AudioFlinger.cpp revision c1dae24a08b67b98e18e4239d4f3a74d600d353c
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// ---------------------------------------------------------------------------- 165 166#ifdef ADD_BATTERY_DATA 167// To collect the amplifier usage 168static void addBatteryData(uint32_t params) { 169 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 170 if (service == NULL) { 171 // it already logged 172 return; 173 } 174 175 service->addBatteryData(params); 176} 177#endif 178 179static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 180{ 181 const hw_module_t *mod; 182 int rc; 183 184 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 185 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 186 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 187 if (rc) { 188 goto out; 189 } 190 rc = audio_hw_device_open(mod, dev); 191 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 197 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 198 rc = BAD_VALUE; 199 goto out; 200 } 201 return 0; 202 203out: 204 *dev = NULL; 205 return rc; 206} 207 208// ---------------------------------------------------------------------------- 209 210AudioFlinger::AudioFlinger() 211 : BnAudioFlinger(), 212 mPrimaryHardwareDev(NULL), 213 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 214 mMasterVolume(1.0f), 215 mMasterVolumeSupportLvl(MVS_NONE), 216 mMasterMute(false), 217 mNextUniqueId(1), 218 mMode(AUDIO_MODE_INVALID), 219 mBtNrecIsOff(false) 220{ 221} 222 223void AudioFlinger::onFirstRef() 224{ 225 int rc = 0; 226 227 Mutex::Autolock _l(mLock); 228 229 /* TODO: move all this work into an Init() function */ 230 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 231 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 232 uint32_t int_val; 233 if (1 == sscanf(val_str, "%u", &int_val)) { 234 mStandbyTimeInNsecs = milliseconds(int_val); 235 ALOGI("Using %u mSec as standby time.", int_val); 236 } else { 237 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 238 ALOGI("Using default %u mSec as standby time.", 239 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 240 } 241 } 242 243 mMode = AUDIO_MODE_NORMAL; 244 mMasterVolumeSW = 1.0; 245 mMasterVolume = 1.0; 246 mHardwareStatus = AUDIO_HW_IDLE; 247} 248 249AudioFlinger::~AudioFlinger() 250{ 251 252 while (!mRecordThreads.isEmpty()) { 253 // closeInput() will remove first entry from mRecordThreads 254 closeInput(mRecordThreads.keyAt(0)); 255 } 256 while (!mPlaybackThreads.isEmpty()) { 257 // closeOutput() will remove first entry from mPlaybackThreads 258 closeOutput(mPlaybackThreads.keyAt(0)); 259 } 260 261 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 262 // no mHardwareLock needed, as there are no other references to this 263 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 264 delete mAudioHwDevs.valueAt(i); 265 } 266} 267 268static const char * const audio_interfaces[] = { 269 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 270 AUDIO_HARDWARE_MODULE_ID_A2DP, 271 AUDIO_HARDWARE_MODULE_ID_USB, 272}; 273#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 274 275audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 276{ 277 // if module is 0, the request comes from an old policy manager and we should load 278 // well known modules 279 if (module == 0) { 280 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 281 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 282 loadHwModule_l(audio_interfaces[i]); 283 } 284 } else { 285 // check a match for the requested module handle 286 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 287 if (audioHwdevice != NULL) { 288 return audioHwdevice->hwDevice(); 289 } 290 } 291 // then try to find a module supporting the requested device. 292 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 293 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 294 if ((dev->get_supported_devices(dev) & devices) == devices) 295 return dev; 296 } 297 298 return NULL; 299} 300 301status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 302{ 303 const size_t SIZE = 256; 304 char buffer[SIZE]; 305 String8 result; 306 307 result.append("Clients:\n"); 308 for (size_t i = 0; i < mClients.size(); ++i) { 309 sp<Client> client = mClients.valueAt(i).promote(); 310 if (client != 0) { 311 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 312 result.append(buffer); 313 } 314 } 315 316 result.append("Global session refs:\n"); 317 result.append(" session pid count\n"); 318 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 319 AudioSessionRef *r = mAudioSessionRefs[i]; 320 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 321 result.append(buffer); 322 } 323 write(fd, result.string(), result.size()); 324 return NO_ERROR; 325} 326 327 328status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 329{ 330 const size_t SIZE = 256; 331 char buffer[SIZE]; 332 String8 result; 333 hardware_call_state hardwareStatus = mHardwareStatus; 334 335 snprintf(buffer, SIZE, "Hardware status: %d\n" 336 "Standby Time mSec: %u\n", 337 hardwareStatus, 338 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 339 result.append(buffer); 340 write(fd, result.string(), result.size()); 341 return NO_ERROR; 342} 343 344status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 345{ 346 const size_t SIZE = 256; 347 char buffer[SIZE]; 348 String8 result; 349 snprintf(buffer, SIZE, "Permission Denial: " 350 "can't dump AudioFlinger from pid=%d, uid=%d\n", 351 IPCThreadState::self()->getCallingPid(), 352 IPCThreadState::self()->getCallingUid()); 353 result.append(buffer); 354 write(fd, result.string(), result.size()); 355 return NO_ERROR; 356} 357 358static bool tryLock(Mutex& mutex) 359{ 360 bool locked = false; 361 for (int i = 0; i < kDumpLockRetries; ++i) { 362 if (mutex.tryLock() == NO_ERROR) { 363 locked = true; 364 break; 365 } 366 usleep(kDumpLockSleepUs); 367 } 368 return locked; 369} 370 371status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 372{ 373 if (!dumpAllowed()) { 374 dumpPermissionDenial(fd, args); 375 } else { 376 // get state of hardware lock 377 bool hardwareLocked = tryLock(mHardwareLock); 378 if (!hardwareLocked) { 379 String8 result(kHardwareLockedString); 380 write(fd, result.string(), result.size()); 381 } else { 382 mHardwareLock.unlock(); 383 } 384 385 bool locked = tryLock(mLock); 386 387 // failed to lock - AudioFlinger is probably deadlocked 388 if (!locked) { 389 String8 result(kDeadlockedString); 390 write(fd, result.string(), result.size()); 391 } 392 393 dumpClients(fd, args); 394 dumpInternals(fd, args); 395 396 // dump playback threads 397 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 398 mPlaybackThreads.valueAt(i)->dump(fd, args); 399 } 400 401 // dump record threads 402 for (size_t i = 0; i < mRecordThreads.size(); i++) { 403 mRecordThreads.valueAt(i)->dump(fd, args); 404 } 405 406 // dump all hardware devs 407 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 408 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 409 dev->dump(dev, fd); 410 } 411 if (locked) mLock.unlock(); 412 } 413 return NO_ERROR; 414} 415 416sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 417{ 418 // If pid is already in the mClients wp<> map, then use that entry 419 // (for which promote() is always != 0), otherwise create a new entry and Client. 420 sp<Client> client = mClients.valueFor(pid).promote(); 421 if (client == 0) { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 return client; 427} 428 429// IAudioFlinger interface 430 431 432sp<IAudioTrack> AudioFlinger::createTrack( 433 pid_t pid, 434 audio_stream_type_t streamType, 435 uint32_t sampleRate, 436 audio_format_t format, 437 uint32_t channelMask, 438 int frameCount, 439 IAudioFlinger::track_flags_t flags, 440 const sp<IMemory>& sharedBuffer, 441 audio_io_handle_t output, 442 pid_t tid, 443 int *sessionId, 444 status_t *status) 445{ 446 sp<PlaybackThread::Track> track; 447 sp<TrackHandle> trackHandle; 448 sp<Client> client; 449 status_t lStatus; 450 int lSessionId; 451 452 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 453 // but if someone uses binder directly they could bypass that and cause us to crash 454 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 455 ALOGE("createTrack() invalid stream type %d", streamType); 456 lStatus = BAD_VALUE; 457 goto Exit; 458 } 459 460 { 461 Mutex::Autolock _l(mLock); 462 PlaybackThread *thread = checkPlaybackThread_l(output); 463 PlaybackThread *effectThread = NULL; 464 if (thread == NULL) { 465 ALOGE("unknown output thread"); 466 lStatus = BAD_VALUE; 467 goto Exit; 468 } 469 470 client = registerPid_l(pid); 471 472 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 473 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 474 // check if an effect chain with the same session ID is present on another 475 // output thread and move it here. 476 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 477 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 478 if (mPlaybackThreads.keyAt(i) != output) { 479 uint32_t sessions = t->hasAudioSession(*sessionId); 480 if (sessions & PlaybackThread::EFFECT_SESSION) { 481 effectThread = t.get(); 482 break; 483 } 484 } 485 } 486 lSessionId = *sessionId; 487 } else { 488 // if no audio session id is provided, create one here 489 lSessionId = nextUniqueId(); 490 if (sessionId != NULL) { 491 *sessionId = lSessionId; 492 } 493 } 494 ALOGV("createTrack() lSessionId: %d", lSessionId); 495 496 track = thread->createTrack_l(client, streamType, sampleRate, format, 497 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 498 499 // move effect chain to this output thread if an effect on same session was waiting 500 // for a track to be created 501 if (lStatus == NO_ERROR && effectThread != NULL) { 502 Mutex::Autolock _dl(thread->mLock); 503 Mutex::Autolock _sl(effectThread->mLock); 504 moveEffectChain_l(lSessionId, effectThread, thread, true); 505 } 506 507 // Look for sync events awaiting for a session to be used. 508 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 509 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 510 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 511 if (lStatus == NO_ERROR) { 512 track->setSyncEvent(mPendingSyncEvents[i]); 513 } else { 514 mPendingSyncEvents[i]->cancel(); 515 } 516 mPendingSyncEvents.removeAt(i); 517 i--; 518 } 519 } 520 } 521 } 522 if (lStatus == NO_ERROR) { 523 trackHandle = new TrackHandle(track); 524 } else { 525 // remove local strong reference to Client before deleting the Track so that the Client 526 // destructor is called by the TrackBase destructor with mLock held 527 client.clear(); 528 track.clear(); 529 } 530 531Exit: 532 if (status != NULL) { 533 *status = lStatus; 534 } 535 return trackHandle; 536} 537 538uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 539{ 540 Mutex::Autolock _l(mLock); 541 PlaybackThread *thread = checkPlaybackThread_l(output); 542 if (thread == NULL) { 543 ALOGW("sampleRate() unknown thread %d", output); 544 return 0; 545 } 546 return thread->sampleRate(); 547} 548 549int AudioFlinger::channelCount(audio_io_handle_t output) const 550{ 551 Mutex::Autolock _l(mLock); 552 PlaybackThread *thread = checkPlaybackThread_l(output); 553 if (thread == NULL) { 554 ALOGW("channelCount() unknown thread %d", output); 555 return 0; 556 } 557 return thread->channelCount(); 558} 559 560audio_format_t AudioFlinger::format(audio_io_handle_t output) const 561{ 562 Mutex::Autolock _l(mLock); 563 PlaybackThread *thread = checkPlaybackThread_l(output); 564 if (thread == NULL) { 565 ALOGW("format() unknown thread %d", output); 566 return AUDIO_FORMAT_INVALID; 567 } 568 return thread->format(); 569} 570 571size_t AudioFlinger::frameCount(audio_io_handle_t output) const 572{ 573 Mutex::Autolock _l(mLock); 574 PlaybackThread *thread = checkPlaybackThread_l(output); 575 if (thread == NULL) { 576 ALOGW("frameCount() unknown thread %d", output); 577 return 0; 578 } 579 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 580 // should examine all callers and fix them to handle smaller counts 581 return thread->frameCount(); 582} 583 584uint32_t AudioFlinger::latency(audio_io_handle_t output) const 585{ 586 Mutex::Autolock _l(mLock); 587 PlaybackThread *thread = checkPlaybackThread_l(output); 588 if (thread == NULL) { 589 ALOGW("latency() unknown thread %d", output); 590 return 0; 591 } 592 return thread->latency(); 593} 594 595status_t AudioFlinger::setMasterVolume(float value) 596{ 597 status_t ret = initCheck(); 598 if (ret != NO_ERROR) { 599 return ret; 600 } 601 602 // check calling permissions 603 if (!settingsAllowed()) { 604 return PERMISSION_DENIED; 605 } 606 607 float swmv = value; 608 609 Mutex::Autolock _l(mLock); 610 611 // when hw supports master volume, don't scale in sw mixer 612 if (MVS_NONE != mMasterVolumeSupportLvl) { 613 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 614 AutoMutex lock(mHardwareLock); 615 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 616 617 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 618 if (NULL != dev->set_master_volume) { 619 dev->set_master_volume(dev, value); 620 } 621 mHardwareStatus = AUDIO_HW_IDLE; 622 } 623 624 swmv = 1.0; 625 } 626 627 mMasterVolume = value; 628 mMasterVolumeSW = swmv; 629 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 630 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 631 632 return NO_ERROR; 633} 634 635status_t AudioFlinger::setMode(audio_mode_t mode) 636{ 637 status_t ret = initCheck(); 638 if (ret != NO_ERROR) { 639 return ret; 640 } 641 642 // check calling permissions 643 if (!settingsAllowed()) { 644 return PERMISSION_DENIED; 645 } 646 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 647 ALOGW("Illegal value: setMode(%d)", mode); 648 return BAD_VALUE; 649 } 650 651 { // scope for the lock 652 AutoMutex lock(mHardwareLock); 653 mHardwareStatus = AUDIO_HW_SET_MODE; 654 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 655 mHardwareStatus = AUDIO_HW_IDLE; 656 } 657 658 if (NO_ERROR == ret) { 659 Mutex::Autolock _l(mLock); 660 mMode = mode; 661 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 662 mPlaybackThreads.valueAt(i)->setMode(mode); 663 } 664 665 return ret; 666} 667 668status_t AudioFlinger::setMicMute(bool state) 669{ 670 status_t ret = initCheck(); 671 if (ret != NO_ERROR) { 672 return ret; 673 } 674 675 // check calling permissions 676 if (!settingsAllowed()) { 677 return PERMISSION_DENIED; 678 } 679 680 AutoMutex lock(mHardwareLock); 681 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 682 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 683 mHardwareStatus = AUDIO_HW_IDLE; 684 return ret; 685} 686 687bool AudioFlinger::getMicMute() const 688{ 689 status_t ret = initCheck(); 690 if (ret != NO_ERROR) { 691 return false; 692 } 693 694 bool state = AUDIO_MODE_INVALID; 695 AutoMutex lock(mHardwareLock); 696 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 697 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 698 mHardwareStatus = AUDIO_HW_IDLE; 699 return state; 700} 701 702status_t AudioFlinger::setMasterMute(bool muted) 703{ 704 // check calling permissions 705 if (!settingsAllowed()) { 706 return PERMISSION_DENIED; 707 } 708 709 Mutex::Autolock _l(mLock); 710 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 711 mMasterMute = muted; 712 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 713 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 714 715 return NO_ERROR; 716} 717 718float AudioFlinger::masterVolume() const 719{ 720 Mutex::Autolock _l(mLock); 721 return masterVolume_l(); 722} 723 724float AudioFlinger::masterVolumeSW() const 725{ 726 Mutex::Autolock _l(mLock); 727 return masterVolumeSW_l(); 728} 729 730bool AudioFlinger::masterMute() const 731{ 732 Mutex::Autolock _l(mLock); 733 return masterMute_l(); 734} 735 736float AudioFlinger::masterVolume_l() const 737{ 738 if (MVS_FULL == mMasterVolumeSupportLvl) { 739 float ret_val; 740 AutoMutex lock(mHardwareLock); 741 742 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 743 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 744 (NULL != mPrimaryHardwareDev->get_master_volume), 745 "can't get master volume"); 746 747 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 748 mHardwareStatus = AUDIO_HW_IDLE; 749 return ret_val; 750 } 751 752 return mMasterVolume; 753} 754 755status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 756 audio_io_handle_t output) 757{ 758 // check calling permissions 759 if (!settingsAllowed()) { 760 return PERMISSION_DENIED; 761 } 762 763 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 764 ALOGE("setStreamVolume() invalid stream %d", stream); 765 return BAD_VALUE; 766 } 767 768 AutoMutex lock(mLock); 769 PlaybackThread *thread = NULL; 770 if (output) { 771 thread = checkPlaybackThread_l(output); 772 if (thread == NULL) { 773 return BAD_VALUE; 774 } 775 } 776 777 mStreamTypes[stream].volume = value; 778 779 if (thread == NULL) { 780 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 781 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 782 } 783 } else { 784 thread->setStreamVolume(stream, value); 785 } 786 787 return NO_ERROR; 788} 789 790status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 791{ 792 // check calling permissions 793 if (!settingsAllowed()) { 794 return PERMISSION_DENIED; 795 } 796 797 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 798 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 799 ALOGE("setStreamMute() invalid stream %d", stream); 800 return BAD_VALUE; 801 } 802 803 AutoMutex lock(mLock); 804 mStreamTypes[stream].mute = muted; 805 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 806 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 807 808 return NO_ERROR; 809} 810 811float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 812{ 813 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 814 return 0.0f; 815 } 816 817 AutoMutex lock(mLock); 818 float volume; 819 if (output) { 820 PlaybackThread *thread = checkPlaybackThread_l(output); 821 if (thread == NULL) { 822 return 0.0f; 823 } 824 volume = thread->streamVolume(stream); 825 } else { 826 volume = streamVolume_l(stream); 827 } 828 829 return volume; 830} 831 832bool AudioFlinger::streamMute(audio_stream_type_t stream) const 833{ 834 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 835 return true; 836 } 837 838 AutoMutex lock(mLock); 839 return streamMute_l(stream); 840} 841 842status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 843{ 844 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 845 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 846 // check calling permissions 847 if (!settingsAllowed()) { 848 return PERMISSION_DENIED; 849 } 850 851 // ioHandle == 0 means the parameters are global to the audio hardware interface 852 if (ioHandle == 0) { 853 Mutex::Autolock _l(mLock); 854 status_t final_result = NO_ERROR; 855 { 856 AutoMutex lock(mHardwareLock); 857 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 858 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 859 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 860 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 861 final_result = result ?: final_result; 862 } 863 mHardwareStatus = AUDIO_HW_IDLE; 864 } 865 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 866 AudioParameter param = AudioParameter(keyValuePairs); 867 String8 value; 868 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 869 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 870 if (mBtNrecIsOff != btNrecIsOff) { 871 for (size_t i = 0; i < mRecordThreads.size(); i++) { 872 sp<RecordThread> thread = mRecordThreads.valueAt(i); 873 RecordThread::RecordTrack *track = thread->track(); 874 if (track != NULL) { 875 audio_devices_t device = (audio_devices_t)( 876 thread->device() & AUDIO_DEVICE_IN_ALL); 877 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 878 thread->setEffectSuspended(FX_IID_AEC, 879 suspend, 880 track->sessionId()); 881 thread->setEffectSuspended(FX_IID_NS, 882 suspend, 883 track->sessionId()); 884 } 885 } 886 mBtNrecIsOff = btNrecIsOff; 887 } 888 } 889 String8 screenState; 890 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 891 bool isOff = screenState == "off"; 892 if (isOff != (gScreenState & 1)) { 893 gScreenState = ((gScreenState & ~1) + 2) | isOff; 894 } 895 } 896 return final_result; 897 } 898 899 // hold a strong ref on thread in case closeOutput() or closeInput() is called 900 // and the thread is exited once the lock is released 901 sp<ThreadBase> thread; 902 { 903 Mutex::Autolock _l(mLock); 904 thread = checkPlaybackThread_l(ioHandle); 905 if (thread == 0) { 906 thread = checkRecordThread_l(ioHandle); 907 } else if (thread == primaryPlaybackThread_l()) { 908 // indicate output device change to all input threads for pre processing 909 AudioParameter param = AudioParameter(keyValuePairs); 910 int value; 911 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 912 (value != 0)) { 913 for (size_t i = 0; i < mRecordThreads.size(); i++) { 914 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 915 } 916 } 917 } 918 } 919 if (thread != 0) { 920 return thread->setParameters(keyValuePairs); 921 } 922 return BAD_VALUE; 923} 924 925String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 926{ 927// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 928// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 929 930 Mutex::Autolock _l(mLock); 931 932 if (ioHandle == 0) { 933 String8 out_s8; 934 935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 936 char *s; 937 { 938 AutoMutex lock(mHardwareLock); 939 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 940 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 941 s = dev->get_parameters(dev, keys.string()); 942 mHardwareStatus = AUDIO_HW_IDLE; 943 } 944 out_s8 += String8(s ? s : ""); 945 free(s); 946 } 947 return out_s8; 948 } 949 950 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 951 if (playbackThread != NULL) { 952 return playbackThread->getParameters(keys); 953 } 954 RecordThread *recordThread = checkRecordThread_l(ioHandle); 955 if (recordThread != NULL) { 956 return recordThread->getParameters(keys); 957 } 958 return String8(""); 959} 960 961size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 962 audio_channel_mask_t channelMask) const 963{ 964 status_t ret = initCheck(); 965 if (ret != NO_ERROR) { 966 return 0; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 971 struct audio_config config = { 972 sample_rate: sampleRate, 973 channel_mask: channelMask, 974 format: format, 975 }; 976 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 977 mHardwareStatus = AUDIO_HW_IDLE; 978 return size; 979} 980 981unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 982{ 983 if (ioHandle == 0) { 984 return 0; 985 } 986 987 Mutex::Autolock _l(mLock); 988 989 RecordThread *recordThread = checkRecordThread_l(ioHandle); 990 if (recordThread != NULL) { 991 return recordThread->getInputFramesLost(); 992 } 993 return 0; 994} 995 996status_t AudioFlinger::setVoiceVolume(float value) 997{ 998 status_t ret = initCheck(); 999 if (ret != NO_ERROR) { 1000 return ret; 1001 } 1002 1003 // check calling permissions 1004 if (!settingsAllowed()) { 1005 return PERMISSION_DENIED; 1006 } 1007 1008 AutoMutex lock(mHardwareLock); 1009 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1010 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1011 mHardwareStatus = AUDIO_HW_IDLE; 1012 1013 return ret; 1014} 1015 1016status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1017 audio_io_handle_t output) const 1018{ 1019 status_t status; 1020 1021 Mutex::Autolock _l(mLock); 1022 1023 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1024 if (playbackThread != NULL) { 1025 return playbackThread->getRenderPosition(halFrames, dspFrames); 1026 } 1027 1028 return BAD_VALUE; 1029} 1030 1031void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1032{ 1033 1034 Mutex::Autolock _l(mLock); 1035 1036 pid_t pid = IPCThreadState::self()->getCallingPid(); 1037 if (mNotificationClients.indexOfKey(pid) < 0) { 1038 sp<NotificationClient> notificationClient = new NotificationClient(this, 1039 client, 1040 pid); 1041 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1042 1043 mNotificationClients.add(pid, notificationClient); 1044 1045 sp<IBinder> binder = client->asBinder(); 1046 binder->linkToDeath(notificationClient); 1047 1048 // the config change is always sent from playback or record threads to avoid deadlock 1049 // with AudioSystem::gLock 1050 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1051 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1052 } 1053 1054 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1055 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1056 } 1057 } 1058} 1059 1060void AudioFlinger::removeNotificationClient(pid_t pid) 1061{ 1062 Mutex::Autolock _l(mLock); 1063 1064 mNotificationClients.removeItem(pid); 1065 1066 ALOGV("%d died, releasing its sessions", pid); 1067 size_t num = mAudioSessionRefs.size(); 1068 bool removed = false; 1069 for (size_t i = 0; i< num; ) { 1070 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1071 ALOGV(" pid %d @ %d", ref->mPid, i); 1072 if (ref->mPid == pid) { 1073 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1074 mAudioSessionRefs.removeAt(i); 1075 delete ref; 1076 removed = true; 1077 num--; 1078 } else { 1079 i++; 1080 } 1081 } 1082 if (removed) { 1083 purgeStaleEffects_l(); 1084 } 1085} 1086 1087// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1089{ 1090 size_t size = mNotificationClients.size(); 1091 for (size_t i = 0; i < size; i++) { 1092 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1093 param2); 1094 } 1095} 1096 1097// removeClient_l() must be called with AudioFlinger::mLock held 1098void AudioFlinger::removeClient_l(pid_t pid) 1099{ 1100 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1101 mClients.removeItem(pid); 1102} 1103 1104// getEffectThread_l() must be called with AudioFlinger::mLock held 1105sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1106{ 1107 sp<PlaybackThread> thread; 1108 1109 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1110 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1111 ALOG_ASSERT(thread == 0); 1112 thread = mPlaybackThreads.valueAt(i); 1113 } 1114 } 1115 1116 return thread; 1117} 1118 1119// ---------------------------------------------------------------------------- 1120 1121AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1122 uint32_t device, type_t type) 1123 : Thread(false), 1124 mType(type), 1125 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1126 // mChannelMask 1127 mChannelCount(0), 1128 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1129 mParamStatus(NO_ERROR), 1130 mStandby(false), mId(id), 1131 mDevice(device), 1132 mDeathRecipient(new PMDeathRecipient(this)) 1133{ 1134} 1135 1136AudioFlinger::ThreadBase::~ThreadBase() 1137{ 1138 mParamCond.broadcast(); 1139 // do not lock the mutex in destructor 1140 releaseWakeLock_l(); 1141 if (mPowerManager != 0) { 1142 sp<IBinder> binder = mPowerManager->asBinder(); 1143 binder->unlinkToDeath(mDeathRecipient); 1144 } 1145} 1146 1147void AudioFlinger::ThreadBase::exit() 1148{ 1149 ALOGV("ThreadBase::exit"); 1150 { 1151 // This lock prevents the following race in thread (uniprocessor for illustration): 1152 // if (!exitPending()) { 1153 // // context switch from here to exit() 1154 // // exit() calls requestExit(), what exitPending() observes 1155 // // exit() calls signal(), which is dropped since no waiters 1156 // // context switch back from exit() to here 1157 // mWaitWorkCV.wait(...); 1158 // // now thread is hung 1159 // } 1160 AutoMutex lock(mLock); 1161 requestExit(); 1162 mWaitWorkCV.signal(); 1163 } 1164 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1165 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1166 requestExitAndWait(); 1167} 1168 1169status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1170{ 1171 status_t status; 1172 1173 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1174 Mutex::Autolock _l(mLock); 1175 1176 mNewParameters.add(keyValuePairs); 1177 mWaitWorkCV.signal(); 1178 // wait condition with timeout in case the thread loop has exited 1179 // before the request could be processed 1180 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1181 status = mParamStatus; 1182 mWaitWorkCV.signal(); 1183 } else { 1184 status = TIMED_OUT; 1185 } 1186 return status; 1187} 1188 1189void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1190{ 1191 Mutex::Autolock _l(mLock); 1192 sendConfigEvent_l(event, param); 1193} 1194 1195// sendConfigEvent_l() must be called with ThreadBase::mLock held 1196void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1197{ 1198 ConfigEvent configEvent; 1199 configEvent.mEvent = event; 1200 configEvent.mParam = param; 1201 mConfigEvents.add(configEvent); 1202 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1203 