AudioFlinger.cpp revision 0ec23ce0d1ff79566c402bc30df3074f6e25a22b
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 302void 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} 326 327 328void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 329{ 330 const size_t SIZE = 256; 331 char buffer[SIZE]; 332 String8 result; 333 hardware_call_state hardwareStatus = mHardwareStatus; 334 335 snprintf(buffer, SIZE, "Hardware status: %d\n" 336 "Standby Time mSec: %u\n", 337 hardwareStatus, 338 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 339 result.append(buffer); 340 write(fd, result.string(), result.size()); 341} 342 343void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 344{ 345 const size_t SIZE = 256; 346 char buffer[SIZE]; 347 String8 result; 348 snprintf(buffer, SIZE, "Permission Denial: " 349 "can't dump AudioFlinger from pid=%d, uid=%d\n", 350 IPCThreadState::self()->getCallingPid(), 351 IPCThreadState::self()->getCallingUid()); 352 result.append(buffer); 353 write(fd, result.string(), result.size()); 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 audio_channel_mask_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 // check if an effect chain with the same session ID is present on another 473 // output thread and move it here. 474 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 475 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 476 if (mPlaybackThreads.keyAt(i) != output) { 477 uint32_t sessions = t->hasAudioSession(*sessionId); 478 if (sessions & PlaybackThread::EFFECT_SESSION) { 479 effectThread = t.get(); 480 break; 481 } 482 } 483 } 484 lSessionId = *sessionId; 485 } else { 486 // if no audio session id is provided, create one here 487 lSessionId = nextUniqueId(); 488 if (sessionId != NULL) { 489 *sessionId = lSessionId; 490 } 491 } 492 ALOGV("createTrack() lSessionId: %d", lSessionId); 493 494 track = thread->createTrack_l(client, streamType, sampleRate, format, 495 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 496 497 // move effect chain to this output thread if an effect on same session was waiting 498 // for a track to be created 499 if (lStatus == NO_ERROR && effectThread != NULL) { 500 Mutex::Autolock _dl(thread->mLock); 501 Mutex::Autolock _sl(effectThread->mLock); 502 moveEffectChain_l(lSessionId, effectThread, thread, true); 503 } 504 505 // Look for sync events awaiting for a session to be used. 506 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 507 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 508 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 509 if (lStatus == NO_ERROR) { 510 track->setSyncEvent(mPendingSyncEvents[i]); 511 } else { 512 mPendingSyncEvents[i]->cancel(); 513 } 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 RecordThread::RecordTrack *track = thread->track(); 872 if (track != NULL) { 873 audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL; 874 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 875 thread->setEffectSuspended(FX_IID_AEC, 876 suspend, 877 track->sessionId()); 878 thread->setEffectSuspended(FX_IID_NS, 879 suspend, 880 track->sessionId()); 881 } 882 } 883 mBtNrecIsOff = btNrecIsOff; 884 } 885 } 886 String8 screenState; 887 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 888 bool isOff = screenState == "off"; 889 if (isOff != (gScreenState & 1)) { 890 gScreenState = ((gScreenState & ~1) + 2) | isOff; 891 } 892 } 893 return final_result; 894 } 895 896 // hold a strong ref on thread in case closeOutput() or closeInput() is called 897 // and the thread is exited once the lock is released 898 sp<ThreadBase> thread; 899 { 900 Mutex::Autolock _l(mLock); 901 thread = checkPlaybackThread_l(ioHandle); 902 if (thread == 0) { 903 thread = checkRecordThread_l(ioHandle); 904 } else if (thread == primaryPlaybackThread_l()) { 905 // indicate output device change to all input threads for pre processing 906 AudioParameter param = AudioParameter(keyValuePairs); 907 int value; 908 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 909 (value != 0)) { 910 for (size_t i = 0; i < mRecordThreads.size(); i++) { 911 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 912 } 913 } 914 } 915 } 916 if (thread != 0) { 917 return thread->setParameters(keyValuePairs); 918 } 919 return BAD_VALUE; 920} 921 922String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 923{ 924// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 925// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 926 927 Mutex::Autolock _l(mLock); 928 929 if (ioHandle == 0) { 930 String8 out_s8; 931 932 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 933 char *s; 934 { 935 AutoMutex lock(mHardwareLock); 936 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 937 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 938 s = dev->get_parameters(dev, keys.string()); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 } 941 out_s8 += String8(s ? s : ""); 942 free(s); 943 } 944 return out_s8; 945 } 946 947 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 948 if (playbackThread != NULL) { 949 return playbackThread->getParameters(keys); 950 } 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getParameters(keys); 954 } 955 return String8(""); 956} 957 958size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 959 audio_channel_mask_t channelMask) const 960{ 961 status_t ret = initCheck(); 962 if (ret != NO_ERROR) { 963 return 0; 964 } 965 966 AutoMutex lock(mHardwareLock); 967 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 968 struct audio_config config = { 969 sample_rate: sampleRate, 970 channel_mask: channelMask, 971 format: format, 972 }; 973 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 974 mHardwareStatus = AUDIO_HW_IDLE; 975 return size; 976} 977 978unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 979{ 980 Mutex::Autolock _l(mLock); 981 982 RecordThread *recordThread = checkRecordThread_l(ioHandle); 983 if (recordThread != NULL) { 984 return recordThread->getInputFramesLost(); 985 } 986 return 0; 987} 988 989status_t AudioFlinger::setVoiceVolume(float value) 990{ 991 status_t ret = initCheck(); 992 if (ret != NO_ERROR) { 993 return ret; 994 } 995 996 // check calling permissions 997 if (!settingsAllowed()) { 998 return PERMISSION_DENIED; 999 } 1000 1001 AutoMutex lock(mHardwareLock); 1002 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1003 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1004 mHardwareStatus = AUDIO_HW_IDLE; 1005 1006 return ret; 1007} 1008 1009status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1010 audio_io_handle_t output) const 1011{ 1012 status_t status; 1013 1014 Mutex::Autolock _l(mLock); 1015 1016 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1017 if (playbackThread != NULL) { 1018 return playbackThread->getRenderPosition(halFrames, dspFrames); 1019 } 1020 1021 return BAD_VALUE; 1022} 1023 1024void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1025{ 1026 1027 Mutex::Autolock _l(mLock); 1028 1029 pid_t pid = IPCThreadState::self()->getCallingPid(); 1030 if (mNotificationClients.indexOfKey(pid) < 0) { 1031 sp<NotificationClient> notificationClient = new NotificationClient(this, 1032 client, 1033 pid); 1034 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1035 1036 mNotificationClients.add(pid, notificationClient); 1037 1038 sp<IBinder> binder = client->asBinder(); 1039 binder->linkToDeath(notificationClient); 1040 1041 // the config change is always sent from playback or record threads to avoid deadlock 1042 // with AudioSystem::gLock 1043 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1044 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1045 } 1046 1047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1048 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1049 } 1050 } 1051} 1052 1053void AudioFlinger::removeNotificationClient(pid_t pid) 1054{ 1055 Mutex::Autolock _l(mLock); 1056 1057 mNotificationClients.removeItem(pid); 1058 1059 ALOGV("%d died, releasing its sessions", pid); 1060 size_t num = mAudioSessionRefs.size(); 1061 bool removed = false; 1062 for (size_t i = 0; i< num; ) { 1063 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1064 ALOGV(" pid %d @ %d", ref->mPid, i); 1065 if (ref->mPid == pid) { 1066 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1067 mAudioSessionRefs.removeAt(i); 1068 delete ref; 1069 removed = true; 1070 num--; 1071 } else { 1072 i++; 1073 } 1074 } 1075 if (removed) { 1076 purgeStaleEffects_l(); 1077 } 1078} 1079 1080// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1081void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1082{ 1083 size_t size = mNotificationClients.size(); 1084 for (size_t i = 0; i < size; i++) { 1085 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1086 param2); 1087 } 1088} 1089 1090// removeClient_l() must be called with AudioFlinger::mLock held 1091void AudioFlinger::removeClient_l(pid_t pid) 1092{ 1093 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1094 mClients.removeItem(pid); 1095} 1096 1097// getEffectThread_l() must be called with AudioFlinger::mLock held 1098sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1099{ 1100 sp<PlaybackThread> thread; 1101 1102 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1103 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1104 ALOG_ASSERT(thread == 0); 1105 thread = mPlaybackThreads.valueAt(i); 1106 } 1107 } 1108 1109 return thread; 1110} 1111 1112// ---------------------------------------------------------------------------- 1113 1114AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1115 audio_devices_t device, type_t type) 1116 : Thread(false), 1117 mType(type), 1118 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1119 // mChannelMask 1120 mChannelCount(0), 1121 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1122 mParamStatus(NO_ERROR), 1123 mStandby(false), mDevice(device), mId(id), 1124 mDeathRecipient(new PMDeathRecipient(this)) 1125{ 1126} 1127 1128AudioFlinger::ThreadBase::~ThreadBase() 1129{ 1130 mParamCond.broadcast(); 1131 // do not lock the mutex in destructor 1132 releaseWakeLock_l(); 1133 if (mPowerManager != 0) { 1134 sp<IBinder> binder = mPowerManager->asBinder(); 1135 binder->unlinkToDeath(mDeathRecipient); 1136 } 1137} 1138 1139void AudioFlinger::ThreadBase::exit() 1140{ 1141 ALOGV("ThreadBase::exit"); 1142 { 1143 // This lock prevents the following race in thread (uniprocessor for illustration): 1144 // if (!exitPending()) { 1145 // // context switch from here to exit() 1146 // // exit() calls requestExit(), what exitPending() observes 1147 // // exit() calls signal(), which is dropped since no waiters 1148 // // context switch back from exit() to here 1149 // mWaitWorkCV.wait(...); 1150 // // now thread is hung 1151 // } 1152 AutoMutex lock(mLock); 1153 requestExit(); 1154 mWaitWorkCV.signal(); 1155 } 1156 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1157 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1158 requestExitAndWait(); 1159} 1160 1161status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1162{ 1163 status_t status; 1164 1165 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1166 Mutex::Autolock _l(mLock); 1167 1168 mNewParameters.add(keyValuePairs); 1169 mWaitWorkCV.signal(); 1170 // wait condition with timeout in case the thread loop has exited 1171 // before the request could be processed 1172 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1173 status = mParamStatus; 1174 mWaitWorkCV.signal(); 1175 } else { 1176 status = TIMED_OUT; 1177 } 1178 return status; 1179} 1180 1181void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1182{ 1183 Mutex::Autolock _l(mLock); 1184 sendConfigEvent_l(event, param); 1185} 1186 1187// sendConfigEvent_l() must be called with ThreadBase::mLock held 1188void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1189{ 1190 ConfigEvent configEvent; 1191 configEvent.mEvent = event; 1192 configEvent.mParam = param; 1193 mConfigEvents.add(configEvent); 1194 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1195 mWaitWorkCV.signal(); 1196} 1197 1198void AudioFlinger::ThreadBase::processConfigEvents() 1199{ 1200 mLock.lock(); 1201 while (!mConfigEvents.isEmpty()) { 1202 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1203 ConfigEvent configEvent = mConfigEvents[0]; 1204 mConfigEvents.removeAt(0); 1205 // release mLock before locking AudioFlinger mLock: lock order is always 1206 // AudioFlinger then ThreadBase to avoid cross deadlock 1207 mLock.unlock(); 1208 mAudioFlinger->mLock.lock(); 1209 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1210 mAudioFlinger->mLock.unlock(); 1211 mLock.lock(); 1212 } 1213 mLock.unlock(); 1214} 1215 1216void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1217{ 1218 const size_t SIZE = 256; 1219 char buffer[SIZE]; 1220 String8 result; 1221 1222 bool locked = tryLock(mLock); 1223 if (!locked) { 1224 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1225 write(fd, buffer, strlen(buffer)); 1226 } 1227 1228 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1229 result.append(buffer); 1230 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1231 result.append(buffer); 1232 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1233 result.append(buffer); 1234 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1235 result.append(buffer); 1236 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1237 result.append(buffer); 1238 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1239 result.append(buffer); 1240 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1245 result.append(buffer); 1246 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1247 result.append(buffer); 1248 1249 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1250 result.append(buffer); 1251 result.append(" Index Command"); 1252 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1253 snprintf(buffer, SIZE, "\n %02d ", i); 1254 result.append(buffer); 1255 result.append(mNewParameters[i]); 1256 } 1257 1258 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1259 result.append(buffer); 1260 snprintf(buffer, SIZE, " Index event param\n"); 1261 result.append(buffer); 1262 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1263 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1264 result.append(buffer); 1265 } 1266 result.append("\n"); 1267 1268 write(fd, result.string(), result.size()); 1269 1270 if (locked) { 1271 mLock.unlock(); 1272 } 1273} 1274 1275void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1276{ 1277 const size_t SIZE = 256; 1278 char buffer[SIZE]; 1279 String8 result; 1280 1281 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1282 write(fd, buffer, strlen(buffer)); 1283 1284 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1285 sp<EffectChain> chain = mEffectChains[i]; 1286 if (chain != 0) { 1287 chain->dump(fd, args); 1288 } 1289 } 1290} 1291 1292void AudioFlinger::ThreadBase::acquireWakeLock() 1293{ 1294 Mutex::Autolock _l(mLock); 1295 acquireWakeLock_l(); 1296} 1297 1298void AudioFlinger::ThreadBase::acquireWakeLock_l() 1299{ 1300 if (mPowerManager == 0) { 1301 // use checkService() to avoid blocking if power service is not up yet 1302 sp<IBinder> binder = 1303 defaultServiceManager()->checkService(String16("power")); 1304 if (binder == 0) { 1305 ALOGW("Thread %s cannot connect to the power manager service", mName); 1306 } else { 1307 mPowerManager = interface_cast<IPowerManager>(binder); 1308 binder->linkToDeath(mDeathRecipient); 1309 } 1310 } 1311 if (mPowerManager != 0) { 1312 sp<IBinder> binder = new BBinder(); 1313 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1314 binder, 1315 String16(mName)); 1316 if (status == NO_ERROR) { 1317 mWakeLockToken = binder; 1318 } 1319 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1320 } 1321} 1322 1323void AudioFlinger::ThreadBase::releaseWakeLock() 1324{ 1325 Mutex::Autolock _l(mLock); 1326 releaseWakeLock_l(); 1327} 1328 1329void AudioFlinger::ThreadBase::releaseWakeLock_l() 1330{ 1331 if (mWakeLockToken != 0) { 1332 ALOGV("releaseWakeLock_l() %s", mName); 1333 if (mPowerManager != 0) { 1334 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1335 } 1336 mWakeLockToken.clear(); 1337 } 1338} 1339 1340void AudioFlinger::ThreadBase::clearPowerManager() 1341{ 1342 Mutex::Autolock _l(mLock); 1343 releaseWakeLock_l(); 1344 mPowerManager.clear(); 1345} 1346 1347void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1348{ 1349 sp<ThreadBase> thread = mThread.promote(); 1350 if (thread != 0) { 1351 thread->clearPowerManager(); 1352 } 1353 ALOGW("power manager service died !!!"); 1354} 1355 1356void AudioFlinger::ThreadBase::setEffectSuspended( 1357 const effect_uuid_t *type, bool suspend, int sessionId) 1358{ 1359 Mutex::Autolock _l(mLock); 1360 setEffectSuspended_l(type, suspend, sessionId); 1361} 1362 1363void AudioFlinger::ThreadBase::setEffectSuspended_l( 1364 const effect_uuid_t *type, bool suspend, int sessionId) 1365{ 1366 sp<EffectChain> chain = getEffectChain_l(sessionId); 1367 if (chain != 0) { 1368 if (type != NULL) { 1369 chain->setEffectSuspended_l(type, suspend); 1370 } else { 1371 chain->setEffectSuspendedAll_l(suspend); 1372 } 1373 } 1374 1375 updateSuspendedSessions_l(type, suspend, sessionId); 1376} 1377 1378void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1379{ 1380 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1381 if (index < 0) { 1382 return; 1383 } 1384 1385 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1386 mSuspendedSessions.valueAt(index); 1387 1388 for (size_t i = 0; i < sessionEffects.size(); i++) { 1389 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1390 for (int j = 0; j < desc->mRefCount; j++) { 1391 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1392 chain->setEffectSuspendedAll_l(true); 1393 } else { 1394 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1395 desc->mType.timeLow); 1396 chain->setEffectSuspended_l(&desc->mType, true); 1397 } 1398 } 1399 } 1400} 1401 1402void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1403 bool suspend, 1404 int sessionId) 1405{ 1406 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1407 1408 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1409 1410 if (suspend) { 1411 if (index >= 0) { 1412 sessionEffects = mSuspendedSessions.valueAt(index); 1413 } else { 1414 mSuspendedSessions.add(sessionId, sessionEffects); 1415 } 1416 } else { 1417 if (index < 0) { 1418 return; 1419 } 1420 sessionEffects = mSuspendedSessions.valueAt(index); 1421 } 1422 1423 1424 int key = EffectChain::kKeyForSuspendAll; 1425 if (type != NULL) { 1426 key = type->timeLow; 1427 } 1428 index = sessionEffects.indexOfKey(key); 1429 1430 sp<SuspendedSessionDesc> desc; 1431 if (suspend) { 1432 if (index >= 0) { 1433 desc = sessionEffects.valueAt(index); 1434 } else { 1435 desc = new SuspendedSessionDesc(); 1436 if (type != NULL) { 1437 desc->mType = *type; 1438 } 1439 sessionEffects.add(key, desc); 1440 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1441 } 1442 desc->mRefCount++; 1443 } else { 1444 if (index < 0) { 1445 return; 1446 } 1447 desc = sessionEffects.valueAt(index); 1448 if (--desc->mRefCount == 0) { 1449 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1450 sessionEffects.removeItemsAt(index); 1451 if (sessionEffects.isEmpty()) { 1452 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1453 sessionId); 1454 mSuspendedSessions.removeItem(sessionId); 1455 } 1456 } 1457 } 1458 if (!sessionEffects.isEmpty()) { 1459 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1460 } 1461} 1462 1463void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1464 bool enabled, 1465 int sessionId) 1466{ 1467 Mutex::Autolock _l(mLock); 1468 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1469} 1470 1471void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1472 bool enabled, 1473 int sessionId) 1474{ 1475 if (mType != RECORD) { 1476 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1477 // another session. This gives the priority to well behaved effect control panels 1478 // and applications not using global effects. 1479 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1480 // global effects 1481 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1482 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1483 } 1484 } 1485 1486 sp<EffectChain> chain = getEffectChain_l(sessionId); 1487 if (chain != 0) { 1488 chain->checkSuspendOnEffectEnabled(effect, enabled); 1489 } 1490} 1491 1492// ---------------------------------------------------------------------------- 1493 1494AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1495 AudioStreamOut* output, 1496 audio_io_handle_t id, 1497 audio_devices_t device, 1498 type_t type) 1499 : ThreadBase(audioFlinger, id, device, type), 1500 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1501 // Assumes constructor is called by AudioFlinger with it's mLock held, 1502 // but it would be safer to explicitly pass initial masterMute as parameter 1503 mMasterMute(audioFlinger->masterMute_l()), 1504 // mStreamTypes[] initialized in constructor body 1505 mOutput(output), 1506 // Assumes constructor is called by AudioFlinger with it's mLock held, 1507 // but it would be safer to explicitly pass initial masterVolume as parameter 1508 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1509 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1510 mMixerStatus(MIXER_IDLE), 1511 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1512 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1513 mScreenState(gScreenState), 1514 // index 0 is reserved for normal mixer's submix 1515 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1516{ 1517 snprintf(mName, kNameLength, "AudioOut_%X", id); 1518 1519 readOutputParameters(); 1520 1521 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1522 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1523 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1524 stream = (audio_stream_type_t) (stream + 1)) { 1525 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1526 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1527 } 1528 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1529 // because mAudioFlinger doesn't have one to copy from 1530} 1531 1532AudioFlinger::PlaybackThread::~PlaybackThread() 1533{ 1534 delete [] mMixBuffer; 1535} 1536 1537void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1538{ 1539 dumpInternals(fd, args); 1540 dumpTracks(fd, args); 1541 dumpEffectChains(fd, args); 1542} 1543 1544void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1545{ 1546 const size_t SIZE = 256; 1547 char buffer[SIZE]; 1548 String8 result; 1549 1550 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1551 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1552 const stream_type_t *st = &mStreamTypes[i]; 1553 if (i > 0) { 1554 result.appendFormat(", "); 1555 } 1556 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1557 if (st->mute) { 1558 result.append("M"); 1559 } 1560 } 1561 result.append("\n"); 1562 write(fd, result.string(), result.length()); 1563 result.clear(); 1564 1565 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1566 result.append(buffer); 1567 Track::appendDumpHeader(result); 1568 for (size_t i = 0; i < mTracks.size(); ++i) { 1569 sp<Track> track = mTracks[i]; 1570 if (track != 0) { 1571 track->dump(buffer, SIZE); 1572 result.append(buffer); 1573 } 1574 } 1575 1576 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1577 result.append(buffer); 1578 Track::appendDumpHeader(result); 1579 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1580 sp<Track> track = mActiveTracks[i].promote(); 1581 if (track != 0) { 1582 track->dump(buffer, SIZE); 1583 result.append(buffer); 1584 } 1585 } 1586 write(fd, result.string(), result.size()); 1587 1588 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1589 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1590 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1591 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1592} 1593 1594void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1595{ 1596 const size_t SIZE = 256; 1597 char buffer[SIZE]; 1598 String8 result; 1599 1600 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1601 result.append(buffer); 1602 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1603 result.append(buffer); 1604 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1605 result.append(buffer); 1606 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1607 result.append(buffer); 1608 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1609 result.append(buffer); 1610 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1611 result.append(buffer); 1612 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1613 result.append(buffer); 1614 write(fd, result.string(), result.size()); 1615 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1616 1617 dumpBase(fd, args); 1618} 1619 1620// Thread virtuals 1621status_t AudioFlinger::PlaybackThread::readyToRun() 1622{ 1623 status_t status = initCheck(); 1624 if (status == NO_ERROR) { 1625 ALOGI("AudioFlinger's thread %p ready to run", this); 1626 } else { 1627 ALOGE("No working audio driver found."); 1628 } 1629 return status; 1630} 1631 1632void AudioFlinger::PlaybackThread::onFirstRef() 1633{ 1634 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1635} 1636 1637// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1638sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1639 const sp<AudioFlinger::Client>& client, 1640 audio_stream_type_t streamType, 1641 uint32_t sampleRate, 1642 audio_format_t format, 1643 audio_channel_mask_t channelMask, 1644 int frameCount, 1645 const sp<IMemory>& sharedBuffer, 1646 int sessionId, 1647 IAudioFlinger::track_flags_t flags, 1648 pid_t tid, 1649 status_t *status) 1650{ 1651 sp<Track> track; 1652 status_t lStatus; 1653 1654 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1655 1656 // client expresses a preference for FAST, but we get the final say 1657 if (flags & IAudioFlinger::TRACK_FAST) { 1658 if ( 1659 // not timed 1660 (!isTimed) && 1661 // either of these use cases: 1662 ( 1663 // use case 1: shared buffer with any frame count 1664 ( 1665 (sharedBuffer != 0) 1666 ) || 1667 // use case 2: callback handler and frame count is default or at least as large as HAL 1668 ( 1669 (tid != -1) && 1670 ((frameCount == 0) || 1671 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1672 ) 1673 ) && 1674 // PCM data 1675 audio_is_linear_pcm(format) && 1676 // mono or stereo 1677 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1678 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1679#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1680 // hardware sample rate 1681 (sampleRate == mSampleRate) && 1682#endif 1683 // normal mixer has an associated fast mixer 1684 hasFastMixer() && 1685 // there are sufficient fast track slots available 1686 (mFastTrackAvailMask != 0) 1687 // FIXME test that MixerThread for this fast track has a capable output HAL 1688 // FIXME add a permission test also? 1689 ) { 1690 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1691 if (frameCount == 0) { 1692 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1693 } 1694 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1695 frameCount, mFrameCount); 1696 } else { 1697 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1698 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1699 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1700 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1701 audio_is_linear_pcm(format), 1702 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1703 flags &= ~IAudioFlinger::TRACK_FAST; 1704 // For compatibility with AudioTrack calculation, buffer depth is forced 1705 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1706 // This is probably too conservative, but legacy application code may depend on it. 1707 // If you change this calculation, also review the start threshold which is related. 1708 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1709 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1710 if (minBufCount < 2) { 1711 minBufCount = 2; 1712 } 1713 int minFrameCount = mNormalFrameCount * minBufCount; 1714 if (frameCount < minFrameCount) { 1715 frameCount = minFrameCount; 1716 } 1717 } 1718 } 1719 1720 if (mType == DIRECT) { 1721 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1722 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1723 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1724 "for output %p with format %d", 1725 sampleRate, format, channelMask, mOutput, mFormat); 1726 lStatus = BAD_VALUE; 1727 goto Exit; 1728 } 1729 } 1730 } else { 1731 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1732 if (sampleRate > mSampleRate*2) { 1733 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1734 lStatus = BAD_VALUE; 1735 goto Exit; 1736 } 1737 } 1738 1739 lStatus = initCheck(); 1740 if (lStatus != NO_ERROR) { 1741 ALOGE("Audio driver not initialized."); 1742 goto Exit; 1743 } 1744 1745 { // scope for mLock 1746 Mutex::Autolock _l(mLock); 1747 1748 // all tracks in same audio session must share the same routing strategy otherwise 1749 // conflicts will happen when tracks are moved from one output to another by audio policy 1750 // manager 1751 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1752 for (size_t i = 0; i < mTracks.size(); ++i) { 1753 sp<Track> t = mTracks[i]; 1754 if (t != 0 && !t->isOutputTrack()) { 1755 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1756 if (sessionId == t->sessionId() && strategy != actual) { 1757 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1758 strategy, actual); 1759 lStatus = BAD_VALUE; 1760 goto Exit; 1761 } 1762 } 1763 } 1764 1765 if (!isTimed) { 1766 track = new Track(this, client, streamType, sampleRate, format, 1767 channelMask, frameCount, sharedBuffer, sessionId, flags); 1768 } else { 1769 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1770 channelMask, frameCount, sharedBuffer, sessionId); 1771 } 1772 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1773 lStatus = NO_MEMORY; 1774 goto Exit; 1775 } 1776 mTracks.add(track); 1777 1778 sp<EffectChain> chain = getEffectChain_l(sessionId); 1779 if (chain != 0) { 1780 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1781 track->setMainBuffer(chain->inBuffer()); 1782 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1783 chain->incTrackCnt(); 1784 } 1785 } 1786 1787 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1788 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1789 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1790 // so ask activity manager to do this on our behalf 1791 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1792 if (err != 0) { 1793 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1794 kPriorityAudioApp, callingPid, tid, err); 1795 } 1796 } 1797 1798 lStatus = NO_ERROR; 1799 1800Exit: 1801 if (status) { 1802 *status = lStatus; 1803 } 1804 return track; 1805} 1806 1807uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1808{ 1809 if (mFastMixer != NULL) { 1810 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1811 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1812 } 1813 return latency; 1814} 1815 1816uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1817{ 1818 return latency; 1819} 1820 1821uint32_t AudioFlinger::PlaybackThread::latency() const 1822{ 1823 Mutex::Autolock _l(mLock); 1824 return latency_l(); 1825} 1826uint32_t AudioFlinger::PlaybackThread::latency_l() const 1827{ 1828 if (initCheck() == NO_ERROR) { 1829 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1830 } else { 1831 return 0; 1832 } 1833} 1834 1835void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1836{ 1837 Mutex::Autolock _l(mLock); 1838 mMasterVolume = value; 1839} 1840 1841void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1842{ 1843 Mutex::Autolock _l(mLock); 1844 setMasterMute_l(muted); 1845} 1846 1847void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1848{ 1849 Mutex::Autolock _l(mLock); 1850 mStreamTypes[stream].volume = value; 1851} 1852 1853void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1854{ 1855 Mutex::Autolock _l(mLock); 1856 mStreamTypes[stream].mute = muted; 1857} 1858 1859float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1860{ 1861 Mutex::Autolock _l(mLock); 1862 return mStreamTypes[stream].volume; 1863} 1864 1865// addTrack_l() must be called with ThreadBase::mLock held 1866status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1867{ 1868 status_t status = ALREADY_EXISTS; 1869 1870 // set retry count for buffer fill 1871 track->mRetryCount = kMaxTrackStartupRetries; 1872 if (mActiveTracks.indexOf(track) < 0) { 1873 // the track is newly added, make sure it fills up all its 1874 // buffers before playing. This is to ensure the client will 1875 // effectively get the latency it requested. 