AudioFlinger.cpp revision bb4350d3b9e9485ae59e084de270f86aecef8066
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// ---------------------------------------------------------------------------- 169 170#ifdef ADD_BATTERY_DATA 171// To collect the amplifier usage 172static void addBatteryData(uint32_t params) { 173 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 174 if (service == NULL) { 175 // it already logged 176 return; 177 } 178 179 service->addBatteryData(params); 180} 181#endif 182 183static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 184{ 185 const hw_module_t *mod; 186 int rc; 187 188 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 189 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 rc = audio_hw_device_open(mod, dev); 195 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 196 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 197 if (rc) { 198 goto out; 199 } 200 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 201 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 202 rc = BAD_VALUE; 203 goto out; 204 } 205 return 0; 206 207out: 208 *dev = NULL; 209 return rc; 210} 211 212// ---------------------------------------------------------------------------- 213 214AudioFlinger::AudioFlinger() 215 : BnAudioFlinger(), 216 mPrimaryHardwareDev(NULL), 217 mHardwareStatus(AUDIO_HW_IDLE), 218 mMasterVolume(1.0f), 219 mMasterVolumeSW(1.0f), 220 mMasterVolumeSupportLvl(MVS_NONE), 221 mMasterMute(false), 222 mNextUniqueId(1), 223 mMode(AUDIO_MODE_INVALID), 224 mBtNrecIsOff(false) 225{ 226} 227 228void AudioFlinger::onFirstRef() 229{ 230 int rc = 0; 231 232 Mutex::Autolock _l(mLock); 233 234 /* TODO: move all this work into an Init() function */ 235 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 236 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 237 uint32_t int_val; 238 if (1 == sscanf(val_str, "%u", &int_val)) { 239 mStandbyTimeInNsecs = milliseconds(int_val); 240 ALOGI("Using %u mSec as standby time.", int_val); 241 } else { 242 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 243 ALOGI("Using default %u mSec as standby time.", 244 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 245 } 246 } 247 248 mMode = AUDIO_MODE_NORMAL; 249} 250 251AudioFlinger::~AudioFlinger() 252{ 253 while (!mRecordThreads.isEmpty()) { 254 // closeInput() will remove first entry from mRecordThreads 255 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 256 } 257 while (!mPlaybackThreads.isEmpty()) { 258 // closeOutput() will remove first entry from mPlaybackThreads 259 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 260 } 261 262 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 263 // no mHardwareLock needed, as there are no other references to this 264 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 265 delete mAudioHwDevs.valueAt(i); 266 } 267} 268 269static const char * const audio_interfaces[] = { 270 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 271 AUDIO_HARDWARE_MODULE_ID_A2DP, 272 AUDIO_HARDWARE_MODULE_ID_USB, 273}; 274#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 275 276audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 } else { 286 // check a match for the requested module handle 287 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 288 if (audioHwdevice != NULL) { 289 return audioHwdevice->hwDevice(); 290 } 291 } 292 // then try to find a module supporting the requested device. 293 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 294 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 295 if ((dev->get_supported_devices(dev) & devices) == devices) 296 return dev; 297 } 298 299 return NULL; 300} 301 302status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 303{ 304 const size_t SIZE = 256; 305 char buffer[SIZE]; 306 String8 result; 307 308 result.append("Clients:\n"); 309 for (size_t i = 0; i < mClients.size(); ++i) { 310 sp<Client> client = mClients.valueAt(i).promote(); 311 if (client != 0) { 312 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 313 result.append(buffer); 314 } 315 } 316 317 result.append("Global session refs:\n"); 318 result.append(" session pid count\n"); 319 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 320 AudioSessionRef *r = mAudioSessionRefs[i]; 321 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 322 result.append(buffer); 323 } 324 write(fd, result.string(), result.size()); 325 return NO_ERROR; 326} 327 328 329status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 330{ 331 const size_t SIZE = 256; 332 char buffer[SIZE]; 333 String8 result; 334 hardware_call_state hardwareStatus = mHardwareStatus; 335 336 snprintf(buffer, SIZE, "Hardware status: %d\n" 337 "Standby Time mSec: %u\n", 338 hardwareStatus, 339 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 340 result.append(buffer); 341 write(fd, result.string(), result.size()); 342 return NO_ERROR; 343} 344 345status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 snprintf(buffer, SIZE, "Permission Denial: " 351 "can't dump AudioFlinger from pid=%d, uid=%d\n", 352 IPCThreadState::self()->getCallingPid(), 353 IPCThreadState::self()->getCallingUid()); 354 result.append(buffer); 355 write(fd, result.string(), result.size()); 356 return NO_ERROR; 357} 358 359static bool tryLock(Mutex& mutex) 360{ 361 bool locked = false; 362 for (int i = 0; i < kDumpLockRetries; ++i) { 363 if (mutex.tryLock() == NO_ERROR) { 364 locked = true; 365 break; 366 } 367 usleep(kDumpLockSleepUs); 368 } 369 return locked; 370} 371 372status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 373{ 374 if (!dumpAllowed()) { 375 dumpPermissionDenial(fd, args); 376 } else { 377 // get state of hardware lock 378 bool hardwareLocked = tryLock(mHardwareLock); 379 if (!hardwareLocked) { 380 String8 result(kHardwareLockedString); 381 write(fd, result.string(), result.size()); 382 } else { 383 mHardwareLock.unlock(); 384 } 385 386 bool locked = tryLock(mLock); 387 388 // failed to lock - AudioFlinger is probably deadlocked 389 if (!locked) { 390 String8 result(kDeadlockedString); 391 write(fd, result.string(), result.size()); 392 } 393 394 dumpClients(fd, args); 395 dumpInternals(fd, args); 396 397 // dump playback threads 398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 399 mPlaybackThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump record threads 403 for (size_t i = 0; i < mRecordThreads.size(); i++) { 404 mRecordThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump all hardware devs 408 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 409 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 410 dev->dump(dev, fd); 411 } 412 if (locked) mLock.unlock(); 413 } 414 return NO_ERROR; 415} 416 417sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 418{ 419 // If pid is already in the mClients wp<> map, then use that entry 420 // (for which promote() is always != 0), otherwise create a new entry and Client. 421 sp<Client> client = mClients.valueFor(pid).promote(); 422 if (client == 0) { 423 client = new Client(this, pid); 424 mClients.add(pid, client); 425 } 426 427 return client; 428} 429 430// IAudioFlinger interface 431 432 433sp<IAudioTrack> AudioFlinger::createTrack( 434 pid_t pid, 435 audio_stream_type_t streamType, 436 uint32_t sampleRate, 437 audio_format_t format, 438 audio_channel_mask_t channelMask, 439 int frameCount, 440 IAudioFlinger::track_flags_t flags, 441 const sp<IMemory>& sharedBuffer, 442 audio_io_handle_t output, 443 pid_t tid, 444 int *sessionId, 445 status_t *status) 446{ 447 sp<PlaybackThread::Track> track; 448 sp<TrackHandle> trackHandle; 449 sp<Client> client; 450 status_t lStatus; 451 int lSessionId; 452 453 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 454 // but if someone uses binder directly they could bypass that and cause us to crash 455 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 456 ALOGE("createTrack() invalid stream type %d", streamType); 457 lStatus = BAD_VALUE; 458 goto Exit; 459 } 460 461 { 462 Mutex::Autolock _l(mLock); 463 PlaybackThread *thread = checkPlaybackThread_l(output); 464 PlaybackThread *effectThread = NULL; 465 if (thread == NULL) { 466 ALOGE("unknown output thread"); 467 lStatus = BAD_VALUE; 468 goto Exit; 469 } 470 471 client = registerPid_l(pid); 472 473 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 474 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 475 // check if an effect chain with the same session ID is present on another 476 // output thread and move it here. 477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 478 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 479 if (mPlaybackThreads.keyAt(i) != output) { 480 uint32_t sessions = t->hasAudioSession(*sessionId); 481 if (sessions & PlaybackThread::EFFECT_SESSION) { 482 effectThread = t.get(); 483 break; 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 508 // Look for sync events awaiting for a session to be used. 509 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 510 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 511 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 512 if (lStatus == NO_ERROR) { 513 track->setSyncEvent(mPendingSyncEvents[i]); 514 } else { 515 mPendingSyncEvents[i]->cancel(); 516 } 517 mPendingSyncEvents.removeAt(i); 518 i--; 519 } 520 } 521 } 522 } 523 if (lStatus == NO_ERROR) { 524 trackHandle = new TrackHandle(track); 525 } else { 526 // remove local strong reference to Client before deleting the Track so that the Client 527 // destructor is called by the TrackBase destructor with mLock held 528 client.clear(); 529 track.clear(); 530 } 531 532Exit: 533 if (status != NULL) { 534 *status = lStatus; 535 } 536 return trackHandle; 537} 538 539uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 540{ 541 Mutex::Autolock _l(mLock); 542 PlaybackThread *thread = checkPlaybackThread_l(output); 543 if (thread == NULL) { 544 ALOGW("sampleRate() unknown thread %d", output); 545 return 0; 546 } 547 return thread->sampleRate(); 548} 549 550int AudioFlinger::channelCount(audio_io_handle_t output) const 551{ 552 Mutex::Autolock _l(mLock); 553 PlaybackThread *thread = checkPlaybackThread_l(output); 554 if (thread == NULL) { 555 ALOGW("channelCount() unknown thread %d", output); 556 return 0; 557 } 558 return thread->channelCount(); 559} 560 561audio_format_t AudioFlinger::format(audio_io_handle_t output) const 562{ 563 Mutex::Autolock _l(mLock); 564 PlaybackThread *thread = checkPlaybackThread_l(output); 565 if (thread == NULL) { 566 ALOGW("format() unknown thread %d", output); 567 return AUDIO_FORMAT_INVALID; 568 } 569 return thread->format(); 570} 571 572size_t AudioFlinger::frameCount(audio_io_handle_t output) const 573{ 574 Mutex::Autolock _l(mLock); 575 PlaybackThread *thread = checkPlaybackThread_l(output); 576 if (thread == NULL) { 577 ALOGW("frameCount() unknown thread %d", output); 578 return 0; 579 } 580 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 581 // should examine all callers and fix them to handle smaller counts 582 return thread->frameCount(); 583} 584 585uint32_t AudioFlinger::latency(audio_io_handle_t output) const 586{ 587 Mutex::Autolock _l(mLock); 588 PlaybackThread *thread = checkPlaybackThread_l(output); 589 if (thread == NULL) { 590 ALOGW("latency() unknown thread %d", output); 591 return 0; 592 } 593 return thread->latency(); 594} 595 596status_t AudioFlinger::setMasterVolume(float value) 597{ 598 status_t ret = initCheck(); 599 if (ret != NO_ERROR) { 600 return ret; 601 } 602 603 // check calling permissions 604 if (!settingsAllowed()) { 605 return PERMISSION_DENIED; 606 } 607 608 float swmv = value; 609 610 Mutex::Autolock _l(mLock); 611 612 // when hw supports master volume, don't scale in sw mixer 613 if (MVS_NONE != mMasterVolumeSupportLvl) { 614 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 615 AutoMutex lock(mHardwareLock); 616 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 617 618 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 619 if (NULL != dev->set_master_volume) { 620 dev->set_master_volume(dev, value); 621 } 622 mHardwareStatus = AUDIO_HW_IDLE; 623 } 624 625 swmv = 1.0; 626 } 627 628 mMasterVolume = value; 629 mMasterVolumeSW = swmv; 630 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 631 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 632 633 return NO_ERROR; 634} 635 636status_t AudioFlinger::setMode(audio_mode_t mode) 637{ 638 status_t ret = initCheck(); 639 if (ret != NO_ERROR) { 640 return ret; 641 } 642 643 // check calling permissions 644 if (!settingsAllowed()) { 645 return PERMISSION_DENIED; 646 } 647 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 648 ALOGW("Illegal value: setMode(%d)", mode); 649 return BAD_VALUE; 650 } 651 652 { // scope for the lock 653 AutoMutex lock(mHardwareLock); 654 mHardwareStatus = AUDIO_HW_SET_MODE; 655 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 656 mHardwareStatus = AUDIO_HW_IDLE; 657 } 658 659 if (NO_ERROR == ret) { 660 Mutex::Autolock _l(mLock); 661 mMode = mode; 662 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 663 mPlaybackThreads.valueAt(i)->setMode(mode); 664 } 665 666 return ret; 667} 668 669status_t AudioFlinger::setMicMute(bool state) 670{ 671 status_t ret = initCheck(); 672 if (ret != NO_ERROR) { 673 return ret; 674 } 675 676 // check calling permissions 677 if (!settingsAllowed()) { 678 return PERMISSION_DENIED; 679 } 680 681 AutoMutex lock(mHardwareLock); 682 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 683 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 684 mHardwareStatus = AUDIO_HW_IDLE; 685 return ret; 686} 687 688bool AudioFlinger::getMicMute() const 689{ 690 status_t ret = initCheck(); 691 if (ret != NO_ERROR) { 692 return false; 693 } 694 695 bool state = AUDIO_MODE_INVALID; 696 AutoMutex lock(mHardwareLock); 697 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 698 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 699 mHardwareStatus = AUDIO_HW_IDLE; 700 return state; 701} 702 703status_t AudioFlinger::setMasterMute(bool muted) 704{ 705 // check calling permissions 706 if (!settingsAllowed()) { 707 return PERMISSION_DENIED; 708 } 709 710 Mutex::Autolock _l(mLock); 711 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 712 mMasterMute = muted; 713 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 714 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 715 716 return NO_ERROR; 717} 718 719float AudioFlinger::masterVolume() const 720{ 721 Mutex::Autolock _l(mLock); 722 return masterVolume_l(); 723} 724 725float AudioFlinger::masterVolumeSW() const 726{ 727 Mutex::Autolock _l(mLock); 728 return masterVolumeSW_l(); 729} 730 731bool AudioFlinger::masterMute() const 732{ 733 Mutex::Autolock _l(mLock); 734 return masterMute_l(); 735} 736 737float AudioFlinger::masterVolume_l() const 738{ 739 if (MVS_FULL == mMasterVolumeSupportLvl) { 740 float ret_val; 741 AutoMutex lock(mHardwareLock); 742 743 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 744 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 745 (NULL != mPrimaryHardwareDev->get_master_volume), 746 "can't get master volume"); 747 748 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 749 mHardwareStatus = AUDIO_HW_IDLE; 750 return ret_val; 751 } 752 753 return mMasterVolume; 754} 755 756status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 757 audio_io_handle_t output) 758{ 759 // check calling permissions 760 if (!settingsAllowed()) { 761 return PERMISSION_DENIED; 762 } 763 764 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 765 ALOGE("setStreamVolume() invalid stream %d", stream); 766 return BAD_VALUE; 767 } 768 769 AutoMutex lock(mLock); 770 PlaybackThread *thread = NULL; 771 if (output) { 772 thread = checkPlaybackThread_l(output); 773 if (thread == NULL) { 774 return BAD_VALUE; 775 } 776 } 777 778 mStreamTypes[stream].volume = value; 779 780 if (thread == NULL) { 781 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 782 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 783 } 784 } else { 785 thread->setStreamVolume(stream, value); 786 } 787 788 return NO_ERROR; 789} 790 791status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 792{ 793 // check calling permissions 794 if (!settingsAllowed()) { 795 return PERMISSION_DENIED; 796 } 797 798 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 799 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 800 ALOGE("setStreamMute() invalid stream %d", stream); 801 return BAD_VALUE; 802 } 803 804 AutoMutex lock(mLock); 805 mStreamTypes[stream].mute = muted; 806 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 807 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 808 809 return NO_ERROR; 810} 811 812float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 813{ 814 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 815 return 0.0f; 816 } 817 818 AutoMutex lock(mLock); 819 float volume; 820 if (output) { 821 PlaybackThread *thread = checkPlaybackThread_l(output); 822 if (thread == NULL) { 823 return 0.0f; 824 } 825 volume = thread->streamVolume(stream); 826 } else { 827 volume = streamVolume_l(stream); 828 } 829 830 return volume; 831} 832 833bool AudioFlinger::streamMute(audio_stream_type_t stream) const 834{ 835 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 836 return true; 837 } 838 839 AutoMutex lock(mLock); 840 return streamMute_l(stream); 841} 842 843status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 844{ 845 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 846 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 847 // check calling permissions 848 if (!settingsAllowed()) { 849 return PERMISSION_DENIED; 850 } 851 852 // ioHandle == 0 means the parameters are global to the audio hardware interface 853 if (ioHandle == 0) { 854 Mutex::Autolock _l(mLock); 855 status_t final_result = NO_ERROR; 856 { 857 AutoMutex lock(mHardwareLock); 858 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 859 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 860 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 861 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 862 final_result = result ?: final_result; 863 } 864 mHardwareStatus = AUDIO_HW_IDLE; 865 } 866 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 867 AudioParameter param = AudioParameter(keyValuePairs); 868 String8 value; 869 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 870 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 871 if (mBtNrecIsOff != btNrecIsOff) { 872 for (size_t i = 0; i < mRecordThreads.size(); i++) { 873 sp<RecordThread> thread = mRecordThreads.valueAt(i); 874 RecordThread::RecordTrack *track = thread->track(); 875 if (track != NULL) { 876 audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL; 877 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 878 thread->setEffectSuspended(FX_IID_AEC, 879 suspend, 880 track->sessionId()); 881 thread->setEffectSuspended(FX_IID_NS, 882 suspend, 883 track->sessionId()); 884 } 885 } 886 mBtNrecIsOff = btNrecIsOff; 887 } 888 } 889 String8 screenState; 890 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 891 bool isOff = screenState == "off"; 892 if (isOff != (gScreenState & 1)) { 893 gScreenState = ((gScreenState & ~1) + 2) | isOff; 894 } 895 } 896 return final_result; 897 } 898 899 // hold a strong ref on thread in case closeOutput() or closeInput() is called 900 // and the thread is exited once the lock is released 901 sp<ThreadBase> thread; 902 { 903 Mutex::Autolock _l(mLock); 904 thread = checkPlaybackThread_l(ioHandle); 905 if (thread == 0) { 906 thread = checkRecordThread_l(ioHandle); 907 } else if (thread == primaryPlaybackThread_l()) { 908 // indicate output device change to all input threads for pre processing 909 AudioParameter param = AudioParameter(keyValuePairs); 910 int value; 911 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 912 (value != 0)) { 913 for (size_t i = 0; i < mRecordThreads.size(); i++) { 914 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 915 } 916 } 917 } 918 } 919 if (thread != 0) { 920 return thread->setParameters(keyValuePairs); 921 } 922 return BAD_VALUE; 923} 924 925String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 926{ 927// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 928// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 929 930 Mutex::Autolock _l(mLock); 931 932 if (ioHandle == 0) { 933 String8 out_s8; 934 935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 936 char *s; 937 { 938 AutoMutex lock(mHardwareLock); 939 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 940 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 941 s = dev->get_parameters(dev, keys.string()); 942 mHardwareStatus = AUDIO_HW_IDLE; 943 } 944 out_s8 += String8(s ? s : ""); 945 free(s); 946 } 947 return out_s8; 948 } 949 950 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 951 if (playbackThread != NULL) { 952 return playbackThread->getParameters(keys); 953 } 954 RecordThread *recordThread = checkRecordThread_l(ioHandle); 955 if (recordThread != NULL) { 956 return recordThread->getParameters(keys); 957 } 958 return String8(""); 959} 960 961size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 962 audio_channel_mask_t channelMask) const 963{ 964 status_t ret = initCheck(); 965 if (ret != NO_ERROR) { 966 return 0; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 971 struct audio_config config = { 972 sample_rate: sampleRate, 973 channel_mask: channelMask, 974 format: format, 975 }; 976 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 977 mHardwareStatus = AUDIO_HW_IDLE; 978 return size; 979} 980 981unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 982{ 983 Mutex::Autolock _l(mLock); 984 985 RecordThread *recordThread = checkRecordThread_l(ioHandle); 986 if (recordThread != NULL) { 987 return recordThread->getInputFramesLost(); 988 } 989 return 0; 990} 991 992status_t AudioFlinger::setVoiceVolume(float value) 993{ 994 status_t ret = initCheck(); 995 if (ret != NO_ERROR) { 996 return ret; 997 } 998 999 // check calling permissions 1000 if (!settingsAllowed()) { 1001 return PERMISSION_DENIED; 1002 } 1003 1004 AutoMutex lock(mHardwareLock); 1005 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1006 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1007 mHardwareStatus = AUDIO_HW_IDLE; 1008 1009 return ret; 1010} 1011 1012status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1013 audio_io_handle_t output) const 1014{ 1015 status_t status; 1016 1017 Mutex::Autolock _l(mLock); 1018 1019 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1020 if (playbackThread != NULL) { 1021 return playbackThread->getRenderPosition(halFrames, dspFrames); 1022 } 1023 1024 return BAD_VALUE; 1025} 1026 1027void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1028{ 1029 1030 Mutex::Autolock _l(mLock); 1031 1032 pid_t pid = IPCThreadState::self()->getCallingPid(); 1033 if (mNotificationClients.indexOfKey(pid) < 0) { 1034 sp<NotificationClient> notificationClient = new NotificationClient(this, 1035 client, 1036 pid); 1037 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1038 1039 mNotificationClients.add(pid, notificationClient); 1040 1041 sp<IBinder> binder = client->asBinder(); 1042 binder->linkToDeath(notificationClient); 1043 1044 // the config change is always sent from playback or record threads to avoid deadlock 1045 // with AudioSystem::gLock 1046 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1047 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1048 } 1049 1050 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1051 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1052 } 1053 } 1054} 1055 1056void AudioFlinger::removeNotificationClient(pid_t pid) 1057{ 1058 Mutex::Autolock _l(mLock); 1059 1060 mNotificationClients.removeItem(pid); 1061 1062 ALOGV("%d died, releasing its sessions", pid); 1063 size_t num = mAudioSessionRefs.size(); 1064 bool removed = false; 1065 for (size_t i = 0; i< num; ) { 1066 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1067 ALOGV(" pid %d @ %d", ref->mPid, i); 1068 if (ref->mPid == pid) { 1069 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1070 mAudioSessionRefs.removeAt(i); 1071 delete ref; 1072 removed = true; 1073 num--; 1074 } else { 1075 i++; 1076 } 1077 } 1078 if (removed) { 1079 purgeStaleEffects_l(); 1080 } 1081} 1082 1083// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1084void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1085{ 1086 size_t size = mNotificationClients.size(); 1087 for (size_t i = 0; i < size; i++) { 1088 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1089 param2); 1090 } 1091} 1092 1093// removeClient_l() must be called with AudioFlinger::mLock held 1094void AudioFlinger::removeClient_l(pid_t pid) 1095{ 1096 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1097 mClients.removeItem(pid); 1098} 1099 1100// getEffectThread_l() must be called with AudioFlinger::mLock held 1101sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1102{ 1103 sp<PlaybackThread> thread; 1104 1105 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1106 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1107 ALOG_ASSERT(thread == 0); 1108 thread = mPlaybackThreads.valueAt(i); 1109 } 1110 } 1111 1112 return thread; 1113} 1114 1115// ---------------------------------------------------------------------------- 1116 1117AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1118 audio_devices_t device, type_t type) 1119 : Thread(false), 1120 mType(type), 1121 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1122 // mChannelMask 1123 mChannelCount(0), 1124 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1125 mParamStatus(NO_ERROR), 1126 mStandby(false), mDevice((audio_devices_t) device), mId(id), 1127 mDeathRecipient(new PMDeathRecipient(this)) 1128{ 1129} 1130 1131AudioFlinger::ThreadBase::~ThreadBase() 1132{ 1133 mParamCond.broadcast(); 1134 // do not lock the mutex in destructor 1135 releaseWakeLock_l(); 1136 if (mPowerManager != 0) { 1137 sp<IBinder> binder = mPowerManager->asBinder(); 1138 binder->unlinkToDeath(mDeathRecipient); 1139 } 1140} 1141 1142void AudioFlinger::ThreadBase::exit() 1143{ 1144 ALOGV("ThreadBase::exit"); 1145 { 1146 // This lock prevents the following race in thread (uniprocessor for illustration): 1147 // if (!exitPending()) { 1148 // // context switch from here to exit() 1149 // // exit() calls requestExit(), what exitPending() observes 1150 // // exit() calls signal(), which is dropped since no waiters 1151 // // context switch back from exit() to here 1152 // mWaitWorkCV.wait(...); 1153 // // now thread is hung 1154 // } 1155 AutoMutex lock(mLock); 1156 requestExit(); 1157 mWaitWorkCV.signal(); 1158 } 1159 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1160 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1161 requestExitAndWait(); 1162} 1163 1164status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1165{ 1166 status_t status; 1167 1168 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1169 Mutex::Autolock _l(mLock); 1170 1171 mNewParameters.add(keyValuePairs); 1172 mWaitWorkCV.signal(); 1173 // wait condition with timeout in case the thread loop has exited 1174 // before the request could be processed 1175 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1176 status = mParamStatus; 1177 mWaitWorkCV.signal(); 1178 } else { 1179 status = TIMED_OUT; 1180 } 1181 return status; 1182} 1183 1184void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1185{ 1186 Mutex::Autolock _l(mLock); 1187 sendConfigEvent_l(event, param); 1188} 1189 1190// sendConfigEvent_l() must be called with ThreadBase::mLock held 1191void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1192{ 1193 ConfigEvent configEvent; 1194 configEvent.mEvent = event; 1195 configEvent.mParam = param; 1196 mConfigEvents.add(configEvent); 1197 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1198 mWaitWorkCV.signal(); 1199} 1200 1201void AudioFlinger::ThreadBase::processConfigEvents() 1202{ 1203 mLock.lock(); 1204 while (!mConfigEvents.isEmpty()) { 1205 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1206 ConfigEvent configEvent = mConfigEvents[0]; 1207 mConfigEvents.removeAt(0); 1208 // release mLock before locking AudioFlinger mLock: lock order is always 1209 // AudioFlinger then ThreadBase to avoid cross deadlock 1210 mLock.unlock(); 1211 mAudioFlinger->mLock.lock(); 1212 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1213 mAudioFlinger->mLock.unlock(); 1214 mLock.lock(); 1215 } 1216 mLock.unlock(); 1217} 1218 1219status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1220{ 1221 const size_t SIZE = 256; 1222 char buffer[SIZE]; 1223 String8 result; 1224 1225 bool locked = tryLock(mLock); 1226 if (!locked) { 1227 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1228 write(fd, buffer, strlen(buffer)); 1229 } 1230 1231 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1246 result.append(buffer); 1247 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1250 result.append(buffer); 1251 1252 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1253 result.append(buffer); 1254 result.append(" Index Command"); 1255 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1256 snprintf(buffer, SIZE, "\n %02d ", i); 1257 result.append(buffer); 1258 result.append(mNewParameters[i]); 1259 } 1260 1261 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1262 result.append(buffer); 1263 snprintf(buffer, SIZE, " Index event param\n"); 1264 result.append(buffer); 1265 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1266 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1267 result.append(buffer); 1268 } 1269 result.append("\n"); 1270 1271 write(fd, result.string(), result.size()); 1272 1273 if (locked) { 1274 mLock.unlock(); 1275 } 1276 return NO_ERROR; 1277} 1278 1279status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1280{ 1281 const size_t SIZE = 256; 1282 char buffer[SIZE]; 1283 String8 result; 1284 1285 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1286 write(fd, buffer, strlen(buffer)); 1287 1288 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1289 sp<EffectChain> chain = mEffectChains[i]; 1290 if (chain != 0) { 1291 chain->dump(fd, args); 1292 } 1293 } 1294 return NO_ERROR; 1295} 1296 1297void AudioFlinger::ThreadBase::acquireWakeLock() 1298{ 1299 Mutex::Autolock _l(mLock); 1300 acquireWakeLock_l(); 1301} 1302 1303void AudioFlinger::ThreadBase::acquireWakeLock_l() 1304{ 1305 if (mPowerManager == 0) { 1306 // use checkService() to avoid blocking if power service is not up yet 1307 sp<IBinder> binder = 1308 defaultServiceManager()->checkService(String16("power")); 1309 if (binder == 0) { 1310 ALOGW("Thread %s cannot connect to the power manager service", mName); 1311 } else { 1312 mPowerManager = interface_cast<IPowerManager>(binder); 1313 binder->linkToDeath(mDeathRecipient); 1314 } 1315 } 1316 if (mPowerManager != 0) { 1317 sp<IBinder> binder = new BBinder(); 1318 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1319 binder, 1320 String16(mName)); 1321 if (status == NO_ERROR) { 1322 mWakeLockToken = binder; 1323 } 1324 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1325 } 1326} 1327 1328void AudioFlinger::ThreadBase::releaseWakeLock() 1329{ 1330 Mutex::Autolock _l(mLock); 1331 releaseWakeLock_l(); 1332} 1333 1334void AudioFlinger::ThreadBase::releaseWakeLock_l() 1335{ 1336 if (mWakeLockToken != 0) { 1337 ALOGV("releaseWakeLock_l() %s", mName); 1338 if (mPowerManager != 0) { 1339 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1340 } 1341 mWakeLockToken.clear(); 1342 } 1343} 1344 1345void AudioFlinger::ThreadBase::clearPowerManager() 1346{ 1347 Mutex::Autolock _l(mLock); 1348 releaseWakeLock_l(); 1349 mPowerManager.clear(); 1350} 1351 1352void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1353{ 1354 sp<ThreadBase> thread = mThread.promote(); 1355 if (thread != 0) { 1356 thread->clearPowerManager(); 1357 } 1358 ALOGW("power manager service died !!!"); 1359} 1360 1361void AudioFlinger::ThreadBase::setEffectSuspended( 1362 const effect_uuid_t *type, bool suspend, int sessionId) 1363{ 1364 Mutex::Autolock _l(mLock); 1365 setEffectSuspended_l(type, suspend, sessionId); 1366} 1367 1368void AudioFlinger::ThreadBase::setEffectSuspended_l( 1369 const effect_uuid_t *type, bool suspend, int sessionId) 1370{ 1371 sp<EffectChain> chain = getEffectChain_l(sessionId); 1372 if (chain != 0) { 1373 if (type != NULL) { 1374 chain->setEffectSuspended_l(type, suspend); 1375 } else { 1376 chain->setEffectSuspendedAll_l(suspend); 1377 } 1378 } 1379 1380 updateSuspendedSessions_l(type, suspend, sessionId); 1381} 1382 1383void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1384{ 1385 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1386 if (index < 0) { 1387 return; 1388 } 1389 1390 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1391 mSuspendedSessions.editValueAt(index); 1392 1393 for (size_t i = 0; i < sessionEffects.size(); i++) { 1394 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1395 for (int j = 0; j < desc->mRefCount; j++) { 1396 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1397 chain->setEffectSuspendedAll_l(true); 1398 } else { 1399 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1400 desc->mType.timeLow); 1401 chain->setEffectSuspended_l(&desc->mType, true); 1402 } 1403 } 1404 } 1405} 1406 1407void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1408 bool suspend, 1409 int sessionId) 1410{ 1411 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1412 1413 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1414 1415 if (suspend) { 1416 if (index >= 0) { 1417 sessionEffects = mSuspendedSessions.editValueAt(index); 1418 } else { 1419 mSuspendedSessions.add(sessionId, sessionEffects); 1420 } 1421 } else { 1422 if (index < 0) { 1423 return; 1424 } 1425 sessionEffects = mSuspendedSessions.editValueAt(index); 1426 } 1427 1428 1429 int key = EffectChain::kKeyForSuspendAll; 1430 if (type != NULL) { 1431 key = type->timeLow; 1432 } 1433 index = sessionEffects.indexOfKey(key); 1434 1435 sp<SuspendedSessionDesc> desc; 1436 if (suspend) { 1437 if (index >= 0) { 1438 desc = sessionEffects.valueAt(index); 1439 } else { 1440 desc = new SuspendedSessionDesc(); 1441 if (type != NULL) { 1442 desc->mType = *type; 1443 } 1444 sessionEffects.add(key, desc); 1445 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1446 } 1447 desc->mRefCount++; 1448 } else { 1449 if (index < 0) { 1450 return; 1451 } 1452 desc = sessionEffects.valueAt(index); 1453 if (--desc->mRefCount == 0) { 1454 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1455 sessionEffects.removeItemsAt(index); 1456 if (sessionEffects.isEmpty()) { 1457 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1458 sessionId); 1459 mSuspendedSessions.removeItem(sessionId); 1460 } 1461 } 1462 } 1463 if (!sessionEffects.isEmpty()) { 1464 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1465 } 1466} 1467 1468void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1469 bool enabled, 1470 int sessionId) 1471{ 1472 Mutex::Autolock _l(mLock); 1473 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1474} 1475 1476void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1477 bool enabled, 1478 int sessionId) 1479{ 1480 if (mType != RECORD) { 1481 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1482 // another session. This gives the priority to well behaved effect control panels 1483 // and applications not using global effects. 