mWaitWorkCV.signal(); 1204} 1205 1206void AudioFlinger::ThreadBase::processConfigEvents() 1207{ 1208 mLock.lock(); 1209 while (!mConfigEvents.isEmpty()) { 1210 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1211 ConfigEvent configEvent = mConfigEvents[0]; 1212 mConfigEvents.removeAt(0); 1213 // release mLock before locking AudioFlinger mLock: lock order is always 1214 // AudioFlinger then ThreadBase to avoid cross deadlock 1215 mLock.unlock(); 1216 mAudioFlinger->mLock.lock(); 1217 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1218 mAudioFlinger->mLock.unlock(); 1219 mLock.lock(); 1220 } 1221 mLock.unlock(); 1222} 1223 1224status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1225{ 1226 const size_t SIZE = 256; 1227 char buffer[SIZE]; 1228 String8 result; 1229 1230 bool locked = tryLock(mLock); 1231 if (!locked) { 1232 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1233 write(fd, buffer, strlen(buffer)); 1234 } 1235 1236 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1237 result.append(buffer); 1238 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1239 result.append(buffer); 1240 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1245 result.append(buffer); 1246 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1247 result.append(buffer); 1248 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1249 result.append(buffer); 1250 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1251 result.append(buffer); 1252 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1253 result.append(buffer); 1254 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1255 result.append(buffer); 1256 1257 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1258 result.append(buffer); 1259 result.append(" Index Command"); 1260 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1261 snprintf(buffer, SIZE, "\n %02d ", i); 1262 result.append(buffer); 1263 result.append(mNewParameters[i]); 1264 } 1265 1266 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1267 result.append(buffer); 1268 snprintf(buffer, SIZE, " Index event param\n"); 1269 result.append(buffer); 1270 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1271 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1272 result.append(buffer); 1273 } 1274 result.append("\n"); 1275 1276 write(fd, result.string(), result.size()); 1277 1278 if (locked) { 1279 mLock.unlock(); 1280 } 1281 return NO_ERROR; 1282} 1283 1284status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1285{ 1286 const size_t SIZE = 256; 1287 char buffer[SIZE]; 1288 String8 result; 1289 1290 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1291 write(fd, buffer, strlen(buffer)); 1292 1293 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1294 sp<EffectChain> chain = mEffectChains[i]; 1295 if (chain != 0) { 1296 chain->dump(fd, args); 1297 } 1298 } 1299 return NO_ERROR; 1300} 1301 1302void AudioFlinger::ThreadBase::acquireWakeLock() 1303{ 1304 Mutex::Autolock _l(mLock); 1305 acquireWakeLock_l(); 1306} 1307 1308void AudioFlinger::ThreadBase::acquireWakeLock_l() 1309{ 1310 if (mPowerManager == 0) { 1311 // use checkService() to avoid blocking if power service is not up yet 1312 sp<IBinder> binder = 1313 defaultServiceManager()->checkService(String16("power")); 1314 if (binder == 0) { 1315 ALOGW("Thread %s cannot connect to the power manager service", mName); 1316 } else { 1317 mPowerManager = interface_cast<IPowerManager>(binder); 1318 binder->linkToDeath(mDeathRecipient); 1319 } 1320 } 1321 if (mPowerManager != 0) { 1322 sp<IBinder> binder = new BBinder(); 1323 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1324 binder, 1325 String16(mName)); 1326 if (status == NO_ERROR) { 1327 mWakeLockToken = binder; 1328 } 1329 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1330 } 1331} 1332 1333void AudioFlinger::ThreadBase::releaseWakeLock() 1334{ 1335 Mutex::Autolock _l(mLock); 1336 releaseWakeLock_l(); 1337} 1338 1339void AudioFlinger::ThreadBase::releaseWakeLock_l() 1340{ 1341 if (mWakeLockToken != 0) { 1342 ALOGV("releaseWakeLock_l() %s", mName); 1343 if (mPowerManager != 0) { 1344 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1345 } 1346 mWakeLockToken.clear(); 1347 } 1348} 1349 1350void AudioFlinger::ThreadBase::clearPowerManager() 1351{ 1352 Mutex::Autolock _l(mLock); 1353 releaseWakeLock_l(); 1354 mPowerManager.clear(); 1355} 1356 1357void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1358{ 1359 sp<ThreadBase> thread = mThread.promote(); 1360 if (thread != 0) { 1361 thread->clearPowerManager(); 1362 } 1363 ALOGW("power manager service died !!!"); 1364} 1365 1366void AudioFlinger::ThreadBase::setEffectSuspended( 1367 const effect_uuid_t *type, bool suspend, int sessionId) 1368{ 1369 Mutex::Autolock _l(mLock); 1370 setEffectSuspended_l(type, suspend, sessionId); 1371} 1372 1373void AudioFlinger::ThreadBase::setEffectSuspended_l( 1374 const effect_uuid_t *type, bool suspend, int sessionId) 1375{ 1376 sp<EffectChain> chain = getEffectChain_l(sessionId); 1377 if (chain != 0) { 1378 if (type != NULL) { 1379 chain->setEffectSuspended_l(type, suspend); 1380 } else { 1381 chain->setEffectSuspendedAll_l(suspend); 1382 } 1383 } 1384 1385 updateSuspendedSessions_l(type, suspend, sessionId); 1386} 1387 1388void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1389{ 1390 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1391 if (index < 0) { 1392 return; 1393 } 1394 1395 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1396 mSuspendedSessions.editValueAt(index); 1397 1398 for (size_t i = 0; i < sessionEffects.size(); i++) { 1399 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1400 for (int j = 0; j < desc->mRefCount; j++) { 1401 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1402 chain->setEffectSuspendedAll_l(true); 1403 } else { 1404 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1405 desc->mType.timeLow); 1406 chain->setEffectSuspended_l(&desc->mType, true); 1407 } 1408 } 1409 } 1410} 1411 1412void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1413 bool suspend, 1414 int sessionId) 1415{ 1416 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1417 1418 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1419 1420 if (suspend) { 1421 if (index >= 0) { 1422 sessionEffects = mSuspendedSessions.editValueAt(index); 1423 } else { 1424 mSuspendedSessions.add(sessionId, sessionEffects); 1425 } 1426 } else { 1427 if (index < 0) { 1428 return; 1429 } 1430 sessionEffects = mSuspendedSessions.editValueAt(index); 1431 } 1432 1433 1434 int key = EffectChain::kKeyForSuspendAll; 1435 if (type != NULL) { 1436 key = type->timeLow; 1437 } 1438 index = sessionEffects.indexOfKey(key); 1439 1440 sp<SuspendedSessionDesc> desc; 1441 if (suspend) { 1442 if (index >= 0) { 1443 desc = sessionEffects.valueAt(index); 1444 } else { 1445 desc = new SuspendedSessionDesc(); 1446 if (type != NULL) { 1447 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1448 } 1449 sessionEffects.add(key, desc); 1450 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1451 } 1452 desc->mRefCount++; 1453 } else { 1454 if (index < 0) { 1455 return; 1456 } 1457 desc = sessionEffects.valueAt(index); 1458 if (--desc->mRefCount == 0) { 1459 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1460 sessionEffects.removeItemsAt(index); 1461 if (sessionEffects.isEmpty()) { 1462 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1463 sessionId); 1464 mSuspendedSessions.removeItem(sessionId); 1465 } 1466 } 1467 } 1468 if (!sessionEffects.isEmpty()) { 1469 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1470 } 1471} 1472 1473void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1474 bool enabled, 1475 int sessionId) 1476{ 1477 Mutex::Autolock _l(mLock); 1478 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1479} 1480 1481void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1482 bool enabled, 1483 int sessionId) 1484{ 1485 if (mType != RECORD) { 1486 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1487 // another session. This gives the priority to well behaved effect control panels 1488 // and applications not using global effects. 1489 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1490 // global effects 1491 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1492 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1493 } 1494 } 1495 1496 sp<EffectChain> chain = getEffectChain_l(sessionId); 1497 if (chain != 0) { 1498 chain->checkSuspendOnEffectEnabled(effect, enabled); 1499 } 1500} 1501 1502// ---------------------------------------------------------------------------- 1503 1504AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1505 AudioStreamOut* output, 1506 audio_io_handle_t id, 1507 uint32_t device, 1508 type_t type) 1509 : ThreadBase(audioFlinger, id, device, type), 1510 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1511 // Assumes constructor is called by AudioFlinger with it's mLock held, 1512 // but it would be safer to explicitly pass initial masterMute as parameter 1513 mMasterMute(audioFlinger->masterMute_l()), 1514 // mStreamTypes[] initialized in constructor body 1515 mOutput(output), 1516 // Assumes constructor is called by AudioFlinger with it's mLock held, 1517 // but it would be safer to explicitly pass initial masterVolume as parameter 1518 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1519 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1520 mMixerStatus(MIXER_IDLE), 1521 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1522 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1523 mScreenState(gScreenState), 1524 // index 0 is reserved for normal mixer's submix 1525 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1526{ 1527 snprintf(mName, kNameLength, "AudioOut_%X", id); 1528 1529 readOutputParameters(); 1530 1531 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1532 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1533 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1534 stream = (audio_stream_type_t) (stream + 1)) { 1535 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1536 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1537 } 1538 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1539 // because mAudioFlinger doesn't have one to copy from 1540} 1541 1542AudioFlinger::PlaybackThread::~PlaybackThread() 1543{ 1544 delete [] mMixBuffer; 1545} 1546 1547status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1548{ 1549 dumpInternals(fd, args); 1550 dumpTracks(fd, args); 1551 dumpEffectChains(fd, args); 1552 return NO_ERROR; 1553} 1554 1555status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1556{ 1557 const size_t SIZE = 256; 1558 char buffer[SIZE]; 1559 String8 result; 1560 1561 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1562 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1563 const stream_type_t *st = &mStreamTypes[i]; 1564 if (i > 0) { 1565 result.appendFormat(", "); 1566 } 1567 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1568 if (st->mute) { 1569 result.append("M"); 1570 } 1571 } 1572 result.append("\n"); 1573 write(fd, result.string(), result.length()); 1574 result.clear(); 1575 1576 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1577 result.append(buffer); 1578 Track::appendDumpHeader(result); 1579 for (size_t i = 0; i < mTracks.size(); ++i) { 1580 sp<Track> track = mTracks[i]; 1581 if (track != 0) { 1582 track->dump(buffer, SIZE); 1583 result.append(buffer); 1584 } 1585 } 1586 1587 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1588 result.append(buffer); 1589 Track::appendDumpHeader(result); 1590 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1591 sp<Track> track = mActiveTracks[i].promote(); 1592 if (track != 0) { 1593 track->dump(buffer, SIZE); 1594 result.append(buffer); 1595 } 1596 } 1597 write(fd, result.string(), result.size()); 1598 1599 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1600 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1601 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1602 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1603 1604 return NO_ERROR; 1605} 1606 1607status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1608{ 1609 const size_t SIZE = 256; 1610 char buffer[SIZE]; 1611 String8 result; 1612 1613 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1614 result.append(buffer); 1615 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1616 result.append(buffer); 1617 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1618 result.append(buffer); 1619 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1620 result.append(buffer); 1621 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1622 result.append(buffer); 1623 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1624 result.append(buffer); 1625 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1626 result.append(buffer); 1627 write(fd, result.string(), result.size()); 1628 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1629 1630 dumpBase(fd, args); 1631 1632 return NO_ERROR; 1633} 1634 1635// Thread virtuals 1636status_t AudioFlinger::PlaybackThread::readyToRun() 1637{ 1638 status_t status = initCheck(); 1639 if (status == NO_ERROR) { 1640 ALOGI("AudioFlinger's thread %p ready to run", this); 1641 } else { 1642 ALOGE("No working audio driver found."); 1643 } 1644 return status; 1645} 1646 1647void AudioFlinger::PlaybackThread::onFirstRef() 1648{ 1649 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1650} 1651 1652// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1653sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1654 const sp<AudioFlinger::Client>& client, 1655 audio_stream_type_t streamType, 1656 uint32_t sampleRate, 1657 audio_format_t format, 1658 uint32_t channelMask, 1659 int frameCount, 1660 const sp<IMemory>& sharedBuffer, 1661 int sessionId, 1662 IAudioFlinger::track_flags_t flags, 1663 pid_t tid, 1664 status_t *status) 1665{ 1666 sp<Track> track; 1667 status_t lStatus; 1668 1669 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1670 1671 // client expresses a preference for FAST, but we get the final say 1672 if (flags & IAudioFlinger::TRACK_FAST) { 1673 if ( 1674 // not timed 1675 (!isTimed) && 1676 // either of these use cases: 1677 ( 1678 // use case 1: shared buffer with any frame count 1679 ( 1680 (sharedBuffer != 0) 1681 ) || 1682 // use case 2: callback handler and frame count is default or at least as large as HAL 1683 ( 1684 (tid != -1) && 1685 ((frameCount == 0) || 1686 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1687 ) 1688 ) && 1689 // PCM data 1690 audio_is_linear_pcm(format) && 1691 // mono or stereo 1692 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1693 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1694#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1695 // hardware sample rate 1696 (sampleRate == mSampleRate) && 1697#endif 1698 // normal mixer has an associated fast mixer 1699 hasFastMixer() && 1700 // there are sufficient fast track slots available 1701 (mFastTrackAvailMask != 0) 1702 // FIXME test that MixerThread for this fast track has a capable output HAL 1703 // FIXME add a permission test also? 1704 ) { 1705 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1706 if (frameCount == 0) { 1707 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1708 } 1709 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1710 frameCount, mFrameCount); 1711 } else { 1712 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1713 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1714 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1715 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1716 audio_is_linear_pcm(format), 1717 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1718 flags &= ~IAudioFlinger::TRACK_FAST; 1719 // For compatibility with AudioTrack calculation, buffer depth is forced 1720 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1721 // This is probably too conservative, but legacy application code may depend on it. 1722 // If you change this calculation, also review the start threshold which is related. 1723 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1724 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1725 if (minBufCount < 2) { 1726 minBufCount = 2; 1727 } 1728 int minFrameCount = mNormalFrameCount * minBufCount; 1729 if (frameCount < minFrameCount) { 1730 frameCount = minFrameCount; 1731 } 1732 } 1733 } 1734 1735 if (mType == DIRECT) { 1736 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1737 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1738 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1739 "for output %p with format %d", 1740 sampleRate, format, channelMask, mOutput, mFormat); 1741 lStatus = BAD_VALUE; 1742 goto Exit; 1743 } 1744 } 1745 } else { 1746 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1747 if (sampleRate > mSampleRate*2) { 1748 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1749 lStatus = BAD_VALUE; 1750 goto Exit; 1751 } 1752 } 1753 1754 lStatus = initCheck(); 1755 if (lStatus != NO_ERROR) { 1756 ALOGE("Audio driver not initialized."); 1757 goto Exit; 1758 } 1759 1760 { // scope for mLock 1761 Mutex::Autolock _l(mLock); 1762 1763 // all tracks in same audio session must share the same routing strategy otherwise 1764 // conflicts will happen when tracks are moved from one output to another by audio policy 1765 // manager 1766 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1767 for (size_t i = 0; i < mTracks.size(); ++i) { 1768 sp<Track> t = mTracks[i]; 1769 if (t != 0 && !t->isOutputTrack()) { 1770 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1771 if (sessionId == t->sessionId() && strategy != actual) { 1772 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1773 strategy, actual); 1774 lStatus = BAD_VALUE; 1775 goto Exit; 1776 } 1777 } 1778 } 1779 1780 if (!isTimed) { 1781 track = new Track(this, client, streamType, sampleRate, format, 1782 channelMask, frameCount, sharedBuffer, sessionId, flags); 1783 } else { 1784 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1785 channelMask, frameCount, sharedBuffer, sessionId); 1786 } 1787 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1788 lStatus = NO_MEMORY; 1789 goto Exit; 1790 } 1791 mTracks.add(track); 1792 1793 sp<EffectChain> chain = getEffectChain_l(sessionId); 1794 if (chain != 0) { 1795 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1796 track->setMainBuffer(chain->inBuffer()); 1797 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1798 chain->incTrackCnt(); 1799 } 1800 } 1801 1802 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1803 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1804 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1805 // so ask activity manager to do this on our behalf 1806 int err = requestPriority(callingPid, tid, 1); 1807 if (err != 0) { 1808 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1809 1, callingPid, tid, err); 1810 } 1811 } 1812 1813 lStatus = NO_ERROR; 1814 1815Exit: 1816 if (status) { 1817 *status = lStatus; 1818 } 1819 return track; 1820} 1821 1822uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1823{ 1824 if (mFastMixer != NULL) { 1825 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1826 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1827 } 1828 return latency; 1829} 1830 1831uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1832{ 1833 return latency; 1834} 1835 1836uint32_t AudioFlinger::PlaybackThread::latency() const 1837{ 1838 Mutex::Autolock _l(mLock); 1839 return latency_l(); 1840} 1841uint32_t AudioFlinger::PlaybackThread::latency_l() const 1842{ 1843 if (initCheck() == NO_ERROR) { 1844 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1845 } else { 1846 return 0; 1847 } 1848} 1849 1850void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1851{ 1852 Mutex::Autolock _l(mLock); 1853 mMasterVolume = value; 1854} 1855 1856void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1857{ 1858 Mutex::Autolock _l(mLock); 1859 setMasterMute_l(muted); 1860} 1861 1862void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1863{ 1864 Mutex::Autolock _l(mLock); 1865 mStreamTypes[stream].volume = value; 1866} 1867 1868void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1869{ 1870 Mutex::Autolock _l(mLock); 1871 mStreamTypes[stream].mute = muted; 1872} 1873 1874float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1875{ 1876 Mutex::Autolock _l(mLock); 1877 return mStreamTypes[stream].volume; 1878} 1879 1880// addTrack_l() must be called with ThreadBase::mLock held 1881status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1882{ 1883 status_t status = ALREADY_EXISTS; 1884 1885 // set retry count for buffer fill 1886 track->mRetryCount = kMaxTrackStartupRetries; 1887 if (mActiveTracks.indexOf(track) < 0) { 1888 // the track is newly added, make sure it fills up all its 1889 // buffers before playing. This is to ensure the client will 1890 // effectively get the latency it requested. 1891 track->mFillingUpStatus = Track::FS_FILLING; 1892 track->mResetDone = false; 1893 track->mPresentationCompleteFrames = 0; 1894 mActiveTracks.add(track); 1895 if (track->mainBuffer() != mMixBuffer) { 1896 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1897 if (chain != 0) { 1898 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1899 chain->incActiveTrackCnt(); 1900 } 1901 } 1902 1903 status = NO_ERROR; 1904 } 1905 1906 ALOGV("mWaitWorkCV.broadcast"); 1907 mWaitWorkCV.broadcast(); 1908 1909 return status; 1910} 1911 1912// destroyTrack_l() must be called with ThreadBase::mLock held 1913void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1914{ 1915 track->mState = TrackBase::TERMINATED; 1916 // active tracks are removed by threadLoop() 1917 if (mActiveTracks.indexOf(track) < 0) { 1918 removeTrack_l(track); 1919 } 1920} 1921 1922void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1923{ 1924 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1925 mTracks.remove(track); 1926 deleteTrackName_l(track->name()); 1927 // redundant as track is about to be destroyed, for dumpsys only 1928 track->mName = -1; 1929 if (track->isFastTrack()) { 1930 int index = track->mFastIndex; 1931 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1932 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1933 mFastTrackAvailMask |= 1 << index; 1934 // redundant as track is about to be destroyed, for dumpsys only 1935 track->mFastIndex = -1; 1936 } 1937 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1938 if (chain != 0) { 1939 chain->decTrackCnt(); 1940 } 1941} 1942 1943String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1944{ 1945 String8 out_s8 = String8(""); 1946 char *s; 1947 1948 Mutex::Autolock _l(mLock); 1949 if (initCheck() != NO_ERROR) { 1950 return out_s8; 1951 } 1952 1953 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1954 out_s8 = String8(s); 1955 free(s); 1956 return out_s8; 1957} 1958 1959// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1960void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1961 AudioSystem::OutputDescriptor desc; 1962 void *param2 = NULL; 1963 1964 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1965 1966 switch (event) { 1967 case AudioSystem::OUTPUT_OPENED: 1968 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1969 desc.channels = mChannelMask; 1970 desc.samplingRate = mSampleRate; 1971 desc.format = mFormat; 1972 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1973 desc.latency = latency(); 1974 param2 = &desc; 1975 break; 1976 1977 case AudioSystem::STREAM_CONFIG_CHANGED: 1978 param2 = ¶m; 1979 case AudioSystem::OUTPUT_CLOSED: 1980 default: 1981 break; 1982 } 1983 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1984} 1985 1986void AudioFlinger::PlaybackThread::readOutputParameters() 1987{ 1988 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1989 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1990 mChannelCount = (uint16_t)popcount(mChannelMask); 1991 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1992 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1993 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1994 if (mFrameCount & 15) { 1995 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1996 mFrameCount); 1997 } 1998 1999 // Calculate size of normal mix buffer relative to the HAL output buffer size 2000 double multiplier = 1.0; 2001 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2002 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2003 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2004 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2005 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2006 maxNormalFrameCount = maxNormalFrameCount & ~15; 2007 if (maxNormalFrameCount < minNormalFrameCount) { 2008 maxNormalFrameCount = minNormalFrameCount; 2009 } 2010 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2011 if (multiplier <= 1.0) { 2012 multiplier = 1.0; 2013 } else if (multiplier <= 2.0) { 2014 if (2 * mFrameCount <= maxNormalFrameCount) { 2015 multiplier = 2.0; 2016 } else { 2017 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2018 } 2019 } else { 2020 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2021 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2022 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2023 // FIXME this rounding up should not be done if no HAL SRC 2024 uint32_t truncMult = (uint32_t) multiplier; 2025 if ((truncMult & 1)) { 2026 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2027 ++truncMult; 2028 } 2029 } 2030 multiplier = (double) truncMult; 2031 } 2032 } 2033 mNormalFrameCount = multiplier * mFrameCount; 2034 // round up to nearest 16 frames to satisfy AudioMixer 2035 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2036 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2037 2038 delete[] mMixBuffer; 2039 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2040 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2041 2042 // force reconfiguration of effect chains and engines to take new buffer size and audio 2043 // parameters into account 2044 // Note that mLock is not held when readOutputParameters() is called from the constructor 2045 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2046 // matter. 