1876 track->mFillingUpStatus = Track::FS_FILLING; 1877 track->mResetDone = false; 1878 track->mPresentationCompleteFrames = 0; 1879 mActiveTracks.add(track); 1880 if (track->mainBuffer() != mMixBuffer) { 1881 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1882 if (chain != 0) { 1883 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1884 chain->incActiveTrackCnt(); 1885 } 1886 } 1887 1888 status = NO_ERROR; 1889 } 1890 1891 ALOGV("mWaitWorkCV.broadcast"); 1892 mWaitWorkCV.broadcast(); 1893 1894 return status; 1895} 1896 1897// destroyTrack_l() must be called with ThreadBase::mLock held 1898void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1899{ 1900 track->mState = TrackBase::TERMINATED; 1901 // active tracks are removed by threadLoop() 1902 if (mActiveTracks.indexOf(track) < 0) { 1903 removeTrack_l(track); 1904 } 1905} 1906 1907void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1908{ 1909 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1910 mTracks.remove(track); 1911 deleteTrackName_l(track->name()); 1912 // redundant as track is about to be destroyed, for dumpsys only 1913 track->mName = -1; 1914 if (track->isFastTrack()) { 1915 int index = track->mFastIndex; 1916 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1917 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1918 mFastTrackAvailMask |= 1 << index; 1919 // redundant as track is about to be destroyed, for dumpsys only 1920 track->mFastIndex = -1; 1921 } 1922 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1923 if (chain != 0) { 1924 chain->decTrackCnt(); 1925 } 1926} 1927 1928String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1929{ 1930 String8 out_s8 = String8(""); 1931 char *s; 1932 1933 Mutex::Autolock _l(mLock); 1934 if (initCheck() != NO_ERROR) { 1935 return out_s8; 1936 } 1937 1938 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1939 out_s8 = String8(s); 1940 free(s); 1941 return out_s8; 1942} 1943 1944// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1945void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1946 AudioSystem::OutputDescriptor desc; 1947 void *param2 = NULL; 1948 1949 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1950 1951 switch (event) { 1952 case AudioSystem::OUTPUT_OPENED: 1953 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1954 desc.channels = mChannelMask; 1955 desc.samplingRate = mSampleRate; 1956 desc.format = mFormat; 1957 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1958 desc.latency = latency(); 1959 param2 = &desc; 1960 break; 1961 1962 case AudioSystem::STREAM_CONFIG_CHANGED: 1963 param2 = ¶m; 1964 case AudioSystem::OUTPUT_CLOSED: 1965 default: 1966 break; 1967 } 1968 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1969} 1970 1971void AudioFlinger::PlaybackThread::readOutputParameters() 1972{ 1973 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1974 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1975 mChannelCount = (uint16_t)popcount(mChannelMask); 1976 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1977 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1978 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1979 if (mFrameCount & 15) { 1980 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1981 mFrameCount); 1982 } 1983 1984 // Calculate size of normal mix buffer relative to the HAL output buffer size 1985 double multiplier = 1.0; 1986 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1987 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1988 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1989 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1990 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1991 maxNormalFrameCount = maxNormalFrameCount & ~15; 1992 if (maxNormalFrameCount < minNormalFrameCount) { 1993 maxNormalFrameCount = minNormalFrameCount; 1994 } 1995 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1996 if (multiplier <= 1.0) { 1997 multiplier = 1.0; 1998 } else if (multiplier <= 2.0) { 1999 if (2 * mFrameCount <= maxNormalFrameCount) { 2000 multiplier = 2.0; 2001 } else { 2002 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2003 } 2004 } else { 2005 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2006 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2007 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2008 // FIXME this rounding up should not be done if no HAL SRC 2009 uint32_t truncMult = (uint32_t) multiplier; 2010 if ((truncMult & 1)) { 2011 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2012 ++truncMult; 2013 } 2014 } 2015 multiplier = (double) truncMult; 2016 } 2017 } 2018 mNormalFrameCount = multiplier * mFrameCount; 2019 // round up to nearest 16 frames to satisfy AudioMixer 2020 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2021 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2022 2023 delete[] mMixBuffer; 2024 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2025 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2026 2027 // force reconfiguration of effect chains and engines to take new buffer size and audio 2028 // parameters into account 2029 // Note that mLock is not held when readOutputParameters() is called from the constructor 2030 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2031 // matter. 2032 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2033 Vector< sp<EffectChain> > effectChains = mEffectChains; 2034 for (size_t i = 0; i < effectChains.size(); i ++) { 2035 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2036 } 2037} 2038 2039 2040status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2041{ 2042 if (halFrames == NULL || dspFrames == NULL) { 2043 return BAD_VALUE; 2044 } 2045 Mutex::Autolock _l(mLock); 2046 if (initCheck() != NO_ERROR) { 2047 return INVALID_OPERATION; 2048 } 2049 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2050 2051 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2052} 2053 2054uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2055{ 2056 Mutex::Autolock _l(mLock); 2057 uint32_t result = 0; 2058 if (getEffectChain_l(sessionId) != 0) { 2059 result = EFFECT_SESSION; 2060 } 2061 2062 for (size_t i = 0; i < mTracks.size(); ++i) { 2063 sp<Track> track = mTracks[i]; 2064 if (sessionId == track->sessionId() && 2065 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2066 result |= TRACK_SESSION; 2067 break; 2068 } 2069 } 2070 2071 return result; 2072} 2073 2074uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2075{ 2076 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2077 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2078 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2079 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2080 } 2081 for (size_t i = 0; i < mTracks.size(); i++) { 2082 sp<Track> track = mTracks[i]; 2083 if (sessionId == track->sessionId() && 2084 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2085 return AudioSystem::getStrategyForStream(track->streamType()); 2086 } 2087 } 2088 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2089} 2090 2091 2092AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2093{ 2094 Mutex::Autolock _l(mLock); 2095 return mOutput; 2096} 2097 2098AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2099{ 2100 Mutex::Autolock _l(mLock); 2101 AudioStreamOut *output = mOutput; 2102 mOutput = NULL; 2103 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2104 // must push a NULL and wait for ack 2105 mOutputSink.clear(); 2106 mPipeSink.clear(); 2107 mNormalSink.clear(); 2108 return output; 2109} 2110 2111// this method must always be called either with ThreadBase mLock held or inside the thread loop 2112audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2113{ 2114 if (mOutput == NULL) { 2115 return NULL; 2116 } 2117 return &mOutput->stream->common; 2118} 2119 2120uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2121{ 2122 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2123} 2124 2125status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2126{ 2127 if (!isValidSyncEvent(event)) { 2128 return BAD_VALUE; 2129 } 2130 2131 Mutex::Autolock _l(mLock); 2132 2133 for (size_t i = 0; i < mTracks.size(); ++i) { 2134 sp<Track> track = mTracks[i]; 2135 if (event->triggerSession() == track->sessionId()) { 2136 track->setSyncEvent(event); 2137 return NO_ERROR; 2138 } 2139 } 2140 2141 return NAME_NOT_FOUND; 2142} 2143 2144bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2145{ 2146 switch (event->type()) { 2147 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2148 return true; 2149 default: 2150 break; 2151 } 2152 return false; 2153} 2154 2155void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2156{ 2157 size_t count = tracksToRemove.size(); 2158 if (CC_UNLIKELY(count)) { 2159 for (size_t i = 0 ; i < count ; i++) { 2160 const sp<Track>& track = tracksToRemove.itemAt(i); 2161 if ((track->sharedBuffer() != 0) && 2162 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2163 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2164 } 2165 } 2166 } 2167 2168} 2169 2170// ---------------------------------------------------------------------------- 2171 2172AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2173 audio_io_handle_t id, audio_devices_t device, type_t type) 2174 : PlaybackThread(audioFlinger, output, id, device, type), 2175 // mAudioMixer below 2176 // mFastMixer below 2177 mFastMixerFutex(0) 2178 // mOutputSink below 2179 // mPipeSink below 2180 // mNormalSink below 2181{ 2182 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2183 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2184 "mFrameCount=%d, mNormalFrameCount=%d", 2185 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2186 mNormalFrameCount); 2187 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2188 2189 // FIXME - Current mixer implementation only supports stereo output 2190 if (mChannelCount != FCC_2) { 2191 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2192 } 2193 2194 // create an NBAIO sink for the HAL output stream, and negotiate 2195 mOutputSink = new AudioStreamOutSink(output->stream); 2196 size_t numCounterOffers = 0; 2197 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2198 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2199 ALOG_ASSERT(index == 0); 2200 2201 // initialize fast mixer depending on configuration 2202 bool initFastMixer; 2203 switch (kUseFastMixer) { 2204 case FastMixer_Never: 2205 initFastMixer = false; 2206 break; 2207 case FastMixer_Always: 2208 initFastMixer = true; 2209 break; 2210 case FastMixer_Static: 2211 case FastMixer_Dynamic: 2212 initFastMixer = mFrameCount < mNormalFrameCount; 2213 break; 2214 } 2215 if (initFastMixer) { 2216 2217 // create a MonoPipe to connect our submix to FastMixer 2218 NBAIO_Format format = mOutputSink->format(); 2219 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2220 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2221 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2222 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2223 const NBAIO_Format offers[1] = {format}; 2224 size_t numCounterOffers = 0; 2225 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2226 ALOG_ASSERT(index == 0); 2227 monoPipe->setAvgFrames((mScreenState & 1) ? 2228 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2229 mPipeSink = monoPipe; 2230 2231#ifdef TEE_SINK_FRAMES 2232 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2233 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2234 numCounterOffers = 0; 2235 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2236 ALOG_ASSERT(index == 0); 2237 mTeeSink = teeSink; 2238 PipeReader *teeSource = new PipeReader(*teeSink); 2239 numCounterOffers = 0; 2240 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2241 ALOG_ASSERT(index == 0); 2242 mTeeSource = teeSource; 2243#endif 2244 2245 // create fast mixer and configure it initially with just one fast track for our submix 2246 mFastMixer = new FastMixer(); 2247 FastMixerStateQueue *sq = mFastMixer->sq(); 2248#ifdef STATE_QUEUE_DUMP 2249 sq->setObserverDump(&mStateQueueObserverDump); 2250 sq->setMutatorDump(&mStateQueueMutatorDump); 2251#endif 2252 FastMixerState *state = sq->begin(); 2253 FastTrack *fastTrack = &state->mFastTracks[0]; 2254 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2255 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2256 fastTrack->mVolumeProvider = NULL; 2257 fastTrack->mGeneration++; 2258 state->mFastTracksGen++; 2259 state->mTrackMask = 1; 2260 // fast mixer will use the HAL output sink 2261 state->mOutputSink = mOutputSink.get(); 2262 state->mOutputSinkGen++; 2263 state->mFrameCount = mFrameCount; 2264 state->mCommand = FastMixerState::COLD_IDLE; 2265 // already done in constructor initialization list 2266 //mFastMixerFutex = 0; 2267 state->mColdFutexAddr = &mFastMixerFutex; 2268 state->mColdGen++; 2269 state->mDumpState = &mFastMixerDumpState; 2270 state->mTeeSink = mTeeSink.get(); 2271 sq->end(); 2272 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2273 2274 // start the fast mixer 2275 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2276 pid_t tid = mFastMixer->getTid(); 2277 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2278 if (err != 0) { 2279 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2280 kPriorityFastMixer, getpid_cached, tid, err); 2281 } 2282 2283#ifdef AUDIO_WATCHDOG 2284 // create and start the watchdog 2285 mAudioWatchdog = new AudioWatchdog(); 2286 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2287 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2288 tid = mAudioWatchdog->getTid(); 2289 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2290 if (err != 0) { 2291 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2292 kPriorityFastMixer, getpid_cached, tid, err); 2293 } 2294#endif 2295 2296 } else { 2297 mFastMixer = NULL; 2298 } 2299 2300 switch (kUseFastMixer) { 2301 case FastMixer_Never: 2302 case FastMixer_Dynamic: 2303 mNormalSink = mOutputSink; 2304 break; 2305 case FastMixer_Always: 2306 mNormalSink = mPipeSink; 2307 break; 2308 case FastMixer_Static: 2309 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2310 break; 2311 } 2312} 2313 2314AudioFlinger::MixerThread::~MixerThread() 2315{ 2316 if (mFastMixer != NULL) { 2317 FastMixerStateQueue *sq = mFastMixer->sq(); 2318 FastMixerState *state = sq->begin(); 2319 if (state->mCommand == FastMixerState::COLD_IDLE) { 2320 int32_t old = android_atomic_inc(&mFastMixerFutex); 2321 if (old == -1) { 2322 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2323 } 2324 } 2325 state->mCommand = FastMixerState::EXIT; 2326 sq->end(); 2327 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2328 mFastMixer->join(); 2329 // Though the fast mixer thread has exited, it's state queue is still valid. 2330 // We'll use that extract the final state which contains one remaining fast track 2331 // corresponding to our sub-mix. 2332 state = sq->begin(); 2333 ALOG_ASSERT(state->mTrackMask == 1); 2334 FastTrack *fastTrack = &state->mFastTracks[0]; 2335 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2336 delete fastTrack->mBufferProvider; 2337 sq->end(false /*didModify*/); 2338 delete mFastMixer; 2339 if (mAudioWatchdog != 0) { 2340 mAudioWatchdog->requestExit(); 2341 mAudioWatchdog->requestExitAndWait(); 2342 mAudioWatchdog.clear(); 2343 } 2344 } 2345 delete mAudioMixer; 2346} 2347 2348class CpuStats { 2349public: 2350 CpuStats(); 2351 void sample(const String8 &title); 2352#ifdef DEBUG_CPU_USAGE 2353private: 2354 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2355 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2356 2357 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2358 2359 int mCpuNum; // thread's current CPU number 2360 int mCpukHz; // frequency of thread's current CPU in kHz 2361#endif 2362}; 2363 2364CpuStats::CpuStats() 2365#ifdef DEBUG_CPU_USAGE 2366 : mCpuNum(-1), mCpukHz(-1) 2367#endif 2368{ 2369} 2370 2371void CpuStats::sample(const String8 &title) { 2372#ifdef DEBUG_CPU_USAGE 2373 // get current thread's delta CPU time in wall clock ns 2374 double wcNs; 2375 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2376 2377 // record sample for wall clock statistics 2378 if (valid) { 2379 mWcStats.sample(wcNs); 2380 } 2381 2382 // get the current CPU number 2383 int cpuNum = sched_getcpu(); 2384 2385 // get the current CPU frequency in kHz 2386 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2387 2388 // check if either CPU number or frequency changed 2389 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2390 mCpuNum = cpuNum; 2391 mCpukHz = cpukHz; 2392 // ignore sample for purposes of cycles 2393 valid = false; 2394 } 2395 2396 // if no change in CPU number or frequency, then record sample for cycle statistics 2397 if (valid && mCpukHz > 0) { 2398 double cycles = wcNs * cpukHz * 0.000001; 2399 mHzStats.sample(cycles); 2400 } 2401 2402 unsigned n = mWcStats.n(); 2403 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2404 if ((n & 127) == 1) { 2405 long long elapsed = mCpuUsage.elapsed(); 2406 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2407 double perLoop = elapsed / (double) n; 2408 double perLoop100 = perLoop * 0.01; 2409 double perLoop1k = perLoop * 0.001; 2410 double mean = mWcStats.mean(); 2411 double stddev = mWcStats.stddev(); 2412 double minimum = mWcStats.minimum(); 2413 double maximum = mWcStats.maximum(); 2414 double meanCycles = mHzStats.mean(); 2415 double stddevCycles = mHzStats.stddev(); 2416 double minCycles = mHzStats.minimum(); 2417 double maxCycles = mHzStats.maximum(); 2418 mCpuUsage.resetElapsed(); 2419 mWcStats.reset(); 2420 mHzStats.reset(); 2421 ALOGD("CPU usage for %s over past %.1f secs\n" 2422 " (%u mixer loops at %.1f mean ms per loop):\n" 2423 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2424 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2425 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2426 title.string(), 2427 elapsed * .000000001, n, perLoop * .000001, 2428 mean * .001, 2429 stddev * .001, 2430 minimum * .001, 2431 maximum * .001, 2432 mean / perLoop100, 2433 stddev / perLoop100, 2434 minimum / perLoop100, 2435 maximum / perLoop100, 2436 meanCycles / perLoop1k, 2437 stddevCycles / perLoop1k, 2438 minCycles / perLoop1k, 2439 maxCycles / perLoop1k); 2440 2441 } 2442 } 2443#endif 2444}; 2445 2446void AudioFlinger::PlaybackThread::checkSilentMode_l() 2447{ 2448 if (!mMasterMute) { 2449 char value[PROPERTY_VALUE_MAX]; 2450 if (property_get("ro.audio.silent", value, "0") > 0) { 2451 char *endptr; 2452 unsigned long ul = strtoul(value, &endptr, 0); 2453 if (*endptr == '\0' && ul != 0) { 2454 ALOGD("Silence is golden"); 2455 // The setprop command will not allow a property to be changed after 2456 // the first time it is set, so we don't have to worry about un-muting. 2457 setMasterMute_l(true); 2458 } 2459 } 2460 } 2461} 2462 2463bool AudioFlinger::PlaybackThread::threadLoop() 2464{ 2465 Vector< sp<Track> > tracksToRemove; 2466 2467 standbyTime = systemTime(); 2468 2469 // MIXER 2470 nsecs_t lastWarning = 0; 2471 2472 // DUPLICATING 2473 // FIXME could this be made local to while loop? 2474 writeFrames = 0; 2475 2476 cacheParameters_l(); 2477 sleepTime = idleSleepTime; 2478 2479 if (mType == MIXER) { 2480 sleepTimeShift = 0; 2481 } 2482 2483 CpuStats cpuStats; 2484 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2485 2486 acquireWakeLock(); 2487 2488 while (!exitPending()) 2489 { 2490 cpuStats.sample(myName); 2491 2492 Vector< sp<EffectChain> > effectChains; 2493 2494 processConfigEvents(); 2495 2496 { // scope for mLock 2497 2498 Mutex::Autolock _l(mLock); 2499 2500 if (checkForNewParameters_l()) { 2501 cacheParameters_l(); 2502 } 2503 2504 saveOutputTracks(); 2505 2506 // put audio hardware into standby after short delay 2507 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2508 isSuspended())) { 2509 if (!mStandby) { 2510 2511 threadLoop_standby(); 2512 2513 mStandby = true; 2514 mBytesWritten = 0; 2515 } 2516 2517 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2518 // we're about to wait, flush the binder command buffer 2519 IPCThreadState::self()->flushCommands(); 2520 2521 clearOutputTracks(); 2522 2523 if (exitPending()) break; 2524 2525 releaseWakeLock_l(); 2526 // wait until we have something to do... 2527 ALOGV("%s going to sleep", myName.string()); 2528 mWaitWorkCV.wait(mLock); 2529 ALOGV("%s waking up", myName.string()); 2530 acquireWakeLock_l(); 2531 2532 mMixerStatus = MIXER_IDLE; 2533 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2534 2535 checkSilentMode_l(); 2536 2537 standbyTime = systemTime() + standbyDelay; 2538 sleepTime = idleSleepTime; 2539 if (mType == MIXER) { 2540 sleepTimeShift = 0; 2541 } 2542 2543 continue; 2544 } 2545 } 2546 2547 // mMixerStatusIgnoringFastTracks is also updated internally 2548 mMixerStatus = prepareTracks_l(&tracksToRemove); 2549 2550 // prevent any changes in effect chain list and in each effect chain 2551 // during mixing and effect process as the audio buffers could be deleted 2552 // or modified if an effect is created or deleted 2553 lockEffectChains_l(effectChains); 2554 } 2555 2556 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2557 threadLoop_mix(); 2558 } else { 2559 threadLoop_sleepTime(); 2560 } 2561 2562 if (isSuspended()) { 2563 sleepTime = suspendSleepTimeUs(); 2564 } 2565 2566 // only process effects if we're going to write 2567 if (sleepTime == 0) { 2568 for (size_t i = 0; i < effectChains.size(); i ++) { 2569 effectChains[i]->process_l(); 2570 } 2571 } 2572 2573 // enable changes in effect chain 2574 unlockEffectChains(effectChains); 2575 2576 // sleepTime == 0 means we must write to audio hardware 2577 if (sleepTime == 0) { 2578 2579 threadLoop_write(); 2580 2581if (mType == MIXER) { 2582 // write blocked detection 2583 nsecs_t now = systemTime(); 2584 nsecs_t delta = now - mLastWriteTime; 2585 if (!mStandby && delta > maxPeriod) { 2586 mNumDelayedWrites++; 2587 if ((now - lastWarning) > kWarningThrottleNs) { 2588#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2589 ScopedTrace st(ATRACE_TAG, "underrun"); 2590#endif 2591 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2592 ns2ms(delta), mNumDelayedWrites, this); 2593 lastWarning = now; 2594 } 2595 } 2596} 2597 2598 mStandby = false; 2599 } else { 2600 usleep(sleepTime); 2601 } 2602 2603 // Finally let go of removed track(s), without the lock held 2604 // since we can't guarantee the destructors won't acquire that 2605 // same lock. This will also mutate and push a new fast mixer state. 2606 threadLoop_removeTracks(tracksToRemove); 2607 tracksToRemove.clear(); 2608 2609 // FIXME I don't understand the need for this here; 2610 // it was in the original code but maybe the 2611 // assignment in saveOutputTracks() makes this unnecessary? 2612 clearOutputTracks(); 2613 2614 // Effect chains will be actually deleted here if they were removed from 2615 // mEffectChains list during mixing or effects processing 2616 effectChains.clear(); 2617 2618 // FIXME Note that the above .clear() is no longer necessary since effectChains 2619 // is now local to this block, but will keep it for now (at least until merge done). 2620 } 2621 2622 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2623 if (mType == MIXER || mType == DIRECT) { 2624 // put output stream into standby mode 2625 if (!mStandby) { 2626 mOutput->stream->common.standby(&mOutput->stream->common); 2627 } 2628 } 2629 2630 releaseWakeLock(); 2631 2632 ALOGV("Thread %p type %d exiting", this, mType); 2633 return false; 2634} 2635 2636void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2637{ 2638 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2639} 2640 2641void AudioFlinger::MixerThread::threadLoop_write() 2642{ 2643 // FIXME we should only do one push per cycle; confirm this is true 2644 // Start the fast mixer if it's not already running 2645 if (mFastMixer != NULL) { 2646 FastMixerStateQueue *sq = mFastMixer->sq(); 2647 FastMixerState *state = sq->begin(); 2648 if (state->mCommand != FastMixerState::MIX_WRITE && 2649 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2650 if (state->mCommand == FastMixerState::COLD_IDLE) { 2651 int32_t old = android_atomic_inc(&mFastMixerFutex); 2652 if (old == -1) { 2653 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2654 } 2655 if (mAudioWatchdog != 0) { 2656 mAudioWatchdog->resume(); 2657 } 2658 } 2659 state->mCommand = FastMixerState::MIX_WRITE; 2660 sq->end(); 2661 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2662 if (kUseFastMixer == FastMixer_Dynamic) { 2663 mNormalSink = mPipeSink; 2664 } 2665 } else { 2666 sq->end(false /*didModify*/); 2667 } 2668 } 2669 PlaybackThread::threadLoop_write(); 2670} 2671 2672// shared by MIXER and DIRECT, overridden by DUPLICATING 2673void AudioFlinger::PlaybackThread::threadLoop_write() 2674{ 2675 // FIXME rewrite to reduce number of system calls 2676 mLastWriteTime = systemTime(); 2677 mInWrite = true; 2678 int bytesWritten; 2679 2680 // If an NBAIO sink is present, use it to write the normal mixer's submix 2681 if (mNormalSink != 0) { 2682#define mBitShift 2 // FIXME 2683 size_t count = mixBufferSize >> mBitShift; 2684#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2685 Tracer::traceBegin(ATRACE_TAG, "write"); 2686#endif 2687 // update the setpoint when gScreenState changes 2688 uint32_t screenState = gScreenState; 2689 if (screenState != mScreenState) { 2690 mScreenState = screenState; 2691 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2692 if (pipe != NULL) { 2693 pipe->setAvgFrames((mScreenState & 1) ? 2694 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2695 } 2696 } 2697 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2698#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2699 Tracer::traceEnd(ATRACE_TAG); 2700#endif 2701 if (framesWritten > 0) { 2702 bytesWritten = framesWritten << mBitShift; 2703 } else { 2704 bytesWritten = framesWritten; 2705 } 2706 // otherwise use the HAL / AudioStreamOut directly 2707 } else { 2708 // Direct output thread. 2709 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2710 } 2711 2712 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2713 mNumWrites++; 2714 mInWrite = false; 2715} 2716 2717void AudioFlinger::MixerThread::threadLoop_standby() 2718{ 2719 // Idle the fast mixer if it's currently running 2720 if (mFastMixer != NULL) { 2721 FastMixerStateQueue *sq = mFastMixer->sq(); 2722 FastMixerState *state = sq->begin(); 2723 if (!(state->mCommand & FastMixerState::IDLE)) { 2724 state->mCommand = FastMixerState::COLD_IDLE; 2725 state->mColdFutexAddr = &mFastMixerFutex; 2726 state->mColdGen++; 2727 mFastMixerFutex = 0; 2728 sq->end(); 2729 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2730 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2731 if (kUseFastMixer == FastMixer_Dynamic) { 2732 mNormalSink = mOutputSink; 2733 } 2734 if (mAudioWatchdog != 0) { 2735 mAudioWatchdog->pause(); 2736 } 2737 } else { 2738 sq->end(false /*didModify*/); 2739 } 2740 } 2741 PlaybackThread::threadLoop_standby(); 2742} 2743 2744// shared by MIXER and DIRECT, overridden by DUPLICATING 2745void AudioFlinger::PlaybackThread::threadLoop_standby() 2746{ 2747 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2748 mOutput->stream->common.standby(&mOutput->stream->common); 2749} 2750 2751void AudioFlinger::MixerThread::threadLoop_mix() 2752{ 2753 // obtain the presentation timestamp of the next output buffer 2754 int64_t pts; 2755 status_t status = INVALID_OPERATION; 2756 2757 if (NULL != mOutput->stream->get_next_write_timestamp) { 2758 status = mOutput->stream->get_next_write_timestamp( 2759 mOutput->stream, &pts); 2760 } 2761 2762 if (status != NO_ERROR) { 2763 pts = AudioBufferProvider::kInvalidPTS; 2764 } 2765 2766 // mix buffers... 2767 mAudioMixer->process(pts); 2768 // increase sleep time progressively when application underrun condition clears. 2769 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2770 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2771 // such that we would underrun the audio HAL. 2772 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2773 sleepTimeShift--; 2774 } 2775 sleepTime = 0; 2776 standbyTime = systemTime() + standbyDelay; 2777 //TODO: delay standby when effects have a tail 2778} 2779 2780void AudioFlinger::MixerThread::threadLoop_sleepTime() 2781{ 2782 // If no tracks are ready, sleep once for the duration of an output 2783 // buffer size, then write 0s to the output 2784 if (sleepTime == 0) { 2785 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2786 sleepTime = activeSleepTime >> sleepTimeShift; 2787 if (sleepTime < kMinThreadSleepTimeUs) { 2788 sleepTime = kMinThreadSleepTimeUs; 2789 } 2790 // reduce sleep time in case of consecutive application underruns to avoid 2791 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2792 // duration we would end up writing less data than needed by the audio HAL if 2793 // the condition persists. 2794 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2795 sleepTimeShift++; 2796 } 2797 } else { 2798 sleepTime = idleSleepTime; 2799 } 2800 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2801 memset (mMixBuffer, 0, mixBufferSize); 2802 sleepTime = 0; 2803 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2804 } 2805 // TODO add standby time extension fct of effect tail 2806} 2807 2808// prepareTracks_l() must be called with ThreadBase::mLock held 2809AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2810 Vector< sp<Track> > *tracksToRemove) 2811{ 2812 2813 mixer_state mixerStatus = MIXER_IDLE; 2814 // find out which tracks need to be processed 2815 size_t count = mActiveTracks.size(); 2816 size_t mixedTracks = 0; 2817 size_t tracksWithEffect = 0; 2818 // counts only _active_ fast tracks 2819 size_t fastTracks = 0; 2820 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2821 2822 float masterVolume = mMasterVolume; 2823 bool masterMute = mMasterMute; 2824 2825 if (masterMute) { 2826 masterVolume = 0; 2827 } 2828 // Delegate master volume control to effect in output mix effect chain if needed 2829 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2830 if (chain != 0) { 2831 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2832 chain->setVolume_l(&v, &v); 2833 masterVolume = (float)((v + (1 << 23)) >> 24); 2834 chain.clear(); 2835 } 2836 2837 // prepare a new state to push 2838 FastMixerStateQueue *sq = NULL; 2839 FastMixerState *state = NULL; 2840 bool didModify = false; 2841 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2842 if (mFastMixer != NULL) { 2843 sq = mFastMixer->sq(); 2844 state = sq->begin(); 2845 } 2846 2847 for (size_t i=0 ; i<count ; i++) { 2848 sp<Track> t = mActiveTracks[i].promote(); 2849 if (t == 0) continue; 2850 2851 // this const just means the local variable doesn't change 2852 Track* const track = t.get(); 2853 2854 // process fast tracks 2855 if (track->isFastTrack()) { 2856 2857 // It's theoretically possible (though unlikely) for a fast track to be created 2858 // and then removed within the same normal mix cycle. This is not a problem, as 2859 // the track never becomes active so it's fast mixer slot is never touched. 2860 // The converse, of removing an (active) track and then creating a new track 2861 // at the identical fast mixer slot within the same normal mix cycle, 2862 // is impossible because the slot isn't marked available until the end of each cycle. 2863 int j = track->mFastIndex; 2864 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2865 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2866 FastTrack *fastTrack = &state->mFastTracks[j]; 2867 2868 // Determine whether the track is currently in underrun condition, 2869 // and whether it had a recent underrun. 2870 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2871 FastTrackUnderruns underruns = ftDump->mUnderruns; 2872 uint32_t recentFull = (underruns.mBitFields.mFull - 2873 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2874 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2875 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2876 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2877 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2878 uint32_t recentUnderruns = recentPartial + recentEmpty; 2879 track->mObservedUnderruns = underruns; 2880 // don't count underruns that occur while stopping or pausing 2881 // or stopped which can occur when flush() is called while active 2882 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2883 track->mUnderrunCount += recentUnderruns; 2884 } 2885 2886 // This is similar to the state machine for normal tracks, 2887 // with a few modifications for fast tracks. 2888 bool isActive = true; 2889 switch (track->mState) { 2890 case TrackBase::STOPPING_1: 2891 // track stays active in STOPPING_1 state until first underrun 2892 if (recentUnderruns > 0) { 2893 track->mState = TrackBase::STOPPING_2; 2894 } 2895 break; 2896 case TrackBase::PAUSING: 2897 // ramp down is not yet implemented 2898 track->setPaused(); 2899 break; 2900 case TrackBase::RESUMING: 2901 // ramp up is not yet implemented 2902 track->mState = TrackBase::ACTIVE; 2903 break; 2904 case TrackBase::ACTIVE: 2905 if (recentFull > 0 || recentPartial > 0) { 2906 // track has provided at least some frames recently: reset retry count 2907 track->mRetryCount = kMaxTrackRetries; 2908 } 2909 if (recentUnderruns == 0) { 2910 // no recent underruns: stay active 2911 break; 2912 } 2913 // there has recently been an underrun of some kind 2914 if (track->sharedBuffer() == 0) { 2915 // were any of the recent underruns "empty" (no frames available)? 2916 if (recentEmpty == 0) { 2917 // no, then ignore the partial underruns as they are allowed indefinitely 2918 break; 2919 } 2920 // there has recently been an "empty" underrun: decrement the retry counter 2921 if (--(track->mRetryCount) > 0) { 2922 break; 2923 } 2924 // indicate to client process that the track was disabled because of underrun; 2925 // it will then automatically call start() when data is available 2926 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2927 // remove from active list, but state remains ACTIVE [confusing but true] 2928 isActive = false; 2929 break; 2930 } 2931 // fall through 2932 case TrackBase::STOPPING_2: 2933 case TrackBase::PAUSED: 2934 case TrackBase::TERMINATED: 2935 case TrackBase::STOPPED: 2936 case TrackBase::FLUSHED: // flush() while active 2937 // Check for presentation complete if track is inactive 2938 // We have consumed all the buffers of this track. 2939 // This would be incomplete if we auto-paused on underrun 2940 { 2941 size_t audioHALFrames = 2942 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2943 size_t framesWritten = 2944 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2945 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2946 // track stays in active list until presentation is complete 2947 break; 2948 } 2949 } 2950 if (track->isStopping_2()) { 2951 track->mState = TrackBase::STOPPED; 2952 } 2953 if (track->isStopped()) { 2954 // Can't reset directly, as fast mixer is still polling this track 2955 // track->reset(); 2956 // So instead mark this track as needing to be reset after push with ack 2957 resetMask |= 1 << i; 2958 } 2959 isActive = false; 2960 break; 2961 case TrackBase::IDLE: 2962 default: 2963 LOG_FATAL("unexpected track state %d", track->mState); 2964 } 2965 2966 if (isActive) { 2967 // was it previously inactive? 