_flags_t *flags, 455 const sp<IMemory>& sharedBuffer, 456 audio_io_handle_t output, 457 pid_t tid, 458 int *sessionId, 459 status_t *status) 460{ 461 sp<PlaybackThread::Track> track; 462 sp<TrackHandle> trackHandle; 463 sp<Client> client; 464 status_t lStatus; 465 int lSessionId; 466 467 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 468 // but if someone uses binder directly they could bypass that and cause us to crash 469 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 470 ALOGE("createTrack() invalid stream type %d", streamType); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 476 // and we don't yet support 8.24 or 32-bit PCM 477 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 478 ALOGE("createTrack() invalid format %d", format); 479 lStatus = BAD_VALUE; 480 goto Exit; 481 } 482 483 { 484 Mutex::Autolock _l(mLock); 485 PlaybackThread *thread = checkPlaybackThread_l(output); 486 PlaybackThread *effectThread = NULL; 487 if (thread == NULL) { 488 ALOGE("unknown output thread"); 489 lStatus = BAD_VALUE; 490 goto Exit; 491 } 492 493 client = registerPid_l(pid); 494 495 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 496 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 497 // check if an effect chain with the same session ID is present on another 498 // output thread and move it here. 499 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 500 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 501 if (mPlaybackThreads.keyAt(i) != output) { 502 uint32_t sessions = t->hasAudioSession(*sessionId); 503 if (sessions & PlaybackThread::EFFECT_SESSION) { 504 effectThread = t.get(); 505 break; 506 } 507 } 508 } 509 lSessionId = *sessionId; 510 } else { 511 // if no audio session id is provided, create one here 512 lSessionId = nextUniqueId(); 513 if (sessionId != NULL) { 514 *sessionId = lSessionId; 515 } 516 } 517 ALOGV("createTrack() lSessionId: %d", lSessionId); 518 519 track = thread->createTrack_l(client, streamType, sampleRate, format, 520 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 521 522 // move effect chain to this output thread if an effect on same session was waiting 523 // for a track to be created 524 if (lStatus == NO_ERROR && effectThread != NULL) { 525 Mutex::Autolock _dl(thread->mLock); 526 Mutex::Autolock _sl(effectThread->mLock); 527 moveEffectChain_l(lSessionId, effectThread, thread, true); 528 } 529 530 // Look for sync events awaiting for a session to be used. 531 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 532 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 533 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 534 if (lStatus == NO_ERROR) { 535 (void) track->setSyncEvent(mPendingSyncEvents[i]); 536 } else { 537 mPendingSyncEvents[i]->cancel(); 538 } 539 mPendingSyncEvents.removeAt(i); 540 i--; 541 } 542 } 543 } 544 } 545 if (lStatus == NO_ERROR) { 546 trackHandle = new TrackHandle(track); 547 } else { 548 // remove local strong reference to Client before deleting the Track so that the Client 549 // destructor is called by the TrackBase destructor with mLock held 550 client.clear(); 551 track.clear(); 552 } 553 554Exit: 555 if (status != NULL) { 556 *status = lStatus; 557 } 558 return trackHandle; 559} 560 561uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 562{ 563 Mutex::Autolock _l(mLock); 564 PlaybackThread *thread = checkPlaybackThread_l(output); 565 if (thread == NULL) { 566 ALOGW("sampleRate() unknown thread %d", output); 567 return 0; 568 } 569 return thread->sampleRate(); 570} 571 572int AudioFlinger::channelCount(audio_io_handle_t output) const 573{ 574 Mutex::Autolock _l(mLock); 575 PlaybackThread *thread = checkPlaybackThread_l(output); 576 if (thread == NULL) { 577 ALOGW("channelCount() unknown thread %d", output); 578 return 0; 579 } 580 return thread->channelCount(); 581} 582 583audio_format_t AudioFlinger::format(audio_io_handle_t output) const 584{ 585 Mutex::Autolock _l(mLock); 586 PlaybackThread *thread = checkPlaybackThread_l(output); 587 if (thread == NULL) { 588 ALOGW("format() unknown thread %d", output); 589 return AUDIO_FORMAT_INVALID; 590 } 591 return thread->format(); 592} 593 594size_t AudioFlinger::frameCount(audio_io_handle_t output) const 595{ 596 Mutex::Autolock _l(mLock); 597 PlaybackThread *thread = checkPlaybackThread_l(output); 598 if (thread == NULL) { 599 ALOGW("frameCount() unknown thread %d", output); 600 return 0; 601 } 602 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 603 // should examine all callers and fix them to handle smaller counts 604 return thread->frameCount(); 605} 606 607uint32_t AudioFlinger::latency(audio_io_handle_t output) const 608{ 609 Mutex::Autolock _l(mLock); 610 PlaybackThread *thread = checkPlaybackThread_l(output); 611 if (thread == NULL) { 612 ALOGW("latency() unknown thread %d", output); 613 return 0; 614 } 615 return thread->latency(); 616} 617 618status_t AudioFlinger::setMasterVolume(float value) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 630 Mutex::Autolock _l(mLock); 631 mMasterVolume = value; 632 633 // Set master volume in the HALs which support it. 634 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 635 AutoMutex lock(mHardwareLock); 636 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 637 638 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 639 if (dev->canSetMasterVolume()) { 640 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 641 } 642 mHardwareStatus = AUDIO_HW_IDLE; 643 } 644 645 // Now set the master volume in each playback thread. Playback threads 646 // assigned to HALs which do not have master volume support will apply 647 // master volume during the mix operation. Threads with HALs which do 648 // support master volume will simply ignore the setting. 649 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 650 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 651 652 return NO_ERROR; 653} 654 655status_t AudioFlinger::setMode(audio_mode_t mode) 656{ 657 status_t ret = initCheck(); 658 if (ret != NO_ERROR) { 659 return ret; 660 } 661 662 // check calling permissions 663 if (!settingsAllowed()) { 664 return PERMISSION_DENIED; 665 } 666 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 667 ALOGW("Illegal value: setMode(%d)", mode); 668 return BAD_VALUE; 669 } 670 671 { // scope for the lock 672 AutoMutex lock(mHardwareLock); 673 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 674 mHardwareStatus = AUDIO_HW_SET_MODE; 675 ret = dev->set_mode(dev, mode); 676 mHardwareStatus = AUDIO_HW_IDLE; 677 } 678 679 if (NO_ERROR == ret) { 680 Mutex::Autolock _l(mLock); 681 mMode = mode; 682 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 683 mPlaybackThreads.valueAt(i)->setMode(mode); 684 } 685 686 return ret; 687} 688 689status_t AudioFlinger::setMicMute(bool state) 690{ 691 status_t ret = initCheck(); 692 if (ret != NO_ERROR) { 693 return ret; 694 } 695 696 // check calling permissions 697 if (!settingsAllowed()) { 698 return PERMISSION_DENIED; 699 } 700 701 AutoMutex lock(mHardwareLock); 702 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 703 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 704 ret = dev->set_mic_mute(dev, state); 705 mHardwareStatus = AUDIO_HW_IDLE; 706 return ret; 707} 708 709bool AudioFlinger::getMicMute() const 710{ 711 status_t ret = initCheck(); 712 if (ret != NO_ERROR) { 713 return false; 714 } 715 716 bool state = AUDIO_MODE_INVALID; 717 AutoMutex lock(mHardwareLock); 718 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 719 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 720 dev->get_mic_mute(dev, &state); 721 mHardwareStatus = AUDIO_HW_IDLE; 722 return state; 723} 724 725status_t AudioFlinger::setMasterMute(bool muted) 726{ 727 status_t ret = initCheck(); 728 if (ret != NO_ERROR) { 729 return ret; 730 } 731 732 // check calling permissions 733 if (!settingsAllowed()) { 734 return PERMISSION_DENIED; 735 } 736 737 Mutex::Autolock _l(mLock); 738 mMasterMute = muted; 739 740 // Set master mute in the HALs which support it. 741 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 742 AutoMutex lock(mHardwareLock); 743 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 744 745 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 746 if (dev->canSetMasterMute()) { 747 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 748 } 749 mHardwareStatus = AUDIO_HW_IDLE; 750 } 751 752 // Now set the master mute in each playback thread. Playback threads 753 // assigned to HALs which do not have master mute support will apply master 754 // mute during the mix operation. Threads with HALs which do support master 755 // mute will simply ignore the setting. 756 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 757 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 758 759 return NO_ERROR; 760} 761 762float AudioFlinger::masterVolume() const 763{ 764 Mutex::Autolock _l(mLock); 765 return masterVolume_l(); 766} 767 768bool AudioFlinger::masterMute() const 769{ 770 Mutex::Autolock _l(mLock); 771 return masterMute_l(); 772} 773 774float AudioFlinger::masterVolume_l() const 775{ 776 return mMasterVolume; 777} 778 779bool AudioFlinger::masterMute_l() const 780{ 781 return mMasterMute; 782} 783 784status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 785 audio_io_handle_t output) 786{ 787 // check calling permissions 788 if (!settingsAllowed()) { 789 return PERMISSION_DENIED; 790 } 791 792 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 793 ALOGE("setStreamVolume() invalid stream %d", stream); 794 return BAD_VALUE; 795 } 796 797 AutoMutex lock(mLock); 798 PlaybackThread *thread = NULL; 799 if (output) { 800 thread = checkPlaybackThread_l(output); 801 if (thread == NULL) { 802 return BAD_VALUE; 803 } 804 } 805 806 mStreamTypes[stream].volume = value; 807 808 if (thread == NULL) { 809 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 810 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 811 } 812 } else { 813 thread->setStreamVolume(stream, value); 814 } 815 816 return NO_ERROR; 817} 818 819status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 820{ 821 // check calling permissions 822 if (!settingsAllowed()) { 823 return PERMISSION_DENIED; 824 } 825 826 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 827 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 828 ALOGE("setStreamMute() invalid stream %d", stream); 829 return BAD_VALUE; 830 } 831 832 AutoMutex lock(mLock); 833 mStreamTypes[stream].mute = muted; 834 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 835 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 836 837 return NO_ERROR; 838} 839 840float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 841{ 842 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 843 return 0.0f; 844 } 845 846 AutoMutex lock(mLock); 847 float volume; 848 if (output) { 849 PlaybackThread *thread = checkPlaybackThread_l(output); 850 if (thread == NULL) { 851 return 0.0f; 852 } 853 volume = thread->streamVolume(stream); 854 } else { 855 volume = streamVolume_l(stream); 856 } 857 858 return volume; 859} 860 861bool AudioFlinger::streamMute(audio_stream_type_t stream) const 862{ 863 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 864 return true; 865 } 866 867 AutoMutex lock(mLock); 868 return streamMute_l(stream); 869} 870 871status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 872{ 873 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 874 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 875 // check calling permissions 876 if (!settingsAllowed()) { 877 return PERMISSION_DENIED; 878 } 879 880 // ioHandle == 0 means the parameters are global to the audio hardware interface 881 if (ioHandle == 0) { 882 Mutex::Autolock _l(mLock); 883 status_t final_result = NO_ERROR; 884 { 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 887 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 888 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 889 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 890 final_result = result ?: final_result; 891 } 892 mHardwareStatus = AUDIO_HW_IDLE; 893 } 894 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 895 AudioParameter param = AudioParameter(keyValuePairs); 896 String8 value; 897 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 898 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 899 if (mBtNrecIsOff != btNrecIsOff) { 900 for (size_t i = 0; i < mRecordThreads.size(); i++) { 901 sp<RecordThread> thread = mRecordThreads.valueAt(i); 902 audio_devices_t device = thread->inDevice(); 903 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 904 // collect all of the thread's session IDs 905 KeyedVector<int, bool> ids = thread->sessionIds(); 906 // suspend effects associated with those session IDs 907 for (size_t j = 0; j < ids.size(); ++j) { 908 int sessionId = ids.keyAt(j); 909 thread->setEffectSuspended(FX_IID_AEC, 910 suspend, 911 sessionId); 912 thread->setEffectSuspended(FX_IID_NS, 913 suspend, 914 sessionId); 915 } 916 } 917 mBtNrecIsOff = btNrecIsOff; 918 } 919 } 920 String8 screenState; 921 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 922 bool isOff = screenState == "off"; 923 if (isOff != (gScreenState & 1)) { 924 gScreenState = ((gScreenState & ~1) + 2) | isOff; 925 } 926 } 927 return final_result; 928 } 929 930 // hold a strong ref on thread in case closeOutput() or closeInput() is called 931 // and the thread is exited once the lock is released 932 sp<ThreadBase> thread; 933 { 934 Mutex::Autolock _l(mLock); 935 thread = checkPlaybackThread_l(ioHandle); 936 if (thread == 0) { 937 thread = checkRecordThread_l(ioHandle); 938 } else if (thread == primaryPlaybackThread_l()) { 939 // indicate output device change to all input threads for pre processing 940 AudioParameter param = AudioParameter(keyValuePairs); 941 int value; 942 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 943 (value != 0)) { 944 for (size_t i = 0; i < mRecordThreads.size(); i++) { 945 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 946 } 947 } 948 } 949 } 950 if (thread != 0) { 951 return thread->setParameters(keyValuePairs); 952 } 953 return BAD_VALUE; 954} 955 956String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 957{ 958 ALOGVV("getParameters() io %d, keys %s, tid %d, calling pid %d", 959 ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 960 961 Mutex::Autolock _l(mLock); 962 963 if (ioHandle == 0) { 964 String8 out_s8; 965 966 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 967 char *s; 968 { 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 971 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 972 s = dev->get_parameters(dev, keys.string()); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 } 975 out_s8 += String8(s ? s : ""); 976 free(s); 977 } 978 return out_s8; 979 } 980 981 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 982 if (playbackThread != NULL) { 983 return playbackThread->getParameters(keys); 984 } 985 RecordThread *recordThread = checkRecordThread_l(ioHandle); 986 if (recordThread != NULL) { 987 return recordThread->getParameters(keys); 988 } 989 return String8(""); 990} 991 992size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 993 audio_channel_mask_t channelMask) const 994{ 995 status_t ret = initCheck(); 996 if (ret != NO_ERROR) { 997 return 0; 998 } 999 1000 AutoMutex lock(mHardwareLock); 1001 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1002 struct audio_config config = { 1003 sample_rate: sampleRate, 1004 channel_mask: channelMask, 1005 format: format, 1006 }; 1007 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1008 size_t size = dev->get_input_buffer_size(dev, &config); 1009 mHardwareStatus = AUDIO_HW_IDLE; 1010 return size; 1011} 1012 1013unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1014{ 1015 Mutex::Autolock _l(mLock); 1016 1017 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1018 if (recordThread != NULL) { 1019 return recordThread->getInputFramesLost(); 1020 } 1021 return 0; 1022} 1023 1024status_t AudioFlinger::setVoiceVolume(float value) 1025{ 1026 status_t ret = initCheck(); 1027 if (ret != NO_ERROR) { 1028 return ret; 1029 } 1030 1031 // check calling permissions 1032 if (!settingsAllowed()) { 1033 return PERMISSION_DENIED; 1034 } 1035 1036 AutoMutex lock(mHardwareLock); 1037 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1038 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1039 ret = dev->set_voice_volume(dev, value); 1040 mHardwareStatus = AUDIO_HW_IDLE; 1041 1042 return ret; 1043} 1044 1045status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1046 audio_io_handle_t output) const 1047{ 1048 status_t status; 1049 1050 Mutex::Autolock _l(mLock); 1051 1052 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1053 if (playbackThread != NULL) { 1054 return playbackThread->getRenderPosition(halFrames, dspFrames); 1055 } 1056 1057 return BAD_VALUE; 1058} 1059 1060void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1061{ 1062 1063 Mutex::Autolock _l(mLock); 1064 1065 pid_t pid = IPCThreadState::self()->getCallingPid(); 1066 if (mNotificationClients.indexOfKey(pid) < 0) { 1067 sp<NotificationClient> notificationClient = new NotificationClient(this, 1068 client, 1069 pid); 1070 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1071 1072 mNotificationClients.add(pid, notificationClient); 1073 1074 sp<IBinder> binder = client->asBinder(); 1075 binder->linkToDeath(notificationClient); 1076 1077 // the config change is always sent from playback or record threads to avoid deadlock 1078 // with AudioSystem::gLock 1079 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1080 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1081 } 1082 1083 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1084 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1085 } 1086 } 1087} 1088 1089void AudioFlinger::removeNotificationClient(pid_t pid) 1090{ 1091 Mutex::Autolock _l(mLock); 1092 1093 mNotificationClients.removeItem(pid); 1094 1095 ALOGV("%d died, releasing its sessions", pid); 1096 size_t num = mAudioSessionRefs.size(); 1097 bool removed = false; 1098 for (size_t i = 0; i< num; ) { 1099 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1100 ALOGV(" pid %d @ %d", ref->mPid, i); 1101 if (ref->mPid == pid) { 1102 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1103 mAudioSessionRefs.removeAt(i); 1104 delete ref; 1105 removed = true; 1106 num--; 1107 } else { 1108 i++; 1109 } 1110 } 1111 if (removed) { 1112 purgeStaleEffects_l(); 1113 } 1114} 1115 1116// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1117void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1118{ 1119 size_t size = mNotificationClients.size(); 1120 for (size_t i = 0; i < size; i++) { 1121 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1122 param2); 1123 } 1124} 1125 1126// removeClient_l() must be called with AudioFlinger::mLock held 1127void AudioFlinger::removeClient_l(pid_t pid) 1128{ 1129 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), 1130 IPCThreadState::self()->getCallingPid()); 1131 mClients.removeItem(pid); 1132} 1133 1134// getEffectThread_l() must be called with AudioFlinger::mLock held 1135sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1136{ 1137 sp<PlaybackThread> thread; 1138 1139 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1140 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1141 ALOG_ASSERT(thread == 0); 1142 thread = mPlaybackThreads.valueAt(i); 1143 } 1144 } 1145 1146 return thread; 1147} 1148 1149// ---------------------------------------------------------------------------- 1150 1151AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1152 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1153 : Thread(false /*canCallJava*/), 1154 mType(type), 1155 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1156 // mChannelMask 1157 mChannelCount(0), 1158 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1159 mParamStatus(NO_ERROR), 1160 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1161 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1162 // mName will be set by concrete (non-virtual) subclass 1163 mDeathRecipient(new PMDeathRecipient(this)) 1164{ 1165} 1166 1167AudioFlinger::ThreadBase::~ThreadBase() 1168{ 1169 mParamCond.broadcast(); 1170 // do not lock the mutex in destructor 1171 releaseWakeLock_l(); 1172 if (mPowerManager != 0) { 1173 sp<IBinder> binder = mPowerManager->asBinder(); 1174 binder->unlinkToDeath(mDeathRecipient); 1175 } 1176} 1177 1178void AudioFlinger::ThreadBase::exit() 1179{ 1180 ALOGV("ThreadBase::exit"); 1181 // do any cleanup required for exit to succeed 1182 preExit(); 1183 { 1184 // This lock prevents the following race in thread (uniprocessor for illustration): 1185 // if (!exitPending()) { 1186 // // context switch from here to exit() 1187 // // exit() calls requestExit(), what exitPending() observes 1188 // // exit() calls signal(), which is dropped since no waiters 1189 // // context switch back from exit() to here 1190 // mWaitWorkCV.wait(...); 1191 // // now thread is hung 1192 // } 1193 AutoMutex lock(mLock); 1194 requestExit(); 1195 mWaitWorkCV.broadcast(); 1196 } 1197 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1198 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1199 requestExitAndWait(); 1200} 1201 1202status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1203{ 1204 status_t status; 1205 1206 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1207 Mutex::Autolock _l(mLock); 1208 1209 mNewParameters.add(keyValuePairs); 1210 mWaitWorkCV.signal(); 1211 // wait condition with timeout in case the thread loop has exited 1212 // before the request could be processed 1213 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1214 status = mParamStatus; 1215 mWaitWorkCV.signal(); 1216 } else { 1217 status = TIMED_OUT; 1218 } 1219 return status; 1220} 1221 1222void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 1223{ 1224 Mutex::Autolock _l(mLock); 1225 sendIoConfigEvent_l(event, param); 1226} 1227 1228// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 1229void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 1230{ 1231 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 1232 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 1233 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 1234 param); 1235 mWaitWorkCV.signal(); 1236} 1237 1238// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 1239void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 1240{ 1241 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 1242 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 1243 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 1244 mConfigEvents.size(), pid, tid, prio); 1245 mWaitWorkCV.signal(); 1246} 1247 1248void AudioFlinger::ThreadBase::processConfigEvents() 1249{ 1250 mLock.lock(); 1251 while (!mConfigEvents.isEmpty()) { 1252 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1253 ConfigEvent *event = mConfigEvents[0]; 1254 mConfigEvents.removeAt(0); 1255 // release mLock before locking AudioFlinger mLock: lock order is always 1256 // AudioFlinger then ThreadBase to avoid cross deadlock 1257 mLock.unlock(); 1258 switch(event->type()) { 1259 case CFG_EVENT_PRIO: { 1260 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 1261 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 1262 if (err != 0) { 1263 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 1264 "error %d", 1265 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 1266 } 1267 } break; 1268 case CFG_EVENT_IO: { 1269 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 1270 mAudioFlinger->mLock.lock(); 1271 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 1272 mAudioFlinger->mLock.unlock(); 1273 } break; 1274 default: 1275 ALOGE("processConfigEvents() unknown event type %d", event->type()); 1276 break; 1277 } 1278 delete event; 1279 mLock.lock(); 1280 } 1281 mLock.unlock(); 1282} 1283 1284void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1285{ 1286 const size_t SIZE = 256; 1287 char buffer[SIZE]; 1288 String8 result; 1289 1290 bool locked = tryLock(mLock); 1291 if (!locked) { 1292 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1293 write(fd, buffer, strlen(buffer)); 1294 } 1295 1296 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1297 result.append(buffer); 1298 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1299 result.append(buffer); 1300 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1301 result.append(buffer); 1302 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 1303 result.append(buffer); 1304 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1305 result.append(buffer); 1306 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1307 result.append(buffer); 1308 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1309 result.append(buffer); 1310 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1311 result.append(buffer); 1312 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1313 result.append(buffer); 1314 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1315 result.append(buffer); 1316 1317 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1318 result.append(buffer); 1319 result.append(" Index Command"); 1320 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1321 snprintf(buffer, SIZE, "\n %02d ", i); 1322 result.append(buffer); 1323 result.append(mNewParameters[i]); 1324 } 1325 1326 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1327 result.append(buffer); 1328 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1329 mConfigEvents[i]->dump(buffer, SIZE); 1330 result.append(buffer); 1331 } 1332 result.append("\n"); 1333 1334 write(fd, result.string(), result.size()); 1335 1336 if (locked) { 1337 mLock.unlock(); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1342{ 1343 const size_t SIZE = 256; 1344 char buffer[SIZE]; 1345 String8 result; 1346 1347 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1348 write(fd, buffer, strlen(buffer)); 1349 1350 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1351 sp<EffectChain> chain = mEffectChains[i]; 1352 if (chain != 0) { 1353 chain->dump(fd, args); 1354 } 1355 } 1356} 1357 1358void AudioFlinger::ThreadBase::acquireWakeLock() 1359{ 1360 Mutex::Autolock _l(mLock); 1361 acquireWakeLock_l(); 1362} 1363 1364void AudioFlinger::ThreadBase::acquireWakeLock_l() 1365{ 1366 if (mPowerManager == 0) { 1367 // use checkService() to avoid blocking if power service is not up yet 1368 sp<IBinder> binder = 1369 defaultServiceManager()->checkService(String16("power")); 1370 if (binder == 0) { 1371 ALOGW("Thread %s cannot connect to the power manager service", mName); 1372 } else { 1373 mPowerManager = interface_cast<IPowerManager>(binder); 1374 binder->linkToDeath(mDeathRecipient); 1375 } 1376 } 1377 if (mPowerManager != 0) { 1378 sp<IBinder> binder = new BBinder(); 1379 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1380 binder, 1381 String16(mName)); 1382 if (status == NO_ERROR) { 1383 mWakeLockToken = binder; 1384 } 1385 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1386 } 1387} 1388 1389void AudioFlinger::ThreadBase::releaseWakeLock() 1390{ 1391 Mutex::Autolock _l(mLock); 1392 releaseWakeLock_l(); 1393} 1394 1395void AudioFlinger::ThreadBase::releaseWakeLock_l() 1396{ 1397 if (mWakeLockToken != 0) { 1398 ALOGV("releaseWakeLock_l() %s", mName); 1399 if (mPowerManager != 0) { 1400 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1401 } 1402 mWakeLockToken.clear(); 1403 } 1404} 1405 1406void AudioFlinger::ThreadBase::clearPowerManager() 1407{ 1408 Mutex::Autolock _l(mLock); 1409 releaseWakeLock_l(); 1410 mPowerManager.clear(); 1411} 1412 1413void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1414{ 1415 sp<ThreadBase> thread = mThread.promote(); 1416 if (thread != 0) { 1417 thread->clearPowerManager(); 1418 } 1419 ALOGW("power manager service died !!!"); 1420} 1421 1422void AudioFlinger::ThreadBase::setEffectSuspended( 1423 const effect_uuid_t *type, bool suspend, int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 setEffectSuspended_l(type, suspend, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::setEffectSuspended_l( 1430 const effect_uuid_t *type, bool suspend, int sessionId) 1431{ 1432 sp<EffectChain> chain = getEffectChain_l(sessionId); 1433 if (chain != 0) { 1434 if (type != NULL) { 1435 chain->setEffectSuspended_l(type, suspend); 1436 } else { 1437 chain->setEffectSuspendedAll_l(suspend); 1438 } 1439 } 1440 1441 updateSuspendedSessions_l(type, suspend, sessionId); 1442} 1443 1444void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1445{ 1446 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1447 if (index < 0) { 1448 return; 1449 } 1450 1451 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1452 mSuspendedSessions.valueAt(index); 1453 1454 for (size_t i = 0; i < sessionEffects.size(); i++) { 1455 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1456 for (int j = 0; j < desc->mRefCount; j++) { 1457 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1458 chain->setEffectSuspendedAll_l(true); 1459 } else { 1460 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1461 desc->mType.timeLow); 1462 chain->setEffectSuspended_l(&desc->mType, true); 1463 } 1464 } 1465 } 1466} 1467 1468void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1469 bool suspend, 1470 int sessionId) 1471{ 1472 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1473 1474 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1475 1476 if (suspend) { 1477 if (index >= 0) { 1478 sessionEffects = mSuspendedSessions.valueAt(index); 1479 } else { 1480 mSuspendedSessions.add(sessionId, sessionEffects); 1481 } 1482 } else { 1483 if (index < 0) { 1484 return; 1485 } 1486 sessionEffects = mSuspendedSessions.valueAt(index); 1487 } 1488 1489 1490 int key = EffectChain::kKeyForSuspendAll; 1491 if (type != NULL) { 1492 key = type->timeLow; 1493 } 1494 index = sessionEffects.indexOfKey(key); 1495 1496 sp<SuspendedSessionDesc> desc; 1497 if (suspend) { 1498 if (index >= 0) { 1499 desc = sessionEffects.valueAt(index); 1500 } else { 1501 desc = new SuspendedSessionDesc(); 1502 if (type != NULL) { 1503 desc->mType = *type; 1504 } 1505 sessionEffects.add(key, desc); 1506 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1507 } 1508 desc->mRefCount++; 1509 } else { 1510 if (index < 0) { 1511 return; 1512 } 1513 desc = sessionEffects.valueAt(index); 1514 if (--desc->mRefCount == 0) { 1515 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1516 sessionEffects.removeItemsAt(index); 1517 if (sessionEffects.isEmpty()) { 1518 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1519 sessionId); 1520 mSuspendedSessions.removeItem(sessionId); 1521 } 1522 } 1523 } 1524 if (!sessionEffects.isEmpty()) { 1525 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1526 } 1527} 1528 1529void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1530 bool enabled, 1531 int sessionId) 1532{ 1533 Mutex::Autolock _l(mLock); 1534 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1535} 1536 1537void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1538 bool enabled, 1539 int sessionId) 1540{ 1541 if (mType != RECORD) { 1542 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1543 // another session. This gives the priority to well behaved effect control panels 1544 // and applications not using global effects. 1545 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1546 // global effects 1547 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1548 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1549 } 1550 } 1551 1552 sp<EffectChain> chain = getEffectChain_l(sessionId); 1553 if (chain != 0) { 1554 chain->checkSuspendOnEffectEnabled(effect, enabled); 1555 } 1556} 1557 1558// ---------------------------------------------------------------------------- 1559 1560AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1561 AudioStreamOut* output, 1562 audio_io_handle_t id, 1563 audio_devices_t device, 1564 type_t type) 1565 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1566 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1567 // mStreamTypes[] initialized in constructor body 1568 mOutput(output), 1569 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1570 mMixerStatus(MIXER_IDLE), 1571 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1572 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1573 mScreenState(gScreenState), 1574 // index 0 is reserved for normal mixer's submix 1575 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1576{ 1577 snprintf(mName, kNameLength, "AudioOut_%X", id); 1578 1579 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1580 // it would be safer to explicitly pass initial masterVolume/masterMute as 1581 // parameter. 1582 // 1583 // If the HAL we are using has support for master volume or master mute, 1584 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1585 // and the mute set to false). 