2047 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2048 Vector< sp<EffectChain> > effectChains = mEffectChains; 2049 for (size_t i = 0; i < effectChains.size(); i ++) { 2050 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2051 } 2052} 2053 2054 2055status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2056{ 2057 if (halFrames == NULL || dspFrames == NULL) { 2058 return BAD_VALUE; 2059 } 2060 Mutex::Autolock _l(mLock); 2061 if (initCheck() != NO_ERROR) { 2062 return INVALID_OPERATION; 2063 } 2064 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2065 2066 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2067} 2068 2069uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2070{ 2071 Mutex::Autolock _l(mLock); 2072 uint32_t result = 0; 2073 if (getEffectChain_l(sessionId) != 0) { 2074 result = EFFECT_SESSION; 2075 } 2076 2077 for (size_t i = 0; i < mTracks.size(); ++i) { 2078 sp<Track> track = mTracks[i]; 2079 if (sessionId == track->sessionId() && 2080 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2081 result |= TRACK_SESSION; 2082 break; 2083 } 2084 } 2085 2086 return result; 2087} 2088 2089uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2090{ 2091 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2092 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2093 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2094 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2095 } 2096 for (size_t i = 0; i < mTracks.size(); i++) { 2097 sp<Track> track = mTracks[i]; 2098 if (sessionId == track->sessionId() && 2099 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2100 return AudioSystem::getStrategyForStream(track->streamType()); 2101 } 2102 } 2103 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2104} 2105 2106 2107AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2108{ 2109 Mutex::Autolock _l(mLock); 2110 return mOutput; 2111} 2112 2113AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2114{ 2115 Mutex::Autolock _l(mLock); 2116 AudioStreamOut *output = mOutput; 2117 mOutput = NULL; 2118 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2119 // must push a NULL and wait for ack 2120 mOutputSink.clear(); 2121 mPipeSink.clear(); 2122 mNormalSink.clear(); 2123 return output; 2124} 2125 2126// this method must always be called either with ThreadBase mLock held or inside the thread loop 2127audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2128{ 2129 if (mOutput == NULL) { 2130 return NULL; 2131 } 2132 return &mOutput->stream->common; 2133} 2134 2135uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2136{ 2137 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2138} 2139 2140status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2141{ 2142 if (!isValidSyncEvent(event)) { 2143 return BAD_VALUE; 2144 } 2145 2146 Mutex::Autolock _l(mLock); 2147 2148 for (size_t i = 0; i < mTracks.size(); ++i) { 2149 sp<Track> track = mTracks[i]; 2150 if (event->triggerSession() == track->sessionId()) { 2151 track->setSyncEvent(event); 2152 return NO_ERROR; 2153 } 2154 } 2155 2156 return NAME_NOT_FOUND; 2157} 2158 2159bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2160{ 2161 switch (event->type()) { 2162 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2163 return true; 2164 default: 2165 break; 2166 } 2167 return false; 2168} 2169 2170void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2171{ 2172 size_t count = tracksToRemove.size(); 2173 if (CC_UNLIKELY(count)) { 2174 for (size_t i = 0 ; i < count ; i++) { 2175 const sp<Track>& track = tracksToRemove.itemAt(i); 2176 if ((track->sharedBuffer() != 0) && 2177 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2178 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2179 } 2180 } 2181 } 2182 2183} 2184 2185// ---------------------------------------------------------------------------- 2186 2187AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2188 audio_io_handle_t id, uint32_t device, type_t type) 2189 : PlaybackThread(audioFlinger, output, id, device, type), 2190 // mAudioMixer below 2191 // mFastMixer below 2192 mFastMixerFutex(0) 2193 // mOutputSink below 2194 // mPipeSink below 2195 // mNormalSink below 2196{ 2197 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2198 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2199 "mFrameCount=%d, mNormalFrameCount=%d", 2200 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2201 mNormalFrameCount); 2202 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2203 2204 // FIXME - Current mixer implementation only supports stereo output 2205 if (mChannelCount == 1) { 2206 ALOGE("Invalid audio hardware channel count"); 2207 } 2208 2209 // create an NBAIO sink for the HAL output stream, and negotiate 2210 mOutputSink = new AudioStreamOutSink(output->stream); 2211 size_t numCounterOffers = 0; 2212 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2213 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2214 ALOG_ASSERT(index == 0); 2215 2216 // initialize fast mixer depending on configuration 2217 bool initFastMixer; 2218 switch (kUseFastMixer) { 2219 case FastMixer_Never: 2220 initFastMixer = false; 2221 break; 2222 case FastMixer_Always: 2223 initFastMixer = true; 2224 break; 2225 case FastMixer_Static: 2226 case FastMixer_Dynamic: 2227 initFastMixer = mFrameCount < mNormalFrameCount; 2228 break; 2229 } 2230 if (initFastMixer) { 2231 2232 // create a MonoPipe to connect our submix to FastMixer 2233 NBAIO_Format format = mOutputSink->format(); 2234 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2235 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2236 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2237 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2238 const NBAIO_Format offers[1] = {format}; 2239 size_t numCounterOffers = 0; 2240 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2241 ALOG_ASSERT(index == 0); 2242 monoPipe->setAvgFrames((mScreenState & 1) ? 2243 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2244 mPipeSink = monoPipe; 2245 2246#ifdef TEE_SINK_FRAMES 2247 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2248 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2249 numCounterOffers = 0; 2250 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2251 ALOG_ASSERT(index == 0); 2252 mTeeSink = teeSink; 2253 PipeReader *teeSource = new PipeReader(*teeSink); 2254 numCounterOffers = 0; 2255 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2256 ALOG_ASSERT(index == 0); 2257 mTeeSource = teeSource; 2258#endif 2259 2260 // create fast mixer and configure it initially with just one fast track for our submix 2261 mFastMixer = new FastMixer(); 2262 FastMixerStateQueue *sq = mFastMixer->sq(); 2263#ifdef STATE_QUEUE_DUMP 2264 sq->setObserverDump(&mStateQueueObserverDump); 2265 sq->setMutatorDump(&mStateQueueMutatorDump); 2266#endif 2267 FastMixerState *state = sq->begin(); 2268 FastTrack *fastTrack = &state->mFastTracks[0]; 2269 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2270 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2271 fastTrack->mVolumeProvider = NULL; 2272 fastTrack->mGeneration++; 2273 state->mFastTracksGen++; 2274 state->mTrackMask = 1; 2275 // fast mixer will use the HAL output sink 2276 state->mOutputSink = mOutputSink.get(); 2277 state->mOutputSinkGen++; 2278 state->mFrameCount = mFrameCount; 2279 state->mCommand = FastMixerState::COLD_IDLE; 2280 // already done in constructor initialization list 2281 //mFastMixerFutex = 0; 2282 state->mColdFutexAddr = &mFastMixerFutex; 2283 state->mColdGen++; 2284 state->mDumpState = &mFastMixerDumpState; 2285 state->mTeeSink = mTeeSink.get(); 2286 sq->end(); 2287 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2288 2289 // start the fast mixer 2290 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2291 pid_t tid = mFastMixer->getTid(); 2292 int err = requestPriority(getpid_cached, tid, 2); 2293 if (err != 0) { 2294 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2295 2, getpid_cached, tid, err); 2296 } 2297 2298#ifdef AUDIO_WATCHDOG 2299 // create and start the watchdog 2300 mAudioWatchdog = new AudioWatchdog(); 2301 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2302 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2303 tid = mAudioWatchdog->getTid(); 2304 err = requestPriority(getpid_cached, tid, 1); 2305 if (err != 0) { 2306 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2307 1, getpid_cached, tid, err); 2308 } 2309#endif 2310 2311 } else { 2312 mFastMixer = NULL; 2313 } 2314 2315 switch (kUseFastMixer) { 2316 case FastMixer_Never: 2317 case FastMixer_Dynamic: 2318 mNormalSink = mOutputSink; 2319 break; 2320 case FastMixer_Always: 2321 mNormalSink = mPipeSink; 2322 break; 2323 case FastMixer_Static: 2324 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2325 break; 2326 } 2327} 2328 2329AudioFlinger::MixerThread::~MixerThread() 2330{ 2331 if (mFastMixer != NULL) { 2332 FastMixerStateQueue *sq = mFastMixer->sq(); 2333 FastMixerState *state = sq->begin(); 2334 if (state->mCommand == FastMixerState::COLD_IDLE) { 2335 int32_t old = android_atomic_inc(&mFastMixerFutex); 2336 if (old == -1) { 2337 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2338 } 2339 } 2340 state->mCommand = FastMixerState::EXIT; 2341 sq->end(); 2342 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2343 mFastMixer->join(); 2344 // Though the fast mixer thread has exited, it's state queue is still valid. 2345 // We'll use that extract the final state which contains one remaining fast track 2346 // corresponding to our sub-mix. 2347 state = sq->begin(); 2348 ALOG_ASSERT(state->mTrackMask == 1); 2349 FastTrack *fastTrack = &state->mFastTracks[0]; 2350 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2351 delete fastTrack->mBufferProvider; 2352 sq->end(false /*didModify*/); 2353 delete mFastMixer; 2354 if (mAudioWatchdog != 0) { 2355 mAudioWatchdog->requestExit(); 2356 mAudioWatchdog->requestExitAndWait(); 2357 mAudioWatchdog.clear(); 2358 } 2359 } 2360 delete mAudioMixer; 2361} 2362 2363class CpuStats { 2364public: 2365 CpuStats(); 2366 void sample(const String8 &title); 2367#ifdef DEBUG_CPU_USAGE 2368private: 2369 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2370 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2371 2372 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2373 2374 int mCpuNum; // thread's current CPU number 2375 int mCpukHz; // frequency of thread's current CPU in kHz 2376#endif 2377}; 2378 2379CpuStats::CpuStats() 2380#ifdef DEBUG_CPU_USAGE 2381 : mCpuNum(-1), mCpukHz(-1) 2382#endif 2383{ 2384} 2385 2386void CpuStats::sample(const String8 &title) { 2387#ifdef DEBUG_CPU_USAGE 2388 // get current thread's delta CPU time in wall clock ns 2389 double wcNs; 2390 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2391 2392 // record sample for wall clock statistics 2393 if (valid) { 2394 mWcStats.sample(wcNs); 2395 } 2396 2397 // get the current CPU number 2398 int cpuNum = sched_getcpu(); 2399 2400 // get the current CPU frequency in kHz 2401 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2402 2403 // check if either CPU number or frequency changed 2404 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2405 mCpuNum = cpuNum; 2406 mCpukHz = cpukHz; 2407 // ignore sample for purposes of cycles 2408 valid = false; 2409 } 2410 2411 // if no change in CPU number or frequency, then record sample for cycle statistics 2412 if (valid && mCpukHz > 0) { 2413 double cycles = wcNs * cpukHz * 0.000001; 2414 mHzStats.sample(cycles); 2415 } 2416 2417 unsigned n = mWcStats.n(); 2418 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2419 if ((n & 127) == 1) { 2420 long long elapsed = mCpuUsage.elapsed(); 2421 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2422 double perLoop = elapsed / (double) n; 2423 double perLoop100 = perLoop * 0.01; 2424 double perLoop1k = perLoop * 0.001; 2425 double mean = mWcStats.mean(); 2426 double stddev = mWcStats.stddev(); 2427 double minimum = mWcStats.minimum(); 2428 double maximum = mWcStats.maximum(); 2429 double meanCycles = mHzStats.mean(); 2430 double stddevCycles = mHzStats.stddev(); 2431 double minCycles = mHzStats.minimum(); 2432 double maxCycles = mHzStats.maximum(); 2433 mCpuUsage.resetElapsed(); 2434 mWcStats.reset(); 2435 mHzStats.reset(); 2436 ALOGD("CPU usage for %s over past %.1f secs\n" 2437 " (%u mixer loops at %.1f mean ms per loop):\n" 2438 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2439 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2440 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2441 title.string(), 2442 elapsed * .000000001, n, perLoop * .000001, 2443 mean * .001, 2444 stddev * .001, 2445 minimum * .001, 2446 maximum * .001, 2447 mean / perLoop100, 2448 stddev / perLoop100, 2449 minimum / perLoop100, 2450 maximum / perLoop100, 2451 meanCycles / perLoop1k, 2452 stddevCycles / perLoop1k, 2453 minCycles / perLoop1k, 2454 maxCycles / perLoop1k); 2455 2456 } 2457 } 2458#endif 2459}; 2460 2461void AudioFlinger::PlaybackThread::checkSilentMode_l() 2462{ 2463 if (!mMasterMute) { 2464 char value[PROPERTY_VALUE_MAX]; 2465 if (property_get("ro.audio.silent", value, "0") > 0) { 2466 char *endptr; 2467 unsigned long ul = strtoul(value, &endptr, 0); 2468 if (*endptr == '\0' && ul != 0) { 2469 ALOGD("Silence is golden"); 2470 // The setprop command will not allow a property to be changed after 2471 // the first time it is set, so we don't have to worry about un-muting. 2472 setMasterMute_l(true); 2473 } 2474 } 2475 } 2476} 2477 2478bool AudioFlinger::PlaybackThread::threadLoop() 2479{ 2480 Vector< sp<Track> > tracksToRemove; 2481 2482 standbyTime = systemTime(); 2483 2484 // MIXER 2485 nsecs_t lastWarning = 0; 2486 2487 // DUPLICATING 2488 // FIXME could this be made local to while loop? 2489 writeFrames = 0; 2490 2491 cacheParameters_l(); 2492 sleepTime = idleSleepTime; 2493 2494if (mType == MIXER) { 2495 sleepTimeShift = 0; 2496} 2497 2498 CpuStats cpuStats; 2499 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2500 2501 acquireWakeLock(); 2502 2503 while (!exitPending()) 2504 { 2505 cpuStats.sample(myName); 2506 2507 Vector< sp<EffectChain> > effectChains; 2508 2509 processConfigEvents(); 2510 2511 { // scope for mLock 2512 2513 Mutex::Autolock _l(mLock); 2514 2515 if (checkForNewParameters_l()) { 2516 cacheParameters_l(); 2517 } 2518 2519 saveOutputTracks(); 2520 2521 // put audio hardware into standby after short delay 2522 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2523 mSuspended > 0)) { 2524 if (!mStandby) { 2525 2526 threadLoop_standby(); 2527 2528 mStandby = true; 2529 mBytesWritten = 0; 2530 } 2531 2532 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2533 // we're about to wait, flush the binder command buffer 2534 IPCThreadState::self()->flushCommands(); 2535 2536 clearOutputTracks(); 2537 2538 if (exitPending()) break; 2539 2540 releaseWakeLock_l(); 2541 // wait until we have something to do... 2542 ALOGV("%s going to sleep", myName.string()); 2543 mWaitWorkCV.wait(mLock); 2544 ALOGV("%s waking up", myName.string()); 2545 acquireWakeLock_l(); 2546 2547 mMixerStatus = MIXER_IDLE; 2548 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2549 2550 checkSilentMode_l(); 2551 2552 standbyTime = systemTime() + standbyDelay; 2553 sleepTime = idleSleepTime; 2554 if (mType == MIXER) { 2555 sleepTimeShift = 0; 2556 } 2557 2558 continue; 2559 } 2560 } 2561 2562 // mMixerStatusIgnoringFastTracks is also updated internally 2563 mMixerStatus = prepareTracks_l(&tracksToRemove); 2564 2565 // prevent any changes in effect chain list and in each effect chain 2566 // during mixing and effect process as the audio buffers could be deleted 2567 // or modified if an effect is created or deleted 2568 lockEffectChains_l(effectChains); 2569 } 2570 2571 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2572 threadLoop_mix(); 2573 } else { 2574 threadLoop_sleepTime(); 2575 } 2576 2577 if (mSuspended > 0) { 2578 sleepTime = suspendSleepTimeUs(); 2579 } 2580 2581 // only process effects if we're going to write 2582 if (sleepTime == 0) { 2583 for (size_t i = 0; i < effectChains.size(); i ++) { 2584 effectChains[i]->process_l(); 2585 } 2586 } 2587 2588 // enable changes in effect chain 2589 unlockEffectChains(effectChains); 2590 2591 // sleepTime == 0 means we must write to audio hardware 2592 if (sleepTime == 0) { 2593 2594 threadLoop_write(); 2595 2596if (mType == MIXER) { 2597 // write blocked detection 2598 nsecs_t now = systemTime(); 2599 nsecs_t delta = now - mLastWriteTime; 2600 if (!mStandby && delta > maxPeriod) { 2601 mNumDelayedWrites++; 2602 if ((now - lastWarning) > kWarningThrottleNs) { 2603#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2604 ScopedTrace st(ATRACE_TAG, "underrun"); 2605#endif 2606 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2607 ns2ms(delta), mNumDelayedWrites, this); 2608 lastWarning = now; 2609 } 2610 } 2611} 2612 2613 mStandby = false; 2614 } else { 2615 usleep(sleepTime); 2616 } 2617 2618 // Finally let go of removed track(s), without the lock held 2619 // since we can't guarantee the destructors won't acquire that 2620 // same lock. This will also mutate and push a new fast mixer state. 2621 threadLoop_removeTracks(tracksToRemove); 2622 tracksToRemove.clear(); 2623 2624 // FIXME I don't understand the need for this here; 2625 // it was in the original code but maybe the 2626 // assignment in saveOutputTracks() makes this unnecessary? 2627 clearOutputTracks(); 2628 2629 // Effect chains will be actually deleted here if they were removed from 2630 // mEffectChains list during mixing or effects processing 2631 effectChains.clear(); 2632 2633 // FIXME Note that the above .clear() is no longer necessary since effectChains 2634 // is now local to this block, but will keep it for now (at least until merge done). 2635 } 2636 2637if (mType == MIXER || mType == DIRECT) { 2638 // put output stream into standby mode 2639 if (!mStandby) { 2640 mOutput->stream->common.standby(&mOutput->stream->common); 2641 } 2642} 2643if (mType == DUPLICATING) { 2644 // for DuplicatingThread, standby mode is handled by the outputTracks 2645} 2646 2647 releaseWakeLock(); 2648 2649 ALOGV("Thread %p type %d exiting", this, mType); 2650 return false; 2651} 2652 2653void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2654{ 2655 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2656} 2657 2658void AudioFlinger::MixerThread::threadLoop_write() 2659{ 2660 // FIXME we should only do one push per cycle; confirm this is true 2661 // Start the fast mixer if it's not already running 2662 if (mFastMixer != NULL) { 2663 FastMixerStateQueue *sq = mFastMixer->sq(); 2664 FastMixerState *state = sq->begin(); 2665 if (state->mCommand != FastMixerState::MIX_WRITE && 2666 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2667 if (state->mCommand == FastMixerState::COLD_IDLE) { 2668 int32_t old = android_atomic_inc(&mFastMixerFutex); 2669 if (old == -1) { 2670 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2671 } 2672 if (mAudioWatchdog != 0) { 2673 mAudioWatchdog->resume(); 2674 } 2675 } 2676 state->mCommand = FastMixerState::MIX_WRITE; 2677 sq->end(); 2678 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2679 if (kUseFastMixer == FastMixer_Dynamic) { 2680 mNormalSink = mPipeSink; 2681 } 2682 } else { 2683 sq->end(false /*didModify*/); 2684 } 2685 } 2686 PlaybackThread::threadLoop_write(); 2687} 2688 2689// shared by MIXER and DIRECT, overridden by DUPLICATING 2690void AudioFlinger::PlaybackThread::threadLoop_write() 2691{ 2692 // FIXME rewrite to reduce number of system calls 2693 mLastWriteTime = systemTime(); 2694 mInWrite = true; 2695 int bytesWritten; 2696 2697 // If an NBAIO sink is present, use it to write the normal mixer's submix 2698 if (mNormalSink != 0) { 2699#define mBitShift 2 // FIXME 2700 size_t count = mixBufferSize >> mBitShift; 2701#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2702 Tracer::traceBegin(ATRACE_TAG, "write"); 2703#endif 2704 // update the setpoint when gScreenState changes 2705 uint32_t screenState = gScreenState; 2706 if (screenState != mScreenState) { 2707 mScreenState = screenState; 2708 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2709 if (pipe != NULL) { 2710 pipe->setAvgFrames((mScreenState & 1) ? 2711 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2712 } 2713 } 2714 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2715#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2716 Tracer::traceEnd(ATRACE_TAG); 2717#endif 2718 if (framesWritten > 0) { 2719 bytesWritten = framesWritten << mBitShift; 2720 } else { 2721 bytesWritten = framesWritten; 2722 } 2723 // otherwise use the HAL / AudioStreamOut directly 2724 } else { 2725 // Direct output thread. 2726 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2727 } 2728 2729 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2730 mNumWrites++; 2731 mInWrite = false; 2732} 2733 2734void AudioFlinger::MixerThread::threadLoop_standby() 2735{ 2736 // Idle the fast mixer if it's currently running 2737 if (mFastMixer != NULL) { 2738 FastMixerStateQueue *sq = mFastMixer->sq(); 2739 FastMixerState *state = sq->begin(); 2740 if (!(state->mCommand & FastMixerState::IDLE)) { 2741 state->mCommand = FastMixerState::COLD_IDLE; 2742 state->mColdFutexAddr = &mFastMixerFutex; 2743 state->mColdGen++; 2744 mFastMixerFutex = 0; 2745 sq->end(); 2746 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2747 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2748 if (kUseFastMixer == FastMixer_Dynamic) { 2749 mNormalSink = mOutputSink; 2750 } 2751 if (mAudioWatchdog != 0) { 2752 mAudioWatchdog->pause(); 2753 } 2754 } else { 2755 sq->end(false /*didModify*/); 2756 } 2757 } 2758 PlaybackThread::threadLoop_standby(); 2759} 2760 2761// shared by MIXER and DIRECT, overridden by DUPLICATING 2762void AudioFlinger::PlaybackThread::threadLoop_standby() 2763{ 2764 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2765 mOutput->stream->common.standby(&mOutput->stream->common); 2766} 2767 2768void AudioFlinger::MixerThread::threadLoop_mix() 2769{ 2770 // obtain the presentation timestamp of the next output buffer 2771 int64_t pts; 2772 status_t status = INVALID_OPERATION; 2773 2774 if (NULL != mOutput->stream->get_next_write_timestamp) { 2775 status = mOutput->stream->get_next_write_timestamp( 2776 mOutput->stream, &pts); 2777 } 2778 2779 if (status != NO_ERROR) { 2780 pts = AudioBufferProvider::kInvalidPTS; 2781 } 2782 2783 // mix buffers... 2784 mAudioMixer->process(pts); 2785 // increase sleep time progressively when application underrun condition clears. 2786 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2787 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2788 // such that we would underrun the audio HAL. 2789 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2790 sleepTimeShift--; 2791 } 2792 sleepTime = 0; 2793 standbyTime = systemTime() + standbyDelay; 2794 //TODO: delay standby when effects have a tail 2795} 2796 2797void AudioFlinger::MixerThread::threadLoop_sleepTime() 2798{ 2799 // If no tracks are ready, sleep once for the duration of an output 2800 // buffer size, then write 0s to the output 2801 if (sleepTime == 0) { 2802 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2803 sleepTime = activeSleepTime >> sleepTimeShift; 2804 if (sleepTime < kMinThreadSleepTimeUs) { 2805 sleepTime = kMinThreadSleepTimeUs; 2806 } 2807 // reduce sleep time in case of consecutive application underruns to avoid 2808 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2809 // duration we would end up writing less data than needed by the audio HAL if 2810 // the condition persists. 2811 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2812 sleepTimeShift++; 2813 } 2814 } else { 2815 sleepTime = idleSleepTime; 2816 } 2817 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2818 memset (mMixBuffer, 0, mixBufferSize); 2819 sleepTime = 0; 2820 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2821 } 2822 // TODO add standby time extension fct of effect tail 2823} 2824 2825// prepareTracks_l() must be called with ThreadBase::mLock held 2826AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2827 Vector< sp<Track> > *tracksToRemove) 2828{ 2829 2830 mixer_state mixerStatus = MIXER_IDLE; 2831 // find out which tracks need to be processed 2832 size_t count = mActiveTracks.size(); 2833 size_t mixedTracks = 0; 2834 size_t tracksWithEffect = 0; 2835 // counts only _active_ fast tracks 2836 size_t fastTracks = 0; 2837 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2838 2839 float masterVolume = mMasterVolume; 2840 bool masterMute = mMasterMute; 2841 2842 if (masterMute) { 2843 masterVolume = 0; 2844 } 2845 // Delegate master volume control to effect in output mix effect chain if needed 2846 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2847 if (chain != 0) { 2848 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2849 chain->setVolume_l(&v, &v); 2850 masterVolume = (float)((v + (1 << 23)) >> 24); 2851 chain.clear(); 2852 } 2853 2854 // prepare a new state to push 2855 FastMixerStateQueue *sq = NULL; 2856 FastMixerState *state = NULL; 2857 bool didModify = false; 2858 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2859 if (mFastMixer != NULL) { 2860 sq = mFastMixer->sq(); 2861 state = sq->begin(); 2862 } 2863 2864 for (size_t i=0 ; i<count ; i++) { 2865 sp<Track> t = mActiveTracks[i].promote(); 2866 if (t == 0) continue; 2867 2868 // this const just means the local variable doesn't change 2869 Track* const track = t.get(); 2870 2871 // process fast tracks 2872 if (track->isFastTrack()) { 2873 2874 // It's theoretically possible (though unlikely) for a fast track to be created 2875 // and then removed within the same normal mix cycle. This is not a problem, as 2876 // the track never becomes active so it's fast mixer slot is never touched. 2877 // The converse, of removing an (active) track and then creating a new track 2878 // at the identical fast mixer slot within the same normal mix cycle, 2879 // is impossible because the slot isn't marked available until the end of each cycle. 2880 int j = track->mFastIndex; 2881 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2882 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2883 FastTrack *fastTrack = &state->mFastTracks[j]; 2884 2885 // Determine whether the track is currently in underrun condition, 2886 // and whether it had a recent underrun. 2887 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2888 FastTrackUnderruns underruns = ftDump->mUnderruns; 2889 uint32_t recentFull = (underruns.mBitFields.mFull - 2890 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2891 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2892 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2893 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2894 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2895 uint32_t recentUnderruns = recentPartial + recentEmpty; 2896 track->mObservedUnderruns = underruns; 2897 // don't count underruns that occur while stopping or pausing 2898 // or stopped which can occur when flush() is called while active 2899 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2900 track->mUnderrunCount += recentUnderruns; 2901 } 2902 2903 // This is similar to the state machine for normal tracks, 2904 // with a few modifications for fast tracks. 2905 bool isActive = true; 2906 switch (track->mState) { 2907 case TrackBase::STOPPING_1: 2908 // track stays active in STOPPING_1 state until first underrun 2909 if (recentUnderruns > 0) { 2910 track->mState = TrackBase::STOPPING_2; 2911 } 2912 break; 2913 case TrackBase::PAUSING: 2914 // ramp down is not yet implemented 2915 track->setPaused(); 2916 break; 2917 case TrackBase::RESUMING: 2918 // ramp up is not yet implemented 2919 track->mState = TrackBase::ACTIVE; 2920 break; 2921 case TrackBase::ACTIVE: 2922 if (recentFull > 0 || recentPartial > 0) { 2923 // track has provided at least some frames recently: reset retry count 2924 track->mRetryCount = kMaxTrackRetries; 2925 } 2926 if (recentUnderruns == 0) { 2927 // no recent underruns: stay active 2928 break; 2929 } 2930 // there has recently been an underrun of some kind 2931 if (track->sharedBuffer() == 0) { 2932 // were any of the recent underruns "empty" (no frames available)? 2933 if (recentEmpty == 0) { 2934 // no, then ignore the partial underruns as they are allowed indefinitely 2935 break; 2936 } 2937 // there has recently been an "empty" underrun: decrement the retry counter 2938 if (--(track->mRetryCount) > 0) { 2939 break; 2940 } 2941 // indicate to client process that the track was disabled because of underrun; 2942 // it will then automatically call start() when data is available 2943 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2944 // remove from active list, but state remains ACTIVE [confusing but true] 2945 isActive = false; 2946 break; 2947 } 2948 // fall through 2949 case TrackBase::STOPPING_2: 2950 case TrackBase::PAUSED: 2951 case TrackBase::TERMINATED: 2952 case TrackBase::STOPPED: 2953 case TrackBase::FLUSHED: // flush() while active 2954 // Check for presentation complete if track is inactive 2955 // We have consumed all the buffers of this track. 2956 // This would be incomplete if we auto-paused on underrun 2957 { 2958 size_t audioHALFrames = 2959 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2960 size_t framesWritten = 2961 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2962 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2963 // track stays in active list until presentation is complete 2964 break; 2965 } 2966 } 2967 if (track->isStopping_2()) { 2968 track->mState = TrackBase::STOPPED; 2969 } 2970 if (track->isStopped()) { 2971 // Can't reset directly, as fast mixer is still polling this track 2972 // track->reset(); 2973 // So instead mark this track as needing to be reset after push with ack 2974 resetMask |= 1 << i; 2975 } 2976 isActive = false; 2977 break; 2978 case TrackBase::IDLE: 2979 default: 2980 LOG_FATAL("unexpected track state %d", track->mState); 2981 } 2982 2983 if (isActive) { 2984 // was it previously inactive? 2985 if (!(state->mTrackMask & (1 << j))) { 2986 ExtendedAudioBufferProvider *eabp = track; 2987 VolumeProvider *vp = track; 2988 fastTrack->mBufferProvider = eabp; 2989 fastTrack->mVolumeProvider = vp; 2990 fastTrack->mSampleRate = track->mSampleRate; 2991 fastTrack->mChannelMask = track->mChannelMask; 2992 fastTrack->mGeneration++; 2993 state->mTrackMask |= 1 << j; 2994 didModify = true; 2995 // no acknowledgement required for newly active tracks 2996 } 2997 // cache the combined master volume and stream type volume for fast mixer; this 2998 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2999 track->mCachedVolume = track->isMuted() ? 3000 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3001 ++fastTracks; 3002 } else { 3003 // was it previously active? 3004 if (state->mTrackMask & (1 << j)) { 3005 fastTrack->mBufferProvider = NULL; 3006 fastTrack->mGeneration++; 3007 state->mTrackMask &= ~(1 << j); 3008 didModify = true; 3009 // If any fast tracks were removed, we must wait for acknowledgement 3010 // because we're about to decrement the last sp<> on those tracks. 3011 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3012 } else { 3013 LOG_FATAL("fast track %d should have been active", j); 3014 } 3015 tracksToRemove->add(track); 3016 // Avoids a misleading display in dumpsys 3017 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3018 } 3019 continue; 3020 } 3021 3022 { // local variable scope to avoid goto warning 3023 3024 audio_track_cblk_t* cblk = track->cblk(); 3025 3026 // The first time a track is added we wait 3027 // for all its buffers to be filled before processing it 3028 int name = track->name(); 3029 // make sure that we have enough frames to mix one full buffer. 