2968 if (!(state->mTrackMask & (1 << j))) { 2969 ExtendedAudioBufferProvider *eabp = track; 2970 VolumeProvider *vp = track; 2971 fastTrack->mBufferProvider = eabp; 2972 fastTrack->mVolumeProvider = vp; 2973 fastTrack->mSampleRate = track->mSampleRate; 2974 fastTrack->mChannelMask = track->mChannelMask; 2975 fastTrack->mGeneration++; 2976 state->mTrackMask |= 1 << j; 2977 didModify = true; 2978 // no acknowledgement required for newly active tracks 2979 } 2980 // cache the combined master volume and stream type volume for fast mixer; this 2981 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2982 track->mCachedVolume = track->isMuted() ? 2983 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2984 ++fastTracks; 2985 } else { 2986 // was it previously active? 2987 if (state->mTrackMask & (1 << j)) { 2988 fastTrack->mBufferProvider = NULL; 2989 fastTrack->mGeneration++; 2990 state->mTrackMask &= ~(1 << j); 2991 didModify = true; 2992 // If any fast tracks were removed, we must wait for acknowledgement 2993 // because we're about to decrement the last sp<> on those tracks. 2994 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2995 } else { 2996 LOG_FATAL("fast track %d should have been active", j); 2997 } 2998 tracksToRemove->add(track); 2999 // Avoids a misleading display in dumpsys 3000 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3001 } 3002 continue; 3003 } 3004 3005 { // local variable scope to avoid goto warning 3006 3007 audio_track_cblk_t* cblk = track->cblk(); 3008 3009 // The first time a track is added we wait 3010 // for all its buffers to be filled before processing it 3011 int name = track->name(); 3012 // make sure that we have enough frames to mix one full buffer. 3013 // enforce this condition only once to enable draining the buffer in case the client 3014 // app does not call stop() and relies on underrun to stop: 3015 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3016 // during last round 3017 uint32_t minFrames = 1; 3018 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3019 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3020 if (t->sampleRate() == (int)mSampleRate) { 3021 minFrames = mNormalFrameCount; 3022 } else { 3023 // +1 for rounding and +1 for additional sample needed for interpolation 3024 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3025 // add frames already consumed but not yet released by the resampler 3026 // because cblk->framesReady() will include these frames 3027 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3028 // the minimum track buffer size is normally twice the number of frames necessary 3029 // to fill one buffer and the resampler should not leave more than one buffer worth 3030 // of unreleased frames after each pass, but just in case... 3031 ALOG_ASSERT(minFrames <= cblk->frameCount); 3032 } 3033 } 3034 if ((track->framesReady() >= minFrames) && track->isReady() && 3035 !track->isPaused() && !track->isTerminated()) 3036 { 3037 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3038 3039 mixedTracks++; 3040 3041 // track->mainBuffer() != mMixBuffer means there is an effect chain 3042 // connected to the track 3043 chain.clear(); 3044 if (track->mainBuffer() != mMixBuffer) { 3045 chain = getEffectChain_l(track->sessionId()); 3046 // Delegate volume control to effect in track effect chain if needed 3047 if (chain != 0) { 3048 tracksWithEffect++; 3049 } else { 3050 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3051 name, track->sessionId()); 3052 } 3053 } 3054 3055 3056 int param = AudioMixer::VOLUME; 3057 if (track->mFillingUpStatus == Track::FS_FILLED) { 3058 // no ramp for the first volume setting 3059 track->mFillingUpStatus = Track::FS_ACTIVE; 3060 if (track->mState == TrackBase::RESUMING) { 3061 track->mState = TrackBase::ACTIVE; 3062 param = AudioMixer::RAMP_VOLUME; 3063 } 3064 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3065 } else if (cblk->server != 0) { 3066 // If the track is stopped before the first frame was mixed, 3067 // do not apply ramp 3068 param = AudioMixer::RAMP_VOLUME; 3069 } 3070 3071 // compute volume for this track 3072 uint32_t vl, vr, va; 3073 if (track->isMuted() || track->isPausing() || 3074 mStreamTypes[track->streamType()].mute) { 3075 vl = vr = va = 0; 3076 if (track->isPausing()) { 3077 track->setPaused(); 3078 } 3079 } else { 3080 3081 // read original volumes with volume control 3082 float typeVolume = mStreamTypes[track->streamType()].volume; 3083 float v = masterVolume * typeVolume; 3084 uint32_t vlr = cblk->getVolumeLR(); 3085 vl = vlr & 0xFFFF; 3086 vr = vlr >> 16; 3087 // track volumes come from shared memory, so can't be trusted and must be clamped 3088 if (vl > MAX_GAIN_INT) { 3089 ALOGV("Track left volume out of range: %04X", vl); 3090 vl = MAX_GAIN_INT; 3091 } 3092 if (vr > MAX_GAIN_INT) { 3093 ALOGV("Track right volume out of range: %04X", vr); 3094 vr = MAX_GAIN_INT; 3095 } 3096 // now apply the master volume and stream type volume 3097 vl = (uint32_t)(v * vl) << 12; 3098 vr = (uint32_t)(v * vr) << 12; 3099 // assuming master volume and stream type volume each go up to 1.0, 3100 // vl and vr are now in 8.24 format 3101 3102 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3103 // send level comes from shared memory and so may be corrupt 3104 if (sendLevel > MAX_GAIN_INT) { 3105 ALOGV("Track send level out of range: %04X", sendLevel); 3106 sendLevel = MAX_GAIN_INT; 3107 } 3108 va = (uint32_t)(v * sendLevel); 3109 } 3110 // Delegate volume control to effect in track effect chain if needed 3111 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3112 // Do not ramp volume if volume is controlled by effect 3113 param = AudioMixer::VOLUME; 3114 track->mHasVolumeController = true; 3115 } else { 3116 // force no volume ramp when volume controller was just disabled or removed 3117 // from effect chain to avoid volume spike 3118 if (track->mHasVolumeController) { 3119 param = AudioMixer::VOLUME; 3120 } 3121 track->mHasVolumeController = false; 3122 } 3123 3124 // Convert volumes from 8.24 to 4.12 format 3125 // This additional clamping is needed in case chain->setVolume_l() overshot 3126 vl = (vl + (1 << 11)) >> 12; 3127 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3128 vr = (vr + (1 << 11)) >> 12; 3129 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3130 3131 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3132 3133 // XXX: these things DON'T need to be done each time 3134 mAudioMixer->setBufferProvider(name, track); 3135 mAudioMixer->enable(name); 3136 3137 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3138 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3139 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3140 mAudioMixer->setParameter( 3141 name, 3142 AudioMixer::TRACK, 3143 AudioMixer::FORMAT, (void *)track->format()); 3144 mAudioMixer->setParameter( 3145 name, 3146 AudioMixer::TRACK, 3147 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3148 mAudioMixer->setParameter( 3149 name, 3150 AudioMixer::RESAMPLE, 3151 AudioMixer::SAMPLE_RATE, 3152 (void *)(cblk->sampleRate)); 3153 mAudioMixer->setParameter( 3154 name, 3155 AudioMixer::TRACK, 3156 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3157 mAudioMixer->setParameter( 3158 name, 3159 AudioMixer::TRACK, 3160 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3161 3162 // reset retry count 3163 track->mRetryCount = kMaxTrackRetries; 3164 3165 // If one track is ready, set the mixer ready if: 3166 // - the mixer was not ready during previous round OR 3167 // - no other track is not ready 3168 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3169 mixerStatus != MIXER_TRACKS_ENABLED) { 3170 mixerStatus = MIXER_TRACKS_READY; 3171 } 3172 } else { 3173 // clear effect chain input buffer if an active track underruns to avoid sending 3174 // previous audio buffer again to effects 3175 chain = getEffectChain_l(track->sessionId()); 3176 if (chain != 0) { 3177 chain->clearInputBuffer(); 3178 } 3179 3180 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3181 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3182 track->isStopped() || track->isPaused()) { 3183 // We have consumed all the buffers of this track. 3184 // Remove it from the list of active tracks. 3185 // TODO: use actual buffer filling status instead of latency when available from 3186 // audio HAL 3187 size_t audioHALFrames = 3188 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3189 size_t framesWritten = 3190 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3191 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3192 if (track->isStopped()) { 3193 track->reset(); 3194 } 3195 tracksToRemove->add(track); 3196 } 3197 } else { 3198 track->mUnderrunCount++; 3199 // No buffers for this track. Give it a few chances to 3200 // fill a buffer, then remove it from active list. 3201 if (--(track->mRetryCount) <= 0) { 3202 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3203 tracksToRemove->add(track); 3204 // indicate to client process that the track was disabled because of underrun; 3205 // it will then automatically call start() when data is available 3206 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3207 // If one track is not ready, mark the mixer also not ready if: 3208 // - the mixer was ready during previous round OR 3209 // - no other track is ready 3210 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3211 mixerStatus != MIXER_TRACKS_READY) { 3212 mixerStatus = MIXER_TRACKS_ENABLED; 3213 } 3214 } 3215 mAudioMixer->disable(name); 3216 } 3217 3218 } // local variable scope to avoid goto warning 3219track_is_ready: ; 3220 3221 } 3222 3223 // Push the new FastMixer state if necessary 3224 bool pauseAudioWatchdog = false; 3225 if (didModify) { 3226 state->mFastTracksGen++; 3227 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3228 if (kUseFastMixer == FastMixer_Dynamic && 3229 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3230 state->mCommand = FastMixerState::COLD_IDLE; 3231 state->mColdFutexAddr = &mFastMixerFutex; 3232 state->mColdGen++; 3233 mFastMixerFutex = 0; 3234 if (kUseFastMixer == FastMixer_Dynamic) { 3235 mNormalSink = mOutputSink; 3236 } 3237 // If we go into cold idle, need to wait for acknowledgement 3238 // so that fast mixer stops doing I/O. 3239 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3240 pauseAudioWatchdog = true; 3241 } 3242 sq->end(); 3243 } 3244 if (sq != NULL) { 3245 sq->end(didModify); 3246 sq->push(block); 3247 } 3248 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3249 mAudioWatchdog->pause(); 3250 } 3251 3252 // Now perform the deferred reset on fast tracks that have stopped 3253 while (resetMask != 0) { 3254 size_t i = __builtin_ctz(resetMask); 3255 ALOG_ASSERT(i < count); 3256 resetMask &= ~(1 << i); 3257 sp<Track> t = mActiveTracks[i].promote(); 3258 if (t == 0) continue; 3259 Track* track = t.get(); 3260 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3261 track->reset(); 3262 } 3263 3264 // remove all the tracks that need to be... 3265 count = tracksToRemove->size(); 3266 if (CC_UNLIKELY(count)) { 3267 for (size_t i=0 ; i<count ; i++) { 3268 const sp<Track>& track = tracksToRemove->itemAt(i); 3269 mActiveTracks.remove(track); 3270 if (track->mainBuffer() != mMixBuffer) { 3271 chain = getEffectChain_l(track->sessionId()); 3272 if (chain != 0) { 3273 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3274 chain->decActiveTrackCnt(); 3275 } 3276 } 3277 if (track->isTerminated()) { 3278 removeTrack_l(track); 3279 } 3280 } 3281 } 3282 3283 // mix buffer must be cleared if all tracks are connected to an 3284 // effect chain as in this case the mixer will not write to 3285 // mix buffer and track effects will accumulate into it 3286 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3287 // FIXME as a performance optimization, should remember previous zero status 3288 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3289 } 3290 3291 // if any fast tracks, then status is ready 3292 mMixerStatusIgnoringFastTracks = mixerStatus; 3293 if (fastTracks > 0) { 3294 mixerStatus = MIXER_TRACKS_READY; 3295 } 3296 return mixerStatus; 3297} 3298 3299/* 3300The derived values that are cached: 3301 - mixBufferSize from frame count * frame size 3302 - activeSleepTime from activeSleepTimeUs() 3303 - idleSleepTime from idleSleepTimeUs() 3304 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3305 - maxPeriod from frame count and sample rate (MIXER only) 3306 3307The parameters that affect these derived values are: 3308 - frame count 3309 - frame size 3310 - sample rate 3311 - device type: A2DP or not 3312 - device latency 3313 - format: PCM or not 3314 - active sleep time 3315 - idle sleep time 3316*/ 3317 3318void AudioFlinger::PlaybackThread::cacheParameters_l() 3319{ 3320 mixBufferSize = mNormalFrameCount * mFrameSize; 3321 activeSleepTime = activeSleepTimeUs(); 3322 idleSleepTime = idleSleepTimeUs(); 3323} 3324 3325void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3326{ 3327 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3328 this, streamType, mTracks.size()); 3329 Mutex::Autolock _l(mLock); 3330 3331 size_t size = mTracks.size(); 3332 for (size_t i = 0; i < size; i++) { 3333 sp<Track> t = mTracks[i]; 3334 if (t->streamType() == streamType) { 3335 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3336 t->mCblk->cv.signal(); 3337 } 3338 } 3339} 3340 3341// getTrackName_l() must be called with ThreadBase::mLock held 3342int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3343{ 3344 return mAudioMixer->getTrackName(channelMask); 3345} 3346 3347// deleteTrackName_l() must be called with ThreadBase::mLock held 3348void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3349{ 3350 ALOGV("remove track (%d) and delete from mixer", name); 3351 mAudioMixer->deleteTrackName(name); 3352} 3353 3354// checkForNewParameters_l() must be called with ThreadBase::mLock held 3355bool AudioFlinger::MixerThread::checkForNewParameters_l() 3356{ 3357 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3358 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3359 bool reconfig = false; 3360 3361 while (!mNewParameters.isEmpty()) { 3362 3363 if (mFastMixer != NULL) { 3364 FastMixerStateQueue *sq = mFastMixer->sq(); 3365 FastMixerState *state = sq->begin(); 3366 if (!(state->mCommand & FastMixerState::IDLE)) { 3367 previousCommand = state->mCommand; 3368 state->mCommand = FastMixerState::HOT_IDLE; 3369 sq->end(); 3370 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3371 } else { 3372 sq->end(false /*didModify*/); 3373 } 3374 } 3375 3376 status_t status = NO_ERROR; 3377 String8 keyValuePair = mNewParameters[0]; 3378 AudioParameter param = AudioParameter(keyValuePair); 3379 int value; 3380 3381 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3382 reconfig = true; 3383 } 3384 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3385 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3386 status = BAD_VALUE; 3387 } else { 3388 reconfig = true; 3389 } 3390 } 3391 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3392 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3393 status = BAD_VALUE; 3394 } else { 3395 reconfig = true; 3396 } 3397 } 3398 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3399 // do not accept frame count changes if tracks are open as the track buffer 3400 // size depends on frame count and correct behavior would not be guaranteed 3401 // if frame count is changed after track creation 3402 if (!mTracks.isEmpty()) { 3403 status = INVALID_OPERATION; 3404 } else { 3405 reconfig = true; 3406 } 3407 } 3408 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3409#ifdef ADD_BATTERY_DATA 3410 // when changing the audio output device, call addBatteryData to notify 3411 // the change 3412 if (mDevice != value) { 3413 uint32_t params = 0; 3414 // check whether speaker is on 3415 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3416 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3417 } 3418 3419 audio_devices_t deviceWithoutSpeaker 3420 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3421 // check if any other device (except speaker) is on 3422 if (value & deviceWithoutSpeaker ) { 3423 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3424 } 3425 3426 if (params != 0) { 3427 addBatteryData(params); 3428 } 3429 } 3430#endif 3431 3432 // forward device change to effects that have requested to be 3433 // aware of attached audio device. 3434 mDevice = value; 3435 for (size_t i = 0; i < mEffectChains.size(); i++) { 3436 mEffectChains[i]->setDevice_l(mDevice); 3437 } 3438 } 3439 3440 if (status == NO_ERROR) { 3441 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3442 keyValuePair.string()); 3443 if (!mStandby && status == INVALID_OPERATION) { 3444 mOutput->stream->common.standby(&mOutput->stream->common); 3445 mStandby = true; 3446 mBytesWritten = 0; 3447 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3448 keyValuePair.string()); 3449 } 3450 if (status == NO_ERROR && reconfig) { 3451 delete mAudioMixer; 3452 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3453 mAudioMixer = NULL; 3454 readOutputParameters(); 3455 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3456 for (size_t i = 0; i < mTracks.size() ; i++) { 3457 int name = getTrackName_l(mTracks[i]->mChannelMask); 3458 if (name < 0) break; 3459 mTracks[i]->mName = name; 3460 // limit track sample rate to 2 x new output sample rate 3461 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3462 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3463 } 3464 } 3465 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3466 } 3467 } 3468 3469 mNewParameters.removeAt(0); 3470 3471 mParamStatus = status; 3472 mParamCond.signal(); 3473 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3474 // already timed out waiting for the status and will never signal the condition. 3475 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3476 } 3477 3478 if (!(previousCommand & FastMixerState::IDLE)) { 3479 ALOG_ASSERT(mFastMixer != NULL); 3480 FastMixerStateQueue *sq = mFastMixer->sq(); 3481 FastMixerState *state = sq->begin(); 3482 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3483 state->mCommand = previousCommand; 3484 sq->end(); 3485 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3486 } 3487 3488 return reconfig; 3489} 3490 3491void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3492{ 3493 const size_t SIZE = 256; 3494 char buffer[SIZE]; 3495 String8 result; 3496 3497 PlaybackThread::dumpInternals(fd, args); 3498 3499 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3500 result.append(buffer); 3501 write(fd, result.string(), result.size()); 3502 3503 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3504 FastMixerDumpState copy = mFastMixerDumpState; 3505 copy.dump(fd); 3506 3507#ifdef STATE_QUEUE_DUMP 3508 // Similar for state queue 3509 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3510 observerCopy.dump(fd); 3511 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3512 mutatorCopy.dump(fd); 3513#endif 3514 3515 // Write the tee output to a .wav file 3516 NBAIO_Source *teeSource = mTeeSource.get(); 3517 if (teeSource != NULL) { 3518 char teePath[64]; 3519 struct timeval tv; 3520 gettimeofday(&tv, NULL); 3521 struct tm tm; 3522 localtime_r(&tv.tv_sec, &tm); 3523 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3524 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3525 if (teeFd >= 0) { 3526 char wavHeader[44]; 3527 memcpy(wavHeader, 3528 "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", 3529 sizeof(wavHeader)); 3530 NBAIO_Format format = teeSource->format(); 3531 unsigned channelCount = Format_channelCount(format); 3532 ALOG_ASSERT(channelCount <= FCC_2); 3533 unsigned sampleRate = Format_sampleRate(format); 3534 wavHeader[22] = channelCount; // number of channels 3535 wavHeader[24] = sampleRate; // sample rate 3536 wavHeader[25] = sampleRate >> 8; 3537 wavHeader[32] = channelCount * 2; // block alignment 3538 write(teeFd, wavHeader, sizeof(wavHeader)); 3539 size_t total = 0; 3540 bool firstRead = true; 3541 for (;;) { 3542#define TEE_SINK_READ 1024 3543 short buffer[TEE_SINK_READ * FCC_2]; 3544 size_t count = TEE_SINK_READ; 3545 ssize_t actual = teeSource->read(buffer, count); 3546 bool wasFirstRead = firstRead; 3547 firstRead = false; 3548 if (actual <= 0) { 3549 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3550 continue; 3551 } 3552 break; 3553 } 3554 ALOG_ASSERT(actual <= (ssize_t)count); 3555 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3556 total += actual; 3557 } 3558 lseek(teeFd, (off_t) 4, SEEK_SET); 3559 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3560 write(teeFd, &temp, sizeof(temp)); 3561 lseek(teeFd, (off_t) 40, SEEK_SET); 3562 temp = total * channelCount * sizeof(short); 3563 write(teeFd, &temp, sizeof(temp)); 3564 close(teeFd); 3565 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3566 } else { 3567 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3568 } 3569 } 3570 3571 if (mAudioWatchdog != 0) { 3572 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3573 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3574 wdCopy.dump(fd); 3575 } 3576} 3577 3578uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3579{ 3580 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3581} 3582 3583uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3584{ 3585 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3586} 3587 3588void AudioFlinger::MixerThread::cacheParameters_l() 3589{ 3590 PlaybackThread::cacheParameters_l(); 3591 3592 // FIXME: Relaxed timing because of a certain device that can't meet latency 3593 // Should be reduced to 2x after the vendor fixes the driver issue 3594 // increase threshold again due to low power audio mode. The way this warning 3595 // threshold is calculated and its usefulness should be reconsidered anyway. 3596 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3597} 3598 3599// ---------------------------------------------------------------------------- 3600AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3601 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3602 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3603 // mLeftVolFloat, mRightVolFloat 3604{ 3605} 3606 3607AudioFlinger::DirectOutputThread::~DirectOutputThread() 3608{ 3609} 3610 3611AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3612 Vector< sp<Track> > *tracksToRemove 3613) 3614{ 3615 sp<Track> trackToRemove; 3616 3617 mixer_state mixerStatus = MIXER_IDLE; 3618 3619 // find out which tracks need to be processed 3620 if (mActiveTracks.size() != 0) { 3621 sp<Track> t = mActiveTracks[0].promote(); 3622 // The track died recently 3623 if (t == 0) return MIXER_IDLE; 3624 3625 Track* const track = t.get(); 3626 audio_track_cblk_t* cblk = track->cblk(); 3627 3628 // The first time a track is added we wait 3629 // for all its buffers to be filled before processing it 3630 uint32_t minFrames; 3631 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3632 minFrames = mNormalFrameCount; 3633 } else { 3634 minFrames = 1; 3635 } 3636 if ((track->framesReady() >= minFrames) && track->isReady() && 3637 !track->isPaused() && !track->isTerminated()) 3638 { 3639 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3640 3641 if (track->mFillingUpStatus == Track::FS_FILLED) { 3642 track->mFillingUpStatus = Track::FS_ACTIVE; 3643 mLeftVolFloat = mRightVolFloat = 0; 3644 if (track->mState == TrackBase::RESUMING) { 3645 track->mState = TrackBase::ACTIVE; 3646 } 3647 } 3648 3649 // compute volume for this track 3650 float left, right; 3651 if (track->isMuted() || mMasterMute || track->isPausing() || 3652 mStreamTypes[track->streamType()].mute) { 3653 left = right = 0; 3654 if (track->isPausing()) { 3655 track->setPaused(); 3656 } 3657 } else { 3658 float typeVolume = mStreamTypes[track->streamType()].volume; 3659 float v = mMasterVolume * typeVolume; 3660 uint32_t vlr = cblk->getVolumeLR(); 3661 float v_clamped = v * (vlr & 0xFFFF); 3662 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3663 left = v_clamped/MAX_GAIN; 3664 v_clamped = v * (vlr >> 16); 3665 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3666 right = v_clamped/MAX_GAIN; 3667 } 3668 3669 if (left != mLeftVolFloat || right != mRightVolFloat) { 3670 mLeftVolFloat = left; 3671 mRightVolFloat = right; 3672 3673 // Convert volumes from float to 8.24 3674 uint32_t vl = (uint32_t)(left * (1 << 24)); 3675 uint32_t vr = (uint32_t)(right * (1 << 24)); 3676 3677 // Delegate volume control to effect in track effect chain if needed 3678 // only one effect chain can be present on DirectOutputThread, so if 3679 // there is one, the track is connected to it 3680 if (!mEffectChains.isEmpty()) { 3681 // Do not ramp volume if volume is controlled by effect 3682 mEffectChains[0]->setVolume_l(&vl, &vr); 3683 left = (float)vl / (1 << 24); 3684 right = (float)vr / (1 << 24); 3685 } 3686 mOutput->stream->set_volume(mOutput->stream, left, right); 3687 } 3688 3689 // reset retry count 3690 track->mRetryCount = kMaxTrackRetriesDirect; 3691 mActiveTrack = t; 3692 mixerStatus = MIXER_TRACKS_READY; 3693 } else { 3694 // clear effect chain input buffer if an active track underruns to avoid sending 3695 // previous audio buffer again to effects 3696 if (!mEffectChains.isEmpty()) { 3697 mEffectChains[0]->clearInputBuffer(); 3698 } 3699 3700 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3701 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3702 track->isStopped() || track->isPaused()) { 3703 // We have consumed all the buffers of this track. 3704 // Remove it from the list of active tracks. 3705 // TODO: implement behavior for compressed audio 3706 size_t audioHALFrames = 3707 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3708 size_t framesWritten = 3709 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3710 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3711 if (track->isStopped()) { 3712 track->reset(); 3713 } 3714 trackToRemove = track; 3715 } 3716 } else { 3717 // No buffers for this track. Give it a few chances to 3718 // fill a buffer, then remove it from active list. 3719 if (--(track->mRetryCount) <= 0) { 3720 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3721 trackToRemove = track; 3722 } else { 3723 mixerStatus = MIXER_TRACKS_ENABLED; 3724 } 3725 } 3726 } 3727 } 3728 3729 // FIXME merge this with similar code for removing multiple tracks 3730 // remove all the tracks that need to be... 3731 if (CC_UNLIKELY(trackToRemove != 0)) { 3732 tracksToRemove->add(trackToRemove); 3733 mActiveTracks.remove(trackToRemove); 3734 if (!mEffectChains.isEmpty()) { 3735 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3736 trackToRemove->sessionId()); 3737 mEffectChains[0]->decActiveTrackCnt(); 3738 } 3739 if (trackToRemove->isTerminated()) { 3740 removeTrack_l(trackToRemove); 3741 } 3742 } 3743 3744 return mixerStatus; 3745} 3746 3747void AudioFlinger::DirectOutputThread::threadLoop_mix() 3748{ 3749 AudioBufferProvider::Buffer buffer; 3750 size_t frameCount = mFrameCount; 3751 int8_t *curBuf = (int8_t *)mMixBuffer; 3752 // output audio to hardware 3753 while (frameCount) { 3754 buffer.frameCount = frameCount; 3755 mActiveTrack->getNextBuffer(&buffer); 3756 if (CC_UNLIKELY(buffer.raw == NULL)) { 3757 memset(curBuf, 0, frameCount * mFrameSize); 3758 break; 3759 } 3760 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3761 frameCount -= buffer.frameCount; 3762 curBuf += buffer.frameCount * mFrameSize; 3763 mActiveTrack->releaseBuffer(&buffer); 3764 } 3765 sleepTime = 0; 3766 standbyTime = systemTime() + standbyDelay; 3767 mActiveTrack.clear(); 3768 3769} 3770 3771void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3772{ 3773 if (sleepTime == 0) { 3774 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3775 sleepTime = activeSleepTime; 3776 } else { 3777 sleepTime = idleSleepTime; 3778 } 3779 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3780 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3781 sleepTime = 0; 3782 } 3783} 3784 3785// getTrackName_l() must be called with ThreadBase::mLock held 3786int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3787{ 3788 return 0; 3789} 3790 3791// deleteTrackName_l() must be called with ThreadBase::mLock held 3792void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3793{ 3794} 3795 3796// checkForNewParameters_l() must be called with ThreadBase::mLock held 3797bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3798{ 3799 bool reconfig = false; 3800 3801 while (!mNewParameters.isEmpty()) { 3802 status_t status = NO_ERROR; 3803 String8 keyValuePair = mNewParameters[0]; 3804 AudioParameter param = AudioParameter(keyValuePair); 3805 int value; 3806 3807 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3808 // do not accept frame count changes if tracks are open as the track buffer 3809 // size depends on frame count and correct behavior would not be garantied 3810 // if frame count is changed after track creation 3811 if (!mTracks.isEmpty()) { 3812 status = INVALID_OPERATION; 3813 } else { 3814 reconfig = true; 3815 } 3816 } 3817 if (status == NO_ERROR) { 3818 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3819 keyValuePair.string()); 3820 if (!mStandby && status == INVALID_OPERATION) { 3821 mOutput->stream->common.standby(&mOutput->stream->common); 3822 mStandby = true; 3823 mBytesWritten = 0; 3824 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3825 keyValuePair.string()); 3826 } 3827 if (status == NO_ERROR && reconfig) { 3828 readOutputParameters(); 3829 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3830 } 3831 } 3832 3833 mNewParameters.removeAt(0); 3834 3835 mParamStatus = status; 3836 mParamCond.signal(); 3837 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3838 // already timed out waiting for the status and will never signal the condition. 3839 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3840 } 3841 return reconfig; 3842} 3843 3844uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3845{ 3846 uint32_t time; 3847 if (audio_is_linear_pcm(mFormat)) { 3848 time = PlaybackThread::activeSleepTimeUs(); 3849 } else { 3850 time = 10000; 3851 } 3852 return time; 3853} 3854 3855uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3856{ 3857 uint32_t time; 3858 if (audio_is_linear_pcm(mFormat)) { 3859 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3860 } else { 3861 time = 10000; 3862 } 3863 return time; 3864} 3865 3866uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3867{ 3868 uint32_t time; 3869 if (audio_is_linear_pcm(mFormat)) { 3870 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3871 } else { 3872 time = 10000; 3873 } 3874 return time; 3875} 3876 3877void AudioFlinger::DirectOutputThread::cacheParameters_l() 3878{ 3879 PlaybackThread::cacheParameters_l(); 3880 3881 // use shorter standby delay as on normal output to release 3882 // hardware resources as soon as possible 3883 standbyDelay = microseconds(activeSleepTime*2); 3884} 3885 3886// ---------------------------------------------------------------------------- 3887 3888AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3889 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3890 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3891 mWaitTimeMs(UINT_MAX) 3892{ 3893 addOutputTrack(mainThread); 3894} 3895 3896AudioFlinger::DuplicatingThread::~DuplicatingThread() 3897{ 3898 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3899 mOutputTracks[i]->destroy(); 3900 } 3901} 3902 3903void AudioFlinger::DuplicatingThread::threadLoop_mix() 3904{ 3905 // mix buffers... 