1484 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1485 // global effects 1486 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1487 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1488 } 1489 } 1490 1491 sp<EffectChain> chain = getEffectChain_l(sessionId); 1492 if (chain != 0) { 1493 chain->checkSuspendOnEffectEnabled(effect, enabled); 1494 } 1495} 1496 1497// ---------------------------------------------------------------------------- 1498 1499AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1500 AudioStreamOut* output, 1501 audio_io_handle_t id, 1502 audio_devices_t device, 1503 type_t type) 1504 : ThreadBase(audioFlinger, id, device, type), 1505 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1506 // Assumes constructor is called by AudioFlinger with it's mLock held, 1507 // but it would be safer to explicitly pass initial masterMute as parameter 1508 mMasterMute(audioFlinger->masterMute_l()), 1509 // mStreamTypes[] initialized in constructor body 1510 mOutput(output), 1511 // Assumes constructor is called by AudioFlinger with it's mLock held, 1512 // but it would be safer to explicitly pass initial masterVolume as parameter 1513 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1514 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1515 mMixerStatus(MIXER_IDLE), 1516 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1517 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1518 mScreenState(gScreenState), 1519 // index 0 is reserved for normal mixer's submix 1520 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1521{ 1522 snprintf(mName, kNameLength, "AudioOut_%X", id); 1523 1524 readOutputParameters(); 1525 1526 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1527 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1528 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1529 stream = (audio_stream_type_t) (stream + 1)) { 1530 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1531 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1532 } 1533 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1534 // because mAudioFlinger doesn't have one to copy from 1535} 1536 1537AudioFlinger::PlaybackThread::~PlaybackThread() 1538{ 1539 delete [] mMixBuffer; 1540} 1541 1542status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1543{ 1544 dumpInternals(fd, args); 1545 dumpTracks(fd, args); 1546 dumpEffectChains(fd, args); 1547 return NO_ERROR; 1548} 1549 1550status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1551{ 1552 const size_t SIZE = 256; 1553 char buffer[SIZE]; 1554 String8 result; 1555 1556 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1557 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1558 const stream_type_t *st = &mStreamTypes[i]; 1559 if (i > 0) { 1560 result.appendFormat(", "); 1561 } 1562 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1563 if (st->mute) { 1564 result.append("M"); 1565 } 1566 } 1567 result.append("\n"); 1568 write(fd, result.string(), result.length()); 1569 result.clear(); 1570 1571 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1572 result.append(buffer); 1573 Track::appendDumpHeader(result); 1574 for (size_t i = 0; i < mTracks.size(); ++i) { 1575 sp<Track> track = mTracks[i]; 1576 if (track != 0) { 1577 track->dump(buffer, SIZE); 1578 result.append(buffer); 1579 } 1580 } 1581 1582 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1583 result.append(buffer); 1584 Track::appendDumpHeader(result); 1585 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1586 sp<Track> track = mActiveTracks[i].promote(); 1587 if (track != 0) { 1588 track->dump(buffer, SIZE); 1589 result.append(buffer); 1590 } 1591 } 1592 write(fd, result.string(), result.size()); 1593 1594 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1595 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1596 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1597 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1598 1599 return NO_ERROR; 1600} 1601 1602status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1603{ 1604 const size_t SIZE = 256; 1605 char buffer[SIZE]; 1606 String8 result; 1607 1608 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1609 result.append(buffer); 1610 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1611 result.append(buffer); 1612 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1613 result.append(buffer); 1614 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1615 result.append(buffer); 1616 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1617 result.append(buffer); 1618 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1619 result.append(buffer); 1620 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1621 result.append(buffer); 1622 write(fd, result.string(), result.size()); 1623 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1624 1625 dumpBase(fd, args); 1626 1627 return NO_ERROR; 1628} 1629 1630// Thread virtuals 1631status_t AudioFlinger::PlaybackThread::readyToRun() 1632{ 1633 status_t status = initCheck(); 1634 if (status == NO_ERROR) { 1635 ALOGI("AudioFlinger's thread %p ready to run", this); 1636 } else { 1637 ALOGE("No working audio driver found."); 1638 } 1639 return status; 1640} 1641 1642void AudioFlinger::PlaybackThread::onFirstRef() 1643{ 1644 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1645} 1646 1647// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1648sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1649 const sp<AudioFlinger::Client>& client, 1650 audio_stream_type_t streamType, 1651 uint32_t sampleRate, 1652 audio_format_t format, 1653 audio_channel_mask_t channelMask, 1654 int frameCount, 1655 const sp<IMemory>& sharedBuffer, 1656 int sessionId, 1657 IAudioFlinger::track_flags_t flags, 1658 pid_t tid, 1659 status_t *status) 1660{ 1661 sp<Track> track; 1662 status_t lStatus; 1663 1664 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1665 1666 // client expresses a preference for FAST, but we get the final say 1667 if (flags & IAudioFlinger::TRACK_FAST) { 1668 if ( 1669 // not timed 1670 (!isTimed) && 1671 // either of these use cases: 1672 ( 1673 // use case 1: shared buffer with any frame count 1674 ( 1675 (sharedBuffer != 0) 1676 ) || 1677 // use case 2: callback handler and frame count is default or at least as large as HAL 1678 ( 1679 (tid != -1) && 1680 ((frameCount == 0) || 1681 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1682 ) 1683 ) && 1684 // PCM data 1685 audio_is_linear_pcm(format) && 1686 // mono or stereo 1687 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1688 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1689#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1690 // hardware sample rate 1691 (sampleRate == mSampleRate) && 1692#endif 1693 // normal mixer has an associated fast mixer 1694 hasFastMixer() && 1695 // there are sufficient fast track slots available 1696 (mFastTrackAvailMask != 0) 1697 // FIXME test that MixerThread for this fast track has a capable output HAL 1698 // FIXME add a permission test also? 1699 ) { 1700 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1701 if (frameCount == 0) { 1702 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1703 } 1704 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1705 frameCount, mFrameCount); 1706 } else { 1707 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1708 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1709 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1710 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1711 audio_is_linear_pcm(format), 1712 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1713 flags &= ~IAudioFlinger::TRACK_FAST; 1714 // For compatibility with AudioTrack calculation, buffer depth is forced 1715 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1716 // This is probably too conservative, but legacy application code may depend on it. 1717 // If you change this calculation, also review the start threshold which is related. 1718 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1719 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1720 if (minBufCount < 2) { 1721 minBufCount = 2; 1722 } 1723 int minFrameCount = mNormalFrameCount * minBufCount; 1724 if (frameCount < minFrameCount) { 1725 frameCount = minFrameCount; 1726 } 1727 } 1728 } 1729 1730 if (mType == DIRECT) { 1731 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1732 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1733 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1734 "for output %p with format %d", 1735 sampleRate, format, channelMask, mOutput, mFormat); 1736 lStatus = BAD_VALUE; 1737 goto Exit; 1738 } 1739 } 1740 } else { 1741 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1742 if (sampleRate > mSampleRate*2) { 1743 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1744 lStatus = BAD_VALUE; 1745 goto Exit; 1746 } 1747 } 1748 1749 lStatus = initCheck(); 1750 if (lStatus != NO_ERROR) { 1751 ALOGE("Audio driver not initialized."); 1752 goto Exit; 1753 } 1754 1755 { // scope for mLock 1756 Mutex::Autolock _l(mLock); 1757 1758 // all tracks in same audio session must share the same routing strategy otherwise 1759 // conflicts will happen when tracks are moved from one output to another by audio policy 1760 // manager 1761 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1762 for (size_t i = 0; i < mTracks.size(); ++i) { 1763 sp<Track> t = mTracks[i]; 1764 if (t != 0 && !t->isOutputTrack()) { 1765 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1766 if (sessionId == t->sessionId() && strategy != actual) { 1767 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1768 strategy, actual); 1769 lStatus = BAD_VALUE; 1770 goto Exit; 1771 } 1772 } 1773 } 1774 1775 if (!isTimed) { 1776 track = new Track(this, client, streamType, sampleRate, format, 1777 channelMask, frameCount, sharedBuffer, sessionId, flags); 1778 } else { 1779 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1780 channelMask, frameCount, sharedBuffer, sessionId); 1781 } 1782 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1783 lStatus = NO_MEMORY; 1784 goto Exit; 1785 } 1786 mTracks.add(track); 1787 1788 sp<EffectChain> chain = getEffectChain_l(sessionId); 1789 if (chain != 0) { 1790 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1791 track->setMainBuffer(chain->inBuffer()); 1792 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1793 chain->incTrackCnt(); 1794 } 1795 } 1796 1797 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1798 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1799 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1800 // so ask activity manager to do this on our behalf 1801 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1802 if (err != 0) { 1803 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1804 kPriorityAudioApp, callingPid, tid, err); 1805 } 1806 } 1807 1808 lStatus = NO_ERROR; 1809 1810Exit: 1811 if (status) { 1812 *status = lStatus; 1813 } 1814 return track; 1815} 1816 1817uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1818{ 1819 if (mFastMixer != NULL) { 1820 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1821 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1822 } 1823 return latency; 1824} 1825 1826uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1827{ 1828 return latency; 1829} 1830 1831uint32_t AudioFlinger::PlaybackThread::latency() const 1832{ 1833 Mutex::Autolock _l(mLock); 1834 return latency_l(); 1835} 1836uint32_t AudioFlinger::PlaybackThread::latency_l() const 1837{ 1838 if (initCheck() == NO_ERROR) { 1839 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1840 } else { 1841 return 0; 1842 } 1843} 1844 1845void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 mMasterVolume = value; 1849} 1850 1851void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1852{ 1853 Mutex::Autolock _l(mLock); 1854 setMasterMute_l(muted); 1855} 1856 1857void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1858{ 1859 Mutex::Autolock _l(mLock); 1860 mStreamTypes[stream].volume = value; 1861} 1862 1863void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1864{ 1865 Mutex::Autolock _l(mLock); 1866 mStreamTypes[stream].mute = muted; 1867} 1868 1869float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1870{ 1871 Mutex::Autolock _l(mLock); 1872 return mStreamTypes[stream].volume; 1873} 1874 1875// addTrack_l() must be called with ThreadBase::mLock held 1876status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1877{ 1878 status_t status = ALREADY_EXISTS; 1879 1880 // set retry count for buffer fill 1881 track->mRetryCount = kMaxTrackStartupRetries; 1882 if (mActiveTracks.indexOf(track) < 0) { 1883 // the track is newly added, make sure it fills up all its 1884 // buffers before playing. This is to ensure the client will 1885 // effectively get the latency it requested. 1886 track->mFillingUpStatus = Track::FS_FILLING; 1887 track->mResetDone = false; 1888 track->mPresentationCompleteFrames = 0; 1889 mActiveTracks.add(track); 1890 if (track->mainBuffer() != mMixBuffer) { 1891 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1892 if (chain != 0) { 1893 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1894 chain->incActiveTrackCnt(); 1895 } 1896 } 1897 1898 status = NO_ERROR; 1899 } 1900 1901 ALOGV("mWaitWorkCV.broadcast"); 1902 mWaitWorkCV.broadcast(); 1903 1904 return status; 1905} 1906 1907// destroyTrack_l() must be called with ThreadBase::mLock held 1908void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1909{ 1910 track->mState = TrackBase::TERMINATED; 1911 // active tracks are removed by threadLoop() 1912 if (mActiveTracks.indexOf(track) < 0) { 1913 removeTrack_l(track); 1914 } 1915} 1916 1917void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1918{ 1919 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1920 mTracks.remove(track); 1921 deleteTrackName_l(track->name()); 1922 // redundant as track is about to be destroyed, for dumpsys only 1923 track->mName = -1; 1924 if (track->isFastTrack()) { 1925 int index = track->mFastIndex; 1926 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1927 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1928 mFastTrackAvailMask |= 1 << index; 1929 // redundant as track is about to be destroyed, for dumpsys only 1930 track->mFastIndex = -1; 1931 } 1932 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1933 if (chain != 0) { 1934 chain->decTrackCnt(); 1935 } 1936} 1937 1938String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1939{ 1940 String8 out_s8 = String8(""); 1941 char *s; 1942 1943 Mutex::Autolock _l(mLock); 1944 if (initCheck() != NO_ERROR) { 1945 return out_s8; 1946 } 1947 1948 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1949 out_s8 = String8(s); 1950 free(s); 1951 return out_s8; 1952} 1953 1954// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1955void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1956 AudioSystem::OutputDescriptor desc; 1957 void *param2 = NULL; 1958 1959 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1960 1961 switch (event) { 1962 case AudioSystem::OUTPUT_OPENED: 1963 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1964 desc.channels = mChannelMask; 1965 desc.samplingRate = mSampleRate; 1966 desc.format = mFormat; 1967 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1968 desc.latency = latency(); 1969 param2 = &desc; 1970 break; 1971 1972 case AudioSystem::STREAM_CONFIG_CHANGED: 1973 param2 = ¶m; 1974 case AudioSystem::OUTPUT_CLOSED: 1975 default: 1976 break; 1977 } 1978 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1979} 1980 1981void AudioFlinger::PlaybackThread::readOutputParameters() 1982{ 1983 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1984 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1985 mChannelCount = (uint16_t)popcount(mChannelMask); 1986 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1987 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1988 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1989 if (mFrameCount & 15) { 1990 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1991 mFrameCount); 1992 } 1993 1994 // Calculate size of normal mix buffer relative to the HAL output buffer size 1995 double multiplier = 1.0; 1996 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1997 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1998 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1999 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2000 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2001 maxNormalFrameCount = maxNormalFrameCount & ~15; 2002 if (maxNormalFrameCount < minNormalFrameCount) { 2003 maxNormalFrameCount = minNormalFrameCount; 2004 } 2005 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2006 if (multiplier <= 1.0) { 2007 multiplier = 1.0; 2008 } else if (multiplier <= 2.0) { 2009 if (2 * mFrameCount <= maxNormalFrameCount) { 2010 multiplier = 2.0; 2011 } else { 2012 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2013 } 2014 } else { 2015 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2016 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2017 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2018 // FIXME this rounding up should not be done if no HAL SRC 2019 uint32_t truncMult = (uint32_t) multiplier; 2020 if ((truncMult & 1)) { 2021 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2022 ++truncMult; 2023 } 2024 } 2025 multiplier = (double) truncMult; 2026 } 2027 } 2028 mNormalFrameCount = multiplier * mFrameCount; 2029 // round up to nearest 16 frames to satisfy AudioMixer 2030 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2031 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2032 2033 delete[] mMixBuffer; 2034 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2035 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2036 2037 // force reconfiguration of effect chains and engines to take new buffer size and audio 2038 // parameters into account 2039 // Note that mLock is not held when readOutputParameters() is called from the constructor 2040 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2041 // matter. 2042 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2043 Vector< sp<EffectChain> > effectChains = mEffectChains; 2044 for (size_t i = 0; i < effectChains.size(); i ++) { 2045 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2046 } 2047} 2048 2049 2050status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2051{ 2052 if (halFrames == NULL || dspFrames == NULL) { 2053 return BAD_VALUE; 2054 } 2055 Mutex::Autolock _l(mLock); 2056 if (initCheck() != NO_ERROR) { 2057 return INVALID_OPERATION; 2058 } 2059 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2060 2061 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2062} 2063 2064uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2065{ 2066 Mutex::Autolock _l(mLock); 2067 uint32_t result = 0; 2068 if (getEffectChain_l(sessionId) != 0) { 2069 result = EFFECT_SESSION; 2070 } 2071 2072 for (size_t i = 0; i < mTracks.size(); ++i) { 2073 sp<Track> track = mTracks[i]; 2074 if (sessionId == track->sessionId() && 2075 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2076 result |= TRACK_SESSION; 2077 break; 2078 } 2079 } 2080 2081 return result; 2082} 2083 2084uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2085{ 2086 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2087 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2088 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2089 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2090 } 2091 for (size_t i = 0; i < mTracks.size(); i++) { 2092 sp<Track> track = mTracks[i]; 2093 if (sessionId == track->sessionId() && 2094 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2095 return AudioSystem::getStrategyForStream(track->streamType()); 2096 } 2097 } 2098 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2099} 2100 2101 2102AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2103{ 2104 Mutex::Autolock _l(mLock); 2105 return mOutput; 2106} 2107 2108AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2109{ 2110 Mutex::Autolock _l(mLock); 2111 AudioStreamOut *output = mOutput; 2112 mOutput = NULL; 2113 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2114 // must push a NULL and wait for ack 2115 mOutputSink.clear(); 2116 mPipeSink.clear(); 2117 mNormalSink.clear(); 2118 return output; 2119} 2120 2121// this method must always be called either with ThreadBase mLock held or inside the thread loop 2122audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2123{ 2124 if (mOutput == NULL) { 2125 return NULL; 2126 } 2127 return &mOutput->stream->common; 2128} 2129 2130uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2131{ 2132 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2133} 2134 2135status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2136{ 2137 if (!isValidSyncEvent(event)) { 2138 return BAD_VALUE; 2139 } 2140 2141 Mutex::Autolock _l(mLock); 2142 2143 for (size_t i = 0; i < mTracks.size(); ++i) { 2144 sp<Track> track = mTracks[i]; 2145 if (event->triggerSession() == track->sessionId()) { 2146 track->setSyncEvent(event); 2147 return NO_ERROR; 2148 } 2149 } 2150 2151 return NAME_NOT_FOUND; 2152} 2153 2154bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2155{ 2156 switch (event->type()) { 2157 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2158 return true; 2159 default: 2160 break; 2161 } 2162 return false; 2163} 2164 2165void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2166{ 2167 size_t count = tracksToRemove.size(); 2168 if (CC_UNLIKELY(count)) { 2169 for (size_t i = 0 ; i < count ; i++) { 2170 const sp<Track>& track = tracksToRemove.itemAt(i); 2171 if ((track->sharedBuffer() != 0) && 2172 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2173 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2174 } 2175 } 2176 } 2177 2178} 2179 2180// ---------------------------------------------------------------------------- 2181 2182AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2183 audio_io_handle_t id, audio_devices_t device, type_t type) 2184 : PlaybackThread(audioFlinger, output, id, device, type), 2185 // mAudioMixer below 2186 // mFastMixer below 2187 mFastMixerFutex(0) 2188 // mOutputSink below 2189 // mPipeSink below 2190 // mNormalSink below 2191{ 2192 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2193 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2194 "mFrameCount=%d, mNormalFrameCount=%d", 2195 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2196 mNormalFrameCount); 2197 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2198 2199 // FIXME - Current mixer implementation only supports stereo output 2200 if (mChannelCount != FCC_2) { 2201 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2202 } 2203 2204 // create an NBAIO sink for the HAL output stream, and negotiate 2205 mOutputSink = new AudioStreamOutSink(output->stream); 2206 size_t numCounterOffers = 0; 2207 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2208 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2209 ALOG_ASSERT(index == 0); 2210 2211 // initialize fast mixer depending on configuration 2212 bool initFastMixer; 2213 switch (kUseFastMixer) { 2214 case FastMixer_Never: 2215 initFastMixer = false; 2216 break; 2217 case FastMixer_Always: 2218 initFastMixer = true; 2219 break; 2220 case FastMixer_Static: 2221 case FastMixer_Dynamic: 2222 initFastMixer = mFrameCount < mNormalFrameCount; 2223 break; 2224 } 2225 if (initFastMixer) { 2226 2227 // create a MonoPipe to connect our submix to FastMixer 2228 NBAIO_Format format = mOutputSink->format(); 2229 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2230 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2231 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2232 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2233 const NBAIO_Format offers[1] = {format}; 2234 size_t numCounterOffers = 0; 2235 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2236 ALOG_ASSERT(index == 0); 2237 monoPipe->setAvgFrames((mScreenState & 1) ? 2238 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2239 mPipeSink = monoPipe; 2240 2241#ifdef TEE_SINK_FRAMES 2242 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2243 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2244 numCounterOffers = 0; 2245 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2246 ALOG_ASSERT(index == 0); 2247 mTeeSink = teeSink; 2248 PipeReader *teeSource = new PipeReader(*teeSink); 2249 numCounterOffers = 0; 2250 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2251 ALOG_ASSERT(index == 0); 2252 mTeeSource = teeSource; 2253#endif 2254 2255 // create fast mixer and configure it initially with just one fast track for our submix 2256 mFastMixer = new FastMixer(); 2257 FastMixerStateQueue *sq = mFastMixer->sq(); 2258#ifdef STATE_QUEUE_DUMP 2259 sq->setObserverDump(&mStateQueueObserverDump); 2260 sq->setMutatorDump(&mStateQueueMutatorDump); 2261#endif 2262 FastMixerState *state = sq->begin(); 2263 FastTrack *fastTrack = &state->mFastTracks[0]; 2264 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2265 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2266 fastTrack->mVolumeProvider = NULL; 2267 fastTrack->mGeneration++; 2268 state->mFastTracksGen++; 2269 state->mTrackMask = 1; 2270 // fast mixer will use the HAL output sink 2271 state->mOutputSink = mOutputSink.get(); 2272 state->mOutputSinkGen++; 2273 state->mFrameCount = mFrameCount; 2274 state->mCommand = FastMixerState::COLD_IDLE; 2275 // already done in constructor initialization list 2276 //mFastMixerFutex = 0; 2277 state->mColdFutexAddr = &mFastMixerFutex; 2278 state->mColdGen++; 2279 state->mDumpState = &mFastMixerDumpState; 2280 state->mTeeSink = mTeeSink.get(); 2281 sq->end(); 2282 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2283 2284 // start the fast mixer 2285 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2286 pid_t tid = mFastMixer->getTid(); 2287 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2288 if (err != 0) { 2289 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2290 kPriorityFastMixer, getpid_cached, tid, err); 2291 } 2292 2293#ifdef AUDIO_WATCHDOG 2294 // create and start the watchdog 2295 mAudioWatchdog = new AudioWatchdog(); 2296 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2297 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2298 tid = mAudioWatchdog->getTid(); 2299 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2300 if (err != 0) { 2301 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2302 kPriorityFastMixer, getpid_cached, tid, err); 2303 } 2304#endif 2305 2306 } else { 2307 mFastMixer = NULL; 2308 } 2309 2310 switch (kUseFastMixer) { 2311 case FastMixer_Never: 2312 case FastMixer_Dynamic: 2313 mNormalSink = mOutputSink; 2314 break; 2315 case FastMixer_Always: 2316 mNormalSink = mPipeSink; 2317 break; 2318 case FastMixer_Static: 2319 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2320 break; 2321 } 2322} 2323 2324AudioFlinger::MixerThread::~MixerThread() 2325{ 2326 if (mFastMixer != NULL) { 2327 FastMixerStateQueue *sq = mFastMixer->sq(); 2328 FastMixerState *state = sq->begin(); 2329 if (state->mCommand == FastMixerState::COLD_IDLE) { 2330 int32_t old = android_atomic_inc(&mFastMixerFutex); 2331 if (old == -1) { 2332 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2333 } 2334 } 2335 state->mCommand = FastMixerState::EXIT; 2336 sq->end(); 2337 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2338 mFastMixer->join(); 2339 // Though the fast mixer thread has exited, it's state queue is still valid. 2340 // We'll use that extract the final state which contains one remaining fast track 2341 // corresponding to our sub-mix. 2342 state = sq->begin(); 2343 ALOG_ASSERT(state->mTrackMask == 1); 2344 FastTrack *fastTrack = &state->mFastTracks[0]; 2345 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2346 delete fastTrack->mBufferProvider; 2347 sq->end(false /*didModify*/); 2348 delete mFastMixer; 2349 if (mAudioWatchdog != 0) { 2350 mAudioWatchdog->requestExit(); 2351 mAudioWatchdog->requestExitAndWait(); 2352 mAudioWatchdog.clear(); 2353 } 2354 } 2355 delete mAudioMixer; 2356} 2357 2358class CpuStats { 2359public: 2360 CpuStats(); 2361 void sample(const String8 &title); 2362#ifdef DEBUG_CPU_USAGE 2363private: 2364 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2365 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2366 2367 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2368 2369 int mCpuNum; // thread's current CPU number 2370 int mCpukHz; // frequency of thread's current CPU in kHz 2371#endif 2372}; 2373 2374CpuStats::CpuStats() 2375#ifdef DEBUG_CPU_USAGE 2376 : mCpuNum(-1), mCpukHz(-1) 2377#endif 2378{ 2379} 2380 2381void CpuStats::sample(const String8 &title) { 2382#ifdef DEBUG_CPU_USAGE 2383 // get current thread's delta CPU time in wall clock ns 2384 double wcNs; 2385 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2386 2387 // record sample for wall clock statistics 2388 if (valid) { 2389 mWcStats.sample(wcNs); 2390 } 2391 2392 // get the current CPU number 2393 int cpuNum = sched_getcpu(); 2394 2395 // get the current CPU frequency in kHz 2396 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2397 2398 // check if either CPU number or frequency changed 2399 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2400 mCpuNum = cpuNum; 2401 mCpukHz = cpukHz; 2402 // ignore sample for purposes of cycles 2403 valid = false; 2404 } 2405 2406 // if no change in CPU number or frequency, then record sample for cycle statistics 2407 if (valid && mCpukHz > 0) { 2408 double cycles = wcNs * cpukHz * 0.000001; 2409 mHzStats.sample(cycles); 2410 } 2411 2412 unsigned n = mWcStats.n(); 2413 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2414 if ((n & 127) == 1) { 2415 long long elapsed = mCpuUsage.elapsed(); 2416 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2417 double perLoop = elapsed / (double) n; 2418 double perLoop100 = perLoop * 0.01; 2419 double perLoop1k = perLoop * 0.001; 2420 double mean = mWcStats.mean(); 2421 double stddev = mWcStats.stddev(); 2422 double minimum = mWcStats.minimum(); 2423 double maximum = mWcStats.maximum(); 2424 double meanCycles = mHzStats.mean(); 2425 double stddevCycles = mHzStats.stddev(); 2426 double minCycles = mHzStats.minimum(); 2427 double maxCycles = mHzStats.maximum(); 2428 mCpuUsage.resetElapsed(); 2429 mWcStats.reset(); 2430 mHzStats.reset(); 2431 ALOGD("CPU usage for %s over past %.1f secs\n" 2432 " (%u mixer loops at %.1f mean ms per loop):\n" 2433 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2434 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2435 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2436 title.string(), 2437 elapsed * .000000001, n, perLoop * .000001, 2438 mean * .001, 2439 stddev * .001, 2440 minimum * .001, 2441 maximum * .001, 2442 mean / perLoop100, 2443 stddev / perLoop100, 2444 minimum / perLoop100, 2445 maximum / perLoop100, 2446 meanCycles / perLoop1k, 2447 stddevCycles / perLoop1k, 2448 minCycles / perLoop1k, 2449 maxCycles / perLoop1k); 2450 2451 } 2452 } 2453#endif 2454}; 2455 2456void AudioFlinger::PlaybackThread::checkSilentMode_l() 2457{ 2458 if (!mMasterMute) { 2459 char value[PROPERTY_VALUE_MAX]; 2460 if (property_get("ro.audio.silent", value, "0") > 0) { 2461 char *endptr; 2462 unsigned long ul = strtoul(value, &endptr, 0); 2463 if (*endptr == '\0' && ul != 0) { 2464 ALOGD("Silence is golden"); 2465 // The setprop command will not allow a property to be changed after 2466 // the first time it is set, so we don't have to worry about un-muting. 2467 setMasterMute_l(true); 2468 } 2469 } 2470 } 2471} 2472 2473bool AudioFlinger::PlaybackThread::threadLoop() 2474{ 2475 Vector< sp<Track> > tracksToRemove; 2476 2477 standbyTime = systemTime(); 2478 2479 // MIXER 2480 nsecs_t lastWarning = 0; 2481 2482 // DUPLICATING 2483 // FIXME could this be made local to while loop? 2484 writeFrames = 0; 2485 2486 cacheParameters_l(); 2487 sleepTime = idleSleepTime; 2488 2489 if (mType == MIXER) { 2490 sleepTimeShift = 0; 2491 } 2492 2493 CpuStats cpuStats; 2494 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2495 2496 acquireWakeLock(); 2497 2498 while (!exitPending()) 2499 { 2500 cpuStats.sample(myName); 2501 2502 Vector< sp<EffectChain> > effectChains; 2503 2504 processConfigEvents(); 2505 2506 { // scope for mLock 2507 2508 Mutex::Autolock _l(mLock); 2509 2510 if (checkForNewParameters_l()) { 2511 cacheParameters_l(); 2512 } 2513 2514 saveOutputTracks(); 2515 2516 // put audio hardware into standby after short delay 2517 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2518 isSuspended())) { 2519 if (!mStandby) { 2520 2521 threadLoop_standby(); 2522 2523 mStandby = true; 2524 mBytesWritten = 0; 2525 } 2526 2527 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2528 // we're about to wait, flush the binder command buffer 2529 IPCThreadState::self()->flushCommands(); 2530 2531 clearOutputTracks(); 2532 2533 if (exitPending()) break; 2534 2535 releaseWakeLock_l(); 2536 // wait until we have something to do... 2537 ALOGV("%s going to sleep", myName.string()); 2538 mWaitWorkCV.wait(mLock); 2539 ALOGV("%s waking up", myName.string()); 2540 acquireWakeLock_l(); 2541 2542 mMixerStatus = MIXER_IDLE; 2543 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2544 2545 checkSilentMode_l(); 2546 2547 standbyTime = systemTime() + standbyDelay; 2548 sleepTime = idleSleepTime; 2549 if (mType == MIXER) { 2550 sleepTimeShift = 0; 2551 } 2552 2553 continue; 2554 } 2555 } 2556 2557 // mMixerStatusIgnoringFastTracks is also updated internally 2558 mMixerStatus = prepareTracks_l(&tracksToRemove); 2559 2560 // prevent any changes in effect chain list and in each effect chain 2561 // during mixing and effect process as the audio buffers could be deleted 2562 // or modified if an effect is created or deleted 2563 lockEffectChains_l(effectChains); 2564 } 2565 2566 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2567 threadLoop_mix(); 2568 } else { 2569 threadLoop_sleepTime(); 2570 } 2571 2572 if (isSuspended()) { 2573 sleepTime = suspendSleepTimeUs(); 2574 } 2575 2576 // only process effects if we're going to write 2577 if (sleepTime == 0) { 2578 for (size_t i = 0; i < effectChains.size(); i ++) { 2579 effectChains[i]->process_l(); 2580 } 2581 } 2582 2583 // enable changes in effect chain 2584 unlockEffectChains(effectChains); 2585 2586 // sleepTime == 0 means we must write to audio hardware 2587 if (sleepTime == 0) { 2588 2589 threadLoop_write(); 2590 2591if (mType == MIXER) { 2592 // write blocked detection 2593 nsecs_t now = systemTime(); 2594 nsecs_t delta = now - mLastWriteTime; 2595 if (!mStandby && delta > maxPeriod) { 2596 mNumDelayedWrites++; 2597 if ((now - lastWarning) > kWarningThrottleNs) { 2598#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2599 ScopedTrace st(ATRACE_TAG, "underrun"); 2600#endif 2601 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2602 ns2ms(delta), mNumDelayedWrites, this); 2603 lastWarning = now; 2604 } 2605 } 2606} 2607 2608 mStandby = false; 2609 } else { 2610 usleep(sleepTime); 2611 } 2612 2613 // Finally let go of removed track(s), without the lock held 2614 // since we can't guarantee the destructors won't acquire that 2615 // same lock. This will also mutate and push a new fast mixer state. 2616 threadLoop_removeTracks(tracksToRemove); 2617 tracksToRemove.clear(); 2618 2619 // FIXME I don't understand the need for this here; 2620 // it was in the original code but maybe the 2621 // assignment in saveOutputTracks() makes this unnecessary? 2622 clearOutputTracks(); 2623 2624 // Effect chains will be actually deleted here if they were removed from 2625 // mEffectChains list during mixing or effects processing 2626 effectChains.clear(); 2627 2628 // FIXME Note that the above .clear() is no longer necessary since effectChains 2629 // is now local to this block, but will keep it for now (at least until merge done). 