3030 // enforce this condition only once to enable draining the buffer in case the client 3031 // app does not call stop() and relies on underrun to stop: 3032 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3033 // during last round 3034 uint32_t minFrames = 1; 3035 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3036 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3037 if (t->sampleRate() == (int)mSampleRate) { 3038 minFrames = mNormalFrameCount; 3039 } else { 3040 // +1 for rounding and +1 for additional sample needed for interpolation 3041 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3042 // add frames already consumed but not yet released by the resampler 3043 // because cblk->framesReady() will include these frames 3044 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3045 // the minimum track buffer size is normally twice the number of frames necessary 3046 // to fill one buffer and the resampler should not leave more than one buffer worth 3047 // of unreleased frames after each pass, but just in case... 3048 ALOG_ASSERT(minFrames <= cblk->frameCount); 3049 } 3050 } 3051 if ((track->framesReady() >= minFrames) && track->isReady() && 3052 !track->isPaused() && !track->isTerminated()) 3053 { 3054 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3055 3056 mixedTracks++; 3057 3058 // track->mainBuffer() != mMixBuffer means there is an effect chain 3059 // connected to the track 3060 chain.clear(); 3061 if (track->mainBuffer() != mMixBuffer) { 3062 chain = getEffectChain_l(track->sessionId()); 3063 // Delegate volume control to effect in track effect chain if needed 3064 if (chain != 0) { 3065 tracksWithEffect++; 3066 } else { 3067 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3068 name, track->sessionId()); 3069 } 3070 } 3071 3072 3073 int param = AudioMixer::VOLUME; 3074 if (track->mFillingUpStatus == Track::FS_FILLED) { 3075 // no ramp for the first volume setting 3076 track->mFillingUpStatus = Track::FS_ACTIVE; 3077 if (track->mState == TrackBase::RESUMING) { 3078 track->mState = TrackBase::ACTIVE; 3079 param = AudioMixer::RAMP_VOLUME; 3080 } 3081 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3082 } else if (cblk->server != 0) { 3083 // If the track is stopped before the first frame was mixed, 3084 // do not apply ramp 3085 param = AudioMixer::RAMP_VOLUME; 3086 } 3087 3088 // compute volume for this track 3089 uint32_t vl, vr, va; 3090 if (track->isMuted() || track->isPausing() || 3091 mStreamTypes[track->streamType()].mute) { 3092 vl = vr = va = 0; 3093 if (track->isPausing()) { 3094 track->setPaused(); 3095 } 3096 } else { 3097 3098 // read original volumes with volume control 3099 float typeVolume = mStreamTypes[track->streamType()].volume; 3100 float v = masterVolume * typeVolume; 3101 uint32_t vlr = cblk->getVolumeLR(); 3102 vl = vlr & 0xFFFF; 3103 vr = vlr >> 16; 3104 // track volumes come from shared memory, so can't be trusted and must be clamped 3105 if (vl > MAX_GAIN_INT) { 3106 ALOGV("Track left volume out of range: %04X", vl); 3107 vl = MAX_GAIN_INT; 3108 } 3109 if (vr > MAX_GAIN_INT) { 3110 ALOGV("Track right volume out of range: %04X", vr); 3111 vr = MAX_GAIN_INT; 3112 } 3113 // now apply the master volume and stream type volume 3114 vl = (uint32_t)(v * vl) << 12; 3115 vr = (uint32_t)(v * vr) << 12; 3116 // assuming master volume and stream type volume each go up to 1.0, 3117 // vl and vr are now in 8.24 format 3118 3119 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3120 // send level comes from shared memory and so may be corrupt 3121 if (sendLevel > MAX_GAIN_INT) { 3122 ALOGV("Track send level out of range: %04X", sendLevel); 3123 sendLevel = MAX_GAIN_INT; 3124 } 3125 va = (uint32_t)(v * sendLevel); 3126 } 3127 // Delegate volume control to effect in track effect chain if needed 3128 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3129 // Do not ramp volume if volume is controlled by effect 3130 param = AudioMixer::VOLUME; 3131 track->mHasVolumeController = true; 3132 } else { 3133 // force no volume ramp when volume controller was just disabled or removed 3134 // from effect chain to avoid volume spike 3135 if (track->mHasVolumeController) { 3136 param = AudioMixer::VOLUME; 3137 } 3138 track->mHasVolumeController = false; 3139 } 3140 3141 // Convert volumes from 8.24 to 4.12 format 3142 // This additional clamping is needed in case chain->setVolume_l() overshot 3143 vl = (vl + (1 << 11)) >> 12; 3144 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3145 vr = (vr + (1 << 11)) >> 12; 3146 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3147 3148 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3149 3150 // XXX: these things DON'T need to be done each time 3151 mAudioMixer->setBufferProvider(name, track); 3152 mAudioMixer->enable(name); 3153 3154 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3155 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3156 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3157 mAudioMixer->setParameter( 3158 name, 3159 AudioMixer::TRACK, 3160 AudioMixer::FORMAT, (void *)track->format()); 3161 mAudioMixer->setParameter( 3162 name, 3163 AudioMixer::TRACK, 3164 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3165 mAudioMixer->setParameter( 3166 name, 3167 AudioMixer::RESAMPLE, 3168 AudioMixer::SAMPLE_RATE, 3169 (void *)(cblk->sampleRate)); 3170 mAudioMixer->setParameter( 3171 name, 3172 AudioMixer::TRACK, 3173 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3174 mAudioMixer->setParameter( 3175 name, 3176 AudioMixer::TRACK, 3177 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3178 3179 // reset retry count 3180 track->mRetryCount = kMaxTrackRetries; 3181 3182 // If one track is ready, set the mixer ready if: 3183 // - the mixer was not ready during previous round OR 3184 // - no other track is not ready 3185 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3186 mixerStatus != MIXER_TRACKS_ENABLED) { 3187 mixerStatus = MIXER_TRACKS_READY; 3188 } 3189 } else { 3190 // clear effect chain input buffer if an active track underruns to avoid sending 3191 // previous audio buffer again to effects 3192 chain = getEffectChain_l(track->sessionId()); 3193 if (chain != 0) { 3194 chain->clearInputBuffer(); 3195 } 3196 3197 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3198 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3199 track->isStopped() || track->isPaused()) { 3200 // We have consumed all the buffers of this track. 3201 // Remove it from the list of active tracks. 3202 // TODO: use actual buffer filling status instead of latency when available from 3203 // audio HAL 3204 size_t audioHALFrames = 3205 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3206 size_t framesWritten = 3207 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3208 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3209 if (track->isStopped()) { 3210 track->reset(); 3211 } 3212 tracksToRemove->add(track); 3213 } 3214 } else { 3215 track->mUnderrunCount++; 3216 // No buffers for this track. Give it a few chances to 3217 // fill a buffer, then remove it from active list. 3218 if (--(track->mRetryCount) <= 0) { 3219 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3220 tracksToRemove->add(track); 3221 // indicate to client process that the track was disabled because of underrun; 3222 // it will then automatically call start() when data is available 3223 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3224 // If one track is not ready, mark the mixer also not ready if: 3225 // - the mixer was ready during previous round OR 3226 // - no other track is ready 3227 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3228 mixerStatus != MIXER_TRACKS_READY) { 3229 mixerStatus = MIXER_TRACKS_ENABLED; 3230 } 3231 } 3232 mAudioMixer->disable(name); 3233 } 3234 3235 } // local variable scope to avoid goto warning 3236track_is_ready: ; 3237 3238 } 3239 3240 // Push the new FastMixer state if necessary 3241 bool pauseAudioWatchdog = false; 3242 if (didModify) { 3243 state->mFastTracksGen++; 3244 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3245 if (kUseFastMixer == FastMixer_Dynamic && 3246 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3247 state->mCommand = FastMixerState::COLD_IDLE; 3248 state->mColdFutexAddr = &mFastMixerFutex; 3249 state->mColdGen++; 3250 mFastMixerFutex = 0; 3251 if (kUseFastMixer == FastMixer_Dynamic) { 3252 mNormalSink = mOutputSink; 3253 } 3254 // If we go into cold idle, need to wait for acknowledgement 3255 // so that fast mixer stops doing I/O. 3256 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3257 pauseAudioWatchdog = true; 3258 } 3259 sq->end(); 3260 } 3261 if (sq != NULL) { 3262 sq->end(didModify); 3263 sq->push(block); 3264 } 3265 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3266 mAudioWatchdog->pause(); 3267 } 3268 3269 // Now perform the deferred reset on fast tracks that have stopped 3270 while (resetMask != 0) { 3271 size_t i = __builtin_ctz(resetMask); 3272 ALOG_ASSERT(i < count); 3273 resetMask &= ~(1 << i); 3274 sp<Track> t = mActiveTracks[i].promote(); 3275 if (t == 0) continue; 3276 Track* track = t.get(); 3277 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3278 track->reset(); 3279 } 3280 3281 // remove all the tracks that need to be... 3282 count = tracksToRemove->size(); 3283 if (CC_UNLIKELY(count)) { 3284 for (size_t i=0 ; i<count ; i++) { 3285 const sp<Track>& track = tracksToRemove->itemAt(i); 3286 mActiveTracks.remove(track); 3287 if (track->mainBuffer() != mMixBuffer) { 3288 chain = getEffectChain_l(track->sessionId()); 3289 if (chain != 0) { 3290 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3291 chain->decActiveTrackCnt(); 3292 } 3293 } 3294 if (track->isTerminated()) { 3295 removeTrack_l(track); 3296 } 3297 } 3298 } 3299 3300 // mix buffer must be cleared if all tracks are connected to an 3301 // effect chain as in this case the mixer will not write to 3302 // mix buffer and track effects will accumulate into it 3303 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3304 // FIXME as a performance optimization, should remember previous zero status 3305 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3306 } 3307 3308 // if any fast tracks, then status is ready 3309 mMixerStatusIgnoringFastTracks = mixerStatus; 3310 if (fastTracks > 0) { 3311 mixerStatus = MIXER_TRACKS_READY; 3312 } 3313 return mixerStatus; 3314} 3315 3316/* 3317The derived values that are cached: 3318 - mixBufferSize from frame count * frame size 3319 - activeSleepTime from activeSleepTimeUs() 3320 - idleSleepTime from idleSleepTimeUs() 3321 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3322 - maxPeriod from frame count and sample rate (MIXER only) 3323 3324The parameters that affect these derived values are: 3325 - frame count 3326 - frame size 3327 - sample rate 3328 - device type: A2DP or not 3329 - device latency 3330 - format: PCM or not 3331 - active sleep time 3332 - idle sleep time 3333*/ 3334 3335void AudioFlinger::PlaybackThread::cacheParameters_l() 3336{ 3337 mixBufferSize = mNormalFrameCount * mFrameSize; 3338 activeSleepTime = activeSleepTimeUs(); 3339 idleSleepTime = idleSleepTimeUs(); 3340} 3341 3342void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3343{ 3344 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3345 this, streamType, mTracks.size()); 3346 Mutex::Autolock _l(mLock); 3347 3348 size_t size = mTracks.size(); 3349 for (size_t i = 0; i < size; i++) { 3350 sp<Track> t = mTracks[i]; 3351 if (t->streamType() == streamType) { 3352 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3353 t->mCblk->cv.signal(); 3354 } 3355 } 3356} 3357 3358// getTrackName_l() must be called with ThreadBase::mLock held 3359int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3360{ 3361 return mAudioMixer->getTrackName(channelMask); 3362} 3363 3364// deleteTrackName_l() must be called with ThreadBase::mLock held 3365void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3366{ 3367 ALOGV("remove track (%d) and delete from mixer", name); 3368 mAudioMixer->deleteTrackName(name); 3369} 3370 3371// checkForNewParameters_l() must be called with ThreadBase::mLock held 3372bool AudioFlinger::MixerThread::checkForNewParameters_l() 3373{ 3374 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3375 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3376 bool reconfig = false; 3377 3378 while (!mNewParameters.isEmpty()) { 3379 3380 if (mFastMixer != NULL) { 3381 FastMixerStateQueue *sq = mFastMixer->sq(); 3382 FastMixerState *state = sq->begin(); 3383 if (!(state->mCommand & FastMixerState::IDLE)) { 3384 previousCommand = state->mCommand; 3385 state->mCommand = FastMixerState::HOT_IDLE; 3386 sq->end(); 3387 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3388 } else { 3389 sq->end(false /*didModify*/); 3390 } 3391 } 3392 3393 status_t status = NO_ERROR; 3394 String8 keyValuePair = mNewParameters[0]; 3395 AudioParameter param = AudioParameter(keyValuePair); 3396 int value; 3397 3398 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3399 reconfig = true; 3400 } 3401 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3402 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3403 status = BAD_VALUE; 3404 } else { 3405 reconfig = true; 3406 } 3407 } 3408 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3409 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3410 status = BAD_VALUE; 3411 } else { 3412 reconfig = true; 3413 } 3414 } 3415 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3416 // do not accept frame count changes if tracks are open as the track buffer 3417 // size depends on frame count and correct behavior would not be guaranteed 3418 // if frame count is changed after track creation 3419 if (!mTracks.isEmpty()) { 3420 status = INVALID_OPERATION; 3421 } else { 3422 reconfig = true; 3423 } 3424 } 3425 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3426#ifdef ADD_BATTERY_DATA 3427 // when changing the audio output device, call addBatteryData to notify 3428 // the change 3429 if ((int)mDevice != value) { 3430 uint32_t params = 0; 3431 // check whether speaker is on 3432 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3433 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3434 } 3435 3436 int deviceWithoutSpeaker 3437 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3438 // check if any other device (except speaker) is on 3439 if (value & deviceWithoutSpeaker ) { 3440 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3441 } 3442 3443 if (params != 0) { 3444 addBatteryData(params); 3445 } 3446 } 3447#endif 3448 3449 // forward device change to effects that have requested to be 3450 // aware of attached audio device. 3451 mDevice = (uint32_t)value; 3452 for (size_t i = 0; i < mEffectChains.size(); i++) { 3453 mEffectChains[i]->setDevice_l(mDevice); 3454 } 3455 } 3456 3457 if (status == NO_ERROR) { 3458 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3459 keyValuePair.string()); 3460 if (!mStandby && status == INVALID_OPERATION) { 3461 mOutput->stream->common.standby(&mOutput->stream->common); 3462 mStandby = true; 3463 mBytesWritten = 0; 3464 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3465 keyValuePair.string()); 3466 } 3467 if (status == NO_ERROR && reconfig) { 3468 delete mAudioMixer; 3469 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3470 mAudioMixer = NULL; 3471 readOutputParameters(); 3472 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3473 for (size_t i = 0; i < mTracks.size() ; i++) { 3474 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3475 if (name < 0) break; 3476 mTracks[i]->mName = name; 3477 // limit track sample rate to 2 x new output sample rate 3478 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3479 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3480 } 3481 } 3482 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3483 } 3484 } 3485 3486 mNewParameters.removeAt(0); 3487 3488 mParamStatus = status; 3489 mParamCond.signal(); 3490 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3491 // already timed out waiting for the status and will never signal the condition. 3492 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3493 } 3494 3495 if (!(previousCommand & FastMixerState::IDLE)) { 3496 ALOG_ASSERT(mFastMixer != NULL); 3497 FastMixerStateQueue *sq = mFastMixer->sq(); 3498 FastMixerState *state = sq->begin(); 3499 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3500 state->mCommand = previousCommand; 3501 sq->end(); 3502 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3503 } 3504 3505 return reconfig; 3506} 3507 3508status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3509{ 3510 const size_t SIZE = 256; 3511 char buffer[SIZE]; 3512 String8 result; 3513 3514 PlaybackThread::dumpInternals(fd, args); 3515 3516 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3517 result.append(buffer); 3518 write(fd, result.string(), result.size()); 3519 3520 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3521 FastMixerDumpState copy = mFastMixerDumpState; 3522 copy.dump(fd); 3523 3524#ifdef STATE_QUEUE_DUMP 3525 // Similar for state queue 3526 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3527 observerCopy.dump(fd); 3528 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3529 mutatorCopy.dump(fd); 3530#endif 3531 3532 // Write the tee output to a .wav file 3533 NBAIO_Source *teeSource = mTeeSource.get(); 3534 if (teeSource != NULL) { 3535 char teePath[64]; 3536 struct timeval tv; 3537 gettimeofday(&tv, NULL); 3538 struct tm tm; 3539 localtime_r(&tv.tv_sec, &tm); 3540 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3541 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3542 if (teeFd >= 0) { 3543 char wavHeader[44]; 3544 memcpy(wavHeader, 3545 "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", 3546 sizeof(wavHeader)); 3547 NBAIO_Format format = teeSource->format(); 3548 unsigned channelCount = Format_channelCount(format); 3549 ALOG_ASSERT(channelCount <= FCC_2); 3550 unsigned sampleRate = Format_sampleRate(format); 3551 wavHeader[22] = channelCount; // number of channels 3552 wavHeader[24] = sampleRate; // sample rate 3553 wavHeader[25] = sampleRate >> 8; 3554 wavHeader[32] = channelCount * 2; // block alignment 3555 write(teeFd, wavHeader, sizeof(wavHeader)); 3556 size_t total = 0; 3557 bool firstRead = true; 3558 for (;;) { 3559#define TEE_SINK_READ 1024 3560 short buffer[TEE_SINK_READ * FCC_2]; 3561 size_t count = TEE_SINK_READ; 3562 ssize_t actual = teeSource->read(buffer, count); 3563 bool wasFirstRead = firstRead; 3564 firstRead = false; 3565 if (actual <= 0) { 3566 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3567 continue; 3568 } 3569 break; 3570 } 3571 ALOG_ASSERT(actual <= (ssize_t)count); 3572 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3573 total += actual; 3574 } 3575 lseek(teeFd, (off_t) 4, SEEK_SET); 3576 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3577 write(teeFd, &temp, sizeof(temp)); 3578 lseek(teeFd, (off_t) 40, SEEK_SET); 3579 temp = total * channelCount * sizeof(short); 3580 write(teeFd, &temp, sizeof(temp)); 3581 close(teeFd); 3582 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3583 } else { 3584 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3585 } 3586 } 3587 3588 if (mAudioWatchdog != 0) { 3589 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3590 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3591 wdCopy.dump(fd); 3592 } 3593 3594 return NO_ERROR; 3595} 3596 3597uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3598{ 3599 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3600} 3601 3602uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3603{ 3604 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3605} 3606 3607void AudioFlinger::MixerThread::cacheParameters_l() 3608{ 3609 PlaybackThread::cacheParameters_l(); 3610 3611 // FIXME: Relaxed timing because of a certain device that can't meet latency 3612 // Should be reduced to 2x after the vendor fixes the driver issue 3613 // increase threshold again due to low power audio mode. The way this warning 3614 // threshold is calculated and its usefulness should be reconsidered anyway. 3615 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3616} 3617 3618// ---------------------------------------------------------------------------- 3619AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3620 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3621 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3622 // mLeftVolFloat, mRightVolFloat 3623{ 3624} 3625 3626AudioFlinger::DirectOutputThread::~DirectOutputThread() 3627{ 3628} 3629 3630AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3631 Vector< sp<Track> > *tracksToRemove 3632) 3633{ 3634 sp<Track> trackToRemove; 3635 3636 mixer_state mixerStatus = MIXER_IDLE; 3637 3638 // find out which tracks need to be processed 3639 if (mActiveTracks.size() != 0) { 3640 sp<Track> t = mActiveTracks[0].promote(); 3641 // The track died recently 3642 if (t == 0) return MIXER_IDLE; 3643 3644 Track* const track = t.get(); 3645 audio_track_cblk_t* cblk = track->cblk(); 3646 3647 // The first time a track is added we wait 3648 // for all its buffers to be filled before processing it 3649 uint32_t minFrames; 3650 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3651 minFrames = mNormalFrameCount; 3652 } else { 3653 minFrames = 1; 3654 } 3655 if ((track->framesReady() >= minFrames) && track->isReady() && 3656 !track->isPaused() && !track->isTerminated()) 3657 { 3658 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3659 3660 if (track->mFillingUpStatus == Track::FS_FILLED) { 3661 track->mFillingUpStatus = Track::FS_ACTIVE; 3662 mLeftVolFloat = mRightVolFloat = 0; 3663 if (track->mState == TrackBase::RESUMING) { 3664 track->mState = TrackBase::ACTIVE; 3665 } 3666 } 3667 3668 // compute volume for this track 3669 float left, right; 3670 if (track->isMuted() || mMasterMute || track->isPausing() || 3671 mStreamTypes[track->streamType()].mute) { 3672 left = right = 0; 3673 if (track->isPausing()) { 3674 track->setPaused(); 3675 } 3676 } else { 3677 float typeVolume = mStreamTypes[track->streamType()].volume; 3678 float v = mMasterVolume * typeVolume; 3679 uint32_t vlr = cblk->getVolumeLR(); 3680 float v_clamped = v * (vlr & 0xFFFF); 3681 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3682 left = v_clamped/MAX_GAIN; 3683 v_clamped = v * (vlr >> 16); 3684 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3685 right = v_clamped/MAX_GAIN; 3686 } 3687 3688 if (left != mLeftVolFloat || right != mRightVolFloat) { 3689 mLeftVolFloat = left; 3690 mRightVolFloat = right; 3691 3692 // Convert volumes from float to 8.24 3693 uint32_t vl = (uint32_t)(left * (1 << 24)); 3694 uint32_t vr = (uint32_t)(right * (1 << 24)); 3695 3696 // Delegate volume control to effect in track effect chain if needed 3697 // only one effect chain can be present on DirectOutputThread, so if 3698 // there is one, the track is connected to it 3699 if (!mEffectChains.isEmpty()) { 3700 // Do not ramp volume if volume is controlled by effect 3701 mEffectChains[0]->setVolume_l(&vl, &vr); 3702 left = (float)vl / (1 << 24); 3703 right = (float)vr / (1 << 24); 3704 } 3705 mOutput->stream->set_volume(mOutput->stream, left, right); 3706 } 3707 3708 // reset retry count 3709 track->mRetryCount = kMaxTrackRetriesDirect; 3710 mActiveTrack = t; 3711 mixerStatus = MIXER_TRACKS_READY; 3712 } else { 3713 // clear effect chain input buffer if an active track underruns to avoid sending 3714 // previous audio buffer again to effects 3715 if (!mEffectChains.isEmpty()) { 3716 mEffectChains[0]->clearInputBuffer(); 3717 } 3718 3719 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3720 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3721 track->isStopped() || track->isPaused()) { 3722 // We have consumed all the buffers of this track. 3723 // Remove it from the list of active tracks. 3724 // TODO: implement behavior for compressed audio 3725 size_t audioHALFrames = 3726 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3727 size_t framesWritten = 3728 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3729 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3730 if (track->isStopped()) { 3731 track->reset(); 3732 } 3733 trackToRemove = track; 3734 } 3735 } else { 3736 // No buffers for this track. Give it a few chances to 3737 // fill a buffer, then remove it from active list. 3738 if (--(track->mRetryCount) <= 0) { 3739 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3740 trackToRemove = track; 3741 } else { 3742 mixerStatus = MIXER_TRACKS_ENABLED; 3743 } 3744 } 3745 } 3746 } 3747 3748 // FIXME merge this with similar code for removing multiple tracks 3749 // remove all the tracks that need to be... 3750 if (CC_UNLIKELY(trackToRemove != 0)) { 3751 tracksToRemove->add(trackToRemove); 3752 mActiveTracks.remove(trackToRemove); 3753 if (!mEffectChains.isEmpty()) { 3754 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3755 trackToRemove->sessionId()); 3756 mEffectChains[0]->decActiveTrackCnt(); 3757 } 3758 if (trackToRemove->isTerminated()) { 3759 removeTrack_l(trackToRemove); 3760 } 3761 } 3762 3763 return mixerStatus; 3764} 3765 3766void AudioFlinger::DirectOutputThread::threadLoop_mix() 3767{ 3768 AudioBufferProvider::Buffer buffer; 3769 size_t frameCount = mFrameCount; 3770 int8_t *curBuf = (int8_t *)mMixBuffer; 3771 // output audio to hardware 3772 while (frameCount) { 3773 buffer.frameCount = frameCount; 3774 mActiveTrack->getNextBuffer(&buffer); 3775 if (CC_UNLIKELY(buffer.raw == NULL)) { 3776 memset(curBuf, 0, frameCount * mFrameSize); 3777 break; 3778 } 3779 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3780 frameCount -= buffer.frameCount; 3781 curBuf += buffer.frameCount * mFrameSize; 3782 mActiveTrack->releaseBuffer(&buffer); 3783 } 3784 sleepTime = 0; 3785 standbyTime = systemTime() + standbyDelay; 3786 mActiveTrack.clear(); 3787 3788} 3789 3790void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3791{ 3792 if (sleepTime == 0) { 3793 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3794 sleepTime = activeSleepTime; 3795 } else { 3796 sleepTime = idleSleepTime; 3797 } 3798 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3799 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3800 sleepTime = 0; 3801 } 3802} 3803 3804// getTrackName_l() must be called with ThreadBase::mLock held 3805int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3806{ 3807 return 0; 3808} 3809 3810// deleteTrackName_l() must be called with ThreadBase::mLock held 3811void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3812{ 3813} 3814 3815// checkForNewParameters_l() must be called with ThreadBase::mLock held 3816bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3817{ 3818 bool reconfig = false; 3819 3820 while (!mNewParameters.isEmpty()) { 3821 status_t status = NO_ERROR; 3822 String8 keyValuePair = mNewParameters[0]; 3823 AudioParameter param = AudioParameter(keyValuePair); 3824 int value; 3825 3826 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3827 // do not accept frame count changes if tracks are open as the track buffer 3828 // size depends on frame count and correct behavior would not be garantied 3829 // if frame count is changed after track creation 3830 if (!mTracks.isEmpty()) { 3831 status = INVALID_OPERATION; 3832 } else { 3833 reconfig = true; 3834 } 3835 } 3836 if (status == NO_ERROR) { 3837 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3838 keyValuePair.string()); 3839 if (!mStandby && status == INVALID_OPERATION) { 3840 mOutput->stream->common.standby(&mOutput->stream->common); 3841 mStandby = true; 3842 mBytesWritten = 0; 3843 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3844 keyValuePair.string()); 3845 } 3846 if (status == NO_ERROR && reconfig) { 3847 readOutputParameters(); 3848 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3849 } 3850 } 3851 3852 mNewParameters.removeAt(0); 3853 3854 mParamStatus = status; 3855 mParamCond.signal(); 3856 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3857 // already timed out waiting for the status and will never signal the condition. 