3906 if (outputsReady(outputTracks)) { 3907 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3908 } else { 3909 memset(mMixBuffer, 0, mixBufferSize); 3910 } 3911 sleepTime = 0; 3912 writeFrames = mNormalFrameCount; 3913 standbyTime = systemTime() + standbyDelay; 3914} 3915 3916void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3917{ 3918 if (sleepTime == 0) { 3919 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3920 sleepTime = activeSleepTime; 3921 } else { 3922 sleepTime = idleSleepTime; 3923 } 3924 } else if (mBytesWritten != 0) { 3925 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3926 writeFrames = mNormalFrameCount; 3927 memset(mMixBuffer, 0, mixBufferSize); 3928 } else { 3929 // flush remaining overflow buffers in output tracks 3930 writeFrames = 0; 3931 } 3932 sleepTime = 0; 3933 } 3934} 3935 3936void AudioFlinger::DuplicatingThread::threadLoop_write() 3937{ 3938 for (size_t i = 0; i < outputTracks.size(); i++) { 3939 outputTracks[i]->write(mMixBuffer, writeFrames); 3940 } 3941 mBytesWritten += mixBufferSize; 3942} 3943 3944void AudioFlinger::DuplicatingThread::threadLoop_standby() 3945{ 3946 // DuplicatingThread implements standby by stopping all tracks 3947 for (size_t i = 0; i < outputTracks.size(); i++) { 3948 outputTracks[i]->stop(); 3949 } 3950} 3951 3952void AudioFlinger::DuplicatingThread::saveOutputTracks() 3953{ 3954 outputTracks = mOutputTracks; 3955} 3956 3957void AudioFlinger::DuplicatingThread::clearOutputTracks() 3958{ 3959 outputTracks.clear(); 3960} 3961 3962void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3963{ 3964 Mutex::Autolock _l(mLock); 3965 // FIXME explain this formula 3966 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3967 OutputTrack *outputTrack = new OutputTrack(thread, 3968 this, 3969 mSampleRate, 3970 mFormat, 3971 mChannelMask, 3972 frameCount); 3973 if (outputTrack->cblk() != NULL) { 3974 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3975 mOutputTracks.add(outputTrack); 3976 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3977 updateWaitTime_l(); 3978 } 3979} 3980 3981void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3982{ 3983 Mutex::Autolock _l(mLock); 3984 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3985 if (mOutputTracks[i]->thread() == thread) { 3986 mOutputTracks[i]->destroy(); 3987 mOutputTracks.removeAt(i); 3988 updateWaitTime_l(); 3989 return; 3990 } 3991 } 3992 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3993} 3994 3995// caller must hold mLock 3996void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3997{ 3998 mWaitTimeMs = UINT_MAX; 3999 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4000 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4001 if (strong != 0) { 4002 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4003 if (waitTimeMs < mWaitTimeMs) { 4004 mWaitTimeMs = waitTimeMs; 4005 } 4006 } 4007 } 4008} 4009 4010 4011bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4012{ 4013 for (size_t i = 0; i < outputTracks.size(); i++) { 4014 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4015 if (thread == 0) { 4016 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4017 return false; 4018 } 4019 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4020 // see note at standby() declaration 4021 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4022 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4023 return false; 4024 } 4025 } 4026 return true; 4027} 4028 4029uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4030{ 4031 return (mWaitTimeMs * 1000) / 2; 4032} 4033 4034void AudioFlinger::DuplicatingThread::cacheParameters_l() 4035{ 4036 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4037 updateWaitTime_l(); 4038 4039 MixerThread::cacheParameters_l(); 4040} 4041 4042// ---------------------------------------------------------------------------- 4043 4044// TrackBase constructor must be called with AudioFlinger::mLock held 4045AudioFlinger::ThreadBase::TrackBase::TrackBase( 4046 ThreadBase *thread, 4047 const sp<Client>& client, 4048 uint32_t sampleRate, 4049 audio_format_t format, 4050 audio_channel_mask_t channelMask, 4051 int frameCount, 4052 const sp<IMemory>& sharedBuffer, 4053 int sessionId) 4054 : RefBase(), 4055 mThread(thread), 4056 mClient(client), 4057 mCblk(NULL), 4058 // mBuffer 4059 // mBufferEnd 4060 mFrameCount(0), 4061 mState(IDLE), 4062 mSampleRate(sampleRate), 4063 mFormat(format), 4064 mStepServerFailed(false), 4065 mSessionId(sessionId) 4066 // mChannelCount 4067 // mChannelMask 4068{ 4069 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4070 4071 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4072 size_t size = sizeof(audio_track_cblk_t); 4073 uint8_t channelCount = popcount(channelMask); 4074 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4075 if (sharedBuffer == 0) { 4076 size += bufferSize; 4077 } 4078 4079 if (client != NULL) { 4080 mCblkMemory = client->heap()->allocate(size); 4081 if (mCblkMemory != 0) { 4082 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4083 if (mCblk != NULL) { // construct the shared structure in-place. 4084 new(mCblk) audio_track_cblk_t(); 4085 // clear all buffers 4086 mCblk->frameCount = frameCount; 4087 mCblk->sampleRate = sampleRate; 4088// uncomment the following lines to quickly test 32-bit wraparound 4089// mCblk->user = 0xffff0000; 4090// mCblk->server = 0xffff0000; 4091// mCblk->userBase = 0xffff0000; 4092// mCblk->serverBase = 0xffff0000; 4093 mChannelCount = channelCount; 4094 mChannelMask = channelMask; 4095 if (sharedBuffer == 0) { 4096 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4097 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4098 // Force underrun condition to avoid false underrun callback until first data is 4099 // written to buffer (other flags are cleared) 4100 mCblk->flags = CBLK_UNDERRUN_ON; 4101 } else { 4102 mBuffer = sharedBuffer->pointer(); 4103 } 4104 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4105 } 4106 } else { 4107 ALOGE("not enough memory for AudioTrack size=%u", size); 4108 client->heap()->dump("AudioTrack"); 4109 return; 4110 } 4111 } else { 4112 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4113 // construct the shared structure in-place. 4114 new(mCblk) audio_track_cblk_t(); 4115 // clear all buffers 4116 mCblk->frameCount = frameCount; 4117 mCblk->sampleRate = sampleRate; 4118// uncomment the following lines to quickly test 32-bit wraparound 4119// mCblk->user = 0xffff0000; 4120// mCblk->server = 0xffff0000; 4121// mCblk->userBase = 0xffff0000; 4122// mCblk->serverBase = 0xffff0000; 4123 mChannelCount = channelCount; 4124 mChannelMask = channelMask; 4125 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4126 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4127 // Force underrun condition to avoid false underrun callback until first data is 4128 // written to buffer (other flags are cleared) 4129 mCblk->flags = CBLK_UNDERRUN_ON; 4130 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4131 } 4132} 4133 4134AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4135{ 4136 if (mCblk != NULL) { 4137 if (mClient == 0) { 4138 delete mCblk; 4139 } else { 4140 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4141 } 4142 } 4143 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4144 if (mClient != 0) { 4145 // Client destructor must run with AudioFlinger mutex locked 4146 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4147 // If the client's reference count drops to zero, the associated destructor 4148 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4149 // relying on the automatic clear() at end of scope. 4150 mClient.clear(); 4151 } 4152} 4153 4154// AudioBufferProvider interface 4155// getNextBuffer() = 0; 4156// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4157void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4158{ 4159 buffer->raw = NULL; 4160 mFrameCount = buffer->frameCount; 4161 // FIXME See note at getNextBuffer() 4162 (void) step(); // ignore return value of step() 4163 buffer->frameCount = 0; 4164} 4165 4166bool AudioFlinger::ThreadBase::TrackBase::step() { 4167 bool result; 4168 audio_track_cblk_t* cblk = this->cblk(); 4169 4170 result = cblk->stepServer(mFrameCount); 4171 if (!result) { 4172 ALOGV("stepServer failed acquiring cblk mutex"); 4173 mStepServerFailed = true; 4174 } 4175 return result; 4176} 4177 4178void AudioFlinger::ThreadBase::TrackBase::reset() { 4179 audio_track_cblk_t* cblk = this->cblk(); 4180 4181 cblk->user = 0; 4182 cblk->server = 0; 4183 cblk->userBase = 0; 4184 cblk->serverBase = 0; 4185 mStepServerFailed = false; 4186 ALOGV("TrackBase::reset"); 4187} 4188 4189int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4190 return (int)mCblk->sampleRate; 4191} 4192 4193void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4194 audio_track_cblk_t* cblk = this->cblk(); 4195 size_t frameSize = cblk->frameSize; 4196 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4197 int8_t *bufferEnd = bufferStart + frames * frameSize; 4198 4199 // Check validity of returned pointer in case the track control block would have been corrupted. 4200 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4201 "TrackBase::getBuffer buffer out of range:\n" 4202 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4203 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4204 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4205 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4206 4207 return bufferStart; 4208} 4209 4210status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4211{ 4212 mSyncEvents.add(event); 4213 return NO_ERROR; 4214} 4215 4216// ---------------------------------------------------------------------------- 4217 4218// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4219AudioFlinger::PlaybackThread::Track::Track( 4220 PlaybackThread *thread, 4221 const sp<Client>& client, 4222 audio_stream_type_t streamType, 4223 uint32_t sampleRate, 4224 audio_format_t format, 4225 audio_channel_mask_t channelMask, 4226 int frameCount, 4227 const sp<IMemory>& sharedBuffer, 4228 int sessionId, 4229 IAudioFlinger::track_flags_t flags) 4230 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4231 mMute(false), 4232 mFillingUpStatus(FS_INVALID), 4233 // mRetryCount initialized later when needed 4234 mSharedBuffer(sharedBuffer), 4235 mStreamType(streamType), 4236 mName(-1), // see note below 4237 mMainBuffer(thread->mixBuffer()), 4238 mAuxBuffer(NULL), 4239 mAuxEffectId(0), mHasVolumeController(false), 4240 mPresentationCompleteFrames(0), 4241 mFlags(flags), 4242 mFastIndex(-1), 4243 mUnderrunCount(0), 4244 mCachedVolume(1.0) 4245{ 4246 if (mCblk != NULL) { 4247 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4248 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4249 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4250 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4251 mName = thread->getTrackName_l(channelMask); 4252 mCblk->mName = mName; 4253 if (mName < 0) { 4254 ALOGE("no more track names available"); 4255 return; 4256 } 4257 // only allocate a fast track index if we were able to allocate a normal track name 4258 if (flags & IAudioFlinger::TRACK_FAST) { 4259 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4260 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4261 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4262 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4263 // FIXME This is too eager. We allocate a fast track index before the 4264 // fast track becomes active. Since fast tracks are a scarce resource, 4265 // this means we are potentially denying other more important fast tracks from 4266 // being created. It would be better to allocate the index dynamically. 4267 mFastIndex = i; 4268 mCblk->mName = i; 4269 // Read the initial underruns because this field is never cleared by the fast mixer 4270 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4271 thread->mFastTrackAvailMask &= ~(1 << i); 4272 } 4273 } 4274 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4275} 4276 4277AudioFlinger::PlaybackThread::Track::~Track() 4278{ 4279 ALOGV("PlaybackThread::Track destructor"); 4280 sp<ThreadBase> thread = mThread.promote(); 4281 if (thread != 0) { 4282 Mutex::Autolock _l(thread->mLock); 4283 mState = TERMINATED; 4284 } 4285} 4286 4287void AudioFlinger::PlaybackThread::Track::destroy() 4288{ 4289 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4290 // by removing it from mTracks vector, so there is a risk that this Tracks's 4291 // destructor is called. As the destructor needs to lock mLock, 4292 // we must acquire a strong reference on this Track before locking mLock 4293 // here so that the destructor is called only when exiting this function. 4294 // On the other hand, as long as Track::destroy() is only called by 4295 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4296 // this Track with its member mTrack. 4297 sp<Track> keep(this); 4298 { // scope for mLock 4299 sp<ThreadBase> thread = mThread.promote(); 4300 if (thread != 0) { 4301 if (!isOutputTrack()) { 4302 if (mState == ACTIVE || mState == RESUMING) { 4303 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4304 4305#ifdef ADD_BATTERY_DATA 4306 // to track the speaker usage 4307 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4308#endif 4309 } 4310 AudioSystem::releaseOutput(thread->id()); 4311 } 4312 Mutex::Autolock _l(thread->mLock); 4313 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4314 playbackThread->destroyTrack_l(this); 4315 } 4316 } 4317} 4318 4319/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4320{ 4321 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4322 " Server User Main buf Aux Buf Flags Underruns\n"); 4323} 4324 4325void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4326{ 4327 uint32_t vlr = mCblk->getVolumeLR(); 4328 if (isFastTrack()) { 4329 sprintf(buffer, " F %2d", mFastIndex); 4330 } else { 4331 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4332 } 4333 track_state state = mState; 4334 char stateChar; 4335 switch (state) { 4336 case IDLE: 4337 stateChar = 'I'; 4338 break; 4339 case TERMINATED: 4340 stateChar = 'T'; 4341 break; 4342 case STOPPING_1: 4343 stateChar = 's'; 4344 break; 4345 case STOPPING_2: 4346 stateChar = '5'; 4347 break; 4348 case STOPPED: 4349 stateChar = 'S'; 4350 break; 4351 case RESUMING: 4352 stateChar = 'R'; 4353 break; 4354 case ACTIVE: 4355 stateChar = 'A'; 4356 break; 4357 case PAUSING: 4358 stateChar = 'p'; 4359 break; 4360 case PAUSED: 4361 stateChar = 'P'; 4362 break; 4363 case FLUSHED: 4364 stateChar = 'F'; 4365 break; 4366 default: 4367 stateChar = '?'; 4368 break; 4369 } 4370 char nowInUnderrun; 4371 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4372 case UNDERRUN_FULL: 4373 nowInUnderrun = ' '; 4374 break; 4375 case UNDERRUN_PARTIAL: 4376 nowInUnderrun = '<'; 4377 break; 4378 case UNDERRUN_EMPTY: 4379 nowInUnderrun = '*'; 4380 break; 4381 default: 4382 nowInUnderrun = '?'; 4383 break; 4384 } 4385 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4386 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4387 (mClient == 0) ? getpid_cached : mClient->pid(), 4388 mStreamType, 4389 mFormat, 4390 mChannelMask, 4391 mSessionId, 4392 mFrameCount, 4393 mCblk->frameCount, 4394 stateChar, 4395 mMute, 4396 mFillingUpStatus, 4397 mCblk->sampleRate, 4398 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4399 20.0 * log10((vlr >> 16) / 4096.0), 4400 mCblk->server, 4401 mCblk->user, 4402 (int)mMainBuffer, 4403 (int)mAuxBuffer, 4404 mCblk->flags, 4405 mUnderrunCount, 4406 nowInUnderrun); 4407} 4408 4409// AudioBufferProvider interface 4410status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4411 AudioBufferProvider::Buffer* buffer, int64_t pts) 4412{ 4413 audio_track_cblk_t* cblk = this->cblk(); 4414 uint32_t framesReady; 4415 uint32_t framesReq = buffer->frameCount; 4416 4417 // Check if last stepServer failed, try to step now 4418 if (mStepServerFailed) { 4419 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4420 // Since the fast mixer is higher priority than client callback thread, 4421 // it does not result in priority inversion for client. 4422 // But a non-blocking solution would be preferable to avoid 4423 // fast mixer being unable to tryLock(), and 4424 // to avoid the extra context switches if the client wakes up, 4425 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4426 if (!step()) goto getNextBuffer_exit; 4427 ALOGV("stepServer recovered"); 4428 mStepServerFailed = false; 4429 } 4430 4431 // FIXME Same as above 4432 framesReady = cblk->framesReady(); 4433 4434 if (CC_LIKELY(framesReady)) { 4435 uint32_t s = cblk->server; 4436 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4437 4438 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4439 if (framesReq > framesReady) { 4440 framesReq = framesReady; 4441 } 4442 if (framesReq > bufferEnd - s) { 4443 framesReq = bufferEnd - s; 4444 } 4445 4446 buffer->raw = getBuffer(s, framesReq); 4447 buffer->frameCount = framesReq; 4448 return NO_ERROR; 4449 } 4450 4451getNextBuffer_exit: 4452 buffer->raw = NULL; 4453 buffer->frameCount = 0; 4454 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4455 return NOT_ENOUGH_DATA; 4456} 4457 4458// Note that framesReady() takes a mutex on the control block using tryLock(). 4459// This could result in priority inversion if framesReady() is called by the normal mixer, 4460// as the normal mixer thread runs at lower 4461// priority than the client's callback thread: there is a short window within framesReady() 4462// during which the normal mixer could be preempted, and the client callback would block. 4463// Another problem can occur if framesReady() is called by the fast mixer: 4464// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4465// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4466size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4467 return mCblk->framesReady(); 4468} 4469 4470// Don't call for fast tracks; the framesReady() could result in priority inversion 4471bool AudioFlinger::PlaybackThread::Track::isReady() const { 4472 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4473 4474 if (framesReady() >= mCblk->frameCount || 4475 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4476 mFillingUpStatus = FS_FILLED; 4477 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4478 return true; 4479 } 4480 return false; 4481} 4482 4483status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4484 int triggerSession) 4485{ 4486 status_t status = NO_ERROR; 4487 ALOGV("start(%d), calling pid %d session %d", 4488 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4489 4490 sp<ThreadBase> thread = mThread.promote(); 4491 if (thread != 0) { 4492 Mutex::Autolock _l(thread->mLock); 4493 track_state state = mState; 4494 // here the track could be either new, or restarted 4495 // in both cases "unstop" the track 4496 if (mState == PAUSED) { 4497 mState = TrackBase::RESUMING; 4498 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4499 } else { 4500 mState = TrackBase::ACTIVE; 4501 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4502 } 4503 4504 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4505 thread->mLock.unlock(); 4506 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4507 thread->mLock.lock(); 4508 4509#ifdef ADD_BATTERY_DATA 4510 // to track the speaker usage 4511 if (status == NO_ERROR) { 4512 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4513 } 4514#endif 4515 } 4516 if (status == NO_ERROR) { 4517 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4518 playbackThread->addTrack_l(this); 4519 } else { 4520 mState = state; 4521 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4522 } 4523 } else { 4524 status = BAD_VALUE; 4525 } 4526 return status; 4527} 4528 4529void AudioFlinger::PlaybackThread::Track::stop() 4530{ 4531 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4532 sp<ThreadBase> thread = mThread.promote(); 4533 if (thread != 0) { 4534 Mutex::Autolock _l(thread->mLock); 4535 track_state state = mState; 4536 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4537 // If the track is not active (PAUSED and buffers full), flush buffers 4538 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4539 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4540 reset(); 4541 mState = STOPPED; 4542 } else if (!isFastTrack()) { 4543 mState = STOPPED; 4544 } else { 4545 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4546 // and then to STOPPED and reset() when presentation is complete 4547 mState = STOPPING_1; 4548 } 4549 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4550 } 4551 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4552 thread->mLock.unlock(); 4553 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4554 thread->mLock.lock(); 4555 4556#ifdef ADD_BATTERY_DATA 4557 // to track the speaker usage 4558 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4559#endif 4560 } 4561 } 4562} 4563 4564void AudioFlinger::PlaybackThread::Track::pause() 4565{ 4566 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4567 sp<ThreadBase> thread = mThread.promote(); 4568 if (thread != 0) { 4569 Mutex::Autolock _l(thread->mLock); 4570 if (mState == ACTIVE || mState == RESUMING) { 4571 mState = PAUSING; 4572 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4573 if (!isOutputTrack()) { 4574 thread->mLock.unlock(); 4575 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4576 thread->mLock.lock(); 4577 4578#ifdef ADD_BATTERY_DATA 4579 // to track the speaker usage 4580 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4581#endif 4582 } 4583 } 4584 } 4585} 4586 4587void AudioFlinger::PlaybackThread::Track::flush() 4588{ 4589 ALOGV("flush(%d)", mName); 4590 sp<ThreadBase> thread = mThread.promote(); 4591 if (thread != 0) { 4592 Mutex::Autolock _l(thread->mLock); 4593 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4594 mState != PAUSING) { 4595 return; 4596 } 4597 // No point remaining in PAUSED state after a flush => go to 4598 // FLUSHED state 4599 mState = FLUSHED; 4600 // do not reset the track if it is still in the process of being stopped or paused. 4601 // this will be done by prepareTracks_l() when the track is stopped. 4602 // prepareTracks_l() will see mState == FLUSHED, then 4603 // remove from active track list, reset(), and trigger presentation complete 4604 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4605 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4606 reset(); 4607 } 4608 } 4609} 4610 4611void AudioFlinger::PlaybackThread::Track::reset() 4612{ 4613 // Do not reset twice to avoid discarding data written just after a flush and before 4614 // the audioflinger thread detects the track is stopped. 4615 if (!mResetDone) { 4616 TrackBase::reset(); 4617 // Force underrun condition to avoid false underrun callback until first data is 4618 // written to buffer 4619 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4620 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4621 mFillingUpStatus = FS_FILLING; 4622 mResetDone = true; 4623 if (mState == FLUSHED) { 4624 mState = IDLE; 4625 } 4626 } 4627} 4628 4629void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4630{ 4631 mMute = muted; 4632} 4633 4634status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4635{ 4636 status_t status = DEAD_OBJECT; 4637 sp<ThreadBase> thread = mThread.promote(); 4638 if (thread != 0) { 4639 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4640 sp<AudioFlinger> af = mClient->audioFlinger(); 4641 4642 Mutex::Autolock _l(af->mLock); 4643 4644 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4645 4646 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4647 Mutex::Autolock _dl(playbackThread->mLock); 4648 Mutex::Autolock _sl(srcThread->mLock); 4649 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4650 if (chain == 0) { 4651 return INVALID_OPERATION; 4652 } 4653 4654 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4655 if (effect == 0) { 4656 return INVALID_OPERATION; 4657 } 4658 srcThread->removeEffect_l(effect); 4659 playbackThread->addEffect_l(effect); 4660 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4661 if (effect->state() == EffectModule::ACTIVE || 4662 effect->state() == EffectModule::STOPPING) { 4663 effect->start(); 4664 } 4665 4666 sp<EffectChain> dstChain = effect->chain().promote(); 4667 if (dstChain == 0) { 4668 srcThread->addEffect_l(effect); 4669 return INVALID_OPERATION; 4670 } 4671 AudioSystem::unregisterEffect(effect->id()); 4672 AudioSystem::registerEffect(&effect->desc(), 4673 srcThread->id(), 4674 dstChain->strategy(), 4675 AUDIO_SESSION_OUTPUT_MIX, 4676 effect->id()); 4677 } 4678 status = playbackThread->attachAuxEffect(this, EffectId); 4679 } 4680 return status; 4681} 4682 4683void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4684{ 4685 mAuxEffectId = EffectId; 4686 mAuxBuffer = buffer; 4687} 4688 4689bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4690 size_t audioHalFrames) 4691{ 4692 // a track is considered presented when the total number of frames written to audio HAL 4693 // corresponds to the number of frames written when presentationComplete() is called for the 4694 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4695 if (mPresentationCompleteFrames == 0) { 4696 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4697 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4698 mPresentationCompleteFrames, audioHalFrames); 4699 } 4700 if (framesWritten >= mPresentationCompleteFrames) { 4701 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4702 mSessionId, framesWritten); 4703 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4704 return true; 4705 } 4706 return false; 4707} 4708 4709void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4710{ 4711 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4712 if (mSyncEvents[i]->type() == type) { 4713 mSyncEvents[i]->trigger(); 4714 mSyncEvents.removeAt(i); 4715 i--; 4716 } 4717 } 4718} 4719 4720// implement VolumeBufferProvider interface 4721 4722uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4723{ 4724 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4725 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4726 uint32_t vlr = mCblk->getVolumeLR(); 4727 uint32_t vl = vlr & 0xFFFF; 4728 uint32_t vr = vlr >> 16; 4729 // track volumes come from shared memory, so can't be trusted and must be clamped 4730 if (vl > MAX_GAIN_INT) { 4731 vl = MAX_GAIN_INT; 4732 } 4733 if (vr > MAX_GAIN_INT) { 4734 vr = MAX_GAIN_INT; 4735 } 4736 // now apply the cached master volume and stream type volume; 4737 // this is trusted but lacks any synchronization or barrier so may be stale 4738 float v = mCachedVolume; 4739 vl *= v; 4740 vr *= v; 4741 // re-combine into U4.16 4742 vlr = (vr << 16) | (vl & 0xFFFF); 4743 // FIXME look at mute, pause, and stop flags 4744 return vlr; 4745} 4746 4747status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4748{ 4749 if (mState == TERMINATED || mState == PAUSED || 4750 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4751 (mState == STOPPED)))) { 4752 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4753 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4754 event->cancel(); 4755 return INVALID_OPERATION; 4756 } 4757 TrackBase::setSyncEvent(event); 4758 return NO_ERROR; 4759} 4760 4761// timed audio tracks 4762 4763sp<AudioFlinger::PlaybackThread::TimedTrack> 4764AudioFlinger::PlaybackThread::TimedTrack::create( 4765 PlaybackThread *thread, 4766 const sp<Client>& client, 4767 audio_stream_type_t streamType, 4768 uint32_t sampleRate, 4769 audio_format_t format, 4770 audio_channel_mask_t channelMask, 4771 int frameCount, 4772 const sp<IMemory>& sharedBuffer, 4773 int sessionId) { 4774 if (!client->reserveTimedTrack()) 4775 return 0; 4776 4777 return new TimedTrack( 4778 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4779 sharedBuffer, sessionId); 4780} 4781 4782AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4783 PlaybackThread *thread, 4784 const sp<Client>& client, 4785 audio_stream_type_t streamType, 4786 uint32_t sampleRate, 4787 audio_format_t format, 4788 audio_channel_mask_t channelMask, 4789 int frameCount, 4790 const sp<IMemory>& sharedBuffer, 4791 int sessionId) 4792 : Track(thread, client, streamType, sampleRate, format, channelMask, 4793 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4794 mQueueHeadInFlight(false), 4795 mTrimQueueHeadOnRelease(false), 4796 mFramesPendingInQueue(0), 4797 mTimedSilenceBuffer(NULL), 4798 mTimedSilenceBufferSize(0), 4799 mTimedAudioOutputOnTime(false), 4800 mMediaTimeTransformValid(false) 4801{ 4802 LocalClock lc; 4803 mLocalTimeFreq = lc.getLocalFreq(); 4804 4805 mLocalTimeToSampleTransform.a_zero = 0; 4806 mLocalTimeToSampleTransform.b_zero = 0; 4807 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4808 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4809 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4810 &mLocalTimeToSampleTransform.a_to_b_denom); 4811 4812 mMediaTimeToSampleTransform.a_zero = 0; 4813 mMediaTimeToSampleTransform.b_zero = 0; 4814 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4815 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4816 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4817 &mMediaTimeToSampleTransform.a_to_b_denom); 4818} 4819 4820AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4821 mClient->releaseTimedTrack(); 4822 delete [] mTimedSilenceBuffer; 4823} 4824 4825status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4826 size_t size, sp<IMemory>* buffer) { 4827 4828 Mutex::Autolock _l(mTimedBufferQueueLock); 4829 4830 trimTimedBufferQueue_l(); 4831 4832 // lazily initialize the shared memory heap for timed buffers 4833 if (mTimedMemoryDealer == NULL) { 4834 const int kTimedBufferHeapSize = 512 << 10; 4835 4836 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4837 "AudioFlingerTimed"); 4838 if (mTimedMemoryDealer == NULL) 4839 return NO_MEMORY; 4840 } 4841 4842 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4843 if (newBuffer == NULL) { 4844 newBuffer = mTimedMemoryDealer->allocate(size); 4845 if (newBuffer == NULL) 4846 return NO_MEMORY; 4847 } 4848 4849 *buffer = newBuffer; 4850 return NO_ERROR; 4851} 4852 4853// caller must hold mTimedBufferQueueLock 4854void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4855 int64_t mediaTimeNow; 4856 { 4857 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4858 if (!mMediaTimeTransformValid) 4859 return; 4860 4861 int64_t targetTimeNow; 4862 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4863 ? mCCHelper.getCommonTime(&targetTimeNow) 4864 : mCCHelper.getLocalTime(&targetTimeNow); 4865 4866 if (OK != res) 4867 return; 4868 4869 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4870 &mediaTimeNow)) { 4871 return; 4872 } 4873 } 4874 4875 size_t trimEnd; 4876 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4877 int64_t bufEnd; 4878 4879 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4880 // We have a next buffer. Just use its PTS as the PTS of the frame 4881 // following the last frame in this buffer. If the stream is sparse 4882 // (ie, there are deliberate gaps left in the stream which should be 4883 // filled with silence by the TimedAudioTrack), then this can result 4884 // in one extra buffer being left un-trimmed when it could have 4885 // been. In general, this is not typical, and we would rather 4886 // optimized away the TS calculation below for the more common case 4887 // where PTSes are contiguous. 4888 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4889 } else { 4890 // We have no next buffer. Compute the PTS of the frame following 4891 // the last frame in this buffer by computing the duration of of 4892 // this frame in media time units and adding it to the PTS of the 4893 // buffer. 4894 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4895 / mCblk->frameSize; 4896 4897 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4898 &bufEnd)) { 4899 ALOGE("Failed to convert frame count of %lld to media time" 4900 " duration" " (scale factor %d/%u) in %s", 4901 frameCount, 4902 mMediaTimeToSampleTransform.a_to_b_numer, 4903 mMediaTimeToSampleTransform.a_to_b_denom, 4904 __PRETTY_FUNCTION__); 4905 break; 4906 } 4907 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4908 } 4909 4910 if (bufEnd > mediaTimeNow) 4911 break; 4912 4913 // Is the buffer we want to use in the middle of a mix operation right 4914 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4915 // from the mixer which should be coming back shortly. 4916 if (!trimEnd && mQueueHeadInFlight) { 4917 mTrimQueueHeadOnRelease = true; 4918 } 4919 } 4920 4921 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4922 if (trimStart < trimEnd) { 4923 // Update the bookkeeping for framesReady() 4924 for (size_t i = trimStart; i < trimEnd; ++i) { 4925 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4926 } 4927 4928 // Now actually remove the buffers from the queue. 4929 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4930 } 4931} 4932 4933void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4934 const char* logTag) { 4935 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4936 "%s called (reason \"%s\"), but timed buffer queue has no" 4937 " elements to trim.", __FUNCTION__, logTag); 4938 4939 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4940 mTimedBufferQueue.removeAt(0); 4941} 4942 4943void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4944 const TimedBuffer& buf, 4945 const char* logTag) { 4946 uint32_t bufBytes = buf.buffer()->size(); 4947 uint32_t consumedAlready = buf.position(); 4948 4949 ALOG_ASSERT(consumedAlready <= bufBytes, 4950 "Bad bookkeeping while updating frames pending. Timed buffer is" 4951 " only %u bytes long, but claims to have consumed %u" 4952 " bytes. (update reason: \"%s\")", 4953 bufBytes, consumedAlready, logTag); 4954 4955 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4956 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4957 "Bad bookkeeping while updating frames pending. Should have at" 4958 " least %u queued frames, but we think we have only %u. (update" 4959 " reason: \"%s\")", 4960 bufFrames, mFramesPendingInQueue, logTag); 4961 4962 mFramesPendingInQueue -= bufFrames; 4963} 4964 4965status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4966 const sp<IMemory>& buffer, int64_t pts) { 4967 4968 { 4969 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4970 if (!mMediaTimeTransformValid) 4971 return INVALID_OPERATION; 4972 } 4973 4974 Mutex::Autolock _l(mTimedBufferQueueLock); 4975 4976 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4977 mFramesPendingInQueue += bufFrames; 4978 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4979 4980 return NO_ERROR; 4981} 4982 4983status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4984 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4985 4986 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4987 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4988 target); 4989 4990 if (!