2630 } 2631 2632 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2633 if (mType == MIXER || mType == DIRECT) { 2634 // put output stream into standby mode 2635 if (!mStandby) { 2636 mOutput->stream->common.standby(&mOutput->stream->common); 2637 } 2638 } 2639 2640 releaseWakeLock(); 2641 2642 ALOGV("Thread %p type %d exiting", this, mType); 2643 return false; 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2647{ 2648 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2649} 2650 2651void AudioFlinger::MixerThread::threadLoop_write() 2652{ 2653 // FIXME we should only do one push per cycle; confirm this is true 2654 // Start the fast mixer if it's not already running 2655 if (mFastMixer != NULL) { 2656 FastMixerStateQueue *sq = mFastMixer->sq(); 2657 FastMixerState *state = sq->begin(); 2658 if (state->mCommand != FastMixerState::MIX_WRITE && 2659 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2660 if (state->mCommand == FastMixerState::COLD_IDLE) { 2661 int32_t old = android_atomic_inc(&mFastMixerFutex); 2662 if (old == -1) { 2663 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2664 } 2665 if (mAudioWatchdog != 0) { 2666 mAudioWatchdog->resume(); 2667 } 2668 } 2669 state->mCommand = FastMixerState::MIX_WRITE; 2670 sq->end(); 2671 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2672 if (kUseFastMixer == FastMixer_Dynamic) { 2673 mNormalSink = mPipeSink; 2674 } 2675 } else { 2676 sq->end(false /*didModify*/); 2677 } 2678 } 2679 PlaybackThread::threadLoop_write(); 2680} 2681 2682// shared by MIXER and DIRECT, overridden by DUPLICATING 2683void AudioFlinger::PlaybackThread::threadLoop_write() 2684{ 2685 // FIXME rewrite to reduce number of system calls 2686 mLastWriteTime = systemTime(); 2687 mInWrite = true; 2688 int bytesWritten; 2689 2690 // If an NBAIO sink is present, use it to write the normal mixer's submix 2691 if (mNormalSink != 0) { 2692#define mBitShift 2 // FIXME 2693 size_t count = mixBufferSize >> mBitShift; 2694#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2695 Tracer::traceBegin(ATRACE_TAG, "write"); 2696#endif 2697 // update the setpoint when gScreenState changes 2698 uint32_t screenState = gScreenState; 2699 if (screenState != mScreenState) { 2700 mScreenState = screenState; 2701 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2702 if (pipe != NULL) { 2703 pipe->setAvgFrames((mScreenState & 1) ? 2704 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2705 } 2706 } 2707 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2708#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2709 Tracer::traceEnd(ATRACE_TAG); 2710#endif 2711 if (framesWritten > 0) { 2712 bytesWritten = framesWritten << mBitShift; 2713 } else { 2714 bytesWritten = framesWritten; 2715 } 2716 // otherwise use the HAL / AudioStreamOut directly 2717 } else { 2718 // Direct output thread. 2719 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2720 } 2721 2722 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2723 mNumWrites++; 2724 mInWrite = false; 2725} 2726 2727void AudioFlinger::MixerThread::threadLoop_standby() 2728{ 2729 // Idle the fast mixer if it's currently running 2730 if (mFastMixer != NULL) { 2731 FastMixerStateQueue *sq = mFastMixer->sq(); 2732 FastMixerState *state = sq->begin(); 2733 if (!(state->mCommand & FastMixerState::IDLE)) { 2734 state->mCommand = FastMixerState::COLD_IDLE; 2735 state->mColdFutexAddr = &mFastMixerFutex; 2736 state->mColdGen++; 2737 mFastMixerFutex = 0; 2738 sq->end(); 2739 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2740 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2741 if (kUseFastMixer == FastMixer_Dynamic) { 2742 mNormalSink = mOutputSink; 2743 } 2744 if (mAudioWatchdog != 0) { 2745 mAudioWatchdog->pause(); 2746 } 2747 } else { 2748 sq->end(false /*didModify*/); 2749 } 2750 } 2751 PlaybackThread::threadLoop_standby(); 2752} 2753 2754// shared by MIXER and DIRECT, overridden by DUPLICATING 2755void AudioFlinger::PlaybackThread::threadLoop_standby() 2756{ 2757 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2758 mOutput->stream->common.standby(&mOutput->stream->common); 2759} 2760 2761void AudioFlinger::MixerThread::threadLoop_mix() 2762{ 2763 // obtain the presentation timestamp of the next output buffer 2764 int64_t pts; 2765 status_t status = INVALID_OPERATION; 2766 2767 if (NULL != mOutput->stream->get_next_write_timestamp) { 2768 status = mOutput->stream->get_next_write_timestamp( 2769 mOutput->stream, &pts); 2770 } 2771 2772 if (status != NO_ERROR) { 2773 pts = AudioBufferProvider::kInvalidPTS; 2774 } 2775 2776 // mix buffers... 2777 mAudioMixer->process(pts); 2778 // increase sleep time progressively when application underrun condition clears. 2779 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2780 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2781 // such that we would underrun the audio HAL. 2782 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2783 sleepTimeShift--; 2784 } 2785 sleepTime = 0; 2786 standbyTime = systemTime() + standbyDelay; 2787 //TODO: delay standby when effects have a tail 2788} 2789 2790void AudioFlinger::MixerThread::threadLoop_sleepTime() 2791{ 2792 // If no tracks are ready, sleep once for the duration of an output 2793 // buffer size, then write 0s to the output 2794 if (sleepTime == 0) { 2795 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2796 sleepTime = activeSleepTime >> sleepTimeShift; 2797 if (sleepTime < kMinThreadSleepTimeUs) { 2798 sleepTime = kMinThreadSleepTimeUs; 2799 } 2800 // reduce sleep time in case of consecutive application underruns to avoid 2801 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2802 // duration we would end up writing less data than needed by the audio HAL if 2803 // the condition persists. 2804 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2805 sleepTimeShift++; 2806 } 2807 } else { 2808 sleepTime = idleSleepTime; 2809 } 2810 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2811 memset (mMixBuffer, 0, mixBufferSize); 2812 sleepTime = 0; 2813 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2814 } 2815 // TODO add standby time extension fct of effect tail 2816} 2817 2818// prepareTracks_l() must be called with ThreadBase::mLock held 2819AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2820 Vector< sp<Track> > *tracksToRemove) 2821{ 2822 2823 mixer_state mixerStatus = MIXER_IDLE; 2824 // find out which tracks need to be processed 2825 size_t count = mActiveTracks.size(); 2826 size_t mixedTracks = 0; 2827 size_t tracksWithEffect = 0; 2828 // counts only _active_ fast tracks 2829 size_t fastTracks = 0; 2830 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2831 2832 float masterVolume = mMasterVolume; 2833 bool masterMute = mMasterMute; 2834 2835 if (masterMute) { 2836 masterVolume = 0; 2837 } 2838 // Delegate master volume control to effect in output mix effect chain if needed 2839 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2840 if (chain != 0) { 2841 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2842 chain->setVolume_l(&v, &v); 2843 masterVolume = (float)((v + (1 << 23)) >> 24); 2844 chain.clear(); 2845 } 2846 2847 // prepare a new state to push 2848 FastMixerStateQueue *sq = NULL; 2849 FastMixerState *state = NULL; 2850 bool didModify = false; 2851 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2852 if (mFastMixer != NULL) { 2853 sq = mFastMixer->sq(); 2854 state = sq->begin(); 2855 } 2856 2857 for (size_t i=0 ; i<count ; i++) { 2858 sp<Track> t = mActiveTracks[i].promote(); 2859 if (t == 0) continue; 2860 2861 // this const just means the local variable doesn't change 2862 Track* const track = t.get(); 2863 2864 // process fast tracks 2865 if (track->isFastTrack()) { 2866 2867 // It's theoretically possible (though unlikely) for a fast track to be created 2868 // and then removed within the same normal mix cycle. This is not a problem, as 2869 // the track never becomes active so it's fast mixer slot is never touched. 2870 // The converse, of removing an (active) track and then creating a new track 2871 // at the identical fast mixer slot within the same normal mix cycle, 2872 // is impossible because the slot isn't marked available until the end of each cycle. 2873 int j = track->mFastIndex; 2874 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2875 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2876 FastTrack *fastTrack = &state->mFastTracks[j]; 2877 2878 // Determine whether the track is currently in underrun condition, 2879 // and whether it had a recent underrun. 2880 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2881 FastTrackUnderruns underruns = ftDump->mUnderruns; 2882 uint32_t recentFull = (underruns.mBitFields.mFull - 2883 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2884 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2885 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2886 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2887 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2888 uint32_t recentUnderruns = recentPartial + recentEmpty; 2889 track->mObservedUnderruns = underruns; 2890 // don't count underruns that occur while stopping or pausing 2891 // or stopped which can occur when flush() is called while active 2892 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2893 track->mUnderrunCount += recentUnderruns; 2894 } 2895 2896 // This is similar to the state machine for normal tracks, 2897 // with a few modifications for fast tracks. 2898 bool isActive = true; 2899 switch (track->mState) { 2900 case TrackBase::STOPPING_1: 2901 // track stays active in STOPPING_1 state until first underrun 2902 if (recentUnderruns > 0) { 2903 track->mState = TrackBase::STOPPING_2; 2904 } 2905 break; 2906 case TrackBase::PAUSING: 2907 // ramp down is not yet implemented 2908 track->setPaused(); 2909 break; 2910 case TrackBase::RESUMING: 2911 // ramp up is not yet implemented 2912 track->mState = TrackBase::ACTIVE; 2913 break; 2914 case TrackBase::ACTIVE: 2915 if (recentFull > 0 || recentPartial > 0) { 2916 // track has provided at least some frames recently: reset retry count 2917 track->mRetryCount = kMaxTrackRetries; 2918 } 2919 if (recentUnderruns == 0) { 2920 // no recent underruns: stay active 2921 break; 2922 } 2923 // there has recently been an underrun of some kind 2924 if (track->sharedBuffer() == 0) { 2925 // were any of the recent underruns "empty" (no frames available)? 2926 if (recentEmpty == 0) { 2927 // no, then ignore the partial underruns as they are allowed indefinitely 2928 break; 2929 } 2930 // there has recently been an "empty" underrun: decrement the retry counter 2931 if (--(track->mRetryCount) > 0) { 2932 break; 2933 } 2934 // indicate to client process that the track was disabled because of underrun; 2935 // it will then automatically call start() when data is available 2936 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2937 // remove from active list, but state remains ACTIVE [confusing but true] 2938 isActive = false; 2939 break; 2940 } 2941 // fall through 2942 case TrackBase::STOPPING_2: 2943 case TrackBase::PAUSED: 2944 case TrackBase::TERMINATED: 2945 case TrackBase::STOPPED: 2946 case TrackBase::FLUSHED: // flush() while active 2947 // Check for presentation complete if track is inactive 2948 // We have consumed all the buffers of this track. 2949 // This would be incomplete if we auto-paused on underrun 2950 { 2951 size_t audioHALFrames = 2952 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2953 size_t framesWritten = 2954 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2955 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2956 // track stays in active list until presentation is complete 2957 break; 2958 } 2959 } 2960 if (track->isStopping_2()) { 2961 track->mState = TrackBase::STOPPED; 2962 } 2963 if (track->isStopped()) { 2964 // Can't reset directly, as fast mixer is still polling this track 2965 // track->reset(); 2966 // So instead mark this track as needing to be reset after push with ack 2967 resetMask |= 1 << i; 2968 } 2969 isActive = false; 2970 break; 2971 case TrackBase::IDLE: 2972 default: 2973 LOG_FATAL("unexpected track state %d", track->mState); 2974 } 2975 2976 if (isActive) { 2977 // was it previously inactive? 2978 if (!(state->mTrackMask & (1 << j))) { 2979 ExtendedAudioBufferProvider *eabp = track; 2980 VolumeProvider *vp = track; 2981 fastTrack->mBufferProvider = eabp; 2982 fastTrack->mVolumeProvider = vp; 2983 fastTrack->mSampleRate = track->mSampleRate; 2984 fastTrack->mChannelMask = track->mChannelMask; 2985 fastTrack->mGeneration++; 2986 state->mTrackMask |= 1 << j; 2987 didModify = true; 2988 // no acknowledgement required for newly active tracks 2989 } 2990 // cache the combined master volume and stream type volume for fast mixer; this 2991 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2992 track->mCachedVolume = track->isMuted() ? 2993 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2994 ++fastTracks; 2995 } else { 2996 // was it previously active? 2997 if (state->mTrackMask & (1 << j)) { 2998 fastTrack->mBufferProvider = NULL; 2999 fastTrack->mGeneration++; 3000 state->mTrackMask &= ~(1 << j); 3001 didModify = true; 3002 // If any fast tracks were removed, we must wait for acknowledgement 3003 // because we're about to decrement the last sp<> on those tracks. 3004 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3005 } else { 3006 LOG_FATAL("fast track %d should have been active", j); 3007 } 3008 tracksToRemove->add(track); 3009 // Avoids a misleading display in dumpsys 3010 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3011 } 3012 continue; 3013 } 3014 3015 { // local variable scope to avoid goto warning 3016 3017 audio_track_cblk_t* cblk = track->cblk(); 3018 3019 // The first time a track is added we wait 3020 // for all its buffers to be filled before processing it 3021 int name = track->name(); 3022 // make sure that we have enough frames to mix one full buffer. 3023 // enforce this condition only once to enable draining the buffer in case the client 3024 // app does not call stop() and relies on underrun to stop: 3025 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3026 // during last round 3027 uint32_t minFrames = 1; 3028 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3029 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3030 if (t->sampleRate() == (int)mSampleRate) { 3031 minFrames = mNormalFrameCount; 3032 } else { 3033 // +1 for rounding and +1 for additional sample needed for interpolation 3034 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3035 // add frames already consumed but not yet released by the resampler 3036 // because cblk->framesReady() will include these frames 3037 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3038 // the minimum track buffer size is normally twice the number of frames necessary 3039 // to fill one buffer and the resampler should not leave more than one buffer worth 3040 // of unreleased frames after each pass, but just in case... 3041 ALOG_ASSERT(minFrames <= cblk->frameCount); 3042 } 3043 } 3044 if ((track->framesReady() >= minFrames) && track->isReady() && 3045 !track->isPaused() && !track->isTerminated()) 3046 { 3047 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3048 3049 mixedTracks++; 3050 3051 // track->mainBuffer() != mMixBuffer means there is an effect chain 3052 // connected to the track 3053 chain.clear(); 3054 if (track->mainBuffer() != mMixBuffer) { 3055 chain = getEffectChain_l(track->sessionId()); 3056 // Delegate volume control to effect in track effect chain if needed 3057 if (chain != 0) { 3058 tracksWithEffect++; 3059 } else { 3060 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3061 name, track->sessionId()); 3062 } 3063 } 3064 3065 3066 int param = AudioMixer::VOLUME; 3067 if (track->mFillingUpStatus == Track::FS_FILLED) { 3068 // no ramp for the first volume setting 3069 track->mFillingUpStatus = Track::FS_ACTIVE; 3070 if (track->mState == TrackBase::RESUMING) { 3071 track->mState = TrackBase::ACTIVE; 3072 param = AudioMixer::RAMP_VOLUME; 3073 } 3074 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3075 } else if (cblk->server != 0) { 3076 // If the track is stopped before the first frame was mixed, 3077 // do not apply ramp 3078 param = AudioMixer::RAMP_VOLUME; 3079 } 3080 3081 // compute volume for this track 3082 uint32_t vl, vr, va; 3083 if (track->isMuted() || track->isPausing() || 3084 mStreamTypes[track->streamType()].mute) { 3085 vl = vr = va = 0; 3086 if (track->isPausing()) { 3087 track->setPaused(); 3088 } 3089 } else { 3090 3091 // read original volumes with volume control 3092 float typeVolume = mStreamTypes[track->streamType()].volume; 3093 float v = masterVolume * typeVolume; 3094 uint32_t vlr = cblk->getVolumeLR(); 3095 vl = vlr & 0xFFFF; 3096 vr = vlr >> 16; 3097 // track volumes come from shared memory, so can't be trusted and must be clamped 3098 if (vl > MAX_GAIN_INT) { 3099 ALOGV("Track left volume out of range: %04X", vl); 3100 vl = MAX_GAIN_INT; 3101 } 3102 if (vr > MAX_GAIN_INT) { 3103 ALOGV("Track right volume out of range: %04X", vr); 3104 vr = MAX_GAIN_INT; 3105 } 3106 // now apply the master volume and stream type volume 3107 vl = (uint32_t)(v * vl) << 12; 3108 vr = (uint32_t)(v * vr) << 12; 3109 // assuming master volume and stream type volume each go up to 1.0, 3110 // vl and vr are now in 8.24 format 3111 3112 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3113 // send level comes from shared memory and so may be corrupt 3114 if (sendLevel > MAX_GAIN_INT) { 3115 ALOGV("Track send level out of range: %04X", sendLevel); 3116 sendLevel = MAX_GAIN_INT; 3117 } 3118 va = (uint32_t)(v * sendLevel); 3119 } 3120 // Delegate volume control to effect in track effect chain if needed 3121 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3122 // Do not ramp volume if volume is controlled by effect 3123 param = AudioMixer::VOLUME; 3124 track->mHasVolumeController = true; 3125 } else { 3126 // force no volume ramp when volume controller was just disabled or removed 3127 // from effect chain to avoid volume spike 3128 if (track->mHasVolumeController) { 3129 param = AudioMixer::VOLUME; 3130 } 3131 track->mHasVolumeController = false; 3132 } 3133 3134 // Convert volumes from 8.24 to 4.12 format 3135 // This additional clamping is needed in case chain->setVolume_l() overshot 3136 vl = (vl + (1 << 11)) >> 12; 3137 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3138 vr = (vr + (1 << 11)) >> 12; 3139 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3140 3141 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3142 3143 // XXX: these things DON'T need to be done each time 3144 mAudioMixer->setBufferProvider(name, track); 3145 mAudioMixer->enable(name); 3146 3147 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3148 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3149 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3150 mAudioMixer->setParameter( 3151 name, 3152 AudioMixer::TRACK, 3153 AudioMixer::FORMAT, (void *)track->format()); 3154 mAudioMixer->setParameter( 3155 name, 3156 AudioMixer::TRACK, 3157 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3158 mAudioMixer->setParameter( 3159 name, 3160 AudioMixer::RESAMPLE, 3161 AudioMixer::SAMPLE_RATE, 3162 (void *)(cblk->sampleRate)); 3163 mAudioMixer->setParameter( 3164 name, 3165 AudioMixer::TRACK, 3166 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3167 mAudioMixer->setParameter( 3168 name, 3169 AudioMixer::TRACK, 3170 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3171 3172 // reset retry count 3173 track->mRetryCount = kMaxTrackRetries; 3174 3175 // If one track is ready, set the mixer ready if: 3176 // - the mixer was not ready during previous round OR 3177 // - no other track is not ready 3178 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3179 mixerStatus != MIXER_TRACKS_ENABLED) { 3180 mixerStatus = MIXER_TRACKS_READY; 3181 } 3182 } else { 3183 // clear effect chain input buffer if an active track underruns to avoid sending 3184 // previous audio buffer again to effects 3185 chain = getEffectChain_l(track->sessionId()); 3186 if (chain != 0) { 3187 chain->clearInputBuffer(); 3188 } 3189 3190 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3191 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3192 track->isStopped() || track->isPaused()) { 3193 // We have consumed all the buffers of this track. 3194 // Remove it from the list of active tracks. 3195 // TODO: use actual buffer filling status instead of latency when available from 3196 // audio HAL 3197 size_t audioHALFrames = 3198 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3199 size_t framesWritten = 3200 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3201 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3202 if (track->isStopped()) { 3203 track->reset(); 3204 } 3205 tracksToRemove->add(track); 3206 } 3207 } else { 3208 track->mUnderrunCount++; 3209 // No buffers for this track. Give it a few chances to 3210 // fill a buffer, then remove it from active list. 3211 if (--(track->mRetryCount) <= 0) { 3212 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3213 tracksToRemove->add(track); 3214 // indicate to client process that the track was disabled because of underrun; 3215 // it will then automatically call start() when data is available 3216 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3217 // If one track is not ready, mark the mixer also not ready if: 3218 // - the mixer was ready during previous round OR 3219 // - no other track is ready 3220 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3221 mixerStatus != MIXER_TRACKS_READY) { 3222 mixerStatus = MIXER_TRACKS_ENABLED; 3223 } 3224 } 3225 mAudioMixer->disable(name); 3226 } 3227 3228 } // local variable scope to avoid goto warning 3229track_is_ready: ; 3230 3231 } 3232 3233 // Push the new FastMixer state if necessary 3234 bool pauseAudioWatchdog = false; 3235 if (didModify) { 3236 state->mFastTracksGen++; 3237 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3238 if (kUseFastMixer == FastMixer_Dynamic && 3239 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3240 state->mCommand = FastMixerState::COLD_IDLE; 3241 state->mColdFutexAddr = &mFastMixerFutex; 3242 state->mColdGen++; 3243 mFastMixerFutex = 0; 3244 if (kUseFastMixer == FastMixer_Dynamic) { 3245 mNormalSink = mOutputSink; 3246 } 3247 // If we go into cold idle, need to wait for acknowledgement 3248 // so that fast mixer stops doing I/O. 3249 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3250 pauseAudioWatchdog = true; 3251 } 3252 sq->end(); 3253 } 3254 if (sq != NULL) { 3255 sq->end(didModify); 3256 sq->push(block); 3257 } 3258 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3259 mAudioWatchdog->pause(); 3260 } 3261 3262 // Now perform the deferred reset on fast tracks that have stopped 3263 while (resetMask != 0) { 3264 size_t i = __builtin_ctz(resetMask); 3265 ALOG_ASSERT(i < count); 3266 resetMask &= ~(1 << i); 3267 sp<Track> t = mActiveTracks[i].promote(); 3268 if (t == 0) continue; 3269 Track* track = t.get(); 3270 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3271 track->reset(); 3272 } 3273 3274 // remove all the tracks that need to be... 3275 count = tracksToRemove->size(); 3276 if (CC_UNLIKELY(count)) { 3277 for (size_t i=0 ; i<count ; i++) { 3278 const sp<Track>& track = tracksToRemove->itemAt(i); 3279 mActiveTracks.remove(track); 3280 if (track->mainBuffer() != mMixBuffer) { 3281 chain = getEffectChain_l(track->sessionId()); 3282 if (chain != 0) { 3283 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3284 chain->decActiveTrackCnt(); 3285 } 3286 } 3287 if (track->isTerminated()) { 3288 removeTrack_l(track); 3289 } 3290 } 3291 } 3292 3293 // mix buffer must be cleared if all tracks are connected to an 3294 // effect chain as in this case the mixer will not write to 3295 // mix buffer and track effects will accumulate into it 3296 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3297 // FIXME as a performance optimization, should remember previous zero status 3298 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3299 } 3300 3301 // if any fast tracks, then status is ready 3302 mMixerStatusIgnoringFastTracks = mixerStatus; 3303 if (fastTracks > 0) { 3304 mixerStatus = MIXER_TRACKS_READY; 3305 } 3306 return mixerStatus; 3307} 3308 3309/* 3310The derived values that are cached: 3311 - mixBufferSize from frame count * frame size 3312 - activeSleepTime from activeSleepTimeUs() 3313 - idleSleepTime from idleSleepTimeUs() 3314 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3315 - maxPeriod from frame count and sample rate (MIXER only) 3316 3317The parameters that affect these derived values are: 3318 - frame count 3319 - frame size 3320 - sample rate 3321 - device type: A2DP or not 3322 - device latency 3323 - format: PCM or not 3324 - active sleep time 3325 - idle sleep time 3326*/ 3327 3328void AudioFlinger::PlaybackThread::cacheParameters_l() 3329{ 3330 mixBufferSize = mNormalFrameCount * mFrameSize; 3331 activeSleepTime = activeSleepTimeUs(); 3332 idleSleepTime = idleSleepTimeUs(); 3333} 3334 3335void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3336{ 3337 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3338 this, streamType, mTracks.size()); 3339 Mutex::Autolock _l(mLock); 3340 3341 size_t size = mTracks.size(); 3342 for (size_t i = 0; i < size; i++) { 3343 sp<Track> t = mTracks[i]; 3344 if (t->streamType() == streamType) { 3345 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3346 t->mCblk->cv.signal(); 3347 } 3348 } 3349} 3350 3351// getTrackName_l() must be called with ThreadBase::mLock held 3352int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3353{ 3354 return mAudioMixer->getTrackName(channelMask); 3355} 3356 3357// deleteTrackName_l() must be called with ThreadBase::mLock held 3358void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3359{ 3360 ALOGV("remove track (%d) and delete from mixer", name); 3361 mAudioMixer->deleteTrackName(name); 3362} 3363 3364// checkForNewParameters_l() must be called with ThreadBase::mLock held 3365bool AudioFlinger::MixerThread::checkForNewParameters_l() 3366{ 3367 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3368 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3369 bool reconfig = false; 3370 3371 while (!mNewParameters.isEmpty()) { 3372 3373 if (mFastMixer != NULL) { 3374 FastMixerStateQueue *sq = mFastMixer->sq(); 3375 FastMixerState *state = sq->begin(); 3376 if (!(state->mCommand & FastMixerState::IDLE)) { 3377 previousCommand = state->mCommand; 3378 state->mCommand = FastMixerState::HOT_IDLE; 3379 sq->end(); 3380 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3381 } else { 3382 sq->end(false /*didModify*/); 3383 } 3384 } 3385 3386 status_t status = NO_ERROR; 3387 String8 keyValuePair = mNewParameters[0]; 3388 AudioParameter param = AudioParameter(keyValuePair); 3389 int value; 3390 3391 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3392 reconfig = true; 3393 } 3394 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3395 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3396 status = BAD_VALUE; 3397 } else { 3398 reconfig = true; 3399 } 3400 } 3401 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3402 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3403 status = BAD_VALUE; 3404 } else { 3405 reconfig = true; 3406 } 3407 } 3408 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3409 // do not accept frame count changes if tracks are open as the track buffer 3410 // size depends on frame count and correct behavior would not be guaranteed 3411 // if frame count is changed after track creation 3412 if (!mTracks.isEmpty()) { 3413 status = INVALID_OPERATION; 3414 } else { 3415 reconfig = true; 3416 } 3417 } 3418 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3419#ifdef ADD_BATTERY_DATA 3420 // when changing the audio output device, call addBatteryData to notify 3421 // the change 3422 if ((int)mDevice != value) { 3423 uint32_t params = 0; 3424 // check whether speaker is on 3425 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3426 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3427 } 3428 3429 audio_devices_t deviceWithoutSpeaker 3430 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3431 // check if any other device (except speaker) is on 3432 if (value & deviceWithoutSpeaker ) { 3433 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3434 } 3435 3436 if (params != 0) { 3437 addBatteryData(params); 3438 } 3439 } 3440#endif 3441 3442 // forward device change to effects that have requested to be 3443 // aware of attached audio device. 3444 mDevice = (audio_devices_t) value; 3445 for (size_t i = 0; i < mEffectChains.size(); i++) { 3446 mEffectChains[i]->setDevice_l(mDevice); 3447 } 3448 } 3449 3450 if (status == NO_ERROR) { 3451 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3452 keyValuePair.string()); 3453 if (!mStandby && status == INVALID_OPERATION) { 3454 mOutput->stream->common.standby(&mOutput->stream->common); 3455 mStandby = true; 3456 mBytesWritten = 0; 3457 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3458 keyValuePair.string()); 3459 } 3460 if (status == NO_ERROR && reconfig) { 3461 delete mAudioMixer; 3462 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3463 mAudioMixer = NULL; 3464 readOutputParameters(); 3465 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3466 for (size_t i = 0; i < mTracks.size() ; i++) { 3467 int name = getTrackName_l(mTracks[i]->mChannelMask); 3468 if (name < 0) break; 3469 mTracks[i]->mName = name; 3470 // limit track sample rate to 2 x new output sample rate 3471 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3472 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3473 } 3474 } 3475 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3476 } 3477 } 3478 3479 mNewParameters.removeAt(0); 3480 3481 mParamStatus = status; 3482 mParamCond.signal(); 3483 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3484 // already timed out waiting for the status and will never signal the condition. 3485 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3486 } 3487 3488 if (!(previousCommand & FastMixerState::IDLE)) { 3489 ALOG_ASSERT(mFastMixer != NULL); 3490 FastMixerStateQueue *sq = mFastMixer->sq(); 3491 FastMixerState *state = sq->begin(); 3492 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3493 state->mCommand = previousCommand; 3494 sq->end(); 3495 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3496 } 3497 3498 return reconfig; 3499} 3500 3501status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3502{ 3503 const size_t SIZE = 256; 3504 char buffer[SIZE]; 3505 String8 result; 3506 3507 PlaybackThread::dumpInternals(fd, args); 3508 3509 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3510 result.append(buffer); 3511 write(fd, result.string(), result.size()); 3512 3513 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3514 FastMixerDumpState copy = mFastMixerDumpState; 3515 copy.dump(fd); 3516 3517#ifdef STATE_QUEUE_DUMP 3518 // Similar for state queue 3519 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3520 observerCopy.dump(fd); 3521 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3522 mutatorCopy.dump(fd); 3523#endif 3524 3525 // Write the tee output to a .wav file 3526 NBAIO_Source *teeSource = mTeeSource.get(); 3527 if (teeSource != NULL) { 3528 char teePath[64]; 3529 struct timeval tv; 3530 gettimeofday(&tv, NULL); 3531 struct tm tm; 3532 localtime_r(&tv.tv_sec, &tm); 3533 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3534 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3535 if (teeFd >= 0) { 3536 char wavHeader[44]; 3537 memcpy(wavHeader, 3538 "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", 3539 sizeof(wavHeader)); 3540 NBAIO_Format format = teeSource->format(); 3541 unsigned channelCount = Format_channelCount(format); 3542 ALOG_ASSERT(channelCount <= FCC_2); 3543 unsigned sampleRate = Format_sampleRate(format); 3544 wavHeader[22] = channelCount; // number of channels 3545 wavHeader[24] = sampleRate; // sample rate 3546 wavHeader[25] = sampleRate >> 8; 3547 wavHeader[32] = channelCount * 2; // block alignment 3548 write(teeFd, wavHeader, sizeof(wavHeader)); 3549 size_t total = 0; 3550 bool firstRead = true; 3551 for (;;) { 3552#define TEE_SINK_READ 1024 3553 short buffer[TEE_SINK_READ * FCC_2]; 3554 size_t count = TEE_SINK_READ; 3555 ssize_t actual = teeSource->read(buffer, count); 3556 bool wasFirstRead = firstRead; 3557 firstRead = false; 3558 if (actual <= 0) { 3559 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3560 continue; 3561 } 3562 break; 3563 } 3564 ALOG_ASSERT(actual <= (ssize_t)count); 3565 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3566 total += actual; 3567 } 3568 lseek(teeFd, (off_t) 4, SEEK_SET); 3569 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3570 write(teeFd, &temp, sizeof(temp)); 3571 lseek(teeFd, (off_t) 40, SEEK_SET); 3572 temp = total * channelCount * sizeof(short); 3573 write(teeFd, &temp, sizeof(temp)); 3574 close(teeFd); 3575 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3576 } else { 3577 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3578 } 3579 } 3580 3581 if (mAudioWatchdog != 0) { 3582 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3583 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3584 wdCopy.dump(fd); 3585 } 3586 3587 return NO_ERROR; 3588} 3589 3590uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3591{ 3592 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3593} 3594 3595uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3596{ 3597 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3598} 3599 3600void AudioFlinger::MixerThread::cacheParameters_l() 3601{ 3602 PlaybackThread::cacheParameters_l(); 3603 3604 // FIXME: Relaxed timing because of a certain device that can't meet latency 3605 // Should be reduced to 2x after the vendor fixes the driver issue 3606 // increase threshold again due to low power audio mode. The way this warning 3607 // threshold is calculated and its usefulness should be reconsidered anyway. 3608 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3609} 3610 3611// ---------------------------------------------------------------------------- 3612AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3613 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3614 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3615 // mLeftVolFloat, mRightVolFloat 3616{ 3617} 3618 3619AudioFlinger::DirectOutputThread::~DirectOutputThread() 3620{ 3621} 3622 3623AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3624 Vector< sp<Track> > *tracksToRemove 3625) 3626{ 3627 sp<Track> trackToRemove; 3628 3629 mixer_state mixerStatus = MIXER_IDLE; 3630 3631 // find out which tracks need to be processed 3632 if (mActiveTracks.size() != 0) { 3633 sp<Track> t = mActiveTracks[0].promote(); 3634 // The track died recently 3635 if (t == 0) return MIXER_IDLE; 3636 3637 Track* const track = t.get(); 3638 audio_track_cblk_t* cblk = track->cblk(); 3639 3640 // The first time a track is added we wait 3641 // for all its buffers to be filled before processing it 3642 uint32_t minFrames; 3643 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3644 minFrames = mNormalFrameCount; 3645 } else { 3646 minFrames = 1; 3647 } 3648 if ((track->framesReady() >= minFrames) && track->isReady() && 3649 !track->isPaused() && !track->isTerminated()) 3650 { 3651 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3652 3653 if (track->mFillingUpStatus == Track::FS_FILLED) { 3654 track->mFillingUpStatus = Track::FS_ACTIVE; 3655 mLeftVolFloat = mRightVolFloat = 0; 3656 if (track->mState == TrackBase::RESUMING) { 3657 track->mState = TrackBase::ACTIVE; 3658 } 3659 } 3660 3661 // compute volume for this track 3662 float left, right; 3663 if (track->isMuted() || mMasterMute || track->isPausing() || 3664 mStreamTypes[track->streamType()].mute) { 3665 left = right = 0; 3666 if (track->isPausing()) { 3667 track->setPaused(); 3668 } 3669 } else { 3670 float typeVolume = mStreamTypes[track->streamType()].volume; 3671 float v = mMasterVolume * typeVolume; 3672 uint32_t vlr = cblk->getVolumeLR(); 3673 float v_clamped = v * (vlr & 0xFFFF); 3674 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3675 left = v_clamped/MAX_GAIN; 3676 v_clamped = v * (vlr >> 16); 3677 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3678 right = v_clamped/MAX_GAIN; 3679 } 3680 3681 if (left != mLeftVolFloat || right != mRightVolFloat) { 3682 mLeftVolFloat = left; 3683 mRightVolFloat = right; 3684 3685 // Convert volumes from float to 8.24 3686 uint32_t vl = (uint32_t)(left * (1 << 24)); 3687 uint32_t vr = (uint32_t)(right * (1 << 24)); 3688 3689 // Delegate volume control to effect in track effect chain if needed 3690 // only one effect chain can be present on DirectOutputThread, so if 3691 // there is one, the track is connected to it 3692 if (!mEffectChains.isEmpty()) { 3693 // Do not ramp volume if volume is controlled by effect 3694 mEffectChains[0]->setVolume_l(&vl, &vr); 3695 left = (float)vl / (1 << 24); 3696 right = (float)vr / (1 << 24); 3697 } 3698 mOutput->stream->set_volume(mOutput->stream, left, right); 3699 } 3700 3701 // reset retry count 3702 track->mRetryCount = kMaxTrackRetriesDirect; 3703 mActiveTrack = t; 3704 mixerStatus = MIXER_TRACKS_READY; 3705 } else { 3706 // clear effect chain input buffer if an active track underruns to avoid sending 3707 // previous audio buffer again to effects 3708 if (!mEffectChains.isEmpty()) { 3709 mEffectChains[0]->clearInputBuffer(); 3710 } 3711 3712 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3713 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3714 track->isStopped() || track->isPaused()) { 3715 // We have consumed all the buffers of this track. 