1586 mMasterVolume = audioFlinger->masterVolume_l(); 1587 mMasterMute = audioFlinger->masterMute_l(); 1588 if (mOutput && mOutput->audioHwDev) { 1589 if (mOutput->audioHwDev->canSetMasterVolume()) { 1590 mMasterVolume = 1.0; 1591 } 1592 1593 if (mOutput->audioHwDev->canSetMasterMute()) { 1594 mMasterMute = false; 1595 } 1596 } 1597 1598 readOutputParameters(); 1599 1600 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1601 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1602 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1603 stream = (audio_stream_type_t) (stream + 1)) { 1604 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1605 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1606 } 1607 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1608 // because mAudioFlinger doesn't have one to copy from 1609} 1610 1611AudioFlinger::PlaybackThread::~PlaybackThread() 1612{ 1613 delete [] mMixBuffer; 1614} 1615 1616void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1617{ 1618 dumpInternals(fd, args); 1619 dumpTracks(fd, args); 1620 dumpEffectChains(fd, args); 1621} 1622 1623void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1624{ 1625 const size_t SIZE = 256; 1626 char buffer[SIZE]; 1627 String8 result; 1628 1629 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1630 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1631 const stream_type_t *st = &mStreamTypes[i]; 1632 if (i > 0) { 1633 result.appendFormat(", "); 1634 } 1635 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1636 if (st->mute) { 1637 result.append("M"); 1638 } 1639 } 1640 result.append("\n"); 1641 write(fd, result.string(), result.length()); 1642 result.clear(); 1643 1644 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1645 result.append(buffer); 1646 Track::appendDumpHeader(result); 1647 for (size_t i = 0; i < mTracks.size(); ++i) { 1648 sp<Track> track = mTracks[i]; 1649 if (track != 0) { 1650 track->dump(buffer, SIZE); 1651 result.append(buffer); 1652 } 1653 } 1654 1655 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1656 result.append(buffer); 1657 Track::appendDumpHeader(result); 1658 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1659 sp<Track> track = mActiveTracks[i].promote(); 1660 if (track != 0) { 1661 track->dump(buffer, SIZE); 1662 result.append(buffer); 1663 } 1664 } 1665 write(fd, result.string(), result.size()); 1666 1667 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1668 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1669 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1670 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1671} 1672 1673void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1674{ 1675 const size_t SIZE = 256; 1676 char buffer[SIZE]; 1677 String8 result; 1678 1679 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1680 result.append(buffer); 1681 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1682 ns2ms(systemTime() - mLastWriteTime)); 1683 result.append(buffer); 1684 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1685 result.append(buffer); 1686 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1687 result.append(buffer); 1688 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1689 result.append(buffer); 1690 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1691 result.append(buffer); 1692 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1693 result.append(buffer); 1694 write(fd, result.string(), result.size()); 1695 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1696 1697 dumpBase(fd, args); 1698} 1699 1700// Thread virtuals 1701status_t AudioFlinger::PlaybackThread::readyToRun() 1702{ 1703 status_t status = initCheck(); 1704 if (status == NO_ERROR) { 1705 ALOGI("AudioFlinger's thread %p ready to run", this); 1706 } else { 1707 ALOGE("No working audio driver found."); 1708 } 1709 return status; 1710} 1711 1712void AudioFlinger::PlaybackThread::onFirstRef() 1713{ 1714 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1715} 1716 1717// ThreadBase virtuals 1718void AudioFlinger::PlaybackThread::preExit() 1719{ 1720 ALOGV(" preExit()"); 1721 // FIXME this is using hard-coded strings but in the future, this functionality will be 1722 // converted to use audio HAL extensions required to support tunneling 1723 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1724} 1725 1726// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1727sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1728 const sp<AudioFlinger::Client>& client, 1729 audio_stream_type_t streamType, 1730 uint32_t sampleRate, 1731 audio_format_t format, 1732 audio_channel_mask_t channelMask, 1733 int frameCount, 1734 const sp<IMemory>& sharedBuffer, 1735 int sessionId, 1736 IAudioFlinger::track_flags_t *flags, 1737 pid_t tid, 1738 status_t *status) 1739{ 1740 sp<Track> track; 1741 status_t lStatus; 1742 1743 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1744 1745 // client expresses a preference for FAST, but we get the final say 1746 if (*flags & IAudioFlinger::TRACK_FAST) { 1747 if ( 1748 // not timed 1749 (!isTimed) && 1750 // either of these use cases: 1751 ( 1752 // use case 1: shared buffer with any frame count 1753 ( 1754 (sharedBuffer != 0) 1755 ) || 1756 // use case 2: callback handler and frame count is default or at least as large as HAL 1757 ( 1758 (tid != -1) && 1759 ((frameCount == 0) || 1760 (frameCount >= (int) (mFrameCount * kFastTrackMultiplier))) 1761 ) 1762 ) && 1763 // PCM data 1764 audio_is_linear_pcm(format) && 1765 // mono or stereo 1766 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1767 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1768#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1769 // hardware sample rate 1770 (sampleRate == mSampleRate) && 1771#endif 1772 // normal mixer has an associated fast mixer 1773 hasFastMixer() && 1774 // there are sufficient fast track slots available 1775 (mFastTrackAvailMask != 0) 1776 // FIXME test that MixerThread for this fast track has a capable output HAL 1777 // FIXME add a permission test also? 1778 ) { 1779 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1780 if (frameCount == 0) { 1781 frameCount = mFrameCount * kFastTrackMultiplier; 1782 } 1783 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1784 frameCount, mFrameCount); 1785 } else { 1786 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1787 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1788 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1789 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1790 audio_is_linear_pcm(format), 1791 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1792 *flags &= ~IAudioFlinger::TRACK_FAST; 1793 // For compatibility with AudioTrack calculation, buffer depth is forced 1794 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1795 // This is probably too conservative, but legacy application code may depend on it. 1796 // If you change this calculation, also review the start threshold which is related. 1797 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1798 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1799 if (minBufCount < 2) { 1800 minBufCount = 2; 1801 } 1802 int minFrameCount = mNormalFrameCount * minBufCount; 1803 if (frameCount < minFrameCount) { 1804 frameCount = minFrameCount; 1805 } 1806 } 1807 } 1808 1809 if (mType == DIRECT) { 1810 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1811 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1812 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1813 "for output %p with format %d", 1814 sampleRate, format, channelMask, mOutput, mFormat); 1815 lStatus = BAD_VALUE; 1816 goto Exit; 1817 } 1818 } 1819 } else { 1820 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1821 if (sampleRate > mSampleRate*2) { 1822 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1823 lStatus = BAD_VALUE; 1824 goto Exit; 1825 } 1826 } 1827 1828 lStatus = initCheck(); 1829 if (lStatus != NO_ERROR) { 1830 ALOGE("Audio driver not initialized."); 1831 goto Exit; 1832 } 1833 1834 { // scope for mLock 1835 Mutex::Autolock _l(mLock); 1836 1837 // all tracks in same audio session must share the same routing strategy otherwise 1838 // conflicts will happen when tracks are moved from one output to another by audio policy 1839 // manager 1840 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1841 for (size_t i = 0; i < mTracks.size(); ++i) { 1842 sp<Track> t = mTracks[i]; 1843 if (t != 0 && !t->isOutputTrack()) { 1844 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1845 if (sessionId == t->sessionId() && strategy != actual) { 1846 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1847 strategy, actual); 1848 lStatus = BAD_VALUE; 1849 goto Exit; 1850 } 1851 } 1852 } 1853 1854 if (!isTimed) { 1855 track = new Track(this, client, streamType, sampleRate, format, 1856 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1857 } else { 1858 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1859 channelMask, frameCount, sharedBuffer, sessionId); 1860 } 1861 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1862 lStatus = NO_MEMORY; 1863 goto Exit; 1864 } 1865 mTracks.add(track); 1866 1867 sp<EffectChain> chain = getEffectChain_l(sessionId); 1868 if (chain != 0) { 1869 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1870 track->setMainBuffer(chain->inBuffer()); 1871 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1872 chain->incTrackCnt(); 1873 } 1874 1875 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1876 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1877 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1878 // so ask activity manager to do this on our behalf 1879 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1880 } 1881 } 1882 1883 lStatus = NO_ERROR; 1884 1885Exit: 1886 if (status) { 1887 *status = lStatus; 1888 } 1889 return track; 1890} 1891 1892uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1893{ 1894 if (mFastMixer != NULL) { 1895 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1896 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1897 } 1898 return latency; 1899} 1900 1901uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1902{ 1903 return latency; 1904} 1905 1906uint32_t AudioFlinger::PlaybackThread::latency() const 1907{ 1908 Mutex::Autolock _l(mLock); 1909 return latency_l(); 1910} 1911uint32_t AudioFlinger::PlaybackThread::latency_l() const 1912{ 1913 if (initCheck() == NO_ERROR) { 1914 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1915 } else { 1916 return 0; 1917 } 1918} 1919 1920void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1921{ 1922 Mutex::Autolock _l(mLock); 1923 // Don't apply master volume in SW if our HAL can do it for us. 1924 if (mOutput && mOutput->audioHwDev && 1925 mOutput->audioHwDev->canSetMasterVolume()) { 1926 mMasterVolume = 1.0; 1927 } else { 1928 mMasterVolume = value; 1929 } 1930} 1931 1932void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1933{ 1934 Mutex::Autolock _l(mLock); 1935 // Don't apply master mute in SW if our HAL can do it for us. 1936 if (mOutput && mOutput->audioHwDev && 1937 mOutput->audioHwDev->canSetMasterMute()) { 1938 mMasterMute = false; 1939 } else { 1940 mMasterMute = muted; 1941 } 1942} 1943 1944void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1945{ 1946 Mutex::Autolock _l(mLock); 1947 mStreamTypes[stream].volume = value; 1948} 1949 1950void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1951{ 1952 Mutex::Autolock _l(mLock); 1953 mStreamTypes[stream].mute = muted; 1954} 1955 1956float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1957{ 1958 Mutex::Autolock _l(mLock); 1959 return mStreamTypes[stream].volume; 1960} 1961 1962// addTrack_l() must be called with ThreadBase::mLock held 1963status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1964{ 1965 status_t status = ALREADY_EXISTS; 1966 1967 // set retry count for buffer fill 1968 track->mRetryCount = kMaxTrackStartupRetries; 1969 if (mActiveTracks.indexOf(track) < 0) { 1970 // the track is newly added, make sure it fills up all its 1971 // buffers before playing. This is to ensure the client will 1972 // effectively get the latency it requested. 1973 track->mFillingUpStatus = Track::FS_FILLING; 1974 track->mResetDone = false; 1975 track->mPresentationCompleteFrames = 0; 1976 mActiveTracks.add(track); 1977 if (track->mainBuffer() != mMixBuffer) { 1978 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1979 if (chain != 0) { 1980 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1981 track->sessionId()); 1982 chain->incActiveTrackCnt(); 1983 } 1984 } 1985 1986 status = NO_ERROR; 1987 } 1988 1989 ALOGV("mWaitWorkCV.broadcast"); 1990 mWaitWorkCV.broadcast(); 1991 1992 return status; 1993} 1994 1995// destroyTrack_l() must be called with ThreadBase::mLock held 1996void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1997{ 1998 track->mState = TrackBase::TERMINATED; 1999 // active tracks are removed by threadLoop() 2000 if (mActiveTracks.indexOf(track) < 0) { 2001 removeTrack_l(track); 2002 } 2003} 2004 2005void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2006{ 2007 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2008 mTracks.remove(track); 2009 deleteTrackName_l(track->name()); 2010 // redundant as track is about to be destroyed, for dumpsys only 2011 track->mName = -1; 2012 if (track->isFastTrack()) { 2013 int index = track->mFastIndex; 2014 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2015 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2016 mFastTrackAvailMask |= 1 << index; 2017 // redundant as track is about to be destroyed, for dumpsys only 2018 track->mFastIndex = -1; 2019 } 2020 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2021 if (chain != 0) { 2022 chain->decTrackCnt(); 2023 } 2024} 2025 2026String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2027{ 2028 String8 out_s8 = String8(""); 2029 char *s; 2030 2031 Mutex::Autolock _l(mLock); 2032 if (initCheck() != NO_ERROR) { 2033 return out_s8; 2034 } 2035 2036 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2037 out_s8 = String8(s); 2038 free(s); 2039 return out_s8; 2040} 2041 2042// audioConfigChanged_l() must be called with AudioFlinger::mLock held 2043void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 2044 AudioSystem::OutputDescriptor desc; 2045 void *param2 = NULL; 2046 2047 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 2048 param); 2049 2050 switch (event) { 2051 case AudioSystem::OUTPUT_OPENED: 2052 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2053 desc.channels = mChannelMask; 2054 desc.samplingRate = mSampleRate; 2055 desc.format = mFormat; 2056 desc.frameCount = mNormalFrameCount; // FIXME see 2057 // AudioFlinger::frameCount(audio_io_handle_t) 2058 desc.latency = latency(); 2059 param2 = &desc; 2060 break; 2061 2062 case AudioSystem::STREAM_CONFIG_CHANGED: 2063 param2 = ¶m; 2064 case AudioSystem::OUTPUT_CLOSED: 2065 default: 2066 break; 2067 } 2068 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2069} 2070 2071void AudioFlinger::PlaybackThread::readOutputParameters() 2072{ 2073 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2074 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2075 mChannelCount = (uint16_t)popcount(mChannelMask); 2076 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2077 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2078 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2079 if (mFrameCount & 15) { 2080 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2081 mFrameCount); 2082 } 2083 2084 // Calculate size of normal mix buffer relative to the HAL output buffer size 2085 double multiplier = 1.0; 2086 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2087 kUseFastMixer == FastMixer_Dynamic)) { 2088 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2089 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2090 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2091 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2092 maxNormalFrameCount = maxNormalFrameCount & ~15; 2093 if (maxNormalFrameCount < minNormalFrameCount) { 2094 maxNormalFrameCount = minNormalFrameCount; 2095 } 2096 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2097 if (multiplier <= 1.0) { 2098 multiplier = 1.0; 2099 } else if (multiplier <= 2.0) { 2100 if (2 * mFrameCount <= maxNormalFrameCount) { 2101 multiplier = 2.0; 2102 } else { 2103 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2104 } 2105 } else { 2106 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2107 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 2108 // track, but we sometimes have to do this to satisfy the maximum frame count 2109 // constraint) 2110 // FIXME this rounding up should not be done if no HAL SRC 2111 uint32_t truncMult = (uint32_t) multiplier; 2112 if ((truncMult & 1)) { 2113 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2114 ++truncMult; 2115 } 2116 } 2117 multiplier = (double) truncMult; 2118 } 2119 } 2120 mNormalFrameCount = multiplier * mFrameCount; 2121 // round up to nearest 16 frames to satisfy AudioMixer 2122 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2123 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 2124 mNormalFrameCount); 2125 2126 delete[] mMixBuffer; 2127 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2128 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2129 2130 // force reconfiguration of effect chains and engines to take new buffer size and audio 2131 // parameters into account 2132 // Note that mLock is not held when readOutputParameters() is called from the constructor 2133 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2134 // matter. 2135 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2136 Vector< sp<EffectChain> > effectChains = mEffectChains; 2137 for (size_t i = 0; i < effectChains.size(); i ++) { 2138 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2139 } 2140} 2141 2142 2143status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2144{ 2145 if (halFrames == NULL || dspFrames == NULL) { 2146 return BAD_VALUE; 2147 } 2148 Mutex::Autolock _l(mLock); 2149 if (initCheck() != NO_ERROR) { 2150 return INVALID_OPERATION; 2151 } 2152 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2153 2154 if (isSuspended()) { 2155 // return an estimation of rendered frames when the output is suspended 2156 int32_t frames = mBytesWritten - latency_l(); 2157 if (frames < 0) { 2158 frames = 0; 2159 } 2160 *dspFrames = (uint32_t)frames; 2161 return NO_ERROR; 2162 } else { 2163 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2164 } 2165} 2166 2167uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2168{ 2169 Mutex::Autolock _l(mLock); 2170 uint32_t result = 0; 2171 if (getEffectChain_l(sessionId) != 0) { 2172 result = EFFECT_SESSION; 2173 } 2174 2175 for (size_t i = 0; i < mTracks.size(); ++i) { 2176 sp<Track> track = mTracks[i]; 2177 if (sessionId == track->sessionId() && 2178 !(track->mCblk->flags & CBLK_INVALID)) { 2179 result |= TRACK_SESSION; 2180 break; 2181 } 2182 } 2183 2184 return result; 2185} 2186 2187uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2188{ 2189 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2190 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2191 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2192 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2193 } 2194 for (size_t i = 0; i < mTracks.size(); i++) { 2195 sp<Track> track = mTracks[i]; 2196 if (sessionId == track->sessionId() && 2197 !(track->mCblk->flags & CBLK_INVALID)) { 2198 return AudioSystem::getStrategyForStream(track->streamType()); 2199 } 2200 } 2201 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2202} 2203 2204 2205AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2206{ 2207 Mutex::Autolock _l(mLock); 2208 return mOutput; 2209} 2210 2211AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2212{ 2213 Mutex::Autolock _l(mLock); 2214 AudioStreamOut *output = mOutput; 2215 mOutput = NULL; 2216 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2217 // must push a NULL and wait for ack 2218 mOutputSink.clear(); 2219 mPipeSink.clear(); 2220 mNormalSink.clear(); 2221 return output; 2222} 2223 2224// this method must always be called either with ThreadBase mLock held or inside the thread loop 2225audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2226{ 2227 if (mOutput == NULL) { 2228 return NULL; 2229 } 2230 return &mOutput->stream->common; 2231} 2232 2233uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2234{ 2235 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2236} 2237 2238status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2239{ 2240 if (!isValidSyncEvent(event)) { 2241 return BAD_VALUE; 2242 } 2243 2244 Mutex::Autolock _l(mLock); 2245 2246 for (size_t i = 0; i < mTracks.size(); ++i) { 2247 sp<Track> track = mTracks[i]; 2248 if (event->triggerSession() == track->sessionId()) { 2249 (void) track->setSyncEvent(event); 2250 return NO_ERROR; 2251 } 2252 } 2253 2254 return NAME_NOT_FOUND; 2255} 2256 2257bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2258{ 2259 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2260} 2261 2262void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2263 const Vector< sp<Track> >& tracksToRemove) 2264{ 2265 size_t count = tracksToRemove.size(); 2266 if (CC_UNLIKELY(count)) { 2267 for (size_t i = 0 ; i < count ; i++) { 2268 const sp<Track>& track = tracksToRemove.itemAt(i); 2269 if ((track->sharedBuffer() != 0) && 2270 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2271 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2272 } 2273 } 2274 } 2275 2276} 2277 2278// ---------------------------------------------------------------------------- 2279 2280AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2281 audio_io_handle_t id, audio_devices_t device, type_t type) 2282 : PlaybackThread(audioFlinger, output, id, device, type), 2283 // mAudioMixer below 2284 // mFastMixer below 2285 mFastMixerFutex(0) 2286 // mOutputSink below 2287 // mPipeSink below 2288 // mNormalSink below 2289{ 2290 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2291 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2292 "mFrameCount=%d, mNormalFrameCount=%d", 2293 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2294 mNormalFrameCount); 2295 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2296 2297 // FIXME - Current mixer implementation only supports stereo output 2298 if (mChannelCount != FCC_2) { 2299 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2300 } 2301 2302 // create an NBAIO sink for the HAL output stream, and negotiate 2303 mOutputSink = new AudioStreamOutSink(output->stream); 2304 size_t numCounterOffers = 0; 2305 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2306 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2307 ALOG_ASSERT(index == 0); 2308 2309 // initialize fast mixer depending on configuration 2310 bool initFastMixer; 2311 switch (kUseFastMixer) { 2312 case FastMixer_Never: 2313 initFastMixer = false; 2314 break; 2315 case FastMixer_Always: 2316 initFastMixer = true; 2317 break; 2318 case FastMixer_Static: 2319 case FastMixer_Dynamic: 2320 initFastMixer = mFrameCount < mNormalFrameCount; 2321 break; 2322 } 2323 if (initFastMixer) { 2324 2325 // create a MonoPipe to connect our submix to FastMixer 2326 NBAIO_Format format = mOutputSink->format(); 2327 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2328 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2329 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2330 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2331 const NBAIO_Format offers[1] = {format}; 2332 size_t numCounterOffers = 0; 2333 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2334 ALOG_ASSERT(index == 0); 2335 monoPipe->setAvgFrames((mScreenState & 1) ? 2336 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2337 mPipeSink = monoPipe; 2338 2339#ifdef TEE_SINK_FRAMES 2340 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2341 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2342 numCounterOffers = 0; 2343 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2344 ALOG_ASSERT(index == 0); 2345 mTeeSink = teeSink; 2346 PipeReader *teeSource = new PipeReader(*teeSink); 2347 numCounterOffers = 0; 2348 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2349 ALOG_ASSERT(index == 0); 2350 mTeeSource = teeSource; 2351#endif 2352 2353 // create fast mixer and configure it initially with just one fast track for our submix 2354 mFastMixer = new FastMixer(); 2355 FastMixerStateQueue *sq = mFastMixer->sq(); 2356#ifdef STATE_QUEUE_DUMP 2357 sq->setObserverDump(&mStateQueueObserverDump); 2358 sq->setMutatorDump(&mStateQueueMutatorDump); 2359#endif 2360 FastMixerState *state = sq->begin(); 2361 FastTrack *fastTrack = &state->mFastTracks[0]; 2362 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2363 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2364 fastTrack->mVolumeProvider = NULL; 2365 fastTrack->mGeneration++; 2366 state->mFastTracksGen++; 2367 state->mTrackMask = 1; 2368 // fast mixer will use the HAL output sink 2369 state->mOutputSink = mOutputSink.get(); 2370 state->mOutputSinkGen++; 2371 state->mFrameCount = mFrameCount; 2372 state->mCommand = FastMixerState::COLD_IDLE; 2373 // already done in constructor initialization list 2374 //mFastMixerFutex = 0; 2375 state->mColdFutexAddr = &mFastMixerFutex; 2376 state->mColdGen++; 2377 state->mDumpState = &mFastMixerDumpState; 2378 state->mTeeSink = mTeeSink.get(); 2379 sq->end(); 2380 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2381 2382 // start the fast mixer 2383 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2384 pid_t tid = mFastMixer->getTid(); 2385 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2386 if (err != 0) { 2387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2388 kPriorityFastMixer, getpid_cached, tid, err); 2389 } 2390 2391#ifdef AUDIO_WATCHDOG 2392 // create and start the watchdog 2393 mAudioWatchdog = new AudioWatchdog(); 2394 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2395 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2396 tid = mAudioWatchdog->getTid(); 2397 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2398 if (err != 0) { 2399 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2400 kPriorityFastMixer, getpid_cached, tid, err); 2401 } 2402#endif 2403 2404 } else { 2405 mFastMixer = NULL; 2406 } 2407 2408 switch (kUseFastMixer) { 2409 case FastMixer_Never: 2410 case FastMixer_Dynamic: 2411 mNormalSink = mOutputSink; 2412 break; 2413 case FastMixer_Always: 2414 mNormalSink = mPipeSink; 2415 break; 2416 case FastMixer_Static: 2417 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2418 break; 2419 } 2420} 2421 2422AudioFlinger::MixerThread::~MixerThread() 2423{ 2424 if (mFastMixer != NULL) { 2425 FastMixerStateQueue *sq = mFastMixer->sq(); 2426 FastMixerState *state = sq->begin(); 2427 if (state->mCommand == FastMixerState::COLD_IDLE) { 2428 int32_t old = android_atomic_inc(&mFastMixerFutex); 2429 if (old == -1) { 2430 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2431 } 2432 } 2433 state->mCommand = FastMixerState::EXIT; 2434 sq->end(); 2435 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2436 mFastMixer->join(); 2437 // Though the fast mixer thread has exited, it's state queue is still valid. 2438 // We'll use that extract the final state which contains one remaining fast track 2439 // corresponding to our sub-mix. 2440 state = sq->begin(); 2441 ALOG_ASSERT(state->mTrackMask == 1); 2442 FastTrack *fastTrack = &state->mFastTracks[0]; 2443 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2444 delete fastTrack->mBufferProvider; 2445 sq->end(false /*didModify*/); 2446 delete mFastMixer; 2447#ifdef AUDIO_WATCHDOG 2448 if (mAudioWatchdog != 0) { 2449 mAudioWatchdog->requestExit(); 2450 mAudioWatchdog->requestExitAndWait(); 2451 mAudioWatchdog.clear(); 2452 } 2453#endif 2454 } 2455 delete mAudioMixer; 2456} 2457 2458class CpuStats { 2459public: 2460 CpuStats(); 2461 void sample(const String8 &title); 2462#ifdef DEBUG_CPU_USAGE 2463private: 2464 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2465 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2466 2467 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2468 2469 int mCpuNum; // thread's current CPU number 2470 int mCpukHz; // frequency of thread's current CPU in kHz 2471#endif 2472}; 2473 2474CpuStats::CpuStats() 2475#ifdef DEBUG_CPU_USAGE 2476 : mCpuNum(-1), mCpukHz(-1) 2477#endif 2478{ 2479} 2480 2481void CpuStats::sample(const String8 &title) { 2482#ifdef DEBUG_CPU_USAGE 2483 // get current thread's delta CPU time in wall clock ns 2484 double wcNs; 2485 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2486 2487 // record sample for wall clock statistics 2488 if (valid) { 2489 mWcStats.sample(wcNs); 2490 } 2491 2492 // get the current CPU number 2493 int cpuNum = sched_getcpu(); 2494 2495 // get the current CPU frequency in kHz 2496 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2497 2498 // check if either CPU number or frequency changed 2499 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2500 mCpuNum = cpuNum; 2501 mCpukHz = cpukHz; 2502 // ignore sample for purposes of cycles 2503 valid = false; 2504 } 2505 2506 // if no change in CPU number or frequency, then record sample for cycle statistics 2507 if (valid && mCpukHz > 0) { 2508 double cycles = wcNs * cpukHz * 0.000001; 2509 mHzStats.sample(cycles); 2510 } 2511 2512 unsigned n = mWcStats.n(); 2513 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2514 if ((n & 127) == 1) { 2515 long long elapsed = mCpuUsage.elapsed(); 2516 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2517 double perLoop = elapsed / (double) n; 2518 double perLoop100 = perLoop * 0.01; 2519 double perLoop1k = perLoop * 0.001; 2520 double mean = mWcStats.mean(); 2521 double stddev = mWcStats.stddev(); 2522 double minimum = mWcStats.minimum(); 2523 double maximum = mWcStats.maximum(); 2524 double meanCycles = mHzStats.mean(); 2525 double stddevCycles = mHzStats.stddev(); 2526 double minCycles = mHzStats.minimum(); 2527 double maxCycles = mHzStats.maximum(); 2528 mCpuUsage.resetElapsed(); 2529 mWcStats.reset(); 2530 mHzStats.reset(); 2531 ALOGD("CPU usage for %s over past %.1f secs\n" 2532 " (%u mixer loops at %.1f mean ms per loop):\n" 2533 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2534 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2535 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2536 title.string(), 2537 elapsed * .000000001, n, perLoop * .000001, 2538 mean * .001, 2539 stddev * .001, 2540 minimum * .001, 2541 maximum * .001, 2542 mean / perLoop100, 2543 stddev / perLoop100, 2544 minimum / perLoop100, 2545 maximum / perLoop100, 2546 meanCycles / perLoop1k, 2547 stddevCycles / perLoop1k, 2548 minCycles / perLoop1k, 2549 maxCycles / perLoop1k); 2550 2551 } 2552 } 2553#endif 2554}; 2555 2556void AudioFlinger::PlaybackThread::checkSilentMode_l() 2557{ 2558 if (!mMasterMute) { 2559 char value[PROPERTY_VALUE_MAX]; 2560 if (property_get("ro.audio.silent", value, "0") > 0) { 2561 char *endptr; 2562 unsigned long ul = strtoul(value, &endptr, 0); 2563 if (*endptr == '\0' && ul != 0) { 2564 ALOGD("Silence is golden"); 2565 // The setprop command will not allow a property to be changed after 2566 // the first time it is set, so we don't have to worry about un-muting. 2567 setMasterMute_l(true); 2568 } 2569 } 2570 } 2571} 2572 2573bool AudioFlinger::PlaybackThread::threadLoop() 2574{ 2575 Vector< sp<Track> > tracksToRemove; 2576 2577 standbyTime = systemTime(); 2578 2579 // MIXER 2580 nsecs_t lastWarning = 0; 2581 2582 // DUPLICATING 2583 // FIXME could this be made local to while loop? 2584 writeFrames = 0; 2585 2586 cacheParameters_l(); 2587 sleepTime = idleSleepTime; 2588 2589 if (mType == MIXER) { 2590 sleepTimeShift = 0; 2591 } 2592 2593 CpuStats cpuStats; 2594 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2595 2596 acquireWakeLock(); 2597 2598 while (!exitPending()) 2599 { 2600 cpuStats.sample(myName); 2601 2602 Vector< sp<EffectChain> > effectChains; 2603 2604 processConfigEvents(); 2605 2606 { // scope for mLock 2607 2608 Mutex::Autolock _l(mLock); 2609 2610 if (checkForNewParameters_l()) { 2611 cacheParameters_l(); 2612 } 2613 2614 saveOutputTracks(); 2615 2616 // put audio hardware into standby after short delay 2617 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2618 isSuspended())) { 2619 if (!mStandby) { 2620 2621 threadLoop_standby(); 2622 2623 mStandby = true; 2624 } 2625 2626 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2627 // we're about to wait, flush the binder command buffer 2628 IPCThreadState::self()->flushCommands(); 2629 2630 clearOutputTracks(); 2631 2632 if (exitPending()) break; 2633 2634 releaseWakeLock_l(); 2635 // wait until we have something to do... 2636 ALOGV("%s going to sleep", myName.string()); 2637 mWaitWorkCV.wait(mLock); 2638 ALOGV("%s waking up", myName.string()); 2639 acquireWakeLock_l(); 2640 2641 mMixerStatus = MIXER_IDLE; 2642 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2643 mBytesWritten = 0; 2644 2645 checkSilentMode_l(); 2646 2647 standbyTime = systemTime() + standbyDelay; 2648 sleepTime = idleSleepTime; 2649 if (mType == MIXER) { 2650 sleepTimeShift = 0; 2651 } 2652 2653 continue; 2654 } 2655 } 2656 2657 // mMixerStatusIgnoringFastTracks is also updated internally 2658 mMixerStatus = prepareTracks_l(&tracksToRemove); 2659 2660 // prevent any changes in effect chain list and in each effect chain 2661 // during mixing and effect process as the audio buffers could be deleted 2662 // or modified if an effect is created or deleted 2663 lockEffectChains_l(effectChains); 2664 } 2665 2666 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2667 threadLoop_mix(); 2668 } else { 2669 threadLoop_sleepTime(); 2670 } 2671 2672 if (isSuspended()) { 2673 sleepTime = suspendSleepTimeUs(); 2674 mBytesWritten += mixBufferSize; 2675 } 2676 2677 // only process effects if we're going to write 2678 if (sleepTime == 0) { 2679 for (size_t i = 0; i < effectChains.size(); i ++) { 2680 effectChains[i]->process_l(); 2681 } 2682 } 2683 2684 // enable changes in effect chain 2685 unlockEffectChains(effectChains); 2686 2687 // sleepTime == 0 means we must write to audio hardware 2688 if (sleepTime == 0) { 2689 2690 threadLoop_write(); 2691 2692if (mType == MIXER) { 2693 // write blocked detection 2694 nsecs_t now = systemTime(); 2695 nsecs_t delta = now - mLastWriteTime; 2696 if (!mStandby && delta > maxPeriod) { 2697 mNumDelayedWrites++; 2698 if ((now - lastWarning) > kWarningThrottleNs) { 2699#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2700 ScopedTrace st(ATRACE_TAG, "underrun"); 2701#endif 2702 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2703 ns2ms(delta), mNumDelayedWrites, this); 2704 lastWarning = now; 2705 } 2706 } 2707} 2708 2709 mStandby = false; 2710 } else { 2711 usleep(sleepTime); 2712 } 2713 2714 // Finally let go of removed track(s), without the lock held 2715 // since we can't guarantee the destructors won't acquire that 2716 // same lock. This will also mutate and push a new fast mixer state. 2717 threadLoop_removeTracks(tracksToRemove); 2718 tracksToRemove.clear(); 2719 2720 // FIXME I don't understand the need for this here; 2721 // it was in the original code but maybe the 2722 // assignment in saveOutputTracks() makes this unnecessary? 2723 clearOutputTracks(); 2724 2725 // Effect chains will be actually deleted here if they were removed from 2726 // mEffectChains list during mixing or effects processing 2727 effectChains.clear(); 2728 2729 // FIXME Note that the above .clear() is no longer necessary since effectChains 2730 // is now local to this block, but will keep it for now (at least until merge done). 2731 } 2732 2733 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2734 if (mType == MIXER || mType == DIRECT) { 2735 // put output stream into standby mode 2736 if (!mStandby) { 2737 mOutput->stream->common.standby(&mOutput->stream->common); 2738 } 2739 } 2740 2741 releaseWakeLock(); 2742 2743 ALOGV("Thread %p type %d exiting", this, mType); 2744 return false; 2745} 2746 2747void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2748{ 2749 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2750} 2751 2752void AudioFlinger::MixerThread::threadLoop_write() 2753{ 2754 // FIXME we should only do one push per cycle; confirm this is true 2755 // Start the fast mixer if it's not already running 2756 if (mFastMixer != NULL) { 2757 FastMixerStateQueue *sq = mFastMixer->sq(); 2758 FastMixerState *state = sq->begin(); 2759 if (state->mCommand != FastMixerState::MIX_WRITE && 2760 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2761 if (state->mCommand == FastMixerState::COLD_IDLE) { 2762 int32_t old = android_atomic_inc(&mFastMixerFutex); 2763 if (old == -1) { 2764 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2765 } 2766#ifdef AUDIO_WATCHDOG 2767 if (mAudioWatchdog != 0) { 2768 mAudioWatchdog->resume(); 2769 } 2770#endif 2771 } 2772 state->mCommand = FastMixerState::MIX_WRITE; 2773 sq->end(); 2774 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2775 if (kUseFastMixer == FastMixer_Dynamic) { 2776 mNormalSink = mPipeSink; 2777 } 2778 } else { 2779 sq->end(false /*didModify*/); 2780 } 2781 } 2782 PlaybackThread::threadLoop_write(); 2783} 2784 2785// shared by MIXER and DIRECT, overridden by DUPLICATING 2786void AudioFlinger::PlaybackThread::threadLoop_write() 2787{ 2788 // FIXME rewrite to reduce number of system calls 2789 mLastWriteTime = systemTime(); 2790 mInWrite = true; 2791 int bytesWritten; 2792 2793 // If an NBAIO sink is present, use it to write the normal mixer's submix 2794 if (mNormalSink != 0) { 2795#define mBitShift 2 // FIXME 2796 size_t count = mixBufferSize >> mBitShift; 2797#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2798 Tracer::traceBegin(ATRACE_TAG, "write"); 2799#endif 2800 // update the setpoint when gScreenState changes 2801 uint32_t screenState = gScreenState; 2802 if (screenState != mScreenState) { 2803 mScreenState = screenState; 2804 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2805 if (pipe != NULL) { 2806 pipe->setAvgFrames((mScreenState & 1) ? 2807 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2808 } 2809 } 2810 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2811#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2812 Tracer::traceEnd(ATRACE_TAG); 2813#endif 2814 if (framesWritten > 0) { 2815 bytesWritten = framesWritten << mBitShift; 2816 } else { 2817 bytesWritten = framesWritten; 2818 } 2819 // otherwise use the HAL / AudioStreamOut directly 2820 } else { 2821 // Direct output thread. 2822 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2823 } 2824 2825 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2826 mNumWrites++; 2827 mInWrite = false; 2828} 2829 2830void AudioFlinger::MixerThread::threadLoop_standby() 2831{ 2832 // Idle the fast mixer if it's currently running 2833 if (mFastMixer != NULL) { 2834 FastMixerStateQueue *sq = mFastMixer->sq(); 2835 FastMixerState *state = sq->begin(); 2836 if (!(state->mCommand & FastMixerState::IDLE)) { 2837 state->mCommand = FastMixerState::COLD_IDLE; 2838 state->mColdFutexAddr = &mFastMixerFutex; 2839 state->mColdGen++; 2840 mFastMixerFutex = 0; 2841 sq->end(); 2842 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2843 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2844 if (kUseFastMixer == FastMixer_Dynamic) { 2845 mNormalSink = mOutputSink; 2846 } 2847#ifdef AUDIO_WATCHDOG 2848 if (mAudioWatchdog != 0) { 2849 mAudioWatchdog->pause(); 2850 } 2851#endif 2852 } else { 2853 sq->end(false /*didModify*/); 2854 } 2855 } 2856 PlaybackThread::threadLoop_standby(); 2857} 2858 2859// shared by MIXER and DIRECT, overridden by DUPLICATING 2860void AudioFlinger::PlaybackThread::threadLoop_standby() 2861{ 2862 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2863 mOutput->stream->common.standby(&mOutput->stream->common); 2864} 2865 2866void AudioFlinger::MixerThread::threadLoop_mix() 2867{ 2868 // obtain the presentation timestamp of the next output buffer 2869 int64_t pts; 2870 status_t status = INVALID_OPERATION; 2871 2872 if (mNormalSink != 0) { 2873 status = mNormalSink->getNextWriteTimestamp(&pts); 2874 } else { 2875 status = mOutputSink->getNextWriteTimestamp(&pts); 2876 } 2877 2878 if (status != NO_ERROR) { 2879 pts = AudioBufferProvider::kInvalidPTS; 2880 } 2881 2882 // mix buffers... 