(target == TimedAudioTrack::LOCAL_TIME || 4991 target == TimedAudioTrack::COMMON_TIME)) { 4992 return BAD_VALUE; 4993 } 4994 4995 Mutex::Autolock lock(mMediaTimeTransformLock); 4996 mMediaTimeTransform = xform; 4997 mMediaTimeTransformTarget = target; 4998 mMediaTimeTransformValid = true; 4999 5000 return NO_ERROR; 5001} 5002 5003#define min(a, b) ((a) < (b) ? (a) : (b)) 5004 5005// implementation of getNextBuffer for tracks whose buffers have timestamps 5006status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5007 AudioBufferProvider::Buffer* buffer, int64_t pts) 5008{ 5009 if (pts == AudioBufferProvider::kInvalidPTS) { 5010 buffer->raw = NULL; 5011 buffer->frameCount = 0; 5012 mTimedAudioOutputOnTime = false; 5013 return INVALID_OPERATION; 5014 } 5015 5016 Mutex::Autolock _l(mTimedBufferQueueLock); 5017 5018 ALOG_ASSERT(!mQueueHeadInFlight, 5019 "getNextBuffer called without releaseBuffer!"); 5020 5021 while (true) { 5022 5023 // if we have no timed buffers, then fail 5024 if (mTimedBufferQueue.isEmpty()) { 5025 buffer->raw = NULL; 5026 buffer->frameCount = 0; 5027 return NOT_ENOUGH_DATA; 5028 } 5029 5030 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5031 5032 // calculate the PTS of the head of the timed buffer queue expressed in 5033 // local time 5034 int64_t headLocalPTS; 5035 { 5036 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5037 5038 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5039 5040 if (mMediaTimeTransform.a_to_b_denom == 0) { 5041 // the transform represents a pause, so yield silence 5042 timedYieldSilence_l(buffer->frameCount, buffer); 5043 return NO_ERROR; 5044 } 5045 5046 int64_t transformedPTS; 5047 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5048 &transformedPTS)) { 5049 // the transform failed. this shouldn't happen, but if it does 5050 // then just drop this buffer 5051 ALOGW("timedGetNextBuffer transform failed"); 5052 buffer->raw = NULL; 5053 buffer->frameCount = 0; 5054 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5055 return NO_ERROR; 5056 } 5057 5058 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5059 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5060 &headLocalPTS)) { 5061 buffer->raw = NULL; 5062 buffer->frameCount = 0; 5063 return INVALID_OPERATION; 5064 } 5065 } else { 5066 headLocalPTS = transformedPTS; 5067 } 5068 } 5069 5070 // adjust the head buffer's PTS to reflect the portion of the head buffer 5071 // that has already been consumed 5072 int64_t effectivePTS = headLocalPTS + 5073 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5074 5075 // Calculate the delta in samples between the head of the input buffer 5076 // queue and the start of the next output buffer that will be written. 5077 // If the transformation fails because of over or underflow, it means 5078 // that the sample's position in the output stream is so far out of 5079 // whack that it should just be dropped. 5080 int64_t sampleDelta; 5081 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5082 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5083 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5084 " mix"); 5085 continue; 5086 } 5087 if (!mLocalTimeToSampleTransform.doForwardTransform( 5088 (effectivePTS - pts) << 32, &sampleDelta)) { 5089 ALOGV("*** too late during sample rate transform: dropped buffer"); 5090 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5091 continue; 5092 } 5093 5094 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5095 " sampleDelta=[%d.%08x]", 5096 head.pts(), head.position(), pts, 5097 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5098 + (sampleDelta >> 32)), 5099 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5100 5101 // if the delta between the ideal placement for the next input sample and 5102 // the current output position is within this threshold, then we will 5103 // concatenate the next input samples to the previous output 5104 const int64_t kSampleContinuityThreshold = 5105 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5106 5107 // if this is the first buffer of audio that we're emitting from this track 5108 // then it should be almost exactly on time. 5109 const int64_t kSampleStartupThreshold = 1LL << 32; 5110 5111 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5112 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5113 // the next input is close enough to being on time, so concatenate it 5114 // with the last output 5115 timedYieldSamples_l(buffer); 5116 5117 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5118 head.position(), buffer->frameCount); 5119 return NO_ERROR; 5120 } 5121 5122 // Looks like our output is not on time. Reset our on timed status. 5123 // Next time we mix samples from our input queue, then should be within 5124 // the StartupThreshold. 5125 mTimedAudioOutputOnTime = false; 5126 if (sampleDelta > 0) { 5127 // the gap between the current output position and the proper start of 5128 // the next input sample is too big, so fill it with silence 5129 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5130 5131 timedYieldSilence_l(framesUntilNextInput, buffer); 5132 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5133 return NO_ERROR; 5134 } else { 5135 // the next input sample is late 5136 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5137 size_t onTimeSamplePosition = 5138 head.position() + lateFrames * mCblk->frameSize; 5139 5140 if (onTimeSamplePosition > head.buffer()->size()) { 5141 // all the remaining samples in the head are too late, so 5142 // drop it and move on 5143 ALOGV("*** too late: dropped buffer"); 5144 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5145 continue; 5146 } else { 5147 // skip over the late samples 5148 head.setPosition(onTimeSamplePosition); 5149 5150 // yield the available samples 5151 timedYieldSamples_l(buffer); 5152 5153 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5154 return NO_ERROR; 5155 } 5156 } 5157 } 5158} 5159 5160// Yield samples from the timed buffer queue head up to the given output 5161// buffer's capacity. 5162// 5163// Caller must hold mTimedBufferQueueLock 5164void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5165 AudioBufferProvider::Buffer* buffer) { 5166 5167 const TimedBuffer& head = mTimedBufferQueue[0]; 5168 5169 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5170 head.position()); 5171 5172 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5173 mCblk->frameSize); 5174 size_t framesRequested = buffer->frameCount; 5175 buffer->frameCount = min(framesLeftInHead, framesRequested); 5176 5177 mQueueHeadInFlight = true; 5178 mTimedAudioOutputOnTime = true; 5179} 5180 5181// Yield samples of silence up to the given output buffer's capacity 5182// 5183// Caller must hold mTimedBufferQueueLock 5184void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5185 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5186 5187 // lazily allocate a buffer filled with silence 5188 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5189 delete [] mTimedSilenceBuffer; 5190 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5191 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5192 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5193 } 5194 5195 buffer->raw = mTimedSilenceBuffer; 5196 size_t framesRequested = buffer->frameCount; 5197 buffer->frameCount = min(numFrames, framesRequested); 5198 5199 mTimedAudioOutputOnTime = false; 5200} 5201 5202// AudioBufferProvider interface 5203void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5204 AudioBufferProvider::Buffer* buffer) { 5205 5206 Mutex::Autolock _l(mTimedBufferQueueLock); 5207 5208 // If the buffer which was just released is part of the buffer at the head 5209 // of the queue, be sure to update the amt of the buffer which has been 5210 // consumed. If the buffer being returned is not part of the head of the 5211 // queue, its either because the buffer is part of the silence buffer, or 5212 // because the head of the timed queue was trimmed after the mixer called 5213 // getNextBuffer but before the mixer called releaseBuffer. 5214 if (buffer->raw == mTimedSilenceBuffer) { 5215 ALOG_ASSERT(!mQueueHeadInFlight, 5216 "Queue head in flight during release of silence buffer!"); 5217 goto done; 5218 } 5219 5220 ALOG_ASSERT(mQueueHeadInFlight, 5221 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5222 " head in flight."); 5223 5224 if (mTimedBufferQueue.size()) { 5225 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5226 5227 void* start = head.buffer()->pointer(); 5228 void* end = reinterpret_cast<void*>( 5229 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5230 + head.buffer()->size()); 5231 5232 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5233 "released buffer not within the head of the timed buffer" 5234 " queue; qHead = [%p, %p], released buffer = %p", 5235 start, end, buffer->raw); 5236 5237 head.setPosition(head.position() + 5238 (buffer->frameCount * mCblk->frameSize)); 5239 mQueueHeadInFlight = false; 5240 5241 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5242 "Bad bookkeeping during releaseBuffer! Should have at" 5243 " least %u queued frames, but we think we have only %u", 5244 buffer->frameCount, mFramesPendingInQueue); 5245 5246 mFramesPendingInQueue -= buffer->frameCount; 5247 5248 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5249 || mTrimQueueHeadOnRelease) { 5250 trimTimedBufferQueueHead_l("releaseBuffer"); 5251 mTrimQueueHeadOnRelease = false; 5252 } 5253 } else { 5254 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5255 " buffers in the timed buffer queue"); 5256 } 5257 5258done: 5259 buffer->raw = 0; 5260 buffer->frameCount = 0; 5261} 5262 5263size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5264 Mutex::Autolock _l(mTimedBufferQueueLock); 5265 return mFramesPendingInQueue; 5266} 5267 5268AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5269 : mPTS(0), mPosition(0) {} 5270 5271AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5272 const sp<IMemory>& buffer, int64_t pts) 5273 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5274 5275// ---------------------------------------------------------------------------- 5276 5277// RecordTrack constructor must be called with AudioFlinger::mLock held 5278AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5279 RecordThread *thread, 5280 const sp<Client>& client, 5281 uint32_t sampleRate, 5282 audio_format_t format, 5283 audio_channel_mask_t channelMask, 5284 int frameCount, 5285 int sessionId) 5286 : TrackBase(thread, client, sampleRate, format, 5287 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5288 mOverflow(false) 5289{ 5290 if (mCblk != NULL) { 5291 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5292 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5293 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5294 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5295 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5296 } else { 5297 mCblk->frameSize = sizeof(int8_t); 5298 } 5299 } 5300} 5301 5302AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5303{ 5304 sp<ThreadBase> thread = mThread.promote(); 5305 if (thread != 0) { 5306 AudioSystem::releaseInput(thread->id()); 5307 } 5308} 5309 5310// AudioBufferProvider interface 5311status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5312{ 5313 audio_track_cblk_t* cblk = this->cblk(); 5314 uint32_t framesAvail; 5315 uint32_t framesReq = buffer->frameCount; 5316 5317 // Check if last stepServer failed, try to step now 5318 if (mStepServerFailed) { 5319 if (!step()) goto getNextBuffer_exit; 5320 ALOGV("stepServer recovered"); 5321 mStepServerFailed = false; 5322 } 5323 5324 framesAvail = cblk->framesAvailable_l(); 5325 5326 if (CC_LIKELY(framesAvail)) { 5327 uint32_t s = cblk->server; 5328 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5329 5330 if (framesReq > framesAvail) { 5331 framesReq = framesAvail; 5332 } 5333 if (framesReq > bufferEnd - s) { 5334 framesReq = bufferEnd - s; 5335 } 5336 5337 buffer->raw = getBuffer(s, framesReq); 5338 buffer->frameCount = framesReq; 5339 return NO_ERROR; 5340 } 5341 5342getNextBuffer_exit: 5343 buffer->raw = NULL; 5344 buffer->frameCount = 0; 5345 return NOT_ENOUGH_DATA; 5346} 5347 5348status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5349 int triggerSession) 5350{ 5351 sp<ThreadBase> thread = mThread.promote(); 5352 if (thread != 0) { 5353 RecordThread *recordThread = (RecordThread *)thread.get(); 5354 return recordThread->start(this, event, triggerSession); 5355 } else { 5356 return BAD_VALUE; 5357 } 5358} 5359 5360void AudioFlinger::RecordThread::RecordTrack::stop() 5361{ 5362 sp<ThreadBase> thread = mThread.promote(); 5363 if (thread != 0) { 5364 RecordThread *recordThread = (RecordThread *)thread.get(); 5365 recordThread->mLock.lock(); 5366 bool doStop = recordThread->stop_l(this); 5367 if (doStop) { 5368 TrackBase::reset(); 5369 // Force overrun condition to avoid false overrun callback until first data is 5370 // read from buffer 5371 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5372 } 5373 recordThread->mLock.unlock(); 5374 if (doStop) { 5375 AudioSystem::stopInput(recordThread->id()); 5376 } 5377 } 5378} 5379 5380void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5381{ 5382 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5383 (mClient == 0) ? getpid_cached : mClient->pid(), 5384 mFormat, 5385 mChannelMask, 5386 mSessionId, 5387 mFrameCount, 5388 mState, 5389 mCblk->sampleRate, 5390 mCblk->server, 5391 mCblk->user); 5392} 5393 5394 5395// ---------------------------------------------------------------------------- 5396 5397AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5398 PlaybackThread *playbackThread, 5399 DuplicatingThread *sourceThread, 5400 uint32_t sampleRate, 5401 audio_format_t format, 5402 audio_channel_mask_t channelMask, 5403 int frameCount) 5404 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5405 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5406 mActive(false), mSourceThread(sourceThread) 5407{ 5408 5409 if (mCblk != NULL) { 5410 mCblk->flags |= CBLK_DIRECTION_OUT; 5411 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5412 mOutBuffer.frameCount = 0; 5413 playbackThread->mTracks.add(this); 5414 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5415 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5416 mCblk, mBuffer, mCblk->buffers, 5417 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5418 } else { 5419 ALOGW("Error creating output track on thread %p", playbackThread); 5420 } 5421} 5422 5423AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5424{ 5425 clearBufferQueue(); 5426} 5427 5428status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5429 int triggerSession) 5430{ 5431 status_t status = Track::start(event, triggerSession); 5432 if (status != NO_ERROR) { 5433 return status; 5434 } 5435 5436 mActive = true; 5437 mRetryCount = 127; 5438 return status; 5439} 5440 5441void AudioFlinger::PlaybackThread::OutputTrack::stop() 5442{ 5443 Track::stop(); 5444 clearBufferQueue(); 5445 mOutBuffer.frameCount = 0; 5446 mActive = false; 5447} 5448 5449bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5450{ 5451 Buffer *pInBuffer; 5452 Buffer inBuffer; 5453 uint32_t channelCount = mChannelCount; 5454 bool outputBufferFull = false; 5455 inBuffer.frameCount = frames; 5456 inBuffer.i16 = data; 5457 5458 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5459 5460 if (!mActive && frames != 0) { 5461 start(); 5462 sp<ThreadBase> thread = mThread.promote(); 5463 if (thread != 0) { 5464 MixerThread *mixerThread = (MixerThread *)thread.get(); 5465 if (mCblk->frameCount > frames){ 5466 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5467 uint32_t startFrames = (mCblk->frameCount - frames); 5468 pInBuffer = new Buffer; 5469 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5470 pInBuffer->frameCount = startFrames; 5471 pInBuffer->i16 = pInBuffer->mBuffer; 5472 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5473 mBufferQueue.add(pInBuffer); 5474 } else { 5475 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5476 } 5477 } 5478 } 5479 } 5480 5481 while (waitTimeLeftMs) { 5482 // First write pending buffers, then new data 5483 if (mBufferQueue.size()) { 5484 pInBuffer = mBufferQueue.itemAt(0); 5485 } else { 5486 pInBuffer = &inBuffer; 5487 } 5488 5489 if (pInBuffer->frameCount == 0) { 5490 break; 5491 } 5492 5493 if (mOutBuffer.frameCount == 0) { 5494 mOutBuffer.frameCount = pInBuffer->frameCount; 5495 nsecs_t startTime = systemTime(); 5496 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5497 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5498 outputBufferFull = true; 5499 break; 5500 } 5501 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5502 if (waitTimeLeftMs >= waitTimeMs) { 5503 waitTimeLeftMs -= waitTimeMs; 5504 } else { 5505 waitTimeLeftMs = 0; 5506 } 5507 } 5508 5509 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5510 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5511 mCblk->stepUser(outFrames); 5512 pInBuffer->frameCount -= outFrames; 5513 pInBuffer->i16 += outFrames * channelCount; 5514 mOutBuffer.frameCount -= outFrames; 5515 mOutBuffer.i16 += outFrames * channelCount; 5516 5517 if (pInBuffer->frameCount == 0) { 5518 if (mBufferQueue.size()) { 5519 mBufferQueue.removeAt(0); 5520 delete [] pInBuffer->mBuffer; 5521 delete pInBuffer; 5522 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5523 } else { 5524 break; 5525 } 5526 } 5527 } 5528 5529 // If we could not write all frames, allocate a buffer and queue it for next time. 5530 if (inBuffer.frameCount) { 5531 sp<ThreadBase> thread = mThread.promote(); 5532 if (thread != 0 && !thread->standby()) { 5533 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5534 pInBuffer = new Buffer; 5535 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5536 pInBuffer->frameCount = inBuffer.frameCount; 5537 pInBuffer->i16 = pInBuffer->mBuffer; 5538 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5539 mBufferQueue.add(pInBuffer); 5540 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5541 } else { 5542 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5543 } 5544 } 5545 } 5546 5547 // Calling write() with a 0 length buffer, means that no more data will be written: 5548 // If no more buffers are pending, fill output track buffer to make sure it is started 5549 // by output mixer. 5550 if (frames == 0 && mBufferQueue.size() == 0) { 5551 if (mCblk->user < mCblk->frameCount) { 5552 frames = mCblk->frameCount - mCblk->user; 5553 pInBuffer = new Buffer; 5554 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5555 pInBuffer->frameCount = frames; 5556 pInBuffer->i16 = pInBuffer->mBuffer; 5557 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5558 mBufferQueue.add(pInBuffer); 5559 } else if (mActive) { 5560 stop(); 5561 } 5562 } 5563 5564 return outputBufferFull; 5565} 5566 5567status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5568{ 5569 int active; 5570 status_t result; 5571 audio_track_cblk_t* cblk = mCblk; 5572 uint32_t framesReq = buffer->frameCount; 5573 5574// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5575 buffer->frameCount = 0; 5576 5577 uint32_t framesAvail = cblk->framesAvailable(); 5578 5579 5580 if (framesAvail == 0) { 5581 Mutex::Autolock _l(cblk->lock); 5582 goto start_loop_here; 5583 while (framesAvail == 0) { 5584 active = mActive; 5585 if (CC_UNLIKELY(!active)) { 5586 ALOGV("Not active and NO_MORE_BUFFERS"); 5587 return NO_MORE_BUFFERS; 5588 } 5589 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5590 if (result != NO_ERROR) { 5591 return NO_MORE_BUFFERS; 5592 } 5593 // read the server count again 5594 start_loop_here: 5595 framesAvail = cblk->framesAvailable_l(); 5596 } 5597 } 5598 5599// if (framesAvail < framesReq) { 5600// return NO_MORE_BUFFERS; 5601// } 5602 5603 if (framesReq > framesAvail) { 5604 framesReq = framesAvail; 5605 } 5606 5607 uint32_t u = cblk->user; 5608 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5609 5610 if (framesReq > bufferEnd - u) { 5611 framesReq = bufferEnd - u; 5612 } 5613 5614 buffer->frameCount = framesReq; 5615 buffer->raw = (void *)cblk->buffer(u); 5616 return NO_ERROR; 5617} 5618 5619 5620void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5621{ 5622 size_t size = mBufferQueue.size(); 5623 5624 for (size_t i = 0; i < size; i++) { 5625 Buffer *pBuffer = mBufferQueue.itemAt(i); 5626 delete [] pBuffer->mBuffer; 5627 delete pBuffer; 5628 } 5629 mBufferQueue.clear(); 5630} 5631 5632// ---------------------------------------------------------------------------- 5633 5634AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5635 : RefBase(), 5636 mAudioFlinger(audioFlinger), 5637 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5638 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5639 mPid(pid), 5640 mTimedTrackCount(0) 5641{ 5642 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5643} 5644 5645// Client destructor must be called with AudioFlinger::mLock held 5646AudioFlinger::Client::~Client() 5647{ 5648 mAudioFlinger->removeClient_l(mPid); 5649} 5650 5651sp<MemoryDealer> AudioFlinger::Client::heap() const 5652{ 5653 return mMemoryDealer; 5654} 5655 5656// Reserve one of the limited slots for a timed audio track associated 5657// with this client 5658bool AudioFlinger::Client::reserveTimedTrack() 5659{ 5660 const int kMaxTimedTracksPerClient = 4; 5661 5662 Mutex::Autolock _l(mTimedTrackLock); 5663 5664 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5665 ALOGW("can not create timed track - pid %d has exceeded the limit", 5666 mPid); 5667 return false; 5668 } 5669 5670 mTimedTrackCount++; 5671 return true; 5672} 5673 5674// Release a slot for a timed audio track 5675void AudioFlinger::Client::releaseTimedTrack() 5676{ 5677 Mutex::Autolock _l(mTimedTrackLock); 5678 mTimedTrackCount--; 5679} 5680 5681// ---------------------------------------------------------------------------- 5682 5683AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5684 const sp<IAudioFlingerClient>& client, 5685 pid_t pid) 5686 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5687{ 5688} 5689 5690AudioFlinger::NotificationClient::~NotificationClient() 5691{ 5692} 5693 5694void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5695{ 5696 sp<NotificationClient> keep(this); 5697 mAudioFlinger->removeNotificationClient(mPid); 5698} 5699 5700// ---------------------------------------------------------------------------- 5701 5702AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5703 : BnAudioTrack(), 5704 mTrack(track) 5705{ 5706} 5707 5708AudioFlinger::TrackHandle::~TrackHandle() { 5709 // just stop the track on deletion, associated resources 5710 // will be freed from the main thread once all pending buffers have 5711 // been played. Unless it's not in the active track list, in which 5712 // case we free everything now... 5713 mTrack->destroy(); 5714} 5715 5716sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5717 return mTrack->getCblk(); 5718} 5719 5720status_t AudioFlinger::TrackHandle::start() { 5721 return mTrack->start(); 5722} 5723 5724void AudioFlinger::TrackHandle::stop() { 5725 mTrack->stop(); 5726} 5727 5728void AudioFlinger::TrackHandle::flush() { 5729 mTrack->flush(); 5730} 5731 5732void AudioFlinger::TrackHandle::mute(bool e) { 5733 mTrack->mute(e); 5734} 5735 5736void AudioFlinger::TrackHandle::pause() { 5737 mTrack->pause(); 5738} 5739 5740status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5741{ 5742 return mTrack->attachAuxEffect(EffectId); 5743} 5744 5745status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5746 sp<IMemory>* buffer) { 5747 if (!mTrack->isTimedTrack()) 5748 return INVALID_OPERATION; 5749 5750 PlaybackThread::TimedTrack* tt = 5751 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5752 return tt->allocateTimedBuffer(size, buffer); 5753} 5754 5755status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5756 int64_t pts) { 5757 if (!mTrack->isTimedTrack()) 5758 return INVALID_OPERATION; 5759 5760 PlaybackThread::TimedTrack* tt = 5761 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5762 return tt->queueTimedBuffer(buffer, pts); 5763} 5764 5765status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5766 const LinearTransform& xform, int target) { 5767 5768 if (!mTrack->isTimedTrack()) 5769 return INVALID_OPERATION; 5770 5771 PlaybackThread::TimedTrack* tt = 5772 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5773 return tt->setMediaTimeTransform( 5774 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5775} 5776 5777status_t AudioFlinger::TrackHandle::onTransact( 5778 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5779{ 5780 return BnAudioTrack::onTransact(code, data, reply, flags); 5781} 5782 5783// ---------------------------------------------------------------------------- 5784 5785sp<IAudioRecord> AudioFlinger::openRecord( 5786 pid_t pid, 5787 audio_io_handle_t input, 5788 uint32_t sampleRate, 5789 audio_format_t format, 5790 audio_channel_mask_t channelMask, 5791 int frameCount, 5792 IAudioFlinger::track_flags_t flags, 5793 pid_t tid, 5794 int *sessionId, 5795 status_t *status) 5796{ 5797 sp<RecordThread::RecordTrack> recordTrack; 5798 sp<RecordHandle> recordHandle; 5799 sp<Client> client; 5800 status_t lStatus; 5801 RecordThread *thread; 5802 size_t inFrameCount; 5803 int lSessionId; 5804 5805 // check calling permissions 5806 if (!recordingAllowed()) { 5807 lStatus = PERMISSION_DENIED; 5808 goto Exit; 5809 } 5810 5811 // add client to list 5812 { // scope for mLock 5813 Mutex::Autolock _l(mLock); 5814 thread = checkRecordThread_l(input); 5815 if (thread == NULL) { 5816 lStatus = BAD_VALUE; 5817 goto Exit; 5818 } 5819 5820 client = registerPid_l(pid); 5821 5822 // If no audio session id is provided, create one here 5823 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5824 lSessionId = *sessionId; 5825 } else { 5826 lSessionId = nextUniqueId(); 5827 if (sessionId != NULL) { 5828 *sessionId = lSessionId; 5829 } 5830 } 5831 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5832 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5833 frameCount, lSessionId, flags, tid, &lStatus); 5834 } 5835 if (lStatus != NO_ERROR) { 5836 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5837 // destructor is called by the TrackBase destructor with mLock held 5838 client.clear(); 5839 recordTrack.clear(); 5840 goto Exit; 5841 } 5842 5843 // return to handle to client 5844 recordHandle = new RecordHandle(recordTrack); 5845 lStatus = NO_ERROR; 5846 5847Exit: 5848 if (status) { 5849 *status = lStatus; 5850 } 5851 return recordHandle; 5852} 5853 5854// ---------------------------------------------------------------------------- 5855 5856AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5857 : BnAudioRecord(), 5858 mRecordTrack(recordTrack) 5859{ 5860} 5861 5862AudioFlinger::RecordHandle::~RecordHandle() { 5863 stop_nonvirtual(); 5864} 5865 5866sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5867 return mRecordTrack->getCblk(); 5868} 5869 5870status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5871 ALOGV("RecordHandle::start()"); 5872 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5873} 5874 5875void AudioFlinger::RecordHandle::stop() { 5876 stop_nonvirtual(); 5877} 5878 5879void AudioFlinger::RecordHandle::stop_nonvirtual() { 5880 ALOGV("RecordHandle::stop()"); 5881 mRecordTrack->stop(); 5882} 5883 5884status_t AudioFlinger::RecordHandle::onTransact( 5885 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5886{ 5887 return BnAudioRecord::onTransact(code, data, reply, flags); 5888} 5889 5890// ---------------------------------------------------------------------------- 5891 5892AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5893 AudioStreamIn *input, 5894 uint32_t sampleRate, 5895 audio_channel_mask_t channelMask, 5896 audio_io_handle_t id, 5897 audio_devices_t device) : 5898 ThreadBase(audioFlinger, id, device, RECORD), 5899 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5900 // mRsmpInIndex and mInputBytes set by readInputParameters() 5901 mReqChannelCount(popcount(channelMask)), 5902 mReqSampleRate(sampleRate) 5903 // mBytesRead is only meaningful while active, and so is cleared in start() 5904 // (but might be better to also clear here for dump?) 5905{ 5906 snprintf(mName, kNameLength, "AudioIn_%X", id); 5907 5908 readInputParameters(); 5909} 5910 5911 5912AudioFlinger::RecordThread::~RecordThread() 5913{ 5914 delete[] mRsmpInBuffer; 5915 delete mResampler; 5916 delete[] mRsmpOutBuffer; 5917} 5918 5919void AudioFlinger::RecordThread::onFirstRef() 5920{ 5921 run(mName, PRIORITY_URGENT_AUDIO); 5922} 5923 5924status_t AudioFlinger::RecordThread::readyToRun() 5925{ 5926 status_t status = initCheck(); 5927 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5928 return status; 5929} 5930 5931bool AudioFlinger::RecordThread::threadLoop() 5932{ 5933 AudioBufferProvider::Buffer buffer; 5934 sp<RecordTrack> activeTrack; 5935 Vector< sp<EffectChain> > effectChains; 5936 5937 nsecs_t lastWarning = 0; 5938 5939 inputStandBy(); 5940 acquireWakeLock(); 5941 5942 // start recording 5943 while (!exitPending()) { 5944 5945 processConfigEvents(); 5946 5947 { // scope for mLock 5948 Mutex::Autolock _l(mLock); 5949 checkForNewParameters_l(); 5950 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5951 standby(); 5952 5953 if (exitPending()) break; 5954 5955 releaseWakeLock_l(); 5956 ALOGV("RecordThread: loop stopping"); 5957 // go to sleep 5958 mWaitWorkCV.wait(mLock); 5959 ALOGV("RecordThread: loop starting"); 5960 acquireWakeLock_l(); 5961 continue; 5962 } 5963 if (mActiveTrack != 0) { 5964 if (mActiveTrack->mState == TrackBase::PAUSING) { 5965 standby(); 5966 mActiveTrack.clear(); 5967 mStartStopCond.broadcast(); 5968 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5969 if (mReqChannelCount != mActiveTrack->channelCount()) { 5970 mActiveTrack.clear(); 5971 mStartStopCond.broadcast(); 5972 } else if (mBytesRead != 0) { 5973 // record start succeeds only if first read from audio input 5974 // succeeds 5975 if (mBytesRead > 0) { 5976 mActiveTrack->mState = TrackBase::ACTIVE; 5977 } else { 5978 mActiveTrack.clear(); 5979 } 5980 mStartStopCond.broadcast(); 5981 } 5982 mStandby = false; 5983 } 5984 } 5985 lockEffectChains_l(effectChains); 5986 } 5987 5988 if (mActiveTrack != 0) { 5989 if (mActiveTrack->mState != TrackBase::ACTIVE && 5990 mActiveTrack->mState != TrackBase::RESUMING) { 5991 unlockEffectChains(effectChains); 5992 usleep(kRecordThreadSleepUs); 5993 continue; 5994 } 5995 for (size_t i = 0; i < effectChains.size(); i ++) { 5996 effectChains[i]->process_l(); 5997 } 5998 5999 buffer.frameCount = mFrameCount; 6000 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6001 size_t framesOut = buffer.frameCount; 6002 if (mResampler == NULL) { 6003 // no resampling 6004 while (framesOut) { 6005 size_t framesIn = mFrameCount - mRsmpInIndex; 6006 if (framesIn) { 6007 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6008 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6009 if (framesIn > framesOut) 6010 framesIn = framesOut; 6011 mRsmpInIndex += framesIn; 6012 framesOut -= framesIn; 6013 if ((int)mChannelCount == mReqChannelCount || 6014 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6015 memcpy(dst, src, framesIn * mFrameSize); 6016 } else { 6017 if (mChannelCount == 1) { 6018 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6019 (int16_t *)src, framesIn); 6020 } else { 6021 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6022 (int16_t *)src, framesIn); 6023 } 6024 } 6025 } 6026 if (framesOut && mFrameCount == mRsmpInIndex) { 6027 if (framesOut == mFrameCount && 6028 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6029 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6030 framesOut = 0; 6031 } else { 6032 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6033 mRsmpInIndex = 0; 6034 } 6035 if (mBytesRead < 0) { 6036 ALOGE("Error reading audio input"); 6037 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6038 // Force input into standby so that it tries to 6039 // recover at next read attempt 6040 inputStandBy(); 6041 usleep(kRecordThreadSleepUs); 6042 } 6043 mRsmpInIndex = mFrameCount; 6044 framesOut = 0; 6045 buffer.frameCount = 0; 6046 } 6047 } 6048 } 6049 } else { 6050 // resampling 6051 6052 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6053 // alter output frame count as if we were expecting stereo samples 6054 if (mChannelCount == 1 && mReqChannelCount == 1) { 6055 framesOut >>= 1; 6056 } 6057 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6058 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6059 // are 32 bit aligned which should be always true. 6060 if (mChannelCount == 2 && mReqChannelCount == 1) { 6061 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6062 // the resampler always outputs stereo samples: do post stereo to mono conversion 6063 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6064 framesOut); 6065 } else { 6066 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6067 } 6068 6069 } 6070 if (mFramestoDrop == 0) { 6071 mActiveTrack->releaseBuffer(&buffer); 6072 } else { 6073 if (mFramestoDrop > 0) { 6074 mFramestoDrop -= buffer.frameCount; 6075 if (mFramestoDrop <= 0) { 6076 clearSyncStartEvent(); 6077 } 6078 } else { 6079 mFramestoDrop += buffer.frameCount; 6080 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6081 mSyncStartEvent->isCancelled()) { 6082 ALOGW("Synced record %s, session %d, trigger session %d", 6083 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6084 mActiveTrack->sessionId(), 6085 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6086 clearSyncStartEvent(); 6087 } 6088 } 6089 } 6090 mActiveTrack->clearOverflow(); 6091 } 6092 // client isn't retrieving buffers fast enough 6093 else { 6094 if (!mActiveTrack->setOverflow()) { 6095 nsecs_t now = systemTime(); 6096 if ((now - lastWarning) > kWarningThrottleNs) { 6097 ALOGW("RecordThread: buffer overflow"); 6098 lastWarning = now; 6099 } 6100 } 6101 // Release the processor for a while before asking for a new buffer. 6102 // This will give the application more chance to read from the buffer and 6103 // clear the overflow. 