3716 // Remove it from the list of active tracks. 3717 // TODO: implement behavior for compressed audio 3718 size_t audioHALFrames = 3719 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3720 size_t framesWritten = 3721 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3722 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3723 if (track->isStopped()) { 3724 track->reset(); 3725 } 3726 trackToRemove = track; 3727 } 3728 } else { 3729 // No buffers for this track. Give it a few chances to 3730 // fill a buffer, then remove it from active list. 3731 if (--(track->mRetryCount) <= 0) { 3732 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3733 trackToRemove = track; 3734 } else { 3735 mixerStatus = MIXER_TRACKS_ENABLED; 3736 } 3737 } 3738 } 3739 } 3740 3741 // FIXME merge this with similar code for removing multiple tracks 3742 // remove all the tracks that need to be... 3743 if (CC_UNLIKELY(trackToRemove != 0)) { 3744 tracksToRemove->add(trackToRemove); 3745 mActiveTracks.remove(trackToRemove); 3746 if (!mEffectChains.isEmpty()) { 3747 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3748 trackToRemove->sessionId()); 3749 mEffectChains[0]->decActiveTrackCnt(); 3750 } 3751 if (trackToRemove->isTerminated()) { 3752 removeTrack_l(trackToRemove); 3753 } 3754 } 3755 3756 return mixerStatus; 3757} 3758 3759void AudioFlinger::DirectOutputThread::threadLoop_mix() 3760{ 3761 AudioBufferProvider::Buffer buffer; 3762 size_t frameCount = mFrameCount; 3763 int8_t *curBuf = (int8_t *)mMixBuffer; 3764 // output audio to hardware 3765 while (frameCount) { 3766 buffer.frameCount = frameCount; 3767 mActiveTrack->getNextBuffer(&buffer); 3768 if (CC_UNLIKELY(buffer.raw == NULL)) { 3769 memset(curBuf, 0, frameCount * mFrameSize); 3770 break; 3771 } 3772 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3773 frameCount -= buffer.frameCount; 3774 curBuf += buffer.frameCount * mFrameSize; 3775 mActiveTrack->releaseBuffer(&buffer); 3776 } 3777 sleepTime = 0; 3778 standbyTime = systemTime() + standbyDelay; 3779 mActiveTrack.clear(); 3780 3781} 3782 3783void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3784{ 3785 if (sleepTime == 0) { 3786 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3787 sleepTime = activeSleepTime; 3788 } else { 3789 sleepTime = idleSleepTime; 3790 } 3791 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3792 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3793 sleepTime = 0; 3794 } 3795} 3796 3797// getTrackName_l() must be called with ThreadBase::mLock held 3798int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3799{ 3800 return 0; 3801} 3802 3803// deleteTrackName_l() must be called with ThreadBase::mLock held 3804void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3805{ 3806} 3807 3808// checkForNewParameters_l() must be called with ThreadBase::mLock held 3809bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3810{ 3811 bool reconfig = false; 3812 3813 while (!mNewParameters.isEmpty()) { 3814 status_t status = NO_ERROR; 3815 String8 keyValuePair = mNewParameters[0]; 3816 AudioParameter param = AudioParameter(keyValuePair); 3817 int value; 3818 3819 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3820 // do not accept frame count changes if tracks are open as the track buffer 3821 // size depends on frame count and correct behavior would not be garantied 3822 // if frame count is changed after track creation 3823 if (!mTracks.isEmpty()) { 3824 status = INVALID_OPERATION; 3825 } else { 3826 reconfig = true; 3827 } 3828 } 3829 if (status == NO_ERROR) { 3830 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3831 keyValuePair.string()); 3832 if (!mStandby && status == INVALID_OPERATION) { 3833 mOutput->stream->common.standby(&mOutput->stream->common); 3834 mStandby = true; 3835 mBytesWritten = 0; 3836 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3837 keyValuePair.string()); 3838 } 3839 if (status == NO_ERROR && reconfig) { 3840 readOutputParameters(); 3841 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3842 } 3843 } 3844 3845 mNewParameters.removeAt(0); 3846 3847 mParamStatus = status; 3848 mParamCond.signal(); 3849 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3850 // already timed out waiting for the status and will never signal the condition. 3851 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3852 } 3853 return reconfig; 3854} 3855 3856uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3857{ 3858 uint32_t time; 3859 if (audio_is_linear_pcm(mFormat)) { 3860 time = PlaybackThread::activeSleepTimeUs(); 3861 } else { 3862 time = 10000; 3863 } 3864 return time; 3865} 3866 3867uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3868{ 3869 uint32_t time; 3870 if (audio_is_linear_pcm(mFormat)) { 3871 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3872 } else { 3873 time = 10000; 3874 } 3875 return time; 3876} 3877 3878uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3879{ 3880 uint32_t time; 3881 if (audio_is_linear_pcm(mFormat)) { 3882 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3883 } else { 3884 time = 10000; 3885 } 3886 return time; 3887} 3888 3889void AudioFlinger::DirectOutputThread::cacheParameters_l() 3890{ 3891 PlaybackThread::cacheParameters_l(); 3892 3893 // use shorter standby delay as on normal output to release 3894 // hardware resources as soon as possible 3895 standbyDelay = microseconds(activeSleepTime*2); 3896} 3897 3898// ---------------------------------------------------------------------------- 3899 3900AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3901 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3902 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3903 mWaitTimeMs(UINT_MAX) 3904{ 3905 addOutputTrack(mainThread); 3906} 3907 3908AudioFlinger::DuplicatingThread::~DuplicatingThread() 3909{ 3910 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3911 mOutputTracks[i]->destroy(); 3912 } 3913} 3914 3915void AudioFlinger::DuplicatingThread::threadLoop_mix() 3916{ 3917 // mix buffers... 3918 if (outputsReady(outputTracks)) { 3919 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3920 } else { 3921 memset(mMixBuffer, 0, mixBufferSize); 3922 } 3923 sleepTime = 0; 3924 writeFrames = mNormalFrameCount; 3925 standbyTime = systemTime() + standbyDelay; 3926} 3927 3928void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3929{ 3930 if (sleepTime == 0) { 3931 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3932 sleepTime = activeSleepTime; 3933 } else { 3934 sleepTime = idleSleepTime; 3935 } 3936 } else if (mBytesWritten != 0) { 3937 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3938 writeFrames = mNormalFrameCount; 3939 memset(mMixBuffer, 0, mixBufferSize); 3940 } else { 3941 // flush remaining overflow buffers in output tracks 3942 writeFrames = 0; 3943 } 3944 sleepTime = 0; 3945 } 3946} 3947 3948void AudioFlinger::DuplicatingThread::threadLoop_write() 3949{ 3950 for (size_t i = 0; i < outputTracks.size(); i++) { 3951 outputTracks[i]->write(mMixBuffer, writeFrames); 3952 } 3953 mBytesWritten += mixBufferSize; 3954} 3955 3956void AudioFlinger::DuplicatingThread::threadLoop_standby() 3957{ 3958 // DuplicatingThread implements standby by stopping all tracks 3959 for (size_t i = 0; i < outputTracks.size(); i++) { 3960 outputTracks[i]->stop(); 3961 } 3962} 3963 3964void AudioFlinger::DuplicatingThread::saveOutputTracks() 3965{ 3966 outputTracks = mOutputTracks; 3967} 3968 3969void AudioFlinger::DuplicatingThread::clearOutputTracks() 3970{ 3971 outputTracks.clear(); 3972} 3973 3974void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3975{ 3976 Mutex::Autolock _l(mLock); 3977 // FIXME explain this formula 3978 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3979 OutputTrack *outputTrack = new OutputTrack(thread, 3980 this, 3981 mSampleRate, 3982 mFormat, 3983 mChannelMask, 3984 frameCount); 3985 if (outputTrack->cblk() != NULL) { 3986 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3987 mOutputTracks.add(outputTrack); 3988 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3989 updateWaitTime_l(); 3990 } 3991} 3992 3993void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3994{ 3995 Mutex::Autolock _l(mLock); 3996 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3997 if (mOutputTracks[i]->thread() == thread) { 3998 mOutputTracks[i]->destroy(); 3999 mOutputTracks.removeAt(i); 4000 updateWaitTime_l(); 4001 return; 4002 } 4003 } 4004 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4005} 4006 4007// caller must hold mLock 4008void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4009{ 4010 mWaitTimeMs = UINT_MAX; 4011 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4012 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4013 if (strong != 0) { 4014 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4015 if (waitTimeMs < mWaitTimeMs) { 4016 mWaitTimeMs = waitTimeMs; 4017 } 4018 } 4019 } 4020} 4021 4022 4023bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4024{ 4025 for (size_t i = 0; i < outputTracks.size(); i++) { 4026 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4027 if (thread == 0) { 4028 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4029 return false; 4030 } 4031 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4032 // see note at standby() declaration 4033 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4034 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4035 return false; 4036 } 4037 } 4038 return true; 4039} 4040 4041uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4042{ 4043 return (mWaitTimeMs * 1000) / 2; 4044} 4045 4046void AudioFlinger::DuplicatingThread::cacheParameters_l() 4047{ 4048 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4049 updateWaitTime_l(); 4050 4051 MixerThread::cacheParameters_l(); 4052} 4053 4054// ---------------------------------------------------------------------------- 4055 4056// TrackBase constructor must be called with AudioFlinger::mLock held 4057AudioFlinger::ThreadBase::TrackBase::TrackBase( 4058 ThreadBase *thread, 4059 const sp<Client>& client, 4060 uint32_t sampleRate, 4061 audio_format_t format, 4062 audio_channel_mask_t channelMask, 4063 int frameCount, 4064 const sp<IMemory>& sharedBuffer, 4065 int sessionId) 4066 : RefBase(), 4067 mThread(thread), 4068 mClient(client), 4069 mCblk(NULL), 4070 // mBuffer 4071 // mBufferEnd 4072 mFrameCount(0), 4073 mState(IDLE), 4074 mSampleRate(sampleRate), 4075 mFormat(format), 4076 mStepServerFailed(false), 4077 mSessionId(sessionId) 4078 // mChannelCount 4079 // mChannelMask 4080{ 4081 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4082 4083 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4084 size_t size = sizeof(audio_track_cblk_t); 4085 uint8_t channelCount = popcount(channelMask); 4086 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4087 if (sharedBuffer == 0) { 4088 size += bufferSize; 4089 } 4090 4091 if (client != NULL) { 4092 mCblkMemory = client->heap()->allocate(size); 4093 if (mCblkMemory != 0) { 4094 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4095 if (mCblk != NULL) { // construct the shared structure in-place. 4096 new(mCblk) audio_track_cblk_t(); 4097 // clear all buffers 4098 mCblk->frameCount = frameCount; 4099 mCblk->sampleRate = sampleRate; 4100// uncomment the following lines to quickly test 32-bit wraparound 4101// mCblk->user = 0xffff0000; 4102// mCblk->server = 0xffff0000; 4103// mCblk->userBase = 0xffff0000; 4104// mCblk->serverBase = 0xffff0000; 4105 mChannelCount = channelCount; 4106 mChannelMask = channelMask; 4107 if (sharedBuffer == 0) { 4108 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4109 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4110 // Force underrun condition to avoid false underrun callback until first data is 4111 // written to buffer (other flags are cleared) 4112 mCblk->flags = CBLK_UNDERRUN_ON; 4113 } else { 4114 mBuffer = sharedBuffer->pointer(); 4115 } 4116 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4117 } 4118 } else { 4119 ALOGE("not enough memory for AudioTrack size=%u", size); 4120 client->heap()->dump("AudioTrack"); 4121 return; 4122 } 4123 } else { 4124 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4125 // construct the shared structure in-place. 4126 new(mCblk) audio_track_cblk_t(); 4127 // clear all buffers 4128 mCblk->frameCount = frameCount; 4129 mCblk->sampleRate = sampleRate; 4130// uncomment the following lines to quickly test 32-bit wraparound 4131// mCblk->user = 0xffff0000; 4132// mCblk->server = 0xffff0000; 4133// mCblk->userBase = 0xffff0000; 4134// mCblk->serverBase = 0xffff0000; 4135 mChannelCount = channelCount; 4136 mChannelMask = channelMask; 4137 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4138 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4139 // Force underrun condition to avoid false underrun callback until first data is 4140 // written to buffer (other flags are cleared) 4141 mCblk->flags = CBLK_UNDERRUN_ON; 4142 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4143 } 4144} 4145 4146AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4147{ 4148 if (mCblk != NULL) { 4149 if (mClient == 0) { 4150 delete mCblk; 4151 } else { 4152 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4153 } 4154 } 4155 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4156 if (mClient != 0) { 4157 // Client destructor must run with AudioFlinger mutex locked 4158 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4159 // If the client's reference count drops to zero, the associated destructor 4160 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4161 // relying on the automatic clear() at end of scope. 4162 mClient.clear(); 4163 } 4164} 4165 4166// AudioBufferProvider interface 4167// getNextBuffer() = 0; 4168// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4169void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4170{ 4171 buffer->raw = NULL; 4172 mFrameCount = buffer->frameCount; 4173 // FIXME See note at getNextBuffer() 4174 (void) step(); // ignore return value of step() 4175 buffer->frameCount = 0; 4176} 4177 4178bool AudioFlinger::ThreadBase::TrackBase::step() { 4179 bool result; 4180 audio_track_cblk_t* cblk = this->cblk(); 4181 4182 result = cblk->stepServer(mFrameCount); 4183 if (!result) { 4184 ALOGV("stepServer failed acquiring cblk mutex"); 4185 mStepServerFailed = true; 4186 } 4187 return result; 4188} 4189 4190void AudioFlinger::ThreadBase::TrackBase::reset() { 4191 audio_track_cblk_t* cblk = this->cblk(); 4192 4193 cblk->user = 0; 4194 cblk->server = 0; 4195 cblk->userBase = 0; 4196 cblk->serverBase = 0; 4197 mStepServerFailed = false; 4198 ALOGV("TrackBase::reset"); 4199} 4200 4201int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4202 return (int)mCblk->sampleRate; 4203} 4204 4205void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4206 audio_track_cblk_t* cblk = this->cblk(); 4207 size_t frameSize = cblk->frameSize; 4208 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4209 int8_t *bufferEnd = bufferStart + frames * frameSize; 4210 4211 // Check validity of returned pointer in case the track control block would have been corrupted. 4212 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4213 "TrackBase::getBuffer buffer out of range:\n" 4214 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4215 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4216 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4217 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4218 4219 return bufferStart; 4220} 4221 4222status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4223{ 4224 mSyncEvents.add(event); 4225 return NO_ERROR; 4226} 4227 4228// ---------------------------------------------------------------------------- 4229 4230// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4231AudioFlinger::PlaybackThread::Track::Track( 4232 PlaybackThread *thread, 4233 const sp<Client>& client, 4234 audio_stream_type_t streamType, 4235 uint32_t sampleRate, 4236 audio_format_t format, 4237 audio_channel_mask_t channelMask, 4238 int frameCount, 4239 const sp<IMemory>& sharedBuffer, 4240 int sessionId, 4241 IAudioFlinger::track_flags_t flags) 4242 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4243 mMute(false), 4244 mFillingUpStatus(FS_INVALID), 4245 // mRetryCount initialized later when needed 4246 mSharedBuffer(sharedBuffer), 4247 mStreamType(streamType), 4248 mName(-1), // see note below 4249 mMainBuffer(thread->mixBuffer()), 4250 mAuxBuffer(NULL), 4251 mAuxEffectId(0), mHasVolumeController(false), 4252 mPresentationCompleteFrames(0), 4253 mFlags(flags), 4254 mFastIndex(-1), 4255 mUnderrunCount(0), 4256 mCachedVolume(1.0) 4257{ 4258 if (mCblk != NULL) { 4259 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4260 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4261 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4262 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4263 mName = thread->getTrackName_l(channelMask); 4264 mCblk->mName = mName; 4265 if (mName < 0) { 4266 ALOGE("no more track names available"); 4267 return; 4268 } 4269 // only allocate a fast track index if we were able to allocate a normal track name 4270 if (flags & IAudioFlinger::TRACK_FAST) { 4271 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4272 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4273 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4274 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4275 // FIXME This is too eager. We allocate a fast track index before the 4276 // fast track becomes active. Since fast tracks are a scarce resource, 4277 // this means we are potentially denying other more important fast tracks from 4278 // being created. It would be better to allocate the index dynamically. 4279 mFastIndex = i; 4280 mCblk->mName = i; 4281 // Read the initial underruns because this field is never cleared by the fast mixer 4282 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4283 thread->mFastTrackAvailMask &= ~(1 << i); 4284 } 4285 } 4286 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4287} 4288 4289AudioFlinger::PlaybackThread::Track::~Track() 4290{ 4291 ALOGV("PlaybackThread::Track destructor"); 4292 sp<ThreadBase> thread = mThread.promote(); 4293 if (thread != 0) { 4294 Mutex::Autolock _l(thread->mLock); 4295 mState = TERMINATED; 4296 } 4297} 4298 4299void AudioFlinger::PlaybackThread::Track::destroy() 4300{ 4301 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4302 // by removing it from mTracks vector, so there is a risk that this Tracks's 4303 // destructor is called. As the destructor needs to lock mLock, 4304 // we must acquire a strong reference on this Track before locking mLock 4305 // here so that the destructor is called only when exiting this function. 4306 // On the other hand, as long as Track::destroy() is only called by 4307 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4308 // this Track with its member mTrack. 4309 sp<Track> keep(this); 4310 { // scope for mLock 4311 sp<ThreadBase> thread = mThread.promote(); 4312 if (thread != 0) { 4313 if (!isOutputTrack()) { 4314 if (mState == ACTIVE || mState == RESUMING) { 4315 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4316 4317#ifdef ADD_BATTERY_DATA 4318 // to track the speaker usage 4319 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4320#endif 4321 } 4322 AudioSystem::releaseOutput(thread->id()); 4323 } 4324 Mutex::Autolock _l(thread->mLock); 4325 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4326 playbackThread->destroyTrack_l(this); 4327 } 4328 } 4329} 4330 4331/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4332{ 4333 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4334 " Server User Main buf Aux Buf Flags Underruns\n"); 4335} 4336 4337void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4338{ 4339 uint32_t vlr = mCblk->getVolumeLR(); 4340 if (isFastTrack()) { 4341 sprintf(buffer, " F %2d", mFastIndex); 4342 } else { 4343 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4344 } 4345 track_state state = mState; 4346 char stateChar; 4347 switch (state) { 4348 case IDLE: 4349 stateChar = 'I'; 4350 break; 4351 case TERMINATED: 4352 stateChar = 'T'; 4353 break; 4354 case STOPPING_1: 4355 stateChar = 's'; 4356 break; 4357 case STOPPING_2: 4358 stateChar = '5'; 4359 break; 4360 case STOPPED: 4361 stateChar = 'S'; 4362 break; 4363 case RESUMING: 4364 stateChar = 'R'; 4365 break; 4366 case ACTIVE: 4367 stateChar = 'A'; 4368 break; 4369 case PAUSING: 4370 stateChar = 'p'; 4371 break; 4372 case PAUSED: 4373 stateChar = 'P'; 4374 break; 4375 case FLUSHED: 4376 stateChar = 'F'; 4377 break; 4378 default: 4379 stateChar = '?'; 4380 break; 4381 } 4382 char nowInUnderrun; 4383 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4384 case UNDERRUN_FULL: 4385 nowInUnderrun = ' '; 4386 break; 4387 case UNDERRUN_PARTIAL: 4388 nowInUnderrun = '<'; 4389 break; 4390 case UNDERRUN_EMPTY: 4391 nowInUnderrun = '*'; 4392 break; 4393 default: 4394 nowInUnderrun = '?'; 4395 break; 4396 } 4397 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4398 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4399 (mClient == 0) ? getpid_cached : mClient->pid(), 4400 mStreamType, 4401 mFormat, 4402 mChannelMask, 4403 mSessionId, 4404 mFrameCount, 4405 mCblk->frameCount, 4406 stateChar, 4407 mMute, 4408 mFillingUpStatus, 4409 mCblk->sampleRate, 4410 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4411 20.0 * log10((vlr >> 16) / 4096.0), 4412 mCblk->server, 4413 mCblk->user, 4414 (int)mMainBuffer, 4415 (int)mAuxBuffer, 4416 mCblk->flags, 4417 mUnderrunCount, 4418 nowInUnderrun); 4419} 4420 4421// AudioBufferProvider interface 4422status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4423 AudioBufferProvider::Buffer* buffer, int64_t pts) 4424{ 4425 audio_track_cblk_t* cblk = this->cblk(); 4426 uint32_t framesReady; 4427 uint32_t framesReq = buffer->frameCount; 4428 4429 // Check if last stepServer failed, try to step now 4430 if (mStepServerFailed) { 4431 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4432 // Since the fast mixer is higher priority than client callback thread, 4433 // it does not result in priority inversion for client. 4434 // But a non-blocking solution would be preferable to avoid 4435 // fast mixer being unable to tryLock(), and 4436 // to avoid the extra context switches if the client wakes up, 4437 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4438 if (!step()) goto getNextBuffer_exit; 4439 ALOGV("stepServer recovered"); 4440 mStepServerFailed = false; 4441 } 4442 4443 // FIXME Same as above 4444 framesReady = cblk->framesReady(); 4445 4446 if (CC_LIKELY(framesReady)) { 4447 uint32_t s = cblk->server; 4448 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4449 4450 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4451 if (framesReq > framesReady) { 4452 framesReq = framesReady; 4453 } 4454 if (framesReq > bufferEnd - s) { 4455 framesReq = bufferEnd - s; 4456 } 4457 4458 buffer->raw = getBuffer(s, framesReq); 4459 buffer->frameCount = framesReq; 4460 return NO_ERROR; 4461 } 4462 4463getNextBuffer_exit: 4464 buffer->raw = NULL; 4465 buffer->frameCount = 0; 4466 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4467 return NOT_ENOUGH_DATA; 4468} 4469 4470// Note that framesReady() takes a mutex on the control block using tryLock(). 4471// This could result in priority inversion if framesReady() is called by the normal mixer, 4472// as the normal mixer thread runs at lower 4473// priority than the client's callback thread: there is a short window within framesReady() 4474// during which the normal mixer could be preempted, and the client callback would block. 4475// Another problem can occur if framesReady() is called by the fast mixer: 4476// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4477// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4478size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4479 return mCblk->framesReady(); 4480} 4481 4482// Don't call for fast tracks; the framesReady() could result in priority inversion 4483bool AudioFlinger::PlaybackThread::Track::isReady() const { 4484 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4485 4486 if (framesReady() >= mCblk->frameCount || 4487 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4488 mFillingUpStatus = FS_FILLED; 4489 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4490 return true; 4491 } 4492 return false; 4493} 4494 4495status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4496 int triggerSession) 4497{ 4498 status_t status = NO_ERROR; 4499 ALOGV("start(%d), calling pid %d session %d", 4500 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4501 4502 sp<ThreadBase> thread = mThread.promote(); 4503 if (thread != 0) { 4504 Mutex::Autolock _l(thread->mLock); 4505 track_state state = mState; 4506 // here the track could be either new, or restarted 4507 // in both cases "unstop" the track 4508 if (mState == PAUSED) { 4509 mState = TrackBase::RESUMING; 4510 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4511 } else { 4512 mState = TrackBase::ACTIVE; 4513 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4514 } 4515 4516 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4517 thread->mLock.unlock(); 4518 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4519 thread->mLock.lock(); 4520 4521#ifdef ADD_BATTERY_DATA 4522 // to track the speaker usage 4523 if (status == NO_ERROR) { 4524 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4525 } 4526#endif 4527 } 4528 if (status == NO_ERROR) { 4529 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4530 playbackThread->addTrack_l(this); 4531 } else { 4532 mState = state; 4533 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4534 } 4535 } else { 4536 status = BAD_VALUE; 4537 } 4538 return status; 4539} 4540 4541void AudioFlinger::PlaybackThread::Track::stop() 4542{ 4543 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4544 sp<ThreadBase> thread = mThread.promote(); 4545 if (thread != 0) { 4546 Mutex::Autolock _l(thread->mLock); 4547 track_state state = mState; 4548 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4549 // If the track is not active (PAUSED and buffers full), flush buffers 4550 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4551 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4552 reset(); 4553 mState = STOPPED; 4554 } else if (!isFastTrack()) { 4555 mState = STOPPED; 4556 } else { 4557 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4558 // and then to STOPPED and reset() when presentation is complete 4559 mState = STOPPING_1; 4560 } 4561 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4562 } 4563 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4564 thread->mLock.unlock(); 4565 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4566 thread->mLock.lock(); 4567 4568#ifdef ADD_BATTERY_DATA 4569 // to track the speaker usage 4570 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4571#endif 4572 } 4573 } 4574} 4575 4576void AudioFlinger::PlaybackThread::Track::pause() 4577{ 4578 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4579 sp<ThreadBase> thread = mThread.promote(); 4580 if (thread != 0) { 4581 Mutex::Autolock _l(thread->mLock); 4582 if (mState == ACTIVE || mState == RESUMING) { 4583 mState = PAUSING; 4584 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4585 if (!isOutputTrack()) { 4586 thread->mLock.unlock(); 4587 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4588 thread->mLock.lock(); 4589 4590#ifdef ADD_BATTERY_DATA 4591 // to track the speaker usage 4592 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4593#endif 4594 } 4595 } 4596 } 4597} 4598 4599void AudioFlinger::PlaybackThread::Track::flush() 4600{ 4601 ALOGV("flush(%d)", mName); 4602 sp<ThreadBase> thread = mThread.promote(); 4603 if (thread != 0) { 4604 Mutex::Autolock _l(thread->mLock); 4605 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4606 mState != PAUSING) { 4607 return; 4608 } 4609 // No point remaining in PAUSED state after a flush => go to 4610 // FLUSHED state 4611 mState = FLUSHED; 4612 // do not reset the track if it is still in the process of being stopped or paused. 4613 // this will be done by prepareTracks_l() when the track is stopped. 4614 // prepareTracks_l() will see mState == FLUSHED, then 4615 // remove from active track list, reset(), and trigger presentation complete 4616 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4617 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4618 reset(); 4619 } 4620 } 4621} 4622 4623void AudioFlinger::PlaybackThread::Track::reset() 4624{ 4625 // Do not reset twice to avoid discarding data written just after a flush and before 4626 // the audioflinger thread detects the track is stopped. 4627 if (!mResetDone) { 4628 TrackBase::reset(); 4629 // Force underrun condition to avoid false underrun callback until first data is 4630 // written to buffer 4631 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4632 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4633 mFillingUpStatus = FS_FILLING; 4634 mResetDone = true; 4635 if (mState == FLUSHED) { 4636 mState = IDLE; 4637 } 4638 } 4639} 4640 4641void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4642{ 4643 mMute = muted; 4644} 4645 4646status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4647{ 4648 status_t status = DEAD_OBJECT; 4649 sp<ThreadBase> thread = mThread.promote(); 4650 if (thread != 0) { 4651 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4652 sp<AudioFlinger> af = mClient->audioFlinger(); 4653 4654 Mutex::Autolock _l(af->mLock); 4655 4656 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4657 4658 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4659 Mutex::Autolock _dl(playbackThread->mLock); 4660 Mutex::Autolock _sl(srcThread->mLock); 4661 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4662 if (chain == 0) { 4663 return INVALID_OPERATION; 4664 } 4665 4666 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4667 if (effect == 0) { 4668 return INVALID_OPERATION; 4669 } 4670 srcThread->removeEffect_l(effect); 4671 playbackThread->addEffect_l(effect); 4672 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4673 if (effect->state() == EffectModule::ACTIVE || 4674 effect->state() == EffectModule::STOPPING) { 4675 effect->start(); 4676 } 4677 4678 sp<EffectChain> dstChain = effect->chain().promote(); 4679 if (dstChain == 0) { 4680 srcThread->addEffect_l(effect); 4681 return INVALID_OPERATION; 4682 } 4683 AudioSystem::unregisterEffect(effect->id()); 4684 AudioSystem::registerEffect(&effect->desc(), 4685 srcThread->id(), 4686 dstChain->strategy(), 4687 AUDIO_SESSION_OUTPUT_MIX, 4688 effect->id()); 4689 } 4690 status = playbackThread->attachAuxEffect(this, EffectId); 4691 } 4692 return status; 4693} 4694 4695void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4696{ 4697 mAuxEffectId = EffectId; 4698 mAuxBuffer = buffer; 4699} 4700 4701bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4702 size_t audioHalFrames) 4703{ 4704 // a track is considered presented when the total number of frames written to audio HAL 4705 // corresponds to the number of frames written when presentationComplete() is called for the 4706 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4707 if (mPresentationCompleteFrames == 0) { 4708 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4709 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4710 mPresentationCompleteFrames, audioHalFrames); 4711 } 4712 if (framesWritten >= mPresentationCompleteFrames) { 4713 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4714 mSessionId, framesWritten); 4715 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4716 return true; 4717 } 4718 return false; 4719} 4720 4721void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4722{ 4723 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4724 if (mSyncEvents[i]->type() == type) { 4725 mSyncEvents[i]->trigger(); 4726 mSyncEvents.removeAt(i); 4727 i--; 4728 } 4729 } 4730} 4731 4732// implement VolumeBufferProvider interface 4733 4734uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4735{ 4736 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4737 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4738 uint32_t vlr = mCblk->getVolumeLR(); 4739 uint32_t vl = vlr & 0xFFFF; 4740 uint32_t vr = vlr >> 16; 4741 // track volumes come from shared memory, so can't be trusted and must be clamped 4742 if (vl > MAX_GAIN_INT) { 4743 vl = MAX_GAIN_INT; 4744 } 4745 if (vr > MAX_GAIN_INT) { 4746 vr = MAX_GAIN_INT; 4747 } 4748 // now apply the cached master volume and stream type volume; 4749 // this is trusted but lacks any synchronization or barrier so may be stale 4750 float v = mCachedVolume; 4751 vl *= v; 4752 vr *= v; 4753 // re-combine into U4.16 4754 vlr = (vr << 16) | (vl & 0xFFFF); 4755 // FIXME look at mute, pause, and stop flags 4756 return vlr; 4757} 4758 4759status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4760{ 4761 if (mState == TERMINATED || mState == PAUSED || 4762 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4763 (mState == STOPPED)))) { 4764 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4765 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4766 event->cancel(); 4767 return INVALID_OPERATION; 4768 } 4769 TrackBase::setSyncEvent(event); 4770 return NO_ERROR; 4771} 4772 4773// timed audio tracks 4774 4775sp<AudioFlinger::PlaybackThread::TimedTrack> 4776AudioFlinger::PlaybackThread::TimedTrack::create( 4777 PlaybackThread *thread, 4778 const sp<Client>& client, 4779 audio_stream_type_t streamType, 4780 uint32_t sampleRate, 4781 audio_format_t format, 4782 audio_channel_mask_t channelMask, 4783 int frameCount, 4784 const sp<IMemory>& sharedBuffer, 4785 int sessionId) { 4786 if (!client->reserveTimedTrack()) 4787 return 0; 4788 4789 return new TimedTrack( 4790 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4791 sharedBuffer, sessionId); 4792} 4793 4794AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4795 PlaybackThread *thread, 4796 const sp<Client>& client, 4797 audio_stream_type_t streamType, 4798 uint32_t sampleRate, 4799 audio_format_t format, 4800 audio_channel_mask_t channelMask, 4801 int frameCount, 4802 const sp<IMemory>& sharedBuffer, 4803 int sessionId) 4804 : Track(thread, client, streamType, sampleRate, format, channelMask, 4805 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4806 mQueueHeadInFlight(false), 4807 mTrimQueueHeadOnRelease(false), 4808 mFramesPendingInQueue(0), 4809 mTimedSilenceBuffer(NULL), 4810 mTimedSilenceBufferSize(0), 4811 mTimedAudioOutputOnTime(false), 4812 mMediaTimeTransformValid(false) 4813{ 4814 LocalClock lc; 4815 mLocalTimeFreq = lc.getLocalFreq(); 4816 4817 mLocalTimeToSampleTransform.a_zero = 0; 4818 mLocalTimeToSampleTransform.b_zero = 0; 4819 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4820 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4821 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4822 &mLocalTimeToSampleTransform.a_to_b_denom); 4823 4824 mMediaTimeToSampleTransform.a_zero = 0; 4825 mMediaTimeToSampleTransform.b_zero = 0; 4826 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4827 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4828 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4829 &mMediaTimeToSampleTransform.a_to_b_denom); 4830} 4831 4832AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4833 mClient->releaseTimedTrack(); 4834 delete [] mTimedSilenceBuffer; 4835} 4836 4837status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4838 size_t size, sp<IMemory>* buffer) { 4839 4840 Mutex::Autolock _l(mTimedBufferQueueLock); 4841 4842 trimTimedBufferQueue_l(); 4843 4844 // lazily initialize the shared memory heap for timed buffers 4845 if (mTimedMemoryDealer == NULL) { 4846 const int kTimedBufferHeapSize = 512 << 10; 4847 4848 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4849 "AudioFlingerTimed"); 4850 if (mTimedMemoryDealer == NULL) 4851 return NO_MEMORY; 4852 } 4853 4854 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4855 if (newBuffer == NULL) { 4856 newBuffer = mTimedMemoryDealer->allocate(size); 4857 if (newBuffer == NULL) 4858 return NO_MEMORY; 4859 } 4860 4861 *buffer = newBuffer; 4862 return NO_ERROR; 4863} 4864 4865// caller must hold mTimedBufferQueueLock 4866void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4867 int64_t mediaTimeNow; 4868 { 4869 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4870 if (!mMediaTimeTransformValid) 4871 return; 4872 4873 int64_t targetTimeNow; 4874 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4875 ? mCCHelper.getCommonTime(&targetTimeNow) 4876 : mCCHelper.getLocalTime(&targetTimeNow); 4877 4878 if (OK != res) 4879 return; 4880 4881 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4882 &mediaTimeNow)) { 4883 return; 4884 } 4885 } 4886 4887 size_t trimEnd; 4888 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4889 int64_t bufEnd; 4890 4891 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4892 // We have a next buffer. Just use its PTS as the PTS of the frame 4893 // following the last frame in this buffer. If the stream is sparse 4894 // (ie, there are deliberate gaps left in the stream which should be 4895 // filled with silence by the TimedAudioTrack), then this can result 4896 // in one extra buffer being left un-trimmed when it could have 4897 // been. In general, this is not typical, and we would rather 4898 // optimized away the TS calculation below for the more common case 4899 // where PTSes are contiguous. 