3858 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3859 } 3860 return reconfig; 3861} 3862 3863uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3864{ 3865 uint32_t time; 3866 if (audio_is_linear_pcm(mFormat)) { 3867 time = PlaybackThread::activeSleepTimeUs(); 3868 } else { 3869 time = 10000; 3870 } 3871 return time; 3872} 3873 3874uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3875{ 3876 uint32_t time; 3877 if (audio_is_linear_pcm(mFormat)) { 3878 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3879 } else { 3880 time = 10000; 3881 } 3882 return time; 3883} 3884 3885uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3886{ 3887 uint32_t time; 3888 if (audio_is_linear_pcm(mFormat)) { 3889 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3890 } else { 3891 time = 10000; 3892 } 3893 return time; 3894} 3895 3896void AudioFlinger::DirectOutputThread::cacheParameters_l() 3897{ 3898 PlaybackThread::cacheParameters_l(); 3899 3900 // use shorter standby delay as on normal output to release 3901 // hardware resources as soon as possible 3902 standbyDelay = microseconds(activeSleepTime*2); 3903} 3904 3905// ---------------------------------------------------------------------------- 3906 3907AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3908 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3909 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3910 mWaitTimeMs(UINT_MAX) 3911{ 3912 addOutputTrack(mainThread); 3913} 3914 3915AudioFlinger::DuplicatingThread::~DuplicatingThread() 3916{ 3917 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3918 mOutputTracks[i]->destroy(); 3919 } 3920} 3921 3922void AudioFlinger::DuplicatingThread::threadLoop_mix() 3923{ 3924 // mix buffers... 3925 if (outputsReady(outputTracks)) { 3926 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3927 } else { 3928 memset(mMixBuffer, 0, mixBufferSize); 3929 } 3930 sleepTime = 0; 3931 writeFrames = mNormalFrameCount; 3932 standbyTime = systemTime() + standbyDelay; 3933} 3934 3935void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3936{ 3937 if (sleepTime == 0) { 3938 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3939 sleepTime = activeSleepTime; 3940 } else { 3941 sleepTime = idleSleepTime; 3942 } 3943 } else if (mBytesWritten != 0) { 3944 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3945 writeFrames = mNormalFrameCount; 3946 memset(mMixBuffer, 0, mixBufferSize); 3947 } else { 3948 // flush remaining overflow buffers in output tracks 3949 writeFrames = 0; 3950 } 3951 sleepTime = 0; 3952 } 3953} 3954 3955void AudioFlinger::DuplicatingThread::threadLoop_write() 3956{ 3957 for (size_t i = 0; i < outputTracks.size(); i++) { 3958 outputTracks[i]->write(mMixBuffer, writeFrames); 3959 } 3960 mBytesWritten += mixBufferSize; 3961} 3962 3963void AudioFlinger::DuplicatingThread::threadLoop_standby() 3964{ 3965 // DuplicatingThread implements standby by stopping all tracks 3966 for (size_t i = 0; i < outputTracks.size(); i++) { 3967 outputTracks[i]->stop(); 3968 } 3969} 3970 3971void AudioFlinger::DuplicatingThread::saveOutputTracks() 3972{ 3973 outputTracks = mOutputTracks; 3974} 3975 3976void AudioFlinger::DuplicatingThread::clearOutputTracks() 3977{ 3978 outputTracks.clear(); 3979} 3980 3981void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3982{ 3983 Mutex::Autolock _l(mLock); 3984 // FIXME explain this formula 3985 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3986 OutputTrack *outputTrack = new OutputTrack(thread, 3987 this, 3988 mSampleRate, 3989 mFormat, 3990 mChannelMask, 3991 frameCount); 3992 if (outputTrack->cblk() != NULL) { 3993 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3994 mOutputTracks.add(outputTrack); 3995 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3996 updateWaitTime_l(); 3997 } 3998} 3999 4000void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4001{ 4002 Mutex::Autolock _l(mLock); 4003 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4004 if (mOutputTracks[i]->thread() == thread) { 4005 mOutputTracks[i]->destroy(); 4006 mOutputTracks.removeAt(i); 4007 updateWaitTime_l(); 4008 return; 4009 } 4010 } 4011 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4012} 4013 4014// caller must hold mLock 4015void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4016{ 4017 mWaitTimeMs = UINT_MAX; 4018 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4019 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4020 if (strong != 0) { 4021 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4022 if (waitTimeMs < mWaitTimeMs) { 4023 mWaitTimeMs = waitTimeMs; 4024 } 4025 } 4026 } 4027} 4028 4029 4030bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4031{ 4032 for (size_t i = 0; i < outputTracks.size(); i++) { 4033 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4034 if (thread == 0) { 4035 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4036 return false; 4037 } 4038 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4039 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4040 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4041 return false; 4042 } 4043 } 4044 return true; 4045} 4046 4047uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4048{ 4049 return (mWaitTimeMs * 1000) / 2; 4050} 4051 4052void AudioFlinger::DuplicatingThread::cacheParameters_l() 4053{ 4054 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4055 updateWaitTime_l(); 4056 4057 MixerThread::cacheParameters_l(); 4058} 4059 4060// ---------------------------------------------------------------------------- 4061 4062// TrackBase constructor must be called with AudioFlinger::mLock held 4063AudioFlinger::ThreadBase::TrackBase::TrackBase( 4064 ThreadBase *thread, 4065 const sp<Client>& client, 4066 uint32_t sampleRate, 4067 audio_format_t format, 4068 uint32_t channelMask, 4069 int frameCount, 4070 const sp<IMemory>& sharedBuffer, 4071 int sessionId) 4072 : RefBase(), 4073 mThread(thread), 4074 mClient(client), 4075 mCblk(NULL), 4076 // mBuffer 4077 // mBufferEnd 4078 mFrameCount(0), 4079 mState(IDLE), 4080 mSampleRate(sampleRate), 4081 mFormat(format), 4082 mStepServerFailed(false), 4083 mSessionId(sessionId) 4084 // mChannelCount 4085 // mChannelMask 4086{ 4087 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4088 4089 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4090 size_t size = sizeof(audio_track_cblk_t); 4091 uint8_t channelCount = popcount(channelMask); 4092 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4093 if (sharedBuffer == 0) { 4094 size += bufferSize; 4095 } 4096 4097 if (client != NULL) { 4098 mCblkMemory = client->heap()->allocate(size); 4099 if (mCblkMemory != 0) { 4100 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4101 if (mCblk != NULL) { // construct the shared structure in-place. 4102 new(mCblk) audio_track_cblk_t(); 4103 // clear all buffers 4104 mCblk->frameCount = frameCount; 4105 mCblk->sampleRate = sampleRate; 4106// uncomment the following lines to quickly test 32-bit wraparound 4107// mCblk->user = 0xffff0000; 4108// mCblk->server = 0xffff0000; 4109// mCblk->userBase = 0xffff0000; 4110// mCblk->serverBase = 0xffff0000; 4111 mChannelCount = channelCount; 4112 mChannelMask = channelMask; 4113 if (sharedBuffer == 0) { 4114 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4115 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4116 // Force underrun condition to avoid false underrun callback until first data is 4117 // written to buffer (other flags are cleared) 4118 mCblk->flags = CBLK_UNDERRUN_ON; 4119 } else { 4120 mBuffer = sharedBuffer->pointer(); 4121 } 4122 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4123 } 4124 } else { 4125 ALOGE("not enough memory for AudioTrack size=%u", size); 4126 client->heap()->dump("AudioTrack"); 4127 return; 4128 } 4129 } else { 4130 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4131 // construct the shared structure in-place. 4132 new(mCblk) audio_track_cblk_t(); 4133 // clear all buffers 4134 mCblk->frameCount = frameCount; 4135 mCblk->sampleRate = sampleRate; 4136// uncomment the following lines to quickly test 32-bit wraparound 4137// mCblk->user = 0xffff0000; 4138// mCblk->server = 0xffff0000; 4139// mCblk->userBase = 0xffff0000; 4140// mCblk->serverBase = 0xffff0000; 4141 mChannelCount = channelCount; 4142 mChannelMask = channelMask; 4143 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4144 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4145 // Force underrun condition to avoid false underrun callback until first data is 4146 // written to buffer (other flags are cleared) 4147 mCblk->flags = CBLK_UNDERRUN_ON; 4148 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4149 } 4150} 4151 4152AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4153{ 4154 if (mCblk != NULL) { 4155 if (mClient == 0) { 4156 delete mCblk; 4157 } else { 4158 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4159 } 4160 } 4161 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4162 if (mClient != 0) { 4163 // Client destructor must run with AudioFlinger mutex locked 4164 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4165 // If the client's reference count drops to zero, the associated destructor 4166 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4167 // relying on the automatic clear() at end of scope. 4168 mClient.clear(); 4169 } 4170} 4171 4172// AudioBufferProvider interface 4173// getNextBuffer() = 0; 4174// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4175void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4176{ 4177 buffer->raw = NULL; 4178 mFrameCount = buffer->frameCount; 4179 // FIXME See note at getNextBuffer() 4180 (void) step(); // ignore return value of step() 4181 buffer->frameCount = 0; 4182} 4183 4184bool AudioFlinger::ThreadBase::TrackBase::step() { 4185 bool result; 4186 audio_track_cblk_t* cblk = this->cblk(); 4187 4188 result = cblk->stepServer(mFrameCount); 4189 if (!result) { 4190 ALOGV("stepServer failed acquiring cblk mutex"); 4191 mStepServerFailed = true; 4192 } 4193 return result; 4194} 4195 4196void AudioFlinger::ThreadBase::TrackBase::reset() { 4197 audio_track_cblk_t* cblk = this->cblk(); 4198 4199 cblk->user = 0; 4200 cblk->server = 0; 4201 cblk->userBase = 0; 4202 cblk->serverBase = 0; 4203 mStepServerFailed = false; 4204 ALOGV("TrackBase::reset"); 4205} 4206 4207int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4208 return (int)mCblk->sampleRate; 4209} 4210 4211void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4212 audio_track_cblk_t* cblk = this->cblk(); 4213 size_t frameSize = cblk->frameSize; 4214 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4215 int8_t *bufferEnd = bufferStart + frames * frameSize; 4216 4217 // Check validity of returned pointer in case the track control block would have been corrupted. 4218 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4219 "TrackBase::getBuffer buffer out of range:\n" 4220 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4221 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4222 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4223 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4224 4225 return bufferStart; 4226} 4227 4228status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4229{ 4230 mSyncEvents.add(event); 4231 return NO_ERROR; 4232} 4233 4234// ---------------------------------------------------------------------------- 4235 4236// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4237AudioFlinger::PlaybackThread::Track::Track( 4238 PlaybackThread *thread, 4239 const sp<Client>& client, 4240 audio_stream_type_t streamType, 4241 uint32_t sampleRate, 4242 audio_format_t format, 4243 uint32_t channelMask, 4244 int frameCount, 4245 const sp<IMemory>& sharedBuffer, 4246 int sessionId, 4247 IAudioFlinger::track_flags_t flags) 4248 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4249 mMute(false), 4250 mFillingUpStatus(FS_INVALID), 4251 // mRetryCount initialized later when needed 4252 mSharedBuffer(sharedBuffer), 4253 mStreamType(streamType), 4254 mName(-1), // see note below 4255 mMainBuffer(thread->mixBuffer()), 4256 mAuxBuffer(NULL), 4257 mAuxEffectId(0), mHasVolumeController(false), 4258 mPresentationCompleteFrames(0), 4259 mFlags(flags), 4260 mFastIndex(-1), 4261 mUnderrunCount(0), 4262 mCachedVolume(1.0) 4263{ 4264 if (mCblk != NULL) { 4265 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4266 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4267 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4268 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4269 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4270 mCblk->mName = mName; 4271 if (mName < 0) { 4272 ALOGE("no more track names available"); 4273 return; 4274 } 4275 // only allocate a fast track index if we were able to allocate a normal track name 4276 if (flags & IAudioFlinger::TRACK_FAST) { 4277 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4278 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4279 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4280 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4281 // FIXME This is too eager. We allocate a fast track index before the 4282 // fast track becomes active. Since fast tracks are a scarce resource, 4283 // this means we are potentially denying other more important fast tracks from 4284 // being created. It would be better to allocate the index dynamically. 4285 mFastIndex = i; 4286 mCblk->mName = i; 4287 // Read the initial underruns because this field is never cleared by the fast mixer 4288 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4289 thread->mFastTrackAvailMask &= ~(1 << i); 4290 } 4291 } 4292 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4293} 4294 4295AudioFlinger::PlaybackThread::Track::~Track() 4296{ 4297 ALOGV("PlaybackThread::Track destructor"); 4298 sp<ThreadBase> thread = mThread.promote(); 4299 if (thread != 0) { 4300 Mutex::Autolock _l(thread->mLock); 4301 mState = TERMINATED; 4302 } 4303} 4304 4305void AudioFlinger::PlaybackThread::Track::destroy() 4306{ 4307 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4308 // by removing it from mTracks vector, so there is a risk that this Tracks's 4309 // destructor is called. As the destructor needs to lock mLock, 4310 // we must acquire a strong reference on this Track before locking mLock 4311 // here so that the destructor is called only when exiting this function. 4312 // On the other hand, as long as Track::destroy() is only called by 4313 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4314 // this Track with its member mTrack. 4315 sp<Track> keep(this); 4316 { // scope for mLock 4317 sp<ThreadBase> thread = mThread.promote(); 4318 if (thread != 0) { 4319 if (!isOutputTrack()) { 4320 if (mState == ACTIVE || mState == RESUMING) { 4321 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4322 4323#ifdef ADD_BATTERY_DATA 4324 // to track the speaker usage 4325 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4326#endif 4327 } 4328 AudioSystem::releaseOutput(thread->id()); 4329 } 4330 Mutex::Autolock _l(thread->mLock); 4331 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4332 playbackThread->destroyTrack_l(this); 4333 } 4334 } 4335} 4336 4337/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4338{ 4339 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4340 " Server User Main buf Aux Buf Flags Underruns\n"); 4341} 4342 4343void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4344{ 4345 uint32_t vlr = mCblk->getVolumeLR(); 4346 if (isFastTrack()) { 4347 sprintf(buffer, " F %2d", mFastIndex); 4348 } else { 4349 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4350 } 4351 track_state state = mState; 4352 char stateChar; 4353 switch (state) { 4354 case IDLE: 4355 stateChar = 'I'; 4356 break; 4357 case TERMINATED: 4358 stateChar = 'T'; 4359 break; 4360 case STOPPING_1: 4361 stateChar = 's'; 4362 break; 4363 case STOPPING_2: 4364 stateChar = '5'; 4365 break; 4366 case STOPPED: 4367 stateChar = 'S'; 4368 break; 4369 case RESUMING: 4370 stateChar = 'R'; 4371 break; 4372 case ACTIVE: 4373 stateChar = 'A'; 4374 break; 4375 case PAUSING: 4376 stateChar = 'p'; 4377 break; 4378 case PAUSED: 4379 stateChar = 'P'; 4380 break; 4381 case FLUSHED: 4382 stateChar = 'F'; 4383 break; 4384 default: 4385 stateChar = '?'; 4386 break; 4387 } 4388 char nowInUnderrun; 4389 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4390 case UNDERRUN_FULL: 4391 nowInUnderrun = ' '; 4392 break; 4393 case UNDERRUN_PARTIAL: 4394 nowInUnderrun = '<'; 4395 break; 4396 case UNDERRUN_EMPTY: 4397 nowInUnderrun = '*'; 4398 break; 4399 default: 4400 nowInUnderrun = '?'; 4401 break; 4402 } 4403 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4404 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4405 (mClient == 0) ? getpid_cached : mClient->pid(), 4406 mStreamType, 4407 mFormat, 4408 mChannelMask, 4409 mSessionId, 4410 mFrameCount, 4411 mCblk->frameCount, 4412 stateChar, 4413 mMute, 4414 mFillingUpStatus, 4415 mCblk->sampleRate, 4416 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4417 20.0 * log10((vlr >> 16) / 4096.0), 4418 mCblk->server, 4419 mCblk->user, 4420 (int)mMainBuffer, 4421 (int)mAuxBuffer, 4422 mCblk->flags, 4423 mUnderrunCount, 4424 nowInUnderrun); 4425} 4426 4427// AudioBufferProvider interface 4428status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4429 AudioBufferProvider::Buffer* buffer, int64_t pts) 4430{ 4431 audio_track_cblk_t* cblk = this->cblk(); 4432 uint32_t framesReady; 4433 uint32_t framesReq = buffer->frameCount; 4434 4435 // Check if last stepServer failed, try to step now 4436 if (mStepServerFailed) { 4437 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4438 // Since the fast mixer is higher priority than client callback thread, 4439 // it does not result in priority inversion for client. 4440 // But a non-blocking solution would be preferable to avoid 4441 // fast mixer being unable to tryLock(), and 4442 // to avoid the extra context switches if the client wakes up, 4443 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4444 if (!step()) goto getNextBuffer_exit; 4445 ALOGV("stepServer recovered"); 4446 mStepServerFailed = false; 4447 } 4448 4449 // FIXME Same as above 4450 framesReady = cblk->framesReady(); 4451 4452 if (CC_LIKELY(framesReady)) { 4453 uint32_t s = cblk->server; 4454 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4455 4456 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4457 if (framesReq > framesReady) { 4458 framesReq = framesReady; 4459 } 4460 if (framesReq > bufferEnd - s) { 4461 framesReq = bufferEnd - s; 4462 } 4463 4464 buffer->raw = getBuffer(s, framesReq); 4465 if (buffer->raw == NULL) goto getNextBuffer_exit; 4466 4467 buffer->frameCount = framesReq; 4468 return NO_ERROR; 4469 } 4470 4471getNextBuffer_exit: 4472 buffer->raw = NULL; 4473 buffer->frameCount = 0; 4474 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4475 return NOT_ENOUGH_DATA; 4476} 4477 4478// Note that framesReady() takes a mutex on the control block using tryLock(). 4479// This could result in priority inversion if framesReady() is called by the normal mixer, 4480// as the normal mixer thread runs at lower 4481// priority than the client's callback thread: there is a short window within framesReady() 4482// during which the normal mixer could be preempted, and the client callback would block. 4483// Another problem can occur if framesReady() is called by the fast mixer: 4484// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4485// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4486size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4487 return mCblk->framesReady(); 4488} 4489 4490// Don't call for fast tracks; the framesReady() could result in priority inversion 4491bool AudioFlinger::PlaybackThread::Track::isReady() const { 4492 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4493 4494 if (framesReady() >= mCblk->frameCount || 4495 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4496 mFillingUpStatus = FS_FILLED; 4497 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4498 return true; 4499 } 4500 return false; 4501} 4502 4503status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4504 int triggerSession) 4505{ 4506 status_t status = NO_ERROR; 4507 ALOGV("start(%d), calling pid %d session %d", 4508 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4509 4510 sp<ThreadBase> thread = mThread.promote(); 4511 if (thread != 0) { 4512 Mutex::Autolock _l(thread->mLock); 4513 track_state state = mState; 4514 // here the track could be either new, or restarted 4515 // in both cases "unstop" the track 4516 if (mState == PAUSED) { 4517 mState = TrackBase::RESUMING; 4518 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4519 } else { 4520 mState = TrackBase::ACTIVE; 4521 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4522 } 4523 4524 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4525 thread->mLock.unlock(); 4526 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4527 thread->mLock.lock(); 4528 4529#ifdef ADD_BATTERY_DATA 4530 // to track the speaker usage 4531 if (status == NO_ERROR) { 4532 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4533 } 4534#endif 4535 } 4536 if (status == NO_ERROR) { 4537 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4538 playbackThread->addTrack_l(this); 4539 } else { 4540 mState = state; 4541 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4542 } 4543 } else { 4544 status = BAD_VALUE; 4545 } 4546 return status; 4547} 4548 4549void AudioFlinger::PlaybackThread::Track::stop() 4550{ 4551 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4552 sp<ThreadBase> thread = mThread.promote(); 4553 if (thread != 0) { 4554 Mutex::Autolock _l(thread->mLock); 4555 track_state state = mState; 4556 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4557 // If the track is not active (PAUSED and buffers full), flush buffers 4558 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4559 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4560 reset(); 4561 mState = STOPPED; 4562 } else if (!isFastTrack()) { 4563 mState = STOPPED; 4564 } else { 4565 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4566 // and then to STOPPED and reset() when presentation is complete 4567 mState = STOPPING_1; 4568 } 4569 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4570 } 4571 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4572 thread->mLock.unlock(); 4573 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4574 thread->mLock.lock(); 4575 4576#ifdef ADD_BATTERY_DATA 4577 // to track the speaker usage 4578 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4579#endif 4580 } 4581 } 4582} 4583 4584void AudioFlinger::PlaybackThread::Track::pause() 4585{ 4586 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4587 sp<ThreadBase> thread = mThread.promote(); 4588 if (thread != 0) { 4589 Mutex::Autolock _l(thread->mLock); 4590 if (mState == ACTIVE || mState == RESUMING) { 4591 mState = PAUSING; 4592 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4593 if (!isOutputTrack()) { 4594 thread->mLock.unlock(); 4595 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4596 thread->mLock.lock(); 4597 4598#ifdef ADD_BATTERY_DATA 4599 // to track the speaker usage 4600 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4601#endif 4602 } 4603 } 4604 } 4605} 4606 4607void AudioFlinger::PlaybackThread::Track::flush() 4608{ 4609 ALOGV("flush(%d)", mName); 4610 sp<ThreadBase> thread = mThread.promote(); 4611 if (thread != 0) { 4612 Mutex::Autolock _l(thread->mLock); 4613 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4614 mState != PAUSING) { 4615 return; 4616 } 4617 // No point remaining in PAUSED state after a flush => go to 4618 // FLUSHED state 4619 mState = FLUSHED; 4620 // do not reset the track if it is still in the process of being stopped or paused. 4621 // this will be done by prepareTracks_l() when the track is stopped. 4622 // prepareTracks_l() will see mState == FLUSHED, then 4623 // remove from active track list, reset(), and trigger presentation complete 4624 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4625 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4626 reset(); 4627 } 4628 } 4629} 4630 4631void AudioFlinger::PlaybackThread::Track::reset() 4632{ 4633 // Do not reset twice to avoid discarding data written just after a flush and before 4634 // the audioflinger thread detects the track is stopped. 4635 if (!mResetDone) { 4636 TrackBase::reset(); 4637 // Force underrun condition to avoid false underrun callback until first data is 4638 // written to buffer 4639 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4640 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4641 mFillingUpStatus = FS_FILLING; 4642 mResetDone = true; 4643 if (mState == FLUSHED) { 4644 mState = IDLE; 4645 } 4646 } 4647} 4648 4649void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4650{ 4651 mMute = muted; 4652} 4653 4654status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4655{ 4656 status_t status = DEAD_OBJECT; 4657 sp<ThreadBase> thread = mThread.promote(); 4658 if (thread != 0) { 4659 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4660 sp<AudioFlinger> af = mClient->audioFlinger(); 4661 4662 Mutex::Autolock _l(af->mLock); 4663 4664 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4665 4666 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4667 Mutex::Autolock _dl(playbackThread->mLock); 4668 Mutex::Autolock _sl(srcThread->mLock); 4669 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4670 if (chain == 0) { 4671 return INVALID_OPERATION; 4672 } 4673 4674 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4675 if (effect == 0) { 4676 return INVALID_OPERATION; 4677 } 4678 srcThread->removeEffect_l(effect); 4679 playbackThread->addEffect_l(effect); 4680 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4681 if (effect->state() == EffectModule::ACTIVE || 4682 effect->state() == EffectModule::STOPPING) { 4683 effect->start(); 4684 } 4685 4686 sp<EffectChain> dstChain = effect->chain().promote(); 4687 if (dstChain == 0) { 4688 srcThread->addEffect_l(effect); 4689 return INVALID_OPERATION; 4690 } 4691 AudioSystem::unregisterEffect(effect->id()); 4692 AudioSystem::registerEffect(&effect->desc(), 4693 srcThread->id(), 4694 dstChain->strategy(), 4695 AUDIO_SESSION_OUTPUT_MIX, 4696 effect->id()); 4697 } 4698 status = playbackThread->attachAuxEffect(this, EffectId); 4699 } 4700 return status; 4701} 4702 4703void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4704{ 4705 mAuxEffectId = EffectId; 4706 mAuxBuffer = buffer; 4707} 4708 4709bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4710 size_t audioHalFrames) 4711{ 4712 // a track is considered presented when the total number of frames written to audio HAL 4713 // corresponds to the number of frames written when presentationComplete() is called for the 4714 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4715 if (mPresentationCompleteFrames == 0) { 4716 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4717 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4718 mPresentationCompleteFrames, audioHalFrames); 4719 } 4720 if (framesWritten >= mPresentationCompleteFrames) { 4721 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4722 mSessionId, framesWritten); 4723 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4724 return true; 4725 } 4726 return false; 4727} 4728 4729void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4730{ 4731 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4732 if (mSyncEvents[i]->type() == type) { 4733 mSyncEvents[i]->trigger(); 4734 mSyncEvents.removeAt(i); 4735 i--; 4736 } 4737 } 4738} 4739 4740// implement VolumeBufferProvider interface 4741 4742uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4743{ 4744 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4745 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4746 uint32_t vlr = mCblk->getVolumeLR(); 4747 uint32_t vl = vlr & 0xFFFF; 4748 uint32_t vr = vlr >> 16; 4749 // track volumes come from shared memory, so can't be trusted and must be clamped 4750 if (vl > MAX_GAIN_INT) { 4751 vl = MAX_GAIN_INT; 4752 } 4753 if (vr > MAX_GAIN_INT) { 4754 vr = MAX_GAIN_INT; 4755 } 4756 // now apply the cached master volume and stream type volume; 4757 // this is trusted but lacks any synchronization or barrier so may be stale 4758 float v = mCachedVolume; 4759 vl *= v; 4760 vr *= v; 4761 // re-combine into U4.16 4762 vlr = (vr << 16) | (vl & 0xFFFF); 4763 // FIXME look at mute, pause, and stop flags 4764 return vlr; 4765} 4766 4767status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4768{ 4769 if (mState == TERMINATED || mState == PAUSED || 4770 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4771 (mState == STOPPED)))) { 4772 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4773 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4774 event->cancel(); 4775 return INVALID_OPERATION; 4776 } 4777 TrackBase::setSyncEvent(event); 4778 return NO_ERROR; 4779} 4780 4781// timed audio tracks 4782 4783sp<AudioFlinger::PlaybackThread::TimedTrack> 4784AudioFlinger::PlaybackThread::TimedTrack::create( 4785 PlaybackThread *thread, 4786 const sp<Client>& client, 4787 audio_stream_type_t streamType, 4788 uint32_t sampleRate, 4789 audio_format_t format, 4790 uint32_t channelMask, 4791 int frameCount, 4792 const sp<IMemory>& sharedBuffer, 4793 int sessionId) { 4794 if (!client->reserveTimedTrack()) 4795 return 0; 4796 4797 return new TimedTrack( 4798 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4799 sharedBuffer, sessionId); 4800} 4801 4802AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4803 PlaybackThread *thread, 4804 const sp<Client>& client, 4805 audio_stream_type_t streamType, 4806 uint32_t sampleRate, 4807 audio_format_t format, 4808 uint32_t channelMask, 4809 int frameCount, 4810 const sp<IMemory>& sharedBuffer, 4811 int sessionId) 4812 : Track(thread, client, streamType, sampleRate, format, channelMask, 4813 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4814 mQueueHeadInFlight(false), 4815 mTrimQueueHeadOnRelease(false), 4816 mFramesPendingInQueue(0), 4817 mTimedSilenceBuffer(NULL), 4818 mTimedSilenceBufferSize(0), 4819 mTimedAudioOutputOnTime(false), 4820 mMediaTimeTransformValid(false) 4821{ 4822 LocalClock lc; 4823 mLocalTimeFreq = lc.getLocalFreq(); 4824 4825 mLocalTimeToSampleTransform.a_zero = 0; 4826 mLocalTimeToSampleTransform.b_zero = 0; 4827 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4828 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4829 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4830 &mLocalTimeToSampleTransform.a_to_b_denom); 4831 4832 mMediaTimeToSampleTransform.a_zero = 0; 4833 mMediaTimeToSampleTransform.b_zero = 0; 4834 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4835 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4836 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4837 &mMediaTimeToSampleTransform.a_to_b_denom); 4838} 4839 4840AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4841 mClient->releaseTimedTrack(); 4842 delete [] mTimedSilenceBuffer; 4843} 4844 4845status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4846 size_t size, sp<IMemory>* buffer) { 4847 4848 Mutex::Autolock _l(mTimedBufferQueueLock); 4849 4850 trimTimedBufferQueue_l(); 4851 4852 // lazily initialize the shared memory heap for timed buffers 4853 if (mTimedMemoryDealer == NULL) { 4854 const int kTimedBufferHeapSize = 512 << 10; 4855 4856 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4857 "AudioFlingerTimed"); 4858 if (mTimedMemoryDealer == NULL) 4859 return NO_MEMORY; 4860 } 4861 4862 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4863 if (newBuffer == NULL) { 4864 newBuffer = mTimedMemoryDealer->allocate(size); 4865 if (newBuffer == NULL) 4866 return NO_MEMORY; 4867 } 4868 4869 *buffer = newBuffer; 4870 return NO_ERROR; 4871} 4872 4873// caller must hold mTimedBufferQueueLock 4874void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4875 int64_t mediaTimeNow; 4876 { 4877 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4878 if (!mMediaTimeTransformValid) 4879 return; 4880 4881 int64_t targetTimeNow; 4882 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4883 ? mCCHelper.getCommonTime(&targetTimeNow) 4884 : mCCHelper.getLocalTime(&targetTimeNow); 4885 4886 if (OK != res) 4887 return; 4888 4889 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4890 &mediaTimeNow)) { 4891 return; 4892 } 4893 } 4894 4895 size_t trimEnd; 4896 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4897 int64_t bufEnd; 4898 4899 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4900 // We have a next buffer. Just use its PTS as the PTS of the frame 4901 // following the last frame in this buffer. If the stream is sparse 4902 // (ie, there are deliberate gaps left in the stream which should be 4903 // filled with silence by the TimedAudioTrack), then this can result 4904 // in one extra buffer being left un-trimmed when it could have 4905 // been. In general, this is not typical, and we would rather 4906 // optimized away the TS calculation below for the more common case 4907 // where PTSes are contiguous. 