6104 usleep(kRecordThreadSleepUs); 6105 } 6106 } 6107 // enable changes in effect chain 6108 unlockEffectChains(effectChains); 6109 effectChains.clear(); 6110 } 6111 6112 standby(); 6113 6114 { 6115 Mutex::Autolock _l(mLock); 6116 mActiveTrack.clear(); 6117 mStartStopCond.broadcast(); 6118 } 6119 6120 releaseWakeLock(); 6121 6122 ALOGV("RecordThread %p exiting", this); 6123 return false; 6124} 6125 6126void AudioFlinger::RecordThread::standby() 6127{ 6128 if (!mStandby) { 6129 inputStandBy(); 6130 mStandby = true; 6131 } 6132} 6133 6134void AudioFlinger::RecordThread::inputStandBy() 6135{ 6136 mInput->stream->common.standby(&mInput->stream->common); 6137} 6138 6139sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6140 const sp<AudioFlinger::Client>& client, 6141 uint32_t sampleRate, 6142 audio_format_t format, 6143 audio_channel_mask_t channelMask, 6144 int frameCount, 6145 int sessionId, 6146 IAudioFlinger::track_flags_t flags, 6147 pid_t tid, 6148 status_t *status) 6149{ 6150 sp<RecordTrack> track; 6151 status_t lStatus; 6152 6153 lStatus = initCheck(); 6154 if (lStatus != NO_ERROR) { 6155 ALOGE("Audio driver not initialized."); 6156 goto Exit; 6157 } 6158 6159 // FIXME use flags and tid similar to createTrack_l() 6160 6161 { // scope for mLock 6162 Mutex::Autolock _l(mLock); 6163 6164 track = new RecordTrack(this, client, sampleRate, 6165 format, channelMask, frameCount, sessionId); 6166 6167 if (track->getCblk() == 0) { 6168 lStatus = NO_MEMORY; 6169 goto Exit; 6170 } 6171 6172 mTrack = track.get(); 6173 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6174 bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) && 6175 mAudioFlinger->btNrecIsOff(); 6176 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6177 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6178 } 6179 lStatus = NO_ERROR; 6180 6181Exit: 6182 if (status) { 6183 *status = lStatus; 6184 } 6185 return track; 6186} 6187 6188status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6189 AudioSystem::sync_event_t event, 6190 int triggerSession) 6191{ 6192 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6193 sp<ThreadBase> strongMe = this; 6194 status_t status = NO_ERROR; 6195 6196 if (event == AudioSystem::SYNC_EVENT_NONE) { 6197 clearSyncStartEvent(); 6198 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6199 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6200 triggerSession, 6201 recordTrack->sessionId(), 6202 syncStartEventCallback, 6203 this); 6204 // Sync event can be cancelled by the trigger session if the track is not in a 6205 // compatible state in which case we start record immediately 6206 if (mSyncStartEvent->isCancelled()) { 6207 clearSyncStartEvent(); 6208 } else { 6209 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6210 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6211 } 6212 } 6213 6214 { 6215 AutoMutex lock(mLock); 6216 if (mActiveTrack != 0) { 6217 if (recordTrack != mActiveTrack.get()) { 6218 status = -EBUSY; 6219 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6220 mActiveTrack->mState = TrackBase::ACTIVE; 6221 } 6222 return status; 6223 } 6224 6225 recordTrack->mState = TrackBase::IDLE; 6226 mActiveTrack = recordTrack; 6227 mLock.unlock(); 6228 status_t status = AudioSystem::startInput(mId); 6229 mLock.lock(); 6230 if (status != NO_ERROR) { 6231 mActiveTrack.clear(); 6232 clearSyncStartEvent(); 6233 return status; 6234 } 6235 mRsmpInIndex = mFrameCount; 6236 mBytesRead = 0; 6237 if (mResampler != NULL) { 6238 mResampler->reset(); 6239 } 6240 mActiveTrack->mState = TrackBase::RESUMING; 6241 // signal thread to start 6242 ALOGV("Signal record thread"); 6243 mWaitWorkCV.signal(); 6244 // do not wait for mStartStopCond if exiting 6245 if (exitPending()) { 6246 mActiveTrack.clear(); 6247 status = INVALID_OPERATION; 6248 goto startError; 6249 } 6250 mStartStopCond.wait(mLock); 6251 if (mActiveTrack == 0) { 6252 ALOGV("Record failed to start"); 6253 status = BAD_VALUE; 6254 goto startError; 6255 } 6256 ALOGV("Record started OK"); 6257 return status; 6258 } 6259startError: 6260 AudioSystem::stopInput(mId); 6261 clearSyncStartEvent(); 6262 return status; 6263} 6264 6265void AudioFlinger::RecordThread::clearSyncStartEvent() 6266{ 6267 if (mSyncStartEvent != 0) { 6268 mSyncStartEvent->cancel(); 6269 } 6270 mSyncStartEvent.clear(); 6271 mFramestoDrop = 0; 6272} 6273 6274void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6275{ 6276 sp<SyncEvent> strongEvent = event.promote(); 6277 6278 if (strongEvent != 0) { 6279 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6280 me->handleSyncStartEvent(strongEvent); 6281 } 6282} 6283 6284void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6285{ 6286 if (event == mSyncStartEvent) { 6287 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6288 // from audio HAL 6289 mFramestoDrop = mFrameCount * 2; 6290 } 6291} 6292 6293bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6294 ALOGV("RecordThread::stop"); 6295 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6296 return false; 6297 } 6298 recordTrack->mState = TrackBase::PAUSING; 6299 // do not wait for mStartStopCond if exiting 6300 if (exitPending()) { 6301 return true; 6302 } 6303 mStartStopCond.wait(mLock); 6304 // if we have been restarted, recordTrack == mActiveTrack.get() here 6305 if (exitPending() || recordTrack != mActiveTrack.get()) { 6306 ALOGV("Record stopped OK"); 6307 return true; 6308 } 6309 return false; 6310} 6311 6312bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6313{ 6314 return false; 6315} 6316 6317status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6318{ 6319 if (!isValidSyncEvent(event)) { 6320 return BAD_VALUE; 6321 } 6322 6323 Mutex::Autolock _l(mLock); 6324 6325 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6326 mTrack->setSyncEvent(event); 6327 return NO_ERROR; 6328 } 6329 return NAME_NOT_FOUND; 6330} 6331 6332void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6333{ 6334 const size_t SIZE = 256; 6335 char buffer[SIZE]; 6336 String8 result; 6337 6338 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6339 result.append(buffer); 6340 6341 if (mActiveTrack != 0) { 6342 result.append("Active Track:\n"); 6343 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6344 mActiveTrack->dump(buffer, SIZE); 6345 result.append(buffer); 6346 6347 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6348 result.append(buffer); 6349 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6350 result.append(buffer); 6351 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6352 result.append(buffer); 6353 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6354 result.append(buffer); 6355 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6356 result.append(buffer); 6357 6358 6359 } else { 6360 result.append("No record client\n"); 6361 } 6362 write(fd, result.string(), result.size()); 6363 6364 dumpBase(fd, args); 6365 dumpEffectChains(fd, args); 6366} 6367 6368// AudioBufferProvider interface 6369status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6370{ 6371 size_t framesReq = buffer->frameCount; 6372 size_t framesReady = mFrameCount - mRsmpInIndex; 6373 int channelCount; 6374 6375 if (framesReady == 0) { 6376 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6377 if (mBytesRead < 0) { 6378 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6379 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6380 // Force input into standby so that it tries to 6381 // recover at next read attempt 6382 inputStandBy(); 6383 usleep(kRecordThreadSleepUs); 6384 } 6385 buffer->raw = NULL; 6386 buffer->frameCount = 0; 6387 return NOT_ENOUGH_DATA; 6388 } 6389 mRsmpInIndex = 0; 6390 framesReady = mFrameCount; 6391 } 6392 6393 if (framesReq > framesReady) { 6394 framesReq = framesReady; 6395 } 6396 6397 if (mChannelCount == 1 && mReqChannelCount == 2) { 6398 channelCount = 1; 6399 } else { 6400 channelCount = 2; 6401 } 6402 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6403 buffer->frameCount = framesReq; 6404 return NO_ERROR; 6405} 6406 6407// AudioBufferProvider interface 6408void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6409{ 6410 mRsmpInIndex += buffer->frameCount; 6411 buffer->frameCount = 0; 6412} 6413 6414bool AudioFlinger::RecordThread::checkForNewParameters_l() 6415{ 6416 bool reconfig = false; 6417 6418 while (!mNewParameters.isEmpty()) { 6419 status_t status = NO_ERROR; 6420 String8 keyValuePair = mNewParameters[0]; 6421 AudioParameter param = AudioParameter(keyValuePair); 6422 int value; 6423 audio_format_t reqFormat = mFormat; 6424 int reqSamplingRate = mReqSampleRate; 6425 int reqChannelCount = mReqChannelCount; 6426 6427 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6428 reqSamplingRate = value; 6429 reconfig = true; 6430 } 6431 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6432 reqFormat = (audio_format_t) value; 6433 reconfig = true; 6434 } 6435 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6436 reqChannelCount = popcount(value); 6437 reconfig = true; 6438 } 6439 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6440 // do not accept frame count changes if tracks are open as the track buffer 6441 // size depends on frame count and correct behavior would not be guaranteed 6442 // if frame count is changed after track creation 6443 if (mActiveTrack != 0) { 6444 status = INVALID_OPERATION; 6445 } else { 6446 reconfig = true; 6447 } 6448 } 6449 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6450 // forward device change to effects that have requested to be 6451 // aware of attached audio device. 6452 for (size_t i = 0; i < mEffectChains.size(); i++) { 6453 mEffectChains[i]->setDevice_l(value); 6454 } 6455 // store input device and output device but do not forward output device to audio HAL. 6456 // Note that status is ignored by the caller for output device 6457 // (see AudioFlinger::setParameters() 6458 audio_devices_t newDevice = mDevice; 6459 if (value & AUDIO_DEVICE_OUT_ALL) { 6460 newDevice &= ~(value & AUDIO_DEVICE_OUT_ALL); 6461 status = BAD_VALUE; 6462 } else { 6463 newDevice &= ~(value & AUDIO_DEVICE_IN_ALL); 6464 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6465 if (mTrack != NULL) { 6466 bool suspend = audio_is_bluetooth_sco_device( 6467 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6468 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6469 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6470 } 6471 } 6472 newDevice |= value; 6473 mDevice = newDevice; // since mDevice is read by other threads, only write to it once 6474 } 6475 if (status == NO_ERROR) { 6476 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6477 if (status == INVALID_OPERATION) { 6478 inputStandBy(); 6479 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6480 keyValuePair.string()); 6481 } 6482 if (reconfig) { 6483 if (status == BAD_VALUE && 6484 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6485 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6486 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6487 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6488 (reqChannelCount <= FCC_2)) { 6489 status = NO_ERROR; 6490 } 6491 if (status == NO_ERROR) { 6492 readInputParameters(); 6493 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6494 } 6495 } 6496 } 6497 6498 mNewParameters.removeAt(0); 6499 6500 mParamStatus = status; 6501 mParamCond.signal(); 6502 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6503 // already timed out waiting for the status and will never signal the condition. 6504 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6505 } 6506 return reconfig; 6507} 6508 6509String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6510{ 6511 char *s; 6512 String8 out_s8 = String8(); 6513 6514 Mutex::Autolock _l(mLock); 6515 if (initCheck() != NO_ERROR) { 6516 return out_s8; 6517 } 6518 6519 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6520 out_s8 = String8(s); 6521 free(s); 6522 return out_s8; 6523} 6524 6525void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6526 AudioSystem::OutputDescriptor desc; 6527 void *param2 = NULL; 6528 6529 switch (event) { 6530 case AudioSystem::INPUT_OPENED: 6531 case AudioSystem::INPUT_CONFIG_CHANGED: 6532 desc.channels = mChannelMask; 6533 desc.samplingRate = mSampleRate; 6534 desc.format = mFormat; 6535 desc.frameCount = mFrameCount; 6536 desc.latency = 0; 6537 param2 = &desc; 6538 break; 6539 6540 case AudioSystem::INPUT_CLOSED: 6541 default: 6542 break; 6543 } 6544 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6545} 6546 6547void AudioFlinger::RecordThread::readInputParameters() 6548{ 6549 delete mRsmpInBuffer; 6550 // mRsmpInBuffer is always assigned a new[] below 6551 delete mRsmpOutBuffer; 6552 mRsmpOutBuffer = NULL; 6553 delete mResampler; 6554 mResampler = NULL; 6555 6556 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6557 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6558 mChannelCount = (uint16_t)popcount(mChannelMask); 6559 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6560 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6561 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6562 mFrameCount = mInputBytes / mFrameSize; 6563 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6564 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6565 6566 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6567 { 6568 int channelCount; 6569 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6570 // stereo to mono post process as the resampler always outputs stereo. 6571 if (mChannelCount == 1 && mReqChannelCount == 2) { 6572 channelCount = 1; 6573 } else { 6574 channelCount = 2; 6575 } 6576 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6577 mResampler->setSampleRate(mSampleRate); 6578 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6579 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6580 6581 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6582 if (mChannelCount == 1 && mReqChannelCount == 1) { 6583 mFrameCount >>= 1; 6584 } 6585 6586 } 6587 mRsmpInIndex = mFrameCount; 6588} 6589 6590unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6591{ 6592 Mutex::Autolock _l(mLock); 6593 if (initCheck() != NO_ERROR) { 6594 return 0; 6595 } 6596 6597 return mInput->stream->get_input_frames_lost(mInput->stream); 6598} 6599 6600uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6601{ 6602 Mutex::Autolock _l(mLock); 6603 uint32_t result = 0; 6604 if (getEffectChain_l(sessionId) != 0) { 6605 result = EFFECT_SESSION; 6606 } 6607 6608 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6609 result |= TRACK_SESSION; 6610 } 6611 6612 return result; 6613} 6614 6615AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6616{ 6617 Mutex::Autolock _l(mLock); 6618 return mTrack; 6619} 6620 6621AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6622{ 6623 Mutex::Autolock _l(mLock); 6624 AudioStreamIn *input = mInput; 6625 mInput = NULL; 6626 return input; 6627} 6628 6629// this method must always be called either with ThreadBase mLock held or inside the thread loop 6630audio_stream_t* AudioFlinger::RecordThread::stream() const 6631{ 6632 if (mInput == NULL) { 6633 return NULL; 6634 } 6635 return &mInput->stream->common; 6636} 6637 6638 6639// ---------------------------------------------------------------------------- 6640 6641audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6642{ 6643 if (!settingsAllowed()) { 6644 return 0; 6645 } 6646 Mutex::Autolock _l(mLock); 6647 return loadHwModule_l(name); 6648} 6649 6650// loadHwModule_l() must be called with AudioFlinger::mLock held 6651audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6652{ 6653 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6654 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6655 ALOGW("loadHwModule() module %s already loaded", name); 6656 return mAudioHwDevs.keyAt(i); 6657 } 6658 } 6659 6660 audio_hw_device_t *dev; 6661 6662 int rc = load_audio_interface(name, &dev); 6663 if (rc) { 6664 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6665 return 0; 6666 } 6667 6668 mHardwareStatus = AUDIO_HW_INIT; 6669 rc = dev->init_check(dev); 6670 mHardwareStatus = AUDIO_HW_IDLE; 6671 if (rc) { 6672 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6673 return 0; 6674 } 6675 6676 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6677 (NULL != dev->set_master_volume)) { 6678 AutoMutex lock(mHardwareLock); 6679 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6680 dev->set_master_volume(dev, mMasterVolume); 6681 mHardwareStatus = AUDIO_HW_IDLE; 6682 } 6683 6684 audio_module_handle_t handle = nextUniqueId(); 6685 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6686 6687 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6688 name, dev->common.module->name, dev->common.module->id, handle); 6689 6690 return handle; 6691 6692} 6693 6694audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6695 audio_devices_t *pDevices, 6696 uint32_t *pSamplingRate, 6697 audio_format_t *pFormat, 6698 audio_channel_mask_t *pChannelMask, 6699 uint32_t *pLatencyMs, 6700 audio_output_flags_t flags) 6701{ 6702 status_t status; 6703 PlaybackThread *thread = NULL; 6704 struct audio_config config = { 6705 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6706 channel_mask: pChannelMask ? *pChannelMask : 0, 6707 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6708 }; 6709 audio_stream_out_t *outStream = NULL; 6710 audio_hw_device_t *outHwDev; 6711 6712 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6713 module, 6714 (pDevices != NULL) ? *pDevices : 0, 6715 config.sample_rate, 6716 config.format, 6717 config.channel_mask, 6718 flags); 6719 6720 if (pDevices == NULL || *pDevices == 0) { 6721 return 0; 6722 } 6723 6724 Mutex::Autolock _l(mLock); 6725 6726 outHwDev = findSuitableHwDev_l(module, *pDevices); 6727 if (outHwDev == NULL) 6728 return 0; 6729 6730 audio_io_handle_t id = nextUniqueId(); 6731 6732 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6733 6734 status = outHwDev->open_output_stream(outHwDev, 6735 id, 6736 *pDevices, 6737 (audio_output_flags_t)flags, 6738 &config, 6739 &outStream); 6740 6741 mHardwareStatus = AUDIO_HW_IDLE; 6742 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6743 outStream, 6744 config.sample_rate, 6745 config.format, 6746 config.channel_mask, 6747 status); 6748 6749 if (status == NO_ERROR && outStream != NULL) { 6750 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6751 6752 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6753 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6754 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6755 thread = new DirectOutputThread(this, output, id, *pDevices); 6756 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6757 } else { 6758 thread = new MixerThread(this, output, id, *pDevices); 6759 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6760 } 6761 mPlaybackThreads.add(id, thread); 6762 6763 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6764 if (pFormat != NULL) *pFormat = config.format; 6765 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6766 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6767 6768 // notify client processes of the new output creation 6769 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6770 6771 // the first primary output opened designates the primary hw device 6772 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6773 ALOGI("Using module %d has the primary audio interface", module); 6774 mPrimaryHardwareDev = outHwDev; 6775 6776 AutoMutex lock(mHardwareLock); 6777 mHardwareStatus = AUDIO_HW_SET_MODE; 6778 outHwDev->set_mode(outHwDev, mMode); 6779 6780 // Determine the level of master volume support the primary audio HAL has, 6781 // and set the initial master volume at the same time. 6782 float initialVolume = 1.0; 6783 mMasterVolumeSupportLvl = MVS_NONE; 6784 6785 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6786 if ((NULL != outHwDev->get_master_volume) && 6787 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6788 mMasterVolumeSupportLvl = MVS_FULL; 6789 } else { 6790 mMasterVolumeSupportLvl = MVS_SETONLY; 6791 initialVolume = 1.0; 6792 } 6793 6794 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6795 if ((NULL == outHwDev->set_master_volume) || 6796 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6797 mMasterVolumeSupportLvl = MVS_NONE; 6798 } 6799 // now that we have a primary device, initialize master volume on other devices 6800 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6801 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6802 6803 if ((dev != mPrimaryHardwareDev) && 6804 (NULL != dev->set_master_volume)) { 6805 dev->set_master_volume(dev, initialVolume); 6806 } 6807 } 6808 mHardwareStatus = AUDIO_HW_IDLE; 6809 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6810 ? initialVolume 6811 : 1.0; 6812 mMasterVolume = initialVolume; 6813 } 6814 return id; 6815 } 6816 6817 return 0; 6818} 6819 6820audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6821 audio_io_handle_t output2) 6822{ 6823 Mutex::Autolock _l(mLock); 6824 MixerThread *thread1 = checkMixerThread_l(output1); 6825 MixerThread *thread2 = checkMixerThread_l(output2); 6826 6827 if (thread1 == NULL || thread2 == NULL) { 6828 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6829 return 0; 6830 } 6831 6832 audio_io_handle_t id = nextUniqueId(); 6833 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6834 thread->addOutputTrack(thread2); 6835 mPlaybackThreads.add(id, thread); 6836 // notify client processes of the new output creation 6837 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6838 return id; 6839} 6840 6841status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6842{ 6843 return closeOutput_nonvirtual(output); 6844} 6845 6846status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6847{ 6848 // keep strong reference on the playback thread so that 6849 // it is not destroyed while exit() is executed 6850 sp<PlaybackThread> thread; 6851 { 6852 Mutex::Autolock _l(mLock); 6853 thread = checkPlaybackThread_l(output); 6854 if (thread == NULL) { 6855 return BAD_VALUE; 6856 } 6857 6858 ALOGV("closeOutput() %d", output); 6859 6860 if (thread->type() == ThreadBase::MIXER) { 6861 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6862 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6863 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6864 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6865 } 6866 } 6867 } 6868 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6869 mPlaybackThreads.removeItem(output); 6870 } 6871 thread->exit(); 6872 // The thread entity (active unit of execution) is no longer running here, 6873 // but the ThreadBase container still exists. 6874 6875 if (thread->type() != ThreadBase::DUPLICATING) { 6876 AudioStreamOut *out = thread->clearOutput(); 6877 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6878 // from now on thread->mOutput is NULL 6879 out->hwDev->close_output_stream(out->hwDev, out->stream); 6880 delete out; 6881 } 6882 return NO_ERROR; 6883} 6884 6885status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6886{ 6887 Mutex::Autolock _l(mLock); 6888 PlaybackThread *thread = checkPlaybackThread_l(output); 6889 6890 if (thread == NULL) { 6891 return BAD_VALUE; 6892 } 6893 6894 ALOGV("suspendOutput() %d", output); 6895 thread->suspend(); 6896 6897 return NO_ERROR; 6898} 6899 6900status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6901{ 6902 Mutex::Autolock _l(mLock); 6903 PlaybackThread *thread = checkPlaybackThread_l(output); 6904 6905 if (thread == NULL) { 6906 return BAD_VALUE; 6907 } 6908 6909 ALOGV("restoreOutput() %d", output); 6910 6911 thread->restore(); 6912 6913 return NO_ERROR; 6914} 6915 6916audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6917 audio_devices_t *pDevices, 6918 uint32_t *pSamplingRate, 6919 audio_format_t *pFormat, 6920 audio_channel_mask_t *pChannelMask) 6921{ 6922 status_t status; 6923 RecordThread *thread = NULL; 6924 struct audio_config config = { 6925 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6926 channel_mask: pChannelMask ? *pChannelMask : 0, 6927 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6928 }; 6929 uint32_t reqSamplingRate = config.sample_rate; 6930 audio_format_t reqFormat = config.format; 6931 audio_channel_mask_t reqChannels = config.channel_mask; 6932 audio_stream_in_t *inStream = NULL; 6933 audio_hw_device_t *inHwDev; 6934 6935 if (pDevices == NULL || *pDevices == 0) { 6936 return 0; 6937 } 6938 6939 Mutex::Autolock _l(mLock); 6940 6941 inHwDev = findSuitableHwDev_l(module, *pDevices); 6942 if (inHwDev == NULL) 6943 return 0; 6944 6945 audio_io_handle_t id = nextUniqueId(); 6946 6947 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6948 &inStream); 6949 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6950 inStream, 6951 config.sample_rate, 6952 config.format, 6953 config.channel_mask, 6954 status); 6955 6956 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6957 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6958 // or stereo to mono conversions on 16 bit PCM inputs. 6959 if (status == BAD_VALUE && 6960 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6961 (config.sample_rate <= 2 * reqSamplingRate) && 6962 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6963 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 6964 inStream = NULL; 6965 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6966 } 6967 6968 if (status == NO_ERROR && inStream != NULL) { 6969 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6970 6971 // Start record thread 6972 // RecorThread require both input and output device indication to forward to audio 6973 // pre processing modules 6974 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 6975 thread = new RecordThread(this, 6976 input, 6977 reqSamplingRate, 6978 reqChannels, 6979 id, 6980 device); 6981 mRecordThreads.add(id, thread); 6982 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6983 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6984 if (pFormat != NULL) *pFormat = config.format; 6985 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6986 6987 // notify client processes of the new input creation 6988 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6989 return id; 6990 } 6991 6992 return 0; 6993} 6994 6995status_t AudioFlinger::closeInput(audio_io_handle_t input) 6996{ 6997 return closeInput_nonvirtual(input); 6998} 6999 7000status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7001{ 7002 // keep strong reference on the record thread so that 7003 // it is not destroyed while exit() is executed 7004 sp<RecordThread> thread; 7005 { 7006 Mutex::Autolock _l(mLock); 7007 thread = checkRecordThread_l(input); 7008 if (thread == 0) { 7009 return BAD_VALUE; 7010 } 7011 7012 ALOGV("closeInput() %d", input); 7013 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7014 mRecordThreads.removeItem(input); 7015 } 7016 thread->exit(); 7017 // The thread entity (active unit of execution) is no longer running here, 7018 // but the ThreadBase container still exists. 7019 7020 AudioStreamIn *in = thread->clearInput(); 7021 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7022 // from now on thread->mInput is NULL 7023 in->hwDev->close_input_stream(in->hwDev, in->stream); 7024 delete in; 7025 7026 return NO_ERROR; 7027} 7028 7029status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7030{ 7031 Mutex::Autolock _l(mLock); 7032 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7033 7034 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7035 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7036 thread->invalidateTracks(stream); 7037 } 7038 7039 return NO_ERROR; 7040} 7041 7042 7043int AudioFlinger::newAudioSessionId() 7044{ 7045 return nextUniqueId(); 7046} 7047 7048void AudioFlinger::acquireAudioSessionId(int audioSession) 7049{ 7050 Mutex::Autolock _l(mLock); 7051 pid_t caller = IPCThreadState::self()->getCallingPid(); 7052 ALOGV("acquiring %d from %d", audioSession, caller); 7053 size_t num = mAudioSessionRefs.size(); 7054 for (size_t i = 0; i< num; i++) { 7055 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7056 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7057 ref->mCnt++; 7058 ALOGV(" incremented refcount to %d", ref->mCnt); 7059 return; 7060 } 7061 } 7062 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7063 ALOGV(" added new entry for %d", audioSession); 7064} 7065 7066void AudioFlinger::releaseAudioSessionId(int audioSession) 7067{ 7068 Mutex::Autolock _l(mLock); 7069 pid_t caller = IPCThreadState::self()->getCallingPid(); 7070 ALOGV("releasing %d from %d", audioSession, caller); 7071 size_t num = mAudioSessionRefs.size(); 7072 for (size_t i = 0; i< num; i++) { 7073 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7074 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7075 ref->mCnt--; 7076 ALOGV(" decremented refcount to %d", ref->mCnt); 7077 if (ref->mCnt == 0) { 7078 mAudioSessionRefs.removeAt(i); 7079 delete ref; 7080 purgeStaleEffects_l(); 7081 } 7082 return; 7083 } 7084 } 7085 ALOGW("session id %d not found for pid %d", audioSession, caller); 7086} 7087 7088void AudioFlinger::purgeStaleEffects_l() { 7089 7090 ALOGV("purging stale effects"); 7091 7092 Vector< sp<EffectChain> > chains; 7093 7094 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7095 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7096 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7097 sp<EffectChain> ec = t->mEffectChains[j]; 7098 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7099 chains.push(ec); 7100 } 7101 } 7102 } 7103 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7104 sp<RecordThread> t = mRecordThreads.valueAt(i); 7105 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7106 sp<EffectChain> ec = t->mEffectChains[j]; 7107 chains.push(ec); 7108 } 7109 } 7110 7111 for (size_t i = 0; i < chains.size(); i++) { 7112 sp<EffectChain> ec = chains[i]; 7113 int sessionid = ec->sessionId(); 7114 sp<ThreadBase> t = ec->mThread.promote(); 7115 if (t == 0) { 7116 continue; 7117 } 7118 size_t numsessionrefs = mAudioSessionRefs.size(); 7119 bool found = false; 7120 for (size_t k = 0; k < numsessionrefs; k++) { 7121 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7122 if (ref->mSessionid == sessionid) { 7123 ALOGV(" session %d still exists for %d with %d refs", 7124 sessionid, ref->mPid, ref->mCnt); 7125 found = true; 7126 break; 7127 } 7128 } 7129 if (!found) { 7130 Mutex::Autolock _l (t->mLock); 7131 // remove all effects from the chain 7132 while (ec->mEffects.size()) { 7133 sp<EffectModule> effect = ec->mEffects[0]; 7134 effect->unPin(); 7135 t->removeEffect_l(effect); 7136 if (effect->purgeHandles()) { 7137 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7138 } 7139 AudioSystem::unregisterEffect(effect->id()); 7140 } 7141 } 7142 } 7143 return; 7144} 7145 7146// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7147AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7148{ 7149 return mPlaybackThreads.valueFor(output).get(); 7150} 7151 7152// checkMixerThread_l() must be called with AudioFlinger::mLock held 7153AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7154{ 7155 PlaybackThread *thread = checkPlaybackThread_l(output); 7156 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7157} 7158 7159// checkRecordThread_l() must be called with AudioFlinger::mLock held 7160AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7161{ 7162 return mRecordThreads.valueFor(input).get(); 7163} 7164 7165uint32_t AudioFlinger::nextUniqueId() 7166{ 7167 return android_atomic_inc(&mNextUniqueId); 7168} 7169 7170AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7171{ 7172 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7173 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7174 AudioStreamOut *output = thread->getOutput(); 7175 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7176 return thread; 7177 } 7178 } 7179 return NULL; 7180} 7181 7182audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7183{ 7184 PlaybackThread *thread = primaryPlaybackThread_l(); 7185 7186 if (thread == NULL) { 7187 return 0; 7188 } 7189 7190 return thread->device(); 7191} 7192 7193sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7194 int triggerSession, 7195 int listenerSession, 7196 sync_event_callback_t callBack, 7197 void *cookie) 7198{ 7199 Mutex::Autolock _l(mLock); 7200 7201 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7202 status_t playStatus = NAME_NOT_FOUND; 7203 status_t recStatus = NAME_NOT_FOUND; 7204 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7205 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7206 if (playStatus == NO_ERROR) { 7207 return event; 7208 } 7209 } 7210 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7211 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7212 if (recStatus == NO_ERROR) { 7213 return event; 7214 } 7215 } 7216 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7217 mPendingSyncEvents.add(event); 7218 } else { 7219 ALOGV("createSyncEvent() invalid event %d", event->type()); 7220 event.clear(); 7221 } 7222 return event; 7223} 7224 7225// ---------------------------------------------------------------------------- 7226// Effect management 7227// ---------------------------------------------------------------------------- 7228 7229 7230status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7231{ 7232 Mutex::Autolock _l(mLock); 7233 return EffectQueryNumberEffects(numEffects); 7234} 7235 7236status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7237{ 7238 Mutex::Autolock _l(mLock); 7239 return EffectQueryEffect(index, descriptor); 7240} 7241 7242status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7243 effect_descriptor_t *descriptor) const 7244{ 7245 Mutex::Autolock _l(mLock); 7246 return EffectGetDescriptor(pUuid, descriptor); 7247} 7248 7249 7250sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7251 effect_descriptor_t *pDesc, 7252 const sp<IEffectClient>& effectClient, 7253 int32_t priority, 7254 audio_io_handle_t io, 7255 int sessionId, 7256 status_t *status, 7257 int *id, 7258 int *enabled) 7259{ 7260 status_t lStatus = NO_ERROR; 7261 sp<EffectHandle> handle; 7262 effect_descriptor_t desc; 7263 7264 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7265 pid, effectClient.get(), priority, sessionId, io); 7266 7267 if (pDesc == NULL) { 7268 lStatus = BAD_VALUE; 7269 goto Exit; 7270 } 7271 7272 // check audio settings permission for global effects 7273 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7274 lStatus = PERMISSION_DENIED; 7275 goto Exit; 7276 } 7277 7278 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7279 // that can only be created by audio policy manager (running in same process) 7280 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7281 lStatus = PERMISSION_DENIED; 7282 goto Exit; 7283 } 7284 7285 if (io == 0) { 7286 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7287 // output must be specified by AudioPolicyManager when using session 7288 // AUDIO_SESSION_OUTPUT_STAGE 7289 lStatus = BAD_VALUE; 7290 goto Exit; 7291 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7292 // if the output returned by getOutputForEffect() is removed before we lock the 7293 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7294 // and we will exit safely 7295 io = AudioSystem::getOutputForEffect(&desc); 7296 } 7297 } 7298 7299 { 7300 Mutex::Autolock _l(mLock); 7301 7302 7303 if (!EffectIsNullUuid(&pDesc->uuid)) { 7304 // if uuid is specified, request effect descriptor 7305 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7306 if (lStatus < 0) { 7307 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7308 goto Exit; 7309 } 7310 } else { 7311 // if uuid is not specified, look for an available implementation 7312 // of the required type in effect factory 7313 if (EffectIsNullUuid(&pDesc->type)) { 7314 ALOGW("createEffect() no effect type"); 7315 lStatus = BAD_VALUE; 7316 goto Exit; 7317 } 7318 uint32_t numEffects = 0; 7319 effect_descriptor_t d; 7320 d.flags = 0; // prevent compiler warning 7321 bool found = false; 7322 7323 lStatus = EffectQueryNumberEffects(&numEffects); 7324 if (lStatus < 0) { 7325 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7326 goto Exit; 7327 } 7328 for (uint32_t i = 0; i < numEffects; i++) { 7329 lStatus = EffectQueryEffect(i, &desc); 7330 if (lStatus < 0) { 7331 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7332 continue; 7333 } 7334 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7335 // If matching type found save effect descriptor. If the session is 7336 // 0 and the effect is not auxiliary, continue enumeration in case 7337 // an auxiliary version of this effect type is available 7338 found = true; 7339 d = desc; 7340 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7341 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7342 break; 7343 } 7344 } 7345 } 7346 if (!found) { 7347 lStatus = BAD_VALUE; 7348 ALOGW("createEffect() effect not found"); 7349 goto Exit; 7350 } 7351 // For same effect type, chose auxiliary version over insert version if 7352 // connect to output mix (Compliance to OpenSL ES) 7353 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7354 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7355 desc = d; 7356 } 7357 } 7358 7359 // Do not allow auxiliary effects on a session different from 0 (output mix) 7360 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7361 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7362 lStatus = INVALID_OPERATION; 7363 goto Exit; 7364 } 7365 7366 // check recording permission for visualizer 7367 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7368 !recordingAllowed()) { 7369 lStatus = PERMISSION_DENIED; 7370 goto Exit; 7371 } 7372 7373 // return effect descriptor 7374 *pDesc = desc; 7375 7376 // If output is not specified try to find a matching audio session ID in one of the 7377 // output threads. 