2883 mAudioMixer->process(pts); 2884 // increase sleep time progressively when application underrun condition clears. 2885 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2886 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2887 // such that we would underrun the audio HAL. 2888 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2889 sleepTimeShift--; 2890 } 2891 sleepTime = 0; 2892 standbyTime = systemTime() + standbyDelay; 2893 //TODO: delay standby when effects have a tail 2894} 2895 2896void AudioFlinger::MixerThread::threadLoop_sleepTime() 2897{ 2898 // If no tracks are ready, sleep once for the duration of an output 2899 // buffer size, then write 0s to the output 2900 if (sleepTime == 0) { 2901 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2902 sleepTime = activeSleepTime >> sleepTimeShift; 2903 if (sleepTime < kMinThreadSleepTimeUs) { 2904 sleepTime = kMinThreadSleepTimeUs; 2905 } 2906 // reduce sleep time in case of consecutive application underruns to avoid 2907 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2908 // duration we would end up writing less data than needed by the audio HAL if 2909 // the condition persists. 2910 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2911 sleepTimeShift++; 2912 } 2913 } else { 2914 sleepTime = idleSleepTime; 2915 } 2916 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2917 memset (mMixBuffer, 0, mixBufferSize); 2918 sleepTime = 0; 2919 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), 2920 "anticipated start"); 2921 } 2922 // TODO add standby time extension fct of effect tail 2923} 2924 2925// prepareTracks_l() must be called with ThreadBase::mLock held 2926AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2927 Vector< sp<Track> > *tracksToRemove) 2928{ 2929 2930 mixer_state mixerStatus = MIXER_IDLE; 2931 // find out which tracks need to be processed 2932 size_t count = mActiveTracks.size(); 2933 size_t mixedTracks = 0; 2934 size_t tracksWithEffect = 0; 2935 // counts only _active_ fast tracks 2936 size_t fastTracks = 0; 2937 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2938 2939 float masterVolume = mMasterVolume; 2940 bool masterMute = mMasterMute; 2941 2942 if (masterMute) { 2943 masterVolume = 0; 2944 } 2945 // Delegate master volume control to effect in output mix effect chain if needed 2946 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2947 if (chain != 0) { 2948 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2949 chain->setVolume_l(&v, &v); 2950 masterVolume = (float)((v + (1 << 23)) >> 24); 2951 chain.clear(); 2952 } 2953 2954 // prepare a new state to push 2955 FastMixerStateQueue *sq = NULL; 2956 FastMixerState *state = NULL; 2957 bool didModify = false; 2958 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2959 if (mFastMixer != NULL) { 2960 sq = mFastMixer->sq(); 2961 state = sq->begin(); 2962 } 2963 2964 for (size_t i=0 ; i<count ; i++) { 2965 sp<Track> t = mActiveTracks[i].promote(); 2966 if (t == 0) continue; 2967 2968 // this const just means the local variable doesn't change 2969 Track* const track = t.get(); 2970 2971 // process fast tracks 2972 if (track->isFastTrack()) { 2973 2974 // It's theoretically possible (though unlikely) for a fast track to be created 2975 // and then removed within the same normal mix cycle. This is not a problem, as 2976 // the track never becomes active so it's fast mixer slot is never touched. 2977 // The converse, of removing an (active) track and then creating a new track 2978 // at the identical fast mixer slot within the same normal mix cycle, 2979 // is impossible because the slot isn't marked available until the end of each cycle. 2980 int j = track->mFastIndex; 2981 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2982 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2983 FastTrack *fastTrack = &state->mFastTracks[j]; 2984 2985 // Determine whether the track is currently in underrun condition, 2986 // and whether it had a recent underrun. 2987 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2988 FastTrackUnderruns underruns = ftDump->mUnderruns; 2989 uint32_t recentFull = (underruns.mBitFields.mFull - 2990 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2991 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2992 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2993 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2994 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2995 uint32_t recentUnderruns = recentPartial + recentEmpty; 2996 track->mObservedUnderruns = underruns; 2997 // don't count underruns that occur while stopping or pausing 2998 // or stopped which can occur when flush() is called while active 2999 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 3000 track->mUnderrunCount += recentUnderruns; 3001 } 3002 3003 // This is similar to the state machine for normal tracks, 3004 // with a few modifications for fast tracks. 3005 bool isActive = true; 3006 switch (track->mState) { 3007 case TrackBase::STOPPING_1: 3008 // track stays active in STOPPING_1 state until first underrun 3009 if (recentUnderruns > 0) { 3010 track->mState = TrackBase::STOPPING_2; 3011 } 3012 break; 3013 case TrackBase::PAUSING: 3014 // ramp down is not yet implemented 3015 track->setPaused(); 3016 break; 3017 case TrackBase::RESUMING: 3018 // ramp up is not yet implemented 3019 track->mState = TrackBase::ACTIVE; 3020 break; 3021 case TrackBase::ACTIVE: 3022 if (recentFull > 0 || recentPartial > 0) { 3023 // track has provided at least some frames recently: reset retry count 3024 track->mRetryCount = kMaxTrackRetries; 3025 } 3026 if (recentUnderruns == 0) { 3027 // no recent underruns: stay active 3028 break; 3029 } 3030 // there has recently been an underrun of some kind 3031 if (track->sharedBuffer() == 0) { 3032 // were any of the recent underruns "empty" (no frames available)? 3033 if (recentEmpty == 0) { 3034 // no, then ignore the partial underruns as they are allowed indefinitely 3035 break; 3036 } 3037 // there has recently been an "empty" underrun: decrement the retry counter 3038 if (--(track->mRetryCount) > 0) { 3039 break; 3040 } 3041 // indicate to client process that the track was disabled because of underrun; 3042 // it will then automatically call start() when data is available 3043 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 3044 // remove from active list, but state remains ACTIVE [confusing but true] 3045 isActive = false; 3046 break; 3047 } 3048 // fall through 3049 case TrackBase::STOPPING_2: 3050 case TrackBase::PAUSED: 3051 case TrackBase::TERMINATED: 3052 case TrackBase::STOPPED: 3053 case TrackBase::FLUSHED: // flush() while active 3054 // Check for presentation complete if track is inactive 3055 // We have consumed all the buffers of this track. 3056 // This would be incomplete if we auto-paused on underrun 3057 { 3058 size_t audioHALFrames = 3059 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3060 size_t framesWritten = 3061 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3062 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3063 // track stays in active list until presentation is complete 3064 break; 3065 } 3066 } 3067 if (track->isStopping_2()) { 3068 track->mState = TrackBase::STOPPED; 3069 } 3070 if (track->isStopped()) { 3071 // Can't reset directly, as fast mixer is still polling this track 3072 // track->reset(); 3073 // So instead mark this track as needing to be reset after push with ack 3074 resetMask |= 1 << i; 3075 } 3076 isActive = false; 3077 break; 3078 case TrackBase::IDLE: 3079 default: 3080 LOG_FATAL("unexpected track state %d", track->mState); 3081 } 3082 3083 if (isActive) { 3084 // was it previously inactive? 3085 if (!(state->mTrackMask & (1 << j))) { 3086 ExtendedAudioBufferProvider *eabp = track; 3087 VolumeProvider *vp = track; 3088 fastTrack->mBufferProvider = eabp; 3089 fastTrack->mVolumeProvider = vp; 3090 fastTrack->mSampleRate = track->mSampleRate; 3091 fastTrack->mChannelMask = track->mChannelMask; 3092 fastTrack->mGeneration++; 3093 state->mTrackMask |= 1 << j; 3094 didModify = true; 3095 // no acknowledgement required for newly active tracks 3096 } 3097 // cache the combined master volume and stream type volume for fast mixer; this 3098 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3099 track->mCachedVolume = track->isMuted() ? 3100 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3101 ++fastTracks; 3102 } else { 3103 // was it previously active? 3104 if (state->mTrackMask & (1 << j)) { 3105 fastTrack->mBufferProvider = NULL; 3106 fastTrack->mGeneration++; 3107 state->mTrackMask &= ~(1 << j); 3108 didModify = true; 3109 // If any fast tracks were removed, we must wait for acknowledgement 3110 // because we're about to decrement the last sp<> on those tracks. 3111 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3112 } else { 3113 LOG_FATAL("fast track %d should have been active", j); 3114 } 3115 tracksToRemove->add(track); 3116 // Avoids a misleading display in dumpsys 3117 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3118 } 3119 continue; 3120 } 3121 3122 { // local variable scope to avoid goto warning 3123 3124 audio_track_cblk_t* cblk = track->cblk(); 3125 3126 // The first time a track is added we wait 3127 // for all its buffers to be filled before processing it 3128 int name = track->name(); 3129 // make sure that we have enough frames to mix one full buffer. 3130 // enforce this condition only once to enable draining the buffer in case the client 3131 // app does not call stop() and relies on underrun to stop: 3132 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3133 // during last round 3134 uint32_t minFrames = 1; 3135 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3136 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3137 if (t->sampleRate() == mSampleRate) { 3138 minFrames = mNormalFrameCount; 3139 } else { 3140 // +1 for rounding and +1 for additional sample needed for interpolation 3141 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3142 // add frames already consumed but not yet released by the resampler 3143 // because cblk->framesReady() will include these frames 3144 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3145 // the minimum track buffer size is normally twice the number of frames necessary 3146 // to fill one buffer and the resampler should not leave more than one buffer worth 3147 // of unreleased frames after each pass, but just in case... 3148 ALOG_ASSERT(minFrames <= cblk->frameCount); 3149 } 3150 } 3151 if ((track->framesReady() >= minFrames) && track->isReady() && 3152 !track->isPaused() && !track->isTerminated()) 3153 { 3154 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 3155 this); 3156 3157 mixedTracks++; 3158 3159 // track->mainBuffer() != mMixBuffer means there is an effect chain 3160 // connected to the track 3161 chain.clear(); 3162 if (track->mainBuffer() != mMixBuffer) { 3163 chain = getEffectChain_l(track->sessionId()); 3164 // Delegate volume control to effect in track effect chain if needed 3165 if (chain != 0) { 3166 tracksWithEffect++; 3167 } else { 3168 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3169 "session %d", 3170 name, track->sessionId()); 3171 } 3172 } 3173 3174 3175 int param = AudioMixer::VOLUME; 3176 if (track->mFillingUpStatus == Track::FS_FILLED) { 3177 // no ramp for the first volume setting 3178 track->mFillingUpStatus = Track::FS_ACTIVE; 3179 if (track->mState == TrackBase::RESUMING) { 3180 track->mState = TrackBase::ACTIVE; 3181 param = AudioMixer::RAMP_VOLUME; 3182 } 3183 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3184 } else if (cblk->server != 0) { 3185 // If the track is stopped before the first frame was mixed, 3186 // do not apply ramp 3187 param = AudioMixer::RAMP_VOLUME; 3188 } 3189 3190 // compute volume for this track 3191 uint32_t vl, vr, va; 3192 if (track->isMuted() || track->isPausing() || 3193 mStreamTypes[track->streamType()].mute) { 3194 vl = vr = va = 0; 3195 if (track->isPausing()) { 3196 track->setPaused(); 3197 } 3198 } else { 3199 3200 // read original volumes with volume control 3201 float typeVolume = mStreamTypes[track->streamType()].volume; 3202 float v = masterVolume * typeVolume; 3203 uint32_t vlr = cblk->getVolumeLR(); 3204 vl = vlr & 0xFFFF; 3205 vr = vlr >> 16; 3206 // track volumes come from shared memory, so can't be trusted and must be clamped 3207 if (vl > MAX_GAIN_INT) { 3208 ALOGV("Track left volume out of range: %04X", vl); 3209 vl = MAX_GAIN_INT; 3210 } 3211 if (vr > MAX_GAIN_INT) { 3212 ALOGV("Track right volume out of range: %04X", vr); 3213 vr = MAX_GAIN_INT; 3214 } 3215 // now apply the master volume and stream type volume 3216 vl = (uint32_t)(v * vl) << 12; 3217 vr = (uint32_t)(v * vr) << 12; 3218 // assuming master volume and stream type volume each go up to 1.0, 3219 // vl and vr are now in 8.24 format 3220 3221 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3222 // send level comes from shared memory and so may be corrupt 3223 if (sendLevel > MAX_GAIN_INT) { 3224 ALOGV("Track send level out of range: %04X", sendLevel); 3225 sendLevel = MAX_GAIN_INT; 3226 } 3227 va = (uint32_t)(v * sendLevel); 3228 } 3229 // Delegate volume control to effect in track effect chain if needed 3230 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3231 // Do not ramp volume if volume is controlled by effect 3232 param = AudioMixer::VOLUME; 3233 track->mHasVolumeController = true; 3234 } else { 3235 // force no volume ramp when volume controller was just disabled or removed 3236 // from effect chain to avoid volume spike 3237 if (track->mHasVolumeController) { 3238 param = AudioMixer::VOLUME; 3239 } 3240 track->mHasVolumeController = false; 3241 } 3242 3243 // Convert volumes from 8.24 to 4.12 format 3244 // This additional clamping is needed in case chain->setVolume_l() overshot 3245 vl = (vl + (1 << 11)) >> 12; 3246 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3247 vr = (vr + (1 << 11)) >> 12; 3248 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3249 3250 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3251 3252 // XXX: these things DON'T need to be done each time 3253 mAudioMixer->setBufferProvider(name, track); 3254 mAudioMixer->enable(name); 3255 3256 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3257 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3258 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3259 mAudioMixer->setParameter( 3260 name, 3261 AudioMixer::TRACK, 3262 AudioMixer::FORMAT, (void *)track->format()); 3263 mAudioMixer->setParameter( 3264 name, 3265 AudioMixer::TRACK, 3266 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3267 mAudioMixer->setParameter( 3268 name, 3269 AudioMixer::RESAMPLE, 3270 AudioMixer::SAMPLE_RATE, 3271 (void *)(cblk->sampleRate)); 3272 mAudioMixer->setParameter( 3273 name, 3274 AudioMixer::TRACK, 3275 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3276 mAudioMixer->setParameter( 3277 name, 3278 AudioMixer::TRACK, 3279 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3280 3281 // reset retry count 3282 track->mRetryCount = kMaxTrackRetries; 3283 3284 // If one track is ready, set the mixer ready if: 3285 // - the mixer was not ready during previous round OR 3286 // - no other track is not ready 3287 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3288 mixerStatus != MIXER_TRACKS_ENABLED) { 3289 mixerStatus = MIXER_TRACKS_READY; 3290 } 3291 } else { 3292 // clear effect chain input buffer if an active track underruns to avoid sending 3293 // previous audio buffer again to effects 3294 chain = getEffectChain_l(track->sessionId()); 3295 if (chain != 0) { 3296 chain->clearInputBuffer(); 3297 } 3298 3299 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 3300 cblk->server, this); 3301 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3302 track->isStopped() || track->isPaused()) { 3303 // We have consumed all the buffers of this track. 3304 // Remove it from the list of active tracks. 3305 // TODO: use actual buffer filling status instead of latency when available from 3306 // audio HAL 3307 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3308 size_t framesWritten = 3309 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3310 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3311 if (track->isStopped()) { 3312 track->reset(); 3313 } 3314 tracksToRemove->add(track); 3315 } 3316 } else { 3317 track->mUnderrunCount++; 3318 // No buffers for this track. Give it a few chances to 3319 // fill a buffer, then remove it from active list. 3320 if (--(track->mRetryCount) <= 0) { 3321 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3322 tracksToRemove->add(track); 3323 // indicate to client process that the track was disabled because of underrun; 3324 // it will then automatically call start() when data is available 3325 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3326 // If one track is not ready, mark the mixer also not ready if: 3327 // - the mixer was ready during previous round OR 3328 // - no other track is ready 3329 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3330 mixerStatus != MIXER_TRACKS_READY) { 3331 mixerStatus = MIXER_TRACKS_ENABLED; 3332 } 3333 } 3334 mAudioMixer->disable(name); 3335 } 3336 3337 } // local variable scope to avoid goto warning 3338track_is_ready: ; 3339 3340 } 3341 3342 // Push the new FastMixer state if necessary 3343 bool pauseAudioWatchdog = false; 3344 if (didModify) { 3345 state->mFastTracksGen++; 3346 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3347 if (kUseFastMixer == FastMixer_Dynamic && 3348 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3349 state->mCommand = FastMixerState::COLD_IDLE; 3350 state->mColdFutexAddr = &mFastMixerFutex; 3351 state->mColdGen++; 3352 mFastMixerFutex = 0; 3353 if (kUseFastMixer == FastMixer_Dynamic) { 3354 mNormalSink = mOutputSink; 3355 } 3356 // If we go into cold idle, need to wait for acknowledgement 3357 // so that fast mixer stops doing I/O. 3358 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3359 pauseAudioWatchdog = true; 3360 } 3361 sq->end(); 3362 } 3363 if (sq != NULL) { 3364 sq->end(didModify); 3365 sq->push(block); 3366 } 3367#ifdef AUDIO_WATCHDOG 3368 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3369 mAudioWatchdog->pause(); 3370 } 3371#endif 3372 3373 // Now perform the deferred reset on fast tracks that have stopped 3374 while (resetMask != 0) { 3375 size_t i = __builtin_ctz(resetMask); 3376 ALOG_ASSERT(i < count); 3377 resetMask &= ~(1 << i); 3378 sp<Track> t = mActiveTracks[i].promote(); 3379 if (t == 0) continue; 3380 Track* track = t.get(); 3381 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3382 track->reset(); 3383 } 3384 3385 // remove all the tracks that need to be... 3386 count = tracksToRemove->size(); 3387 if (CC_UNLIKELY(count)) { 3388 for (size_t i=0 ; i<count ; i++) { 3389 const sp<Track>& track = tracksToRemove->itemAt(i); 3390 mActiveTracks.remove(track); 3391 if (track->mainBuffer() != mMixBuffer) { 3392 chain = getEffectChain_l(track->sessionId()); 3393 if (chain != 0) { 3394 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3395 track->sessionId()); 3396 chain->decActiveTrackCnt(); 3397 } 3398 } 3399 if (track->isTerminated()) { 3400 removeTrack_l(track); 3401 } 3402 } 3403 } 3404 3405 // mix buffer must be cleared if all tracks are connected to an 3406 // effect chain as in this case the mixer will not write to 3407 // mix buffer and track effects will accumulate into it 3408 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3409 (mixedTracks == 0 && fastTracks > 0)) { 3410 // FIXME as a performance optimization, should remember previous zero status 3411 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3412 } 3413 3414 // if any fast tracks, then status is ready 3415 mMixerStatusIgnoringFastTracks = mixerStatus; 3416 if (fastTracks > 0) { 3417 mixerStatus = MIXER_TRACKS_READY; 3418 } 3419 return mixerStatus; 3420} 3421 3422/* 3423The derived values that are cached: 3424 - mixBufferSize from frame count * frame size 3425 - activeSleepTime from activeSleepTimeUs() 3426 - idleSleepTime from idleSleepTimeUs() 3427 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3428 - maxPeriod from frame count and sample rate (MIXER only) 3429 3430The parameters that affect these derived values are: 3431 - frame count 3432 - frame size 3433 - sample rate 3434 - device type: A2DP or not 3435 - device latency 3436 - format: PCM or not 3437 - active sleep time 3438 - idle sleep time 3439*/ 3440 3441void AudioFlinger::PlaybackThread::cacheParameters_l() 3442{ 3443 mixBufferSize = mNormalFrameCount * mFrameSize; 3444 activeSleepTime = activeSleepTimeUs(); 3445 idleSleepTime = idleSleepTimeUs(); 3446} 3447 3448void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3449{ 3450 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3451 this, streamType, mTracks.size()); 3452 Mutex::Autolock _l(mLock); 3453 3454 size_t size = mTracks.size(); 3455 for (size_t i = 0; i < size; i++) { 3456 sp<Track> t = mTracks[i]; 3457 if (t->streamType() == streamType) { 3458 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 3459 t->mCblk->cv.signal(); 3460 } 3461 } 3462} 3463 3464// getTrackName_l() must be called with ThreadBase::mLock held 3465int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3466{ 3467 return mAudioMixer->getTrackName(channelMask, sessionId); 3468} 3469 3470// deleteTrackName_l() must be called with ThreadBase::mLock held 3471void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3472{ 3473 ALOGV("remove track (%d) and delete from mixer", name); 3474 mAudioMixer->deleteTrackName(name); 3475} 3476 3477// checkForNewParameters_l() must be called with ThreadBase::mLock held 3478bool AudioFlinger::MixerThread::checkForNewParameters_l() 3479{ 3480 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3481 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3482 bool reconfig = false; 3483 3484 while (!mNewParameters.isEmpty()) { 3485 3486 if (mFastMixer != NULL) { 3487 FastMixerStateQueue *sq = mFastMixer->sq(); 3488 FastMixerState *state = sq->begin(); 3489 if (!(state->mCommand & FastMixerState::IDLE)) { 3490 previousCommand = state->mCommand; 3491 state->mCommand = FastMixerState::HOT_IDLE; 3492 sq->end(); 3493 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3494 } else { 3495 sq->end(false /*didModify*/); 3496 } 3497 } 3498 3499 status_t status = NO_ERROR; 3500 String8 keyValuePair = mNewParameters[0]; 3501 AudioParameter param = AudioParameter(keyValuePair); 3502 int value; 3503 3504 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3505 reconfig = true; 3506 } 3507 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3508 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3509 status = BAD_VALUE; 3510 } else { 3511 reconfig = true; 3512 } 3513 } 3514 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3515 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3516 status = BAD_VALUE; 3517 } else { 3518 reconfig = true; 3519 } 3520 } 3521 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3522 // do not accept frame count changes if tracks are open as the track buffer 3523 // size depends on frame count and correct behavior would not be guaranteed 3524 // if frame count is changed after track creation 3525 if (!mTracks.isEmpty()) { 3526 status = INVALID_OPERATION; 3527 } else { 3528 reconfig = true; 3529 } 3530 } 3531 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3532#ifdef ADD_BATTERY_DATA 3533 // when changing the audio output device, call addBatteryData to notify 3534 // the change 3535 if (mOutDevice != value) { 3536 uint32_t params = 0; 3537 // check whether speaker is on 3538 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3539 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3540 } 3541 3542 audio_devices_t deviceWithoutSpeaker 3543 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3544 // check if any other device (except speaker) is on 3545 if (value & deviceWithoutSpeaker ) { 3546 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3547 } 3548 3549 if (params != 0) { 3550 addBatteryData(params); 3551 } 3552 } 3553#endif 3554 3555 // forward device change to effects that have requested to be 3556 // aware of attached audio device. 3557 mOutDevice = value; 3558 for (size_t i = 0; i < mEffectChains.size(); i++) { 3559 mEffectChains[i]->setDevice_l(mOutDevice); 3560 } 3561 } 3562 3563 if (status == NO_ERROR) { 3564 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3565 keyValuePair.string()); 3566 if (!mStandby && status == INVALID_OPERATION) { 3567 mOutput->stream->common.standby(&mOutput->stream->common); 3568 mStandby = true; 3569 mBytesWritten = 0; 3570 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3571 keyValuePair.string()); 3572 } 3573 if (status == NO_ERROR && reconfig) { 3574 delete mAudioMixer; 3575 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3576 mAudioMixer = NULL; 3577 readOutputParameters(); 3578 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3579 for (size_t i = 0; i < mTracks.size() ; i++) { 3580 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3581 if (name < 0) break; 3582 mTracks[i]->mName = name; 3583 // limit track sample rate to 2 x new output sample rate 3584 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3585 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3586 } 3587 } 3588 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3589 } 3590 } 3591 3592 mNewParameters.removeAt(0); 3593 3594 mParamStatus = status; 3595 mParamCond.signal(); 3596 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3597 // already timed out waiting for the status and will never signal the condition. 3598 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3599 } 3600 3601 if (!(previousCommand & FastMixerState::IDLE)) { 3602 ALOG_ASSERT(mFastMixer != NULL); 3603 FastMixerStateQueue *sq = mFastMixer->sq(); 3604 FastMixerState *state = sq->begin(); 3605 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3606 state->mCommand = previousCommand; 3607 sq->end(); 3608 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3609 } 3610 3611 return reconfig; 3612} 3613 3614void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3615{ 3616 NBAIO_Source *teeSource = source.get(); 3617 if (teeSource != NULL) { 3618 char teeTime[16]; 3619 struct timeval tv; 3620 gettimeofday(&tv, NULL); 3621 struct tm tm; 3622 localtime_r(&tv.tv_sec, &tm); 3623 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3624 char teePath[64]; 3625 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3626 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3627 if (teeFd >= 0) { 3628 char wavHeader[44]; 3629 memcpy(wavHeader, 3630 "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", 3631 sizeof(wavHeader)); 3632 NBAIO_Format format = teeSource->format(); 3633 unsigned channelCount = Format_channelCount(format); 3634 ALOG_ASSERT(channelCount <= FCC_2); 3635 uint32_t sampleRate = Format_sampleRate(format); 3636 wavHeader[22] = channelCount; // number of channels 3637 wavHeader[24] = sampleRate; // sample rate 3638 wavHeader[25] = sampleRate >> 8; 3639 wavHeader[32] = channelCount * 2; // block alignment 3640 write(teeFd, wavHeader, sizeof(wavHeader)); 3641 size_t total = 0; 3642 bool firstRead = true; 3643 for (;;) { 3644#define TEE_SINK_READ 1024 3645 short buffer[TEE_SINK_READ * FCC_2]; 3646 size_t count = TEE_SINK_READ; 3647 ssize_t actual = teeSource->read(buffer, count, 3648 AudioBufferProvider::kInvalidPTS); 3649 bool wasFirstRead = firstRead; 3650 firstRead = false; 3651 if (actual <= 0) { 3652 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3653 continue; 3654 } 3655 break; 3656 } 3657 ALOG_ASSERT(actual <= (ssize_t)count); 3658 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3659 total += actual; 3660 } 3661 lseek(teeFd, (off_t) 4, SEEK_SET); 3662 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3663 write(teeFd, &temp, sizeof(temp)); 3664 lseek(teeFd, (off_t) 40, SEEK_SET); 3665 temp = total * channelCount * sizeof(short); 3666 write(teeFd, &temp, sizeof(temp)); 3667 close(teeFd); 3668 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3669 } else { 3670 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3671 } 3672 } 3673} 3674 3675void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3676{ 3677 const size_t SIZE = 256; 3678 char buffer[SIZE]; 3679 String8 result; 3680 3681 PlaybackThread::dumpInternals(fd, args); 3682 3683 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3684 result.append(buffer); 3685 write(fd, result.string(), result.size()); 3686 3687 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3688 FastMixerDumpState copy = mFastMixerDumpState; 3689 copy.dump(fd); 3690 3691#ifdef STATE_QUEUE_DUMP 3692 // Similar for state queue 3693 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3694 observerCopy.dump(fd); 3695 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3696 mutatorCopy.dump(fd); 3697#endif 3698 3699 // Write the tee output to a .wav file 3700 dumpTee(fd, mTeeSource, mId); 3701 3702#ifdef AUDIO_WATCHDOG 3703 if (mAudioWatchdog != 0) { 3704 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3705 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3706 wdCopy.dump(fd); 3707 } 3708#endif 3709} 3710 3711uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3712{ 3713 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3714} 3715 3716uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3717{ 3718 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3719} 3720 3721void AudioFlinger::MixerThread::cacheParameters_l() 3722{ 3723 PlaybackThread::cacheParameters_l(); 3724 3725 // FIXME: Relaxed timing because of a certain device that can't meet latency 3726 // Should be reduced to 2x after the vendor fixes the driver issue 3727 // increase threshold again due to low power audio mode. The way this warning 3728 // threshold is calculated and its usefulness should be reconsidered anyway. 3729 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3730} 3731 3732// ---------------------------------------------------------------------------- 3733AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3734 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3735 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3736 // mLeftVolFloat, mRightVolFloat 3737{ 3738} 3739 3740AudioFlinger::DirectOutputThread::~DirectOutputThread() 3741{ 3742} 3743 3744AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3745 Vector< sp<Track> > *tracksToRemove 3746) 3747{ 3748 sp<Track> trackToRemove; 3749 3750 mixer_state mixerStatus = MIXER_IDLE; 3751 3752 // find out which tracks need to be processed 3753 if (mActiveTracks.size() != 0) { 3754 sp<Track> t = mActiveTracks[0].promote(); 3755 // The track died recently 3756 if (t == 0) return MIXER_IDLE; 3757 3758 Track* const track = t.get(); 3759 audio_track_cblk_t* cblk = track->cblk(); 3760 3761 // The first time a track is added we wait 3762 // for all its buffers to be filled before processing it 3763 uint32_t minFrames; 3764 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3765 minFrames = mNormalFrameCount; 3766 } else { 3767 minFrames = 1; 3768 } 3769 if ((track->framesReady() >= minFrames) && track->isReady() && 3770 !track->isPaused() && !track->isTerminated()) 3771 { 3772 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3773 3774 if (track->mFillingUpStatus == Track::FS_FILLED) { 3775 track->mFillingUpStatus = Track::FS_ACTIVE; 3776 mLeftVolFloat = mRightVolFloat = 0; 3777 if (track->mState == TrackBase::RESUMING) { 3778 track->mState = TrackBase::ACTIVE; 3779 } 3780 } 3781 3782 // compute volume for this track 3783 float left, right; 3784 if (track->isMuted() || mMasterMute || track->isPausing() || 3785 mStreamTypes[track->streamType()].mute) { 3786 left = right = 0; 3787 if (track->isPausing()) { 3788 track->setPaused(); 3789 } 3790 } else { 3791 float typeVolume = mStreamTypes[track->streamType()].volume; 3792 float v = mMasterVolume * typeVolume; 3793 uint32_t vlr = cblk->getVolumeLR(); 3794 float v_clamped = v * (vlr & 0xFFFF); 3795 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3796 left = v_clamped/MAX_GAIN; 3797 v_clamped = v * (vlr >> 16); 3798 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3799 right = v_clamped/MAX_GAIN; 3800 } 3801 3802 if (left != mLeftVolFloat || right != mRightVolFloat) { 3803 mLeftVolFloat = left; 3804 mRightVolFloat = right; 3805 3806 // Convert volumes from float to 8.24 3807 uint32_t vl = (uint32_t)(left * (1 << 24)); 3808 uint32_t vr = (uint32_t)(right * (1 << 24)); 3809 3810 // Delegate volume control to effect in track effect chain if needed 3811 // only one effect chain can be present on DirectOutputThread, so if 3812 // there is one, the track is connected to it 3813 if (!mEffectChains.isEmpty()) { 3814 // Do not ramp volume if volume is controlled by effect 3815 mEffectChains[0]->setVolume_l(&vl, &vr); 3816 left = (float)vl / (1 << 24); 3817 right = (float)vr / (1 << 24); 3818 } 3819 mOutput->stream->set_volume(mOutput->stream, left, right); 3820 } 3821 3822 // reset retry count 3823 track->mRetryCount = kMaxTrackRetriesDirect; 3824 mActiveTrack = t; 3825 mixerStatus = MIXER_TRACKS_READY; 3826 } else { 3827 // clear effect chain input buffer if an active track underruns to avoid sending 3828 // previous audio buffer again to effects 3829 if (!mEffectChains.isEmpty()) { 3830 mEffectChains[0]->clearInputBuffer(); 3831 } 3832 3833 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3834 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3835 track->isStopped() || track->isPaused()) { 3836 // We have consumed all the buffers of this track. 3837 // Remove it from the list of active tracks. 