4900 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4901 } else { 4902 // We have no next buffer. Compute the PTS of the frame following 4903 // the last frame in this buffer by computing the duration of of 4904 // this frame in media time units and adding it to the PTS of the 4905 // buffer. 4906 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4907 / mCblk->frameSize; 4908 4909 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4910 &bufEnd)) { 4911 ALOGE("Failed to convert frame count of %lld to media time" 4912 " duration" " (scale factor %d/%u) in %s", 4913 frameCount, 4914 mMediaTimeToSampleTransform.a_to_b_numer, 4915 mMediaTimeToSampleTransform.a_to_b_denom, 4916 __PRETTY_FUNCTION__); 4917 break; 4918 } 4919 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4920 } 4921 4922 if (bufEnd > mediaTimeNow) 4923 break; 4924 4925 // Is the buffer we want to use in the middle of a mix operation right 4926 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4927 // from the mixer which should be coming back shortly. 4928 if (!trimEnd && mQueueHeadInFlight) { 4929 mTrimQueueHeadOnRelease = true; 4930 } 4931 } 4932 4933 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4934 if (trimStart < trimEnd) { 4935 // Update the bookkeeping for framesReady() 4936 for (size_t i = trimStart; i < trimEnd; ++i) { 4937 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4938 } 4939 4940 // Now actually remove the buffers from the queue. 4941 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4942 } 4943} 4944 4945void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4946 const char* logTag) { 4947 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4948 "%s called (reason \"%s\"), but timed buffer queue has no" 4949 " elements to trim.", __FUNCTION__, logTag); 4950 4951 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4952 mTimedBufferQueue.removeAt(0); 4953} 4954 4955void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4956 const TimedBuffer& buf, 4957 const char* logTag) { 4958 uint32_t bufBytes = buf.buffer()->size(); 4959 uint32_t consumedAlready = buf.position(); 4960 4961 ALOG_ASSERT(consumedAlready <= bufBytes, 4962 "Bad bookkeeping while updating frames pending. Timed buffer is" 4963 " only %u bytes long, but claims to have consumed %u" 4964 " bytes. (update reason: \"%s\")", 4965 bufBytes, consumedAlready, logTag); 4966 4967 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4968 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4969 "Bad bookkeeping while updating frames pending. Should have at" 4970 " least %u queued frames, but we think we have only %u. (update" 4971 " reason: \"%s\")", 4972 bufFrames, mFramesPendingInQueue, logTag); 4973 4974 mFramesPendingInQueue -= bufFrames; 4975} 4976 4977status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4978 const sp<IMemory>& buffer, int64_t pts) { 4979 4980 { 4981 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4982 if (!mMediaTimeTransformValid) 4983 return INVALID_OPERATION; 4984 } 4985 4986 Mutex::Autolock _l(mTimedBufferQueueLock); 4987 4988 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4989 mFramesPendingInQueue += bufFrames; 4990 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4991 4992 return NO_ERROR; 4993} 4994 4995status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4996 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4997 4998 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4999 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5000 target); 5001 5002 if (!(target == TimedAudioTrack::LOCAL_TIME || 5003 target == TimedAudioTrack::COMMON_TIME)) { 5004 return BAD_VALUE; 5005 } 5006 5007 Mutex::Autolock lock(mMediaTimeTransformLock); 5008 mMediaTimeTransform = xform; 5009 mMediaTimeTransformTarget = target; 5010 mMediaTimeTransformValid = true; 5011 5012 return NO_ERROR; 5013} 5014 5015#define min(a, b) ((a) < (b) ? (a) : (b)) 5016 5017// implementation of getNextBuffer for tracks whose buffers have timestamps 5018status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5019 AudioBufferProvider::Buffer* buffer, int64_t pts) 5020{ 5021 if (pts == AudioBufferProvider::kInvalidPTS) { 5022 buffer->raw = NULL; 5023 buffer->frameCount = 0; 5024 mTimedAudioOutputOnTime = false; 5025 return INVALID_OPERATION; 5026 } 5027 5028 Mutex::Autolock _l(mTimedBufferQueueLock); 5029 5030 ALOG_ASSERT(!mQueueHeadInFlight, 5031 "getNextBuffer called without releaseBuffer!"); 5032 5033 while (true) { 5034 5035 // if we have no timed buffers, then fail 5036 if (mTimedBufferQueue.isEmpty()) { 5037 buffer->raw = NULL; 5038 buffer->frameCount = 0; 5039 return NOT_ENOUGH_DATA; 5040 } 5041 5042 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5043 5044 // calculate the PTS of the head of the timed buffer queue expressed in 5045 // local time 5046 int64_t headLocalPTS; 5047 { 5048 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5049 5050 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5051 5052 if (mMediaTimeTransform.a_to_b_denom == 0) { 5053 // the transform represents a pause, so yield silence 5054 timedYieldSilence_l(buffer->frameCount, buffer); 5055 return NO_ERROR; 5056 } 5057 5058 int64_t transformedPTS; 5059 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5060 &transformedPTS)) { 5061 // the transform failed. this shouldn't happen, but if it does 5062 // then just drop this buffer 5063 ALOGW("timedGetNextBuffer transform failed"); 5064 buffer->raw = NULL; 5065 buffer->frameCount = 0; 5066 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5067 return NO_ERROR; 5068 } 5069 5070 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5071 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5072 &headLocalPTS)) { 5073 buffer->raw = NULL; 5074 buffer->frameCount = 0; 5075 return INVALID_OPERATION; 5076 } 5077 } else { 5078 headLocalPTS = transformedPTS; 5079 } 5080 } 5081 5082 // adjust the head buffer's PTS to reflect the portion of the head buffer 5083 // that has already been consumed 5084 int64_t effectivePTS = headLocalPTS + 5085 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5086 5087 // Calculate the delta in samples between the head of the input buffer 5088 // queue and the start of the next output buffer that will be written. 5089 // If the transformation fails because of over or underflow, it means 5090 // that the sample's position in the output stream is so far out of 5091 // whack that it should just be dropped. 5092 int64_t sampleDelta; 5093 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5094 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5095 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5096 " mix"); 5097 continue; 5098 } 5099 if (!mLocalTimeToSampleTransform.doForwardTransform( 5100 (effectivePTS - pts) << 32, &sampleDelta)) { 5101 ALOGV("*** too late during sample rate transform: dropped buffer"); 5102 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5103 continue; 5104 } 5105 5106 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5107 " sampleDelta=[%d.%08x]", 5108 head.pts(), head.position(), pts, 5109 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5110 + (sampleDelta >> 32)), 5111 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5112 5113 // if the delta between the ideal placement for the next input sample and 5114 // the current output position is within this threshold, then we will 5115 // concatenate the next input samples to the previous output 5116 const int64_t kSampleContinuityThreshold = 5117 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5118 5119 // if this is the first buffer of audio that we're emitting from this track 5120 // then it should be almost exactly on time. 5121 const int64_t kSampleStartupThreshold = 1LL << 32; 5122 5123 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5124 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5125 // the next input is close enough to being on time, so concatenate it 5126 // with the last output 5127 timedYieldSamples_l(buffer); 5128 5129 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5130 head.position(), buffer->frameCount); 5131 return NO_ERROR; 5132 } 5133 5134 // Looks like our output is not on time. Reset our on timed status. 5135 // Next time we mix samples from our input queue, then should be within 5136 // the StartupThreshold. 5137 mTimedAudioOutputOnTime = false; 5138 if (sampleDelta > 0) { 5139 // the gap between the current output position and the proper start of 5140 // the next input sample is too big, so fill it with silence 5141 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5142 5143 timedYieldSilence_l(framesUntilNextInput, buffer); 5144 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5145 return NO_ERROR; 5146 } else { 5147 // the next input sample is late 5148 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5149 size_t onTimeSamplePosition = 5150 head.position() + lateFrames * mCblk->frameSize; 5151 5152 if (onTimeSamplePosition > head.buffer()->size()) { 5153 // all the remaining samples in the head are too late, so 5154 // drop it and move on 5155 ALOGV("*** too late: dropped buffer"); 5156 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5157 continue; 5158 } else { 5159 // skip over the late samples 5160 head.setPosition(onTimeSamplePosition); 5161 5162 // yield the available samples 5163 timedYieldSamples_l(buffer); 5164 5165 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5166 return NO_ERROR; 5167 } 5168 } 5169 } 5170} 5171 5172// Yield samples from the timed buffer queue head up to the given output 5173// buffer's capacity. 5174// 5175// Caller must hold mTimedBufferQueueLock 5176void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5177 AudioBufferProvider::Buffer* buffer) { 5178 5179 const TimedBuffer& head = mTimedBufferQueue[0]; 5180 5181 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5182 head.position()); 5183 5184 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5185 mCblk->frameSize); 5186 size_t framesRequested = buffer->frameCount; 5187 buffer->frameCount = min(framesLeftInHead, framesRequested); 5188 5189 mQueueHeadInFlight = true; 5190 mTimedAudioOutputOnTime = true; 5191} 5192 5193// Yield samples of silence up to the given output buffer's capacity 5194// 5195// Caller must hold mTimedBufferQueueLock 5196void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5197 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5198 5199 // lazily allocate a buffer filled with silence 5200 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5201 delete [] mTimedSilenceBuffer; 5202 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5203 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5204 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5205 } 5206 5207 buffer->raw = mTimedSilenceBuffer; 5208 size_t framesRequested = buffer->frameCount; 5209 buffer->frameCount = min(numFrames, framesRequested); 5210 5211 mTimedAudioOutputOnTime = false; 5212} 5213 5214// AudioBufferProvider interface 5215void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5216 AudioBufferProvider::Buffer* buffer) { 5217 5218 Mutex::Autolock _l(mTimedBufferQueueLock); 5219 5220 // If the buffer which was just released is part of the buffer at the head 5221 // of the queue, be sure to update the amt of the buffer which has been 5222 // consumed. If the buffer being returned is not part of the head of the 5223 // queue, its either because the buffer is part of the silence buffer, or 5224 // because the head of the timed queue was trimmed after the mixer called 5225 // getNextBuffer but before the mixer called releaseBuffer. 5226 if (buffer->raw == mTimedSilenceBuffer) { 5227 ALOG_ASSERT(!mQueueHeadInFlight, 5228 "Queue head in flight during release of silence buffer!"); 5229 goto done; 5230 } 5231 5232 ALOG_ASSERT(mQueueHeadInFlight, 5233 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5234 " head in flight."); 5235 5236 if (mTimedBufferQueue.size()) { 5237 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5238 5239 void* start = head.buffer()->pointer(); 5240 void* end = reinterpret_cast<void*>( 5241 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5242 + head.buffer()->size()); 5243 5244 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5245 "released buffer not within the head of the timed buffer" 5246 " queue; qHead = [%p, %p], released buffer = %p", 5247 start, end, buffer->raw); 5248 5249 head.setPosition(head.position() + 5250 (buffer->frameCount * mCblk->frameSize)); 5251 mQueueHeadInFlight = false; 5252 5253 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5254 "Bad bookkeeping during releaseBuffer! Should have at" 5255 " least %u queued frames, but we think we have only %u", 5256 buffer->frameCount, mFramesPendingInQueue); 5257 5258 mFramesPendingInQueue -= buffer->frameCount; 5259 5260 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5261 || mTrimQueueHeadOnRelease) { 5262 trimTimedBufferQueueHead_l("releaseBuffer"); 5263 mTrimQueueHeadOnRelease = false; 5264 } 5265 } else { 5266 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5267 " buffers in the timed buffer queue"); 5268 } 5269 5270done: 5271 buffer->raw = 0; 5272 buffer->frameCount = 0; 5273} 5274 5275size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5276 Mutex::Autolock _l(mTimedBufferQueueLock); 5277 return mFramesPendingInQueue; 5278} 5279 5280AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5281 : mPTS(0), mPosition(0) {} 5282 5283AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5284 const sp<IMemory>& buffer, int64_t pts) 5285 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5286 5287// ---------------------------------------------------------------------------- 5288 5289// RecordTrack constructor must be called with AudioFlinger::mLock held 5290AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5291 RecordThread *thread, 5292 const sp<Client>& client, 5293 uint32_t sampleRate, 5294 audio_format_t format, 5295 audio_channel_mask_t channelMask, 5296 int frameCount, 5297 int sessionId) 5298 : TrackBase(thread, client, sampleRate, format, 5299 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5300 mOverflow(false) 5301{ 5302 if (mCblk != NULL) { 5303 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5304 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5305 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5306 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5307 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5308 } else { 5309 mCblk->frameSize = sizeof(int8_t); 5310 } 5311 } 5312} 5313 5314AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5315{ 5316 sp<ThreadBase> thread = mThread.promote(); 5317 if (thread != 0) { 5318 AudioSystem::releaseInput(thread->id()); 5319 } 5320} 5321 5322// AudioBufferProvider interface 5323status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5324{ 5325 audio_track_cblk_t* cblk = this->cblk(); 5326 uint32_t framesAvail; 5327 uint32_t framesReq = buffer->frameCount; 5328 5329 // Check if last stepServer failed, try to step now 5330 if (mStepServerFailed) { 5331 if (!step()) goto getNextBuffer_exit; 5332 ALOGV("stepServer recovered"); 5333 mStepServerFailed = false; 5334 } 5335 5336 framesAvail = cblk->framesAvailable_l(); 5337 5338 if (CC_LIKELY(framesAvail)) { 5339 uint32_t s = cblk->server; 5340 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5341 5342 if (framesReq > framesAvail) { 5343 framesReq = framesAvail; 5344 } 5345 if (framesReq > bufferEnd - s) { 5346 framesReq = bufferEnd - s; 5347 } 5348 5349 buffer->raw = getBuffer(s, framesReq); 5350 buffer->frameCount = framesReq; 5351 return NO_ERROR; 5352 } 5353 5354getNextBuffer_exit: 5355 buffer->raw = NULL; 5356 buffer->frameCount = 0; 5357 return NOT_ENOUGH_DATA; 5358} 5359 5360status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5361 int triggerSession) 5362{ 5363 sp<ThreadBase> thread = mThread.promote(); 5364 if (thread != 0) { 5365 RecordThread *recordThread = (RecordThread *)thread.get(); 5366 return recordThread->start(this, event, triggerSession); 5367 } else { 5368 return BAD_VALUE; 5369 } 5370} 5371 5372void AudioFlinger::RecordThread::RecordTrack::stop() 5373{ 5374 sp<ThreadBase> thread = mThread.promote(); 5375 if (thread != 0) { 5376 RecordThread *recordThread = (RecordThread *)thread.get(); 5377 recordThread->stop(this); 5378 TrackBase::reset(); 5379 // Force overrun condition to avoid false overrun callback until first data is 5380 // read from buffer 5381 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5382 } 5383} 5384 5385void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5386{ 5387 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5388 (mClient == 0) ? getpid_cached : mClient->pid(), 5389 mFormat, 5390 mChannelMask, 5391 mSessionId, 5392 mFrameCount, 5393 mState, 5394 mCblk->sampleRate, 5395 mCblk->server, 5396 mCblk->user); 5397} 5398 5399 5400// ---------------------------------------------------------------------------- 5401 5402AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5403 PlaybackThread *playbackThread, 5404 DuplicatingThread *sourceThread, 5405 uint32_t sampleRate, 5406 audio_format_t format, 5407 audio_channel_mask_t channelMask, 5408 int frameCount) 5409 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5410 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5411 mActive(false), mSourceThread(sourceThread) 5412{ 5413 5414 if (mCblk != NULL) { 5415 mCblk->flags |= CBLK_DIRECTION_OUT; 5416 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5417 mOutBuffer.frameCount = 0; 5418 playbackThread->mTracks.add(this); 5419 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5420 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5421 mCblk, mBuffer, mCblk->buffers, 5422 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5423 } else { 5424 ALOGW("Error creating output track on thread %p", playbackThread); 5425 } 5426} 5427 5428AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5429{ 5430 clearBufferQueue(); 5431} 5432 5433status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5434 int triggerSession) 5435{ 5436 status_t status = Track::start(event, triggerSession); 5437 if (status != NO_ERROR) { 5438 return status; 5439 } 5440 5441 mActive = true; 5442 mRetryCount = 127; 5443 return status; 5444} 5445 5446void AudioFlinger::PlaybackThread::OutputTrack::stop() 5447{ 5448 Track::stop(); 5449 clearBufferQueue(); 5450 mOutBuffer.frameCount = 0; 5451 mActive = false; 5452} 5453 5454bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5455{ 5456 Buffer *pInBuffer; 5457 Buffer inBuffer; 5458 uint32_t channelCount = mChannelCount; 5459 bool outputBufferFull = false; 5460 inBuffer.frameCount = frames; 5461 inBuffer.i16 = data; 5462 5463 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5464 5465 if (!mActive && frames != 0) { 5466 start(); 5467 sp<ThreadBase> thread = mThread.promote(); 5468 if (thread != 0) { 5469 MixerThread *mixerThread = (MixerThread *)thread.get(); 5470 if (mCblk->frameCount > frames){ 5471 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5472 uint32_t startFrames = (mCblk->frameCount - frames); 5473 pInBuffer = new Buffer; 5474 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5475 pInBuffer->frameCount = startFrames; 5476 pInBuffer->i16 = pInBuffer->mBuffer; 5477 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5478 mBufferQueue.add(pInBuffer); 5479 } else { 5480 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5481 } 5482 } 5483 } 5484 } 5485 5486 while (waitTimeLeftMs) { 5487 // First write pending buffers, then new data 5488 if (mBufferQueue.size()) { 5489 pInBuffer = mBufferQueue.itemAt(0); 5490 } else { 5491 pInBuffer = &inBuffer; 5492 } 5493 5494 if (pInBuffer->frameCount == 0) { 5495 break; 5496 } 5497 5498 if (mOutBuffer.frameCount == 0) { 5499 mOutBuffer.frameCount = pInBuffer->frameCount; 5500 nsecs_t startTime = systemTime(); 5501 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5502 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5503 outputBufferFull = true; 5504 break; 5505 } 5506 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5507 if (waitTimeLeftMs >= waitTimeMs) { 5508 waitTimeLeftMs -= waitTimeMs; 5509 } else { 5510 waitTimeLeftMs = 0; 5511 } 5512 } 5513 5514 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5515 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5516 mCblk->stepUser(outFrames); 5517 pInBuffer->frameCount -= outFrames; 5518 pInBuffer->i16 += outFrames * channelCount; 5519 mOutBuffer.frameCount -= outFrames; 5520 mOutBuffer.i16 += outFrames * channelCount; 5521 5522 if (pInBuffer->frameCount == 0) { 5523 if (mBufferQueue.size()) { 5524 mBufferQueue.removeAt(0); 5525 delete [] pInBuffer->mBuffer; 5526 delete pInBuffer; 5527 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5528 } else { 5529 break; 5530 } 5531 } 5532 } 5533 5534 // If we could not write all frames, allocate a buffer and queue it for next time. 5535 if (inBuffer.frameCount) { 5536 sp<ThreadBase> thread = mThread.promote(); 5537 if (thread != 0 && !thread->standby()) { 5538 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5539 pInBuffer = new Buffer; 5540 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5541 pInBuffer->frameCount = inBuffer.frameCount; 5542 pInBuffer->i16 = pInBuffer->mBuffer; 5543 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5544 mBufferQueue.add(pInBuffer); 5545 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5546 } else { 5547 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5548 } 5549 } 5550 } 5551 5552 // Calling write() with a 0 length buffer, means that no more data will be written: 5553 // If no more buffers are pending, fill output track buffer to make sure it is started 5554 // by output mixer. 5555 if (frames == 0 && mBufferQueue.size() == 0) { 5556 if (mCblk->user < mCblk->frameCount) { 5557 frames = mCblk->frameCount - mCblk->user; 5558 pInBuffer = new Buffer; 5559 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5560 pInBuffer->frameCount = frames; 5561 pInBuffer->i16 = pInBuffer->mBuffer; 5562 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5563 mBufferQueue.add(pInBuffer); 5564 } else if (mActive) { 5565 stop(); 5566 } 5567 } 5568 5569 return outputBufferFull; 5570} 5571 5572status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5573{ 5574 int active; 5575 status_t result; 5576 audio_track_cblk_t* cblk = mCblk; 5577 uint32_t framesReq = buffer->frameCount; 5578 5579// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5580 buffer->frameCount = 0; 5581 5582 uint32_t framesAvail = cblk->framesAvailable(); 5583 5584 5585 if (framesAvail == 0) { 5586 Mutex::Autolock _l(cblk->lock); 5587 goto start_loop_here; 5588 while (framesAvail == 0) { 5589 active = mActive; 5590 if (CC_UNLIKELY(!active)) { 5591 ALOGV("Not active and NO_MORE_BUFFERS"); 5592 return NO_MORE_BUFFERS; 5593 } 5594 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5595 if (result != NO_ERROR) { 5596 return NO_MORE_BUFFERS; 5597 } 5598 // read the server count again 5599 start_loop_here: 5600 framesAvail = cblk->framesAvailable_l(); 5601 } 5602 } 5603 5604// if (framesAvail < framesReq) { 5605// return NO_MORE_BUFFERS; 5606// } 5607 5608 if (framesReq > framesAvail) { 5609 framesReq = framesAvail; 5610 } 5611 5612 uint32_t u = cblk->user; 5613 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5614 5615 if (framesReq > bufferEnd - u) { 5616 framesReq = bufferEnd - u; 5617 } 5618 5619 buffer->frameCount = framesReq; 5620 buffer->raw = (void *)cblk->buffer(u); 5621 return NO_ERROR; 5622} 5623 5624 5625void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5626{ 5627 size_t size = mBufferQueue.size(); 5628 5629 for (size_t i = 0; i < size; i++) { 5630 Buffer *pBuffer = mBufferQueue.itemAt(i); 5631 delete [] pBuffer->mBuffer; 5632 delete pBuffer; 5633 } 5634 mBufferQueue.clear(); 5635} 5636 5637// ---------------------------------------------------------------------------- 5638 5639AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5640 : RefBase(), 5641 mAudioFlinger(audioFlinger), 5642 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5643 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5644 mPid(pid), 5645 mTimedTrackCount(0) 5646{ 5647 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5648} 5649 5650// Client destructor must be called with AudioFlinger::mLock held 5651AudioFlinger::Client::~Client() 5652{ 5653 mAudioFlinger->removeClient_l(mPid); 5654} 5655 5656sp<MemoryDealer> AudioFlinger::Client::heap() const 5657{ 5658 return mMemoryDealer; 5659} 5660 5661// Reserve one of the limited slots for a timed audio track associated 5662// with this client 5663bool AudioFlinger::Client::reserveTimedTrack() 5664{ 5665 const int kMaxTimedTracksPerClient = 4; 5666 5667 Mutex::Autolock _l(mTimedTrackLock); 5668 5669 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5670 ALOGW("can not create timed track - pid %d has exceeded the limit", 5671 mPid); 5672 return false; 5673 } 5674 5675 mTimedTrackCount++; 5676 return true; 5677} 5678 5679// Release a slot for a timed audio track 5680void AudioFlinger::Client::releaseTimedTrack() 5681{ 5682 Mutex::Autolock _l(mTimedTrackLock); 5683 mTimedTrackCount--; 5684} 5685 5686// ---------------------------------------------------------------------------- 5687 5688AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5689 const sp<IAudioFlingerClient>& client, 5690 pid_t pid) 5691 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5692{ 5693} 5694 5695AudioFlinger::NotificationClient::~NotificationClient() 5696{ 5697} 5698 5699void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5700{ 5701 sp<NotificationClient> keep(this); 5702 mAudioFlinger->removeNotificationClient(mPid); 5703} 5704 5705// ---------------------------------------------------------------------------- 5706 5707AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5708 : BnAudioTrack(), 5709 mTrack(track) 5710{ 5711} 5712 5713AudioFlinger::TrackHandle::~TrackHandle() { 5714 // just stop the track on deletion, associated resources 5715 // will be freed from the main thread once all pending buffers have 5716 // been played. Unless it's not in the active track list, in which 5717 // case we free everything now... 5718 mTrack->destroy(); 5719} 5720 5721sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5722 return mTrack->getCblk(); 5723} 5724 5725status_t AudioFlinger::TrackHandle::start() { 5726 return mTrack->start(); 5727} 5728 5729void AudioFlinger::TrackHandle::stop() { 5730 mTrack->stop(); 5731} 5732 5733void AudioFlinger::TrackHandle::flush() { 5734 mTrack->flush(); 5735} 5736 5737void AudioFlinger::TrackHandle::mute(bool e) { 5738 mTrack->mute(e); 5739} 5740 5741void AudioFlinger::TrackHandle::pause() { 5742 mTrack->pause(); 5743} 5744 5745status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5746{ 5747 return mTrack->attachAuxEffect(EffectId); 5748} 5749 5750status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5751 sp<IMemory>* buffer) { 5752 if (!mTrack->isTimedTrack()) 5753 return INVALID_OPERATION; 5754 5755 PlaybackThread::TimedTrack* tt = 5756 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5757 return tt->allocateTimedBuffer(size, buffer); 5758} 5759 5760status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5761 int64_t pts) { 5762 if (!mTrack->isTimedTrack()) 5763 return INVALID_OPERATION; 5764 5765 PlaybackThread::TimedTrack* tt = 5766 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5767 return tt->queueTimedBuffer(buffer, pts); 5768} 5769 5770status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5771 const LinearTransform& xform, int target) { 5772 5773 if (!mTrack->isTimedTrack()) 5774 return INVALID_OPERATION; 5775 5776 PlaybackThread::TimedTrack* tt = 5777 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5778 return tt->setMediaTimeTransform( 5779 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5780} 5781 5782status_t AudioFlinger::TrackHandle::onTransact( 5783 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5784{ 5785 return BnAudioTrack::onTransact(code, data, reply, flags); 5786} 5787 5788// ---------------------------------------------------------------------------- 5789 5790sp<IAudioRecord> AudioFlinger::openRecord( 5791 pid_t pid, 5792 audio_io_handle_t input, 5793 uint32_t sampleRate, 5794 audio_format_t format, 5795 audio_channel_mask_t channelMask, 5796 int frameCount, 5797 IAudioFlinger::track_flags_t flags, 5798 pid_t tid, 5799 int *sessionId, 5800 status_t *status) 5801{ 5802 sp<RecordThread::RecordTrack> recordTrack; 5803 sp<RecordHandle> recordHandle; 5804 sp<Client> client; 5805 status_t lStatus; 5806 RecordThread *thread; 5807 size_t inFrameCount; 5808 int lSessionId; 5809 5810 // check calling permissions 5811 if (!recordingAllowed()) { 5812 lStatus = PERMISSION_DENIED; 5813 goto Exit; 5814 } 5815 5816 // add client to list 5817 { // scope for mLock 5818 Mutex::Autolock _l(mLock); 5819 thread = checkRecordThread_l(input); 5820 if (thread == NULL) { 5821 lStatus = BAD_VALUE; 5822 goto Exit; 5823 } 5824 5825 client = registerPid_l(pid); 5826 5827 // If no audio session id is provided, create one here 5828 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5829 lSessionId = *sessionId; 5830 } else { 5831 lSessionId = nextUniqueId(); 5832 if (sessionId != NULL) { 5833 *sessionId = lSessionId; 5834 } 5835 } 5836 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5837 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5838 frameCount, lSessionId, flags, tid, &lStatus); 5839 } 5840 if (lStatus != NO_ERROR) { 5841 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5842 // destructor is called by the TrackBase destructor with mLock held 5843 client.clear(); 5844 recordTrack.clear(); 5845 goto Exit; 5846 } 5847 5848 // return to handle to client 5849 recordHandle = new RecordHandle(recordTrack); 5850 lStatus = NO_ERROR; 5851 5852Exit: 5853 if (status) { 5854 *status = lStatus; 5855 } 5856 return recordHandle; 5857} 5858 5859// ---------------------------------------------------------------------------- 5860 5861AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5862 : BnAudioRecord(), 5863 mRecordTrack(recordTrack) 5864{ 5865} 5866 5867AudioFlinger::RecordHandle::~RecordHandle() { 5868 stop_nonvirtual(); 5869} 5870 5871sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5872 return mRecordTrack->getCblk(); 5873} 5874 5875status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5876 ALOGV("RecordHandle::start()"); 5877 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5878} 5879 5880void AudioFlinger::RecordHandle::stop() { 5881 stop_nonvirtual(); 5882} 5883 5884void AudioFlinger::RecordHandle::stop_nonvirtual() { 5885 ALOGV("RecordHandle::stop()"); 5886 mRecordTrack->stop(); 5887} 5888 5889status_t AudioFlinger::RecordHandle::onTransact( 5890 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5891{ 5892 return BnAudioRecord::onTransact(code, data, reply, flags); 5893} 5894 5895// ---------------------------------------------------------------------------- 5896 5897AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5898 AudioStreamIn *input, 5899 uint32_t sampleRate, 5900 audio_channel_mask_t channelMask, 5901 audio_io_handle_t id, 5902 audio_devices_t device) : 5903 ThreadBase(audioFlinger, id, device, RECORD), 5904 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5905 // mRsmpInIndex and mInputBytes set by readInputParameters() 5906 mReqChannelCount(popcount(channelMask)), 5907 mReqSampleRate(sampleRate) 5908 // mBytesRead is only meaningful while active, and so is cleared in start() 5909 // (but might be better to also clear here for dump?) 5910{ 5911 snprintf(mName, kNameLength, "AudioIn_%X", id); 5912 5913 readInputParameters(); 5914} 5915 5916 5917AudioFlinger::RecordThread::~RecordThread() 5918{ 5919 delete[] mRsmpInBuffer; 5920 delete mResampler; 5921 delete[] mRsmpOutBuffer; 5922} 5923 5924void AudioFlinger::RecordThread::onFirstRef() 5925{ 5926 run(mName, PRIORITY_URGENT_AUDIO); 5927} 5928 5929status_t AudioFlinger::RecordThread::readyToRun() 5930{ 5931 status_t status = initCheck(); 5932 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5933 return status; 5934} 5935 5936bool AudioFlinger::RecordThread::threadLoop() 5937{ 5938 AudioBufferProvider::Buffer buffer; 5939 sp<RecordTrack> activeTrack; 5940 Vector< sp<EffectChain> > effectChains; 5941 5942 nsecs_t lastWarning = 0; 5943 5944 acquireWakeLock(); 5945 5946 // start recording 5947 while (!exitPending()) { 5948 5949 processConfigEvents(); 5950 5951 { // scope for mLock 5952 Mutex::Autolock _l(mLock); 5953 checkForNewParameters_l(); 5954 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5955 if (!mStandby) { 5956 mInput->stream->common.standby(&mInput->stream->common); 5957 mStandby = true; 5958 } 5959 5960 if (exitPending()) break; 5961 5962 releaseWakeLock_l(); 5963 ALOGV("RecordThread: loop stopping"); 5964 // go to sleep 5965 mWaitWorkCV.wait(mLock); 5966 ALOGV("RecordThread: loop starting"); 5967 acquireWakeLock_l(); 5968 continue; 5969 } 5970 if (mActiveTrack != 0) { 5971 if (mActiveTrack->mState == TrackBase::PAUSING) { 5972 if (!mStandby) { 5973 mInput->stream->common.standby(&mInput->stream->common); 5974 mStandby = true; 5975 } 5976 mActiveTrack.clear(); 5977 mStartStopCond.broadcast(); 5978 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5979 if (mReqChannelCount != mActiveTrack->channelCount()) { 5980 mActiveTrack.clear(); 5981 mStartStopCond.broadcast(); 5982 } else if (mBytesRead != 0) { 5983 // record start succeeds only if first read from audio input 5984 // succeeds 5985 if (mBytesRead > 0) { 5986 mActiveTrack->mState = TrackBase::ACTIVE; 5987 } else { 5988 mActiveTrack.clear(); 5989 } 5990 mStartStopCond.broadcast(); 5991 } 5992 mStandby = false; 5993 } 5994 } 5995 lockEffectChains_l(effectChains); 5996 } 5997 5998 if (mActiveTrack != 0) { 5999 if (mActiveTrack->mState != TrackBase::ACTIVE && 6000 mActiveTrack->mState != TrackBase::RESUMING) { 6001 unlockEffectChains(effectChains); 6002 usleep(kRecordThreadSleepUs); 6003 continue; 6004 } 6005 for (size_t i = 0; i < effectChains.size(); i ++) { 6006 effectChains[i]->process_l(); 6007 } 6008 6009 buffer.frameCount = mFrameCount; 6010 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6011 size_t framesOut = buffer.frameCount; 6012 if (mResampler == NULL) { 6013 // no resampling 6014 while (framesOut) { 6015 size_t framesIn = mFrameCount - mRsmpInIndex; 6016 if (framesIn) { 6017 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6018 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6019 if (framesIn > framesOut) 6020 framesIn = framesOut; 6021 mRsmpInIndex += framesIn; 6022 framesOut -= framesIn; 6023 if ((int)mChannelCount == mReqChannelCount || 6024 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6025 memcpy(dst, src, framesIn * mFrameSize); 6026 } else { 6027 int16_t *src16 = (int16_t *)src; 6028 int16_t *dst16 = (int16_t *)dst; 6029 if (mChannelCount == 1) { 6030 while (framesIn--) { 6031 *dst16++ = *src16; 6032 *dst16++ = *src16++; 6033 } 6034 } else { 6035 while (framesIn--) { 6036 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6037 src16 += 2; 6038 } 6039 } 6040 } 6041 } 6042 if (framesOut && mFrameCount == mRsmpInIndex) { 6043 if (framesOut == mFrameCount && 6044 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6045 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6046 framesOut = 0; 6047 } else { 6048 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6049 mRsmpInIndex = 0; 6050 } 6051 if (mBytesRead < 0) { 6052 ALOGE("Error reading audio input"); 6053 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6054 // Force input into standby so that it tries to 6055 // recover at next read attempt 6056 mInput->stream->common.standby(&mInput->stream->common); 6057 usleep(kRecordThreadSleepUs); 6058 } 6059 mRsmpInIndex = mFrameCount; 6060 framesOut = 0; 6061 buffer.frameCount = 0; 6062 } 6063 } 6064 } 6065 } else { 6066 // resampling 6067 6068 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6069 // alter output frame count as if we were expecting stereo samples 6070 if (mChannelCount == 1 && mReqChannelCount == 1) { 6071 framesOut >>= 1; 6072 } 6073 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6074 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6075 // are 32 bit aligned which should be always true. 6076 if (mChannelCount == 2 && mReqChannelCount == 1) { 6077 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6078 // the resampler always outputs stereo samples: do post stereo to mono conversion 6079 int16_t *src = (int16_t *)mRsmpOutBuffer; 6080 int16_t *dst = buffer.i16; 6081 while (framesOut--) { 6082 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6083 src += 2; 6084 } 6085 } else { 6086 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6087 } 6088 6089 } 6090 if (mFramestoDrop == 0) { 6091 mActiveTrack->releaseBuffer(&buffer); 6092 } else { 6093 if (mFramestoDrop > 0) { 6094 mFramestoDrop -= buffer.frameCount; 6095 if (mFramestoDrop <= 0) { 6096 clearSyncStartEvent(); 6097 } 6098 } else { 6099 mFramestoDrop += buffer.frameCount; 6100 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6101 mSyncStartEvent->isCancelled()) { 6102 ALOGW("Synced record %s, session %d, trigger session %d", 6103 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6104 mActiveTrack->sessionId(), 6105 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6106 clearSyncStartEvent(); 6107 } 6108 } 6109 } 6110 mActiveTrack->clearOverflow(); 6111 } 6112 // client isn't retrieving buffers fast enough 6113 else { 6114 if (!mActiveTrack->setOverflow()) { 6115 nsecs_t now = systemTime(); 6116 if ((now - lastWarning) > kWarningThrottleNs) { 6117 ALOGW("RecordThread: buffer overflow"); 6118 lastWarning = now; 6119 } 6120 } 6121 // Release the processor for a while before asking for a new buffer. 