7378 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7379 // because of code checking output when entering the function. 7380 // Note: io is never 0 when creating an effect on an input 7381 if (io == 0) { 7382 // look for the thread where the specified audio session is present 7383 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7384 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7385 io = mPlaybackThreads.keyAt(i); 7386 break; 7387 } 7388 } 7389 if (io == 0) { 7390 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7391 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7392 io = mRecordThreads.keyAt(i); 7393 break; 7394 } 7395 } 7396 } 7397 // If no output thread contains the requested session ID, default to 7398 // first output. The effect chain will be moved to the correct output 7399 // thread when a track with the same session ID is created 7400 if (io == 0 && mPlaybackThreads.size()) { 7401 io = mPlaybackThreads.keyAt(0); 7402 } 7403 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7404 } 7405 ThreadBase *thread = checkRecordThread_l(io); 7406 if (thread == NULL) { 7407 thread = checkPlaybackThread_l(io); 7408 if (thread == NULL) { 7409 ALOGE("createEffect() unknown output thread"); 7410 lStatus = BAD_VALUE; 7411 goto Exit; 7412 } 7413 } 7414 7415 sp<Client> client = registerPid_l(pid); 7416 7417 // create effect on selected output thread 7418 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7419 &desc, enabled, &lStatus); 7420 if (handle != 0 && id != NULL) { 7421 *id = handle->id(); 7422 } 7423 } 7424 7425Exit: 7426 if (status != NULL) { 7427 *status = lStatus; 7428 } 7429 return handle; 7430} 7431 7432status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7433 audio_io_handle_t dstOutput) 7434{ 7435 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7436 sessionId, srcOutput, dstOutput); 7437 Mutex::Autolock _l(mLock); 7438 if (srcOutput == dstOutput) { 7439 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7440 return NO_ERROR; 7441 } 7442 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7443 if (srcThread == NULL) { 7444 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7445 return BAD_VALUE; 7446 } 7447 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7448 if (dstThread == NULL) { 7449 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7450 return BAD_VALUE; 7451 } 7452 7453 Mutex::Autolock _dl(dstThread->mLock); 7454 Mutex::Autolock _sl(srcThread->mLock); 7455 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7456 7457 return NO_ERROR; 7458} 7459 7460// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7461status_t AudioFlinger::moveEffectChain_l(int sessionId, 7462 AudioFlinger::PlaybackThread *srcThread, 7463 AudioFlinger::PlaybackThread *dstThread, 7464 bool reRegister) 7465{ 7466 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7467 sessionId, srcThread, dstThread); 7468 7469 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7470 if (chain == 0) { 7471 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7472 sessionId, srcThread); 7473 return INVALID_OPERATION; 7474 } 7475 7476 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7477 // so that a new chain is created with correct parameters when first effect is added. This is 7478 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7479 // removed. 7480 srcThread->removeEffectChain_l(chain); 7481 7482 // transfer all effects one by one so that new effect chain is created on new thread with 7483 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7484 audio_io_handle_t dstOutput = dstThread->id(); 7485 sp<EffectChain> dstChain; 7486 uint32_t strategy = 0; // prevent compiler warning 7487 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7488 while (effect != 0) { 7489 srcThread->removeEffect_l(effect); 7490 dstThread->addEffect_l(effect); 7491 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7492 if (effect->state() == EffectModule::ACTIVE || 7493 effect->state() == EffectModule::STOPPING) { 7494 effect->start(); 7495 } 7496 // if the move request is not received from audio policy manager, the effect must be 7497 // re-registered with the new strategy and output 7498 if (dstChain == 0) { 7499 dstChain = effect->chain().promote(); 7500 if (dstChain == 0) { 7501 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7502 srcThread->addEffect_l(effect); 7503 return NO_INIT; 7504 } 7505 strategy = dstChain->strategy(); 7506 } 7507 if (reRegister) { 7508 AudioSystem::unregisterEffect(effect->id()); 7509 AudioSystem::registerEffect(&effect->desc(), 7510 dstOutput, 7511 strategy, 7512 sessionId, 7513 effect->id()); 7514 } 7515 effect = chain->getEffectFromId_l(0); 7516 } 7517 7518 return NO_ERROR; 7519} 7520 7521 7522// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7523sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7524 const sp<AudioFlinger::Client>& client, 7525 const sp<IEffectClient>& effectClient, 7526 int32_t priority, 7527 int sessionId, 7528 effect_descriptor_t *desc, 7529 int *enabled, 7530 status_t *status 7531 ) 7532{ 7533 sp<EffectModule> effect; 7534 sp<EffectHandle> handle; 7535 status_t lStatus; 7536 sp<EffectChain> chain; 7537 bool chainCreated = false; 7538 bool effectCreated = false; 7539 bool effectRegistered = false; 7540 7541 lStatus = initCheck(); 7542 if (lStatus != NO_ERROR) { 7543 ALOGW("createEffect_l() Audio driver not initialized."); 7544 goto Exit; 7545 } 7546 7547 // Do not allow effects with session ID 0 on direct output or duplicating threads 7548 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7549 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7550 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7551 desc->name, sessionId); 7552 lStatus = BAD_VALUE; 7553 goto Exit; 7554 } 7555 // Only Pre processor effects are allowed on input threads and only on input threads 7556 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7557 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7558 desc->name, desc->flags, mType); 7559 lStatus = BAD_VALUE; 7560 goto Exit; 7561 } 7562 7563 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7564 7565 { // scope for mLock 7566 Mutex::Autolock _l(mLock); 7567 7568 // check for existing effect chain with the requested audio session 7569 chain = getEffectChain_l(sessionId); 7570 if (chain == 0) { 7571 // create a new chain for this session 7572 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7573 chain = new EffectChain(this, sessionId); 7574 addEffectChain_l(chain); 7575 chain->setStrategy(getStrategyForSession_l(sessionId)); 7576 chainCreated = true; 7577 } else { 7578 effect = chain->getEffectFromDesc_l(desc); 7579 } 7580 7581 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7582 7583 if (effect == 0) { 7584 int id = mAudioFlinger->nextUniqueId(); 7585 // Check CPU and memory usage 7586 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7587 if (lStatus != NO_ERROR) { 7588 goto Exit; 7589 } 7590 effectRegistered = true; 7591 // create a new effect module if none present in the chain 7592 effect = new EffectModule(this, chain, desc, id, sessionId); 7593 lStatus = effect->status(); 7594 if (lStatus != NO_ERROR) { 7595 goto Exit; 7596 } 7597 lStatus = chain->addEffect_l(effect); 7598 if (lStatus != NO_ERROR) { 7599 goto Exit; 7600 } 7601 effectCreated = true; 7602 7603 effect->setDevice(mDevice); 7604 effect->setMode(mAudioFlinger->getMode()); 7605 } 7606 // create effect handle and connect it to effect module 7607 handle = new EffectHandle(effect, client, effectClient, priority); 7608 lStatus = effect->addHandle(handle.get()); 7609 if (enabled != NULL) { 7610 *enabled = (int)effect->isEnabled(); 7611 } 7612 } 7613 7614Exit: 7615 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7616 Mutex::Autolock _l(mLock); 7617 if (effectCreated) { 7618 chain->removeEffect_l(effect); 7619 } 7620 if (effectRegistered) { 7621 AudioSystem::unregisterEffect(effect->id()); 7622 } 7623 if (chainCreated) { 7624 removeEffectChain_l(chain); 7625 } 7626 handle.clear(); 7627 } 7628 7629 if (status != NULL) { 7630 *status = lStatus; 7631 } 7632 return handle; 7633} 7634 7635sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7636{ 7637 Mutex::Autolock _l(mLock); 7638 return getEffect_l(sessionId, effectId); 7639} 7640 7641sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7642{ 7643 sp<EffectChain> chain = getEffectChain_l(sessionId); 7644 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7645} 7646 7647// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7648// PlaybackThread::mLock held 7649status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7650{ 7651 // check for existing effect chain with the requested audio session 7652 int sessionId = effect->sessionId(); 7653 sp<EffectChain> chain = getEffectChain_l(sessionId); 7654 bool chainCreated = false; 7655 7656 if (chain == 0) { 7657 // create a new chain for this session 7658 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7659 chain = new EffectChain(this, sessionId); 7660 addEffectChain_l(chain); 7661 chain->setStrategy(getStrategyForSession_l(sessionId)); 7662 chainCreated = true; 7663 } 7664 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7665 7666 if (chain->getEffectFromId_l(effect->id()) != 0) { 7667 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7668 this, effect->desc().name, chain.get()); 7669 return BAD_VALUE; 7670 } 7671 7672 status_t status = chain->addEffect_l(effect); 7673 if (status != NO_ERROR) { 7674 if (chainCreated) { 7675 removeEffectChain_l(chain); 7676 } 7677 return status; 7678 } 7679 7680 effect->setDevice(mDevice); 7681 effect->setMode(mAudioFlinger->getMode()); 7682 return NO_ERROR; 7683} 7684 7685void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7686 7687 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7688 effect_descriptor_t desc = effect->desc(); 7689 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7690 detachAuxEffect_l(effect->id()); 7691 } 7692 7693 sp<EffectChain> chain = effect->chain().promote(); 7694 if (chain != 0) { 7695 // remove effect chain if removing last effect 7696 if (chain->removeEffect_l(effect) == 0) { 7697 removeEffectChain_l(chain); 7698 } 7699 } else { 7700 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7701 } 7702} 7703 7704void AudioFlinger::ThreadBase::lockEffectChains_l( 7705 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7706{ 7707 effectChains = mEffectChains; 7708 for (size_t i = 0; i < mEffectChains.size(); i++) { 7709 mEffectChains[i]->lock(); 7710 } 7711} 7712 7713void AudioFlinger::ThreadBase::unlockEffectChains( 7714 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7715{ 7716 for (size_t i = 0; i < effectChains.size(); i++) { 7717 effectChains[i]->unlock(); 7718 } 7719} 7720 7721sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7722{ 7723 Mutex::Autolock _l(mLock); 7724 return getEffectChain_l(sessionId); 7725} 7726 7727sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7728{ 7729 size_t size = mEffectChains.size(); 7730 for (size_t i = 0; i < size; i++) { 7731 if (mEffectChains[i]->sessionId() == sessionId) { 7732 return mEffectChains[i]; 7733 } 7734 } 7735 return 0; 7736} 7737 7738void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7739{ 7740 Mutex::Autolock _l(mLock); 7741 size_t size = mEffectChains.size(); 7742 for (size_t i = 0; i < size; i++) { 7743 mEffectChains[i]->setMode_l(mode); 7744 } 7745} 7746 7747void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7748 EffectHandle *handle, 7749 bool unpinIfLast) { 7750 7751 Mutex::Autolock _l(mLock); 7752 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7753 // delete the effect module if removing last handle on it 7754 if (effect->removeHandle(handle) == 0) { 7755 if (!effect->isPinned() || unpinIfLast) { 7756 removeEffect_l(effect); 7757 AudioSystem::unregisterEffect(effect->id()); 7758 } 7759 } 7760} 7761 7762status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7763{ 7764 int session = chain->sessionId(); 7765 int16_t *buffer = mMixBuffer; 7766 bool ownsBuffer = false; 7767 7768 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7769 if (session > 0) { 7770 // Only one effect chain can be present in direct output thread and it uses 7771 // the mix buffer as input 7772 if (mType != DIRECT) { 7773 size_t numSamples = mNormalFrameCount * mChannelCount; 7774 buffer = new int16_t[numSamples]; 7775 memset(buffer, 0, numSamples * sizeof(int16_t)); 7776 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7777 ownsBuffer = true; 7778 } 7779 7780 // Attach all tracks with same session ID to this chain. 7781 for (size_t i = 0; i < mTracks.size(); ++i) { 7782 sp<Track> track = mTracks[i]; 7783 if (session == track->sessionId()) { 7784 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7785 track->setMainBuffer(buffer); 7786 chain->incTrackCnt(); 7787 } 7788 } 7789 7790 // indicate all active tracks in the chain 7791 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7792 sp<Track> track = mActiveTracks[i].promote(); 7793 if (track == 0) continue; 7794 if (session == track->sessionId()) { 7795 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7796 chain->incActiveTrackCnt(); 7797 } 7798 } 7799 } 7800 7801 chain->setInBuffer(buffer, ownsBuffer); 7802 chain->setOutBuffer(mMixBuffer); 7803 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7804 // chains list in order to be processed last as it contains output stage effects 7805 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7806 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7807 // after track specific effects and before output stage 7808 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7809 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7810 // Effect chain for other sessions are inserted at beginning of effect 7811 // chains list to be processed before output mix effects. Relative order between other 7812 // sessions is not important 7813 size_t size = mEffectChains.size(); 7814 size_t i = 0; 7815 for (i = 0; i < size; i++) { 7816 if (mEffectChains[i]->sessionId() < session) break; 7817 } 7818 mEffectChains.insertAt(chain, i); 7819 checkSuspendOnAddEffectChain_l(chain); 7820 7821 return NO_ERROR; 7822} 7823 7824size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7825{ 7826 int session = chain->sessionId(); 7827 7828 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7829 7830 for (size_t i = 0; i < mEffectChains.size(); i++) { 7831 if (chain == mEffectChains[i]) { 7832 mEffectChains.removeAt(i); 7833 // detach all active tracks from the chain 7834 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7835 sp<Track> track = mActiveTracks[i].promote(); 7836 if (track == 0) continue; 7837 if (session == track->sessionId()) { 7838 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7839 chain.get(), session); 7840 chain->decActiveTrackCnt(); 7841 } 7842 } 7843 7844 // detach all tracks with same session ID from this chain 7845 for (size_t i = 0; i < mTracks.size(); ++i) { 7846 sp<Track> track = mTracks[i]; 7847 if (session == track->sessionId()) { 7848 track->setMainBuffer(mMixBuffer); 7849 chain->decTrackCnt(); 7850 } 7851 } 7852 break; 7853 } 7854 } 7855 return mEffectChains.size(); 7856} 7857 7858status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7859 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7860{ 7861 Mutex::Autolock _l(mLock); 7862 return attachAuxEffect_l(track, EffectId); 7863} 7864 7865status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7866 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7867{ 7868 status_t status = NO_ERROR; 7869 7870 if (EffectId == 0) { 7871 track->setAuxBuffer(0, NULL); 7872 } else { 7873 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7874 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7875 if (effect != 0) { 7876 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7877 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7878 } else { 7879 status = INVALID_OPERATION; 7880 } 7881 } else { 7882 status = BAD_VALUE; 7883 } 7884 } 7885 return status; 7886} 7887 7888void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7889{ 7890 for (size_t i = 0; i < mTracks.size(); ++i) { 7891 sp<Track> track = mTracks[i]; 7892 if (track->auxEffectId() == effectId) { 7893 attachAuxEffect_l(track, 0); 7894 } 7895 } 7896} 7897 7898status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7899{ 7900 // only one chain per input thread 7901 if (mEffectChains.size() != 0) { 7902 return INVALID_OPERATION; 7903 } 7904 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7905 7906 chain->setInBuffer(NULL); 7907 chain->setOutBuffer(NULL); 7908 7909 checkSuspendOnAddEffectChain_l(chain); 7910 7911 mEffectChains.add(chain); 7912 7913 return NO_ERROR; 7914} 7915 7916size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7917{ 7918 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7919 ALOGW_IF(mEffectChains.size() != 1, 7920 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7921 chain.get(), mEffectChains.size(), this); 7922 if (mEffectChains.size() == 1) { 7923 mEffectChains.removeAt(0); 7924 } 7925 return 0; 7926} 7927 7928// ---------------------------------------------------------------------------- 7929// EffectModule implementation 7930// ---------------------------------------------------------------------------- 7931 7932#undef LOG_TAG 7933#define LOG_TAG "AudioFlinger::EffectModule" 7934 7935AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7936 const wp<AudioFlinger::EffectChain>& chain, 7937 effect_descriptor_t *desc, 7938 int id, 7939 int sessionId) 7940 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7941 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7942 mDescriptor(*desc), 7943 // mConfig is set by configure() and not used before then 7944 mEffectInterface(NULL), 7945 mStatus(NO_INIT), mState(IDLE), 7946 // mMaxDisableWaitCnt is set by configure() and not used before then 7947 // mDisableWaitCnt is set by process() and updateState() and not used before then 7948 mSuspended(false) 7949{ 7950 ALOGV("Constructor %p", this); 7951 int lStatus; 7952 7953 // create effect engine from effect factory 7954 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7955 7956 if (mStatus != NO_ERROR) { 7957 return; 7958 } 7959 lStatus = init(); 7960 if (lStatus < 0) { 7961 mStatus = lStatus; 7962 goto Error; 7963 } 7964 7965 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7966 return; 7967Error: 7968 EffectRelease(mEffectInterface); 7969 mEffectInterface = NULL; 7970 ALOGV("Constructor Error %d", mStatus); 7971} 7972 7973AudioFlinger::EffectModule::~EffectModule() 7974{ 7975 ALOGV("Destructor %p", this); 7976 if (mEffectInterface != NULL) { 7977 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7978 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7979 sp<ThreadBase> thread = mThread.promote(); 7980 if (thread != 0) { 7981 audio_stream_t *stream = thread->stream(); 7982 if (stream != NULL) { 7983 stream->remove_audio_effect(stream, mEffectInterface); 7984 } 7985 } 7986 } 7987 // release effect engine 7988 EffectRelease(mEffectInterface); 7989 } 7990} 7991 7992status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 7993{ 7994 status_t status; 7995 7996 Mutex::Autolock _l(mLock); 7997 int priority = handle->priority(); 7998 size_t size = mHandles.size(); 7999 EffectHandle *controlHandle = NULL; 8000 size_t i; 8001 for (i = 0; i < size; i++) { 8002 EffectHandle *h = mHandles[i]; 8003 if (h == NULL || h->destroyed_l()) continue; 8004 // first non destroyed handle is considered in control 8005 if (controlHandle == NULL) 8006 controlHandle = h; 8007 if (h->priority() <= priority) break; 8008 } 8009 // if inserted in first place, move effect control from previous owner to this handle 8010 if (i == 0) { 8011 bool enabled = false; 8012 if (controlHandle != NULL) { 8013 enabled = controlHandle->enabled(); 8014 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8015 } 8016 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8017 status = NO_ERROR; 8018 } else { 8019 status = ALREADY_EXISTS; 8020 } 8021 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8022 mHandles.insertAt(handle, i); 8023 return status; 8024} 8025 8026size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8027{ 8028 Mutex::Autolock _l(mLock); 8029 size_t size = mHandles.size(); 8030 size_t i; 8031 for (i = 0; i < size; i++) { 8032 if (mHandles[i] == handle) break; 8033 } 8034 if (i == size) { 8035 return size; 8036 } 8037 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8038 8039 mHandles.removeAt(i); 8040 // if removed from first place, move effect control from this handle to next in line 8041 if (i == 0) { 8042 EffectHandle *h = controlHandle_l(); 8043 if (h != NULL) { 8044 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8045 } 8046 } 8047 8048 // Prevent calls to process() and other functions on effect interface from now on. 8049 // The effect engine will be released by the destructor when the last strong reference on 8050 // this object is released which can happen after next process is called. 8051 if (mHandles.size() == 0 && !mPinned) { 8052 mState = DESTROYED; 8053 } 8054 8055 return mHandles.size(); 8056} 8057 8058// must be called with EffectModule::mLock held 8059AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8060{ 8061 // the first valid handle in the list has control over the module 8062 for (size_t i = 0; i < mHandles.size(); i++) { 8063 EffectHandle *h = mHandles[i]; 8064 if (h != NULL && !h->destroyed_l()) { 8065 return h; 8066 } 8067 } 8068 8069 return NULL; 8070} 8071 8072size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8073{ 8074 ALOGV("disconnect() %p handle %p", this, handle); 8075 // keep a strong reference on this EffectModule to avoid calling the 8076 // destructor before we exit 8077 sp<EffectModule> keep(this); 8078 { 8079 sp<ThreadBase> thread = mThread.promote(); 8080 if (thread != 0) { 8081 thread->disconnectEffect(keep, handle, unpinIfLast); 8082 } 8083 } 8084 return mHandles.size(); 8085} 8086 8087void AudioFlinger::EffectModule::updateState() { 8088 Mutex::Autolock _l(mLock); 8089 8090 switch (mState) { 8091 case RESTART: 8092 reset_l(); 8093 // FALL THROUGH 8094 8095 case STARTING: 8096 // clear auxiliary effect input buffer for next accumulation 8097 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8098 memset(mConfig.inputCfg.buffer.raw, 8099 0, 8100 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8101 } 8102 start_l(); 8103 mState = ACTIVE; 8104 break; 8105 case STOPPING: 8106 stop_l(); 8107 mDisableWaitCnt = mMaxDisableWaitCnt; 8108 mState = STOPPED; 8109 break; 8110 case STOPPED: 8111 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8112 // turn off sequence. 8113 if (--mDisableWaitCnt == 0) { 8114 reset_l(); 8115 mState = IDLE; 8116 } 8117 break; 8118 default: //IDLE , ACTIVE, DESTROYED 8119 break; 8120 } 8121} 8122 8123void AudioFlinger::EffectModule::process() 8124{ 8125 Mutex::Autolock _l(mLock); 8126 8127 if (mState == DESTROYED || mEffectInterface == NULL || 8128 mConfig.inputCfg.buffer.raw == NULL || 8129 mConfig.outputCfg.buffer.raw == NULL) { 8130 return; 8131 } 8132 8133 if (isProcessEnabled()) { 8134 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8135 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8136 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8137 mConfig.inputCfg.buffer.s32, 8138 mConfig.inputCfg.buffer.frameCount/2); 8139 } 8140 8141 // do the actual processing in the effect engine 8142 int ret = (*mEffectInterface)->process(mEffectInterface, 8143 &mConfig.inputCfg.buffer, 8144 &mConfig.outputCfg.buffer); 8145 8146 // force transition to IDLE state when engine is ready 8147 if (mState == STOPPED && ret == -ENODATA) { 8148 mDisableWaitCnt = 1; 8149 } 8150 8151 // clear auxiliary effect input buffer for next accumulation 8152 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8153 memset(mConfig.inputCfg.buffer.raw, 0, 8154 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8155 } 8156 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8157 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8158 // If an insert effect is idle and input buffer is different from output buffer, 8159 // accumulate input onto output 8160 sp<EffectChain> chain = mChain.promote(); 8161 if (chain != 0 && chain->activeTrackCnt() != 0) { 8162 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8163 int16_t *in = mConfig.inputCfg.buffer.s16; 8164 int16_t *out = mConfig.outputCfg.buffer.s16; 8165 for (size_t i = 0; i < frameCnt; i++) { 8166 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8167 } 8168 } 8169 } 8170} 8171 8172void AudioFlinger::EffectModule::reset_l() 8173{ 8174 if (mEffectInterface == NULL) { 8175 return; 8176 } 8177 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8178} 8179 8180status_t AudioFlinger::EffectModule::configure() 8181{ 8182 if (mEffectInterface == NULL) { 8183 return NO_INIT; 8184 } 8185 8186 sp<ThreadBase> thread = mThread.promote(); 8187 if (thread == 0) { 8188 return DEAD_OBJECT; 8189 } 8190 8191 // TODO: handle configuration of effects replacing track process 8192 audio_channel_mask_t channelMask = thread->channelMask(); 8193 8194 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8195 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8196 } else { 8197 mConfig.inputCfg.channels = channelMask; 8198 } 8199 mConfig.outputCfg.channels = channelMask; 8200 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8201 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8202 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8203 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8204 mConfig.inputCfg.bufferProvider.cookie = NULL; 8205 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8206 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8207 mConfig.outputCfg.bufferProvider.cookie = NULL; 8208 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8209 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8210 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8211 // Insert effect: 8212 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8213 // always overwrites output buffer: input buffer == output buffer 8214 // - in other sessions: 8215 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8216 // other effect: overwrites output buffer: input buffer == output buffer 8217 // Auxiliary effect: 8218 // accumulates in output buffer: input buffer != output buffer 8219 // Therefore: accumulate <=> input buffer != output buffer 8220 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8221 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8222 } else { 8223 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8224 } 8225 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8226 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8227 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8228 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8229 8230 ALOGV("configure() %p thread %p buffer %p framecount %d", 8231 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8232 8233 status_t cmdStatus; 8234 uint32_t size = sizeof(int); 8235 status_t status = (*mEffectInterface)->command(mEffectInterface, 8236 EFFECT_CMD_SET_CONFIG, 8237 sizeof(effect_config_t), 8238 &mConfig, 8239 &size, 8240 &cmdStatus); 8241 if (status == 0) { 8242 status = cmdStatus; 8243 } 8244 8245 if (status == 0 && 8246 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8247 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8248 effect_param_t *p = (effect_param_t *)buf32; 8249 8250 p->psize = sizeof(uint32_t); 8251 p->vsize = sizeof(uint32_t); 8252 size = sizeof(int); 8253 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8254 8255 uint32_t latency = 0; 8256 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8257 if (pbt != NULL) { 8258 latency = pbt->latency_l(); 8259 } 8260 8261 *((int32_t *)p->data + 1)= latency; 8262 (*mEffectInterface)->command(mEffectInterface, 8263 EFFECT_CMD_SET_PARAM, 8264 sizeof(effect_param_t) + 8, 8265 &buf32, 8266 &size, 8267 &cmdStatus); 8268 } 8269 8270 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8271 (1000 * mConfig.outputCfg.buffer.frameCount); 8272 8273 return status; 8274} 8275 8276status_t AudioFlinger::EffectModule::init() 8277{ 8278 Mutex::Autolock _l(mLock); 8279 if (mEffectInterface == NULL) { 8280 return NO_INIT; 8281 } 8282 status_t cmdStatus; 8283 uint32_t size = sizeof(status_t); 8284 status_t status = (*mEffectInterface)->command(mEffectInterface, 8285 EFFECT_CMD_INIT, 8286 0, 8287 NULL, 8288 &size, 8289 &cmdStatus); 8290 if (status == 0) { 8291 status = cmdStatus; 8292 } 8293 return status; 8294} 8295 8296status_t AudioFlinger::EffectModule::start() 8297{ 8298 Mutex::Autolock _l(mLock); 8299 return start_l(); 8300} 8301 8302status_t AudioFlinger::EffectModule::start_l() 8303{ 8304 if (mEffectInterface == NULL) { 8305 return NO_INIT; 8306 } 8307 status_t cmdStatus; 8308 uint32_t size = sizeof(status_t); 8309 status_t status = (*mEffectInterface)->command(mEffectInterface, 8310 EFFECT_CMD_ENABLE, 8311 0, 8312 NULL, 8313 &size, 8314 &cmdStatus); 8315 if (status == 0) { 8316 status = cmdStatus; 8317 } 8318 if (status == 0 && 8319 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8320 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8321 sp<ThreadBase> thread = mThread.promote(); 8322 if (thread != 0) { 8323 audio_stream_t *stream = thread->stream(); 8324 if (stream != NULL) { 8325 stream->add_audio_effect(stream, mEffectInterface); 8326 } 8327 } 8328 } 8329 return status; 8330} 8331 8332status_t AudioFlinger::EffectModule::stop() 8333{ 8334 Mutex::Autolock _l(mLock); 8335 return stop_l(); 8336} 8337 8338status_t AudioFlinger::EffectModule::stop_l() 8339{ 8340 if (mEffectInterface == NULL) { 8341 return NO_INIT; 8342 } 8343 status_t cmdStatus; 8344 uint32_t size = sizeof(status_t); 8345 status_t status = (*mEffectInterface)->command(mEffectInterface, 8346 EFFECT_CMD_DISABLE, 8347 0, 8348 NULL, 8349 &size, 8350 &cmdStatus); 8351 if (status == 0) { 8352 status = cmdStatus; 8353 } 8354 if (status == 0 && 8355 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8356 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8357 sp<ThreadBase> thread = mThread.promote(); 8358 if (thread != 0) { 8359 audio_stream_t *stream = thread->stream(); 8360 if (stream != NULL) { 8361 stream->remove_audio_effect(stream, mEffectInterface); 8362 } 8363 } 8364 } 8365 return status; 8366} 8367 8368status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8369 uint32_t cmdSize, 8370 void *pCmdData, 8371 uint32_t *replySize, 8372 void *pReplyData) 8373{ 8374 Mutex::Autolock _l(mLock); 8375// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8376 8377 if (mState == DESTROYED || mEffectInterface == NULL) { 8378 