3838 // TODO: implement behavior for compressed audio 3839 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3840 size_t framesWritten = 3841 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3842 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3843 if (track->isStopped()) { 3844 track->reset(); 3845 } 3846 trackToRemove = track; 3847 } 3848 } else { 3849 // No buffers for this track. Give it a few chances to 3850 // fill a buffer, then remove it from active list. 3851 if (--(track->mRetryCount) <= 0) { 3852 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3853 trackToRemove = track; 3854 } else { 3855 mixerStatus = MIXER_TRACKS_ENABLED; 3856 } 3857 } 3858 } 3859 } 3860 3861 // FIXME merge this with similar code for removing multiple tracks 3862 // remove all the tracks that need to be... 3863 if (CC_UNLIKELY(trackToRemove != 0)) { 3864 tracksToRemove->add(trackToRemove); 3865 mActiveTracks.remove(trackToRemove); 3866 if (!mEffectChains.isEmpty()) { 3867 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3868 trackToRemove->sessionId()); 3869 mEffectChains[0]->decActiveTrackCnt(); 3870 } 3871 if (trackToRemove->isTerminated()) { 3872 removeTrack_l(trackToRemove); 3873 } 3874 } 3875 3876 return mixerStatus; 3877} 3878 3879void AudioFlinger::DirectOutputThread::threadLoop_mix() 3880{ 3881 AudioBufferProvider::Buffer buffer; 3882 size_t frameCount = mFrameCount; 3883 int8_t *curBuf = (int8_t *)mMixBuffer; 3884 // output audio to hardware 3885 while (frameCount) { 3886 buffer.frameCount = frameCount; 3887 mActiveTrack->getNextBuffer(&buffer); 3888 if (CC_UNLIKELY(buffer.raw == NULL)) { 3889 memset(curBuf, 0, frameCount * mFrameSize); 3890 break; 3891 } 3892 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3893 frameCount -= buffer.frameCount; 3894 curBuf += buffer.frameCount * mFrameSize; 3895 mActiveTrack->releaseBuffer(&buffer); 3896 } 3897 sleepTime = 0; 3898 standbyTime = systemTime() + standbyDelay; 3899 mActiveTrack.clear(); 3900 3901} 3902 3903void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3904{ 3905 if (sleepTime == 0) { 3906 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3907 sleepTime = activeSleepTime; 3908 } else { 3909 sleepTime = idleSleepTime; 3910 } 3911 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3912 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3913 sleepTime = 0; 3914 } 3915} 3916 3917// getTrackName_l() must be called with ThreadBase::mLock held 3918int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3919 int sessionId) 3920{ 3921 return 0; 3922} 3923 3924// deleteTrackName_l() must be called with ThreadBase::mLock held 3925void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3926{ 3927} 3928 3929// checkForNewParameters_l() must be called with ThreadBase::mLock held 3930bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3931{ 3932 bool reconfig = false; 3933 3934 while (!mNewParameters.isEmpty()) { 3935 status_t status = NO_ERROR; 3936 String8 keyValuePair = mNewParameters[0]; 3937 AudioParameter param = AudioParameter(keyValuePair); 3938 int value; 3939 3940 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3941 // do not accept frame count changes if tracks are open as the track buffer 3942 // size depends on frame count and correct behavior would not be garantied 3943 // if frame count is changed after track creation 3944 if (!mTracks.isEmpty()) { 3945 status = INVALID_OPERATION; 3946 } else { 3947 reconfig = true; 3948 } 3949 } 3950 if (status == NO_ERROR) { 3951 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3952 keyValuePair.string()); 3953 if (!mStandby && status == INVALID_OPERATION) { 3954 mOutput->stream->common.standby(&mOutput->stream->common); 3955 mStandby = true; 3956 mBytesWritten = 0; 3957 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3958 keyValuePair.string()); 3959 } 3960 if (status == NO_ERROR && reconfig) { 3961 readOutputParameters(); 3962 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3963 } 3964 } 3965 3966 mNewParameters.removeAt(0); 3967 3968 mParamStatus = status; 3969 mParamCond.signal(); 3970 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3971 // already timed out waiting for the status and will never signal the condition. 3972 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3973 } 3974 return reconfig; 3975} 3976 3977uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3978{ 3979 uint32_t time; 3980 if (audio_is_linear_pcm(mFormat)) { 3981 time = PlaybackThread::activeSleepTimeUs(); 3982 } else { 3983 time = 10000; 3984 } 3985 return time; 3986} 3987 3988uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3989{ 3990 uint32_t time; 3991 if (audio_is_linear_pcm(mFormat)) { 3992 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3993 } else { 3994 time = 10000; 3995 } 3996 return time; 3997} 3998 3999uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4000{ 4001 uint32_t time; 4002 if (audio_is_linear_pcm(mFormat)) { 4003 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4004 } else { 4005 time = 10000; 4006 } 4007 return time; 4008} 4009 4010void AudioFlinger::DirectOutputThread::cacheParameters_l() 4011{ 4012 PlaybackThread::cacheParameters_l(); 4013 4014 // use shorter standby delay as on normal output to release 4015 // hardware resources as soon as possible 4016 standbyDelay = microseconds(activeSleepTime*2); 4017} 4018 4019// ---------------------------------------------------------------------------- 4020 4021AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4022 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4023 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4024 DUPLICATING), 4025 mWaitTimeMs(UINT_MAX) 4026{ 4027 addOutputTrack(mainThread); 4028} 4029 4030AudioFlinger::DuplicatingThread::~DuplicatingThread() 4031{ 4032 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4033 mOutputTracks[i]->destroy(); 4034 } 4035} 4036 4037void AudioFlinger::DuplicatingThread::threadLoop_mix() 4038{ 4039 // mix buffers... 4040 if (outputsReady(outputTracks)) { 4041 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4042 } else { 4043 memset(mMixBuffer, 0, mixBufferSize); 4044 } 4045 sleepTime = 0; 4046 writeFrames = mNormalFrameCount; 4047 standbyTime = systemTime() + standbyDelay; 4048} 4049 4050void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4051{ 4052 if (sleepTime == 0) { 4053 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4054 sleepTime = activeSleepTime; 4055 } else { 4056 sleepTime = idleSleepTime; 4057 } 4058 } else if (mBytesWritten != 0) { 4059 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4060 writeFrames = mNormalFrameCount; 4061 memset(mMixBuffer, 0, mixBufferSize); 4062 } else { 4063 // flush remaining overflow buffers in output tracks 4064 writeFrames = 0; 4065 } 4066 sleepTime = 0; 4067 } 4068} 4069 4070void AudioFlinger::DuplicatingThread::threadLoop_write() 4071{ 4072 for (size_t i = 0; i < outputTracks.size(); i++) { 4073 outputTracks[i]->write(mMixBuffer, writeFrames); 4074 } 4075 mBytesWritten += mixBufferSize; 4076} 4077 4078void AudioFlinger::DuplicatingThread::threadLoop_standby() 4079{ 4080 // DuplicatingThread implements standby by stopping all tracks 4081 for (size_t i = 0; i < outputTracks.size(); i++) { 4082 outputTracks[i]->stop(); 4083 } 4084} 4085 4086void AudioFlinger::DuplicatingThread::saveOutputTracks() 4087{ 4088 outputTracks = mOutputTracks; 4089} 4090 4091void AudioFlinger::DuplicatingThread::clearOutputTracks() 4092{ 4093 outputTracks.clear(); 4094} 4095 4096void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4097{ 4098 Mutex::Autolock _l(mLock); 4099 // FIXME explain this formula 4100 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4101 OutputTrack *outputTrack = new OutputTrack(thread, 4102 this, 4103 mSampleRate, 4104 mFormat, 4105 mChannelMask, 4106 frameCount); 4107 if (outputTrack->cblk() != NULL) { 4108 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4109 mOutputTracks.add(outputTrack); 4110 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4111 updateWaitTime_l(); 4112 } 4113} 4114 4115void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4116{ 4117 Mutex::Autolock _l(mLock); 4118 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4119 if (mOutputTracks[i]->thread() == thread) { 4120 mOutputTracks[i]->destroy(); 4121 mOutputTracks.removeAt(i); 4122 updateWaitTime_l(); 4123 return; 4124 } 4125 } 4126 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4127} 4128 4129// caller must hold mLock 4130void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4131{ 4132 mWaitTimeMs = UINT_MAX; 4133 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4134 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4135 if (strong != 0) { 4136 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4137 if (waitTimeMs < mWaitTimeMs) { 4138 mWaitTimeMs = waitTimeMs; 4139 } 4140 } 4141 } 4142} 4143 4144 4145bool AudioFlinger::DuplicatingThread::outputsReady( 4146 const SortedVector< sp<OutputTrack> > &outputTracks) 4147{ 4148 for (size_t i = 0; i < outputTracks.size(); i++) { 4149 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4150 if (thread == 0) { 4151 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4152 outputTracks[i].get()); 4153 return false; 4154 } 4155 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4156 // see note at standby() declaration 4157 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4158 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4159 thread.get()); 4160 return false; 4161 } 4162 } 4163 return true; 4164} 4165 4166uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4167{ 4168 return (mWaitTimeMs * 1000) / 2; 4169} 4170 4171void AudioFlinger::DuplicatingThread::cacheParameters_l() 4172{ 4173 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4174 updateWaitTime_l(); 4175 4176 MixerThread::cacheParameters_l(); 4177} 4178 4179// ---------------------------------------------------------------------------- 4180 4181// TrackBase constructor must be called with AudioFlinger::mLock held 4182AudioFlinger::ThreadBase::TrackBase::TrackBase( 4183 ThreadBase *thread, 4184 const sp<Client>& client, 4185 uint32_t sampleRate, 4186 audio_format_t format, 4187 audio_channel_mask_t channelMask, 4188 int frameCount, 4189 const sp<IMemory>& sharedBuffer, 4190 int sessionId) 4191 : RefBase(), 4192 mThread(thread), 4193 mClient(client), 4194 mCblk(NULL), 4195 // mBuffer 4196 // mBufferEnd 4197 mStepCount(0), 4198 mState(IDLE), 4199 mSampleRate(sampleRate), 4200 mFormat(format), 4201 mChannelMask(channelMask), 4202 mChannelCount(popcount(channelMask)), 4203 mFrameSize(audio_is_linear_pcm(format) ? 4204 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 4205 mStepServerFailed(false), 4206 mSessionId(sessionId) 4207{ 4208 // client == 0 implies sharedBuffer == 0 4209 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4210 4211 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4212 sharedBuffer->size()); 4213 4214 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4215 size_t size = sizeof(audio_track_cblk_t); 4216 size_t bufferSize = frameCount * mFrameSize; 4217 if (sharedBuffer == 0) { 4218 size += bufferSize; 4219 } 4220 4221 if (client != 0) { 4222 mCblkMemory = client->heap()->allocate(size); 4223 if (mCblkMemory != 0) { 4224 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4225 // can't assume mCblk != NULL 4226 } else { 4227 ALOGE("not enough memory for AudioTrack size=%u", size); 4228 client->heap()->dump("AudioTrack"); 4229 return; 4230 } 4231 } else { 4232 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4233 // assume mCblk != NULL 4234 } 4235 4236 // construct the shared structure in-place. 4237 if (mCblk != NULL) { 4238 new(mCblk) audio_track_cblk_t(); 4239 // clear all buffers 4240 mCblk->frameCount = frameCount; 4241 mCblk->sampleRate = sampleRate; 4242// uncomment the following lines to quickly test 32-bit wraparound 4243// mCblk->user = 0xffff0000; 4244// mCblk->server = 0xffff0000; 4245// mCblk->userBase = 0xffff0000; 4246// mCblk->serverBase = 0xffff0000; 4247 if (sharedBuffer == 0) { 4248 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4249 memset(mBuffer, 0, bufferSize); 4250 // Force underrun condition to avoid false underrun callback until first data is 4251 // written to buffer (other flags are cleared) 4252 mCblk->flags = CBLK_UNDERRUN; 4253 } else { 4254 mBuffer = sharedBuffer->pointer(); 4255 } 4256 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4257 } 4258} 4259 4260AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4261{ 4262 if (mCblk != NULL) { 4263 if (mClient == 0) { 4264 delete mCblk; 4265 } else { 4266 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4267 } 4268 } 4269 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4270 if (mClient != 0) { 4271 // Client destructor must run with AudioFlinger mutex locked 4272 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4273 // If the client's reference count drops to zero, the associated destructor 4274 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4275 // relying on the automatic clear() at end of scope. 4276 mClient.clear(); 4277 } 4278} 4279 4280// AudioBufferProvider interface 4281// getNextBuffer() = 0; 4282// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4283void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4284{ 4285 buffer->raw = NULL; 4286 mStepCount = buffer->frameCount; 4287 // FIXME See note at getNextBuffer() 4288 (void) step(); // ignore return value of step() 4289 buffer->frameCount = 0; 4290} 4291 4292bool AudioFlinger::ThreadBase::TrackBase::step() { 4293 bool result; 4294 audio_track_cblk_t* cblk = this->cblk(); 4295 4296 result = cblk->stepServer(mStepCount, isOut()); 4297 if (!result) { 4298 ALOGV("stepServer failed acquiring cblk mutex"); 4299 mStepServerFailed = true; 4300 } 4301 return result; 4302} 4303 4304void AudioFlinger::ThreadBase::TrackBase::reset() { 4305 audio_track_cblk_t* cblk = this->cblk(); 4306 4307 cblk->user = 0; 4308 cblk->server = 0; 4309 cblk->userBase = 0; 4310 cblk->serverBase = 0; 4311 mStepServerFailed = false; 4312 ALOGV("TrackBase::reset"); 4313} 4314 4315uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4316 return mCblk->sampleRate; 4317} 4318 4319void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4320 audio_track_cblk_t* cblk = this->cblk(); 4321 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; 4322 int8_t *bufferEnd = bufferStart + frames * mFrameSize; 4323 4324 // Check validity of returned pointer in case the track control block would have been corrupted. 4325 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4326 "TrackBase::getBuffer buffer out of range:\n" 4327 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4328 " server %u, serverBase %u, user %u, userBase %u, frameSize %u", 4329 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4330 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); 4331 4332 return bufferStart; 4333} 4334 4335status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4336{ 4337 mSyncEvents.add(event); 4338 return NO_ERROR; 4339} 4340 4341// ---------------------------------------------------------------------------- 4342 4343// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4344AudioFlinger::PlaybackThread::Track::Track( 4345 PlaybackThread *thread, 4346 const sp<Client>& client, 4347 audio_stream_type_t streamType, 4348 uint32_t sampleRate, 4349 audio_format_t format, 4350 audio_channel_mask_t channelMask, 4351 int frameCount, 4352 const sp<IMemory>& sharedBuffer, 4353 int sessionId, 4354 IAudioFlinger::track_flags_t flags) 4355 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4356 sessionId), 4357 mMute(false), 4358 mFillingUpStatus(FS_INVALID), 4359 // mRetryCount initialized later when needed 4360 mSharedBuffer(sharedBuffer), 4361 mStreamType(streamType), 4362 mName(-1), // see note below 4363 mMainBuffer(thread->mixBuffer()), 4364 mAuxBuffer(NULL), 4365 mAuxEffectId(0), mHasVolumeController(false), 4366 mPresentationCompleteFrames(0), 4367 mFlags(flags), 4368 mFastIndex(-1), 4369 mUnderrunCount(0), 4370 mCachedVolume(1.0) 4371{ 4372 if (mCblk != NULL) { 4373 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4374 mName = thread->getTrackName_l(channelMask, sessionId); 4375 mCblk->mName = mName; 4376 if (mName < 0) { 4377 ALOGE("no more track names available"); 4378 return; 4379 } 4380 // only allocate a fast track index if we were able to allocate a normal track name 4381 if (flags & IAudioFlinger::TRACK_FAST) { 4382 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4383 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4384 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4385 // FIXME This is too eager. We allocate a fast track index before the 4386 // fast track becomes active. Since fast tracks are a scarce resource, 4387 // this means we are potentially denying other more important fast tracks from 4388 // being created. It would be better to allocate the index dynamically. 4389 mFastIndex = i; 4390 mCblk->mName = i; 4391 // Read the initial underruns because this field is never cleared by the fast mixer 4392 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4393 thread->mFastTrackAvailMask &= ~(1 << i); 4394 } 4395 } 4396 ALOGV("Track constructor name %d, calling pid %d", mName, 4397 IPCThreadState::self()->getCallingPid()); 4398} 4399 4400AudioFlinger::PlaybackThread::Track::~Track() 4401{ 4402 ALOGV("PlaybackThread::Track destructor"); 4403} 4404 4405void AudioFlinger::PlaybackThread::Track::destroy() 4406{ 4407 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4408 // by removing it from mTracks vector, so there is a risk that this Tracks's 4409 // destructor is called. As the destructor needs to lock mLock, 4410 // we must acquire a strong reference on this Track before locking mLock 4411 // here so that the destructor is called only when exiting this function. 4412 // On the other hand, as long as Track::destroy() is only called by 4413 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4414 // this Track with its member mTrack. 4415 sp<Track> keep(this); 4416 { // scope for mLock 4417 sp<ThreadBase> thread = mThread.promote(); 4418 if (thread != 0) { 4419 if (!isOutputTrack()) { 4420 if (mState == ACTIVE || mState == RESUMING) { 4421 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4422 4423#ifdef ADD_BATTERY_DATA 4424 // to track the speaker usage 4425 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4426#endif 4427 } 4428 AudioSystem::releaseOutput(thread->id()); 4429 } 4430 Mutex::Autolock _l(thread->mLock); 4431 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4432 playbackThread->destroyTrack_l(this); 4433 } 4434 } 4435} 4436 4437/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4438{ 4439 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 4440 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4441} 4442 4443void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4444{ 4445 uint32_t vlr = mCblk->getVolumeLR(); 4446 if (isFastTrack()) { 4447 sprintf(buffer, " F %2d", mFastIndex); 4448 } else { 4449 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4450 } 4451 track_state state = mState; 4452 char stateChar; 4453 switch (state) { 4454 case IDLE: 4455 stateChar = 'I'; 4456 break; 4457 case TERMINATED: 4458 stateChar = 'T'; 4459 break; 4460 case STOPPING_1: 4461 stateChar = 's'; 4462 break; 4463 case STOPPING_2: 4464 stateChar = '5'; 4465 break; 4466 case STOPPED: 4467 stateChar = 'S'; 4468 break; 4469 case RESUMING: 4470 stateChar = 'R'; 4471 break; 4472 case ACTIVE: 4473 stateChar = 'A'; 4474 break; 4475 case PAUSING: 4476 stateChar = 'p'; 4477 break; 4478 case PAUSED: 4479 stateChar = 'P'; 4480 break; 4481 case FLUSHED: 4482 stateChar = 'F'; 4483 break; 4484 default: 4485 stateChar = '?'; 4486 break; 4487 } 4488 char nowInUnderrun; 4489 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4490 case UNDERRUN_FULL: 4491 nowInUnderrun = ' '; 4492 break; 4493 case UNDERRUN_PARTIAL: 4494 nowInUnderrun = '<'; 4495 break; 4496 case UNDERRUN_EMPTY: 4497 nowInUnderrun = '*'; 4498 break; 4499 default: 4500 nowInUnderrun = '?'; 4501 break; 4502 } 4503 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4504 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4505 (mClient == 0) ? getpid_cached : mClient->pid(), 4506 mStreamType, 4507 mFormat, 4508 mChannelMask, 4509 mSessionId, 4510 mStepCount, 4511 mCblk->frameCount, 4512 stateChar, 4513 mMute, 4514 mFillingUpStatus, 4515 mCblk->sampleRate, 4516 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4517 20.0 * log10((vlr >> 16) / 4096.0), 4518 mCblk->server, 4519 mCblk->user, 4520 (int)mMainBuffer, 4521 (int)mAuxBuffer, 4522 mCblk->flags, 4523 mUnderrunCount, 4524 nowInUnderrun); 4525} 4526 4527// AudioBufferProvider interface 4528status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4529 AudioBufferProvider::Buffer* buffer, int64_t pts) 4530{ 4531 audio_track_cblk_t* cblk = this->cblk(); 4532 uint32_t framesReady; 4533 uint32_t framesReq = buffer->frameCount; 4534 4535 // Check if last stepServer failed, try to step now 4536 if (mStepServerFailed) { 4537 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4538 // Since the fast mixer is higher priority than client callback thread, 4539 // it does not result in priority inversion for client. 4540 // But a non-blocking solution would be preferable to avoid 4541 // fast mixer being unable to tryLock(), and 4542 // to avoid the extra context switches if the client wakes up, 4543 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4544 if (!step()) goto getNextBuffer_exit; 4545 ALOGV("stepServer recovered"); 4546 mStepServerFailed = false; 4547 } 4548 4549 // FIXME Same as above 4550 framesReady = cblk->framesReadyOut(); 4551 4552 if (CC_LIKELY(framesReady)) { 4553 uint32_t s = cblk->server; 4554 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4555 4556 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4557 if (framesReq > framesReady) { 4558 framesReq = framesReady; 4559 } 4560 if (framesReq > bufferEnd - s) { 4561 framesReq = bufferEnd - s; 4562 } 4563 4564 buffer->raw = getBuffer(s, framesReq); 4565 buffer->frameCount = framesReq; 4566 return NO_ERROR; 4567 } 4568 4569getNextBuffer_exit: 4570 buffer->raw = NULL; 4571 buffer->frameCount = 0; 4572 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4573 return NOT_ENOUGH_DATA; 4574} 4575 4576// Note that framesReady() takes a mutex on the control block using tryLock(). 4577// This could result in priority inversion if framesReady() is called by the normal mixer, 4578// as the normal mixer thread runs at lower 4579// priority than the client's callback thread: there is a short window within framesReady() 4580// during which the normal mixer could be preempted, and the client callback would block. 4581// Another problem can occur if framesReady() is called by the fast mixer: 4582// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4583// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4584size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4585 return mCblk->framesReadyOut(); 4586} 4587 4588// Don't call for fast tracks; the framesReady() could result in priority inversion 4589bool AudioFlinger::PlaybackThread::Track::isReady() const { 4590 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4591 4592 if (framesReady() >= mCblk->frameCount || 4593 (mCblk->flags & CBLK_FORCEREADY)) { 4594 mFillingUpStatus = FS_FILLED; 4595 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4596 return true; 4597 } 4598 return false; 4599} 4600 4601status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4602 int triggerSession) 4603{ 4604 status_t status = NO_ERROR; 4605 ALOGV("start(%d), calling pid %d session %d", 4606 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4607 4608 sp<ThreadBase> thread = mThread.promote(); 4609 if (thread != 0) { 4610 Mutex::Autolock _l(thread->mLock); 4611 track_state state = mState; 4612 // here the track could be either new, or restarted 4613 // in both cases "unstop" the track 4614 if (mState == PAUSED) { 4615 mState = TrackBase::RESUMING; 4616 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4617 } else { 4618 mState = TrackBase::ACTIVE; 4619 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4620 } 4621 4622 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4623 thread->mLock.unlock(); 4624 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4625 thread->mLock.lock(); 4626 4627#ifdef ADD_BATTERY_DATA 4628 // to track the speaker usage 4629 if (status == NO_ERROR) { 4630 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4631 } 4632#endif 4633 } 4634 if (status == NO_ERROR) { 4635 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4636 playbackThread->addTrack_l(this); 4637 } else { 4638 mState = state; 4639 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4640 } 4641 } else { 4642 status = BAD_VALUE; 4643 } 4644 return status; 4645} 4646 4647void AudioFlinger::PlaybackThread::Track::stop() 4648{ 4649 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4650 sp<ThreadBase> thread = mThread.promote(); 4651 if (thread != 0) { 4652 Mutex::Autolock _l(thread->mLock); 4653 track_state state = mState; 4654 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4655 // If the track is not active (PAUSED and buffers full), flush buffers 4656 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4657 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4658 reset(); 4659 mState = STOPPED; 4660 } else if (!isFastTrack()) { 4661 mState = STOPPED; 4662 } else { 4663 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4664 // and then to STOPPED and reset() when presentation is complete 4665 mState = STOPPING_1; 4666 } 4667 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4668 playbackThread); 4669 } 4670 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4671 thread->mLock.unlock(); 4672 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4673 thread->mLock.lock(); 4674 4675#ifdef ADD_BATTERY_DATA 4676 // to track the speaker usage 4677 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4678#endif 4679 } 4680 } 4681} 4682 4683void AudioFlinger::PlaybackThread::Track::pause() 4684{ 4685 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4686 sp<ThreadBase> thread = mThread.promote(); 4687 if (thread != 0) { 4688 Mutex::Autolock _l(thread->mLock); 4689 if (mState == ACTIVE || mState == RESUMING) { 4690 mState = PAUSING; 4691 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4692 if (!isOutputTrack()) { 4693 thread->mLock.unlock(); 4694 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4695 thread->mLock.lock(); 4696 4697#ifdef ADD_BATTERY_DATA 4698 // to track the speaker usage 4699 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4700#endif 4701 } 4702 } 4703 } 4704} 4705 4706void AudioFlinger::PlaybackThread::Track::flush() 4707{ 4708 ALOGV("flush(%d)", mName); 4709 sp<ThreadBase> thread = mThread.promote(); 4710 if (thread != 0) { 4711 Mutex::Autolock _l(thread->mLock); 4712 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4713 mState != PAUSING) { 4714 return; 4715 } 4716 // No point remaining in PAUSED state after a flush => go to 4717 // FLUSHED state 4718 mState = FLUSHED; 4719 // do not reset the track if it is still in the process of being stopped or paused. 4720 // this will be done by prepareTracks_l() when the track is stopped. 4721 // prepareTracks_l() will see mState == FLUSHED, then 4722 // remove from active track list, reset(), and trigger presentation complete 4723 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4724 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4725 reset(); 4726 } 4727 } 4728} 4729 4730void AudioFlinger::PlaybackThread::Track::reset() 4731{ 4732 // Do not reset twice to avoid discarding data written just after a flush and before 4733 // the audioflinger thread detects the track is stopped. 4734 if (!mResetDone) { 4735 TrackBase::reset(); 4736 // Force underrun condition to avoid false underrun callback until first data is 4737 // written to buffer 4738 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4739 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4740 mFillingUpStatus = FS_FILLING; 4741 mResetDone = true; 4742 if (mState == FLUSHED) { 4743 mState = IDLE; 4744 } 4745 } 4746} 4747 4748void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4749{ 4750 mMute = muted; 4751} 4752 4753status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4754{ 4755 status_t status = DEAD_OBJECT; 4756 sp<ThreadBase> thread = mThread.promote(); 4757 if (thread != 0) { 4758 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4759 sp<AudioFlinger> af = mClient->audioFlinger(); 4760 4761 Mutex::Autolock _l(af->mLock); 4762 4763 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4764 4765 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4766 Mutex::Autolock _dl(playbackThread->mLock); 4767 Mutex::Autolock _sl(srcThread->mLock); 4768 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4769 if (chain == 0) { 4770 return INVALID_OPERATION; 4771 } 4772 4773 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4774 if (effect == 0) { 4775 return INVALID_OPERATION; 4776 } 4777 srcThread->removeEffect_l(effect); 4778 playbackThread->addEffect_l(effect); 4779 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4780 if (effect->state() == EffectModule::ACTIVE || 4781 effect->state() == EffectModule::STOPPING) { 4782 effect->start(); 4783 } 4784 4785 sp<EffectChain> dstChain = effect->chain().promote(); 4786 if (dstChain == 0) { 4787 srcThread->addEffect_l(effect); 4788 return INVALID_OPERATION; 4789 } 4790 AudioSystem::unregisterEffect(effect->id()); 4791 AudioSystem::registerEffect(&effect->desc(), 4792 srcThread->id(), 4793 dstChain->strategy(), 4794 AUDIO_SESSION_OUTPUT_MIX, 4795 effect->id()); 4796 } 4797 status = playbackThread->attachAuxEffect(this, EffectId); 4798 } 4799 return status; 4800} 4801 4802void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4803{ 4804 mAuxEffectId = EffectId; 4805 mAuxBuffer = buffer; 4806} 4807 4808bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4809 size_t audioHalFrames) 4810{ 4811 // a track is considered presented when the total number of frames written to audio HAL 4812 // corresponds to the number of frames written when presentationComplete() is called for the 4813 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4814 if (mPresentationCompleteFrames == 0) { 4815 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4816 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4817 mPresentationCompleteFrames, audioHalFrames); 4818 } 4819 if (framesWritten >= mPresentationCompleteFrames) { 4820 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4821 mSessionId, framesWritten); 4822 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4823 return true; 4824 } 4825 return false; 4826} 4827 4828void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4829{ 4830 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4831 if (mSyncEvents[i]->type() == type) { 4832 mSyncEvents[i]->trigger(); 4833 mSyncEvents.removeAt(i); 4834 i--; 4835 } 4836 } 4837} 4838 4839// implement VolumeBufferProvider interface 4840 4841uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4842{ 4843 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4844 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4845 uint32_t vlr = mCblk->getVolumeLR(); 4846 uint32_t vl = vlr & 0xFFFF; 4847 uint32_t vr = vlr >> 16; 4848 // track volumes come from shared memory, so can't be trusted and must be clamped 4849 if (vl > MAX_GAIN_INT) { 4850 vl = MAX_GAIN_INT; 4851 } 4852 if (vr > MAX_GAIN_INT) { 4853 vr = MAX_GAIN_INT; 4854 } 4855 // now apply the cached master volume and stream type volume; 4856 // this is trusted but lacks any synchronization or barrier so may be stale 4857 float v = mCachedVolume; 4858 vl *= v; 4859 vr *= v; 4860 // re-combine into U4.16 4861 vlr = (vr << 16) | (vl & 0xFFFF); 4862 // FIXME look at mute, pause, and stop flags 4863 return vlr; 4864} 4865 4866status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4867{ 4868 if (mState == TERMINATED || mState == PAUSED || 4869 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4870 (mState == STOPPED)))) { 4871 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4872 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4873 event->cancel(); 4874 return INVALID_OPERATION; 4875 } 4876 (void) TrackBase::setSyncEvent(event); 4877 return NO_ERROR; 4878} 4879 4880bool AudioFlinger::PlaybackThread::Track::isOut() const 4881{ 4882 return true; 4883} 4884 4885// timed audio tracks 4886 4887sp<AudioFlinger::PlaybackThread::TimedTrack> 4888AudioFlinger::PlaybackThread::TimedTrack::create( 4889 PlaybackThread *thread, 4890 const sp<Client>& client, 4891 audio_stream_type_t streamType, 4892 uint32_t sampleRate, 4893 audio_format_t format, 4894 audio_channel_mask_t channelMask, 4895 int frameCount, 4896 const sp<IMemory>& sharedBuffer, 4897 int sessionId) { 4898 if (!client->reserveTimedTrack()) 4899 return 0; 4900 4901 return new TimedTrack( 4902 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4903 sharedBuffer, sessionId); 4904} 4905 4906AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4907 PlaybackThread *thread, 4908 const sp<Client>& client, 4909 audio_stream_type_t streamType, 4910 uint32_t sampleRate, 4911 audio_format_t format, 4912 audio_channel_mask_t channelMask, 4913 int frameCount, 4914 const sp<IMemory>& sharedBuffer, 4915 int sessionId) 4916 : Track(thread, client, streamType, sampleRate, format, channelMask, 4917 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4918 mQueueHeadInFlight(false), 4919 mTrimQueueHeadOnRelease(false), 4920 mFramesPendingInQueue(0), 4921 mTimedSilenceBuffer(NULL), 4922 mTimedSilenceBufferSize(0), 4923 mTimedAudioOutputOnTime(false), 4924 mMediaTimeTransformValid(false) 4925{ 4926 LocalClock lc; 4927 mLocalTimeFreq = lc.getLocalFreq(); 4928 4929 mLocalTimeToSampleTransform.a_zero = 0; 4930 mLocalTimeToSampleTransform.b_zero = 0; 4931 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4932 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4933 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4934 &mLocalTimeToSampleTransform.a_to_b_denom); 4935 4936 mMediaTimeToSampleTransform.a_zero = 0; 4937 mMediaTimeToSampleTransform.b_zero = 0; 4938 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4939 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4940 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4941 &mMediaTimeToSampleTransform.a_to_b_denom); 4942} 4943 4944AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4945 mClient->releaseTimedTrack(); 4946 delete [] mTimedSilenceBuffer; 4947} 4948 4949status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4950 size_t size, sp<IMemory>* buffer) { 4951 4952 Mutex::Autolock _l(mTimedBufferQueueLock); 4953 4954 trimTimedBufferQueue_l(); 4955 4956 // lazily initialize the shared memory heap for timed buffers 4957 if (mTimedMemoryDealer == NULL) { 4958 const int kTimedBufferHeapSize = 512 << 10; 4959 4960 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4961 "AudioFlingerTimed"); 4962 if (mTimedMemoryDealer == NULL) 4963 return NO_MEMORY; 4964 } 4965 4966 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4967 if (newBuffer == NULL) { 4968 newBuffer = mTimedMemoryDealer->allocate(size); 4969 if (newBuffer == NULL) 4970 return NO_MEMORY; 4971 } 4972 4973 *buffer = newBuffer; 4974 return NO_ERROR; 4975} 4976 4977// caller must hold mTimedBufferQueueLock 4978void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4979 int64_t mediaTimeNow; 4980 { 4981 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4982 if (!mMediaTimeTransformValid) 4983 return; 4984 4985 int64_t targetTimeNow; 4986 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4987 ? mCCHelper.getCommonTime(&targetTimeNow) 4988 : mCCHelper.getLocalTime(&targetTimeNow); 4989 4990 if (OK != res) 4991 return; 4992 4993 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4994 &mediaTimeNow)) { 4995 return; 4996 } 4997 } 4998 4999 size_t trimEnd; 5000 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 5001 int64_t bufEnd; 5002 5003 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 5004 // We have a next buffer. Just use its PTS as the PTS of the frame 5005 // following the last frame in this buffer. If the stream is sparse 5006 // (ie, there are deliberate gaps left in the stream which should be 5007 // filled with silence by the TimedAudioTrack), then this can result 5008 // in one extra buffer being left un-trimmed when it could have 5009 // been. In general, this is not typical, and we would rather 5010 // optimized away the TS calculation below for the more common case 5011 // where PTSes are contiguous. 5012 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5013 } else { 5014 // We have no next buffer. Compute the PTS of the frame following 5015 // the last frame in this buffer by computing the duration of of 5016 // this frame in media time units and adding it to the PTS of the 5017 // buffer. 5018 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5019 / mFrameSize; 5020 5021 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5022 &bufEnd)) { 5023 ALOGE("Failed to convert frame count of %lld to media time" 5024 " duration" " (scale factor %d/%u) in %s", 5025 frameCount, 5026 mMediaTimeToSampleTransform.a_to_b_numer, 5027 mMediaTimeToSampleTransform.a_to_b_denom, 5028 __PRETTY_FUNCTION__); 5029 break; 5030 } 5031 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5032 } 5033 5034 if (bufEnd > mediaTimeNow) 5035 break; 5036 5037 // Is the buffer we want to use in the middle of a mix operation right 5038 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5039 // from the mixer which should be coming back shortly. 5040 if (!trimEnd && mQueueHeadInFlight) { 5041 mTrimQueueHeadOnRelease = true; 5042 } 5043 } 5044 5045 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5046 if (trimStart < trimEnd) { 5047 // Update the bookkeeping for framesReady() 5048 for (size_t i = trimStart; i < trimEnd; ++i) { 5049 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5050 } 5051 5052 // Now actually remove the buffers from the queue. 