4908 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4909 } else { 4910 // We have no next buffer. Compute the PTS of the frame following 4911 // the last frame in this buffer by computing the duration of of 4912 // this frame in media time units and adding it to the PTS of the 4913 // buffer. 4914 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4915 / mCblk->frameSize; 4916 4917 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4918 &bufEnd)) { 4919 ALOGE("Failed to convert frame count of %lld to media time" 4920 " duration" " (scale factor %d/%u) in %s", 4921 frameCount, 4922 mMediaTimeToSampleTransform.a_to_b_numer, 4923 mMediaTimeToSampleTransform.a_to_b_denom, 4924 __PRETTY_FUNCTION__); 4925 break; 4926 } 4927 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4928 } 4929 4930 if (bufEnd > mediaTimeNow) 4931 break; 4932 4933 // Is the buffer we want to use in the middle of a mix operation right 4934 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4935 // from the mixer which should be coming back shortly. 4936 if (!trimEnd && mQueueHeadInFlight) { 4937 mTrimQueueHeadOnRelease = true; 4938 } 4939 } 4940 4941 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4942 if (trimStart < trimEnd) { 4943 // Update the bookkeeping for framesReady() 4944 for (size_t i = trimStart; i < trimEnd; ++i) { 4945 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4946 } 4947 4948 // Now actually remove the buffers from the queue. 4949 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4950 } 4951} 4952 4953void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4954 const char* logTag) { 4955 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4956 "%s called (reason \"%s\"), but timed buffer queue has no" 4957 " elements to trim.", __FUNCTION__, logTag); 4958 4959 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4960 mTimedBufferQueue.removeAt(0); 4961} 4962 4963void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4964 const TimedBuffer& buf, 4965 const char* logTag) { 4966 uint32_t bufBytes = buf.buffer()->size(); 4967 uint32_t consumedAlready = buf.position(); 4968 4969 ALOG_ASSERT(consumedAlready <= bufBytes, 4970 "Bad bookkeeping while updating frames pending. Timed buffer is" 4971 " only %u bytes long, but claims to have consumed %u" 4972 " bytes. (update reason: \"%s\")", 4973 bufBytes, consumedAlready, logTag); 4974 4975 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4976 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4977 "Bad bookkeeping while updating frames pending. Should have at" 4978 " least %u queued frames, but we think we have only %u. (update" 4979 " reason: \"%s\")", 4980 bufFrames, mFramesPendingInQueue, logTag); 4981 4982 mFramesPendingInQueue -= bufFrames; 4983} 4984 4985status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4986 const sp<IMemory>& buffer, int64_t pts) { 4987 4988 { 4989 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4990 if (!mMediaTimeTransformValid) 4991 return INVALID_OPERATION; 4992 } 4993 4994 Mutex::Autolock _l(mTimedBufferQueueLock); 4995 4996 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4997 mFramesPendingInQueue += bufFrames; 4998 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4999 5000 return NO_ERROR; 5001} 5002 5003status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5004 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5005 5006 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5007 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5008 target); 5009 5010 if (!(target == TimedAudioTrack::LOCAL_TIME || 5011 target == TimedAudioTrack::COMMON_TIME)) { 5012 return BAD_VALUE; 5013 } 5014 5015 Mutex::Autolock lock(mMediaTimeTransformLock); 5016 mMediaTimeTransform = xform; 5017 mMediaTimeTransformTarget = target; 5018 mMediaTimeTransformValid = true; 5019 5020 return NO_ERROR; 5021} 5022 5023#define min(a, b) ((a) < (b) ? (a) : (b)) 5024 5025// implementation of getNextBuffer for tracks whose buffers have timestamps 5026status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5027 AudioBufferProvider::Buffer* buffer, int64_t pts) 5028{ 5029 if (pts == AudioBufferProvider::kInvalidPTS) { 5030 buffer->raw = NULL; 5031 buffer->frameCount = 0; 5032 mTimedAudioOutputOnTime = false; 5033 return INVALID_OPERATION; 5034 } 5035 5036 Mutex::Autolock _l(mTimedBufferQueueLock); 5037 5038 ALOG_ASSERT(!mQueueHeadInFlight, 5039 "getNextBuffer called without releaseBuffer!"); 5040 5041 while (true) { 5042 5043 // if we have no timed buffers, then fail 5044 if (mTimedBufferQueue.isEmpty()) { 5045 buffer->raw = NULL; 5046 buffer->frameCount = 0; 5047 return NOT_ENOUGH_DATA; 5048 } 5049 5050 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5051 5052 // calculate the PTS of the head of the timed buffer queue expressed in 5053 // local time 5054 int64_t headLocalPTS; 5055 { 5056 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5057 5058 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5059 5060 if (mMediaTimeTransform.a_to_b_denom == 0) { 5061 // the transform represents a pause, so yield silence 5062 timedYieldSilence_l(buffer->frameCount, buffer); 5063 return NO_ERROR; 5064 } 5065 5066 int64_t transformedPTS; 5067 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5068 &transformedPTS)) { 5069 // the transform failed. this shouldn't happen, but if it does 5070 // then just drop this buffer 5071 ALOGW("timedGetNextBuffer transform failed"); 5072 buffer->raw = NULL; 5073 buffer->frameCount = 0; 5074 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5075 return NO_ERROR; 5076 } 5077 5078 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5079 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5080 &headLocalPTS)) { 5081 buffer->raw = NULL; 5082 buffer->frameCount = 0; 5083 return INVALID_OPERATION; 5084 } 5085 } else { 5086 headLocalPTS = transformedPTS; 5087 } 5088 } 5089 5090 // adjust the head buffer's PTS to reflect the portion of the head buffer 5091 // that has already been consumed 5092 int64_t effectivePTS = headLocalPTS + 5093 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5094 5095 // Calculate the delta in samples between the head of the input buffer 5096 // queue and the start of the next output buffer that will be written. 5097 // If the transformation fails because of over or underflow, it means 5098 // that the sample's position in the output stream is so far out of 5099 // whack that it should just be dropped. 5100 int64_t sampleDelta; 5101 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5102 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5103 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5104 " mix"); 5105 continue; 5106 } 5107 if (!mLocalTimeToSampleTransform.doForwardTransform( 5108 (effectivePTS - pts) << 32, &sampleDelta)) { 5109 ALOGV("*** too late during sample rate transform: dropped buffer"); 5110 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5111 continue; 5112 } 5113 5114 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5115 " sampleDelta=[%d.%08x]", 5116 head.pts(), head.position(), pts, 5117 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5118 + (sampleDelta >> 32)), 5119 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5120 5121 // if the delta between the ideal placement for the next input sample and 5122 // the current output position is within this threshold, then we will 5123 // concatenate the next input samples to the previous output 5124 const int64_t kSampleContinuityThreshold = 5125 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5126 5127 // if this is the first buffer of audio that we're emitting from this track 5128 // then it should be almost exactly on time. 5129 const int64_t kSampleStartupThreshold = 1LL << 32; 5130 5131 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5132 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5133 // the next input is close enough to being on time, so concatenate it 5134 // with the last output 5135 timedYieldSamples_l(buffer); 5136 5137 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5138 head.position(), buffer->frameCount); 5139 return NO_ERROR; 5140 } 5141 5142 // Looks like our output is not on time. Reset our on timed status. 5143 // Next time we mix samples from our input queue, then should be within 5144 // the StartupThreshold. 5145 mTimedAudioOutputOnTime = false; 5146 if (sampleDelta > 0) { 5147 // the gap between the current output position and the proper start of 5148 // the next input sample is too big, so fill it with silence 5149 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5150 5151 timedYieldSilence_l(framesUntilNextInput, buffer); 5152 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5153 return NO_ERROR; 5154 } else { 5155 // the next input sample is late 5156 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5157 size_t onTimeSamplePosition = 5158 head.position() + lateFrames * mCblk->frameSize; 5159 5160 if (onTimeSamplePosition > head.buffer()->size()) { 5161 // all the remaining samples in the head are too late, so 5162 // drop it and move on 5163 ALOGV("*** too late: dropped buffer"); 5164 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5165 continue; 5166 } else { 5167 // skip over the late samples 5168 head.setPosition(onTimeSamplePosition); 5169 5170 // yield the available samples 5171 timedYieldSamples_l(buffer); 5172 5173 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5174 return NO_ERROR; 5175 } 5176 } 5177 } 5178} 5179 5180// Yield samples from the timed buffer queue head up to the given output 5181// buffer's capacity. 5182// 5183// Caller must hold mTimedBufferQueueLock 5184void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5185 AudioBufferProvider::Buffer* buffer) { 5186 5187 const TimedBuffer& head = mTimedBufferQueue[0]; 5188 5189 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5190 head.position()); 5191 5192 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5193 mCblk->frameSize); 5194 size_t framesRequested = buffer->frameCount; 5195 buffer->frameCount = min(framesLeftInHead, framesRequested); 5196 5197 mQueueHeadInFlight = true; 5198 mTimedAudioOutputOnTime = true; 5199} 5200 5201// Yield samples of silence up to the given output buffer's capacity 5202// 5203// Caller must hold mTimedBufferQueueLock 5204void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5205 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5206 5207 // lazily allocate a buffer filled with silence 5208 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5209 delete [] mTimedSilenceBuffer; 5210 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5211 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5212 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5213 } 5214 5215 buffer->raw = mTimedSilenceBuffer; 5216 size_t framesRequested = buffer->frameCount; 5217 buffer->frameCount = min(numFrames, framesRequested); 5218 5219 mTimedAudioOutputOnTime = false; 5220} 5221 5222// AudioBufferProvider interface 5223void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5224 AudioBufferProvider::Buffer* buffer) { 5225 5226 Mutex::Autolock _l(mTimedBufferQueueLock); 5227 5228 // If the buffer which was just released is part of the buffer at the head 5229 // of the queue, be sure to update the amt of the buffer which has been 5230 // consumed. If the buffer being returned is not part of the head of the 5231 // queue, its either because the buffer is part of the silence buffer, or 5232 // because the head of the timed queue was trimmed after the mixer called 5233 // getNextBuffer but before the mixer called releaseBuffer. 5234 if (buffer->raw == mTimedSilenceBuffer) { 5235 ALOG_ASSERT(!mQueueHeadInFlight, 5236 "Queue head in flight during release of silence buffer!"); 5237 goto done; 5238 } 5239 5240 ALOG_ASSERT(mQueueHeadInFlight, 5241 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5242 " head in flight."); 5243 5244 if (mTimedBufferQueue.size()) { 5245 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5246 5247 void* start = head.buffer()->pointer(); 5248 void* end = reinterpret_cast<void*>( 5249 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5250 + head.buffer()->size()); 5251 5252 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5253 "released buffer not within the head of the timed buffer" 5254 " queue; qHead = [%p, %p], released buffer = %p", 5255 start, end, buffer->raw); 5256 5257 head.setPosition(head.position() + 5258 (buffer->frameCount * mCblk->frameSize)); 5259 mQueueHeadInFlight = false; 5260 5261 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5262 "Bad bookkeeping during releaseBuffer! Should have at" 5263 " least %u queued frames, but we think we have only %u", 5264 buffer->frameCount, mFramesPendingInQueue); 5265 5266 mFramesPendingInQueue -= buffer->frameCount; 5267 5268 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5269 || mTrimQueueHeadOnRelease) { 5270 trimTimedBufferQueueHead_l("releaseBuffer"); 5271 mTrimQueueHeadOnRelease = false; 5272 } 5273 } else { 5274 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5275 " buffers in the timed buffer queue"); 5276 } 5277 5278done: 5279 buffer->raw = 0; 5280 buffer->frameCount = 0; 5281} 5282 5283size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5284 Mutex::Autolock _l(mTimedBufferQueueLock); 5285 return mFramesPendingInQueue; 5286} 5287 5288AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5289 : mPTS(0), mPosition(0) {} 5290 5291AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5292 const sp<IMemory>& buffer, int64_t pts) 5293 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5294 5295// ---------------------------------------------------------------------------- 5296 5297// RecordTrack constructor must be called with AudioFlinger::mLock held 5298AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5299 RecordThread *thread, 5300 const sp<Client>& client, 5301 uint32_t sampleRate, 5302 audio_format_t format, 5303 uint32_t channelMask, 5304 int frameCount, 5305 int sessionId) 5306 : TrackBase(thread, client, sampleRate, format, 5307 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5308 mOverflow(false) 5309{ 5310 if (mCblk != NULL) { 5311 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5312 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5313 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5314 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5315 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5316 } else { 5317 mCblk->frameSize = sizeof(int8_t); 5318 } 5319 } 5320} 5321 5322AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5323{ 5324 sp<ThreadBase> thread = mThread.promote(); 5325 if (thread != 0) { 5326 AudioSystem::releaseInput(thread->id()); 5327 } 5328} 5329 5330// AudioBufferProvider interface 5331status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5332{ 5333 audio_track_cblk_t* cblk = this->cblk(); 5334 uint32_t framesAvail; 5335 uint32_t framesReq = buffer->frameCount; 5336 5337 // Check if last stepServer failed, try to step now 5338 if (mStepServerFailed) { 5339 if (!step()) goto getNextBuffer_exit; 5340 ALOGV("stepServer recovered"); 5341 mStepServerFailed = false; 5342 } 5343 5344 framesAvail = cblk->framesAvailable_l(); 5345 5346 if (CC_LIKELY(framesAvail)) { 5347 uint32_t s = cblk->server; 5348 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5349 5350 if (framesReq > framesAvail) { 5351 framesReq = framesAvail; 5352 } 5353 if (framesReq > bufferEnd - s) { 5354 framesReq = bufferEnd - s; 5355 } 5356 5357 buffer->raw = getBuffer(s, framesReq); 5358 if (buffer->raw == NULL) goto getNextBuffer_exit; 5359 5360 buffer->frameCount = framesReq; 5361 return NO_ERROR; 5362 } 5363 5364getNextBuffer_exit: 5365 buffer->raw = NULL; 5366 buffer->frameCount = 0; 5367 return NOT_ENOUGH_DATA; 5368} 5369 5370status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5371 int triggerSession) 5372{ 5373 sp<ThreadBase> thread = mThread.promote(); 5374 if (thread != 0) { 5375 RecordThread *recordThread = (RecordThread *)thread.get(); 5376 return recordThread->start(this, event, triggerSession); 5377 } else { 5378 return BAD_VALUE; 5379 } 5380} 5381 5382void AudioFlinger::RecordThread::RecordTrack::stop() 5383{ 5384 sp<ThreadBase> thread = mThread.promote(); 5385 if (thread != 0) { 5386 RecordThread *recordThread = (RecordThread *)thread.get(); 5387 recordThread->stop(this); 5388 TrackBase::reset(); 5389 // Force overrun condition to avoid false overrun callback until first data is 5390 // read from buffer 5391 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5392 } 5393} 5394 5395void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5396{ 5397 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5398 (mClient == 0) ? getpid_cached : mClient->pid(), 5399 mFormat, 5400 mChannelMask, 5401 mSessionId, 5402 mFrameCount, 5403 mState, 5404 mCblk->sampleRate, 5405 mCblk->server, 5406 mCblk->user); 5407} 5408 5409 5410// ---------------------------------------------------------------------------- 5411 5412AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5413 PlaybackThread *playbackThread, 5414 DuplicatingThread *sourceThread, 5415 uint32_t sampleRate, 5416 audio_format_t format, 5417 uint32_t channelMask, 5418 int frameCount) 5419 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5420 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5421 mActive(false), mSourceThread(sourceThread) 5422{ 5423 5424 if (mCblk != NULL) { 5425 mCblk->flags |= CBLK_DIRECTION_OUT; 5426 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5427 mOutBuffer.frameCount = 0; 5428 playbackThread->mTracks.add(this); 5429 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5430 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5431 mCblk, mBuffer, mCblk->buffers, 5432 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5433 } else { 5434 ALOGW("Error creating output track on thread %p", playbackThread); 5435 } 5436} 5437 5438AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5439{ 5440 clearBufferQueue(); 5441} 5442 5443status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5444 int triggerSession) 5445{ 5446 status_t status = Track::start(event, triggerSession); 5447 if (status != NO_ERROR) { 5448 return status; 5449 } 5450 5451 mActive = true; 5452 mRetryCount = 127; 5453 return status; 5454} 5455 5456void AudioFlinger::PlaybackThread::OutputTrack::stop() 5457{ 5458 Track::stop(); 5459 clearBufferQueue(); 5460 mOutBuffer.frameCount = 0; 5461 mActive = false; 5462} 5463 5464bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5465{ 5466 Buffer *pInBuffer; 5467 Buffer inBuffer; 5468 uint32_t channelCount = mChannelCount; 5469 bool outputBufferFull = false; 5470 inBuffer.frameCount = frames; 5471 inBuffer.i16 = data; 5472 5473 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5474 5475 if (!mActive && frames != 0) { 5476 start(); 5477 sp<ThreadBase> thread = mThread.promote(); 5478 if (thread != 0) { 5479 MixerThread *mixerThread = (MixerThread *)thread.get(); 5480 if (mCblk->frameCount > frames){ 5481 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5482 uint32_t startFrames = (mCblk->frameCount - frames); 5483 pInBuffer = new Buffer; 5484 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5485 pInBuffer->frameCount = startFrames; 5486 pInBuffer->i16 = pInBuffer->mBuffer; 5487 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5488 mBufferQueue.add(pInBuffer); 5489 } else { 5490 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5491 } 5492 } 5493 } 5494 } 5495 5496 while (waitTimeLeftMs) { 5497 // First write pending buffers, then new data 5498 if (mBufferQueue.size()) { 5499 pInBuffer = mBufferQueue.itemAt(0); 5500 } else { 5501 pInBuffer = &inBuffer; 5502 } 5503 5504 if (pInBuffer->frameCount == 0) { 5505 break; 5506 } 5507 5508 if (mOutBuffer.frameCount == 0) { 5509 mOutBuffer.frameCount = pInBuffer->frameCount; 5510 nsecs_t startTime = systemTime(); 5511 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5512 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5513 outputBufferFull = true; 5514 break; 5515 } 5516 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5517 if (waitTimeLeftMs >= waitTimeMs) { 5518 waitTimeLeftMs -= waitTimeMs; 5519 } else { 5520 waitTimeLeftMs = 0; 5521 } 5522 } 5523 5524 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5525 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5526 mCblk->stepUser(outFrames); 5527 pInBuffer->frameCount -= outFrames; 5528 pInBuffer->i16 += outFrames * channelCount; 5529 mOutBuffer.frameCount -= outFrames; 5530 mOutBuffer.i16 += outFrames * channelCount; 5531 5532 if (pInBuffer->frameCount == 0) { 5533 if (mBufferQueue.size()) { 5534 mBufferQueue.removeAt(0); 5535 delete [] pInBuffer->mBuffer; 5536 delete pInBuffer; 5537 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5538 } else { 5539 break; 5540 } 5541 } 5542 } 5543 5544 // If we could not write all frames, allocate a buffer and queue it for next time. 5545 if (inBuffer.frameCount) { 5546 sp<ThreadBase> thread = mThread.promote(); 5547 if (thread != 0 && !thread->standby()) { 5548 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5549 pInBuffer = new Buffer; 5550 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5551 pInBuffer->frameCount = inBuffer.frameCount; 5552 pInBuffer->i16 = pInBuffer->mBuffer; 5553 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5554 mBufferQueue.add(pInBuffer); 5555 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5556 } else { 5557 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5558 } 5559 } 5560 } 5561 5562 // Calling write() with a 0 length buffer, means that no more data will be written: 5563 // If no more buffers are pending, fill output track buffer to make sure it is started 5564 // by output mixer. 