6122 // This will give the application more chance to read from the buffer and 6123 // clear the overflow. 6124 usleep(kRecordThreadSleepUs); 6125 } 6126 } 6127 // enable changes in effect chain 6128 unlockEffectChains(effectChains); 6129 effectChains.clear(); 6130 } 6131 6132 if (!mStandby) { 6133 mInput->stream->common.standby(&mInput->stream->common); 6134 } 6135 mActiveTrack.clear(); 6136 6137 mStartStopCond.broadcast(); 6138 6139 releaseWakeLock(); 6140 6141 ALOGV("RecordThread %p exiting", this); 6142 return false; 6143} 6144 6145 6146sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6147 const sp<AudioFlinger::Client>& client, 6148 uint32_t sampleRate, 6149 audio_format_t format, 6150 audio_channel_mask_t channelMask, 6151 int frameCount, 6152 int sessionId, 6153 IAudioFlinger::track_flags_t flags, 6154 pid_t tid, 6155 status_t *status) 6156{ 6157 sp<RecordTrack> track; 6158 status_t lStatus; 6159 6160 lStatus = initCheck(); 6161 if (lStatus != NO_ERROR) { 6162 ALOGE("Audio driver not initialized."); 6163 goto Exit; 6164 } 6165 6166 // FIXME use flags and tid similar to createTrack_l() 6167 6168 { // scope for mLock 6169 Mutex::Autolock _l(mLock); 6170 6171 track = new RecordTrack(this, client, sampleRate, 6172 format, channelMask, frameCount, sessionId); 6173 6174 if (track->getCblk() == 0) { 6175 lStatus = NO_MEMORY; 6176 goto Exit; 6177 } 6178 6179 mTrack = track.get(); 6180 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6181 bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) && 6182 mAudioFlinger->btNrecIsOff(); 6183 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6184 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6185 } 6186 lStatus = NO_ERROR; 6187 6188Exit: 6189 if (status) { 6190 *status = lStatus; 6191 } 6192 return track; 6193} 6194 6195status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6196 AudioSystem::sync_event_t event, 6197 int triggerSession) 6198{ 6199 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6200 sp<ThreadBase> strongMe = this; 6201 status_t status = NO_ERROR; 6202 6203 if (event == AudioSystem::SYNC_EVENT_NONE) { 6204 clearSyncStartEvent(); 6205 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6206 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6207 triggerSession, 6208 recordTrack->sessionId(), 6209 syncStartEventCallback, 6210 this); 6211 // Sync event can be cancelled by the trigger session if the track is not in a 6212 // compatible state in which case we start record immediately 6213 if (mSyncStartEvent->isCancelled()) { 6214 clearSyncStartEvent(); 6215 } else { 6216 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6217 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6218 } 6219 } 6220 6221 { 6222 AutoMutex lock(mLock); 6223 if (mActiveTrack != 0) { 6224 if (recordTrack != mActiveTrack.get()) { 6225 status = -EBUSY; 6226 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6227 mActiveTrack->mState = TrackBase::ACTIVE; 6228 } 6229 return status; 6230 } 6231 6232 recordTrack->mState = TrackBase::IDLE; 6233 mActiveTrack = recordTrack; 6234 mLock.unlock(); 6235 status_t status = AudioSystem::startInput(mId); 6236 mLock.lock(); 6237 if (status != NO_ERROR) { 6238 mActiveTrack.clear(); 6239 clearSyncStartEvent(); 6240 return status; 6241 } 6242 mRsmpInIndex = mFrameCount; 6243 mBytesRead = 0; 6244 if (mResampler != NULL) { 6245 mResampler->reset(); 6246 } 6247 mActiveTrack->mState = TrackBase::RESUMING; 6248 // signal thread to start 6249 ALOGV("Signal record thread"); 6250 mWaitWorkCV.signal(); 6251 // do not wait for mStartStopCond if exiting 6252 if (exitPending()) { 6253 mActiveTrack.clear(); 6254 status = INVALID_OPERATION; 6255 goto startError; 6256 } 6257 mStartStopCond.wait(mLock); 6258 if (mActiveTrack == 0) { 6259 ALOGV("Record failed to start"); 6260 status = BAD_VALUE; 6261 goto startError; 6262 } 6263 ALOGV("Record started OK"); 6264 return status; 6265 } 6266startError: 6267 AudioSystem::stopInput(mId); 6268 clearSyncStartEvent(); 6269 return status; 6270} 6271 6272void AudioFlinger::RecordThread::clearSyncStartEvent() 6273{ 6274 if (mSyncStartEvent != 0) { 6275 mSyncStartEvent->cancel(); 6276 } 6277 mSyncStartEvent.clear(); 6278 mFramestoDrop = 0; 6279} 6280 6281void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6282{ 6283 sp<SyncEvent> strongEvent = event.promote(); 6284 6285 if (strongEvent != 0) { 6286 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6287 me->handleSyncStartEvent(strongEvent); 6288 } 6289} 6290 6291void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6292{ 6293 if (event == mSyncStartEvent) { 6294 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6295 // from audio HAL 6296 mFramestoDrop = mFrameCount * 2; 6297 } 6298} 6299 6300void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6301 ALOGV("RecordThread::stop"); 6302 sp<ThreadBase> strongMe = this; 6303 { 6304 AutoMutex lock(mLock); 6305 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6306 mActiveTrack->mState = TrackBase::PAUSING; 6307 // do not wait for mStartStopCond if exiting 6308 if (exitPending()) { 6309 return; 6310 } 6311 mStartStopCond.wait(mLock); 6312 // if we have been restarted, recordTrack == mActiveTrack.get() here 6313 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6314 mLock.unlock(); 6315 AudioSystem::stopInput(mId); 6316 mLock.lock(); 6317 ALOGV("Record stopped OK"); 6318 } 6319 } 6320 } 6321} 6322 6323bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6324{ 6325 return false; 6326} 6327 6328status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6329{ 6330 if (!isValidSyncEvent(event)) { 6331 return BAD_VALUE; 6332 } 6333 6334 Mutex::Autolock _l(mLock); 6335 6336 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6337 mTrack->setSyncEvent(event); 6338 return NO_ERROR; 6339 } 6340 return NAME_NOT_FOUND; 6341} 6342 6343status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6344{ 6345 const size_t SIZE = 256; 6346 char buffer[SIZE]; 6347 String8 result; 6348 6349 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6350 result.append(buffer); 6351 6352 if (mActiveTrack != 0) { 6353 result.append("Active Track:\n"); 6354 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6355 mActiveTrack->dump(buffer, SIZE); 6356 result.append(buffer); 6357 6358 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6359 result.append(buffer); 6360 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6361 result.append(buffer); 6362 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6363 result.append(buffer); 6364 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6365 result.append(buffer); 6366 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6367 result.append(buffer); 6368 6369 6370 } else { 6371 result.append("No record client\n"); 6372 } 6373 write(fd, result.string(), result.size()); 6374 6375 dumpBase(fd, args); 6376 dumpEffectChains(fd, args); 6377 6378 return NO_ERROR; 6379} 6380 6381// AudioBufferProvider interface 6382status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6383{ 6384 size_t framesReq = buffer->frameCount; 6385 size_t framesReady = mFrameCount - mRsmpInIndex; 6386 int channelCount; 6387 6388 if (framesReady == 0) { 6389 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6390 if (mBytesRead < 0) { 6391 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6392 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6393 // Force input into standby so that it tries to 6394 // recover at next read attempt 6395 mInput->stream->common.standby(&mInput->stream->common); 6396 usleep(kRecordThreadSleepUs); 6397 } 6398 buffer->raw = NULL; 6399 buffer->frameCount = 0; 6400 return NOT_ENOUGH_DATA; 6401 } 6402 mRsmpInIndex = 0; 6403 framesReady = mFrameCount; 6404 } 6405 6406 if (framesReq > framesReady) { 6407 framesReq = framesReady; 6408 } 6409 6410 if (mChannelCount == 1 && mReqChannelCount == 2) { 6411 channelCount = 1; 6412 } else { 6413 channelCount = 2; 6414 } 6415 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6416 buffer->frameCount = framesReq; 6417 return NO_ERROR; 6418} 6419 6420// AudioBufferProvider interface 6421void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6422{ 6423 mRsmpInIndex += buffer->frameCount; 6424 buffer->frameCount = 0; 6425} 6426 6427bool AudioFlinger::RecordThread::checkForNewParameters_l() 6428{ 6429 bool reconfig = false; 6430 6431 while (!mNewParameters.isEmpty()) { 6432 status_t status = NO_ERROR; 6433 String8 keyValuePair = mNewParameters[0]; 6434 AudioParameter param = AudioParameter(keyValuePair); 6435 int value; 6436 audio_format_t reqFormat = mFormat; 6437 int reqSamplingRate = mReqSampleRate; 6438 int reqChannelCount = mReqChannelCount; 6439 6440 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6441 reqSamplingRate = value; 6442 reconfig = true; 6443 } 6444 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6445 reqFormat = (audio_format_t) value; 6446 reconfig = true; 6447 } 6448 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6449 reqChannelCount = popcount(value); 6450 reconfig = true; 6451 } 6452 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6453 // do not accept frame count changes if tracks are open as the track buffer 6454 // size depends on frame count and correct behavior would not be guaranteed 6455 // if frame count is changed after track creation 6456 if (mActiveTrack != 0) { 6457 status = INVALID_OPERATION; 6458 } else { 6459 reconfig = true; 6460 } 6461 } 6462 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6463 // forward device change to effects that have requested to be 6464 // aware of attached audio device. 6465 for (size_t i = 0; i < mEffectChains.size(); i++) { 6466 mEffectChains[i]->setDevice_l(value); 6467 } 6468 // store input device and output device but do not forward output device to audio HAL. 6469 // Note that status is ignored by the caller for output device 6470 // (see AudioFlinger::setParameters() 6471 uint32_t /*audio_devices_t*/ newDevice = mDevice; 6472 if (value & AUDIO_DEVICE_OUT_ALL) { 6473 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6474 status = BAD_VALUE; 6475 } else { 6476 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6477 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6478 if (mTrack != NULL) { 6479 bool suspend = audio_is_bluetooth_sco_device( 6480 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6481 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6482 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6483 } 6484 } 6485 newDevice |= value; 6486 mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once 6487 } 6488 if (status == NO_ERROR) { 6489 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6490 if (status == INVALID_OPERATION) { 6491 mInput->stream->common.standby(&mInput->stream->common); 6492 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6493 keyValuePair.string()); 6494 } 6495 if (reconfig) { 6496 if (status == BAD_VALUE && 6497 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6498 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6499 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6500 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6501 (reqChannelCount <= FCC_2)) { 6502 status = NO_ERROR; 6503 } 6504 if (status == NO_ERROR) { 6505 readInputParameters(); 6506 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6507 } 6508 } 6509 } 6510 6511 mNewParameters.removeAt(0); 6512 6513 mParamStatus = status; 6514 mParamCond.signal(); 6515 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6516 // already timed out waiting for the status and will never signal the condition. 6517 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6518 } 6519 return reconfig; 6520} 6521 6522String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6523{ 6524 char *s; 6525 String8 out_s8 = String8(); 6526 6527 Mutex::Autolock _l(mLock); 6528 if (initCheck() != NO_ERROR) { 6529 return out_s8; 6530 } 6531 6532 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6533 out_s8 = String8(s); 6534 free(s); 6535 return out_s8; 6536} 6537 6538void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6539 AudioSystem::OutputDescriptor desc; 6540 void *param2 = NULL; 6541 6542 switch (event) { 6543 case AudioSystem::INPUT_OPENED: 6544 case AudioSystem::INPUT_CONFIG_CHANGED: 6545 desc.channels = mChannelMask; 6546 desc.samplingRate = mSampleRate; 6547 desc.format = mFormat; 6548 desc.frameCount = mFrameCount; 6549 desc.latency = 0; 6550 param2 = &desc; 6551 break; 6552 6553 case AudioSystem::INPUT_CLOSED: 6554 default: 6555 break; 6556 } 6557 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6558} 6559 6560void AudioFlinger::RecordThread::readInputParameters() 6561{ 6562 delete mRsmpInBuffer; 6563 // mRsmpInBuffer is always assigned a new[] below 6564 delete mRsmpOutBuffer; 6565 mRsmpOutBuffer = NULL; 6566 delete mResampler; 6567 mResampler = NULL; 6568 6569 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6570 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6571 mChannelCount = (uint16_t)popcount(mChannelMask); 6572 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6573 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6574 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6575 mFrameCount = mInputBytes / mFrameSize; 6576 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6577 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6578 6579 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6580 { 6581 int channelCount; 6582 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6583 // stereo to mono post process as the resampler always outputs stereo. 6584 if (mChannelCount == 1 && mReqChannelCount == 2) { 6585 channelCount = 1; 6586 } else { 6587 channelCount = 2; 6588 } 6589 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6590 mResampler->setSampleRate(mSampleRate); 6591 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6592 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6593 6594 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6595 if (mChannelCount == 1 && mReqChannelCount == 1) { 6596 mFrameCount >>= 1; 6597 } 6598 6599 } 6600 mRsmpInIndex = mFrameCount; 6601} 6602 6603unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6604{ 6605 Mutex::Autolock _l(mLock); 6606 if (initCheck() != NO_ERROR) { 6607 return 0; 6608 } 6609 6610 return mInput->stream->get_input_frames_lost(mInput->stream); 6611} 6612 6613uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6614{ 6615 Mutex::Autolock _l(mLock); 6616 uint32_t result = 0; 6617 if (getEffectChain_l(sessionId) != 0) { 6618 result = EFFECT_SESSION; 6619 } 6620 6621 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6622 result |= TRACK_SESSION; 6623 } 6624 6625 return result; 6626} 6627 6628AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6629{ 6630 Mutex::Autolock _l(mLock); 6631 return mTrack; 6632} 6633 6634AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6635{ 6636 Mutex::Autolock _l(mLock); 6637 AudioStreamIn *input = mInput; 6638 mInput = NULL; 6639 return input; 6640} 6641 6642// this method must always be called either with ThreadBase mLock held or inside the thread loop 6643audio_stream_t* AudioFlinger::RecordThread::stream() const 6644{ 6645 if (mInput == NULL) { 6646 return NULL; 6647 } 6648 return &mInput->stream->common; 6649} 6650 6651 6652// ---------------------------------------------------------------------------- 6653 6654audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6655{ 6656 if (!settingsAllowed()) { 6657 return 0; 6658 } 6659 Mutex::Autolock _l(mLock); 6660 return loadHwModule_l(name); 6661} 6662 6663// loadHwModule_l() must be called with AudioFlinger::mLock held 6664audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6665{ 6666 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6667 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6668 ALOGW("loadHwModule() module %s already loaded", name); 6669 return mAudioHwDevs.keyAt(i); 6670 } 6671 } 6672 6673 audio_hw_device_t *dev; 6674 6675 int rc = load_audio_interface(name, &dev); 6676 if (rc) { 6677 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6678 return 0; 6679 } 6680 6681 mHardwareStatus = AUDIO_HW_INIT; 6682 rc = dev->init_check(dev); 6683 mHardwareStatus = AUDIO_HW_IDLE; 6684 if (rc) { 6685 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6686 return 0; 6687 } 6688 6689 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6690 (NULL != dev->set_master_volume)) { 6691 AutoMutex lock(mHardwareLock); 6692 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6693 dev->set_master_volume(dev, mMasterVolume); 6694 mHardwareStatus = AUDIO_HW_IDLE; 6695 } 6696 6697 audio_module_handle_t handle = nextUniqueId(); 6698 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6699 6700 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6701 name, dev->common.module->name, dev->common.module->id, handle); 6702 6703 return handle; 6704 6705} 6706 6707audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6708 audio_devices_t *pDevices, 6709 uint32_t *pSamplingRate, 6710 audio_format_t *pFormat, 6711 audio_channel_mask_t *pChannelMask, 6712 uint32_t *pLatencyMs, 6713 audio_output_flags_t flags) 6714{ 6715 status_t status; 6716 PlaybackThread *thread = NULL; 6717 struct audio_config config = { 6718 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6719 channel_mask: pChannelMask ? *pChannelMask : 0, 6720 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6721 }; 6722 audio_stream_out_t *outStream = NULL; 6723 audio_hw_device_t *outHwDev; 6724 6725 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6726 module, 6727 (pDevices != NULL) ? *pDevices : 0, 6728 config.sample_rate, 6729 config.format, 6730 config.channel_mask, 6731 flags); 6732 6733 if (pDevices == NULL || *pDevices == 0) { 6734 return 0; 6735 } 6736 6737 Mutex::Autolock _l(mLock); 6738 6739 outHwDev = findSuitableHwDev_l(module, *pDevices); 6740 if (outHwDev == NULL) 6741 return 0; 6742 6743 audio_io_handle_t id = nextUniqueId(); 6744 6745 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6746 6747 status = outHwDev->open_output_stream(outHwDev, 6748 id, 6749 *pDevices, 6750 (audio_output_flags_t)flags, 6751 &config, 6752 &outStream); 6753 6754 mHardwareStatus = AUDIO_HW_IDLE; 6755 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6756 outStream, 6757 config.sample_rate, 6758 config.format, 6759 config.channel_mask, 6760 status); 6761 6762 if (status == NO_ERROR && outStream != NULL) { 6763 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6764 6765 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6766 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6767 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6768 thread = new DirectOutputThread(this, output, id, *pDevices); 6769 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6770 } else { 6771 thread = new MixerThread(this, output, id, *pDevices); 6772 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6773 } 6774 mPlaybackThreads.add(id, thread); 6775 6776 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6777 if (pFormat != NULL) *pFormat = config.format; 6778 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6779 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6780 6781 // notify client processes of the new output creation 6782 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6783 6784 // the first primary output opened designates the primary hw device 6785 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6786 ALOGI("Using module %d has the primary audio interface", module); 6787 mPrimaryHardwareDev = outHwDev; 6788 6789 AutoMutex lock(mHardwareLock); 6790 mHardwareStatus = AUDIO_HW_SET_MODE; 6791 outHwDev->set_mode(outHwDev, mMode); 6792 6793 // Determine the level of master volume support the primary audio HAL has, 6794 // and set the initial master volume at the same time. 6795 float initialVolume = 1.0; 6796 mMasterVolumeSupportLvl = MVS_NONE; 6797 6798 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6799 if ((NULL != outHwDev->get_master_volume) && 6800 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6801 mMasterVolumeSupportLvl = MVS_FULL; 6802 } else { 6803 mMasterVolumeSupportLvl = MVS_SETONLY; 6804 initialVolume = 1.0; 6805 } 6806 6807 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6808 if ((NULL == outHwDev->set_master_volume) || 6809 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6810 mMasterVolumeSupportLvl = MVS_NONE; 6811 } 6812 // now that we have a primary device, initialize master volume on other devices 6813 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6814 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6815 6816 if ((dev != mPrimaryHardwareDev) && 6817 (NULL != dev->set_master_volume)) { 6818 dev->set_master_volume(dev, initialVolume); 6819 } 6820 } 6821 mHardwareStatus = AUDIO_HW_IDLE; 6822 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6823 ? initialVolume 6824 : 1.0; 6825 mMasterVolume = initialVolume; 6826 } 6827 return id; 6828 } 6829 6830 return 0; 6831} 6832 6833audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6834 audio_io_handle_t output2) 6835{ 6836 Mutex::Autolock _l(mLock); 6837 MixerThread *thread1 = checkMixerThread_l(output1); 6838 MixerThread *thread2 = checkMixerThread_l(output2); 6839 6840 if (thread1 == NULL || thread2 == NULL) { 6841 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6842 return 0; 6843 } 6844 6845 audio_io_handle_t id = nextUniqueId(); 6846 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6847 thread->addOutputTrack(thread2); 6848 mPlaybackThreads.add(id, thread); 6849 // notify client processes of the new output creation 6850 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6851 return id; 6852} 6853 6854status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6855{ 6856 return closeOutput_nonvirtual(output); 6857} 6858 6859status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6860{ 6861 // keep strong reference on the playback thread so that 6862 // it is not destroyed while exit() is executed 6863 sp<PlaybackThread> thread; 6864 { 6865 Mutex::Autolock _l(mLock); 6866 thread = checkPlaybackThread_l(output); 6867 if (thread == NULL) { 6868 return BAD_VALUE; 6869 } 6870 6871 ALOGV("closeOutput() %d", output); 6872 6873 if (thread->type() == ThreadBase::MIXER) { 6874 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6875 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6876 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6877 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6878 } 6879 } 6880 } 6881 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6882 mPlaybackThreads.removeItem(output); 6883 } 6884 thread->exit(); 6885 // The thread entity (active unit of execution) is no longer running here, 6886 // but the ThreadBase container still exists. 6887 6888 if (thread->type() != ThreadBase::DUPLICATING) { 6889 AudioStreamOut *out = thread->clearOutput(); 6890 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6891 // from now on thread->mOutput is NULL 6892 out->hwDev->close_output_stream(out->hwDev, out->stream); 6893 delete out; 6894 } 6895 return NO_ERROR; 6896} 6897 6898status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6899{ 6900 Mutex::Autolock _l(mLock); 6901 PlaybackThread *thread = checkPlaybackThread_l(output); 6902 6903 if (thread == NULL) { 6904 return BAD_VALUE; 6905 } 6906 6907 ALOGV("suspendOutput() %d", output); 6908 thread->suspend(); 6909 6910 return NO_ERROR; 6911} 6912 6913status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6914{ 6915 Mutex::Autolock _l(mLock); 6916 PlaybackThread *thread = checkPlaybackThread_l(output); 6917 6918 if (thread == NULL) { 6919 return BAD_VALUE; 6920 } 6921 6922 ALOGV("restoreOutput() %d", output); 6923 6924 thread->restore(); 6925 6926 return NO_ERROR; 6927} 6928 6929audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6930 audio_devices_t *pDevices, 6931 uint32_t *pSamplingRate, 6932 audio_format_t *pFormat, 6933 audio_channel_mask_t *pChannelMask) 6934{ 6935 status_t status; 6936 RecordThread *thread = NULL; 6937 struct audio_config config = { 6938 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6939 channel_mask: pChannelMask ? *pChannelMask : 0, 6940 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6941 }; 6942 uint32_t reqSamplingRate = config.sample_rate; 6943 audio_format_t reqFormat = config.format; 6944 audio_channel_mask_t reqChannels = config.channel_mask; 6945 audio_stream_in_t *inStream = NULL; 6946 audio_hw_device_t *inHwDev; 6947 6948 if (pDevices == NULL || *pDevices == 0) { 6949 return 0; 6950 } 6951 6952 Mutex::Autolock _l(mLock); 6953 6954 inHwDev = findSuitableHwDev_l(module, *pDevices); 6955 if (inHwDev == NULL) 6956 return 0; 6957 6958 audio_io_handle_t id = nextUniqueId(); 6959 6960 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6961 &inStream); 6962 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6963 inStream, 6964 config.sample_rate, 6965 config.format, 6966 config.channel_mask, 6967 status); 6968 6969 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6970 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6971 // or stereo to mono conversions on 16 bit PCM inputs. 6972 if (status == BAD_VALUE && 6973 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6974 (config.sample_rate <= 2 * reqSamplingRate) && 6975 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6976 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 6977 inStream = NULL; 6978 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6979 } 6980 6981 if (status == NO_ERROR && inStream != NULL) { 6982 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6983 6984 // Start record thread 6985 // RecorThread require both input and output device indication to forward to audio 6986 // pre processing modules 6987 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 6988 thread = new RecordThread(this, 6989 input, 6990 reqSamplingRate, 6991 reqChannels, 6992 id, 6993 device); 6994 mRecordThreads.add(id, thread); 6995 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6996 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6997 if (pFormat != NULL) *pFormat = config.format; 6998 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6999 7000 input->stream->common.standby(&input->stream->common); 7001 7002 // notify client processes of the new input creation 7003 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7004 return id; 7005 } 7006 7007 return 0; 7008} 7009 7010status_t AudioFlinger::closeInput(audio_io_handle_t input) 7011{ 7012 return closeInput_nonvirtual(input); 7013} 7014 7015status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7016{ 7017 // keep strong reference on the record thread so that 7018 // it is not destroyed while exit() is executed 7019 sp<RecordThread> thread; 7020 { 7021 Mutex::Autolock _l(mLock); 7022 thread = checkRecordThread_l(input); 7023 if (thread == 0) { 7024 return BAD_VALUE; 7025 } 7026 7027 ALOGV("closeInput() %d", input); 7028 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7029 mRecordThreads.removeItem(input); 7030 } 7031 thread->exit(); 7032 // The thread entity (active unit of execution) is no longer running here, 7033 // but the ThreadBase container still exists. 