5053 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5054 } 5055} 5056 5057void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5058 const char* logTag) { 5059 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5060 "%s called (reason \"%s\"), but timed buffer queue has no" 5061 " elements to trim.", __FUNCTION__, logTag); 5062 5063 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5064 mTimedBufferQueue.removeAt(0); 5065} 5066 5067void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5068 const TimedBuffer& buf, 5069 const char* logTag) { 5070 uint32_t bufBytes = buf.buffer()->size(); 5071 uint32_t consumedAlready = buf.position(); 5072 5073 ALOG_ASSERT(consumedAlready <= bufBytes, 5074 "Bad bookkeeping while updating frames pending. Timed buffer is" 5075 " only %u bytes long, but claims to have consumed %u" 5076 " bytes. (update reason: \"%s\")", 5077 bufBytes, consumedAlready, logTag); 5078 5079 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 5080 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5081 "Bad bookkeeping while updating frames pending. Should have at" 5082 " least %u queued frames, but we think we have only %u. (update" 5083 " reason: \"%s\")", 5084 bufFrames, mFramesPendingInQueue, logTag); 5085 5086 mFramesPendingInQueue -= bufFrames; 5087} 5088 5089status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5090 const sp<IMemory>& buffer, int64_t pts) { 5091 5092 { 5093 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5094 if (!mMediaTimeTransformValid) 5095 return INVALID_OPERATION; 5096 } 5097 5098 Mutex::Autolock _l(mTimedBufferQueueLock); 5099 5100 uint32_t bufFrames = buffer->size() / mFrameSize; 5101 mFramesPendingInQueue += bufFrames; 5102 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5103 5104 return NO_ERROR; 5105} 5106 5107status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5108 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5109 5110 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5111 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5112 target); 5113 5114 if (!(target == TimedAudioTrack::LOCAL_TIME || 5115 target == TimedAudioTrack::COMMON_TIME)) { 5116 return BAD_VALUE; 5117 } 5118 5119 Mutex::Autolock lock(mMediaTimeTransformLock); 5120 mMediaTimeTransform = xform; 5121 mMediaTimeTransformTarget = target; 5122 mMediaTimeTransformValid = true; 5123 5124 return NO_ERROR; 5125} 5126 5127#define min(a, b) ((a) < (b) ? (a) : (b)) 5128 5129// implementation of getNextBuffer for tracks whose buffers have timestamps 5130status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5131 AudioBufferProvider::Buffer* buffer, int64_t pts) 5132{ 5133 if (pts == AudioBufferProvider::kInvalidPTS) { 5134 buffer->raw = NULL; 5135 buffer->frameCount = 0; 5136 mTimedAudioOutputOnTime = false; 5137 return INVALID_OPERATION; 5138 } 5139 5140 Mutex::Autolock _l(mTimedBufferQueueLock); 5141 5142 ALOG_ASSERT(!mQueueHeadInFlight, 5143 "getNextBuffer called without releaseBuffer!"); 5144 5145 while (true) { 5146 5147 // if we have no timed buffers, then fail 5148 if (mTimedBufferQueue.isEmpty()) { 5149 buffer->raw = NULL; 5150 buffer->frameCount = 0; 5151 return NOT_ENOUGH_DATA; 5152 } 5153 5154 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5155 5156 // calculate the PTS of the head of the timed buffer queue expressed in 5157 // local time 5158 int64_t headLocalPTS; 5159 { 5160 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5161 5162 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5163 5164 if (mMediaTimeTransform.a_to_b_denom == 0) { 5165 // the transform represents a pause, so yield silence 5166 timedYieldSilence_l(buffer->frameCount, buffer); 5167 return NO_ERROR; 5168 } 5169 5170 int64_t transformedPTS; 5171 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5172 &transformedPTS)) { 5173 // the transform failed. this shouldn't happen, but if it does 5174 // then just drop this buffer 5175 ALOGW("timedGetNextBuffer transform failed"); 5176 buffer->raw = NULL; 5177 buffer->frameCount = 0; 5178 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5179 return NO_ERROR; 5180 } 5181 5182 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5183 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5184 &headLocalPTS)) { 5185 buffer->raw = NULL; 5186 buffer->frameCount = 0; 5187 return INVALID_OPERATION; 5188 } 5189 } else { 5190 headLocalPTS = transformedPTS; 5191 } 5192 } 5193 5194 // adjust the head buffer's PTS to reflect the portion of the head buffer 5195 // that has already been consumed 5196 int64_t effectivePTS = headLocalPTS + 5197 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 5198 5199 // Calculate the delta in samples between the head of the input buffer 5200 // queue and the start of the next output buffer that will be written. 5201 // If the transformation fails because of over or underflow, it means 5202 // that the sample's position in the output stream is so far out of 5203 // whack that it should just be dropped. 5204 int64_t sampleDelta; 5205 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5206 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5207 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5208 " mix"); 5209 continue; 5210 } 5211 if (!mLocalTimeToSampleTransform.doForwardTransform( 5212 (effectivePTS - pts) << 32, &sampleDelta)) { 5213 ALOGV("*** too late during sample rate transform: dropped buffer"); 5214 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5215 continue; 5216 } 5217 5218 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5219 " sampleDelta=[%d.%08x]", 5220 head.pts(), head.position(), pts, 5221 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5222 + (sampleDelta >> 32)), 5223 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5224 5225 // if the delta between the ideal placement for the next input sample and 5226 // the current output position is within this threshold, then we will 5227 // concatenate the next input samples to the previous output 5228 const int64_t kSampleContinuityThreshold = 5229 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5230 5231 // if this is the first buffer of audio that we're emitting from this track 5232 // then it should be almost exactly on time. 5233 const int64_t kSampleStartupThreshold = 1LL << 32; 5234 5235 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5236 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5237 // the next input is close enough to being on time, so concatenate it 5238 // with the last output 5239 timedYieldSamples_l(buffer); 5240 5241 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5242 head.position(), buffer->frameCount); 5243 return NO_ERROR; 5244 } 5245 5246 // Looks like our output is not on time. Reset our on timed status. 5247 // Next time we mix samples from our input queue, then should be within 5248 // the StartupThreshold. 5249 mTimedAudioOutputOnTime = false; 5250 if (sampleDelta > 0) { 5251 // the gap between the current output position and the proper start of 5252 // the next input sample is too big, so fill it with silence 5253 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5254 5255 timedYieldSilence_l(framesUntilNextInput, buffer); 5256 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5257 return NO_ERROR; 5258 } else { 5259 // the next input sample is late 5260 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5261 size_t onTimeSamplePosition = 5262 head.position() + lateFrames * mFrameSize; 5263 5264 if (onTimeSamplePosition > head.buffer()->size()) { 5265 // all the remaining samples in the head are too late, so 5266 // drop it and move on 5267 ALOGV("*** too late: dropped buffer"); 5268 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5269 continue; 5270 } else { 5271 // skip over the late samples 5272 head.setPosition(onTimeSamplePosition); 5273 5274 // yield the available samples 5275 timedYieldSamples_l(buffer); 5276 5277 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5278 return NO_ERROR; 5279 } 5280 } 5281 } 5282} 5283 5284// Yield samples from the timed buffer queue head up to the given output 5285// buffer's capacity. 5286// 5287// Caller must hold mTimedBufferQueueLock 5288void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5289 AudioBufferProvider::Buffer* buffer) { 5290 5291 const TimedBuffer& head = mTimedBufferQueue[0]; 5292 5293 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5294 head.position()); 5295 5296 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5297 mFrameSize); 5298 size_t framesRequested = buffer->frameCount; 5299 buffer->frameCount = min(framesLeftInHead, framesRequested); 5300 5301 mQueueHeadInFlight = true; 5302 mTimedAudioOutputOnTime = true; 5303} 5304 5305// Yield samples of silence up to the given output buffer's capacity 5306// 5307// Caller must hold mTimedBufferQueueLock 5308void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5309 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5310 5311 // lazily allocate a buffer filled with silence 5312 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 5313 delete [] mTimedSilenceBuffer; 5314 mTimedSilenceBufferSize = numFrames * mFrameSize; 5315 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5316 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5317 } 5318 5319 buffer->raw = mTimedSilenceBuffer; 5320 size_t framesRequested = buffer->frameCount; 5321 buffer->frameCount = min(numFrames, framesRequested); 5322 5323 mTimedAudioOutputOnTime = false; 5324} 5325 5326// AudioBufferProvider interface 5327void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5328 AudioBufferProvider::Buffer* buffer) { 5329 5330 Mutex::Autolock _l(mTimedBufferQueueLock); 5331 5332 // If the buffer which was just released is part of the buffer at the head 5333 // of the queue, be sure to update the amt of the buffer which has been 5334 // consumed. If the buffer being returned is not part of the head of the 5335 // queue, its either because the buffer is part of the silence buffer, or 5336 // because the head of the timed queue was trimmed after the mixer called 5337 // getNextBuffer but before the mixer called releaseBuffer. 5338 if (buffer->raw == mTimedSilenceBuffer) { 5339 ALOG_ASSERT(!mQueueHeadInFlight, 5340 "Queue head in flight during release of silence buffer!"); 5341 goto done; 5342 } 5343 5344 ALOG_ASSERT(mQueueHeadInFlight, 5345 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5346 " head in flight."); 5347 5348 if (mTimedBufferQueue.size()) { 5349 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5350 5351 void* start = head.buffer()->pointer(); 5352 void* end = reinterpret_cast<void*>( 5353 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5354 + head.buffer()->size()); 5355 5356 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5357 "released buffer not within the head of the timed buffer" 5358 " queue; qHead = [%p, %p], released buffer = %p", 5359 start, end, buffer->raw); 5360 5361 head.setPosition(head.position() + 5362 (buffer->frameCount * mFrameSize)); 5363 mQueueHeadInFlight = false; 5364 5365 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5366 "Bad bookkeeping during releaseBuffer! Should have at" 5367 " least %u queued frames, but we think we have only %u", 5368 buffer->frameCount, mFramesPendingInQueue); 5369 5370 mFramesPendingInQueue -= buffer->frameCount; 5371 5372 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5373 || mTrimQueueHeadOnRelease) { 5374 trimTimedBufferQueueHead_l("releaseBuffer"); 5375 mTrimQueueHeadOnRelease = false; 5376 } 5377 } else { 5378 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5379 " buffers in the timed buffer queue"); 5380 } 5381 5382done: 5383 buffer->raw = 0; 5384 buffer->frameCount = 0; 5385} 5386 5387size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5388 Mutex::Autolock _l(mTimedBufferQueueLock); 5389 return mFramesPendingInQueue; 5390} 5391 5392AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5393 : mPTS(0), mPosition(0) {} 5394 5395AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5396 const sp<IMemory>& buffer, int64_t pts) 5397 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5398 5399// ---------------------------------------------------------------------------- 5400 5401// RecordTrack constructor must be called with AudioFlinger::mLock held 5402AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5403 RecordThread *thread, 5404 const sp<Client>& client, 5405 uint32_t sampleRate, 5406 audio_format_t format, 5407 audio_channel_mask_t channelMask, 5408 int frameCount, 5409 int sessionId) 5410 : TrackBase(thread, client, sampleRate, format, 5411 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5412 mOverflow(false) 5413{ 5414 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5415} 5416 5417AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5418{ 5419 ALOGV("%s", __func__); 5420} 5421 5422// AudioBufferProvider interface 5423status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5424 int64_t pts) 5425{ 5426 audio_track_cblk_t* cblk = this->cblk(); 5427 uint32_t framesAvail; 5428 uint32_t framesReq = buffer->frameCount; 5429 5430 // Check if last stepServer failed, try to step now 5431 if (mStepServerFailed) { 5432 if (!step()) goto getNextBuffer_exit; 5433 ALOGV("stepServer recovered"); 5434 mStepServerFailed = false; 5435 } 5436 5437 // FIXME lock is not actually held, so overrun is possible 5438 framesAvail = cblk->framesAvailableIn_l(); 5439 5440 if (CC_LIKELY(framesAvail)) { 5441 uint32_t s = cblk->server; 5442 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5443 5444 if (framesReq > framesAvail) { 5445 framesReq = framesAvail; 5446 } 5447 if (framesReq > bufferEnd - s) { 5448 framesReq = bufferEnd - s; 5449 } 5450 5451 buffer->raw = getBuffer(s, framesReq); 5452 buffer->frameCount = framesReq; 5453 return NO_ERROR; 5454 } 5455 5456getNextBuffer_exit: 5457 buffer->raw = NULL; 5458 buffer->frameCount = 0; 5459 return NOT_ENOUGH_DATA; 5460} 5461 5462status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5463 int triggerSession) 5464{ 5465 sp<ThreadBase> thread = mThread.promote(); 5466 if (thread != 0) { 5467 RecordThread *recordThread = (RecordThread *)thread.get(); 5468 return recordThread->start(this, event, triggerSession); 5469 } else { 5470 return BAD_VALUE; 5471 } 5472} 5473 5474void AudioFlinger::RecordThread::RecordTrack::stop() 5475{ 5476 sp<ThreadBase> thread = mThread.promote(); 5477 if (thread != 0) { 5478 RecordThread *recordThread = (RecordThread *)thread.get(); 5479 recordThread->mLock.lock(); 5480 bool doStop = recordThread->stop_l(this); 5481 if (doStop) { 5482 TrackBase::reset(); 5483 // Force overrun condition to avoid false overrun callback until first data is 5484 // read from buffer 5485 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5486 } 5487 recordThread->mLock.unlock(); 5488 if (doStop) { 5489 AudioSystem::stopInput(recordThread->id()); 5490 } 5491 } 5492} 5493 5494/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5495{ 5496 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 5497} 5498 5499void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5500{ 5501 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5502 (mClient == 0) ? getpid_cached : mClient->pid(), 5503 mFormat, 5504 mChannelMask, 5505 mSessionId, 5506 mStepCount, 5507 mState, 5508 mCblk->sampleRate, 5509 mCblk->server, 5510 mCblk->user, 5511 mCblk->frameCount); 5512} 5513 5514bool AudioFlinger::RecordThread::RecordTrack::isOut() const 5515{ 5516 return false; 5517} 5518 5519// ---------------------------------------------------------------------------- 5520 5521AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5522 PlaybackThread *playbackThread, 5523 DuplicatingThread *sourceThread, 5524 uint32_t sampleRate, 5525 audio_format_t format, 5526 audio_channel_mask_t channelMask, 5527 int frameCount) 5528 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5529 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5530 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 5531{ 5532 5533 if (mCblk != NULL) { 5534 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5535 mOutBuffer.frameCount = 0; 5536 playbackThread->mTracks.add(this); 5537 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5538 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 5539 mCblk, mBuffer, mCblk->buffers, 5540 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5541 } else { 5542 ALOGW("Error creating output track on thread %p", playbackThread); 5543 } 5544} 5545 5546AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5547{ 5548 clearBufferQueue(); 5549} 5550 5551status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5552 int triggerSession) 5553{ 5554 status_t status = Track::start(event, triggerSession); 5555 if (status != NO_ERROR) { 5556 return status; 5557 } 5558 5559 mActive = true; 5560 mRetryCount = 127; 5561 return status; 5562} 5563 5564void AudioFlinger::PlaybackThread::OutputTrack::stop() 5565{ 5566 Track::stop(); 5567 clearBufferQueue(); 5568 mOutBuffer.frameCount = 0; 5569 mActive = false; 5570} 5571 5572bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5573{ 5574 Buffer *pInBuffer; 5575 Buffer inBuffer; 5576 uint32_t channelCount = mChannelCount; 5577 bool outputBufferFull = false; 5578 inBuffer.frameCount = frames; 5579 inBuffer.i16 = data; 5580 5581 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5582 5583 if (!mActive && frames != 0) { 5584 start(); 5585 sp<ThreadBase> thread = mThread.promote(); 5586 if (thread != 0) { 5587 MixerThread *mixerThread = (MixerThread *)thread.get(); 5588 if (mCblk->frameCount > frames){ 5589 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5590 uint32_t startFrames = (mCblk->frameCount - frames); 5591 pInBuffer = new Buffer; 5592 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5593 pInBuffer->frameCount = startFrames; 5594 pInBuffer->i16 = pInBuffer->mBuffer; 5595 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5596 mBufferQueue.add(pInBuffer); 5597 } else { 5598 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5599 } 5600 } 5601 } 5602 } 5603 5604 while (waitTimeLeftMs) { 5605 // First write pending buffers, then new data 5606 if (mBufferQueue.size()) { 5607 pInBuffer = mBufferQueue.itemAt(0); 5608 } else { 5609 pInBuffer = &inBuffer; 5610 } 5611 5612 if (pInBuffer->frameCount == 0) { 5613 break; 5614 } 5615 5616 if (mOutBuffer.frameCount == 0) { 5617 mOutBuffer.frameCount = pInBuffer->frameCount; 5618 nsecs_t startTime = systemTime(); 5619 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5620 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5621 mThread.unsafe_get()); 5622 outputBufferFull = true; 5623 break; 5624 } 5625 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5626 if (waitTimeLeftMs >= waitTimeMs) { 5627 waitTimeLeftMs -= waitTimeMs; 5628 } else { 5629 waitTimeLeftMs = 0; 5630 } 5631 } 5632 5633 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5634 pInBuffer->frameCount; 5635 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5636 mCblk->stepUserOut(outFrames); 5637 pInBuffer->frameCount -= outFrames; 5638 pInBuffer->i16 += outFrames * channelCount; 5639 mOutBuffer.frameCount -= outFrames; 5640 mOutBuffer.i16 += outFrames * channelCount; 5641 5642 if (pInBuffer->frameCount == 0) { 5643 if (mBufferQueue.size()) { 5644 mBufferQueue.removeAt(0); 5645 delete [] pInBuffer->mBuffer; 5646 delete pInBuffer; 5647 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5648 mThread.unsafe_get(), mBufferQueue.size()); 5649 } else { 5650 break; 5651 } 5652 } 5653 } 5654 5655 // If we could not write all frames, allocate a buffer and queue it for next time. 5656 if (inBuffer.frameCount) { 5657 sp<ThreadBase> thread = mThread.promote(); 5658 if (thread != 0 && !thread->standby()) { 5659 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5660 pInBuffer = new Buffer; 5661 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5662 pInBuffer->frameCount = inBuffer.frameCount; 5663 pInBuffer->i16 = pInBuffer->mBuffer; 5664 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5665 sizeof(int16_t)); 5666 mBufferQueue.add(pInBuffer); 5667 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5668 mThread.unsafe_get(), mBufferQueue.size()); 5669 } else { 5670 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5671 mThread.unsafe_get(), this); 5672 } 5673 } 5674 } 5675 5676 // Calling write() with a 0 length buffer, means that no more data will be written: 5677 // If no more buffers are pending, fill output track buffer to make sure it is started 5678 // by output mixer. 5679 if (frames == 0 && mBufferQueue.size() == 0) { 5680 if (mCblk->user < mCblk->frameCount) { 5681 frames = mCblk->frameCount - mCblk->user; 5682 pInBuffer = new Buffer; 5683 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5684 pInBuffer->frameCount = frames; 5685 pInBuffer->i16 = pInBuffer->mBuffer; 5686 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5687 mBufferQueue.add(pInBuffer); 5688 } else if (mActive) { 5689 stop(); 5690 } 5691 } 5692 5693 return outputBufferFull; 5694} 5695 5696status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5697 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5698{ 5699 int active; 5700 status_t result; 5701 audio_track_cblk_t* cblk = mCblk; 5702 uint32_t framesReq = buffer->frameCount; 5703 5704 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5705 buffer->frameCount = 0; 5706 5707 uint32_t framesAvail = cblk->framesAvailableOut(); 5708 5709 5710 if (framesAvail == 0) { 5711 Mutex::Autolock _l(cblk->lock); 5712 goto start_loop_here; 5713 while (framesAvail == 0) { 5714 active = mActive; 5715 if (CC_UNLIKELY(!active)) { 5716 ALOGV("Not active and NO_MORE_BUFFERS"); 5717 return NO_MORE_BUFFERS; 5718 } 5719 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5720 if (result != NO_ERROR) { 5721 return NO_MORE_BUFFERS; 5722 } 5723 // read the server count again 5724 start_loop_here: 5725 framesAvail = cblk->framesAvailableOut_l(); 5726 } 5727 } 5728 5729// if (framesAvail < framesReq) { 5730// return NO_MORE_BUFFERS; 5731// } 5732 5733 if (framesReq > framesAvail) { 5734 framesReq = framesAvail; 5735 } 5736 5737 uint32_t u = cblk->user; 5738 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5739 5740 if (framesReq > bufferEnd - u) { 5741 framesReq = bufferEnd - u; 5742 } 5743 5744 buffer->frameCount = framesReq; 5745 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 5746 return NO_ERROR; 5747} 5748 5749 5750void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5751{ 5752 size_t size = mBufferQueue.size(); 5753 5754 for (size_t i = 0; i < size; i++) { 5755 Buffer *pBuffer = mBufferQueue.itemAt(i); 5756 delete [] pBuffer->mBuffer; 5757 delete pBuffer; 5758 } 5759 mBufferQueue.clear(); 5760} 5761 5762// ---------------------------------------------------------------------------- 5763 5764AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5765 : RefBase(), 5766 mAudioFlinger(audioFlinger), 5767 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5768 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5769 mPid(pid), 5770 mTimedTrackCount(0) 5771{ 5772 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5773} 5774 5775// Client destructor must be called with AudioFlinger::mLock held 5776AudioFlinger::Client::~Client() 5777{ 5778 mAudioFlinger->removeClient_l(mPid); 5779} 5780 5781sp<MemoryDealer> AudioFlinger::Client::heap() const 5782{ 5783 return mMemoryDealer; 5784} 5785 5786// Reserve one of the limited slots for a timed audio track associated 5787// with this client 5788bool AudioFlinger::Client::reserveTimedTrack() 5789{ 5790 const int kMaxTimedTracksPerClient = 4; 5791 5792 Mutex::Autolock _l(mTimedTrackLock); 5793 5794 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5795 ALOGW("can not create timed track - pid %d has exceeded the limit", 5796 mPid); 5797 return false; 5798 } 5799 5800 mTimedTrackCount++; 5801 return true; 5802} 5803 5804// Release a slot for a timed audio track 5805void AudioFlinger::Client::releaseTimedTrack() 5806{ 5807 Mutex::Autolock _l(mTimedTrackLock); 5808 mTimedTrackCount--; 5809} 5810 5811// ---------------------------------------------------------------------------- 5812 5813AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5814 const sp<IAudioFlingerClient>& client, 5815 pid_t pid) 5816 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5817{ 5818} 5819 5820AudioFlinger::NotificationClient::~NotificationClient() 5821{ 5822} 5823 5824void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5825{ 5826 sp<NotificationClient> keep(this); 5827 mAudioFlinger->removeNotificationClient(mPid); 5828} 5829 5830// ---------------------------------------------------------------------------- 5831 5832AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5833 : BnAudioTrack(), 5834 mTrack(track) 5835{ 5836} 5837 5838AudioFlinger::TrackHandle::~TrackHandle() { 5839 // just stop the track on deletion, associated resources 5840 // will be freed from the main thread once all pending buffers have 5841 // been played. Unless it's not in the active track list, in which 5842 // case we free everything now... 5843 mTrack->destroy(); 5844} 5845 5846sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5847 return mTrack->getCblk(); 5848} 5849 5850status_t AudioFlinger::TrackHandle::start() { 5851 return mTrack->start(); 5852} 5853 5854void AudioFlinger::TrackHandle::stop() { 5855 mTrack->stop(); 5856} 5857 5858void AudioFlinger::TrackHandle::flush() { 5859 mTrack->flush(); 5860} 5861 5862void AudioFlinger::TrackHandle::mute(bool e) { 5863 mTrack->mute(e); 5864} 5865 5866void AudioFlinger::TrackHandle::pause() { 5867 mTrack->pause(); 5868} 5869 5870status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5871{ 5872 return mTrack->attachAuxEffect(EffectId); 5873} 5874 5875status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5876 sp<IMemory>* buffer) { 5877 if (!mTrack->isTimedTrack()) 5878 return INVALID_OPERATION; 5879 5880 PlaybackThread::TimedTrack* tt = 5881 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5882 return tt->allocateTimedBuffer(size, buffer); 5883} 5884 5885status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5886 int64_t pts) { 5887 if (!mTrack->isTimedTrack()) 5888 return INVALID_OPERATION; 5889 5890 PlaybackThread::TimedTrack* tt = 5891 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5892 return tt->queueTimedBuffer(buffer, pts); 5893} 5894 5895status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5896 const LinearTransform& xform, int target) { 5897 5898 if (!mTrack->isTimedTrack()) 5899 return INVALID_OPERATION; 5900 5901 PlaybackThread::TimedTrack* tt = 5902 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5903 return tt->setMediaTimeTransform( 5904 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5905} 5906 5907status_t AudioFlinger::TrackHandle::onTransact( 5908 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5909{ 5910 return BnAudioTrack::onTransact(code, data, reply, flags); 5911} 5912 5913// ---------------------------------------------------------------------------- 5914 5915sp<IAudioRecord> AudioFlinger::openRecord( 5916 pid_t pid, 5917 audio_io_handle_t input, 5918 uint32_t sampleRate, 5919 audio_format_t format, 5920 audio_channel_mask_t channelMask, 5921 int frameCount, 5922 IAudioFlinger::track_flags_t flags, 5923 pid_t tid, 5924 int *sessionId, 5925 status_t *status) 5926{ 5927 sp<RecordThread::RecordTrack> recordTrack; 5928 sp<RecordHandle> recordHandle; 5929 sp<Client> client; 5930 status_t lStatus; 5931 RecordThread *thread; 5932 size_t inFrameCount; 5933 int lSessionId; 5934 5935 // check calling permissions 5936 if (!recordingAllowed()) { 5937 lStatus = PERMISSION_DENIED; 5938 goto Exit; 5939 } 5940 5941 // add client to list 5942 { // scope for mLock 5943 Mutex::Autolock _l(mLock); 5944 thread = checkRecordThread_l(input); 5945 if (thread == NULL) { 5946 lStatus = BAD_VALUE; 5947 goto Exit; 5948 } 5949 5950 client = registerPid_l(pid); 5951 5952 // If no audio session id is provided, create one here 5953 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5954 lSessionId = *sessionId; 5955 } else { 5956 lSessionId = nextUniqueId(); 5957 if (sessionId != NULL) { 5958 *sessionId = lSessionId; 5959 } 5960 } 5961 // create new record track. 5962 // The record track uses one track in mHardwareMixerThread by convention. 5963 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5964 frameCount, lSessionId, flags, tid, &lStatus); 5965 } 5966 if (lStatus != NO_ERROR) { 5967 // remove local strong reference to Client before deleting the RecordTrack so that the 5968 // Client destructor is called by the TrackBase destructor with mLock held 5969 client.clear(); 5970 recordTrack.clear(); 5971 goto Exit; 5972 } 5973 5974 // return to handle to client 5975 recordHandle = new RecordHandle(recordTrack); 5976 lStatus = NO_ERROR; 5977 5978Exit: 5979 if (status) { 5980 *status = lStatus; 5981 } 5982 return recordHandle; 5983} 5984 5985// ---------------------------------------------------------------------------- 5986 5987AudioFlinger::RecordHandle::RecordHandle( 5988 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5989 : BnAudioRecord(), 5990 mRecordTrack(recordTrack) 5991{ 5992} 5993 5994AudioFlinger::RecordHandle::~RecordHandle() { 5995 stop_nonvirtual(); 5996 mRecordTrack->destroy(); 5997} 5998 5999sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 6000 return mRecordTrack->getCblk(); 6001} 6002 6003status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6004 int triggerSession) { 6005 ALOGV("RecordHandle::start()"); 6006 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6007} 6008 6009void AudioFlinger::RecordHandle::stop() { 6010 stop_nonvirtual(); 6011} 6012 6013void AudioFlinger::RecordHandle::stop_nonvirtual() { 6014 ALOGV("RecordHandle::stop()"); 6015 mRecordTrack->stop(); 6016} 6017 6018status_t AudioFlinger::RecordHandle::onTransact( 6019 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6020{ 6021 return BnAudioRecord::onTransact(code, data, reply, flags); 6022} 6023 6024// ---------------------------------------------------------------------------- 6025 6026AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6027 AudioStreamIn *input, 6028 uint32_t sampleRate, 6029 audio_channel_mask_t channelMask, 6030 audio_io_handle_t id, 6031 audio_devices_t device, 6032 const sp<NBAIO_Sink>& teeSink) : 6033 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6034 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6035 // mRsmpInIndex and mInputBytes set by readInputParameters() 6036 mReqChannelCount(popcount(channelMask)), 6037 mReqSampleRate(sampleRate), 6038 // mBytesRead is only meaningful while active, and so is cleared in start() 6039 // (but might be better to also clear here for dump?) 6040 mTeeSink(teeSink) 6041{ 6042 snprintf(mName, kNameLength, "AudioIn_%X", id); 6043 6044 readInputParameters(); 6045 6046} 6047 6048 6049AudioFlinger::RecordThread::~RecordThread() 6050{ 6051 delete[] mRsmpInBuffer; 6052 delete mResampler; 6053 delete[] mRsmpOutBuffer; 6054} 6055 6056void AudioFlinger::RecordThread::onFirstRef() 6057{ 6058 run(mName, PRIORITY_URGENT_AUDIO); 6059} 6060 6061status_t AudioFlinger::RecordThread::readyToRun() 6062{ 6063 status_t status = initCheck(); 6064 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6065 return status; 6066} 6067 6068bool AudioFlinger::RecordThread::threadLoop() 6069{ 6070 AudioBufferProvider::Buffer buffer; 6071 sp<RecordTrack> activeTrack; 6072 Vector< sp<EffectChain> > effectChains; 6073 6074 nsecs_t lastWarning = 0; 6075 6076 inputStandBy(); 6077 acquireWakeLock(); 6078 6079 // used to verify we've read at least once before evaluating how many bytes were read 6080 bool readOnce = false; 6081 6082 // start recording 6083 while (!exitPending()) { 6084 6085 processConfigEvents(); 6086 6087 { // scope for mLock 6088 Mutex::Autolock _l(mLock); 6089 checkForNewParameters_l(); 6090 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6091 standby(); 6092 6093 if (exitPending()) break; 6094 6095 releaseWakeLock_l(); 6096 ALOGV("RecordThread: loop stopping"); 6097 // go to sleep 6098 mWaitWorkCV.wait(mLock); 6099 ALOGV("RecordThread: loop starting"); 6100 acquireWakeLock_l(); 6101 continue; 6102 } 6103 if (mActiveTrack != 0) { 6104 if (mActiveTrack->mState == TrackBase::PAUSING) { 6105 standby(); 6106 mActiveTrack.clear(); 6107 mStartStopCond.broadcast(); 6108 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6109 if (mReqChannelCount != mActiveTrack->channelCount()) { 6110 mActiveTrack.clear(); 6111 mStartStopCond.broadcast(); 6112 } else if (readOnce) { 6113 // record start succeeds only if first read from audio input 6114 // succeeds 6115 if (mBytesRead >= 0) { 6116 mActiveTrack->mState = TrackBase::ACTIVE; 6117 } else { 6118 mActiveTrack.clear(); 6119 } 6120 mStartStopCond.broadcast(); 6121 } 6122 mStandby = false; 6123 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6124 removeTrack_l(mActiveTrack); 6125 mActiveTrack.clear(); 6126 } 6127 } 6128 lockEffectChains_l(effectChains); 6129 } 6130 6131 if (mActiveTrack != 0) { 6132 if (mActiveTrack->mState != TrackBase::ACTIVE && 6133 mActiveTrack->mState != TrackBase::RESUMING) { 6134 unlockEffectChains(effectChains); 6135 usleep(kRecordThreadSleepUs); 6136 continue; 6137 } 6138 for (size_t i = 0; i < effectChains.size(); i ++) { 6139 effectChains[i]->process_l(); 6140 } 6141 6142 buffer.frameCount = mFrameCount; 6143 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6144 readOnce = true; 6145 size_t framesOut = buffer.frameCount; 6146 if (mResampler == NULL) { 6147 // no resampling 6148 while (framesOut) { 6149 size_t framesIn = mFrameCount - mRsmpInIndex; 6150 if (framesIn) { 6151 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6152 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6153 mActiveTrack->mFrameSize; 6154 if (framesIn > framesOut) 6155 framesIn = framesOut; 6156 mRsmpInIndex += framesIn; 6157 framesOut -= framesIn; 6158 if ((int)mChannelCount == mReqChannelCount || 6159 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6160 memcpy(dst, src, framesIn * mFrameSize); 6161 } else { 6162 if (mChannelCount == 1) { 6163 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6164 (int16_t *)src, framesIn); 6165 } else { 6166 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6167 (int16_t *)src, framesIn); 6168 } 6169 } 6170 } 6171 if (framesOut && mFrameCount == mRsmpInIndex) { 6172 void *readInto; 6173 if (framesOut == mFrameCount && 6174 ((int)mChannelCount == mReqChannelCount || 6175 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6176 readInto = buffer.raw; 6177 framesOut = 0; 6178 } else { 6179 readInto = mRsmpInBuffer; 6180 mRsmpInIndex = 0; 6181 } 6182 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6183 if (mBytesRead <= 0) { 6184 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6185 { 6186 ALOGE("Error reading audio input"); 6187 // Force input into standby so that it tries to 6188 // recover at next read attempt 6189 inputStandBy(); 6190 usleep(kRecordThreadSleepUs); 6191 } 6192 mRsmpInIndex = mFrameCount; 6193 framesOut = 0; 6194 buffer.frameCount = 0; 6195 } else if (mTeeSink != 0) { 6196 (void) mTeeSink->write(readInto, 6197 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6198 } 6199 } 6200 } 6201 } else { 6202 // resampling 6203 6204 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6205 // alter output frame count as if we were expecting stereo samples 6206 if (mChannelCount == 1 && mReqChannelCount == 1) { 6207 framesOut >>= 1; 6208 } 6209 mResampler->resample(mRsmpOutBuffer, framesOut, 6210 this /* AudioBufferProvider* */); 6211 // ditherAndClamp() works as long as all buffers returned by 6212 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6213 if (mChannelCount == 2 && mReqChannelCount == 1) { 6214 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6215 // the resampler always outputs stereo samples: 6216 // do post stereo to mono conversion 6217 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6218 framesOut); 6219 } else { 6220 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6221 } 6222 6223 } 6224 if (mFramestoDrop == 0) { 6225 mActiveTrack->releaseBuffer(&buffer); 6226 } else { 6227 if (mFramestoDrop > 0) { 6228 mFramestoDrop -= buffer.frameCount; 6229 if (mFramestoDrop <= 0) { 6230 clearSyncStartEvent(); 6231 } 6232 } else { 6233 mFramestoDrop += buffer.frameCount; 6234 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6235 mSyncStartEvent->isCancelled()) { 6236 ALOGW("Synced record %s, session %d, trigger session %d", 6237 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6238 mActiveTrack->sessionId(), 6239 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6240 clearSyncStartEvent(); 6241 } 6242 } 6243 } 6244 mActiveTrack->clearOverflow(); 6245 } 6246 // client isn't retrieving buffers fast enough 6247 else { 6248 if (!mActiveTrack->setOverflow()) { 6249 nsecs_t now = systemTime(); 6250 if ((now - lastWarning) > kWarningThrottleNs) { 6251 ALOGW("RecordThread: buffer overflow"); 6252 lastWarning = now; 6253 } 6254 } 6255 // Release the processor for a while before asking for a new buffer. 