return NO_INIT; 8379 } 8380 status_t status = (*mEffectInterface)->command(mEffectInterface, 8381 cmdCode, 8382 cmdSize, 8383 pCmdData, 8384 replySize, 8385 pReplyData); 8386 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8387 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8388 for (size_t i = 1; i < mHandles.size(); i++) { 8389 EffectHandle *h = mHandles[i]; 8390 if (h != NULL && !h->destroyed_l()) { 8391 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8392 } 8393 } 8394 } 8395 return status; 8396} 8397 8398status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8399{ 8400 Mutex::Autolock _l(mLock); 8401 return setEnabled_l(enabled); 8402} 8403 8404// must be called with EffectModule::mLock held 8405status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8406{ 8407 8408 ALOGV("setEnabled %p enabled %d", this, enabled); 8409 8410 if (enabled != isEnabled()) { 8411 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8412 if (enabled && status != NO_ERROR) { 8413 return status; 8414 } 8415 8416 switch (mState) { 8417 // going from disabled to enabled 8418 case IDLE: 8419 mState = STARTING; 8420 break; 8421 case STOPPED: 8422 mState = RESTART; 8423 break; 8424 case STOPPING: 8425 mState = ACTIVE; 8426 break; 8427 8428 // going from enabled to disabled 8429 case RESTART: 8430 mState = STOPPED; 8431 break; 8432 case STARTING: 8433 mState = IDLE; 8434 break; 8435 case ACTIVE: 8436 mState = STOPPING; 8437 break; 8438 case DESTROYED: 8439 return NO_ERROR; // simply ignore as we are being destroyed 8440 } 8441 for (size_t i = 1; i < mHandles.size(); i++) { 8442 EffectHandle *h = mHandles[i]; 8443 if (h != NULL && !h->destroyed_l()) { 8444 h->setEnabled(enabled); 8445 } 8446 } 8447 } 8448 return NO_ERROR; 8449} 8450 8451bool AudioFlinger::EffectModule::isEnabled() const 8452{ 8453 switch (mState) { 8454 case RESTART: 8455 case STARTING: 8456 case ACTIVE: 8457 return true; 8458 case IDLE: 8459 case STOPPING: 8460 case STOPPED: 8461 case DESTROYED: 8462 default: 8463 return false; 8464 } 8465} 8466 8467bool AudioFlinger::EffectModule::isProcessEnabled() const 8468{ 8469 switch (mState) { 8470 case RESTART: 8471 case ACTIVE: 8472 case STOPPING: 8473 case STOPPED: 8474 return true; 8475 case IDLE: 8476 case STARTING: 8477 case DESTROYED: 8478 default: 8479 return false; 8480 } 8481} 8482 8483status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8484{ 8485 Mutex::Autolock _l(mLock); 8486 status_t status = NO_ERROR; 8487 8488 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8489 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8490 if (isProcessEnabled() && 8491 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8492 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8493 status_t cmdStatus; 8494 uint32_t volume[2]; 8495 uint32_t *pVolume = NULL; 8496 uint32_t size = sizeof(volume); 8497 volume[0] = *left; 8498 volume[1] = *right; 8499 if (controller) { 8500 pVolume = volume; 8501 } 8502 status = (*mEffectInterface)->command(mEffectInterface, 8503 EFFECT_CMD_SET_VOLUME, 8504 size, 8505 volume, 8506 &size, 8507 pVolume); 8508 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8509 *left = volume[0]; 8510 *right = volume[1]; 8511 } 8512 } 8513 return status; 8514} 8515 8516status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8517{ 8518 Mutex::Autolock _l(mLock); 8519 status_t status = NO_ERROR; 8520 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8521 // audio pre processing modules on RecordThread can receive both output and 8522 // input device indication in the same call 8523 audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL; 8524 if (dev) { 8525 status_t cmdStatus; 8526 uint32_t size = sizeof(status_t); 8527 8528 status = (*mEffectInterface)->command(mEffectInterface, 8529 EFFECT_CMD_SET_DEVICE, 8530 sizeof(uint32_t), 8531 &dev, 8532 &size, 8533 &cmdStatus); 8534 if (status == NO_ERROR) { 8535 status = cmdStatus; 8536 } 8537 } 8538 dev = device & AUDIO_DEVICE_IN_ALL; 8539 if (dev) { 8540 status_t cmdStatus; 8541 uint32_t size = sizeof(status_t); 8542 8543 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8544 EFFECT_CMD_SET_INPUT_DEVICE, 8545 sizeof(uint32_t), 8546 &dev, 8547 &size, 8548 &cmdStatus); 8549 if (status2 == NO_ERROR) { 8550 status2 = cmdStatus; 8551 } 8552 if (status == NO_ERROR) { 8553 status = status2; 8554 } 8555 } 8556 } 8557 return status; 8558} 8559 8560status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8561{ 8562 Mutex::Autolock _l(mLock); 8563 status_t status = NO_ERROR; 8564 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8565 status_t cmdStatus; 8566 uint32_t size = sizeof(status_t); 8567 status = (*mEffectInterface)->command(mEffectInterface, 8568 EFFECT_CMD_SET_AUDIO_MODE, 8569 sizeof(audio_mode_t), 8570 &mode, 8571 &size, 8572 &cmdStatus); 8573 if (status == NO_ERROR) { 8574 status = cmdStatus; 8575 } 8576 } 8577 return status; 8578} 8579 8580void AudioFlinger::EffectModule::setSuspended(bool suspended) 8581{ 8582 Mutex::Autolock _l(mLock); 8583 mSuspended = suspended; 8584} 8585 8586bool AudioFlinger::EffectModule::suspended() const 8587{ 8588 Mutex::Autolock _l(mLock); 8589 return mSuspended; 8590} 8591 8592bool AudioFlinger::EffectModule::purgeHandles() 8593{ 8594 bool enabled = false; 8595 Mutex::Autolock _l(mLock); 8596 for (size_t i = 0; i < mHandles.size(); i++) { 8597 EffectHandle *handle = mHandles[i]; 8598 if (handle != NULL && !handle->destroyed_l()) { 8599 handle->effect().clear(); 8600 if (handle->hasControl()) { 8601 enabled = handle->enabled(); 8602 } 8603 } 8604 } 8605 return enabled; 8606} 8607 8608void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8609{ 8610 const size_t SIZE = 256; 8611 char buffer[SIZE]; 8612 String8 result; 8613 8614 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8615 result.append(buffer); 8616 8617 bool locked = tryLock(mLock); 8618 // failed to lock - AudioFlinger is probably deadlocked 8619 if (!locked) { 8620 result.append("\t\tCould not lock Fx mutex:\n"); 8621 } 8622 8623 result.append("\t\tSession Status State Engine:\n"); 8624 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8625 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8626 result.append(buffer); 8627 8628 result.append("\t\tDescriptor:\n"); 8629 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8630 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8631 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8632 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8633 result.append(buffer); 8634 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8635 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8636 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8637 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8638 result.append(buffer); 8639 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8640 mDescriptor.apiVersion, 8641 mDescriptor.flags); 8642 result.append(buffer); 8643 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8644 mDescriptor.name); 8645 result.append(buffer); 8646 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8647 mDescriptor.implementor); 8648 result.append(buffer); 8649 8650 result.append("\t\t- Input configuration:\n"); 8651 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8652 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8653 (uint32_t)mConfig.inputCfg.buffer.raw, 8654 mConfig.inputCfg.buffer.frameCount, 8655 mConfig.inputCfg.samplingRate, 8656 mConfig.inputCfg.channels, 8657 mConfig.inputCfg.format); 8658 result.append(buffer); 8659 8660 result.append("\t\t- Output configuration:\n"); 8661 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8662 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8663 (uint32_t)mConfig.outputCfg.buffer.raw, 8664 mConfig.outputCfg.buffer.frameCount, 8665 mConfig.outputCfg.samplingRate, 8666 mConfig.outputCfg.channels, 8667 mConfig.outputCfg.format); 8668 result.append(buffer); 8669 8670 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8671 result.append(buffer); 8672 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8673 for (size_t i = 0; i < mHandles.size(); ++i) { 8674 EffectHandle *handle = mHandles[i]; 8675 if (handle != NULL && !handle->destroyed_l()) { 8676 handle->dump(buffer, SIZE); 8677 result.append(buffer); 8678 } 8679 } 8680 8681 result.append("\n"); 8682 8683 write(fd, result.string(), result.length()); 8684 8685 if (locked) { 8686 mLock.unlock(); 8687 } 8688} 8689 8690// ---------------------------------------------------------------------------- 8691// EffectHandle implementation 8692// ---------------------------------------------------------------------------- 8693 8694#undef LOG_TAG 8695#define LOG_TAG "AudioFlinger::EffectHandle" 8696 8697AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8698 const sp<AudioFlinger::Client>& client, 8699 const sp<IEffectClient>& effectClient, 8700 int32_t priority) 8701 : BnEffect(), 8702 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8703 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8704{ 8705 ALOGV("constructor %p", this); 8706 8707 if (client == 0) { 8708 return; 8709 } 8710 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8711 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8712 if (mCblkMemory != 0) { 8713 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8714 8715 if (mCblk != NULL) { 8716 new(mCblk) effect_param_cblk_t(); 8717 mBuffer = (uint8_t *)mCblk + bufOffset; 8718 } 8719 } else { 8720 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8721 return; 8722 } 8723} 8724 8725AudioFlinger::EffectHandle::~EffectHandle() 8726{ 8727 ALOGV("Destructor %p", this); 8728 8729 if (mEffect == 0) { 8730 mDestroyed = true; 8731 return; 8732 } 8733 mEffect->lock(); 8734 mDestroyed = true; 8735 mEffect->unlock(); 8736 disconnect(false); 8737} 8738 8739status_t AudioFlinger::EffectHandle::enable() 8740{ 8741 ALOGV("enable %p", this); 8742 if (!mHasControl) return INVALID_OPERATION; 8743 if (mEffect == 0) return DEAD_OBJECT; 8744 8745 if (mEnabled) { 8746 return NO_ERROR; 8747 } 8748 8749 mEnabled = true; 8750 8751 sp<ThreadBase> thread = mEffect->thread().promote(); 8752 if (thread != 0) { 8753 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8754 } 8755 8756 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8757 if (mEffect->suspended()) { 8758 return NO_ERROR; 8759 } 8760 8761 status_t status = mEffect->setEnabled(true); 8762 if (status != NO_ERROR) { 8763 if (thread != 0) { 8764 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8765 } 8766 mEnabled = false; 8767 } 8768 return status; 8769} 8770 8771status_t AudioFlinger::EffectHandle::disable() 8772{ 8773 ALOGV("disable %p", this); 8774 if (!mHasControl) return INVALID_OPERATION; 8775 if (mEffect == 0) return DEAD_OBJECT; 8776 8777 if (!mEnabled) { 8778 return NO_ERROR; 8779 } 8780 mEnabled = false; 8781 8782 if (mEffect->suspended()) { 8783 return NO_ERROR; 8784 } 8785 8786 status_t status = mEffect->setEnabled(false); 8787 8788 sp<ThreadBase> thread = mEffect->thread().promote(); 8789 if (thread != 0) { 8790 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8791 } 8792 8793 return status; 8794} 8795 8796void AudioFlinger::EffectHandle::disconnect() 8797{ 8798 disconnect(true); 8799} 8800 8801void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8802{ 8803 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8804 if (mEffect == 0) { 8805 return; 8806 } 8807 // restore suspended effects if the disconnected handle was enabled and the last one. 8808 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8809 sp<ThreadBase> thread = mEffect->thread().promote(); 8810 if (thread != 0) { 8811 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8812 } 8813 } 8814 8815 // release sp on module => module destructor can be called now 8816 mEffect.clear(); 8817 if (mClient != 0) { 8818 if (mCblk != NULL) { 8819 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8820 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8821 } 8822 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8823 // Client destructor must run with AudioFlinger mutex locked 8824 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8825 mClient.clear(); 8826 } 8827} 8828 8829status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8830 uint32_t cmdSize, 8831 void *pCmdData, 8832 uint32_t *replySize, 8833 void *pReplyData) 8834{ 8835// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8836// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8837 8838 // only get parameter command is permitted for applications not controlling the effect 8839 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8840 return INVALID_OPERATION; 8841 } 8842 if (mEffect == 0) return DEAD_OBJECT; 8843 if (mClient == 0) return INVALID_OPERATION; 8844 8845 // handle commands that are not forwarded transparently to effect engine 8846 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8847 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8848 // no risk to block the whole media server process or mixer threads is we are stuck here 8849 Mutex::Autolock _l(mCblk->lock); 8850 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8851 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8852 mCblk->serverIndex = 0; 8853 mCblk->clientIndex = 0; 8854 return BAD_VALUE; 8855 } 8856 status_t status = NO_ERROR; 8857 while (mCblk->serverIndex < mCblk->clientIndex) { 8858 int reply; 8859 uint32_t rsize = sizeof(int); 8860 int *p = (int *)(mBuffer + mCblk->serverIndex); 8861 int size = *p++; 8862 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8863 ALOGW("command(): invalid parameter block size"); 8864 break; 8865 } 8866 effect_param_t *param = (effect_param_t *)p; 8867 if (param->psize == 0 || param->vsize == 0) { 8868 ALOGW("command(): null parameter or value size"); 8869 mCblk->serverIndex += size; 8870 continue; 8871 } 8872 uint32_t psize = sizeof(effect_param_t) + 8873 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8874 param->vsize; 8875 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8876 psize, 8877 p, 8878 &rsize, 8879 &reply); 8880 // stop at first error encountered 8881 if (ret != NO_ERROR) { 8882 status = ret; 8883 *(int *)pReplyData = reply; 8884 break; 8885 } else if (reply != NO_ERROR) { 8886 *(int *)pReplyData = reply; 8887 break; 8888 } 8889 mCblk->serverIndex += size; 8890 } 8891 mCblk->serverIndex = 0; 8892 mCblk->clientIndex = 0; 8893 return status; 8894 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8895 *(int *)pReplyData = NO_ERROR; 8896 return enable(); 8897 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8898 *(int *)pReplyData = NO_ERROR; 8899 return disable(); 8900 } 8901 8902 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8903} 8904 8905void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8906{ 8907 ALOGV("setControl %p control %d", this, hasControl); 8908 8909 mHasControl = hasControl; 8910 mEnabled = enabled; 8911 8912 if (signal && mEffectClient != 0) { 8913 mEffectClient->controlStatusChanged(hasControl); 8914 } 8915} 8916 8917void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8918 uint32_t cmdSize, 8919 void *pCmdData, 8920 uint32_t replySize, 8921 void *pReplyData) 8922{ 8923 if (mEffectClient != 0) { 8924 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8925 } 8926} 8927 8928 8929 8930void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8931{ 8932 if (mEffectClient != 0) { 8933 mEffectClient->enableStatusChanged(enabled); 8934 } 8935} 8936 8937status_t AudioFlinger::EffectHandle::onTransact( 8938 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8939{ 8940 return BnEffect::onTransact(code, data, reply, flags); 8941} 8942 8943 8944void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8945{ 8946 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8947 8948 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8949 (mClient == 0) ? getpid_cached : mClient->pid(), 8950 mPriority, 8951 mHasControl, 8952 !locked, 8953 mCblk ? mCblk->clientIndex : 0, 8954 mCblk ? mCblk->serverIndex : 0 8955 ); 8956 8957 if (locked) { 8958 mCblk->lock.unlock(); 8959 } 8960} 8961 8962#undef LOG_TAG 8963#define LOG_TAG "AudioFlinger::EffectChain" 8964 8965AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8966 int sessionId) 8967 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8968 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8969 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8970{ 8971 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8972 if (thread == NULL) { 8973 return; 8974 } 8975 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8976 thread->frameCount(); 8977} 8978 8979AudioFlinger::EffectChain::~EffectChain() 8980{ 8981 if (mOwnInBuffer) { 8982 delete mInBuffer; 8983 } 8984 8985} 8986 8987// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8988sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8989{ 8990 size_t size = mEffects.size(); 8991 8992 for (size_t i = 0; i < size; i++) { 8993 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8994 return mEffects[i]; 8995 } 8996 } 8997 return 0; 8998} 8999 9000// getEffectFromId_l() must be called with ThreadBase::mLock held 9001sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9002{ 9003 size_t size = mEffects.size(); 9004 9005 for (size_t i = 0; i < size; i++) { 9006 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9007 if (id == 0 || mEffects[i]->id() == id) { 9008 return mEffects[i]; 9009 } 9010 } 9011 return 0; 9012} 9013 9014// getEffectFromType_l() must be called with ThreadBase::mLock held 9015sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9016 const effect_uuid_t *type) 9017{ 9018 size_t size = mEffects.size(); 9019 9020 for (size_t i = 0; i < size; i++) { 9021 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9022 return mEffects[i]; 9023 } 9024 } 9025 return 0; 9026} 9027 9028void AudioFlinger::EffectChain::clearInputBuffer() 9029{ 9030 Mutex::Autolock _l(mLock); 9031 sp<ThreadBase> thread = mThread.promote(); 9032 if (thread == 0) { 9033 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9034 return; 9035 } 9036 clearInputBuffer_l(thread); 9037} 9038 9039// Must be called with EffectChain::mLock locked 9040void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9041{ 9042 size_t numSamples = thread->frameCount() * thread->channelCount(); 9043 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9044 9045} 9046 9047// Must be called with EffectChain::mLock locked 9048void AudioFlinger::EffectChain::process_l() 9049{ 9050 sp<ThreadBase> thread = mThread.promote(); 9051 if (thread == 0) { 9052 ALOGW("process_l(): cannot promote mixer thread"); 9053 return; 9054 } 9055 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9056 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9057 // always process effects unless no more tracks are on the session and the effect tail 9058 // has been rendered 9059 bool doProcess = true; 9060 if (!isGlobalSession) { 9061 bool tracksOnSession = (trackCnt() != 0); 9062 9063 if (!tracksOnSession && mTailBufferCount == 0) { 9064 doProcess = false; 9065 } 9066 9067 if (activeTrackCnt() == 0) { 9068 // if no track is active and the effect tail has not been rendered, 9069 // the input buffer must be cleared here as the mixer process will not do it 9070 if (tracksOnSession || mTailBufferCount > 0) { 9071 clearInputBuffer_l(thread); 9072 if (mTailBufferCount > 0) { 9073 mTailBufferCount--; 9074 } 9075 } 9076 } 9077 } 9078 9079 size_t size = mEffects.size(); 9080 if (doProcess) { 9081 for (size_t i = 0; i < size; i++) { 9082 mEffects[i]->process(); 9083 } 9084 } 9085 for (size_t i = 0; i < size; i++) { 9086 mEffects[i]->updateState(); 9087 } 9088} 9089 9090// addEffect_l() must be called with PlaybackThread::mLock held 9091status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9092{ 9093 effect_descriptor_t desc = effect->desc(); 9094 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9095 9096 Mutex::Autolock _l(mLock); 9097 effect->setChain(this); 9098 sp<ThreadBase> thread = mThread.promote(); 9099 if (thread == 0) { 9100 return NO_INIT; 9101 } 9102 effect->setThread(thread); 9103 9104 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9105 // Auxiliary effects are inserted at the beginning of mEffects vector as 9106 // they are processed first and accumulated in chain input buffer 9107 mEffects.insertAt(effect, 0); 9108 9109 // the input buffer for auxiliary effect contains mono samples in 9110 // 32 bit format. This is to avoid saturation in AudoMixer 9111 // accumulation stage. Saturation is done in EffectModule::process() before 9112 // calling the process in effect engine 9113 size_t numSamples = thread->frameCount(); 9114 int32_t *buffer = new int32_t[numSamples]; 9115 memset(buffer, 0, numSamples * sizeof(int32_t)); 9116 effect->setInBuffer((int16_t *)buffer); 9117 // auxiliary effects output samples to chain input buffer for further processing 9118 // by insert effects 9119 effect->setOutBuffer(mInBuffer); 9120 } else { 9121 // Insert effects are inserted at the end of mEffects vector as they are processed 9122 // after track and auxiliary effects. 9123 // Insert effect order as a function of indicated preference: 9124 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9125 // another effect is present 9126 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9127 // last effect claiming first position 9128 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9129 // first effect claiming last position 9130 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9131 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9132 // already present 9133 9134 size_t size = mEffects.size(); 9135 size_t idx_insert = size; 9136 ssize_t idx_insert_first = -1; 9137 ssize_t idx_insert_last = -1; 9138 9139 for (size_t i = 0; i < size; i++) { 9140 effect_descriptor_t d = mEffects[i]->desc(); 9141 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9142 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9143 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9144 // check invalid effect chaining combinations 9145 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9146 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9147 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9148 return INVALID_OPERATION; 9149 } 9150 // remember position of first insert effect and by default 9151 // select this as insert position for new effect 9152 if (idx_insert == size) { 9153 idx_insert = i; 9154 } 9155 // remember position of last insert effect claiming 9156 // first position 9157 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9158 idx_insert_first = i; 9159 } 9160 // remember position of first insert effect claiming 9161 // last position 9162 if (iPref == EFFECT_FLAG_INSERT_LAST && 9163 idx_insert_last == -1) { 9164 idx_insert_last = i; 9165 } 9166 } 9167 } 9168 9169 // modify idx_insert from first position if needed 9170 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9171 if (idx_insert_last != -1) { 9172 idx_insert = idx_insert_last; 9173 } else { 9174 idx_insert = size; 9175 } 9176 } else { 9177 if (idx_insert_first != -1) { 9178 idx_insert = idx_insert_first + 1; 9179 } 9180 } 9181 9182 // always read samples from chain input buffer 9183 effect->setInBuffer(mInBuffer); 9184 9185 // if last effect in the chain, output samples to chain 9186 // output buffer, otherwise to chain input buffer 9187 if (idx_insert == size) { 9188 if (idx_insert != 0) { 9189 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9190 mEffects[idx_insert-1]->configure(); 9191 } 9192 effect->setOutBuffer(mOutBuffer); 9193 } else { 9194 effect->setOutBuffer(mInBuffer); 9195 } 9196 mEffects.insertAt(effect, idx_insert); 9197 9198 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9199 } 9200 effect->configure(); 9201 return NO_ERROR; 9202} 9203 9204// removeEffect_l() must be called with PlaybackThread::mLock held 9205size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9206{ 9207 Mutex::Autolock _l(mLock); 9208 size_t size = mEffects.size(); 9209 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9210 9211 for (size_t i = 0; i < size; i++) { 9212 if (effect == mEffects[i]) { 9213 // calling stop here will remove pre-processing effect from the audio HAL. 9214 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9215 // the middle of a read from audio HAL 9216 if (mEffects[i]->state() == EffectModule::ACTIVE || 9217 mEffects[i]->state() == EffectModule::STOPPING) { 9218 mEffects[i]->stop(); 9219 } 9220 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9221 delete[] effect->inBuffer(); 9222 } else { 9223 if (i == size - 1 && i != 0) { 9224 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9225 mEffects[i - 1]->configure(); 9226 } 9227 } 9228 mEffects.removeAt(i); 9229 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9230 break; 9231 } 9232 } 9233 9234 return mEffects.size(); 9235} 9236 9237// setDevice_l() must be called with PlaybackThread::mLock held 9238void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9239{ 9240 size_t size = mEffects.size(); 9241 for (size_t i = 0; i < size; i++) { 9242 mEffects[i]->setDevice(device); 9243 } 9244} 9245 9246// setMode_l() must be called with PlaybackThread::mLock held 9247void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9248{ 9249 size_t size = mEffects.size(); 9250 for (size_t i = 0; i < size; i++) { 9251 mEffects[i]->setMode(mode); 9252 } 9253} 9254 9255// setVolume_l() must be called with PlaybackThread::mLock held 9256bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9257{ 9258 uint32_t newLeft = *left; 9259 uint32_t newRight = *right; 9260 bool hasControl = false; 9261 int ctrlIdx = -1; 9262 size_t size = mEffects.size(); 9263 9264 // first update volume controller 9265 for (size_t i = size; i > 0; i--) { 9266 if (mEffects[i - 1]->isProcessEnabled() && 9267 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9268 ctrlIdx = i - 1; 9269 hasControl = true; 9270 break; 9271 } 9272 } 9273 9274 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9275 if (hasControl) { 9276 *left = mNewLeftVolume; 9277 *right = mNewRightVolume; 9278 } 9279 return hasControl; 9280 } 9281 9282 mVolumeCtrlIdx = ctrlIdx; 9283 mLeftVolume = newLeft; 9284 mRightVolume = newRight; 9285 9286 // second get volume update from volume controller 9287 if (ctrlIdx >= 0) { 9288 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9289 mNewLeftVolume = newLeft; 9290 mNewRightVolume = newRight; 9291 } 9292 // then indicate volume to all other effects in chain. 9293 // Pass altered volume to effects before volume controller 9294 // and requested volume to effects after controller 9295 uint32_t lVol = newLeft; 9296 uint32_t rVol = newRight; 9297 9298 for (size_t i = 0; i < size; i++) { 9299 if ((int)i == ctrlIdx) continue; 9300 // this also works for ctrlIdx == -1 when there is no volume controller 9301 if ((int)i > ctrlIdx) { 9302 lVol = *left; 9303 rVol = *right; 9304 } 9305 mEffects[i]->setVolume(&lVol, &rVol, false); 9306 } 9307 *left = newLeft; 9308 *right = newRight; 9309 9310 return hasControl; 9311} 9312 9313void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9314{ 9315 const size_t SIZE = 256; 9316 char buffer[SIZE]; 9317 String8 result; 9318 9319 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9320 result.append(buffer); 9321 9322 bool locked = tryLock(mLock); 9323 // failed to lock - AudioFlinger is probably deadlocked 9324 if (!locked) { 9325 result.append("\tCould not lock mutex:\n"); 9326 } 9327 9328 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9329 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9330 mEffects.size(), 9331 (uint32_t)mInBuffer, 9332 (uint32_t)mOutBuffer, 9333 mActiveTrackCnt); 9334 result.append(buffer); 9335 write(fd, result.string(), result.size()); 9336 9337 for (size_t i = 0; i < mEffects.size(); ++i) { 9338 sp<EffectModule> effect = mEffects[i]; 9339 if (effect != 0) { 9340 effect->dump(fd, args); 9341 } 9342 } 9343 9344 if (locked) { 9345 mLock.unlock(); 9346 } 9347} 9348 9349// must be called with ThreadBase::mLock held 9350void AudioFlinger::EffectChain::setEffectSuspended_l( 9351 const effect_uuid_t *type, bool suspend) 9352{ 9353 sp<SuspendedEffectDesc> desc; 9354 // use effect type UUID timelow as key as there is no real risk of identical 9355 // timeLow fields among effect type UUIDs. 9356 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9357 if (suspend) { 9358 if (index >= 0) { 9359 desc = mSuspendedEffects.valueAt(index); 9360 } else { 9361 desc = new SuspendedEffectDesc(); 9362 desc->mType = *type; 9363 mSuspendedEffects.add(type->timeLow, desc); 9364 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9365 } 9366 if (desc->mRefCount++ == 0) { 9367 sp<EffectModule> effect = getEffectIfEnabled(type); 9368 if (effect != 0) { 9369 desc->mEffect = effect; 9370 effect->setSuspended(true); 9371 effect->setEnabled(false); 9372 } 9373 } 9374 } else { 9375 if (index < 0) { 9376 return; 9377 } 9378 desc = mSuspendedEffects.valueAt(index); 9379 if (desc->mRefCount <= 0) { 9380 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9381 desc->mRefCount = 1; 9382 } 9383 if (--desc->mRefCount == 0) { 9384 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9385 if (desc->mEffect != 0) { 9386 sp<EffectModule> effect = desc->mEffect.promote(); 9387 if (effect != 0) { 9388 effect->setSuspended(false); 9389 effect->lock(); 9390 EffectHandle *handle = effect->controlHandle_l(); 9391 if (handle != NULL && !handle->destroyed_l()) { 9392 effect->setEnabled_l(handle->enabled()); 9393 } 9394 effect->unlock(); 9395 } 9396 desc->mEffect.clear(); 9397 } 9398 mSuspendedEffects.removeItemsAt(index); 9399 } 9400 } 9401} 9402 9403// must be called with ThreadBase::mLock held 9404void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9405{ 9406 sp<SuspendedEffectDesc> desc; 9407 9408 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9409 if (suspend) { 9410 if (index >= 0) { 9411 desc = mSuspendedEffects.valueAt(index); 9412 } else { 9413 desc = new SuspendedEffectDesc(); 9414 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9415 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9416 } 9417 if (desc->mRefCount++ == 0) { 9418 Vector< sp<EffectModule> > effects; 9419 getSuspendEligibleEffects(effects); 9420 for (size_t i = 0; i < effects.size(); i++) { 9421 setEffectSuspended_l(&effects[i]->desc().type, true); 9422 } 9423 } 9424 } else { 9425 if (index < 0) { 9426 return; 9427 } 9428 desc = mSuspendedEffects.valueAt(index); 9429 if (desc->mRefCount <= 0) { 9430 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9431 desc->mRefCount = 1; 9432 } 9433 if (--desc->mRefCount == 0) { 9434 Vector<const effect_uuid_t *> types; 9435 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9436 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9437 continue; 9438 } 9439 types.add(&mSuspendedEffects.valueAt(i)->mType); 9440 } 9441 for (size_t i = 0; i < types.size(); i++) { 9442 setEffectSuspended_l(types[i], false); 9443 } 9444 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9445 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9446 } 9447 } 9448} 9449 9450 9451// The volume effect is used for automated tests only 9452#ifndef OPENSL_ES_H_ 9453static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9454 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9455const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9456#endif //OPENSL_ES_H_ 9457 9458bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9459{ 9460 // auxiliary effects and visualizer are never suspended on output mix 9461 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9462 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9463 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9464 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9465 return false; 9466 } 9467 return true; 9468} 9469 9470void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9471{ 9472 effects.clear(); 9473 for (size_t i = 0; i < mEffects.size(); i++) { 9474 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9475 effects.add(mEffects[i]); 9476 } 9477 } 9478} 9479 9480sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9481 const effect_uuid_t *type) 9482{ 9483 sp<EffectModule> effect = getEffectFromType_l(type); 9484 return effect != 0 && effect->isEnabled() ? effect : 0; 9485} 9486 9487void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9488 bool enabled) 9489{ 9490 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9491 if (enabled) { 9492 if (index < 0) { 9493 // if the effect is not suspend check if all effects are suspended 9494 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9495 if (index < 0) { 9496 return; 9497 } 9498 if (!isEffectEligibleForSuspend(effect->desc())) { 9499 return; 9500 } 9501 setEffectSuspended_l(&effect->desc().type, enabled); 9502 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9503 if (index < 0) { 9504 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9505 return; 9506 } 9507 } 9508 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9509 effect->desc().type.timeLow); 9510 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9511 // if effect is requested to suspended but was not yet enabled, supend it now. 9512 if (desc->mEffect == 0) { 9513 desc->mEffect = effect; 9514 effect->setEnabled(false); 9515 effect->setSuspended(true); 9516 } 9517 } else { 9518 if (index < 0) { 9519 return; 9520 } 9521 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9522 effect->desc().type.timeLow); 9523 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9524 desc->mEffect.clear(); 9525 effect->setSuspended(false); 9526 } 9527} 9528 9529#undef LOG_TAG 9530#define LOG_TAG "AudioFlinger" 9531 9532// ---------------------------------------------------------------------------- 9533 9534status_t AudioFlinger::onTransact( 9535 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9536{ 9537 return BnAudioFlinger::onTransact(code, data, reply, flags); 9538} 9539 9540}; // namespace android 9541