5565 if (frames == 0 && mBufferQueue.size() == 0) { 5566 if (mCblk->user < mCblk->frameCount) { 5567 frames = mCblk->frameCount - mCblk->user; 5568 pInBuffer = new Buffer; 5569 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5570 pInBuffer->frameCount = frames; 5571 pInBuffer->i16 = pInBuffer->mBuffer; 5572 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5573 mBufferQueue.add(pInBuffer); 5574 } else if (mActive) { 5575 stop(); 5576 } 5577 } 5578 5579 return outputBufferFull; 5580} 5581 5582status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5583{ 5584 int active; 5585 status_t result; 5586 audio_track_cblk_t* cblk = mCblk; 5587 uint32_t framesReq = buffer->frameCount; 5588 5589// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5590 buffer->frameCount = 0; 5591 5592 uint32_t framesAvail = cblk->framesAvailable(); 5593 5594 5595 if (framesAvail == 0) { 5596 Mutex::Autolock _l(cblk->lock); 5597 goto start_loop_here; 5598 while (framesAvail == 0) { 5599 active = mActive; 5600 if (CC_UNLIKELY(!active)) { 5601 ALOGV("Not active and NO_MORE_BUFFERS"); 5602 return NO_MORE_BUFFERS; 5603 } 5604 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5605 if (result != NO_ERROR) { 5606 return NO_MORE_BUFFERS; 5607 } 5608 // read the server count again 5609 start_loop_here: 5610 framesAvail = cblk->framesAvailable_l(); 5611 } 5612 } 5613 5614// if (framesAvail < framesReq) { 5615// return NO_MORE_BUFFERS; 5616// } 5617 5618 if (framesReq > framesAvail) { 5619 framesReq = framesAvail; 5620 } 5621 5622 uint32_t u = cblk->user; 5623 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5624 5625 if (framesReq > bufferEnd - u) { 5626 framesReq = bufferEnd - u; 5627 } 5628 5629 buffer->frameCount = framesReq; 5630 buffer->raw = (void *)cblk->buffer(u); 5631 return NO_ERROR; 5632} 5633 5634 5635void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5636{ 5637 size_t size = mBufferQueue.size(); 5638 5639 for (size_t i = 0; i < size; i++) { 5640 Buffer *pBuffer = mBufferQueue.itemAt(i); 5641 delete [] pBuffer->mBuffer; 5642 delete pBuffer; 5643 } 5644 mBufferQueue.clear(); 5645} 5646 5647// ---------------------------------------------------------------------------- 5648 5649AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5650 : RefBase(), 5651 mAudioFlinger(audioFlinger), 5652 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5653 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5654 mPid(pid), 5655 mTimedTrackCount(0) 5656{ 5657 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5658} 5659 5660// Client destructor must be called with AudioFlinger::mLock held 5661AudioFlinger::Client::~Client() 5662{ 5663 mAudioFlinger->removeClient_l(mPid); 5664} 5665 5666sp<MemoryDealer> AudioFlinger::Client::heap() const 5667{ 5668 return mMemoryDealer; 5669} 5670 5671// Reserve one of the limited slots for a timed audio track associated 5672// with this client 5673bool AudioFlinger::Client::reserveTimedTrack() 5674{ 5675 const int kMaxTimedTracksPerClient = 4; 5676 5677 Mutex::Autolock _l(mTimedTrackLock); 5678 5679 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5680 ALOGW("can not create timed track - pid %d has exceeded the limit", 5681 mPid); 5682 return false; 5683 } 5684 5685 mTimedTrackCount++; 5686 return true; 5687} 5688 5689// Release a slot for a timed audio track 5690void AudioFlinger::Client::releaseTimedTrack() 5691{ 5692 Mutex::Autolock _l(mTimedTrackLock); 5693 mTimedTrackCount--; 5694} 5695 5696// ---------------------------------------------------------------------------- 5697 5698AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5699 const sp<IAudioFlingerClient>& client, 5700 pid_t pid) 5701 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5702{ 5703} 5704 5705AudioFlinger::NotificationClient::~NotificationClient() 5706{ 5707} 5708 5709void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5710{ 5711 sp<NotificationClient> keep(this); 5712 mAudioFlinger->removeNotificationClient(mPid); 5713} 5714 5715// ---------------------------------------------------------------------------- 5716 5717AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5718 : BnAudioTrack(), 5719 mTrack(track) 5720{ 5721} 5722 5723AudioFlinger::TrackHandle::~TrackHandle() { 5724 // just stop the track on deletion, associated resources 5725 // will be freed from the main thread once all pending buffers have 5726 // been played. Unless it's not in the active track list, in which 5727 // case we free everything now... 5728 mTrack->destroy(); 5729} 5730 5731sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5732 return mTrack->getCblk(); 5733} 5734 5735status_t AudioFlinger::TrackHandle::start() { 5736 return mTrack->start(); 5737} 5738 5739void AudioFlinger::TrackHandle::stop() { 5740 mTrack->stop(); 5741} 5742 5743void AudioFlinger::TrackHandle::flush() { 5744 mTrack->flush(); 5745} 5746 5747void AudioFlinger::TrackHandle::mute(bool e) { 5748 mTrack->mute(e); 5749} 5750 5751void AudioFlinger::TrackHandle::pause() { 5752 mTrack->pause(); 5753} 5754 5755status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5756{ 5757 return mTrack->attachAuxEffect(EffectId); 5758} 5759 5760status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5761 sp<IMemory>* buffer) { 5762 if (!mTrack->isTimedTrack()) 5763 return INVALID_OPERATION; 5764 5765 PlaybackThread::TimedTrack* tt = 5766 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5767 return tt->allocateTimedBuffer(size, buffer); 5768} 5769 5770status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5771 int64_t pts) { 5772 if (!mTrack->isTimedTrack()) 5773 return INVALID_OPERATION; 5774 5775 PlaybackThread::TimedTrack* tt = 5776 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5777 return tt->queueTimedBuffer(buffer, pts); 5778} 5779 5780status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5781 const LinearTransform& xform, int target) { 5782 5783 if (!mTrack->isTimedTrack()) 5784 return INVALID_OPERATION; 5785 5786 PlaybackThread::TimedTrack* tt = 5787 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5788 return tt->setMediaTimeTransform( 5789 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5790} 5791 5792status_t AudioFlinger::TrackHandle::onTransact( 5793 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5794{ 5795 return BnAudioTrack::onTransact(code, data, reply, flags); 5796} 5797 5798// ---------------------------------------------------------------------------- 5799 5800sp<IAudioRecord> AudioFlinger::openRecord( 5801 pid_t pid, 5802 audio_io_handle_t input, 5803 uint32_t sampleRate, 5804 audio_format_t format, 5805 uint32_t channelMask, 5806 int frameCount, 5807 IAudioFlinger::track_flags_t flags, 5808 int *sessionId, 5809 status_t *status) 5810{ 5811 sp<RecordThread::RecordTrack> recordTrack; 5812 sp<RecordHandle> recordHandle; 5813 sp<Client> client; 5814 status_t lStatus; 5815 RecordThread *thread; 5816 size_t inFrameCount; 5817 int lSessionId; 5818 5819 // check calling permissions 5820 if (!recordingAllowed()) { 5821 lStatus = PERMISSION_DENIED; 5822 goto Exit; 5823 } 5824 5825 // add client to list 5826 { // scope for mLock 5827 Mutex::Autolock _l(mLock); 5828 thread = checkRecordThread_l(input); 5829 if (thread == NULL) { 5830 lStatus = BAD_VALUE; 5831 goto Exit; 5832 } 5833 5834 client = registerPid_l(pid); 5835 5836 // If no audio session id is provided, create one here 5837 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5838 lSessionId = *sessionId; 5839 } else { 5840 lSessionId = nextUniqueId(); 5841 if (sessionId != NULL) { 5842 *sessionId = lSessionId; 5843 } 5844 } 5845 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5846 recordTrack = thread->createRecordTrack_l(client, 5847 sampleRate, 5848 format, 5849 channelMask, 5850 frameCount, 5851 lSessionId, 5852 &lStatus); 5853 } 5854 if (lStatus != NO_ERROR) { 5855 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5856 // destructor is called by the TrackBase destructor with mLock held 5857 client.clear(); 5858 recordTrack.clear(); 5859 goto Exit; 5860 } 5861 5862 // return to handle to client 5863 recordHandle = new RecordHandle(recordTrack); 5864 lStatus = NO_ERROR; 5865 5866Exit: 5867 if (status) { 5868 *status = lStatus; 5869 } 5870 return recordHandle; 5871} 5872 5873// ---------------------------------------------------------------------------- 5874 5875AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5876 : BnAudioRecord(), 5877 mRecordTrack(recordTrack) 5878{ 5879} 5880 5881AudioFlinger::RecordHandle::~RecordHandle() { 5882 stop(); 5883} 5884 5885sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5886 return mRecordTrack->getCblk(); 5887} 5888 5889status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5890 ALOGV("RecordHandle::start()"); 5891 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5892} 5893 5894void AudioFlinger::RecordHandle::stop() { 5895 ALOGV("RecordHandle::stop()"); 5896 mRecordTrack->stop(); 5897} 5898 5899status_t AudioFlinger::RecordHandle::onTransact( 5900 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5901{ 5902 return BnAudioRecord::onTransact(code, data, reply, flags); 5903} 5904 5905// ---------------------------------------------------------------------------- 5906 5907AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5908 AudioStreamIn *input, 5909 uint32_t sampleRate, 5910 uint32_t channels, 5911 audio_io_handle_t id, 5912 uint32_t device) : 5913 ThreadBase(audioFlinger, id, device, RECORD), 5914 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5915 // mRsmpInIndex and mInputBytes set by readInputParameters() 5916 mReqChannelCount(popcount(channels)), 5917 mReqSampleRate(sampleRate) 5918 // mBytesRead is only meaningful while active, and so is cleared in start() 5919 // (but might be better to also clear here for dump?) 5920{ 5921 snprintf(mName, kNameLength, "AudioIn_%X", id); 5922 5923 readInputParameters(); 5924} 5925 5926 5927AudioFlinger::RecordThread::~RecordThread() 5928{ 5929 delete[] mRsmpInBuffer; 5930 delete mResampler; 5931 delete[] mRsmpOutBuffer; 5932} 5933 5934void AudioFlinger::RecordThread::onFirstRef() 5935{ 5936 run(mName, PRIORITY_URGENT_AUDIO); 5937} 5938 5939status_t AudioFlinger::RecordThread::readyToRun() 5940{ 5941 status_t status = initCheck(); 5942 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5943 return status; 5944} 5945 5946bool AudioFlinger::RecordThread::threadLoop() 5947{ 5948 AudioBufferProvider::Buffer buffer; 5949 sp<RecordTrack> activeTrack; 5950 Vector< sp<EffectChain> > effectChains; 5951 5952 nsecs_t lastWarning = 0; 5953 5954 acquireWakeLock(); 5955 5956 // start recording 5957 while (!exitPending()) { 5958 5959 processConfigEvents(); 5960 5961 { // scope for mLock 5962 Mutex::Autolock _l(mLock); 5963 checkForNewParameters_l(); 5964 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5965 if (!mStandby) { 5966 mInput->stream->common.standby(&mInput->stream->common); 5967 mStandby = true; 5968 } 5969 5970 if (exitPending()) break; 5971 5972 releaseWakeLock_l(); 5973 ALOGV("RecordThread: loop stopping"); 5974 // go to sleep 5975 mWaitWorkCV.wait(mLock); 5976 ALOGV("RecordThread: loop starting"); 5977 acquireWakeLock_l(); 5978 continue; 5979 } 5980 if (mActiveTrack != 0) { 5981 if (mActiveTrack->mState == TrackBase::PAUSING) { 5982 if (!mStandby) { 5983 mInput->stream->common.standby(&mInput->stream->common); 5984 mStandby = true; 5985 } 5986 mActiveTrack.clear(); 5987 mStartStopCond.broadcast(); 5988 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5989 if (mReqChannelCount != mActiveTrack->channelCount()) { 5990 mActiveTrack.clear(); 5991 mStartStopCond.broadcast(); 5992 } else if (mBytesRead != 0) { 5993 // record start succeeds only if first read from audio input 5994 // succeeds 5995 if (mBytesRead > 0) { 5996 mActiveTrack->mState = TrackBase::ACTIVE; 5997 } else { 5998 mActiveTrack.clear(); 5999 } 6000 mStartStopCond.broadcast(); 6001 } 6002 mStandby = false; 6003 } 6004 } 6005 lockEffectChains_l(effectChains); 6006 } 6007 6008 if (mActiveTrack != 0) { 6009 if (mActiveTrack->mState != TrackBase::ACTIVE && 6010 mActiveTrack->mState != TrackBase::RESUMING) { 6011 unlockEffectChains(effectChains); 6012 usleep(kRecordThreadSleepUs); 6013 continue; 6014 } 6015 for (size_t i = 0; i < effectChains.size(); i ++) { 6016 effectChains[i]->process_l(); 6017 } 6018 6019 buffer.frameCount = mFrameCount; 6020 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6021 size_t framesOut = buffer.frameCount; 6022 if (mResampler == NULL) { 6023 // no resampling 6024 while (framesOut) { 6025 size_t framesIn = mFrameCount - mRsmpInIndex; 6026 if (framesIn) { 6027 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6028 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6029 if (framesIn > framesOut) 6030 framesIn = framesOut; 6031 mRsmpInIndex += framesIn; 6032 framesOut -= framesIn; 6033 if ((int)mChannelCount == mReqChannelCount || 6034 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6035 memcpy(dst, src, framesIn * mFrameSize); 6036 } else { 6037 int16_t *src16 = (int16_t *)src; 6038 int16_t *dst16 = (int16_t *)dst; 6039 if (mChannelCount == 1) { 6040 while (framesIn--) { 6041 *dst16++ = *src16; 6042 *dst16++ = *src16++; 6043 } 6044 } else { 6045 while (framesIn--) { 6046 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6047 src16 += 2; 6048 } 6049 } 6050 } 6051 } 6052 if (framesOut && mFrameCount == mRsmpInIndex) { 6053 if (framesOut == mFrameCount && 6054 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6055 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6056 framesOut = 0; 6057 } else { 6058 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6059 mRsmpInIndex = 0; 6060 } 6061 if (mBytesRead < 0) { 6062 ALOGE("Error reading audio input"); 6063 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6064 // Force input into standby so that it tries to 6065 // recover at next read attempt 6066 mInput->stream->common.standby(&mInput->stream->common); 6067 usleep(kRecordThreadSleepUs); 6068 } 6069 mRsmpInIndex = mFrameCount; 6070 framesOut = 0; 6071 buffer.frameCount = 0; 6072 } 6073 } 6074 } 6075 } else { 6076 // resampling 6077 6078 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6079 // alter output frame count as if we were expecting stereo samples 6080 if (mChannelCount == 1 && mReqChannelCount == 1) { 6081 framesOut >>= 1; 6082 } 6083 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6084 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6085 // are 32 bit aligned which should be always true. 6086 if (mChannelCount == 2 && mReqChannelCount == 1) { 6087 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6088 // the resampler always outputs stereo samples: do post stereo to mono conversion 6089 int16_t *src = (int16_t *)mRsmpOutBuffer; 6090 int16_t *dst = buffer.i16; 6091 while (framesOut--) { 6092 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6093 src += 2; 6094 } 6095 } else { 6096 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6097 } 6098 6099 } 6100 if (mFramestoDrop == 0) { 6101 mActiveTrack->releaseBuffer(&buffer); 6102 } else { 6103 if (mFramestoDrop > 0) { 6104 mFramestoDrop -= buffer.frameCount; 6105 if (mFramestoDrop <= 0) { 6106 clearSyncStartEvent(); 6107 } 6108 } else { 6109 mFramestoDrop += buffer.frameCount; 6110 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6111 mSyncStartEvent->isCancelled()) { 6112 ALOGW("Synced record %s, session %d, trigger session %d", 6113 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6114 mActiveTrack->sessionId(), 6115 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6116 clearSyncStartEvent(); 6117 } 6118 } 6119 } 6120 mActiveTrack->overflow(); 6121 } 6122 // client isn't retrieving buffers fast enough 6123 else { 6124 if (!mActiveTrack->setOverflow()) { 6125 nsecs_t now = systemTime(); 6126 if ((now - lastWarning) > kWarningThrottleNs) { 6127 ALOGW("RecordThread: buffer overflow"); 6128 lastWarning = now; 6129 } 6130 } 6131 // Release the processor for a while before asking for a new buffer. 6132 // This will give the application more chance to read from the buffer and 6133 // clear the overflow. 6134 usleep(kRecordThreadSleepUs); 6135 } 6136 } 6137 // enable changes in effect chain 6138 unlockEffectChains(effectChains); 6139 effectChains.clear(); 6140 } 6141 6142 if (!mStandby) { 6143 mInput->stream->common.standby(&mInput->stream->common); 6144 } 6145 mActiveTrack.clear(); 6146 6147 mStartStopCond.broadcast(); 6148 6149 releaseWakeLock(); 6150 6151 ALOGV("RecordThread %p exiting", this); 6152 return false; 6153} 6154 6155 6156sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6157 const sp<AudioFlinger::Client>& client, 6158 uint32_t sampleRate, 6159 audio_format_t format, 6160 int channelMask, 6161 int frameCount, 6162 int sessionId, 6163 status_t *status) 6164{ 6165 sp<RecordTrack> track; 6166 status_t lStatus; 6167 6168 lStatus = initCheck(); 6169 if (lStatus != NO_ERROR) { 6170 ALOGE("Audio driver not initialized."); 6171 goto Exit; 6172 } 6173 6174 { // scope for mLock 6175 Mutex::Autolock _l(mLock); 6176 6177 track = new RecordTrack(this, client, sampleRate, 6178 format, channelMask, frameCount, sessionId); 6179 6180 if (track->getCblk() == 0) { 6181 lStatus = NO_MEMORY; 6182 goto Exit; 6183 } 6184 6185 mTrack = track.get(); 6186 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6187 bool suspend = audio_is_bluetooth_sco_device( 6188 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6189 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6190 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6191 } 6192 lStatus = NO_ERROR; 6193 6194Exit: 6195 if (status) { 6196 *status = lStatus; 6197 } 6198 return track; 6199} 6200 6201status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6202 AudioSystem::sync_event_t event, 6203 int triggerSession) 6204{ 6205 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6206 sp<ThreadBase> strongMe = this; 6207 status_t status = NO_ERROR; 6208 6209 if (event == AudioSystem::SYNC_EVENT_NONE) { 6210 clearSyncStartEvent(); 6211 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6212 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6213 triggerSession, 6214 recordTrack->sessionId(), 6215 syncStartEventCallback, 6216 this); 6217 // Sync event can be cancelled by the trigger session if the track is not in a 6218 // compatible state in which case we start record immediately 6219 if (mSyncStartEvent->isCancelled()) { 6220 clearSyncStartEvent(); 6221 } else { 6222 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6223 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6224 } 6225 } 6226 6227 { 6228 AutoMutex lock(mLock); 6229 if (mActiveTrack != 0) { 6230 if (recordTrack != mActiveTrack.get()) { 6231 status = -EBUSY; 6232 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6233 mActiveTrack->mState = TrackBase::ACTIVE; 6234 } 6235 return status; 6236 } 6237 6238 recordTrack->mState = TrackBase::IDLE; 6239 mActiveTrack = recordTrack; 6240 mLock.unlock(); 6241 status_t status = AudioSystem::startInput(mId); 6242 mLock.lock(); 6243 if (status != NO_ERROR) { 6244 mActiveTrack.clear(); 6245 clearSyncStartEvent(); 6246 return status; 6247 } 6248 mRsmpInIndex = mFrameCount; 6249 mBytesRead = 0; 6250 if (mResampler != NULL) { 6251 mResampler->reset(); 6252 } 6253 mActiveTrack->mState = TrackBase::RESUMING; 6254 // signal thread to start 6255 ALOGV("Signal record thread"); 6256 mWaitWorkCV.signal(); 6257 // do not wait for mStartStopCond if exiting 6258 if (exitPending()) { 6259 mActiveTrack.clear(); 6260 status = INVALID_OPERATION; 6261 goto startError; 6262 } 6263 mStartStopCond.wait(mLock); 6264 if (mActiveTrack == 0) { 6265 ALOGV("Record failed to start"); 6266 status = BAD_VALUE; 6267 goto startError; 6268 } 6269 ALOGV("Record started OK"); 6270 return status; 6271 } 6272startError: 6273 AudioSystem::stopInput(mId); 6274 clearSyncStartEvent(); 6275 return status; 6276} 6277 6278void AudioFlinger::RecordThread::clearSyncStartEvent() 6279{ 6280 if (mSyncStartEvent != 0) { 6281 mSyncStartEvent->cancel(); 6282 } 6283 mSyncStartEvent.clear(); 6284 mFramestoDrop = 0; 6285} 6286 6287void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6288{ 6289 sp<SyncEvent> strongEvent = event.promote(); 6290 6291 if (strongEvent != 0) { 6292 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6293 me->handleSyncStartEvent(strongEvent); 6294 } 6295} 6296 6297void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6298{ 6299 if (event == mSyncStartEvent) { 6300 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6301 // from audio HAL 6302 mFramestoDrop = mFrameCount * 2; 6303 } 6304} 6305 6306void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6307 ALOGV("RecordThread::stop"); 6308 sp<ThreadBase> strongMe = this; 6309 { 6310 AutoMutex lock(mLock); 6311 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6312 mActiveTrack->mState = TrackBase::PAUSING; 6313 // do not wait for mStartStopCond if exiting 6314 if (exitPending()) { 6315 return; 6316 } 6317 mStartStopCond.wait(mLock); 6318 // if we have been restarted, recordTrack == mActiveTrack.get() here 6319 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6320 mLock.unlock(); 6321 AudioSystem::stopInput(mId); 6322 mLock.lock(); 6323 ALOGV("Record stopped OK"); 6324 } 6325 } 6326 } 6327} 6328 6329bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6330{ 6331 return false; 6332} 6333 6334status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6335{ 6336 if (!isValidSyncEvent(event)) { 6337 return BAD_VALUE; 6338 } 6339 6340 Mutex::Autolock _l(mLock); 6341 6342 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6343 mTrack->setSyncEvent(event); 6344 return NO_ERROR; 6345 } 6346 return NAME_NOT_FOUND; 6347} 6348 6349status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6350{ 6351 const size_t SIZE = 256; 6352 char buffer[SIZE]; 6353 String8 result; 6354 6355 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6356 result.append(buffer); 6357 6358 if (mActiveTrack != 0) { 6359 result.append("Active Track:\n"); 6360 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6361 mActiveTrack->dump(buffer, SIZE); 6362 result.append(buffer); 6363 6364 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6365 result.append(buffer); 6366 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6367 result.append(buffer); 6368 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6369 result.append(buffer); 6370 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6371 result.append(buffer); 6372 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6373 result.append(buffer); 6374 6375 6376 } else { 6377 result.append("No record client\n"); 6378 } 6379 write(fd, result.string(), result.size()); 6380 6381 dumpBase(fd, args); 6382 dumpEffectChains(fd, args); 6383 6384 return NO_ERROR; 6385} 6386 6387// AudioBufferProvider interface 6388status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6389{ 6390 size_t framesReq = buffer->frameCount; 6391 size_t framesReady = mFrameCount - mRsmpInIndex; 6392 int channelCount; 6393 6394 if (framesReady == 0) { 6395 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6396 if (mBytesRead < 0) { 6397 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6398 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6399 // Force input into standby so that it tries to 6400 // recover at next read attempt 6401 mInput->stream->common.standby(&mInput->stream->common); 6402 usleep(kRecordThreadSleepUs); 6403 } 6404 buffer->raw = NULL; 6405 buffer->frameCount = 0; 6406 return NOT_ENOUGH_DATA; 6407 } 6408 mRsmpInIndex = 0; 6409 framesReady = mFrameCount; 6410 } 6411 6412 if (framesReq > framesReady) { 6413 framesReq = framesReady; 6414 } 6415 6416 if (mChannelCount == 1 && mReqChannelCount == 2) { 6417 channelCount = 1; 6418 } else { 6419 channelCount = 2; 6420 } 6421 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6422 buffer->frameCount = framesReq; 6423 return NO_ERROR; 6424} 6425 6426// AudioBufferProvider interface 6427void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6428{ 6429 mRsmpInIndex += buffer->frameCount; 6430 buffer->frameCount = 0; 6431} 6432 6433bool AudioFlinger::RecordThread::checkForNewParameters_l() 6434{ 6435 bool reconfig = false; 6436 6437 while (!mNewParameters.isEmpty()) { 6438 status_t status = NO_ERROR; 6439 String8 keyValuePair = mNewParameters[0]; 6440 AudioParameter param = AudioParameter(keyValuePair); 6441 int value; 6442 audio_format_t reqFormat = mFormat; 6443 int reqSamplingRate = mReqSampleRate; 6444 int reqChannelCount = mReqChannelCount; 6445 6446 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6447 reqSamplingRate = value; 6448 reconfig = true; 6449 } 6450 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6451 reqFormat = (audio_format_t) value; 6452 reconfig = true; 6453 } 6454 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6455 reqChannelCount = popcount(value); 6456 reconfig = true; 6457 } 6458 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6459 // do not accept frame count changes if tracks are open as the track buffer 6460 // size depends on frame count and correct behavior would not be guaranteed 6461 // if frame count is changed after track creation 6462 if (mActiveTrack != 0) { 6463 status = INVALID_OPERATION; 6464 } else { 6465 reconfig = true; 6466 } 6467 } 6468 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6469 // forward device change to effects that have requested to be 6470 // aware of attached audio device. 6471 for (size_t i = 0; i < mEffectChains.size(); i++) { 6472 mEffectChains[i]->setDevice_l(value); 6473 } 6474 // store input device and output device but do not forward output device to audio HAL. 6475 // Note that status is ignored by the caller for output device 6476 // (see AudioFlinger::setParameters() 6477 if (value & AUDIO_DEVICE_OUT_ALL) { 6478 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6479 status = BAD_VALUE; 6480 } else { 6481 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6482 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6483 if (mTrack != NULL) { 6484 bool suspend = audio_is_bluetooth_sco_device( 6485 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6486 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6487 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6488 } 6489 } 6490 mDevice |= (uint32_t)value; 6491 } 6492 if (status == NO_ERROR) { 6493 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6494 if (status == INVALID_OPERATION) { 6495 mInput->stream->common.standby(&mInput->stream->common); 6496 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6497 keyValuePair.string()); 6498 } 6499 if (reconfig) { 6500 if (status == BAD_VALUE && 6501 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6502 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6503 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6504 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6505 (reqChannelCount <= FCC_2)) { 6506 status = NO_ERROR; 6507 } 6508 if (status == NO_ERROR) { 6509 readInputParameters(); 6510 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6511 } 6512 } 6513 } 6514 6515 mNewParameters.removeAt(0); 6516 6517 mParamStatus = status; 6518 mParamCond.signal(); 6519 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6520 // already timed out waiting for the status and will never signal the condition. 6521 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6522 } 6523 return reconfig; 6524} 6525 6526String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6527{ 6528 char *s; 6529 String8 out_s8 = String8(); 6530 6531 Mutex::Autolock _l(mLock); 6532 if (initCheck() != NO_ERROR) { 6533 return out_s8; 6534 } 6535 6536 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6537 out_s8 = String8(s); 6538 free(s); 6539 return out_s8; 6540} 6541 6542void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6543 AudioSystem::OutputDescriptor desc; 6544 void *param2 = NULL; 6545 6546 switch (event) { 6547 case AudioSystem::INPUT_OPENED: 6548 case AudioSystem::INPUT_CONFIG_CHANGED: 6549 desc.channels = mChannelMask; 6550 desc.samplingRate = mSampleRate; 6551 desc.format = mFormat; 6552 desc.frameCount = mFrameCount; 6553 desc.latency = 0; 6554 param2 = &desc; 6555 break; 6556 6557 case AudioSystem::INPUT_CLOSED: 6558 default: 6559 break; 6560 } 6561 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6562} 6563 6564void AudioFlinger::RecordThread::readInputParameters() 6565{ 6566 delete mRsmpInBuffer; 6567 // mRsmpInBuffer is always assigned a new[] below 6568 delete mRsmpOutBuffer; 6569 mRsmpOutBuffer = NULL; 6570 delete mResampler; 6571 mResampler = NULL; 6572 6573 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6574 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6575 mChannelCount = (uint16_t)popcount(mChannelMask); 6576 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6577 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6578 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6579 mFrameCount = mInputBytes / mFrameSize; 6580 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6581 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6582 6583 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6584 { 6585 int channelCount; 6586 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6587 // stereo to mono post process as the resampler always outputs stereo. 