6256 // This will give the application more chance to read from the buffer and 6257 // clear the overflow. 6258 usleep(kRecordThreadSleepUs); 6259 } 6260 } 6261 // enable changes in effect chain 6262 unlockEffectChains(effectChains); 6263 effectChains.clear(); 6264 } 6265 6266 standby(); 6267 6268 { 6269 Mutex::Autolock _l(mLock); 6270 mActiveTrack.clear(); 6271 mStartStopCond.broadcast(); 6272 } 6273 6274 releaseWakeLock(); 6275 6276 ALOGV("RecordThread %p exiting", this); 6277 return false; 6278} 6279 6280void AudioFlinger::RecordThread::standby() 6281{ 6282 if (!mStandby) { 6283 inputStandBy(); 6284 mStandby = true; 6285 } 6286} 6287 6288void AudioFlinger::RecordThread::inputStandBy() 6289{ 6290 mInput->stream->common.standby(&mInput->stream->common); 6291} 6292 6293sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6294 const sp<AudioFlinger::Client>& client, 6295 uint32_t sampleRate, 6296 audio_format_t format, 6297 audio_channel_mask_t channelMask, 6298 int frameCount, 6299 int sessionId, 6300 IAudioFlinger::track_flags_t flags, 6301 pid_t tid, 6302 status_t *status) 6303{ 6304 sp<RecordTrack> track; 6305 status_t lStatus; 6306 6307 lStatus = initCheck(); 6308 if (lStatus != NO_ERROR) { 6309 ALOGE("Audio driver not initialized."); 6310 goto Exit; 6311 } 6312 6313 // FIXME use flags and tid similar to createTrack_l() 6314 6315 { // scope for mLock 6316 Mutex::Autolock _l(mLock); 6317 6318 track = new RecordTrack(this, client, sampleRate, 6319 format, channelMask, frameCount, sessionId); 6320 6321 if (track->getCblk() == 0) { 6322 lStatus = NO_MEMORY; 6323 goto Exit; 6324 } 6325 mTracks.add(track); 6326 6327 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6328 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6329 mAudioFlinger->btNrecIsOff(); 6330 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6331 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6332 } 6333 lStatus = NO_ERROR; 6334 6335Exit: 6336 if (status) { 6337 *status = lStatus; 6338 } 6339 return track; 6340} 6341 6342status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6343 AudioSystem::sync_event_t event, 6344 int triggerSession) 6345{ 6346 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6347 sp<ThreadBase> strongMe = this; 6348 status_t status = NO_ERROR; 6349 6350 if (event == AudioSystem::SYNC_EVENT_NONE) { 6351 clearSyncStartEvent(); 6352 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6353 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6354 triggerSession, 6355 recordTrack->sessionId(), 6356 syncStartEventCallback, 6357 this); 6358 // Sync event can be cancelled by the trigger session if the track is not in a 6359 // compatible state in which case we start record immediately 6360 if (mSyncStartEvent->isCancelled()) { 6361 clearSyncStartEvent(); 6362 } else { 6363 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6364 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6365 } 6366 } 6367 6368 { 6369 AutoMutex lock(mLock); 6370 if (mActiveTrack != 0) { 6371 if (recordTrack != mActiveTrack.get()) { 6372 status = -EBUSY; 6373 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6374 mActiveTrack->mState = TrackBase::ACTIVE; 6375 } 6376 return status; 6377 } 6378 6379 recordTrack->mState = TrackBase::IDLE; 6380 mActiveTrack = recordTrack; 6381 mLock.unlock(); 6382 status_t status = AudioSystem::startInput(mId); 6383 mLock.lock(); 6384 if (status != NO_ERROR) { 6385 mActiveTrack.clear(); 6386 clearSyncStartEvent(); 6387 return status; 6388 } 6389 mRsmpInIndex = mFrameCount; 6390 mBytesRead = 0; 6391 if (mResampler != NULL) { 6392 mResampler->reset(); 6393 } 6394 mActiveTrack->mState = TrackBase::RESUMING; 6395 // signal thread to start 6396 ALOGV("Signal record thread"); 6397 mWaitWorkCV.broadcast(); 6398 // do not wait for mStartStopCond if exiting 6399 if (exitPending()) { 6400 mActiveTrack.clear(); 6401 status = INVALID_OPERATION; 6402 goto startError; 6403 } 6404 mStartStopCond.wait(mLock); 6405 if (mActiveTrack == 0) { 6406 ALOGV("Record failed to start"); 6407 status = BAD_VALUE; 6408 goto startError; 6409 } 6410 ALOGV("Record started OK"); 6411 return status; 6412 } 6413startError: 6414 AudioSystem::stopInput(mId); 6415 clearSyncStartEvent(); 6416 return status; 6417} 6418 6419void AudioFlinger::RecordThread::clearSyncStartEvent() 6420{ 6421 if (mSyncStartEvent != 0) { 6422 mSyncStartEvent->cancel(); 6423 } 6424 mSyncStartEvent.clear(); 6425 mFramestoDrop = 0; 6426} 6427 6428void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6429{ 6430 sp<SyncEvent> strongEvent = event.promote(); 6431 6432 if (strongEvent != 0) { 6433 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6434 me->handleSyncStartEvent(strongEvent); 6435 } 6436} 6437 6438void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6439{ 6440 if (event == mSyncStartEvent) { 6441 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6442 // from audio HAL 6443 mFramestoDrop = mFrameCount * 2; 6444 } 6445} 6446 6447bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6448 ALOGV("RecordThread::stop"); 6449 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6450 return false; 6451 } 6452 recordTrack->mState = TrackBase::PAUSING; 6453 // do not wait for mStartStopCond if exiting 6454 if (exitPending()) { 6455 return true; 6456 } 6457 mStartStopCond.wait(mLock); 6458 // if we have been restarted, recordTrack == mActiveTrack.get() here 6459 if (exitPending() || recordTrack != mActiveTrack.get()) { 6460 ALOGV("Record stopped OK"); 6461 return true; 6462 } 6463 return false; 6464} 6465 6466bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6467{ 6468 return false; 6469} 6470 6471status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6472{ 6473#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6474 if (!isValidSyncEvent(event)) { 6475 return BAD_VALUE; 6476 } 6477 6478 int eventSession = event->triggerSession(); 6479 status_t ret = NAME_NOT_FOUND; 6480 6481 Mutex::Autolock _l(mLock); 6482 6483 for (size_t i = 0; i < mTracks.size(); i++) { 6484 sp<RecordTrack> track = mTracks[i]; 6485 if (eventSession == track->sessionId()) { 6486 (void) track->setSyncEvent(event); 6487 ret = NO_ERROR; 6488 } 6489 } 6490 return ret; 6491#else 6492 return BAD_VALUE; 6493#endif 6494} 6495 6496void AudioFlinger::RecordThread::RecordTrack::destroy() 6497{ 6498 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6499 sp<RecordTrack> keep(this); 6500 { 6501 sp<ThreadBase> thread = mThread.promote(); 6502 if (thread != 0) { 6503 if (mState == ACTIVE || mState == RESUMING) { 6504 AudioSystem::stopInput(thread->id()); 6505 } 6506 AudioSystem::releaseInput(thread->id()); 6507 Mutex::Autolock _l(thread->mLock); 6508 RecordThread *recordThread = (RecordThread *) thread.get(); 6509 recordThread->destroyTrack_l(this); 6510 } 6511 } 6512} 6513 6514// destroyTrack_l() must be called with ThreadBase::mLock held 6515void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6516{ 6517 track->mState = TrackBase::TERMINATED; 6518 // active tracks are removed by threadLoop() 6519 if (mActiveTrack != track) { 6520 removeTrack_l(track); 6521 } 6522} 6523 6524void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6525{ 6526 mTracks.remove(track); 6527 // need anything related to effects here? 6528} 6529 6530void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6531{ 6532 dumpInternals(fd, args); 6533 dumpTracks(fd, args); 6534 dumpEffectChains(fd, args); 6535} 6536 6537void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6538{ 6539 const size_t SIZE = 256; 6540 char buffer[SIZE]; 6541 String8 result; 6542 6543 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6544 result.append(buffer); 6545 6546 if (mActiveTrack != 0) { 6547 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6548 result.append(buffer); 6549 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6550 result.append(buffer); 6551 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6552 result.append(buffer); 6553 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6554 result.append(buffer); 6555 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 6556 result.append(buffer); 6557 } else { 6558 result.append("No active record client\n"); 6559 } 6560 6561 write(fd, result.string(), result.size()); 6562 6563 dumpBase(fd, args); 6564} 6565 6566void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6567{ 6568 const size_t SIZE = 256; 6569 char buffer[SIZE]; 6570 String8 result; 6571 6572 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6573 result.append(buffer); 6574 RecordTrack::appendDumpHeader(result); 6575 for (size_t i = 0; i < mTracks.size(); ++i) { 6576 sp<RecordTrack> track = mTracks[i]; 6577 if (track != 0) { 6578 track->dump(buffer, SIZE); 6579 result.append(buffer); 6580 } 6581 } 6582 6583 if (mActiveTrack != 0) { 6584 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6585 result.append(buffer); 6586 RecordTrack::appendDumpHeader(result); 6587 mActiveTrack->dump(buffer, SIZE); 6588 result.append(buffer); 6589 6590 } 6591 write(fd, result.string(), result.size()); 6592} 6593 6594// AudioBufferProvider interface 6595status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6596{ 6597 size_t framesReq = buffer->frameCount; 6598 size_t framesReady = mFrameCount - mRsmpInIndex; 6599 int channelCount; 6600 6601 if (framesReady == 0) { 6602 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6603 if (mBytesRead <= 0) { 6604 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6605 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6606 // Force input into standby so that it tries to 6607 // recover at next read attempt 6608 inputStandBy(); 6609 usleep(kRecordThreadSleepUs); 6610 } 6611 buffer->raw = NULL; 6612 buffer->frameCount = 0; 6613 return NOT_ENOUGH_DATA; 6614 } 6615 mRsmpInIndex = 0; 6616 framesReady = mFrameCount; 6617 } 6618 6619 if (framesReq > framesReady) { 6620 framesReq = framesReady; 6621 } 6622 6623 if (mChannelCount == 1 && mReqChannelCount == 2) { 6624 channelCount = 1; 6625 } else { 6626 channelCount = 2; 6627 } 6628 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6629 buffer->frameCount = framesReq; 6630 return NO_ERROR; 6631} 6632 6633// AudioBufferProvider interface 6634void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6635{ 6636 mRsmpInIndex += buffer->frameCount; 6637 buffer->frameCount = 0; 6638} 6639 6640bool AudioFlinger::RecordThread::checkForNewParameters_l() 6641{ 6642 bool reconfig = false; 6643 6644 while (!mNewParameters.isEmpty()) { 6645 status_t status = NO_ERROR; 6646 String8 keyValuePair = mNewParameters[0]; 6647 AudioParameter param = AudioParameter(keyValuePair); 6648 int value; 6649 audio_format_t reqFormat = mFormat; 6650 uint32_t reqSamplingRate = mReqSampleRate; 6651 int reqChannelCount = mReqChannelCount; 6652 6653 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6654 reqSamplingRate = value; 6655 reconfig = true; 6656 } 6657 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6658 reqFormat = (audio_format_t) value; 6659 reconfig = true; 6660 } 6661 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6662 reqChannelCount = popcount(value); 6663 reconfig = true; 6664 } 6665 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6666 // do not accept frame count changes if tracks are open as the track buffer 6667 // size depends on frame count and correct behavior would not be guaranteed 6668 // if frame count is changed after track creation 6669 if (mActiveTrack != 0) { 6670 status = INVALID_OPERATION; 6671 } else { 6672 reconfig = true; 6673 } 6674 } 6675 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6676 // forward device change to effects that have requested to be 6677 // aware of attached audio device. 6678 for (size_t i = 0; i < mEffectChains.size(); i++) { 6679 mEffectChains[i]->setDevice_l(value); 6680 } 6681 6682 // store input device and output device but do not forward output device to audio HAL. 6683 // Note that status is ignored by the caller for output device 6684 // (see AudioFlinger::setParameters() 6685 if (audio_is_output_devices(value)) { 6686 mOutDevice = value; 6687 status = BAD_VALUE; 6688 } else { 6689 mInDevice = value; 6690 // disable AEC and NS if the device is a BT SCO headset supporting those 6691 // pre processings 6692 if (mTracks.size() > 0) { 6693 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6694 mAudioFlinger->btNrecIsOff(); 6695 for (size_t i = 0; i < mTracks.size(); i++) { 6696 sp<RecordTrack> track = mTracks[i]; 6697 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6698 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6699 } 6700 } 6701 } 6702 } 6703 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6704 mAudioSource != (audio_source_t)value) { 6705 // forward device change to effects that have requested to be 6706 // aware of attached audio device. 6707 for (size_t i = 0; i < mEffectChains.size(); i++) { 6708 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6709 } 6710 mAudioSource = (audio_source_t)value; 6711 } 6712 if (status == NO_ERROR) { 6713 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6714 keyValuePair.string()); 6715 if (status == INVALID_OPERATION) { 6716 inputStandBy(); 6717 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6718 keyValuePair.string()); 6719 } 6720 if (reconfig) { 6721 if (status == BAD_VALUE && 6722 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6723 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6724 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6725 <= (2 * reqSamplingRate)) && 6726 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6727 <= FCC_2 && 6728 (reqChannelCount <= FCC_2)) { 6729 status = NO_ERROR; 6730 } 6731 if (status == NO_ERROR) { 6732 readInputParameters(); 6733 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6734 } 6735 } 6736 } 6737 6738 mNewParameters.removeAt(0); 6739 6740 mParamStatus = status; 6741 mParamCond.signal(); 6742 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6743 // already timed out waiting for the status and will never signal the condition. 6744 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6745 } 6746 return reconfig; 6747} 6748 6749String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6750{ 6751 char *s; 6752 String8 out_s8 = String8(); 6753 6754 Mutex::Autolock _l(mLock); 6755 if (initCheck() != NO_ERROR) { 6756 return out_s8; 6757 } 6758 6759 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6760 out_s8 = String8(s); 6761 free(s); 6762 return out_s8; 6763} 6764 6765void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6766 AudioSystem::OutputDescriptor desc; 6767 void *param2 = NULL; 6768 6769 switch (event) { 6770 case AudioSystem::INPUT_OPENED: 6771 case AudioSystem::INPUT_CONFIG_CHANGED: 6772 desc.channels = mChannelMask; 6773 desc.samplingRate = mSampleRate; 6774 desc.format = mFormat; 6775 desc.frameCount = mFrameCount; 6776 desc.latency = 0; 6777 param2 = &desc; 6778 break; 6779 6780 case AudioSystem::INPUT_CLOSED: 6781 default: 6782 break; 6783 } 6784 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6785} 6786 6787void AudioFlinger::RecordThread::readInputParameters() 6788{ 6789 delete mRsmpInBuffer; 6790 // mRsmpInBuffer is always assigned a new[] below 6791 delete mRsmpOutBuffer; 6792 mRsmpOutBuffer = NULL; 6793 delete mResampler; 6794 mResampler = NULL; 6795 6796 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6797 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6798 mChannelCount = (uint16_t)popcount(mChannelMask); 6799 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6800 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6801 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6802 mFrameCount = mInputBytes / mFrameSize; 6803 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6804 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6805 6806 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6807 { 6808 int channelCount; 6809 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6810 // stereo to mono post process as the resampler always outputs stereo. 6811 if (mChannelCount == 1 && mReqChannelCount == 2) { 6812 channelCount = 1; 6813 } else { 6814 channelCount = 2; 6815 } 6816 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6817 mResampler->setSampleRate(mSampleRate); 6818 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6819 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6820 6821 // optmization: if mono to mono, alter input frame count as if we were inputing 6822 // stereo samples 6823 if (mChannelCount == 1 && mReqChannelCount == 1) { 6824 mFrameCount >>= 1; 6825 } 6826 6827 } 6828 mRsmpInIndex = mFrameCount; 6829} 6830 6831unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6832{ 6833 Mutex::Autolock _l(mLock); 6834 if (initCheck() != NO_ERROR) { 6835 return 0; 6836 } 6837 6838 return mInput->stream->get_input_frames_lost(mInput->stream); 6839} 6840 6841uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6842{ 6843 Mutex::Autolock _l(mLock); 6844 uint32_t result = 0; 6845 if (getEffectChain_l(sessionId) != 0) { 6846 result = EFFECT_SESSION; 6847 } 6848 6849 for (size_t i = 0; i < mTracks.size(); ++i) { 6850 if (sessionId == mTracks[i]->sessionId()) { 6851 result |= TRACK_SESSION; 6852 break; 6853 } 6854 } 6855 6856 return result; 6857} 6858 6859KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6860{ 6861 KeyedVector<int, bool> ids; 6862 Mutex::Autolock _l(mLock); 6863 for (size_t j = 0; j < mTracks.size(); ++j) { 6864 sp<RecordThread::RecordTrack> track = mTracks[j]; 6865 int sessionId = track->sessionId(); 6866 if (ids.indexOfKey(sessionId) < 0) { 6867 ids.add(sessionId, true); 6868 } 6869 } 6870 return ids; 6871} 6872 6873AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6874{ 6875 Mutex::Autolock _l(mLock); 6876 AudioStreamIn *input = mInput; 6877 mInput = NULL; 6878 return input; 6879} 6880 6881// this method must always be called either with ThreadBase mLock held or inside the thread loop 6882audio_stream_t* AudioFlinger::RecordThread::stream() const 6883{ 6884 if (mInput == NULL) { 6885 return NULL; 6886 } 6887 return &mInput->stream->common; 6888} 6889 6890 6891// ---------------------------------------------------------------------------- 6892 6893audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6894{ 6895 if (!settingsAllowed()) { 6896 return 0; 6897 } 6898 Mutex::Autolock _l(mLock); 6899 return loadHwModule_l(name); 6900} 6901 6902// loadHwModule_l() must be called with AudioFlinger::mLock held 6903audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6904{ 6905 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6906 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6907 ALOGW("loadHwModule() module %s already loaded", name); 6908 return mAudioHwDevs.keyAt(i); 6909 } 6910 } 6911 6912 audio_hw_device_t *dev; 6913 6914 int rc = load_audio_interface(name, &dev); 6915 if (rc) { 6916 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6917 return 0; 6918 } 6919 6920 mHardwareStatus = AUDIO_HW_INIT; 6921 rc = dev->init_check(dev); 6922 mHardwareStatus = AUDIO_HW_IDLE; 6923 if (rc) { 6924 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6925 return 0; 6926 } 6927 6928 // Check and cache this HAL's level of support for master mute and master 6929 // volume. If this is the first HAL opened, and it supports the get 6930 // methods, use the initial values provided by the HAL as the current 6931 // master mute and volume settings. 6932 6933 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6934 { // scope for auto-lock pattern 6935 AutoMutex lock(mHardwareLock); 6936 6937 if (0 == mAudioHwDevs.size()) { 6938 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6939 if (NULL != dev->get_master_volume) { 6940 float mv; 6941 if (OK == dev->get_master_volume(dev, &mv)) { 6942 mMasterVolume = mv; 6943 } 6944 } 6945 6946 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6947 if (NULL != dev->get_master_mute) { 6948 bool mm; 6949 if (OK == dev->get_master_mute(dev, &mm)) { 6950 mMasterMute = mm; 6951 } 6952 } 6953 } 6954 6955 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6956 if ((NULL != dev->set_master_volume) && 6957 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6958 flags = static_cast<AudioHwDevice::Flags>(flags | 6959 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6960 } 6961 6962 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6963 if ((NULL != dev->set_master_mute) && 6964 (OK == dev->set_master_mute(dev, mMasterMute))) { 6965 flags = static_cast<AudioHwDevice::Flags>(flags | 6966 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6967 } 6968 6969 mHardwareStatus = AUDIO_HW_IDLE; 6970 } 6971 6972 audio_module_handle_t handle = nextUniqueId(); 6973 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6974 6975 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6976 name, dev->common.module->name, dev->common.module->id, handle); 6977 6978 return handle; 6979 6980} 6981 6982// ---------------------------------------------------------------------------- 6983 6984uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 6985{ 6986 Mutex::Autolock _l(mLock); 6987 PlaybackThread *thread = primaryPlaybackThread_l(); 6988 return thread != NULL ? thread->sampleRate() : 0; 6989} 6990 6991int32_t AudioFlinger::getPrimaryOutputFrameCount() 6992{ 6993 Mutex::Autolock _l(mLock); 6994 PlaybackThread *thread = primaryPlaybackThread_l(); 6995 return thread != NULL ? thread->frameCountHAL() : 0; 6996} 6997 6998// ---------------------------------------------------------------------------- 6999 7000audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 7001 audio_devices_t *pDevices, 7002 uint32_t *pSamplingRate, 7003 audio_format_t *pFormat, 7004 audio_channel_mask_t *pChannelMask, 7005 uint32_t *pLatencyMs, 7006 audio_output_flags_t flags) 7007{ 7008 status_t status; 7009 PlaybackThread *thread = NULL; 7010 struct audio_config config = { 7011 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7012 channel_mask: pChannelMask ? *pChannelMask : 0, 7013 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7014 }; 7015 audio_stream_out_t *outStream = NULL; 7016 AudioHwDevice *outHwDev; 7017 7018 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7019 module, 7020 (pDevices != NULL) ? *pDevices : 0, 7021 config.sample_rate, 7022 config.format, 7023 config.channel_mask, 7024 flags); 7025 7026 if (pDevices == NULL || *pDevices == 0) { 7027 return 0; 7028 } 7029 7030 Mutex::Autolock _l(mLock); 7031 7032 outHwDev = findSuitableHwDev_l(module, *pDevices); 7033 if (outHwDev == NULL) 7034 return 0; 7035 7036 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7037 audio_io_handle_t id = nextUniqueId(); 7038 7039 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7040 7041 status = hwDevHal->open_output_stream(hwDevHal, 7042 id, 7043 *pDevices, 7044 (audio_output_flags_t)flags, 7045 &config, 7046 &outStream); 7047 7048 mHardwareStatus = AUDIO_HW_IDLE; 7049 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7050 "Channels %x, status %d", 7051 outStream, 7052 config.sample_rate, 7053 config.format, 7054 config.channel_mask, 7055 status); 7056 7057 if (status == NO_ERROR && outStream != NULL) { 7058 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7059 7060 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7061 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7062 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7063 thread = new DirectOutputThread(this, output, id, *pDevices); 7064 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7065 } else { 7066 thread = new MixerThread(this, output, id, *pDevices); 7067 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7068 } 7069 mPlaybackThreads.add(id, thread); 7070 7071 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7072 if (pFormat != NULL) *pFormat = config.format; 7073 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7074 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7075 7076 // notify client processes of the new output creation 7077 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7078 7079 // the first primary output opened designates the primary hw device 7080 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7081 ALOGI("Using module %d has the primary audio interface", module); 7082 mPrimaryHardwareDev = outHwDev; 7083 7084 AutoMutex lock(mHardwareLock); 7085 mHardwareStatus = AUDIO_HW_SET_MODE; 7086 hwDevHal->set_mode(hwDevHal, mMode); 7087 mHardwareStatus = AUDIO_HW_IDLE; 7088 } 7089 return id; 7090 } 7091 7092 return 0; 7093} 7094 7095audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7096 audio_io_handle_t output2) 7097{ 7098 Mutex::Autolock _l(mLock); 7099 MixerThread *thread1 = checkMixerThread_l(output1); 7100 MixerThread *thread2 = checkMixerThread_l(output2); 7101 7102 if (thread1 == NULL || thread2 == NULL) { 7103 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7104 output2); 7105 return 0; 7106 } 7107 7108 audio_io_handle_t id = nextUniqueId(); 7109 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7110 thread->addOutputTrack(thread2); 7111 mPlaybackThreads.add(id, thread); 7112 // notify client processes of the new output creation 7113 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7114 return id; 7115} 7116 7117status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7118{ 7119 return closeOutput_nonvirtual(output); 7120} 7121 7122status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7123{ 7124 // keep strong reference on the playback thread so that 7125 // it is not destroyed while exit() is executed 7126 sp<PlaybackThread> thread; 7127 { 7128 Mutex::Autolock _l(mLock); 7129 thread = checkPlaybackThread_l(output); 7130 if (thread == NULL) { 7131 return BAD_VALUE; 7132 } 7133 7134 ALOGV("closeOutput() %d", output); 7135 7136 if (thread->type() == ThreadBase::MIXER) { 7137 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7138 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7139 DuplicatingThread *dupThread = 7140 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7141 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7142 } 7143 } 7144 } 7145 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7146 mPlaybackThreads.removeItem(output); 7147 } 7148 thread->exit(); 7149 // The thread entity (active unit of execution) is no longer running here, 7150 // but the ThreadBase container still exists. 7151 7152 if (thread->type() != ThreadBase::DUPLICATING) { 7153 AudioStreamOut *out = thread->clearOutput(); 7154 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7155 // from now on thread->mOutput is NULL 7156 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7157 delete out; 7158 } 7159 return NO_ERROR; 7160} 7161 7162status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7163{ 7164 Mutex::Autolock _l(mLock); 7165 PlaybackThread *thread = checkPlaybackThread_l(output); 7166 7167 if (thread == NULL) { 7168 return BAD_VALUE; 7169 } 7170 7171 ALOGV("suspendOutput() %d", output); 7172 thread->suspend(); 7173 7174 return NO_ERROR; 7175} 7176 7177status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7178{ 7179 Mutex::Autolock _l(mLock); 7180 PlaybackThread *thread = checkPlaybackThread_l(output); 7181 7182 if (thread == NULL) { 7183 return BAD_VALUE; 7184 } 7185 7186 ALOGV("restoreOutput() %d", output); 7187 7188 thread->restore(); 7189 7190 return NO_ERROR; 7191} 7192 7193audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7194 audio_devices_t *pDevices, 7195 uint32_t *pSamplingRate, 7196 audio_format_t *pFormat, 7197 audio_channel_mask_t *pChannelMask) 7198{ 7199 status_t status; 7200 RecordThread *thread = NULL; 7201 struct audio_config config = { 7202 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7203 channel_mask: pChannelMask ? *pChannelMask : 0, 7204 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7205 }; 7206 uint32_t reqSamplingRate = config.sample_rate; 7207 audio_format_t reqFormat = config.format; 7208 audio_channel_mask_t reqChannels = config.channel_mask; 7209 audio_stream_in_t *inStream = NULL; 7210 AudioHwDevice *inHwDev; 7211 7212 if (pDevices == NULL || *pDevices == 0) { 7213 return 0; 7214 } 7215 7216 Mutex::Autolock _l(mLock); 7217 7218 inHwDev = findSuitableHwDev_l(module, *pDevices); 7219 if (inHwDev == NULL) 7220 return 0; 7221 7222 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7223 audio_io_handle_t id = nextUniqueId(); 7224 7225 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7226 &inStream); 7227 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7228 "status %d", 7229 inStream, 7230 config.sample_rate, 7231 config.format, 7232 config.channel_mask, 7233 status); 7234 7235 // If the input could not be opened with the requested parameters and we can handle the 7236 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7237 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7238 if (status == BAD_VALUE && 7239 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7240 (config.sample_rate <= 2 * reqSamplingRate) && 7241 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7242 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7243 inStream = NULL; 7244 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7245 } 7246 7247 if (status == NO_ERROR && inStream != NULL) { 7248 7249 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7250 // or (re-)create if current Pipe is idle and does not match the new format 7251 sp<NBAIO_Sink> teeSink; 7252#ifdef TEE_SINK_INPUT_FRAMES 7253 enum { 7254 TEE_SINK_NO, // don't copy input 7255 TEE_SINK_NEW, // copy input using a new pipe 7256 TEE_SINK_OLD, // copy input using an existing pipe 7257 } kind; 7258 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7259 popcount(inStream->common.get_channels(&inStream->common))); 7260 if (format == Format_Invalid) { 7261 kind = TEE_SINK_NO; 7262 } else if (mRecordTeeSink == 0) { 7263 kind = TEE_SINK_NEW; 7264 } else if (mRecordTeeSink->getStrongCount() != 1) { 7265 kind = TEE_SINK_NO; 7266 } else if (format == mRecordTeeSink->format()) { 7267 kind = TEE_SINK_OLD; 7268 } else { 7269 kind = TEE_SINK_NEW; 7270 } 7271 switch (kind) { 7272 case TEE_SINK_NEW: { 7273 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7274 size_t numCounterOffers = 0; 7275 const NBAIO_Format offers[1] = {format}; 7276 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7277 ALOG_ASSERT(index == 0); 7278 PipeReader *pipeReader = new PipeReader(*pipe); 7279 numCounterOffers = 0; 7280 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7281 ALOG_ASSERT(index == 0); 7282 mRecordTeeSink = pipe; 7283 mRecordTeeSource = pipeReader; 7284 teeSink = pipe; 7285 } 7286 break; 7287 case TEE_SINK_OLD: 7288 teeSink = mRecordTeeSink; 7289 break; 7290 case TEE_SINK_NO: 7291 default: 7292 break; 7293 } 7294#endif 7295 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7296 7297 // Start record thread 7298 // RecorThread require both input and output device indication to forward to audio 7299 // pre processing modules 7300 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7301 7302 thread = new RecordThread(this, 7303 input, 7304 reqSamplingRate, 7305 reqChannels, 7306 id, 7307 device, teeSink); 7308 mRecordThreads.add(id, thread); 7309 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7310 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7311 if (pFormat != NULL) *pFormat = config.format; 7312 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7313 7314 // notify client processes of the new input creation 7315 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7316 return id; 7317 } 7318 7319 return 0; 7320} 7321 7322status_t AudioFlinger::closeInput(audio_io_handle_t input) 7323{ 7324 return closeInput_nonvirtual(input); 7325} 7326 7327status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7328{ 7329 // keep strong reference on the record thread so that 7330 // it is not destroyed while exit() is executed 7331 sp<RecordThread> thread; 7332 { 7333 Mutex::Autolock _l(mLock); 7334 thread = checkRecordThread_l(input); 7335 if (thread == 0) { 7336 return BAD_VALUE; 7337 } 7338 7339 ALOGV("closeInput() %d", input); 7340 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7341 mRecordThreads.removeItem(input); 7342 } 7343 thread->exit(); 7344 // The thread entity (active unit of execution) is no longer running here, 7345 // but the ThreadBase container still exists. 