7034 7035 AudioStreamIn *in = thread->clearInput(); 7036 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7037 // from now on thread->mInput is NULL 7038 in->hwDev->close_input_stream(in->hwDev, in->stream); 7039 delete in; 7040 7041 return NO_ERROR; 7042} 7043 7044status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7045{ 7046 Mutex::Autolock _l(mLock); 7047 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7048 7049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7050 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7051 thread->invalidateTracks(stream); 7052 } 7053 7054 return NO_ERROR; 7055} 7056 7057 7058int AudioFlinger::newAudioSessionId() 7059{ 7060 return nextUniqueId(); 7061} 7062 7063void AudioFlinger::acquireAudioSessionId(int audioSession) 7064{ 7065 Mutex::Autolock _l(mLock); 7066 pid_t caller = IPCThreadState::self()->getCallingPid(); 7067 ALOGV("acquiring %d from %d", audioSession, caller); 7068 size_t num = mAudioSessionRefs.size(); 7069 for (size_t i = 0; i< num; i++) { 7070 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7071 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7072 ref->mCnt++; 7073 ALOGV(" incremented refcount to %d", ref->mCnt); 7074 return; 7075 } 7076 } 7077 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7078 ALOGV(" added new entry for %d", audioSession); 7079} 7080 7081void AudioFlinger::releaseAudioSessionId(int audioSession) 7082{ 7083 Mutex::Autolock _l(mLock); 7084 pid_t caller = IPCThreadState::self()->getCallingPid(); 7085 ALOGV("releasing %d from %d", audioSession, caller); 7086 size_t num = mAudioSessionRefs.size(); 7087 for (size_t i = 0; i< num; i++) { 7088 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7089 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7090 ref->mCnt--; 7091 ALOGV(" decremented refcount to %d", ref->mCnt); 7092 if (ref->mCnt == 0) { 7093 mAudioSessionRefs.removeAt(i); 7094 delete ref; 7095 purgeStaleEffects_l(); 7096 } 7097 return; 7098 } 7099 } 7100 ALOGW("session id %d not found for pid %d", audioSession, caller); 7101} 7102 7103void AudioFlinger::purgeStaleEffects_l() { 7104 7105 ALOGV("purging stale effects"); 7106 7107 Vector< sp<EffectChain> > chains; 7108 7109 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7110 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7111 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7112 sp<EffectChain> ec = t->mEffectChains[j]; 7113 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7114 chains.push(ec); 7115 } 7116 } 7117 } 7118 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7119 sp<RecordThread> t = mRecordThreads.valueAt(i); 7120 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7121 sp<EffectChain> ec = t->mEffectChains[j]; 7122 chains.push(ec); 7123 } 7124 } 7125 7126 for (size_t i = 0; i < chains.size(); i++) { 7127 sp<EffectChain> ec = chains[i]; 7128 int sessionid = ec->sessionId(); 7129 sp<ThreadBase> t = ec->mThread.promote(); 7130 if (t == 0) { 7131 continue; 7132 } 7133 size_t numsessionrefs = mAudioSessionRefs.size(); 7134 bool found = false; 7135 for (size_t k = 0; k < numsessionrefs; k++) { 7136 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7137 if (ref->mSessionid == sessionid) { 7138 ALOGV(" session %d still exists for %d with %d refs", 7139 sessionid, ref->mPid, ref->mCnt); 7140 found = true; 7141 break; 7142 } 7143 } 7144 if (!found) { 7145 Mutex::Autolock _l (t->mLock); 7146 // remove all effects from the chain 7147 while (ec->mEffects.size()) { 7148 sp<EffectModule> effect = ec->mEffects[0]; 7149 effect->unPin(); 7150 t->removeEffect_l(effect); 7151 if (effect->purgeHandles()) { 7152 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7153 } 7154 AudioSystem::unregisterEffect(effect->id()); 7155 } 7156 } 7157 } 7158 return; 7159} 7160 7161// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7162AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7163{ 7164 return mPlaybackThreads.valueFor(output).get(); 7165} 7166 7167// checkMixerThread_l() must be called with AudioFlinger::mLock held 7168AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7169{ 7170 PlaybackThread *thread = checkPlaybackThread_l(output); 7171 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7172} 7173 7174// checkRecordThread_l() must be called with AudioFlinger::mLock held 7175AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7176{ 7177 return mRecordThreads.valueFor(input).get(); 7178} 7179 7180uint32_t AudioFlinger::nextUniqueId() 7181{ 7182 return android_atomic_inc(&mNextUniqueId); 7183} 7184 7185AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7186{ 7187 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7188 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7189 AudioStreamOut *output = thread->getOutput(); 7190 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7191 return thread; 7192 } 7193 } 7194 return NULL; 7195} 7196 7197audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7198{ 7199 PlaybackThread *thread = primaryPlaybackThread_l(); 7200 7201 if (thread == NULL) { 7202 return 0; 7203 } 7204 7205 return thread->device(); 7206} 7207 7208sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7209 int triggerSession, 7210 int listenerSession, 7211 sync_event_callback_t callBack, 7212 void *cookie) 7213{ 7214 Mutex::Autolock _l(mLock); 7215 7216 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7217 status_t playStatus = NAME_NOT_FOUND; 7218 status_t recStatus = NAME_NOT_FOUND; 7219 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7220 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7221 if (playStatus == NO_ERROR) { 7222 return event; 7223 } 7224 } 7225 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7226 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7227 if (recStatus == NO_ERROR) { 7228 return event; 7229 } 7230 } 7231 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7232 mPendingSyncEvents.add(event); 7233 } else { 7234 ALOGV("createSyncEvent() invalid event %d", event->type()); 7235 event.clear(); 7236 } 7237 return event; 7238} 7239 7240// ---------------------------------------------------------------------------- 7241// Effect management 7242// ---------------------------------------------------------------------------- 7243 7244 7245status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7246{ 7247 Mutex::Autolock _l(mLock); 7248 return EffectQueryNumberEffects(numEffects); 7249} 7250 7251status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7252{ 7253 Mutex::Autolock _l(mLock); 7254 return EffectQueryEffect(index, descriptor); 7255} 7256 7257status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7258 effect_descriptor_t *descriptor) const 7259{ 7260 Mutex::Autolock _l(mLock); 7261 return EffectGetDescriptor(pUuid, descriptor); 7262} 7263 7264 7265sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7266 effect_descriptor_t *pDesc, 7267 const sp<IEffectClient>& effectClient, 7268 int32_t priority, 7269 audio_io_handle_t io, 7270 int sessionId, 7271 status_t *status, 7272 int *id, 7273 int *enabled) 7274{ 7275 status_t lStatus = NO_ERROR; 7276 sp<EffectHandle> handle; 7277 effect_descriptor_t desc; 7278 7279 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7280 pid, effectClient.get(), priority, sessionId, io); 7281 7282 if (pDesc == NULL) { 7283 lStatus = BAD_VALUE; 7284 goto Exit; 7285 } 7286 7287 // check audio settings permission for global effects 7288 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7289 lStatus = PERMISSION_DENIED; 7290 goto Exit; 7291 } 7292 7293 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7294 // that can only be created by audio policy manager (running in same process) 7295 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7296 lStatus = PERMISSION_DENIED; 7297 goto Exit; 7298 } 7299 7300 if (io == 0) { 7301 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7302 // output must be specified by AudioPolicyManager when using session 7303 // AUDIO_SESSION_OUTPUT_STAGE 7304 lStatus = BAD_VALUE; 7305 goto Exit; 7306 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7307 // if the output returned by getOutputForEffect() is removed before we lock the 7308 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7309 // and we will exit safely 7310 io = AudioSystem::getOutputForEffect(&desc); 7311 } 7312 } 7313 7314 { 7315 Mutex::Autolock _l(mLock); 7316 7317 7318 if (!EffectIsNullUuid(&pDesc->uuid)) { 7319 // if uuid is specified, request effect descriptor 7320 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7321 if (lStatus < 0) { 7322 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7323 goto Exit; 7324 } 7325 } else { 7326 // if uuid is not specified, look for an available implementation 7327 // of the required type in effect factory 7328 if (EffectIsNullUuid(&pDesc->type)) { 7329 ALOGW("createEffect() no effect type"); 7330 lStatus = BAD_VALUE; 7331 goto Exit; 7332 } 7333 uint32_t numEffects = 0; 7334 effect_descriptor_t d; 7335 d.flags = 0; // prevent compiler warning 7336 bool found = false; 7337 7338 lStatus = EffectQueryNumberEffects(&numEffects); 7339 if (lStatus < 0) { 7340 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7341 goto Exit; 7342 } 7343 for (uint32_t i = 0; i < numEffects; i++) { 7344 lStatus = EffectQueryEffect(i, &desc); 7345 if (lStatus < 0) { 7346 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7347 continue; 7348 } 7349 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7350 // If matching type found save effect descriptor. If the session is 7351 // 0 and the effect is not auxiliary, continue enumeration in case 7352 // an auxiliary version of this effect type is available 7353 found = true; 7354 d = desc; 7355 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7356 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7357 break; 7358 } 7359 } 7360 } 7361 if (!found) { 7362 lStatus = BAD_VALUE; 7363 ALOGW("createEffect() effect not found"); 7364 goto Exit; 7365 } 7366 // For same effect type, chose auxiliary version over insert version if 7367 // connect to output mix (Compliance to OpenSL ES) 7368 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7369 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7370 desc = d; 7371 } 7372 } 7373 7374 // Do not allow auxiliary effects on a session different from 0 (output mix) 7375 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7376 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7377 lStatus = INVALID_OPERATION; 7378 goto Exit; 7379 } 7380 7381 // check recording permission for visualizer 7382 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7383 !recordingAllowed()) { 7384 lStatus = PERMISSION_DENIED; 7385 goto Exit; 7386 } 7387 7388 // return effect descriptor 7389 *pDesc = desc; 7390 7391 // If output is not specified try to find a matching audio session ID in one of the 7392 // output threads. 7393 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7394 // because of code checking output when entering the function. 7395 // Note: io is never 0 when creating an effect on an input 7396 if (io == 0) { 7397 // look for the thread where the specified audio session is present 7398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7399 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7400 io = mPlaybackThreads.keyAt(i); 7401 break; 7402 } 7403 } 7404 if (io == 0) { 7405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7406 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7407 io = mRecordThreads.keyAt(i); 7408 break; 7409 } 7410 } 7411 } 7412 // If no output thread contains the requested session ID, default to 7413 // first output. The effect chain will be moved to the correct output 7414 // thread when a track with the same session ID is created 7415 if (io == 0 && mPlaybackThreads.size()) { 7416 io = mPlaybackThreads.keyAt(0); 7417 } 7418 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7419 } 7420 ThreadBase *thread = checkRecordThread_l(io); 7421 if (thread == NULL) { 7422 thread = checkPlaybackThread_l(io); 7423 if (thread == NULL) { 7424 ALOGE("createEffect() unknown output thread"); 7425 lStatus = BAD_VALUE; 7426 goto Exit; 7427 } 7428 } 7429 7430 sp<Client> client = registerPid_l(pid); 7431 7432 // create effect on selected output thread 7433 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7434 &desc, enabled, &lStatus); 7435 if (handle != 0 && id != NULL) { 7436 *id = handle->id(); 7437 } 7438 } 7439 7440Exit: 7441 if (status != NULL) { 7442 *status = lStatus; 7443 } 7444 return handle; 7445} 7446 7447status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7448 audio_io_handle_t dstOutput) 7449{ 7450 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7451 sessionId, srcOutput, dstOutput); 7452 Mutex::Autolock _l(mLock); 7453 if (srcOutput == dstOutput) { 7454 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7455 return NO_ERROR; 7456 } 7457 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7458 if (srcThread == NULL) { 7459 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7460 return BAD_VALUE; 7461 } 7462 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7463 if (dstThread == NULL) { 7464 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7465 return BAD_VALUE; 7466 } 7467 7468 Mutex::Autolock _dl(dstThread->mLock); 7469 Mutex::Autolock _sl(srcThread->mLock); 7470 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7471 7472 return NO_ERROR; 7473} 7474 7475// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7476status_t AudioFlinger::moveEffectChain_l(int sessionId, 7477 AudioFlinger::PlaybackThread *srcThread, 7478 AudioFlinger::PlaybackThread *dstThread, 7479 bool reRegister) 7480{ 7481 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7482 sessionId, srcThread, dstThread); 7483 7484 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7485 if (chain == 0) { 7486 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7487 sessionId, srcThread); 7488 return INVALID_OPERATION; 7489 } 7490 7491 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7492 // so that a new chain is created with correct parameters when first effect is added. This is 7493 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7494 // removed. 7495 srcThread->removeEffectChain_l(chain); 7496 7497 // transfer all effects one by one so that new effect chain is created on new thread with 7498 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7499 audio_io_handle_t dstOutput = dstThread->id(); 7500 sp<EffectChain> dstChain; 7501 uint32_t strategy = 0; // prevent compiler warning 7502 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7503 while (effect != 0) { 7504 srcThread->removeEffect_l(effect); 7505 dstThread->addEffect_l(effect); 7506 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7507 if (effect->state() == EffectModule::ACTIVE || 7508 effect->state() == EffectModule::STOPPING) { 7509 effect->start(); 7510 } 7511 // if the move request is not received from audio policy manager, the effect must be 7512 // re-registered with the new strategy and output 7513 if (dstChain == 0) { 7514 dstChain = effect->chain().promote(); 7515 if (dstChain == 0) { 7516 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7517 srcThread->addEffect_l(effect); 7518 return NO_INIT; 7519 } 7520 strategy = dstChain->strategy(); 7521 } 7522 if (reRegister) { 7523 AudioSystem::unregisterEffect(effect->id()); 7524 AudioSystem::registerEffect(&effect->desc(), 7525 dstOutput, 7526 strategy, 7527 sessionId, 7528 effect->id()); 7529 } 7530 effect = chain->getEffectFromId_l(0); 7531 } 7532 7533 return NO_ERROR; 7534} 7535 7536 7537// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7538sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7539 const sp<AudioFlinger::Client>& client, 7540 const sp<IEffectClient>& effectClient, 7541 int32_t priority, 7542 int sessionId, 7543 effect_descriptor_t *desc, 7544 int *enabled, 7545 status_t *status 7546 ) 7547{ 7548 sp<EffectModule> effect; 7549 sp<EffectHandle> handle; 7550 status_t lStatus; 7551 sp<EffectChain> chain; 7552 bool chainCreated = false; 7553 bool effectCreated = false; 7554 bool effectRegistered = false; 7555 7556 lStatus = initCheck(); 7557 if (lStatus != NO_ERROR) { 7558 ALOGW("createEffect_l() Audio driver not initialized."); 7559 goto Exit; 7560 } 7561 7562 // Do not allow effects with session ID 0 on direct output or duplicating threads 7563 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7564 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7565 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7566 desc->name, sessionId); 7567 lStatus = BAD_VALUE; 7568 goto Exit; 7569 } 7570 // Only Pre processor effects are allowed on input threads and only on input threads 7571 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7572 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7573 desc->name, desc->flags, mType); 7574 lStatus = BAD_VALUE; 7575 goto Exit; 7576 } 7577 7578 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7579 7580 { // scope for mLock 7581 Mutex::Autolock _l(mLock); 7582 7583 // check for existing effect chain with the requested audio session 7584 chain = getEffectChain_l(sessionId); 7585 if (chain == 0) { 7586 // create a new chain for this session 7587 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7588 chain = new EffectChain(this, sessionId); 7589 addEffectChain_l(chain); 7590 chain->setStrategy(getStrategyForSession_l(sessionId)); 7591 chainCreated = true; 7592 } else { 7593 effect = chain->getEffectFromDesc_l(desc); 7594 } 7595 7596 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7597 7598 if (effect == 0) { 7599 int id = mAudioFlinger->nextUniqueId(); 7600 // Check CPU and memory usage 7601 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7602 if (lStatus != NO_ERROR) { 7603 goto Exit; 7604 } 7605 effectRegistered = true; 7606 // create a new effect module if none present in the chain 7607 effect = new EffectModule(this, chain, desc, id, sessionId); 7608 lStatus = effect->status(); 7609 if (lStatus != NO_ERROR) { 7610 goto Exit; 7611 } 7612 lStatus = chain->addEffect_l(effect); 7613 if (lStatus != NO_ERROR) { 7614 goto Exit; 7615 } 7616 effectCreated = true; 7617 7618 effect->setDevice(mDevice); 7619 effect->setMode(mAudioFlinger->getMode()); 7620 } 7621 // create effect handle and connect it to effect module 7622 handle = new EffectHandle(effect, client, effectClient, priority); 7623 lStatus = effect->addHandle(handle.get()); 7624 if (enabled != NULL) { 7625 *enabled = (int)effect->isEnabled(); 7626 } 7627 } 7628 7629Exit: 7630 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7631 Mutex::Autolock _l(mLock); 7632 if (effectCreated) { 7633 chain->removeEffect_l(effect); 7634 } 7635 if (effectRegistered) { 7636 AudioSystem::unregisterEffect(effect->id()); 7637 } 7638 if (chainCreated) { 7639 removeEffectChain_l(chain); 7640 } 7641 handle.clear(); 7642 } 7643 7644 if (status != NULL) { 7645 *status = lStatus; 7646 } 7647 return handle; 7648} 7649 7650sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7651{ 7652 Mutex::Autolock _l(mLock); 7653 return getEffect_l(sessionId, effectId); 7654} 7655 7656sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7657{ 7658 sp<EffectChain> chain = getEffectChain_l(sessionId); 7659 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7660} 7661 7662// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7663// PlaybackThread::mLock held 7664status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7665{ 7666 // check for existing effect chain with the requested audio session 7667 int sessionId = effect->sessionId(); 7668 sp<EffectChain> chain = getEffectChain_l(sessionId); 7669 bool chainCreated = false; 7670 7671 if (chain == 0) { 7672 // create a new chain for this session 7673 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7674 chain = new EffectChain(this, sessionId); 7675 addEffectChain_l(chain); 7676 chain->setStrategy(getStrategyForSession_l(sessionId)); 7677 chainCreated = true; 7678 } 7679 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7680 7681 if (chain->getEffectFromId_l(effect->id()) != 0) { 7682 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7683 this, effect->desc().name, chain.get()); 7684 return BAD_VALUE; 7685 } 7686 7687 status_t status = chain->addEffect_l(effect); 7688 if (status != NO_ERROR) { 7689 if (chainCreated) { 7690 removeEffectChain_l(chain); 7691 } 7692 return status; 7693 } 7694 7695 effect->setDevice(mDevice); 7696 effect->setMode(mAudioFlinger->getMode()); 7697 return NO_ERROR; 7698} 7699 7700void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7701 7702 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7703 effect_descriptor_t desc = effect->desc(); 7704 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7705 detachAuxEffect_l(effect->id()); 7706 } 7707 7708 sp<EffectChain> chain = effect->chain().promote(); 7709 if (chain != 0) { 7710 // remove effect chain if removing last effect 7711 if (chain->removeEffect_l(effect) == 0) { 7712 removeEffectChain_l(chain); 7713 } 7714 } else { 7715 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7716 } 7717} 7718 7719void AudioFlinger::ThreadBase::lockEffectChains_l( 7720 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7721{ 7722 effectChains = mEffectChains; 7723 for (size_t i = 0; i < mEffectChains.size(); i++) { 7724 mEffectChains[i]->lock(); 7725 } 7726} 7727 7728void AudioFlinger::ThreadBase::unlockEffectChains( 7729 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7730{ 7731 for (size_t i = 0; i < effectChains.size(); i++) { 7732 effectChains[i]->unlock(); 7733 } 7734} 7735 7736sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7737{ 7738 Mutex::Autolock _l(mLock); 7739 return getEffectChain_l(sessionId); 7740} 7741 7742sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7743{ 7744 size_t size = mEffectChains.size(); 7745 for (size_t i = 0; i < size; i++) { 7746 if (mEffectChains[i]->sessionId() == sessionId) { 7747 return mEffectChains[i]; 7748 } 7749 } 7750 return 0; 7751} 7752 7753void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7754{ 7755 Mutex::Autolock _l(mLock); 7756 size_t size = mEffectChains.size(); 7757 for (size_t i = 0; i < size; i++) { 7758 mEffectChains[i]->setMode_l(mode); 7759 } 7760} 7761 7762void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7763 EffectHandle *handle, 7764 bool unpinIfLast) { 7765 7766 Mutex::Autolock _l(mLock); 7767 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7768 // delete the effect module if removing last handle on it 7769 if (effect->removeHandle(handle) == 0) { 7770 if (!effect->isPinned() || unpinIfLast) { 7771 removeEffect_l(effect); 7772 AudioSystem::unregisterEffect(effect->id()); 7773 } 7774 } 7775} 7776 7777status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7778{ 7779 int session = chain->sessionId(); 7780 int16_t *buffer = mMixBuffer; 7781 bool ownsBuffer = false; 7782 7783 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7784 if (session > 0) { 7785 // Only one effect chain can be present in direct output thread and it uses 7786 // the mix buffer as input 7787 if (mType != DIRECT) { 7788 size_t numSamples = mNormalFrameCount * mChannelCount; 7789 buffer = new int16_t[numSamples]; 7790 memset(buffer, 0, numSamples * sizeof(int16_t)); 7791 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7792 ownsBuffer = true; 7793 } 7794 7795 // Attach all tracks with same session ID to this chain. 7796 for (size_t i = 0; i < mTracks.size(); ++i) { 7797 sp<Track> track = mTracks[i]; 7798 if (session == track->sessionId()) { 7799 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7800 track->setMainBuffer(buffer); 7801 chain->incTrackCnt(); 7802 } 7803 } 7804 7805 // indicate all active tracks in the chain 7806 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7807 sp<Track> track = mActiveTracks[i].promote(); 7808 if (track == 0) continue; 7809 if (session == track->sessionId()) { 7810 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7811 chain->incActiveTrackCnt(); 7812 } 7813 } 7814 } 7815 7816 chain->setInBuffer(buffer, ownsBuffer); 7817 chain->setOutBuffer(mMixBuffer); 7818 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7819 // chains list in order to be processed last as it contains output stage effects 7820 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7821 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7822 // after track specific effects and before output stage 7823 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7824 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7825 // Effect chain for other sessions are inserted at beginning of effect 7826 // chains list to be processed before output mix effects. Relative order between other 7827 // sessions is not important 7828 size_t size = mEffectChains.size(); 7829 size_t i = 0; 7830 for (i = 0; i < size; i++) { 7831 if (mEffectChains[i]->sessionId() < session) break; 7832 } 7833 mEffectChains.insertAt(chain, i); 7834 checkSuspendOnAddEffectChain_l(chain); 7835 7836 return NO_ERROR; 7837} 7838 7839size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7840{ 7841 int session = chain->sessionId(); 7842 7843 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7844 7845 for (size_t i = 0; i < mEffectChains.size(); i++) { 7846 if (chain == mEffectChains[i]) { 7847 mEffectChains.removeAt(i); 7848 // detach all active tracks from the chain 7849 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7850 sp<Track> track = mActiveTracks[i].promote(); 7851 if (track == 0) continue; 7852 if (session == track->sessionId()) { 7853 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7854 chain.get(), session); 7855 chain->decActiveTrackCnt(); 7856 } 7857 } 7858 7859 // detach all tracks with same session ID from this chain 7860 for (size_t i = 0; i < mTracks.size(); ++i) { 7861 sp<Track> track = mTracks[i]; 7862 if (session == track->sessionId()) { 7863 track->setMainBuffer(mMixBuffer); 7864 chain->decTrackCnt(); 7865 } 7866 } 7867 break; 7868 } 7869 } 7870 return mEffectChains.size(); 7871} 7872 7873status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7874 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7875{ 7876 Mutex::Autolock _l(mLock); 7877 return attachAuxEffect_l(track, EffectId); 7878} 7879 7880status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7881 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7882{ 7883 status_t status = NO_ERROR; 7884 7885 if (EffectId == 0) { 7886 track->setAuxBuffer(0, NULL); 7887 } else { 7888 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7889 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7890 if (effect != 0) { 7891 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7892 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7893 } else { 7894 status = INVALID_OPERATION; 7895 } 7896 } else { 7897 status = BAD_VALUE; 7898 } 7899 } 7900 return status; 7901} 7902 7903void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7904{ 7905 for (size_t i = 0; i < mTracks.size(); ++i) { 7906 sp<Track> track = mTracks[i]; 7907 if (track->auxEffectId() == effectId) { 7908 attachAuxEffect_l(track, 0); 7909 } 7910 } 7911} 7912 7913status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7914{ 7915 // only one chain per input thread 7916 if (mEffectChains.size() != 0) { 7917 return INVALID_OPERATION; 7918 } 7919 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7920 7921 chain->setInBuffer(NULL); 7922 chain->setOutBuffer(NULL); 7923 7924 checkSuspendOnAddEffectChain_l(chain); 7925 7926 mEffectChains.add(chain); 7927 7928 return NO_ERROR; 7929} 7930 7931size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7932{ 7933 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7934 ALOGW_IF(mEffectChains.size() != 1, 7935 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7936 chain.get(), mEffectChains.size(), this); 7937 if (mEffectChains.size() == 1) { 7938 mEffectChains.removeAt(0); 7939 } 7940 return 0; 7941} 7942 7943// ---------------------------------------------------------------------------- 7944// EffectModule implementation 7945// ---------------------------------------------------------------------------- 7946 7947#undef LOG_TAG 7948#define LOG_TAG "AudioFlinger::EffectModule" 7949 7950AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7951 const wp<AudioFlinger::EffectChain>& chain, 7952 effect_descriptor_t *desc, 7953 int id, 7954 int sessionId) 7955 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7956 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7957 // mDescriptor is set below 7958 // mConfig is set by configure() and not used before then 7959 mEffectInterface(NULL), 7960 mStatus(NO_INIT), mState(IDLE), 7961 // mMaxDisableWaitCnt is set by configure() and not used before then 7962 // mDisableWaitCnt is set by process() and updateState() and not used before then 7963 mSuspended(false) 7964{ 7965 ALOGV("Constructor %p", this); 7966 int lStatus; 7967 if (thread == NULL) { 7968 return; 7969 } 7970 7971 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7972 7973 // create effect engine from effect factory 7974 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7975 7976 if (mStatus != NO_ERROR) { 7977 return; 7978 } 7979 lStatus = init(); 7980 if (lStatus < 0) { 7981 mStatus = lStatus; 7982 goto Error; 7983 } 7984 7985 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7986 return; 7987Error: 7988 EffectRelease(mEffectInterface); 7989 mEffectInterface = NULL; 7990 ALOGV("Constructor Error %d", mStatus); 7991} 7992 7993AudioFlinger::EffectModule::~EffectModule() 7994{ 7995 ALOGV("Destructor %p", this); 7996 if (mEffectInterface != NULL) { 7997 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7998 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7999 sp<ThreadBase> thread = mThread.promote(); 8000 if (thread != 0) { 8001 audio_stream_t *stream = thread->stream(); 8002 if (stream != NULL) { 8003 stream->remove_audio_effect(stream, mEffectInterface); 8004 } 8005 } 8006 } 8007 // release effect engine 8008 EffectRelease(mEffectInterface); 8009 } 8010} 8011 8012status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8013{ 8014 status_t status; 8015 8016 Mutex::Autolock _l(mLock); 8017 int priority = handle->priority(); 8018 size_t size = mHandles.size(); 8019 EffectHandle *controlHandle = NULL; 8020 size_t i; 8021 for (i = 0; i < size; i++) { 8022 EffectHandle *h = mHandles[i]; 8023 if (h == NULL || h->destroyed_l()) continue; 8024 // first non destroyed handle is considered in control 8025 if (controlHandle == NULL) 8026 controlHandle = h; 8027 if (h->priority() <= priority) break; 8028 } 8029 // if inserted in first place, move effect control from previous owner to this handle 8030 if (i == 0) { 8031 bool enabled = false; 8032 if (controlHandle != NULL) { 8033 enabled = controlHandle->enabled(); 8034 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8035 } 8036 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8037 status = NO_ERROR; 8038 } else { 8039 status = ALREADY_EXISTS; 8040 } 8041 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8042 mHandles.insertAt(handle, i); 8043 return status; 8044} 8045 8046size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8047{ 8048 Mutex::Autolock _l(mLock); 8049 size_t size = mHandles.size(); 8050 size_t i; 8051 for (i = 0; i < size; i++) { 8052 if (mHandles[i] == handle) break; 8053 } 8054 if (i == size) { 8055 return size; 8056 } 8057 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8058 8059 mHandles.removeAt(i); 8060 // if removed from first place, move effect control from this handle to next in line 8061 if (i == 0) { 8062 EffectHandle *h = controlHandle_l(); 8063 if (h != NULL) { 8064 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8065 } 8066 } 8067 8068 // Prevent calls to process() and other functions on effect interface from now on. 8069 // The effect engine will be released by the destructor when the last strong reference on 8070 // this object is released which can happen after next process is called. 8071 if (mHandles.size() == 0 && !mPinned) { 8072 mState = DESTROYED; 8073 } 8074 8075 return size; 8076} 8077 8078// must be called with EffectModule::mLock held 8079AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8080{ 8081 // the first valid handle in the list has control over the module 8082 for (size_t i = 0; i < mHandles.size(); i++) { 8083 EffectHandle *h = mHandles[i]; 8084 if (h != NULL && !h->destroyed_l()) { 8085 return h; 8086 } 8087 } 8088 8089 return NULL; 8090} 8091 8092size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8093{ 8094 ALOGV("disconnect() %p handle %p", this, handle); 8095 // keep a strong reference on this EffectModule to avoid calling the 8096 // destructor before we exit 8097 sp<EffectModule> keep(this); 8098 { 8099 sp<ThreadBase> thread = mThread.promote(); 8100 if (thread != 0) { 8101 thread->disconnectEffect(keep, handle, unpinIfLast); 8102 } 8103 } 8104 return mHandles.size(); 8105} 8106 8107void AudioFlinger::EffectModule::updateState() { 8108 Mutex::Autolock _l(mLock); 8109 8110 switch (mState) { 8111 case RESTART: 8112 reset_l(); 8113 // FALL THROUGH 8114 8115 case STARTING: 8116 // clear auxiliary effect input buffer for next accumulation 8117 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8118 memset(mConfig.inputCfg.buffer.raw, 8119 0, 8120 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8121 } 8122 start_l(); 8123 mState = ACTIVE; 8124 break; 8125 case STOPPING: 8126 stop_l(); 8127 mDisableWaitCnt = mMaxDisableWaitCnt; 8128 mState = STOPPED; 8129 break; 8130 case STOPPED: 8131 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8132 // turn off sequence. 8133 if (--mDisableWaitCnt == 0) { 8134 reset_l(); 8135 mState = IDLE; 8136 } 8137 break; 8138 default: //IDLE , ACTIVE, DESTROYED 8139 break; 8140 } 8141} 8142 8143void AudioFlinger::EffectModule::process() 8144{ 8145 Mutex::Autolock _l(mLock); 8146 8147 if (mState == DESTROYED || mEffectInterface == NULL || 8148 mConfig.inputCfg.buffer.raw == NULL || 8149 mConfig.outputCfg.buffer.raw == NULL) { 8150 return; 8151 } 8152 8153 if (isProcessEnabled()) { 8154 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8155 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8156 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8157 mConfig.inputCfg.buffer.s32, 8158 mConfig.inputCfg.buffer.frameCount/2); 8159 } 8160 8161 // do the actual processing in the effect engine 8162 int ret = (*mEffectInterface)->process(mEffectInterface, 8163 &mConfig.inputCfg.buffer, 8164 &mConfig.outputCfg.buffer); 8165 8166 // force transition to IDLE state when engine is ready 8167 if (mState == STOPPED && ret == -ENODATA) { 8168 mDisableWaitCnt = 1; 8169 } 8170 8171 // clear auxiliary effect input buffer for next accumulation 8172 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8173 memset(mConfig.inputCfg.buffer.raw, 0, 8174 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8175 } 8176 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8177 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8178 // If an insert effect is idle and input buffer is different from output buffer, 8179 // accumulate input onto output 8180 sp<EffectChain> chain = mChain.promote(); 8181 if (chain != 0 && chain->activeTrackCnt() != 0) { 8182 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8183 int16_t *in = mConfig.inputCfg.buffer.s16; 8184 int16_t *out = mConfig.outputCfg.buffer.s16; 8185 for (size_t i = 0; i < frameCnt; i++) { 8186 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8187 } 8188 } 8189 } 8190} 8191 8192void AudioFlinger::EffectModule::reset_l() 8193{ 8194 if (mEffectInterface == NULL) { 8195 return; 8196 } 8197 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8198} 8199 8200status_t AudioFlinger::EffectModule::configure() 8201{ 8202 if (mEffectInterface == NULL) { 8203 return NO_INIT; 8204 } 8205 8206 sp<ThreadBase> thread = mThread.promote(); 8207 if (thread == 0) { 8208 return DEAD_OBJECT; 8209 } 8210 8211 // TODO: handle configuration of effects replacing track process 8212 audio_channel_mask_t channelMask = thread->channelMask(); 8213 8214 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8215 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8216 } else { 8217 mConfig.inputCfg.channels = channelMask; 8218 } 8219 mConfig.outputCfg.channels = channelMask; 8220 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8221 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8222 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8223 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8224 mConfig.inputCfg.bufferProvider.cookie = NULL; 8225 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8226 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8227 mConfig.outputCfg.bufferProvider.cookie = NULL; 8228 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8229 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8230 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8231 // Insert effect: 8232 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8233 // always overwrites output buffer: input buffer == output buffer 8234 // - in other sessions: 8235 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8236 // other effect: overwrites output buffer: input buffer == output buffer 8237 // Auxiliary effect: 8238 // accumulates in output buffer: input buffer != output buffer 8239 // Therefore: accumulate <=> input buffer != output buffer 8240 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8241 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8242 } else { 8243 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8244 } 8245 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8246 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8247 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8248 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8249 8250 ALOGV("configure() %p thread %p buffer %p framecount %d", 8251 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8252 8253 status_t cmdStatus; 8254 uint32_t size = sizeof(int); 8255 status_t status = (*mEffectInterface)->command(mEffectInterface, 8256 EFFECT_CMD_SET_CONFIG, 8257 sizeof(effect_config_t), 8258 &mConfig, 8259 &size, 8260 &cmdStatus); 8261 if (status == 0) { 8262 status = cmdStatus; 8263 } 8264 8265 if (status == 0 && 8266 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8267 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8268 effect_param_t *p = (effect_param_t *)buf32; 8269 8270 p->psize = sizeof(uint32_t); 8271 p->vsize = sizeof(uint32_t); 8272 size = sizeof(int); 8273 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8274 8275 uint32_t latency = 0; 8276 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8277 if (pbt != NULL) { 8278 latency = pbt->latency_l(); 8279 } 8280 8281 *((int32_t *)p->data + 1)= latency; 8282 (*mEffectInterface)->command(mEffectInterface, 8283 EFFECT_CMD_SET_PARAM, 8284 sizeof(effect_param_t) + 8, 8285 &buf32, 8286 &size, 8287 &cmdStatus); 8288 } 8289 8290 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8291 (1000 * mConfig.outputCfg.buffer.frameCount); 8292 8293 return status; 8294} 8295 8296status_t AudioFlinger::EffectModule::init() 8297{ 8298 Mutex::Autolock _l(mLock); 8299 if (mEffectInterface == NULL) { 8300 return NO_INIT; 8301 } 8302 status_t cmdStatus; 8303 uint32_t size = sizeof(status_t); 8304 status_t status = (*mEffectInterface)->command(mEffectInterface, 8305 EFFECT_CMD_INIT, 8306 0, 8307 NULL, 8308 &size, 8309 &cmdStatus); 8310 if (status == 0) { 8311 status = cmdStatus; 8312 } 8313 return status; 8314} 8315 8316status_t AudioFlinger::EffectModule::start() 8317{ 8318 Mutex::Autolock _l(mLock); 8319 return start_l(); 8320} 8321 8322status_t AudioFlinger::EffectModule::start_l() 8323{ 8324 if (mEffectInterface == NULL) { 8325 return NO_INIT; 8326 } 8327 status_t cmdStatus; 8328 uint32_t size = sizeof(status_t); 8329 status_t status = (*mEffectInterface)->command(mEffectInterface, 8330 EFFECT_CMD_ENABLE, 8331 0, 8332 NULL, 8333 &size, 8334 &cmdStatus); 8335 if (status == 0) { 8336 status = cmdStatus; 8337 } 8338 if (status == 0 && 8339 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8340 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8341 sp<ThreadBase> thread = mThread.promote(); 8342 if (thread != 0) { 8343 audio_stream_t *stream = thread->stream(); 8344 if (stream != NULL) { 8345 stream->add_audio_effect(stream, mEffectInterface); 8346 } 8347 } 8348 } 8349 return status; 8350} 8351 8352status_t AudioFlinger::EffectModule::stop() 8353{ 8354 Mutex::Autolock _l(mLock); 8355 return stop_l(); 8356} 8357 8358status_t AudioFlinger::EffectModule::stop_l() 8359{ 8360 if (mEffectInterface == NULL) { 8361 return NO_INIT; 8362 } 8363 status_t cmdStatus; 8364 uint32_t size = sizeof(status_t); 8365 status_t status = (*mEffectInterface)->command(mEffectInterface, 8366 EFFECT_CMD_DISABLE, 8367 0, 8368 NULL, 8369 &size, 8370 &cmdStatus); 8371 if (status == 0) { 8372 status = cmdStatus; 8373 } 8374 if (status == 0 && 8375 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8376 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8377 sp<ThreadBase> thread = mThread.promote(); 8378 if (thread != 0) { 8379 audio_stream_t *stream = thread->stream(); 8380 if (stream != NULL) { 8381 stream->remove_audio_effect(stream, mEffectInterface); 8382 } 8383 } 8384 } 8385 return status; 8386} 8387 8388status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8389 uint32_t cmdSize, 8390 void *pCmdData, 8391 uint32_t *replySize, 8392 void *pReplyData) 8393{ 8394 Mutex::Autolock _l(mLock); 8395// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8396 8397 if (mState == DESTROYED || mEffectInterface == NULL) { 8398 return NO_INIT; 8399 } 8400 status_t status = (*mEffectInterface)->command(mEffectInterface, 8401 cmdCode, 8402 cmdSize, 8403 pCmdData, 8404 replySize, 8405 pReplyData); 8406 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8407 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8408 for (size_t i = 1; i < mHandles.size(); i++) { 8409 EffectHandle *h = mHandles[i]; 8410 if (h != NULL && !h->destroyed_l()) { 8411 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8412 } 8413 } 8414 } 8415 return status; 8416} 8417 8418status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8419{ 8420 Mutex::Autolock _l(mLock); 8421 return setEnabled_l(enabled); 8422} 8423 8424// must be called with EffectModule::mLock held 8425status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8426{ 8427 8428 ALOGV("setEnabled %p enabled %d", this, enabled); 8429 8430 if (enabled != isEnabled()) { 8431 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8432 if (enabled && status != NO_ERROR) { 8433 return status; 8434 } 8435 8436 switch (mState) { 8437 // going from disabled to enabled 8438 case IDLE: 8439 mState = STARTING; 8440 break; 8441 case STOPPED: 8442 mState = RESTART; 8443 break; 8444 case STOPPING: 8445 mState = ACTIVE; 8446 break; 8447 8448 // going from enabled to disabled 8449 case RESTART: 8450 mState = STOPPED; 8451 break; 8452 case STARTING: 8453 mState = IDLE; 8454 break; 8455 case ACTIVE: 8456 mState = STOPPING; 8457 break; 8458 case DESTROYED: 8459 return NO_ERROR; // simply ignore as we are being destroyed 8460 } 8461 for (size_t i = 1; i < mHandles.size(); i++) { 8462 EffectHandle *h = mHandles[i]; 8463 if (h != NULL && !h->destroyed_l()) { 8464 h->setEnabled(enabled); 8465 } 8466 } 8467 } 8468 return NO_ERROR; 8469} 8470 8471bool AudioFlinger::EffectModule::isEnabled() const 8472{ 8473 switch (mState) { 8474 case RESTART: 8475 case STARTING: 8476 case ACTIVE: 8477 return true; 8478 case IDLE: 8479 case STOPPING: 8480 case STOPPED: 8481 case DESTROYED: 8482 default: 8483 return false; 8484 } 8485} 8486 8487bool AudioFlinger::EffectModule::isProcessEnabled() const 8488{ 8489 switch (mState) { 8490 case RESTART: 8491 case ACTIVE: 8492 case STOPPING: 8493 case STOPPED: 8494 return true; 8495 case IDLE: 8496 case STARTING: 8497 case DESTROYED: 8498 default: 8499 return false; 8500 } 8501} 8502 8503status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8504{ 8505 Mutex::Autolock _l(mLock); 8506 status_t status = NO_ERROR; 8507 8508 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8509 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8510 if (isProcessEnabled() && 8511 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8512 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8513 status_t cmdStatus; 8514 uint32_t volume[2]; 8515 uint32_t *pVolume = NULL; 8516 uint32_t size = sizeof(volume); 8517 volume[0] = *left; 8518 volume[1] = *right; 8519 if (controller) { 8520 pVolume = volume; 8521 } 8522 status = (*mEffectInterface)->command(mEffectInterface, 8523 EFFECT_CMD_SET_VOLUME, 8524 size, 8525 volume, 8526 &size, 8527 pVolume); 8528 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8529 *left = volume[0]; 8530 *right = volume[1]; 8531 } 8532 } 8533 return status; 8534} 8535 8536status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8537{ 8538 Mutex::Autolock _l(mLock); 8539 status_t status = NO_ERROR; 8540 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8541 // audio pre processing modules on RecordThread can receive both output and 8542 // input device indication in the same call 8543 audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL; 8544 if (dev) { 8545 status_t cmdStatus; 8546 uint32_t size = sizeof(status_t); 8547 8548 status = (*mEffectInterface)->command(mEffectInterface, 8549 EFFECT_CMD_SET_DEVICE, 8550 sizeof(uint32_t), 8551 &dev, 8552 &size, 8553 &cmdStatus); 8554 if (status == NO_ERROR) { 8555 status = cmdStatus; 8556 } 8557 } 8558 dev = device & AUDIO_DEVICE_IN_ALL; 8559 if (dev) { 8560 status_t cmdStatus; 8561 uint32_t size = sizeof(status_t); 8562 8563 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8564 EFFECT_CMD_SET_INPUT_DEVICE, 8565 sizeof(uint32_t), 8566 &dev, 8567 &size, 8568 &cmdStatus); 8569 if (status2 == NO_ERROR) { 8570 status2 = cmdStatus; 8571 } 8572 if (status == NO_ERROR) { 8573 status = status2; 8574 } 8575 } 8576 } 8577 return status; 8578} 8579 8580status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8581{ 8582 Mutex::Autolock _l(mLock); 8583 status_t status = NO_ERROR; 8584 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8585 status_t cmdStatus; 8586 uint32_t size = sizeof(status_t); 8587 status = (*mEffectInterface)->command(mEffectInterface, 8588 EFFECT_CMD_SET_AUDIO_MODE, 8589 sizeof(audio_mode_t), 8590 &mode, 8591 &size, 8592 &cmdStatus); 8593 if (status == NO_ERROR) { 8594 status = cmdStatus; 8595 } 8596 } 8597 return status; 8598} 8599 8600void AudioFlinger::EffectModule::setSuspended(bool suspended) 8601{ 8602 Mutex::Autolock _l(mLock); 8603 mSuspended = suspended; 8604} 8605 8606bool AudioFlinger::EffectModule::suspended() const 8607{ 8608 Mutex::Autolock _l(mLock); 8609 return mSuspended; 8610} 8611 8612bool AudioFlinger::EffectModule::purgeHandles() 8613{ 8614 bool enabled = false; 8615 Mutex::Autolock _l(mLock); 8616 for (size_t i = 0; i < mHandles.size(); i++) { 8617 EffectHandle *handle = mHandles[i]; 8618 if (handle != NULL && !handle->destroyed_l()) { 8619 handle->effect().clear(); 8620 if (handle->hasControl()) { 8621 enabled = handle->enabled(); 8622 } 8623 } 8624 } 8625 return enabled; 8626} 8627 8628status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8629{ 8630 const size_t SIZE = 256; 8631 char buffer[SIZE]; 8632 String8 result; 8633 8634 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8635 result.append(buffer); 8636 8637 bool locked = tryLock(mLock); 8638 // failed to lock - AudioFlinger is probably deadlocked 8639 if (!locked) { 8640 result.append("\t\tCould not lock Fx mutex:\n"); 8641 } 8642 8643 result.append("\t\tSession Status State Engine:\n"); 8644 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8645 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8646 result.append(buffer); 8647 8648 result.append("\t\tDescriptor:\n"); 8649 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8650 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8651 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8652 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8653 result.append(buffer); 8654 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8655 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8656 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8657 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8658 result.append(buffer); 8659 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8660 mDescriptor.apiVersion, 8661 mDescriptor.flags); 8662 result.append(buffer); 8663 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8664 mDescriptor.name); 8665 result.append(buffer); 8666 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8667 mDescriptor.implementor); 8668 result.append(buffer); 8669 8670 result.append("\t\t- Input configuration:\n"); 8671 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8672 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8673 (uint32_t)mConfig.inputCfg.buffer.raw, 8674 mConfig.inputCfg.buffer.frameCount, 8675 mConfig.inputCfg.samplingRate, 8676 mConfig.inputCfg.channels, 8677 mConfig.inputCfg.format); 8678 result.append(buffer); 8679 8680 result.append("\t\t- Output configuration:\n"); 8681 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8682 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8683 (uint32_t)mConfig.outputCfg.buffer.raw, 8684 mConfig.outputCfg.buffer.frameCount, 8685 mConfig.outputCfg.samplingRate, 8686 mConfig.outputCfg.channels, 8687 mConfig.outputCfg.format); 8688 result.append(buffer); 8689 8690 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8691 result.append(buffer); 8692 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8693 for (size_t i = 0; i < mHandles.size(); ++i) { 8694 EffectHandle *handle = mHandles[i]; 8695 if (handle != NULL && !handle->destroyed_l()) { 8696 handle->dump(buffer, SIZE); 8697 result.append(buffer); 8698 } 8699 } 8700 8701 result.append("\n"); 8702 8703 write(fd, result.string(), result.length()); 8704 8705 if (locked) { 8706 mLock.unlock(); 8707 } 8708 8709 return NO_ERROR; 8710} 8711 8712// ---------------------------------------------------------------------------- 8713// EffectHandle implementation 8714// ---------------------------------------------------------------------------- 8715 8716#undef LOG_TAG 8717#define LOG_TAG "AudioFlinger::EffectHandle" 8718 8719AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8720 const sp<AudioFlinger::Client>& client, 8721 const sp<IEffectClient>& effectClient, 8722 int32_t priority) 8723 : BnEffect(), 8724 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8725 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8726{ 8727 ALOGV("constructor %p", this); 8728 8729 if (client == 0) { 8730 return; 8731 } 8732 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8733 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8734 if (mCblkMemory != 0) { 8735 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8736 8737 if (mCblk != NULL) { 8738 new(mCblk) effect_param_cblk_t(); 8739 mBuffer = (uint8_t *)mCblk + bufOffset; 8740 } 8741 } else { 8742 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8743 return; 8744 } 8745} 8746 8747AudioFlinger::EffectHandle::~EffectHandle() 8748{ 8749 ALOGV("Destructor %p", this); 8750 8751 if (mEffect == 0) { 8752 mDestroyed = true; 8753 return; 8754 } 8755 mEffect->lock(); 8756 mDestroyed = true; 8757 mEffect->unlock(); 8758 disconnect(false); 8759} 8760 8761status_t AudioFlinger::EffectHandle::enable() 8762{ 8763 ALOGV("enable %p", this); 8764 if (!mHasControl) return INVALID_OPERATION; 8765 if (mEffect == 0) return DEAD_OBJECT; 8766 8767 if (mEnabled) { 8768 return NO_ERROR; 8769 } 8770 8771 mEnabled = true; 8772 8773 sp<ThreadBase> thread = mEffect->thread().promote(); 8774 if (thread != 0) { 8775 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8776 } 8777 8778 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8779 if (mEffect->suspended()) { 8780 return NO_ERROR; 8781 } 8782 8783 status_t status = mEffect->setEnabled(true); 8784 if (status != NO_ERROR) { 8785 if (thread != 0) { 8786 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8787 } 8788 mEnabled = false; 8789 } 8790 return status; 8791} 8792 8793status_t AudioFlinger::EffectHandle::disable() 8794{ 8795 ALOGV("disable %p", this); 8796 if (!mHasControl) return INVALID_OPERATION; 8797 if (mEffect == 0) return DEAD_OBJECT; 8798 8799 if (!mEnabled) { 8800 return NO_ERROR; 8801 } 8802 mEnabled = false; 8803 8804 if (mEffect->suspended()) { 8805 return NO_ERROR; 8806 } 8807 8808 status_t status = mEffect->setEnabled(false); 8809 8810 sp<ThreadBase> thread = mEffect->thread().promote(); 8811 if (thread != 0) { 8812 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8813 } 8814 8815 return status; 8816} 8817 8818void AudioFlinger::EffectHandle::disconnect() 8819{ 8820 disconnect(true); 8821} 8822 8823void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8824{ 8825 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8826 if (mEffect == 0) { 8827 return; 8828 } 8829 // restore suspended effects if the disconnected handle was enabled and the last one. 8830 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8831 sp<ThreadBase> thread = mEffect->thread().promote(); 8832 if (thread != 0) { 8833 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8834 } 8835 } 8836 8837 // release sp on module => module destructor can be called now 8838 mEffect.clear(); 8839 if (mClient != 0) { 8840 if (mCblk != NULL) { 8841 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8842 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8843 } 8844 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8845 // Client destructor must run with AudioFlinger mutex locked 8846 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8847 mClient.clear(); 8848 } 8849} 8850 8851status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8852 uint32_t cmdSize, 8853 void *pCmdData, 8854 uint32_t *replySize, 8855 void *pReplyData) 8856{ 8857// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8858// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8859 8860 // only get parameter command is permitted for applications not controlling the effect 8861 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8862 return INVALID_OPERATION; 8863 } 8864 if (mEffect == 0) return DEAD_OBJECT; 8865 if (mClient == 0) return INVALID_OPERATION; 8866 8867 // handle commands that are not forwarded transparently to effect engine 8868 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8869 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8870 // no risk to block the whole media server process or mixer threads is we are stuck here 8871 Mutex::Autolock _l(mCblk->lock); 8872 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8873 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8874 mCblk->serverIndex = 0; 8875 mCblk->clientIndex = 0; 8876 return BAD_VALUE; 8877 } 8878 status_t status = NO_ERROR; 8879 while (mCblk->serverIndex < mCblk->clientIndex) { 8880 int reply; 8881 uint32_t rsize = sizeof(int); 8882 int *p = (int *)(mBuffer + mCblk->serverIndex); 8883 int size = *p++; 8884 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8885 ALOGW("command(): invalid parameter block size"); 8886 break; 8887 } 8888 effect_param_t *param = (effect_param_t *)p; 8889 if (param->psize == 0 || param->vsize == 0) { 8890 ALOGW("command(): null parameter or value size"); 8891 mCblk->serverIndex += size; 8892 continue; 8893 } 8894 uint32_t psize = sizeof(effect_param_t) + 8895 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8896 param->vsize; 8897 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8898 psize, 8899 p, 8900 &rsize, 8901 &reply); 8902 // stop at first error encountered 8903 if (ret != NO_ERROR) { 8904 status = ret; 8905 *(int *)pReplyData = reply; 8906 break; 8907 } else if (reply != NO_ERROR) { 8908 *(int *)pReplyData = reply; 8909 break; 8910 } 8911 mCblk->serverIndex += size; 8912 } 8913 mCblk->serverIndex = 0; 8914 mCblk->clientIndex = 0; 8915 return status; 8916 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8917 *(int *)pReplyData = NO_ERROR; 8918 return enable(); 8919 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8920 *(int *)pReplyData = NO_ERROR; 8921 return disable(); 8922 } 8923 8924 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8925} 8926 8927void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8928{ 8929 ALOGV("setControl %p control %d", this, hasControl); 8930 8931 mHasControl = hasControl; 8932 mEnabled = enabled; 8933 8934 if (signal && mEffectClient != 0) { 8935 mEffectClient->controlStatusChanged(hasControl); 8936 } 8937} 8938 8939void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8940 uint32_t cmdSize, 8941 void *pCmdData, 8942 uint32_t replySize, 8943 void *pReplyData) 8944{ 8945 if (mEffectClient != 0) { 8946 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8947 } 8948} 8949 8950 8951 8952void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8953{ 8954 if (mEffectClient != 0) { 8955 mEffectClient->enableStatusChanged(enabled); 8956 } 8957} 8958 8959status_t AudioFlinger::EffectHandle::onTransact( 8960 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8961{ 8962 return BnEffect::onTransact(code, data, reply, flags); 8963} 8964 8965 8966void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8967{ 8968 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8969 8970 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8971 (mClient == 0) ? getpid_cached : mClient->pid(), 8972 mPriority, 8973 mHasControl, 8974 !locked, 8975 mCblk ? mCblk->clientIndex : 0, 8976 mCblk ? mCblk->serverIndex : 0 8977 ); 8978 8979 if (locked) { 8980 mCblk->lock.unlock(); 8981 } 8982} 8983 8984#undef LOG_TAG 8985#define LOG_TAG "AudioFlinger::EffectChain" 8986 8987AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8988 int sessionId) 8989 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8990 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8991 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8992{ 8993 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8994 if (thread == NULL) { 8995 return; 8996 } 8997 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8998 thread->frameCount(); 8999} 9000 9001AudioFlinger::EffectChain::~EffectChain() 9002{ 9003 if (mOwnInBuffer) { 9004 delete mInBuffer; 9005 } 9006 9007} 9008 9009// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9010sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9011{ 9012 size_t size = mEffects.size(); 9013 9014 for (size_t i = 0; i < size; i++) { 9015 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9016 return mEffects[i]; 9017 } 9018 } 9019 return 0; 9020} 9021 9022// getEffectFromId_l() must be called with ThreadBase::mLock held 9023sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9024{ 9025 size_t size = mEffects.size(); 9026 9027 for (size_t i = 0; i < size; i++) { 9028 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9029 if (id == 0 || mEffects[i]->id() == id) { 9030 return mEffects[i]; 9031 } 9032 } 9033 return 0; 9034} 9035 9036// getEffectFromType_l() must be called with ThreadBase::mLock held 9037sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9038 const effect_uuid_t *type) 9039{ 9040 size_t size = mEffects.size(); 9041 9042 for (size_t i = 0; i < size; i++) { 9043 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9044 return mEffects[i]; 9045 } 9046 } 9047 return 0; 9048} 9049 9050void AudioFlinger::EffectChain::clearInputBuffer() 9051{ 9052 Mutex::Autolock _l(mLock); 9053 sp<ThreadBase> thread = mThread.promote(); 9054 if (thread == 0) { 9055 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9056 return; 9057 } 9058 clearInputBuffer_l(thread); 9059} 9060 9061// Must be called with EffectChain::mLock locked 9062void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9063{ 9064 size_t numSamples = thread->frameCount() * thread->channelCount(); 9065 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9066 9067} 9068 9069// Must be called with EffectChain::mLock locked 9070void AudioFlinger::EffectChain::process_l() 9071{ 9072 sp<ThreadBase> thread = mThread.promote(); 9073 if (thread == 0) { 9074 ALOGW("process_l(): cannot promote mixer thread"); 9075 return; 9076 } 9077 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9078 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9079 // always process effects unless no more tracks are on the session and the effect tail 9080 // has been rendered 9081 bool doProcess = true; 9082 if (!isGlobalSession) { 9083 bool tracksOnSession = (trackCnt() != 0); 9084 9085 if (!tracksOnSession && mTailBufferCount == 0) { 9086 doProcess = false; 9087 } 9088 9089 if (activeTrackCnt() == 0) { 9090 // if no track is active and the effect tail has not been rendered, 9091 // the input buffer must be cleared here as the mixer process will not do it 9092 if (tracksOnSession || mTailBufferCount > 0) { 9093 clearInputBuffer_l(thread); 9094 if (mTailBufferCount > 0) { 9095 mTailBufferCount--; 9096 } 9097 } 9098 } 9099 } 9100 9101 size_t size = mEffects.size(); 9102 if (doProcess) { 9103 for (size_t i = 0; i < size; i++) { 9104 mEffects[i]->process(); 9105 } 9106 } 9107 for (size_t i = 0; i < size; i++) { 9108 mEffects[i]->updateState(); 9109 } 9110} 9111 9112// addEffect_l() must be called with PlaybackThread::mLock held 9113status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9114{ 9115 effect_descriptor_t desc = effect->desc(); 9116 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9117 9118 Mutex::Autolock _l(mLock); 9119 effect->setChain(this); 9120 sp<ThreadBase> thread = mThread.promote(); 9121 if (thread == 0) { 9122 return NO_INIT; 9123 } 9124 effect->setThread(thread); 9125 9126 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9127 // Auxiliary effects are inserted at the beginning of mEffects vector as 9128 // they are processed first and accumulated in chain input buffer 9129 mEffects.insertAt(effect, 0); 9130 9131 // the input buffer for auxiliary effect contains mono samples in 9132 // 32 bit format. This is to avoid saturation in AudoMixer 9133 // accumulation stage. Saturation is done in EffectModule::process() before 9134 // calling the process in effect engine 9135 size_t numSamples = thread->frameCount(); 9136 int32_t *buffer = new int32_t[numSamples]; 9137 memset(buffer, 0, numSamples * sizeof(int32_t)); 9138 effect->setInBuffer((int16_t *)buffer); 9139 // auxiliary effects output samples to chain input buffer for further processing 9140 // by insert effects 9141 effect->setOutBuffer(mInBuffer); 9142 } else { 9143 // Insert effects are inserted at the end of mEffects vector as they are processed 9144 // after track and auxiliary effects. 9145 // Insert effect order as a function of indicated preference: 9146 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9147 // another effect is present 9148 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9149 // last effect claiming first position 9150 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9151 // first effect claiming last position 9152 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9153 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9154 // already present 9155 9156 size_t size = mEffects.size(); 9157 size_t idx_insert = size; 9158 ssize_t idx_insert_first = -1; 9159 ssize_t idx_insert_last = -1; 9160 9161 for (size_t i = 0; i < size; i++) { 9162 effect_descriptor_t d = mEffects[i]->desc(); 9163 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9164 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9165 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9166 // check invalid effect chaining combinations 9167 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9168 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9169 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9170 return INVALID_OPERATION; 9171 } 9172 // remember position of first insert effect and by default 9173 // select this as insert position for new effect 9174 if (idx_insert == size) { 9175 idx_insert = i; 9176 } 9177 // remember position of last insert effect claiming 9178 // first position 9179 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9180 idx_insert_first = i; 9181 } 9182 // remember position of first insert effect claiming 9183 // last position 9184 if (iPref == EFFECT_FLAG_INSERT_LAST && 9185 idx_insert_last == -1) { 9186 idx_insert_last = i; 9187 } 9188 } 9189 } 9190 9191 // modify idx_insert from first position if needed 9192 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9193 if (idx_insert_last != -1) { 9194 idx_insert = idx_insert_last; 9195 } else { 9196 idx_insert = size; 9197 } 9198 } else { 9199 if (idx_insert_first != -1) { 9200 idx_insert = idx_insert_first + 1; 9201 } 9202 } 9203 9204 // always read samples from chain input buffer 9205 effect->setInBuffer(mInBuffer); 9206 9207 // if last effect in the chain, output samples to chain 9208 // output buffer, otherwise to chain input buffer 9209 if (idx_insert == size) { 9210 if (idx_insert != 0) { 9211 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9212 mEffects[idx_insert-1]->configure(); 9213 } 9214 effect->setOutBuffer(mOutBuffer); 9215 } else { 9216 effect->setOutBuffer(mInBuffer); 9217 } 9218 mEffects.insertAt(effect, idx_insert); 9219 9220 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9221 } 9222 effect->configure(); 9223 return NO_ERROR; 9224} 9225 9226// removeEffect_l() must be called with PlaybackThread::mLock held 9227size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9228{ 9229 Mutex::Autolock _l(mLock); 9230 size_t size = mEffects.size(); 9231 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9232 9233 for (size_t i = 0; i < size; i++) { 9234 if (effect == mEffects[i]) { 9235 // calling stop here will remove pre-processing effect from the audio HAL. 9236 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9237 // the middle of a read from audio HAL 9238 if (mEffects[i]->state() == EffectModule::ACTIVE || 9239 mEffects[i]->state() == EffectModule::STOPPING) { 9240 mEffects[i]->stop(); 9241 } 9242 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9243 delete[] effect->inBuffer(); 9244 } else { 9245 if (i == size - 1 && i != 0) { 9246 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9247 mEffects[i - 1]->configure(); 9248 } 9249 } 9250 mEffects.removeAt(i); 9251 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9252 break; 9253 } 9254 } 9255 9256 return mEffects.size(); 9257} 9258 9259// setDevice_l() must be called with PlaybackThread::mLock held 9260void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9261{ 9262 size_t size = mEffects.size(); 9263 for (size_t i = 0; i < size; i++) { 9264 mEffects[i]->setDevice(device); 9265 } 9266} 9267 9268// setMode_l() must be called with PlaybackThread::mLock held 9269void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9270{ 9271 size_t size = mEffects.size(); 9272 for (size_t i = 0; i < size; i++) { 9273 mEffects[i]->setMode(mode); 9274 } 9275} 9276 9277// setVolume_l() must be called with PlaybackThread::mLock held 9278bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9279{ 9280 uint32_t newLeft = *left; 9281 uint32_t newRight = *right; 9282 bool hasControl = false; 9283 int ctrlIdx = -1; 9284 size_t size = mEffects.size(); 9285 9286 // first update volume controller 9287 for (size_t i = size; i > 0; i--) { 9288 if (mEffects[i - 1]->isProcessEnabled() && 9289 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9290 ctrlIdx = i - 1; 9291 hasControl = true; 9292 break; 9293 } 9294 } 9295 9296 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9297 if (hasControl) { 9298 *left = mNewLeftVolume; 9299 *right = mNewRightVolume; 9300 } 9301 return hasControl; 9302 } 9303 9304 mVolumeCtrlIdx = ctrlIdx; 9305 mLeftVolume = newLeft; 9306 mRightVolume = newRight; 9307 9308 // second get volume update from volume controller 9309 if (ctrlIdx >= 0) { 9310 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9311 mNewLeftVolume = newLeft; 9312 mNewRightVolume = newRight; 9313 } 9314 // then indicate volume to all other effects in chain. 9315 // Pass altered volume to effects before volume controller 9316 // and requested volume to effects after controller 9317 uint32_t lVol = newLeft; 9318 uint32_t rVol = newRight; 9319 9320 for (size_t i = 0; i < size; i++) { 9321 if ((int)i == ctrlIdx) continue; 9322 // this also works for ctrlIdx == -1 when there is no volume controller 9323 if ((int)i > ctrlIdx) { 9324 lVol = *left; 9325 rVol = *right; 9326 } 9327 mEffects[i]->setVolume(&lVol, &rVol, false); 9328 } 9329 *left = newLeft; 9330 *right = newRight; 9331 9332 return hasControl; 9333} 9334 9335status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9336{ 9337 const size_t SIZE = 256; 9338 char buffer[SIZE]; 9339 String8 result; 9340 9341 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9342 result.append(buffer); 9343 9344 bool locked = tryLock(mLock); 9345 // failed to lock - AudioFlinger is probably deadlocked 9346 if (!locked) { 9347 result.append("\tCould not lock mutex:\n"); 9348 } 9349 9350 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9351 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9352 mEffects.size(), 9353 (uint32_t)mInBuffer, 9354 (uint32_t)mOutBuffer, 9355 mActiveTrackCnt); 9356 result.append(buffer); 9357 write(fd, result.string(), result.size()); 9358 9359 for (size_t i = 0; i < mEffects.size(); ++i) { 9360 sp<EffectModule> effect = mEffects[i]; 9361 if (effect != 0) { 9362 effect->dump(fd, args); 9363 } 9364 } 9365 9366 if (locked) { 9367 mLock.unlock(); 9368 } 9369 9370 return NO_ERROR; 9371} 9372 9373// must be called with ThreadBase::mLock held 9374void AudioFlinger::EffectChain::setEffectSuspended_l( 9375 const effect_uuid_t *type, bool suspend) 9376{ 9377 sp<SuspendedEffectDesc> desc; 9378 // use effect type UUID timelow as key as there is no real risk of identical 9379 // timeLow fields among effect type UUIDs. 9380 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9381 if (suspend) { 9382 if (index >= 0) { 9383 desc = mSuspendedEffects.valueAt(index); 9384 } else { 9385 desc = new SuspendedEffectDesc(); 9386 desc->mType = *type; 9387 mSuspendedEffects.add(type->timeLow, desc); 9388 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9389 } 9390 if (desc->mRefCount++ == 0) { 9391 sp<EffectModule> effect = getEffectIfEnabled(type); 9392 if (effect != 0) { 9393 desc->mEffect = effect; 9394 effect->setSuspended(true); 9395 effect->setEnabled(false); 9396 } 9397 } 9398 } else { 9399 if (index < 0) { 9400 return; 9401 } 9402 desc = mSuspendedEffects.valueAt(index); 9403 if (desc->mRefCount <= 0) { 9404 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9405 desc->mRefCount = 1; 9406 } 9407 if (--desc->mRefCount == 0) { 9408 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9409 if (desc->mEffect != 0) { 9410 sp<EffectModule> effect = desc->mEffect.promote(); 9411 if (effect != 0) { 9412 effect->setSuspended(false); 9413 effect->lock(); 9414 EffectHandle *handle = effect->controlHandle_l(); 9415 if (handle != NULL && !handle->destroyed_l()) { 9416 effect->setEnabled_l(handle->enabled()); 9417 } 9418 effect->unlock(); 9419 } 9420 desc->mEffect.clear(); 9421 } 9422 mSuspendedEffects.removeItemsAt(index); 9423 } 9424 } 9425} 9426 9427// must be called with ThreadBase::mLock held 9428void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9429{ 9430 sp<SuspendedEffectDesc> desc; 9431 9432 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9433 if (suspend) { 9434 if (index >= 0) { 9435 desc = mSuspendedEffects.valueAt(index); 9436 } else { 9437 desc = new SuspendedEffectDesc(); 9438 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9439 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9440 } 9441 if (desc->mRefCount++ == 0) { 9442 Vector< sp<EffectModule> > effects; 9443 getSuspendEligibleEffects(effects); 9444 for (size_t i = 0; i < effects.size(); i++) { 9445 setEffectSuspended_l(&effects[i]->desc().type, true); 9446 } 9447 } 9448 } else { 9449 if (index < 0) { 9450 return; 9451 } 9452 desc = mSuspendedEffects.valueAt(index); 9453 if (desc->mRefCount <= 0) { 9454 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9455 desc->mRefCount = 1; 9456 } 9457 if (--desc->mRefCount == 0) { 9458 Vector<const effect_uuid_t *> types; 9459 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9460 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9461 continue; 9462 } 9463 types.add(&mSuspendedEffects.valueAt(i)->mType); 9464 } 9465 for (size_t i = 0; i < types.size(); i++) { 9466 setEffectSuspended_l(types[i], false); 9467 } 9468 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9469 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9470 } 9471 } 9472} 9473 9474 9475// The volume effect is used for automated tests only 9476#ifndef OPENSL_ES_H_ 9477static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9478 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9479const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9480#endif //OPENSL_ES_H_ 9481 9482bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9483{ 9484 // auxiliary effects and visualizer are never suspended on output mix 9485 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9486 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9487 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9488 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9489 return false; 9490 } 9491 return true; 9492} 9493 9494void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9495{ 9496 effects.clear(); 9497 for (size_t i = 0; i < mEffects.size(); i++) { 9498 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9499 effects.add(mEffects[i]); 9500 } 9501 } 9502} 9503 9504sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9505 const effect_uuid_t *type) 9506{ 9507 sp<EffectModule> effect = getEffectFromType_l(type); 9508 return effect != 0 && effect->isEnabled() ? effect : 0; 9509} 9510 9511void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9512 bool enabled) 9513{ 9514 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9515 if (enabled) { 9516 if (index < 0) { 9517 // if the effect is not suspend check if all effects are suspended 9518 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9519 if (index < 0) { 9520 return; 9521 } 9522 if (!isEffectEligibleForSuspend(effect->desc())) { 9523 return; 9524 } 9525 setEffectSuspended_l(&effect->desc().type, enabled); 9526 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9527 if (index < 0) { 9528 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9529 return; 9530 } 9531 } 9532 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9533 effect->desc().type.timeLow); 9534 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9535 // if effect is requested to suspended but was not yet enabled, supend it now. 9536 if (desc->mEffect == 0) { 9537 desc->mEffect = effect; 9538 effect->setEnabled(false); 9539 effect->setSuspended(true); 9540 } 9541 } else { 9542 if (index < 0) { 9543 return; 9544 } 9545 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9546 effect->desc().type.timeLow); 9547 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9548 desc->mEffect.clear(); 9549 effect->setSuspended(false); 9550 } 9551} 9552 9553#undef LOG_TAG 9554#define LOG_TAG "AudioFlinger" 9555 9556// ---------------------------------------------------------------------------- 9557 9558status_t AudioFlinger::onTransact( 9559 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9560{ 9561 return BnAudioFlinger::onTransact(code, data, reply, flags); 9562} 9563 9564}; // namespace android 9565