7346 7347 AudioStreamIn *in = thread->clearInput(); 7348 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7349 // from now on thread->mInput is NULL 7350 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7351 delete in; 7352 7353 return NO_ERROR; 7354} 7355 7356status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7357{ 7358 Mutex::Autolock _l(mLock); 7359 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7360 7361 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7362 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7363 thread->invalidateTracks(stream); 7364 } 7365 7366 return NO_ERROR; 7367} 7368 7369 7370int AudioFlinger::newAudioSessionId() 7371{ 7372 return nextUniqueId(); 7373} 7374 7375void AudioFlinger::acquireAudioSessionId(int audioSession) 7376{ 7377 Mutex::Autolock _l(mLock); 7378 pid_t caller = IPCThreadState::self()->getCallingPid(); 7379 ALOGV("acquiring %d from %d", audioSession, caller); 7380 size_t num = mAudioSessionRefs.size(); 7381 for (size_t i = 0; i< num; i++) { 7382 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7383 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7384 ref->mCnt++; 7385 ALOGV(" incremented refcount to %d", ref->mCnt); 7386 return; 7387 } 7388 } 7389 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7390 ALOGV(" added new entry for %d", audioSession); 7391} 7392 7393void AudioFlinger::releaseAudioSessionId(int audioSession) 7394{ 7395 Mutex::Autolock _l(mLock); 7396 pid_t caller = IPCThreadState::self()->getCallingPid(); 7397 ALOGV("releasing %d from %d", audioSession, caller); 7398 size_t num = mAudioSessionRefs.size(); 7399 for (size_t i = 0; i< num; i++) { 7400 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7401 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7402 ref->mCnt--; 7403 ALOGV(" decremented refcount to %d", ref->mCnt); 7404 if (ref->mCnt == 0) { 7405 mAudioSessionRefs.removeAt(i); 7406 delete ref; 7407 purgeStaleEffects_l(); 7408 } 7409 return; 7410 } 7411 } 7412 ALOGW("session id %d not found for pid %d", audioSession, caller); 7413} 7414 7415void AudioFlinger::purgeStaleEffects_l() { 7416 7417 ALOGV("purging stale effects"); 7418 7419 Vector< sp<EffectChain> > chains; 7420 7421 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7422 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7423 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7424 sp<EffectChain> ec = t->mEffectChains[j]; 7425 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7426 chains.push(ec); 7427 } 7428 } 7429 } 7430 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7431 sp<RecordThread> t = mRecordThreads.valueAt(i); 7432 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7433 sp<EffectChain> ec = t->mEffectChains[j]; 7434 chains.push(ec); 7435 } 7436 } 7437 7438 for (size_t i = 0; i < chains.size(); i++) { 7439 sp<EffectChain> ec = chains[i]; 7440 int sessionid = ec->sessionId(); 7441 sp<ThreadBase> t = ec->mThread.promote(); 7442 if (t == 0) { 7443 continue; 7444 } 7445 size_t numsessionrefs = mAudioSessionRefs.size(); 7446 bool found = false; 7447 for (size_t k = 0; k < numsessionrefs; k++) { 7448 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7449 if (ref->mSessionid == sessionid) { 7450 ALOGV(" session %d still exists for %d with %d refs", 7451 sessionid, ref->mPid, ref->mCnt); 7452 found = true; 7453 break; 7454 } 7455 } 7456 if (!found) { 7457 Mutex::Autolock _l (t->mLock); 7458 // remove all effects from the chain 7459 while (ec->mEffects.size()) { 7460 sp<EffectModule> effect = ec->mEffects[0]; 7461 effect->unPin(); 7462 t->removeEffect_l(effect); 7463 if (effect->purgeHandles()) { 7464 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7465 } 7466 AudioSystem::unregisterEffect(effect->id()); 7467 } 7468 } 7469 } 7470 return; 7471} 7472 7473// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7474AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7475{ 7476 return mPlaybackThreads.valueFor(output).get(); 7477} 7478 7479// checkMixerThread_l() must be called with AudioFlinger::mLock held 7480AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7481{ 7482 PlaybackThread *thread = checkPlaybackThread_l(output); 7483 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7484} 7485 7486// checkRecordThread_l() must be called with AudioFlinger::mLock held 7487AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7488{ 7489 return mRecordThreads.valueFor(input).get(); 7490} 7491 7492uint32_t AudioFlinger::nextUniqueId() 7493{ 7494 return android_atomic_inc(&mNextUniqueId); 7495} 7496 7497AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7498{ 7499 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7500 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7501 AudioStreamOut *output = thread->getOutput(); 7502 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7503 return thread; 7504 } 7505 } 7506 return NULL; 7507} 7508 7509audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7510{ 7511 PlaybackThread *thread = primaryPlaybackThread_l(); 7512 7513 if (thread == NULL) { 7514 return 0; 7515 } 7516 7517 return thread->outDevice(); 7518} 7519 7520sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7521 int triggerSession, 7522 int listenerSession, 7523 sync_event_callback_t callBack, 7524 void *cookie) 7525{ 7526 Mutex::Autolock _l(mLock); 7527 7528 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7529 status_t playStatus = NAME_NOT_FOUND; 7530 status_t recStatus = NAME_NOT_FOUND; 7531 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7532 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7533 if (playStatus == NO_ERROR) { 7534 return event; 7535 } 7536 } 7537 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7538 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7539 if (recStatus == NO_ERROR) { 7540 return event; 7541 } 7542 } 7543 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7544 mPendingSyncEvents.add(event); 7545 } else { 7546 ALOGV("createSyncEvent() invalid event %d", event->type()); 7547 event.clear(); 7548 } 7549 return event; 7550} 7551 7552// ---------------------------------------------------------------------------- 7553// Effect management 7554// ---------------------------------------------------------------------------- 7555 7556 7557status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7558{ 7559 Mutex::Autolock _l(mLock); 7560 return EffectQueryNumberEffects(numEffects); 7561} 7562 7563status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7564{ 7565 Mutex::Autolock _l(mLock); 7566 return EffectQueryEffect(index, descriptor); 7567} 7568 7569status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7570 effect_descriptor_t *descriptor) const 7571{ 7572 Mutex::Autolock _l(mLock); 7573 return EffectGetDescriptor(pUuid, descriptor); 7574} 7575 7576 7577sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7578 effect_descriptor_t *pDesc, 7579 const sp<IEffectClient>& effectClient, 7580 int32_t priority, 7581 audio_io_handle_t io, 7582 int sessionId, 7583 status_t *status, 7584 int *id, 7585 int *enabled) 7586{ 7587 status_t lStatus = NO_ERROR; 7588 sp<EffectHandle> handle; 7589 effect_descriptor_t desc; 7590 7591 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7592 pid, effectClient.get(), priority, sessionId, io); 7593 7594 if (pDesc == NULL) { 7595 lStatus = BAD_VALUE; 7596 goto Exit; 7597 } 7598 7599 // check audio settings permission for global effects 7600 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7601 lStatus = PERMISSION_DENIED; 7602 goto Exit; 7603 } 7604 7605 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7606 // that can only be created by audio policy manager (running in same process) 7607 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7608 lStatus = PERMISSION_DENIED; 7609 goto Exit; 7610 } 7611 7612 if (io == 0) { 7613 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7614 // output must be specified by AudioPolicyManager when using session 7615 // AUDIO_SESSION_OUTPUT_STAGE 7616 lStatus = BAD_VALUE; 7617 goto Exit; 7618 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7619 // if the output returned by getOutputForEffect() is removed before we lock the 7620 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7621 // and we will exit safely 7622 io = AudioSystem::getOutputForEffect(&desc); 7623 } 7624 } 7625 7626 { 7627 Mutex::Autolock _l(mLock); 7628 7629 7630 if (!EffectIsNullUuid(&pDesc->uuid)) { 7631 // if uuid is specified, request effect descriptor 7632 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7633 if (lStatus < 0) { 7634 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7635 goto Exit; 7636 } 7637 } else { 7638 // if uuid is not specified, look for an available implementation 7639 // of the required type in effect factory 7640 if (EffectIsNullUuid(&pDesc->type)) { 7641 ALOGW("createEffect() no effect type"); 7642 lStatus = BAD_VALUE; 7643 goto Exit; 7644 } 7645 uint32_t numEffects = 0; 7646 effect_descriptor_t d; 7647 d.flags = 0; // prevent compiler warning 7648 bool found = false; 7649 7650 lStatus = EffectQueryNumberEffects(&numEffects); 7651 if (lStatus < 0) { 7652 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7653 goto Exit; 7654 } 7655 for (uint32_t i = 0; i < numEffects; i++) { 7656 lStatus = EffectQueryEffect(i, &desc); 7657 if (lStatus < 0) { 7658 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7659 continue; 7660 } 7661 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7662 // If matching type found save effect descriptor. If the session is 7663 // 0 and the effect is not auxiliary, continue enumeration in case 7664 // an auxiliary version of this effect type is available 7665 found = true; 7666 d = desc; 7667 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7668 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7669 break; 7670 } 7671 } 7672 } 7673 if (!found) { 7674 lStatus = BAD_VALUE; 7675 ALOGW("createEffect() effect not found"); 7676 goto Exit; 7677 } 7678 // For same effect type, chose auxiliary version over insert version if 7679 // connect to output mix (Compliance to OpenSL ES) 7680 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7681 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7682 desc = d; 7683 } 7684 } 7685 7686 // Do not allow auxiliary effects on a session different from 0 (output mix) 7687 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7688 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7689 lStatus = INVALID_OPERATION; 7690 goto Exit; 7691 } 7692 7693 // check recording permission for visualizer 7694 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7695 !recordingAllowed()) { 7696 lStatus = PERMISSION_DENIED; 7697 goto Exit; 7698 } 7699 7700 // return effect descriptor 7701 *pDesc = desc; 7702 7703 // If output is not specified try to find a matching audio session ID in one of the 7704 // output threads. 7705 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7706 // because of code checking output when entering the function. 7707 // Note: io is never 0 when creating an effect on an input 7708 if (io == 0) { 7709 // look for the thread where the specified audio session is present 7710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7711 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7712 io = mPlaybackThreads.keyAt(i); 7713 break; 7714 } 7715 } 7716 if (io == 0) { 7717 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7718 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7719 io = mRecordThreads.keyAt(i); 7720 break; 7721 } 7722 } 7723 } 7724 // If no output thread contains the requested session ID, default to 7725 // first output. The effect chain will be moved to the correct output 7726 // thread when a track with the same session ID is created 7727 if (io == 0 && mPlaybackThreads.size()) { 7728 io = mPlaybackThreads.keyAt(0); 7729 } 7730 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7731 } 7732 ThreadBase *thread = checkRecordThread_l(io); 7733 if (thread == NULL) { 7734 thread = checkPlaybackThread_l(io); 7735 if (thread == NULL) { 7736 ALOGE("createEffect() unknown output thread"); 7737 lStatus = BAD_VALUE; 7738 goto Exit; 7739 } 7740 } 7741 7742 sp<Client> client = registerPid_l(pid); 7743 7744 // create effect on selected output thread 7745 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7746 &desc, enabled, &lStatus); 7747 if (handle != 0 && id != NULL) { 7748 *id = handle->id(); 7749 } 7750 } 7751 7752Exit: 7753 if (status != NULL) { 7754 *status = lStatus; 7755 } 7756 return handle; 7757} 7758 7759status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7760 audio_io_handle_t dstOutput) 7761{ 7762 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7763 sessionId, srcOutput, dstOutput); 7764 Mutex::Autolock _l(mLock); 7765 if (srcOutput == dstOutput) { 7766 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7767 return NO_ERROR; 7768 } 7769 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7770 if (srcThread == NULL) { 7771 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7772 return BAD_VALUE; 7773 } 7774 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7775 if (dstThread == NULL) { 7776 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7777 return BAD_VALUE; 7778 } 7779 7780 Mutex::Autolock _dl(dstThread->mLock); 7781 Mutex::Autolock _sl(srcThread->mLock); 7782 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7783 7784 return NO_ERROR; 7785} 7786 7787// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7788status_t AudioFlinger::moveEffectChain_l(int sessionId, 7789 AudioFlinger::PlaybackThread *srcThread, 7790 AudioFlinger::PlaybackThread *dstThread, 7791 bool reRegister) 7792{ 7793 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7794 sessionId, srcThread, dstThread); 7795 7796 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7797 if (chain == 0) { 7798 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7799 sessionId, srcThread); 7800 return INVALID_OPERATION; 7801 } 7802 7803 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7804 // so that a new chain is created with correct parameters when first effect is added. This is 7805 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7806 // removed. 7807 srcThread->removeEffectChain_l(chain); 7808 7809 // transfer all effects one by one so that new effect chain is created on new thread with 7810 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7811 audio_io_handle_t dstOutput = dstThread->id(); 7812 sp<EffectChain> dstChain; 7813 uint32_t strategy = 0; // prevent compiler warning 7814 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7815 while (effect != 0) { 7816 srcThread->removeEffect_l(effect); 7817 dstThread->addEffect_l(effect); 7818 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7819 if (effect->state() == EffectModule::ACTIVE || 7820 effect->state() == EffectModule::STOPPING) { 7821 effect->start(); 7822 } 7823 // if the move request is not received from audio policy manager, the effect must be 7824 // re-registered with the new strategy and output 7825 if (dstChain == 0) { 7826 dstChain = effect->chain().promote(); 7827 if (dstChain == 0) { 7828 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7829 srcThread->addEffect_l(effect); 7830 return NO_INIT; 7831 } 7832 strategy = dstChain->strategy(); 7833 } 7834 if (reRegister) { 7835 AudioSystem::unregisterEffect(effect->id()); 7836 AudioSystem::registerEffect(&effect->desc(), 7837 dstOutput, 7838 strategy, 7839 sessionId, 7840 effect->id()); 7841 } 7842 effect = chain->getEffectFromId_l(0); 7843 } 7844 7845 return NO_ERROR; 7846} 7847 7848 7849// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7850sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7851 const sp<AudioFlinger::Client>& client, 7852 const sp<IEffectClient>& effectClient, 7853 int32_t priority, 7854 int sessionId, 7855 effect_descriptor_t *desc, 7856 int *enabled, 7857 status_t *status 7858 ) 7859{ 7860 sp<EffectModule> effect; 7861 sp<EffectHandle> handle; 7862 status_t lStatus; 7863 sp<EffectChain> chain; 7864 bool chainCreated = false; 7865 bool effectCreated = false; 7866 bool effectRegistered = false; 7867 7868 lStatus = initCheck(); 7869 if (lStatus != NO_ERROR) { 7870 ALOGW("createEffect_l() Audio driver not initialized."); 7871 goto Exit; 7872 } 7873 7874 // Do not allow effects with session ID 0 on direct output or duplicating threads 7875 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7876 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7877 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7878 desc->name, sessionId); 7879 lStatus = BAD_VALUE; 7880 goto Exit; 7881 } 7882 // Only Pre processor effects are allowed on input threads and only on input threads 7883 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7884 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7885 desc->name, desc->flags, mType); 7886 lStatus = BAD_VALUE; 7887 goto Exit; 7888 } 7889 7890 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7891 7892 { // scope for mLock 7893 Mutex::Autolock _l(mLock); 7894 7895 // check for existing effect chain with the requested audio session 7896 chain = getEffectChain_l(sessionId); 7897 if (chain == 0) { 7898 // create a new chain for this session 7899 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7900 chain = new EffectChain(this, sessionId); 7901 addEffectChain_l(chain); 7902 chain->setStrategy(getStrategyForSession_l(sessionId)); 7903 chainCreated = true; 7904 } else { 7905 effect = chain->getEffectFromDesc_l(desc); 7906 } 7907 7908 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7909 7910 if (effect == 0) { 7911 int id = mAudioFlinger->nextUniqueId(); 7912 // Check CPU and memory usage 7913 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7914 if (lStatus != NO_ERROR) { 7915 goto Exit; 7916 } 7917 effectRegistered = true; 7918 // create a new effect module if none present in the chain 7919 effect = new EffectModule(this, chain, desc, id, sessionId); 7920 lStatus = effect->status(); 7921 if (lStatus != NO_ERROR) { 7922 goto Exit; 7923 } 7924 lStatus = chain->addEffect_l(effect); 7925 if (lStatus != NO_ERROR) { 7926 goto Exit; 7927 } 7928 effectCreated = true; 7929 7930 effect->setDevice(mOutDevice); 7931 effect->setDevice(mInDevice); 7932 effect->setMode(mAudioFlinger->getMode()); 7933 effect->setAudioSource(mAudioSource); 7934 } 7935 // create effect handle and connect it to effect module 7936 handle = new EffectHandle(effect, client, effectClient, priority); 7937 lStatus = effect->addHandle(handle.get()); 7938 if (enabled != NULL) { 7939 *enabled = (int)effect->isEnabled(); 7940 } 7941 } 7942 7943Exit: 7944 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7945 Mutex::Autolock _l(mLock); 7946 if (effectCreated) { 7947 chain->removeEffect_l(effect); 7948 } 7949 if (effectRegistered) { 7950 AudioSystem::unregisterEffect(effect->id()); 7951 } 7952 if (chainCreated) { 7953 removeEffectChain_l(chain); 7954 } 7955 handle.clear(); 7956 } 7957 7958 if (status != NULL) { 7959 *status = lStatus; 7960 } 7961 return handle; 7962} 7963 7964sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7965{ 7966 Mutex::Autolock _l(mLock); 7967 return getEffect_l(sessionId, effectId); 7968} 7969 7970sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7971{ 7972 sp<EffectChain> chain = getEffectChain_l(sessionId); 7973 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7974} 7975 7976// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7977// PlaybackThread::mLock held 7978status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7979{ 7980 // check for existing effect chain with the requested audio session 7981 int sessionId = effect->sessionId(); 7982 sp<EffectChain> chain = getEffectChain_l(sessionId); 7983 bool chainCreated = false; 7984 7985 if (chain == 0) { 7986 // create a new chain for this session 7987 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7988 chain = new EffectChain(this, sessionId); 7989 addEffectChain_l(chain); 7990 chain->setStrategy(getStrategyForSession_l(sessionId)); 7991 chainCreated = true; 7992 } 7993 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7994 7995 if (chain->getEffectFromId_l(effect->id()) != 0) { 7996 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7997 this, effect->desc().name, chain.get()); 7998 return BAD_VALUE; 7999 } 8000 8001 status_t status = chain->addEffect_l(effect); 8002 if (status != NO_ERROR) { 8003 if (chainCreated) { 8004 removeEffectChain_l(chain); 8005 } 8006 return status; 8007 } 8008 8009 effect->setDevice(mOutDevice); 8010 effect->setDevice(mInDevice); 8011 effect->setMode(mAudioFlinger->getMode()); 8012 effect->setAudioSource(mAudioSource); 8013 return NO_ERROR; 8014} 8015 8016void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8017 8018 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8019 effect_descriptor_t desc = effect->desc(); 8020 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8021 detachAuxEffect_l(effect->id()); 8022 } 8023 8024 sp<EffectChain> chain = effect->chain().promote(); 8025 if (chain != 0) { 8026 // remove effect chain if removing last effect 8027 if (chain->removeEffect_l(effect) == 0) { 8028 removeEffectChain_l(chain); 8029 } 8030 } else { 8031 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8032 } 8033} 8034 8035void AudioFlinger::ThreadBase::lockEffectChains_l( 8036 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8037{ 8038 effectChains = mEffectChains; 8039 for (size_t i = 0; i < mEffectChains.size(); i++) { 8040 mEffectChains[i]->lock(); 8041 } 8042} 8043 8044void AudioFlinger::ThreadBase::unlockEffectChains( 8045 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8046{ 8047 for (size_t i = 0; i < effectChains.size(); i++) { 8048 effectChains[i]->unlock(); 8049 } 8050} 8051 8052sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8053{ 8054 Mutex::Autolock _l(mLock); 8055 return getEffectChain_l(sessionId); 8056} 8057 8058sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8059{ 8060 size_t size = mEffectChains.size(); 8061 for (size_t i = 0; i < size; i++) { 8062 if (mEffectChains[i]->sessionId() == sessionId) { 8063 return mEffectChains[i]; 8064 } 8065 } 8066 return 0; 8067} 8068 8069void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8070{ 8071 Mutex::Autolock _l(mLock); 8072 size_t size = mEffectChains.size(); 8073 for (size_t i = 0; i < size; i++) { 8074 mEffectChains[i]->setMode_l(mode); 8075 } 8076} 8077 8078void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8079 EffectHandle *handle, 8080 bool unpinIfLast) { 8081 8082 Mutex::Autolock _l(mLock); 8083 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8084 // delete the effect module if removing last handle on it 8085 if (effect->removeHandle(handle) == 0) { 8086 if (!effect->isPinned() || unpinIfLast) { 8087 removeEffect_l(effect); 8088 AudioSystem::unregisterEffect(effect->id()); 8089 } 8090 } 8091} 8092 8093status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8094{ 8095 int session = chain->sessionId(); 8096 int16_t *buffer = mMixBuffer; 8097 bool ownsBuffer = false; 8098 8099 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8100 if (session > 0) { 8101 // Only one effect chain can be present in direct output thread and it uses 8102 // the mix buffer as input 8103 if (mType != DIRECT) { 8104 size_t numSamples = mNormalFrameCount * mChannelCount; 8105 buffer = new int16_t[numSamples]; 8106 memset(buffer, 0, numSamples * sizeof(int16_t)); 8107 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8108 ownsBuffer = true; 8109 } 8110 8111 // Attach all tracks with same session ID to this chain. 8112 for (size_t i = 0; i < mTracks.size(); ++i) { 8113 sp<Track> track = mTracks[i]; 8114 if (session == track->sessionId()) { 8115 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8116 buffer); 8117 track->setMainBuffer(buffer); 8118 chain->incTrackCnt(); 8119 } 8120 } 8121 8122 // indicate all active tracks in the chain 8123 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8124 sp<Track> track = mActiveTracks[i].promote(); 8125 if (track == 0) continue; 8126 if (session == track->sessionId()) { 8127 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8128 chain->incActiveTrackCnt(); 8129 } 8130 } 8131 } 8132 8133 chain->setInBuffer(buffer, ownsBuffer); 8134 chain->setOutBuffer(mMixBuffer); 8135 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8136 // chains list in order to be processed last as it contains output stage effects 8137 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8138 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8139 // after track specific effects and before output stage 8140 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8141 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8142 // Effect chain for other sessions are inserted at beginning of effect 8143 // chains list to be processed before output mix effects. Relative order between other 8144 // sessions is not important 8145 size_t size = mEffectChains.size(); 8146 size_t i = 0; 8147 for (i = 0; i < size; i++) { 8148 if (mEffectChains[i]->sessionId() < session) break; 8149 } 8150 mEffectChains.insertAt(chain, i); 8151 checkSuspendOnAddEffectChain_l(chain); 8152 8153 return NO_ERROR; 8154} 8155 8156size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8157{ 8158 int session = chain->sessionId(); 8159 8160 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8161 8162 for (size_t i = 0; i < mEffectChains.size(); i++) { 8163 if (chain == mEffectChains[i]) { 8164 mEffectChains.removeAt(i); 8165 // detach all active tracks from the chain 8166 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8167 sp<Track> track = mActiveTracks[i].promote(); 8168 if (track == 0) continue; 8169 if (session == track->sessionId()) { 8170 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8171 chain.get(), session); 8172 chain->decActiveTrackCnt(); 8173 } 8174 } 8175 8176 // detach all tracks with same session ID from this chain 8177 for (size_t i = 0; i < mTracks.size(); ++i) { 8178 sp<Track> track = mTracks[i]; 8179 if (session == track->sessionId()) { 8180 track->setMainBuffer(mMixBuffer); 8181 chain->decTrackCnt(); 8182 } 8183 } 8184 break; 8185 } 8186 } 8187 return mEffectChains.size(); 8188} 8189 8190status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8191 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8192{ 8193 Mutex::Autolock _l(mLock); 8194 return attachAuxEffect_l(track, EffectId); 8195} 8196 8197status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8198 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8199{ 8200 status_t status = NO_ERROR; 8201 8202 if (EffectId == 0) { 8203 track->setAuxBuffer(0, NULL); 8204 } else { 8205 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8206 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8207 if (effect != 0) { 8208 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8209 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8210 } else { 8211 status = INVALID_OPERATION; 8212 } 8213 } else { 8214 status = BAD_VALUE; 8215 } 8216 } 8217 return status; 8218} 8219 8220void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8221{ 8222 for (size_t i = 0; i < mTracks.size(); ++i) { 8223 sp<Track> track = mTracks[i]; 8224 if (track->auxEffectId() == effectId) { 8225 attachAuxEffect_l(track, 0); 8226 } 8227 } 8228} 8229 8230status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8231{ 8232 // only one chain per input thread 8233 if (mEffectChains.size() != 0) { 8234 return INVALID_OPERATION; 8235 } 8236 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8237 8238 chain->setInBuffer(NULL); 8239 chain->setOutBuffer(NULL); 8240 8241 checkSuspendOnAddEffectChain_l(chain); 8242 8243 mEffectChains.add(chain); 8244 8245 return NO_ERROR; 8246} 8247 8248size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8249{ 8250 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8251 ALOGW_IF(mEffectChains.size() != 1, 8252 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8253 chain.get(), mEffectChains.size(), this); 8254 if (mEffectChains.size() == 1) { 8255 mEffectChains.removeAt(0); 8256 } 8257 return 0; 8258} 8259 8260// ---------------------------------------------------------------------------- 8261// EffectModule implementation 8262// ---------------------------------------------------------------------------- 8263 8264#undef LOG_TAG 8265#define LOG_TAG "AudioFlinger::EffectModule" 8266 8267AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8268 const wp<AudioFlinger::EffectChain>& chain, 8269 effect_descriptor_t *desc, 8270 int id, 8271 int sessionId) 8272 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8273 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8274 mDescriptor(*desc), 8275 // mConfig is set by configure() and not used before then 8276 mEffectInterface(NULL), 8277 mStatus(NO_INIT), mState(IDLE), 8278 // mMaxDisableWaitCnt is set by configure() and not used before then 8279 // mDisableWaitCnt is set by process() and updateState() and not used before then 8280 mSuspended(false) 8281{ 8282 ALOGV("Constructor %p", this); 8283 int lStatus; 8284 8285 // create effect engine from effect factory 8286 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8287 8288 if (mStatus != NO_ERROR) { 8289 return; 8290 } 8291 lStatus = init(); 8292 if (lStatus < 0) { 8293 mStatus = lStatus; 8294 goto Error; 8295 } 8296 8297 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8298 return; 8299Error: 8300 EffectRelease(mEffectInterface); 8301 mEffectInterface = NULL; 8302 ALOGV("Constructor Error %d", mStatus); 8303} 8304 8305AudioFlinger::EffectModule::~EffectModule() 8306{ 8307 ALOGV("Destructor %p", this); 8308 if (mEffectInterface != NULL) { 8309 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8310 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8311 sp<ThreadBase> thread = mThread.promote(); 8312 if (thread != 0) { 8313 audio_stream_t *stream = thread->stream(); 8314 if (stream != NULL) { 8315 stream->remove_audio_effect(stream, mEffectInterface); 8316 } 8317 } 8318 } 8319 // release effect engine 8320 EffectRelease(mEffectInterface); 8321 } 8322} 8323 8324status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8325{ 8326 status_t status; 8327 8328 Mutex::Autolock _l(mLock); 8329 int priority = handle->priority(); 8330 size_t size = mHandles.size(); 8331 EffectHandle *controlHandle = NULL; 8332 size_t i; 8333 for (i = 0; i < size; i++) { 8334 EffectHandle *h = mHandles[i]; 8335 if (h == NULL || h->destroyed_l()) continue; 8336 // first non destroyed handle is considered in control 8337 if (controlHandle == NULL) 8338 controlHandle = h; 8339 if (h->priority() <= priority) break; 8340 } 8341 // if inserted in first place, move effect control from previous owner to this handle 8342 if (i == 0) { 8343 bool enabled = false; 8344 if (controlHandle != NULL) { 8345 enabled = controlHandle->enabled(); 8346 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8347 } 8348 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8349 status = NO_ERROR; 8350 } else { 8351 status = ALREADY_EXISTS; 8352 } 8353 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8354 mHandles.insertAt(handle, i); 8355 return status; 8356} 8357 8358size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8359{ 8360 Mutex::Autolock _l(mLock); 8361 size_t size = mHandles.size(); 8362 size_t i; 8363 for (i = 0; i < size; i++) { 8364 if (mHandles[i] == handle) break; 8365 } 8366 if (i == size) { 8367 return size; 8368 } 8369 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8370 8371 mHandles.removeAt(i); 8372 // if removed from first place, move effect control from this handle to next in line 8373 if (i == 0) { 8374 EffectHandle *h = controlHandle_l(); 8375 if (h != NULL) { 8376 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8377 } 8378 } 8379 8380 // Prevent calls to process() and other functions on effect interface from now on. 8381 // The effect engine will be released by the destructor when the last strong reference on 8382 // this object is released which can happen after next process is called. 8383 if (mHandles.size() == 0 && !mPinned) { 8384 mState = DESTROYED; 8385 } 8386 8387 return mHandles.size(); 8388} 8389 8390// must be called with EffectModule::mLock held 8391AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8392{ 8393 // the first valid handle in the list has control over the module 8394 for (size_t i = 0; i < mHandles.size(); i++) { 8395 EffectHandle *h = mHandles[i]; 8396 if (h != NULL && !h->destroyed_l()) { 8397 return h; 8398 } 8399 } 8400 8401 return NULL; 8402} 8403 8404size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8405{ 8406 ALOGV("disconnect() %p handle %p", this, handle); 8407 // keep a strong reference on this EffectModule to avoid calling the 8408 // destructor before we exit 8409 sp<EffectModule> keep(this); 8410 { 8411 sp<ThreadBase> thread = mThread.promote(); 8412 if (thread != 0) { 8413 thread->disconnectEffect(keep, handle, unpinIfLast); 8414 } 8415 } 8416 return mHandles.size(); 8417} 8418 8419void AudioFlinger::EffectModule::updateState() { 8420 Mutex::Autolock _l(mLock); 8421 8422 switch (mState) { 8423 case RESTART: 8424 reset_l(); 8425 // FALL THROUGH 8426 8427 case STARTING: 8428 // clear auxiliary effect input buffer for next accumulation 8429 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8430 memset(mConfig.inputCfg.buffer.raw, 8431 0, 8432 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8433 } 8434 start_l(); 8435 mState = ACTIVE; 8436 break; 8437 case STOPPING: 8438 stop_l(); 8439 mDisableWaitCnt = mMaxDisableWaitCnt; 8440 mState = STOPPED; 8441 break; 8442 case STOPPED: 8443 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8444 // turn off sequence. 8445 if (--mDisableWaitCnt == 0) { 8446 reset_l(); 8447 mState = IDLE; 8448 } 8449 break; 8450 default: //IDLE , ACTIVE, DESTROYED 8451 break; 8452 } 8453} 8454 8455void AudioFlinger::EffectModule::process() 8456{ 8457 Mutex::Autolock _l(mLock); 8458 8459 if (mState == DESTROYED || mEffectInterface == NULL || 8460 mConfig.inputCfg.buffer.raw == NULL || 8461 mConfig.outputCfg.buffer.raw == NULL) { 8462 return; 8463 } 8464 8465 if (isProcessEnabled()) { 8466 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8467 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8468 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8469 mConfig.inputCfg.buffer.s32, 8470 mConfig.inputCfg.buffer.frameCount/2); 8471 } 8472 8473 // do the actual processing in the effect engine 8474 int ret = (*mEffectInterface)->process(mEffectInterface, 8475 &mConfig.inputCfg.buffer, 8476 &mConfig.outputCfg.buffer); 8477 8478 // force transition to IDLE state when engine is ready 8479 if (mState == STOPPED && ret == -ENODATA) { 8480 mDisableWaitCnt = 1; 8481 } 8482 8483 // clear auxiliary effect input buffer for next accumulation 8484 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8485 memset(mConfig.inputCfg.buffer.raw, 0, 8486 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8487 } 8488 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8489 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8490 // If an insert effect is idle and input buffer is different from output buffer, 8491 // accumulate input onto output 8492 sp<EffectChain> chain = mChain.promote(); 8493 if (chain != 0 && chain->activeTrackCnt() != 0) { 8494 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8495 int16_t *in = mConfig.inputCfg.buffer.s16; 8496 int16_t *out = mConfig.outputCfg.buffer.s16; 8497 for (size_t i = 0; i < frameCnt; i++) { 8498 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8499 } 8500 } 8501 } 8502} 8503 8504void AudioFlinger::EffectModule::reset_l() 8505{ 8506 if (mEffectInterface == NULL) { 8507 return; 8508 } 8509 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8510} 8511 8512status_t AudioFlinger::EffectModule::configure() 8513{ 8514 if (mEffectInterface == NULL) { 8515 return NO_INIT; 8516 } 8517 8518 sp<ThreadBase> thread = mThread.promote(); 8519 if (thread == 0) { 8520 return DEAD_OBJECT; 8521 } 8522 8523 // TODO: handle configuration of effects replacing track process 8524 audio_channel_mask_t channelMask = thread->channelMask(); 8525 8526 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8527 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8528 } else { 8529 mConfig.inputCfg.channels = channelMask; 8530 } 8531 mConfig.outputCfg.channels = channelMask; 8532 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8533 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8534 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8535 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8536 mConfig.inputCfg.bufferProvider.cookie = NULL; 8537 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8538 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8539 mConfig.outputCfg.bufferProvider.cookie = NULL; 8540 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8541 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8542 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8543 // Insert effect: 8544 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8545 // always overwrites output buffer: input buffer == output buffer 8546 // - in other sessions: 8547 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8548 // other effect: overwrites output buffer: input buffer == output buffer 8549 // Auxiliary effect: 8550 // accumulates in output buffer: input buffer != output buffer 8551 // Therefore: accumulate <=> input buffer != output buffer 8552 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8553 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8554 } else { 8555 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8556 } 8557 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8558 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8559 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8560 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8561 8562 ALOGV("configure() %p thread %p buffer %p framecount %d", 8563 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8564 8565 status_t cmdStatus; 8566 uint32_t size = sizeof(int); 8567 status_t status = (*mEffectInterface)->command(mEffectInterface, 8568 EFFECT_CMD_SET_CONFIG, 8569 sizeof(effect_config_t), 8570 &mConfig, 8571 &size, 8572 &cmdStatus); 8573 if (status == 0) { 8574 status = cmdStatus; 8575 } 8576 8577 if (status == 0 && 8578 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8579 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8580 effect_param_t *p = (effect_param_t *)buf32; 8581 8582 p->psize = sizeof(uint32_t); 8583 p->vsize = sizeof(uint32_t); 8584 size = sizeof(int); 8585 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8586 8587 uint32_t latency = 0; 8588 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8589 if (pbt != NULL) { 8590 latency = pbt->latency_l(); 8591 } 8592 8593 *((int32_t *)p->data + 1)= latency; 8594 (*mEffectInterface)->command(mEffectInterface, 8595 EFFECT_CMD_SET_PARAM, 8596 sizeof(effect_param_t) + 8, 8597 &buf32, 8598 &size, 8599 &cmdStatus); 8600 } 8601 8602 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8603 (1000 * mConfig.outputCfg.buffer.frameCount); 8604 8605 return status; 8606} 8607 8608status_t AudioFlinger::EffectModule::init() 8609{ 8610 Mutex::Autolock _l(mLock); 8611 if (mEffectInterface == NULL) { 8612 return NO_INIT; 8613 } 8614 status_t cmdStatus; 8615 uint32_t size = sizeof(status_t); 8616 status_t status = (*mEffectInterface)->command(mEffectInterface, 8617 EFFECT_CMD_INIT, 8618 0, 8619 NULL, 8620 &size, 8621 &cmdStatus); 8622 if (status == 0) { 8623 status = cmdStatus; 8624 } 8625 return status; 8626} 8627 8628status_t AudioFlinger::EffectModule::start() 8629{ 8630 Mutex::Autolock _l(mLock); 8631 return start_l(); 8632} 8633 8634status_t AudioFlinger::EffectModule::start_l() 8635{ 8636 if (mEffectInterface == NULL) { 8637 return NO_INIT; 8638 } 8639 status_t cmdStatus; 8640 uint32_t size = sizeof(status_t); 8641 status_t status = (*mEffectInterface)->command(mEffectInterface, 8642 EFFECT_CMD_ENABLE, 8643 0, 8644 NULL, 8645 &size, 8646 &cmdStatus); 8647 if (status == 0) { 8648 status = cmdStatus; 8649 } 8650 if (status == 0 && 8651 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8652 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8653 sp<ThreadBase> thread = mThread.promote(); 8654 if (thread != 0) { 8655 audio_stream_t *stream = thread->stream(); 8656 if (stream != NULL) { 8657 stream->add_audio_effect(stream, mEffectInterface); 8658 } 8659 } 8660 } 8661 return status; 8662} 8663 8664status_t AudioFlinger::EffectModule::stop() 8665{ 8666 Mutex::Autolock _l(mLock); 8667 return stop_l(); 8668} 8669 8670status_t AudioFlinger::EffectModule::stop_l() 8671{ 8672 if (mEffectInterface == NULL) { 8673 return NO_INIT; 8674 } 8675 status_t cmdStatus; 8676 uint32_t size = sizeof(status_t); 8677 status_t status = (*mEffectInterface)->command(mEffectInterface, 8678 EFFECT_CMD_DISABLE, 8679 0, 8680 NULL, 8681 &size, 8682 &cmdStatus); 8683 if (status == 0) { 8684 status = cmdStatus; 8685 } 8686 if (status == 0 && 8687 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8688 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8689 sp<ThreadBase> thread = mThread.promote(); 8690 if (thread != 0) { 8691 audio_stream_t *stream = thread->stream(); 8692 if (stream != NULL) { 8693 stream->remove_audio_effect(stream, mEffectInterface); 8694 } 8695 } 8696 } 8697 return status; 8698} 8699 8700status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8701 uint32_t cmdSize, 8702 void *pCmdData, 8703 uint32_t *replySize, 8704 void *pReplyData) 8705{ 8706 Mutex::Autolock _l(mLock); 8707 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8708 8709 if (mState == DESTROYED || mEffectInterface == NULL) { 8710 return NO_INIT; 8711 } 8712 status_t status = (*mEffectInterface)->command(mEffectInterface, 8713 cmdCode, 8714 cmdSize, 8715 pCmdData, 8716 replySize, 8717 pReplyData); 8718 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8719 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8720 for (size_t i = 1; i < mHandles.size(); i++) { 8721 EffectHandle *h = mHandles[i]; 8722 if (h != NULL && !h->destroyed_l()) { 8723 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8724 } 8725 } 8726 } 8727 return status; 8728} 8729 8730status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8731{ 8732 Mutex::Autolock _l(mLock); 8733 return setEnabled_l(enabled); 8734} 8735 8736// must be called with EffectModule::mLock held 8737status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8738{ 8739 8740 ALOGV("setEnabled %p enabled %d", this, enabled); 8741 8742 if (enabled != isEnabled()) { 8743 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8744 if (enabled && status != NO_ERROR) { 8745 return status; 8746 } 8747 8748 switch (mState) { 8749 // going from disabled to enabled 8750 case IDLE: 8751 mState = STARTING; 8752 break; 8753 case STOPPED: 8754 mState = RESTART; 8755 break; 8756 case STOPPING: 8757 mState = ACTIVE; 8758 break; 8759 8760 // going from enabled to disabled 8761 case RESTART: 8762 mState = STOPPED; 8763 break; 8764 case STARTING: 8765 mState = IDLE; 8766 break; 8767 case ACTIVE: 8768 mState = STOPPING; 8769 break; 8770 case DESTROYED: 8771 return NO_ERROR; // simply ignore as we are being destroyed 8772 } 8773 for (size_t i = 1; i < mHandles.size(); i++) { 8774 EffectHandle *h = mHandles[i]; 8775 if (h != NULL && !h->destroyed_l()) { 8776 h->setEnabled(enabled); 8777 } 8778 } 8779 } 8780 return NO_ERROR; 8781} 8782 8783bool AudioFlinger::EffectModule::isEnabled() const 8784{ 8785 switch (mState) { 8786 case RESTART: 8787 case STARTING: 8788 case ACTIVE: 8789 return true; 8790 case IDLE: 8791 case STOPPING: 8792 case STOPPED: 8793 case DESTROYED: 8794 default: 8795 return false; 8796 } 8797} 8798 8799bool AudioFlinger::EffectModule::isProcessEnabled() const 8800{ 8801 switch (mState) { 8802 case RESTART: 8803 case ACTIVE: 8804 case STOPPING: 8805 case STOPPED: 8806 return true; 8807 case IDLE: 8808 case STARTING: 8809 case DESTROYED: 8810 default: 8811 return false; 8812 } 8813} 8814 8815status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8816{ 8817 Mutex::Autolock _l(mLock); 8818 status_t status = NO_ERROR; 8819 8820 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8821 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8822 if (isProcessEnabled() && 8823 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8824 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8825 status_t cmdStatus; 8826 uint32_t volume[2]; 8827 uint32_t *pVolume = NULL; 8828 uint32_t size = sizeof(volume); 8829 volume[0] = *left; 8830 volume[1] = *right; 8831 if (controller) { 8832 pVolume = volume; 8833 } 8834 status = (*mEffectInterface)->command(mEffectInterface, 8835 EFFECT_CMD_SET_VOLUME, 8836 size, 8837 volume, 8838 &size, 8839 pVolume); 8840 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8841 *left = volume[0]; 8842 *right = volume[1]; 8843 } 8844 } 8845 return status; 8846} 8847 8848status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8849{ 8850 if (device == AUDIO_DEVICE_NONE) { 8851 return NO_ERROR; 8852 } 8853 8854 Mutex::Autolock _l(mLock); 8855 status_t status = NO_ERROR; 8856 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8857 status_t cmdStatus; 8858 uint32_t size = sizeof(status_t); 8859 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8860 EFFECT_CMD_SET_INPUT_DEVICE; 8861 status = (*mEffectInterface)->command(mEffectInterface, 8862 cmd, 8863 sizeof(uint32_t), 8864 &device, 8865 &size, 8866 &cmdStatus); 8867 } 8868 return status; 8869} 8870 8871status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8872{ 8873 Mutex::Autolock _l(mLock); 8874 status_t status = NO_ERROR; 8875 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8876 status_t cmdStatus; 8877 uint32_t size = sizeof(status_t); 8878 status = (*mEffectInterface)->command(mEffectInterface, 8879 EFFECT_CMD_SET_AUDIO_MODE, 8880 sizeof(audio_mode_t), 8881 &mode, 8882 &size, 8883 &cmdStatus); 8884 if (status == NO_ERROR) { 8885 status = cmdStatus; 8886 } 8887 } 8888 return status; 8889} 8890 8891status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8892{ 8893 Mutex::Autolock _l(mLock); 8894 status_t status = NO_ERROR; 8895 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8896 uint32_t size = 0; 8897 status = (*mEffectInterface)->command(mEffectInterface, 8898 EFFECT_CMD_SET_AUDIO_SOURCE, 8899 sizeof(audio_source_t), 8900 &source, 8901 &size, 8902 NULL); 8903 } 8904 return status; 8905} 8906 8907void AudioFlinger::EffectModule::setSuspended(bool suspended) 8908{ 8909 Mutex::Autolock _l(mLock); 8910 mSuspended = suspended; 8911} 8912 8913bool AudioFlinger::EffectModule::suspended() const 8914{ 8915 Mutex::Autolock _l(mLock); 8916 return mSuspended; 8917} 8918 8919bool AudioFlinger::EffectModule::purgeHandles() 8920{ 8921 bool enabled = false; 8922 Mutex::Autolock _l(mLock); 8923 for (size_t i = 0; i < mHandles.size(); i++) { 8924 EffectHandle *handle = mHandles[i]; 8925 if (handle != NULL && !handle->destroyed_l()) { 8926 handle->effect().clear(); 8927 if (handle->hasControl()) { 8928 enabled = handle->enabled(); 8929 } 8930 } 8931 } 8932 return enabled; 8933} 8934 8935void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8936{ 8937 const size_t SIZE = 256; 8938 char buffer[SIZE]; 8939 String8 result; 8940 8941 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8942 result.append(buffer); 8943 8944 bool locked = tryLock(mLock); 8945 // failed to lock - AudioFlinger is probably deadlocked 8946 if (!locked) { 8947 result.append("\t\tCould not lock Fx mutex:\n"); 8948 } 8949 8950 result.append("\t\tSession Status State Engine:\n"); 8951 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8952 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8953 result.append(buffer); 8954 8955 result.append("\t\tDescriptor:\n"); 8956 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8957 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8958 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 8959 mDescriptor.uuid.node[2], 8960 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8961 result.append(buffer); 8962 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8963 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 8964 mDescriptor.type.timeHiAndVersion, 8965 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 8966 mDescriptor.type.node[2], 8967 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8968 result.append(buffer); 8969 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8970 mDescriptor.apiVersion, 8971 mDescriptor.flags); 8972 result.append(buffer); 8973 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8974 mDescriptor.name); 8975 result.append(buffer); 8976 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8977 mDescriptor.implementor); 8978 result.append(buffer); 8979 8980 result.append("\t\t- Input configuration:\n"); 8981 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8982 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8983 (uint32_t)mConfig.inputCfg.buffer.raw, 8984 mConfig.inputCfg.buffer.frameCount, 8985 mConfig.inputCfg.samplingRate, 8986 mConfig.inputCfg.channels, 8987 mConfig.inputCfg.format); 8988 result.append(buffer); 8989 8990 result.append("\t\t- Output configuration:\n"); 8991 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8992 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8993 (uint32_t)mConfig.outputCfg.buffer.raw, 8994 mConfig.outputCfg.buffer.frameCount, 8995 mConfig.outputCfg.samplingRate, 8996 mConfig.outputCfg.channels, 8997 mConfig.outputCfg.format); 8998 result.append(buffer); 8999 9000 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 9001 result.append(buffer); 9002 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 9003 for (size_t i = 0; i < mHandles.size(); ++i) { 9004 EffectHandle *handle = mHandles[i]; 9005 if (handle != NULL && !handle->destroyed_l()) { 9006 handle->dump(buffer, SIZE); 9007 result.append(buffer); 9008 } 9009 } 9010 9011 result.append("\n"); 9012 9013 write(fd, result.string(), result.length()); 9014 9015 if (locked) { 9016 mLock.unlock(); 9017 } 9018} 9019 9020// ---------------------------------------------------------------------------- 9021// EffectHandle implementation 9022// ---------------------------------------------------------------------------- 9023 9024#undef LOG_TAG 9025#define LOG_TAG "AudioFlinger::EffectHandle" 9026 9027AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9028 const sp<AudioFlinger::Client>& client, 9029 const sp<IEffectClient>& effectClient, 9030 int32_t priority) 9031 : BnEffect(), 9032 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9033 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9034{ 9035 ALOGV("constructor %p", this); 9036 9037 if (client == 0) { 9038 return; 9039 } 9040 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9041 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9042 if (mCblkMemory != 0) { 9043 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9044 9045 if (mCblk != NULL) { 9046 new(mCblk) effect_param_cblk_t(); 9047 mBuffer = (uint8_t *)mCblk + bufOffset; 9048 } 9049 } else { 9050 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9051 sizeof(effect_param_cblk_t)); 9052 return; 9053 } 9054} 9055 9056AudioFlinger::EffectHandle::~EffectHandle() 9057{ 9058 ALOGV("Destructor %p", this); 9059 9060 if (mEffect == 0) { 9061 mDestroyed = true; 9062 return; 9063 } 9064 mEffect->lock(); 9065 mDestroyed = true; 9066 mEffect->unlock(); 9067 disconnect(false); 9068} 9069 9070status_t AudioFlinger::EffectHandle::enable() 9071{ 9072 ALOGV("enable %p", this); 9073 if (!mHasControl) return INVALID_OPERATION; 9074 if (mEffect == 0) return DEAD_OBJECT; 9075 9076 if (mEnabled) { 9077 return NO_ERROR; 9078 } 9079 9080 mEnabled = true; 9081 9082 sp<ThreadBase> thread = mEffect->thread().promote(); 9083 if (thread != 0) { 9084 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9085 } 9086 9087 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9088 if (mEffect->suspended()) { 9089 return NO_ERROR; 9090 } 9091 9092 status_t status = mEffect->setEnabled(true); 9093 if (status != NO_ERROR) { 9094 if (thread != 0) { 9095 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9096 } 9097 mEnabled = false; 9098 } 9099 return status; 9100} 9101 9102status_t AudioFlinger::EffectHandle::disable() 9103{ 9104 ALOGV("disable %p", this); 9105 if (!mHasControl) return INVALID_OPERATION; 9106 if (mEffect == 0) return DEAD_OBJECT; 9107 9108 if (!mEnabled) { 9109 return NO_ERROR; 9110 } 9111 mEnabled = false; 9112 9113 if (mEffect->suspended()) { 9114 return NO_ERROR; 9115 } 9116 9117 status_t status = mEffect->setEnabled(false); 9118 9119 sp<ThreadBase> thread = mEffect->thread().promote(); 9120 if (thread != 0) { 9121 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9122 } 9123 9124 return status; 9125} 9126 9127void AudioFlinger::EffectHandle::disconnect() 9128{ 9129 disconnect(true); 9130} 9131 9132void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9133{ 9134 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9135 if (mEffect == 0) { 9136 return; 9137 } 9138 // restore suspended effects if the disconnected handle was enabled and the last one. 9139 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9140 sp<ThreadBase> thread = mEffect->thread().promote(); 9141 if (thread != 0) { 9142 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9143 } 9144 } 9145 9146 // release sp on module => module destructor can be called now 9147 mEffect.clear(); 9148 if (mClient != 0) { 9149 if (mCblk != NULL) { 9150 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9151 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9152 } 9153 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9154 // Client destructor must run with AudioFlinger mutex locked 9155 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9156 mClient.clear(); 9157 } 9158} 9159 9160status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9161 uint32_t cmdSize, 9162 void *pCmdData, 9163 uint32_t *replySize, 9164 void *pReplyData) 9165{ 9166 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9167 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9168 9169 // only get parameter command is permitted for applications not controlling the effect 9170 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9171 return INVALID_OPERATION; 9172 } 9173 if (mEffect == 0) return DEAD_OBJECT; 9174 if (mClient == 0) return INVALID_OPERATION; 9175 9176 // handle commands that are not forwarded transparently to effect engine 9177 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9178 // No need to trylock() here as this function is executed in the binder thread serving a 9179 // particular client process: no risk to block the whole media server process or mixer 9180 // threads if we are stuck here 9181 Mutex::Autolock _l(mCblk->lock); 9182 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9183 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9184 mCblk->serverIndex = 0; 9185 mCblk->clientIndex = 0; 9186 return BAD_VALUE; 9187 } 9188 status_t status = NO_ERROR; 9189 while (mCblk->serverIndex < mCblk->clientIndex) { 9190 int reply; 9191 uint32_t rsize = sizeof(int); 9192 int *p = (int *)(mBuffer + mCblk->serverIndex); 9193 int size = *p++; 9194 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9195 ALOGW("command(): invalid parameter block size"); 9196 break; 9197 } 9198 effect_param_t *param = (effect_param_t *)p; 9199 if (param->psize == 0 || param->vsize == 0) { 9200 ALOGW("command(): null parameter or value size"); 9201 mCblk->serverIndex += size; 9202 continue; 9203 } 9204 uint32_t psize = sizeof(effect_param_t) + 9205 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9206 param->vsize; 9207 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9208 psize, 9209 p, 9210 &rsize, 9211 &reply); 9212 // stop at first error encountered 9213 if (ret != NO_ERROR) { 9214 status = ret; 9215 *(int *)pReplyData = reply; 9216 break; 9217 } else if (reply != NO_ERROR) { 9218 *(int *)pReplyData = reply; 9219 break; 9220 } 9221 mCblk->serverIndex += size; 9222 } 9223 mCblk->serverIndex = 0; 9224 mCblk->clientIndex = 0; 9225 return status; 9226 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9227 *(int *)pReplyData = NO_ERROR; 9228 return enable(); 9229 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9230 *(int *)pReplyData = NO_ERROR; 9231 return disable(); 9232 } 9233 9234 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9235} 9236 9237void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9238{ 9239 ALOGV("setControl %p control %d", this, hasControl); 9240 9241 mHasControl = hasControl; 9242 mEnabled = enabled; 9243 9244 if (signal && mEffectClient != 0) { 9245 mEffectClient->controlStatusChanged(hasControl); 9246 } 9247} 9248 9249void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9250 uint32_t cmdSize, 9251 void *pCmdData, 9252 uint32_t replySize, 9253 void *pReplyData) 9254{ 9255 if (mEffectClient != 0) { 9256 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9257 } 9258} 9259 9260 9261 9262void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9263{ 9264 if (mEffectClient != 0) { 9265 mEffectClient->enableStatusChanged(enabled); 9266 } 9267} 9268 9269status_t AudioFlinger::EffectHandle::onTransact( 9270 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9271{ 9272 return BnEffect::onTransact(code, data, reply, flags); 9273} 9274 9275 9276void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9277{ 9278 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9279 9280 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9281 (mClient == 0) ? getpid_cached : mClient->pid(), 9282 mPriority, 9283 mHasControl, 9284 !locked, 9285 mCblk ? mCblk->clientIndex : 0, 9286 mCblk ? mCblk->serverIndex : 0 9287 ); 9288 9289 if (locked) { 9290 mCblk->lock.unlock(); 9291 } 9292} 9293 9294#undef LOG_TAG 9295#define LOG_TAG "AudioFlinger::EffectChain" 9296 9297AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9298 int sessionId) 9299 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9300 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9301 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9302{ 9303 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9304 if (thread == NULL) { 9305 return; 9306 } 9307 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9308 thread->frameCount(); 9309} 9310 9311AudioFlinger::EffectChain::~EffectChain() 9312{ 9313 if (mOwnInBuffer) { 9314 delete mInBuffer; 9315 } 9316 9317} 9318 9319// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9320sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9321 effect_descriptor_t *descriptor) 9322{ 9323 size_t size = mEffects.size(); 9324 9325 for (size_t i = 0; i < size; i++) { 9326 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9327 return mEffects[i]; 9328 } 9329 } 9330 return 0; 9331} 9332 9333// getEffectFromId_l() must be called with ThreadBase::mLock held 9334sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9335{ 9336 size_t size = mEffects.size(); 9337 9338 for (size_t i = 0; i < size; i++) { 9339 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9340 if (id == 0 || mEffects[i]->id() == id) { 9341 return mEffects[i]; 9342 } 9343 } 9344 return 0; 9345} 9346 9347// getEffectFromType_l() must be called with ThreadBase::mLock held 9348sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9349 const effect_uuid_t *type) 9350{ 9351 size_t size = mEffects.size(); 9352 9353 for (size_t i = 0; i < size; i++) { 9354 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9355 return mEffects[i]; 9356 } 9357 } 9358 return 0; 9359} 9360 9361void AudioFlinger::EffectChain::clearInputBuffer() 9362{ 9363 Mutex::Autolock _l(mLock); 9364 sp<ThreadBase> thread = mThread.promote(); 9365 if (thread == 0) { 9366 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9367 return; 9368 } 9369 clearInputBuffer_l(thread); 9370} 9371 9372// Must be called with EffectChain::mLock locked 9373void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9374{ 9375 size_t numSamples = thread->frameCount() * thread->channelCount(); 9376 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9377 9378} 9379 9380// Must be called with EffectChain::mLock locked 9381void AudioFlinger::EffectChain::process_l() 9382{ 9383 sp<ThreadBase> thread = mThread.promote(); 9384 if (thread == 0) { 9385 ALOGW("process_l(): cannot promote mixer thread"); 9386 return; 9387 } 9388 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9389 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9390 // always process effects unless no more tracks are on the session and the effect tail 9391 // has been rendered 9392 bool doProcess = true; 9393 if (!isGlobalSession) { 9394 bool tracksOnSession = (trackCnt() != 0); 9395 9396 if (!tracksOnSession && mTailBufferCount == 0) { 9397 doProcess = false; 9398 } 9399 9400 if (activeTrackCnt() == 0) { 9401 // if no track is active and the effect tail has not been rendered, 9402 // the input buffer must be cleared here as the mixer process will not do it 9403 if (tracksOnSession || mTailBufferCount > 0) { 9404 clearInputBuffer_l(thread); 9405 if (mTailBufferCount > 0) { 9406 mTailBufferCount--; 9407 } 9408 } 9409 } 9410 } 9411 9412 size_t size = mEffects.size(); 9413 if (doProcess) { 9414 for (size_t i = 0; i < size; i++) { 9415 mEffects[i]->process(); 9416 } 9417 } 9418 for (size_t i = 0; i < size; i++) { 9419 mEffects[i]->updateState(); 9420 } 9421} 9422 9423// addEffect_l() must be called with PlaybackThread::mLock held 9424status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9425{ 9426 effect_descriptor_t desc = effect->desc(); 9427 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9428 9429 Mutex::Autolock _l(mLock); 9430 effect->setChain(this); 9431 sp<ThreadBase> thread = mThread.promote(); 9432 if (thread == 0) { 9433 return NO_INIT; 9434 } 9435 effect->setThread(thread); 9436 9437 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9438 // Auxiliary effects are inserted at the beginning of mEffects vector as 9439 // they are processed first and accumulated in chain input buffer 9440 mEffects.insertAt(effect, 0); 9441 9442 // the input buffer for auxiliary effect contains mono samples in 9443 // 32 bit format. This is to avoid saturation in AudoMixer 9444 // accumulation stage. Saturation is done in EffectModule::process() before 9445 // calling the process in effect engine 9446 size_t numSamples = thread->frameCount(); 9447 int32_t *buffer = new int32_t[numSamples]; 9448 memset(buffer, 0, numSamples * sizeof(int32_t)); 9449 effect->setInBuffer((int16_t *)buffer); 9450 // auxiliary effects output samples to chain input buffer for further processing 9451 // by insert effects 9452 effect->setOutBuffer(mInBuffer); 9453 } else { 9454 // Insert effects are inserted at the end of mEffects vector as they are processed 9455 // after track and auxiliary effects. 9456 // Insert effect order as a function of indicated preference: 9457 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9458 // another effect is present 9459 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9460 // last effect claiming first position 9461 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9462 // first effect claiming last position 9463 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9464 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9465 // already present 9466 9467 size_t size = mEffects.size(); 9468 size_t idx_insert = size; 9469 ssize_t idx_insert_first = -1; 9470 ssize_t idx_insert_last = -1; 9471 9472 for (size_t i = 0; i < size; i++) { 9473 effect_descriptor_t d = mEffects[i]->desc(); 9474 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9475 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9476 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9477 // check invalid effect chaining combinations 9478 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9479 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9480 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9481 desc.name, d.name); 9482 return INVALID_OPERATION; 9483 } 9484 // remember position of first insert effect and by default 9485 // select this as insert position for new effect 9486 if (idx_insert == size) { 9487 idx_insert = i; 9488 } 9489 // remember position of last insert effect claiming 9490 // first position 9491 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9492 idx_insert_first = i; 9493 } 9494 // remember position of first insert effect claiming 9495 // last position 9496 if (iPref == EFFECT_FLAG_INSERT_LAST && 9497 idx_insert_last == -1) { 9498 idx_insert_last = i; 9499 } 9500 } 9501 } 9502 9503 // modify idx_insert from first position if needed 9504 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9505 if (idx_insert_last != -1) { 9506 idx_insert = idx_insert_last; 9507 } else { 9508 idx_insert = size; 9509 } 9510 } else { 9511 if (idx_insert_first != -1) { 9512 idx_insert = idx_insert_first + 1; 9513 } 9514 } 9515 9516 // always read samples from chain input buffer 9517 effect->setInBuffer(mInBuffer); 9518 9519 // if last effect in the chain, output samples to chain 9520 // output buffer, otherwise to chain input buffer 9521 if (idx_insert == size) { 9522 if (idx_insert != 0) { 9523 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9524 mEffects[idx_insert-1]->configure(); 9525 } 9526 effect->setOutBuffer(mOutBuffer); 9527 } else { 9528 effect->setOutBuffer(mInBuffer); 9529 } 9530 mEffects.insertAt(effect, idx_insert); 9531 9532 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9533 idx_insert); 9534 } 9535 effect->configure(); 9536 return NO_ERROR; 9537} 9538 9539// removeEffect_l() must be called with PlaybackThread::mLock held 9540size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9541{ 9542 Mutex::Autolock _l(mLock); 9543 size_t size = mEffects.size(); 9544 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9545 9546 for (size_t i = 0; i < size; i++) { 9547 if (effect == mEffects[i]) { 9548 // calling stop here will remove pre-processing effect from the audio HAL. 9549 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9550 // the middle of a read from audio HAL 9551 if (mEffects[i]->state() == EffectModule::ACTIVE || 9552 mEffects[i]->state() == EffectModule::STOPPING) { 9553 mEffects[i]->stop(); 9554 } 9555 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9556 delete[] effect->inBuffer(); 9557 } else { 9558 if (i == size - 1 && i != 0) { 9559 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9560 mEffects[i - 1]->configure(); 9561 } 9562 } 9563 mEffects.removeAt(i); 9564 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9565 this, i); 9566 break; 9567 } 9568 } 9569 9570 return mEffects.size(); 9571} 9572 9573// setDevice_l() must be called with PlaybackThread::mLock held 9574void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9575{ 9576 size_t size = mEffects.size(); 9577 for (size_t i = 0; i < size; i++) { 9578 mEffects[i]->setDevice(device); 9579 } 9580} 9581 9582// setMode_l() must be called with PlaybackThread::mLock held 9583void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9584{ 9585 size_t size = mEffects.size(); 9586 for (size_t i = 0; i < size; i++) { 9587 mEffects[i]->setMode(mode); 9588 } 9589} 9590 9591// setAudioSource_l() must be called with PlaybackThread::mLock held 9592void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9593{ 9594 size_t size = mEffects.size(); 9595 for (size_t i = 0; i < size; i++) { 9596 mEffects[i]->setAudioSource(source); 9597 } 9598} 9599 9600// setVolume_l() must be called with PlaybackThread::mLock held 9601bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9602{ 9603 uint32_t newLeft = *left; 9604 uint32_t newRight = *right; 9605 bool hasControl = false; 9606 int ctrlIdx = -1; 9607 size_t size = mEffects.size(); 9608 9609 // first update volume controller 9610 for (size_t i = size; i > 0; i--) { 9611 if (mEffects[i - 1]->isProcessEnabled() && 9612 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9613 ctrlIdx = i - 1; 9614 hasControl = true; 9615 break; 9616 } 9617 } 9618 9619 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9620 if (hasControl) { 9621 *left = mNewLeftVolume; 9622 *right = mNewRightVolume; 9623 } 9624 return hasControl; 9625 } 9626 9627 mVolumeCtrlIdx = ctrlIdx; 9628 mLeftVolume = newLeft; 9629 mRightVolume = newRight; 9630 9631 // second get volume update from volume controller 9632 if (ctrlIdx >= 0) { 9633 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9634 mNewLeftVolume = newLeft; 9635 mNewRightVolume = newRight; 9636 } 9637 // then indicate volume to all other effects in chain. 9638 // Pass altered volume to effects before volume controller 9639 // and requested volume to effects after controller 9640 uint32_t lVol = newLeft; 9641 uint32_t rVol = newRight; 9642 9643 for (size_t i = 0; i < size; i++) { 9644 if ((int)i == ctrlIdx) continue; 9645 // this also works for ctrlIdx == -1 when there is no volume controller 9646 if ((int)i > ctrlIdx) { 9647 lVol = *left; 9648 rVol = *right; 9649 } 9650 mEffects[i]->setVolume(&lVol, &rVol, false); 9651 } 9652 *left = newLeft; 9653 *right = newRight; 9654 9655 return hasControl; 9656} 9657 9658void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9659{ 9660 const size_t SIZE = 256; 9661 char buffer[SIZE]; 9662 String8 result; 9663 9664 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9665 result.append(buffer); 9666 9667 bool locked = tryLock(mLock); 9668 // failed to lock - AudioFlinger is probably deadlocked 9669 if (!locked) { 9670 result.append("\tCould not lock mutex:\n"); 9671 } 9672 9673 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9674 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9675 mEffects.size(), 9676 (uint32_t)mInBuffer, 9677 (uint32_t)mOutBuffer, 9678 mActiveTrackCnt); 9679 result.append(buffer); 9680 write(fd, result.string(), result.size()); 9681 9682 for (size_t i = 0; i < mEffects.size(); ++i) { 9683 sp<EffectModule> effect = mEffects[i]; 9684 if (effect != 0) { 9685 effect->dump(fd, args); 9686 } 9687 } 9688 9689 if (locked) { 9690 mLock.unlock(); 9691 } 9692} 9693 9694// must be called with ThreadBase::mLock held 9695void AudioFlinger::EffectChain::setEffectSuspended_l( 9696 const effect_uuid_t *type, bool suspend) 9697{ 9698 sp<SuspendedEffectDesc> desc; 9699 // use effect type UUID timelow as key as there is no real risk of identical 9700 // timeLow fields among effect type UUIDs. 9701 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9702 if (suspend) { 9703 if (index >= 0) { 9704 desc = mSuspendedEffects.valueAt(index); 9705 } else { 9706 desc = new SuspendedEffectDesc(); 9707 desc->mType = *type; 9708 mSuspendedEffects.add(type->timeLow, desc); 9709 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9710 } 9711 if (desc->mRefCount++ == 0) { 9712 sp<EffectModule> effect = getEffectIfEnabled(type); 9713 if (effect != 0) { 9714 desc->mEffect = effect; 9715 effect->setSuspended(true); 9716 effect->setEnabled(false); 9717 } 9718 } 9719 } else { 9720 if (index < 0) { 9721 return; 9722 } 9723 desc = mSuspendedEffects.valueAt(index); 9724 if (desc->mRefCount <= 0) { 9725 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9726 desc->mRefCount = 1; 9727 } 9728 if (--desc->mRefCount == 0) { 9729 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9730 if (desc->mEffect != 0) { 9731 sp<EffectModule> effect = desc->mEffect.promote(); 9732 if (effect != 0) { 9733 effect->setSuspended(false); 9734 effect->lock(); 9735 EffectHandle *handle = effect->controlHandle_l(); 9736 if (handle != NULL && !handle->destroyed_l()) { 9737 effect->setEnabled_l(handle->enabled()); 9738 } 9739 effect->unlock(); 9740 } 9741 desc->mEffect.clear(); 9742 } 9743 mSuspendedEffects.removeItemsAt(index); 9744 } 9745 } 9746} 9747 9748// must be called with ThreadBase::mLock held 9749void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9750{ 9751 sp<SuspendedEffectDesc> desc; 9752 9753 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9754 if (suspend) { 9755 if (index >= 0) { 9756 desc = mSuspendedEffects.valueAt(index); 9757 } else { 9758 desc = new SuspendedEffectDesc(); 9759 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9760 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9761 } 9762 if (desc->mRefCount++ == 0) { 9763 Vector< sp<EffectModule> > effects; 9764 getSuspendEligibleEffects(effects); 9765 for (size_t i = 0; i < effects.size(); i++) { 9766 setEffectSuspended_l(&effects[i]->desc().type, true); 9767 } 9768 } 9769 } else { 9770 if (index < 0) { 9771 return; 9772 } 9773 desc = mSuspendedEffects.valueAt(index); 9774 if (desc->mRefCount <= 0) { 9775 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9776 desc->mRefCount = 1; 9777 } 9778 if (--desc->mRefCount == 0) { 9779 Vector<const effect_uuid_t *> types; 9780 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9781 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9782 continue; 9783 } 9784 types.add(&mSuspendedEffects.valueAt(i)->mType); 9785 } 9786 for (size_t i = 0; i < types.size(); i++) { 9787 setEffectSuspended_l(types[i], false); 9788 } 9789 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9790 mSuspendedEffects.keyAt(index)); 9791 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9792 } 9793 } 9794} 9795 9796 9797// The volume effect is used for automated tests only 9798#ifndef OPENSL_ES_H_ 9799static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9800 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9801const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9802#endif //OPENSL_ES_H_ 9803 9804bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9805{ 9806 // auxiliary effects and visualizer are never suspended on output mix 9807 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9808 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9809 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9810 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9811 return false; 9812 } 9813 return true; 9814} 9815 9816void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9817 Vector< sp<AudioFlinger::EffectModule> > &effects) 9818{ 9819 effects.clear(); 9820 for (size_t i = 0; i < mEffects.size(); i++) { 9821 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9822 effects.add(mEffects[i]); 9823 } 9824 } 9825} 9826 9827sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9828 const effect_uuid_t *type) 9829{ 9830 sp<EffectModule> effect = getEffectFromType_l(type); 9831 return effect != 0 && effect->isEnabled() ? effect : 0; 9832} 9833 9834void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9835 bool enabled) 9836{ 9837 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9838 if (enabled) { 9839 if (index < 0) { 9840 // if the effect is not suspend check if all effects are suspended 9841 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9842 if (index < 0) { 9843 return; 9844 } 9845 if (!isEffectEligibleForSuspend(effect->desc())) { 9846 return; 9847 } 9848 setEffectSuspended_l(&effect->desc().type, enabled); 9849 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9850 if (index < 0) { 9851 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9852 return; 9853 } 9854 } 9855 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9856 effect->desc().type.timeLow); 9857 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9858 // if effect is requested to suspended but was not yet enabled, supend it now. 9859 if (desc->mEffect == 0) { 9860 desc->mEffect = effect; 9861 effect->setEnabled(false); 9862 effect->setSuspended(true); 9863 } 9864 } else { 9865 if (index < 0) { 9866 return; 9867 } 9868 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9869 effect->desc().type.timeLow); 9870 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9871 desc->mEffect.clear(); 9872 effect->setSuspended(false); 9873 } 9874} 9875 9876#undef LOG_TAG 9877#define LOG_TAG "AudioFlinger" 9878 9879// ---------------------------------------------------------------------------- 9880 9881status_t AudioFlinger::onTransact( 9882 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9883{ 9884 return BnAudioFlinger::onTransact(code, data, reply, flags); 9885} 9886 9887}; // namespace android 9888