6588 if (mChannelCount == 1 && mReqChannelCount == 2) { 6589 channelCount = 1; 6590 } else { 6591 channelCount = 2; 6592 } 6593 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6594 mResampler->setSampleRate(mSampleRate); 6595 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6596 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6597 6598 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6599 if (mChannelCount == 1 && mReqChannelCount == 1) { 6600 mFrameCount >>= 1; 6601 } 6602 6603 } 6604 mRsmpInIndex = mFrameCount; 6605} 6606 6607unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6608{ 6609 Mutex::Autolock _l(mLock); 6610 if (initCheck() != NO_ERROR) { 6611 return 0; 6612 } 6613 6614 return mInput->stream->get_input_frames_lost(mInput->stream); 6615} 6616 6617uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6618{ 6619 Mutex::Autolock _l(mLock); 6620 uint32_t result = 0; 6621 if (getEffectChain_l(sessionId) != 0) { 6622 result = EFFECT_SESSION; 6623 } 6624 6625 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6626 result |= TRACK_SESSION; 6627 } 6628 6629 return result; 6630} 6631 6632AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6633{ 6634 Mutex::Autolock _l(mLock); 6635 return mTrack; 6636} 6637 6638AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6639{ 6640 Mutex::Autolock _l(mLock); 6641 return mInput; 6642} 6643 6644AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6645{ 6646 Mutex::Autolock _l(mLock); 6647 AudioStreamIn *input = mInput; 6648 mInput = NULL; 6649 return input; 6650} 6651 6652// this method must always be called either with ThreadBase mLock held or inside the thread loop 6653audio_stream_t* AudioFlinger::RecordThread::stream() const 6654{ 6655 if (mInput == NULL) { 6656 return NULL; 6657 } 6658 return &mInput->stream->common; 6659} 6660 6661 6662// ---------------------------------------------------------------------------- 6663 6664audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6665{ 6666 if (!settingsAllowed()) { 6667 return 0; 6668 } 6669 Mutex::Autolock _l(mLock); 6670 return loadHwModule_l(name); 6671} 6672 6673// loadHwModule_l() must be called with AudioFlinger::mLock held 6674audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6675{ 6676 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6677 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6678 ALOGW("loadHwModule() module %s already loaded", name); 6679 return mAudioHwDevs.keyAt(i); 6680 } 6681 } 6682 6683 audio_hw_device_t *dev; 6684 6685 int rc = load_audio_interface(name, &dev); 6686 if (rc) { 6687 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6688 return 0; 6689 } 6690 6691 mHardwareStatus = AUDIO_HW_INIT; 6692 rc = dev->init_check(dev); 6693 mHardwareStatus = AUDIO_HW_IDLE; 6694 if (rc) { 6695 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6696 return 0; 6697 } 6698 6699 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6700 (NULL != dev->set_master_volume)) { 6701 AutoMutex lock(mHardwareLock); 6702 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6703 dev->set_master_volume(dev, mMasterVolume); 6704 mHardwareStatus = AUDIO_HW_IDLE; 6705 } 6706 6707 audio_module_handle_t handle = nextUniqueId(); 6708 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6709 6710 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6711 name, dev->common.module->name, dev->common.module->id, handle); 6712 6713 return handle; 6714 6715} 6716 6717audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6718 audio_devices_t *pDevices, 6719 uint32_t *pSamplingRate, 6720 audio_format_t *pFormat, 6721 audio_channel_mask_t *pChannelMask, 6722 uint32_t *pLatencyMs, 6723 audio_output_flags_t flags) 6724{ 6725 status_t status; 6726 PlaybackThread *thread = NULL; 6727 struct audio_config config = { 6728 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6729 channel_mask: pChannelMask ? *pChannelMask : 0, 6730 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6731 }; 6732 audio_stream_out_t *outStream = NULL; 6733 audio_hw_device_t *outHwDev; 6734 6735 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6736 module, 6737 (pDevices != NULL) ? (int)*pDevices : 0, 6738 config.sample_rate, 6739 config.format, 6740 config.channel_mask, 6741 flags); 6742 6743 if (pDevices == NULL || *pDevices == 0) { 6744 return 0; 6745 } 6746 6747 Mutex::Autolock _l(mLock); 6748 6749 outHwDev = findSuitableHwDev_l(module, *pDevices); 6750 if (outHwDev == NULL) 6751 return 0; 6752 6753 audio_io_handle_t id = nextUniqueId(); 6754 6755 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6756 6757 status = outHwDev->open_output_stream(outHwDev, 6758 id, 6759 *pDevices, 6760 (audio_output_flags_t)flags, 6761 &config, 6762 &outStream); 6763 6764 mHardwareStatus = AUDIO_HW_IDLE; 6765 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6766 outStream, 6767 config.sample_rate, 6768 config.format, 6769 config.channel_mask, 6770 status); 6771 6772 if (status == NO_ERROR && outStream != NULL) { 6773 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6774 6775 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6776 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6777 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6778 thread = new DirectOutputThread(this, output, id, *pDevices); 6779 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6780 } else { 6781 thread = new MixerThread(this, output, id, *pDevices); 6782 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6783 } 6784 mPlaybackThreads.add(id, thread); 6785 6786 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6787 if (pFormat != NULL) *pFormat = config.format; 6788 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6789 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6790 6791 // notify client processes of the new output creation 6792 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6793 6794 // the first primary output opened designates the primary hw device 6795 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6796 ALOGI("Using module %d has the primary audio interface", module); 6797 mPrimaryHardwareDev = outHwDev; 6798 6799 AutoMutex lock(mHardwareLock); 6800 mHardwareStatus = AUDIO_HW_SET_MODE; 6801 outHwDev->set_mode(outHwDev, mMode); 6802 6803 // Determine the level of master volume support the primary audio HAL has, 6804 // and set the initial master volume at the same time. 6805 float initialVolume = 1.0; 6806 mMasterVolumeSupportLvl = MVS_NONE; 6807 6808 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6809 if ((NULL != outHwDev->get_master_volume) && 6810 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6811 mMasterVolumeSupportLvl = MVS_FULL; 6812 } else { 6813 mMasterVolumeSupportLvl = MVS_SETONLY; 6814 initialVolume = 1.0; 6815 } 6816 6817 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6818 if ((NULL == outHwDev->set_master_volume) || 6819 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6820 mMasterVolumeSupportLvl = MVS_NONE; 6821 } 6822 // now that we have a primary device, initialize master volume on other devices 6823 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6824 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6825 6826 if ((dev != mPrimaryHardwareDev) && 6827 (NULL != dev->set_master_volume)) { 6828 dev->set_master_volume(dev, initialVolume); 6829 } 6830 } 6831 mHardwareStatus = AUDIO_HW_IDLE; 6832 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6833 ? initialVolume 6834 : 1.0; 6835 mMasterVolume = initialVolume; 6836 } 6837 return id; 6838 } 6839 6840 return 0; 6841} 6842 6843audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6844 audio_io_handle_t output2) 6845{ 6846 Mutex::Autolock _l(mLock); 6847 MixerThread *thread1 = checkMixerThread_l(output1); 6848 MixerThread *thread2 = checkMixerThread_l(output2); 6849 6850 if (thread1 == NULL || thread2 == NULL) { 6851 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6852 return 0; 6853 } 6854 6855 audio_io_handle_t id = nextUniqueId(); 6856 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6857 thread->addOutputTrack(thread2); 6858 mPlaybackThreads.add(id, thread); 6859 // notify client processes of the new output creation 6860 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6861 return id; 6862} 6863 6864status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6865{ 6866 // keep strong reference on the playback thread so that 6867 // it is not destroyed while exit() is executed 6868 sp<PlaybackThread> thread; 6869 { 6870 Mutex::Autolock _l(mLock); 6871 thread = checkPlaybackThread_l(output); 6872 if (thread == NULL) { 6873 return BAD_VALUE; 6874 } 6875 6876 ALOGV("closeOutput() %d", output); 6877 6878 if (thread->type() == ThreadBase::MIXER) { 6879 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6880 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6881 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6882 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6883 } 6884 } 6885 } 6886 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6887 mPlaybackThreads.removeItem(output); 6888 } 6889 thread->exit(); 6890 // The thread entity (active unit of execution) is no longer running here, 6891 // but the ThreadBase container still exists. 6892 6893 if (thread->type() != ThreadBase::DUPLICATING) { 6894 AudioStreamOut *out = thread->clearOutput(); 6895 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6896 // from now on thread->mOutput is NULL 6897 out->hwDev->close_output_stream(out->hwDev, out->stream); 6898 delete out; 6899 } 6900 return NO_ERROR; 6901} 6902 6903status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6904{ 6905 Mutex::Autolock _l(mLock); 6906 PlaybackThread *thread = checkPlaybackThread_l(output); 6907 6908 if (thread == NULL) { 6909 return BAD_VALUE; 6910 } 6911 6912 ALOGV("suspendOutput() %d", output); 6913 thread->suspend(); 6914 6915 return NO_ERROR; 6916} 6917 6918status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6919{ 6920 Mutex::Autolock _l(mLock); 6921 PlaybackThread *thread = checkPlaybackThread_l(output); 6922 6923 if (thread == NULL) { 6924 return BAD_VALUE; 6925 } 6926 6927 ALOGV("restoreOutput() %d", output); 6928 6929 thread->restore(); 6930 6931 return NO_ERROR; 6932} 6933 6934audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6935 audio_devices_t *pDevices, 6936 uint32_t *pSamplingRate, 6937 audio_format_t *pFormat, 6938 uint32_t *pChannelMask) 6939{ 6940 status_t status; 6941 RecordThread *thread = NULL; 6942 struct audio_config config = { 6943 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6944 channel_mask: pChannelMask ? *pChannelMask : 0, 6945 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6946 }; 6947 uint32_t reqSamplingRate = config.sample_rate; 6948 audio_format_t reqFormat = config.format; 6949 audio_channel_mask_t reqChannels = config.channel_mask; 6950 audio_stream_in_t *inStream = NULL; 6951 audio_hw_device_t *inHwDev; 6952 6953 if (pDevices == NULL || *pDevices == 0) { 6954 return 0; 6955 } 6956 6957 Mutex::Autolock _l(mLock); 6958 6959 inHwDev = findSuitableHwDev_l(module, *pDevices); 6960 if (inHwDev == NULL) 6961 return 0; 6962 6963 audio_io_handle_t id = nextUniqueId(); 6964 6965 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6966 &inStream); 6967 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6968 inStream, 6969 config.sample_rate, 6970 config.format, 6971 config.channel_mask, 6972 status); 6973 6974 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6975 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6976 // or stereo to mono conversions on 16 bit PCM inputs. 6977 if (status == BAD_VALUE && 6978 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6979 (config.sample_rate <= 2 * reqSamplingRate) && 6980 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6981 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6982 inStream = NULL; 6983 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6984 } 6985 6986 if (status == NO_ERROR && inStream != NULL) { 6987 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6988 6989 // Start record thread 6990 // RecorThread require both input and output device indication to forward to audio 6991 // pre processing modules 6992 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6993 thread = new RecordThread(this, 6994 input, 6995 reqSamplingRate, 6996 reqChannels, 6997 id, 6998 device); 6999 mRecordThreads.add(id, thread); 7000 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7001 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7002 if (pFormat != NULL) *pFormat = config.format; 7003 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7004 7005 input->stream->common.standby(&input->stream->common); 7006 7007 // notify client processes of the new input creation 7008 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7009 return id; 7010 } 7011 7012 return 0; 7013} 7014 7015status_t AudioFlinger::closeInput(audio_io_handle_t input) 7016{ 7017 // keep strong reference on the record thread so that 7018 // it is not destroyed while exit() is executed 7019 sp<RecordThread> thread; 7020 { 7021 Mutex::Autolock _l(mLock); 7022 thread = checkRecordThread_l(input); 7023 if (thread == 0) { 7024 return BAD_VALUE; 7025 } 7026 7027 ALOGV("closeInput() %d", input); 7028 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7029 mRecordThreads.removeItem(input); 7030 } 7031 thread->exit(); 7032 // The thread entity (active unit of execution) is no longer running here, 7033 // but the ThreadBase container still exists. 7034 7035 AudioStreamIn *in = thread->clearInput(); 7036 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7037 // from now on thread->mInput is NULL 7038 in->hwDev->close_input_stream(in->hwDev, in->stream); 7039 delete in; 7040 7041 return NO_ERROR; 7042} 7043 7044status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7045{ 7046 Mutex::Autolock _l(mLock); 7047 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7048 7049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7050 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7051 thread->invalidateTracks(stream); 7052 } 7053 7054 return NO_ERROR; 7055} 7056 7057 7058int AudioFlinger::newAudioSessionId() 7059{ 7060 return nextUniqueId(); 7061} 7062 7063void AudioFlinger::acquireAudioSessionId(int audioSession) 7064{ 7065 Mutex::Autolock _l(mLock); 7066 pid_t caller = IPCThreadState::self()->getCallingPid(); 7067 ALOGV("acquiring %d from %d", audioSession, caller); 7068 size_t num = mAudioSessionRefs.size(); 7069 for (size_t i = 0; i< num; i++) { 7070 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7071 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7072 ref->mCnt++; 7073 ALOGV(" incremented refcount to %d", ref->mCnt); 7074 return; 7075 } 7076 } 7077 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7078 ALOGV(" added new entry for %d", audioSession); 7079} 7080 7081void AudioFlinger::releaseAudioSessionId(int audioSession) 7082{ 7083 Mutex::Autolock _l(mLock); 7084 pid_t caller = IPCThreadState::self()->getCallingPid(); 7085 ALOGV("releasing %d from %d", audioSession, caller); 7086 size_t num = mAudioSessionRefs.size(); 7087 for (size_t i = 0; i< num; i++) { 7088 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7089 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7090 ref->mCnt--; 7091 ALOGV(" decremented refcount to %d", ref->mCnt); 7092 if (ref->mCnt == 0) { 7093 mAudioSessionRefs.removeAt(i); 7094 delete ref; 7095 purgeStaleEffects_l(); 7096 } 7097 return; 7098 } 7099 } 7100 ALOGW("session id %d not found for pid %d", audioSession, caller); 7101} 7102 7103void AudioFlinger::purgeStaleEffects_l() { 7104 7105 ALOGV("purging stale effects"); 7106 7107 Vector< sp<EffectChain> > chains; 7108 7109 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7110 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7111 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7112 sp<EffectChain> ec = t->mEffectChains[j]; 7113 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7114 chains.push(ec); 7115 } 7116 } 7117 } 7118 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7119 sp<RecordThread> t = mRecordThreads.valueAt(i); 7120 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7121 sp<EffectChain> ec = t->mEffectChains[j]; 7122 chains.push(ec); 7123 } 7124 } 7125 7126 for (size_t i = 0; i < chains.size(); i++) { 7127 sp<EffectChain> ec = chains[i]; 7128 int sessionid = ec->sessionId(); 7129 sp<ThreadBase> t = ec->mThread.promote(); 7130 if (t == 0) { 7131 continue; 7132 } 7133 size_t numsessionrefs = mAudioSessionRefs.size(); 7134 bool found = false; 7135 for (size_t k = 0; k < numsessionrefs; k++) { 7136 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7137 if (ref->mSessionid == sessionid) { 7138 ALOGV(" session %d still exists for %d with %d refs", 7139 sessionid, ref->mPid, ref->mCnt); 7140 found = true; 7141 break; 7142 } 7143 } 7144 if (!found) { 7145 Mutex::Autolock _l (t->mLock); 7146 // remove all effects from the chain 7147 while (ec->mEffects.size()) { 7148 sp<EffectModule> effect = ec->mEffects[0]; 7149 effect->unPin(); 7150 t->removeEffect_l(effect); 7151 if (effect->purgeHandles()) { 7152 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7153 } 7154 AudioSystem::unregisterEffect(effect->id()); 7155 } 7156 } 7157 } 7158 return; 7159} 7160 7161// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7162AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7163{ 7164 return mPlaybackThreads.valueFor(output).get(); 7165} 7166 7167// checkMixerThread_l() must be called with AudioFlinger::mLock held 7168AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7169{ 7170 PlaybackThread *thread = checkPlaybackThread_l(output); 7171 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7172} 7173 7174// checkRecordThread_l() must be called with AudioFlinger::mLock held 7175AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7176{ 7177 return mRecordThreads.valueFor(input).get(); 7178} 7179 7180uint32_t AudioFlinger::nextUniqueId() 7181{ 7182 return android_atomic_inc(&mNextUniqueId); 7183} 7184 7185AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7186{ 7187 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7188 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7189 AudioStreamOut *output = thread->getOutput(); 7190 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7191 return thread; 7192 } 7193 } 7194 return NULL; 7195} 7196 7197uint32_t AudioFlinger::primaryOutputDevice_l() const 7198{ 7199 PlaybackThread *thread = primaryPlaybackThread_l(); 7200 7201 if (thread == NULL) { 7202 return 0; 7203 } 7204 7205 return thread->device(); 7206} 7207 7208sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7209 int triggerSession, 7210 int listenerSession, 7211 sync_event_callback_t callBack, 7212 void *cookie) 7213{ 7214 Mutex::Autolock _l(mLock); 7215 7216 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7217 status_t playStatus = NAME_NOT_FOUND; 7218 status_t recStatus = NAME_NOT_FOUND; 7219 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7220 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7221 if (playStatus == NO_ERROR) { 7222 return event; 7223 } 7224 } 7225 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7226 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7227 if (recStatus == NO_ERROR) { 7228 return event; 7229 } 7230 } 7231 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7232 mPendingSyncEvents.add(event); 7233 } else { 7234 ALOGV("createSyncEvent() invalid event %d", event->type()); 7235 event.clear(); 7236 } 7237 return event; 7238} 7239 7240// ---------------------------------------------------------------------------- 7241// Effect management 7242// ---------------------------------------------------------------------------- 7243 7244 7245status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7246{ 7247 Mutex::Autolock _l(mLock); 7248 return EffectQueryNumberEffects(numEffects); 7249} 7250 7251status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7252{ 7253 Mutex::Autolock _l(mLock); 7254 return EffectQueryEffect(index, descriptor); 7255} 7256 7257status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7258 effect_descriptor_t *descriptor) const 7259{ 7260 Mutex::Autolock _l(mLock); 7261 return EffectGetDescriptor(pUuid, descriptor); 7262} 7263 7264 7265sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7266 effect_descriptor_t *pDesc, 7267 const sp<IEffectClient>& effectClient, 7268 int32_t priority, 7269 audio_io_handle_t io, 7270 int sessionId, 7271 status_t *status, 7272 int *id, 7273 int *enabled) 7274{ 7275 status_t lStatus = NO_ERROR; 7276 sp<EffectHandle> handle; 7277 effect_descriptor_t desc; 7278 7279 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7280 pid, effectClient.get(), priority, sessionId, io); 7281 7282 if (pDesc == NULL) { 7283 lStatus = BAD_VALUE; 7284 goto Exit; 7285 } 7286 7287 // check audio settings permission for global effects 7288 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7289 lStatus = PERMISSION_DENIED; 7290 goto Exit; 7291 } 7292 7293 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7294 // that can only be created by audio policy manager (running in same process) 7295 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7296 lStatus = PERMISSION_DENIED; 7297 goto Exit; 7298 } 7299 7300 if (io == 0) { 7301 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7302 // output must be specified by AudioPolicyManager when using session 7303 // AUDIO_SESSION_OUTPUT_STAGE 7304 lStatus = BAD_VALUE; 7305 goto Exit; 7306 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7307 // if the output returned by getOutputForEffect() is removed before we lock the 7308 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7309 // and we will exit safely 7310 io = AudioSystem::getOutputForEffect(&desc); 7311 } 7312 } 7313 7314 { 7315 Mutex::Autolock _l(mLock); 7316 7317 7318 if (!EffectIsNullUuid(&pDesc->uuid)) { 7319 // if uuid is specified, request effect descriptor 7320 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7321 if (lStatus < 0) { 7322 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7323 goto Exit; 7324 } 7325 } else { 7326 // if uuid is not specified, look for an available implementation 7327 // of the required type in effect factory 7328 if (EffectIsNullUuid(&pDesc->type)) { 7329 ALOGW("createEffect() no effect type"); 7330 lStatus = BAD_VALUE; 7331 goto Exit; 7332 } 7333 uint32_t numEffects = 0; 7334 effect_descriptor_t d; 7335 d.flags = 0; // prevent compiler warning 7336 bool found = false; 7337 7338 lStatus = EffectQueryNumberEffects(&numEffects); 7339 if (lStatus < 0) { 7340 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7341 goto Exit; 7342 } 7343 for (uint32_t i = 0; i < numEffects; i++) { 7344 lStatus = EffectQueryEffect(i, &desc); 7345 if (lStatus < 0) { 7346 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7347 continue; 7348 } 7349 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7350 // If matching type found save effect descriptor. If the session is 7351 // 0 and the effect is not auxiliary, continue enumeration in case 7352 // an auxiliary version of this effect type is available 7353 found = true; 7354 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7355 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7356 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7357 break; 7358 } 7359 } 7360 } 7361 if (!found) { 7362 lStatus = BAD_VALUE; 7363 ALOGW("createEffect() effect not found"); 7364 goto Exit; 7365 } 7366 // For same effect type, chose auxiliary version over insert version if 7367 // connect to output mix (Compliance to OpenSL ES) 7368 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7369 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7370 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7371 } 7372 } 7373 7374 // Do not allow auxiliary effects on a session different from 0 (output mix) 7375 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7376 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7377 lStatus = INVALID_OPERATION; 7378 goto Exit; 7379 } 7380 7381 // check recording permission for visualizer 7382 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7383 !recordingAllowed()) { 7384 lStatus = PERMISSION_DENIED; 7385 goto Exit; 7386 } 7387 7388 // return effect descriptor 7389 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7390 7391 // If output is not specified try to find a matching audio session ID in one of the 7392 // output threads. 7393 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7394 // because of code checking output when entering the function. 7395 // Note: io is never 0 when creating an effect on an input 7396 if (io == 0) { 7397 // look for the thread where the specified audio session is present 7398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7399 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7400 io = mPlaybackThreads.keyAt(i); 7401 break; 7402 } 7403 } 7404 if (io == 0) { 7405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7406 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7407 io = mRecordThreads.keyAt(i); 7408 break; 7409 } 7410 } 7411 } 7412 // If no output thread contains the requested session ID, default to 7413 // first output. The effect chain will be moved to the correct output 7414 // thread when a track with the same session ID is created 7415 if (io == 0 && mPlaybackThreads.size()) { 7416 io = mPlaybackThreads.keyAt(0); 7417 } 7418 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7419 } 7420 ThreadBase *thread = checkRecordThread_l(io); 7421 if (thread == NULL) { 7422 thread = checkPlaybackThread_l(io); 7423 if (thread == NULL) { 7424 ALOGE("createEffect() unknown output thread"); 7425 lStatus = BAD_VALUE; 7426 goto Exit; 7427 } 7428 } 7429 7430 sp<Client> client = registerPid_l(pid); 7431 7432 // create effect on selected output thread 7433 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7434 &desc, enabled, &lStatus); 7435 if (handle != 0 && id != NULL) { 7436 *id = handle->id(); 7437 } 7438 } 7439 7440Exit: 7441 if (status != NULL) { 7442 *status = lStatus; 7443 } 7444 return handle; 7445} 7446 7447status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7448 audio_io_handle_t dstOutput) 7449{ 7450 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7451 sessionId, srcOutput, dstOutput); 7452 Mutex::Autolock _l(mLock); 7453 if (srcOutput == dstOutput) { 7454 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7455 return NO_ERROR; 7456 } 7457 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7458 if (srcThread == NULL) { 7459 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7460 return BAD_VALUE; 7461 } 7462 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7463 if (dstThread == NULL) { 7464 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7465 return BAD_VALUE; 7466 } 7467 7468 Mutex::Autolock _dl(dstThread->mLock); 7469 Mutex::Autolock _sl(srcThread->mLock); 7470 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7471 7472 return NO_ERROR; 7473} 7474 7475// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7476status_t AudioFlinger::moveEffectChain_l(int sessionId, 7477 AudioFlinger::PlaybackThread *srcThread, 7478 AudioFlinger::PlaybackThread *dstThread, 7479 bool reRegister) 7480{ 7481 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7482 sessionId, srcThread, dstThread); 7483 7484 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7485 if (chain == 0) { 7486 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7487 sessionId, srcThread); 7488 return INVALID_OPERATION; 7489 } 7490 7491 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7492 // so that a new chain is created with correct parameters when first effect is added. This is 7493 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7494 // removed. 7495 srcThread->removeEffectChain_l(chain); 7496 7497 // transfer all effects one by one so that new effect chain is created on new thread with 7498 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7499 audio_io_handle_t dstOutput = dstThread->id(); 7500 sp<EffectChain> dstChain; 7501 uint32_t strategy = 0; // prevent compiler warning 7502 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7503 while (effect != 0) { 7504 srcThread->removeEffect_l(effect); 7505 dstThread->addEffect_l(effect); 7506 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7507 if (effect->state() == EffectModule::ACTIVE || 7508 effect->state() == EffectModule::STOPPING) { 7509 effect->start(); 7510 } 7511 // if the move request is not received from audio policy manager, the effect must be 7512 // re-registered with the new strategy and output 7513 if (dstChain == 0) { 7514 dstChain = effect->chain().promote(); 7515 if (dstChain == 0) { 7516 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7517 srcThread->addEffect_l(effect); 7518 return NO_INIT; 7519 } 7520 strategy = dstChain->strategy(); 7521 } 7522 if (reRegister) { 7523 AudioSystem::unregisterEffect(effect->id()); 7524 AudioSystem::registerEffect(&effect->desc(), 7525 dstOutput, 7526 strategy, 7527 sessionId, 7528 effect->id()); 7529 } 7530 effect = chain->getEffectFromId_l(0); 7531 } 7532 7533 return NO_ERROR; 7534} 7535 7536 7537// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7538sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7539 const sp<AudioFlinger::Client>& client, 7540 const sp<IEffectClient>& effectClient, 7541 int32_t priority, 7542 int sessionId, 7543 effect_descriptor_t *desc, 7544 int *enabled, 7545 status_t *status 7546 ) 7547{ 7548 sp<EffectModule> effect; 7549 sp<EffectHandle> handle; 7550 status_t lStatus; 7551 sp<EffectChain> chain; 7552 bool chainCreated = false; 7553 bool effectCreated = false; 7554 bool effectRegistered = false; 7555 7556 lStatus = initCheck(); 7557 if (lStatus != NO_ERROR) { 7558 ALOGW("createEffect_l() Audio driver not initialized."); 7559 goto Exit; 7560 } 7561 7562 // Do not allow effects with session ID 0 on direct output or duplicating threads 7563 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7564 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7565 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7566 desc->name, sessionId); 7567 lStatus = BAD_VALUE; 7568 goto Exit; 7569 } 7570 // Only Pre processor effects are allowed on input threads and only on input threads 7571 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7572 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7573 desc->name, desc->flags, mType); 7574 lStatus = BAD_VALUE; 7575 goto Exit; 7576 } 7577 7578 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7579 7580 { // scope for mLock 7581 Mutex::Autolock _l(mLock); 7582 7583 // check for existing effect chain with the requested audio session 7584 chain = getEffectChain_l(sessionId); 7585 if (chain == 0) { 7586 // create a new chain for this session 7587 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7588 chain = new EffectChain(this, sessionId); 7589 addEffectChain_l(chain); 7590 chain->setStrategy(getStrategyForSession_l(sessionId)); 7591 chainCreated = true; 7592 } else { 7593 effect = chain->getEffectFromDesc_l(desc); 7594 } 7595 7596 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7597 7598 if (effect == 0) { 7599 int id = mAudioFlinger->nextUniqueId(); 7600 // Check CPU and memory usage 7601 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7602 if (lStatus != NO_ERROR) { 7603 goto Exit; 7604 } 7605 effectRegistered = true; 7606 // create a new effect module if none present in the chain 7607 effect = new EffectModule(this, chain, desc, id, sessionId); 7608 lStatus = effect->status(); 7609 if (lStatus != NO_ERROR) { 7610 goto Exit; 7611 } 7612 lStatus = chain->addEffect_l(effect); 7613 if (lStatus != NO_ERROR) { 7614 goto Exit; 7615 } 7616 effectCreated = true; 7617 7618 effect->setDevice(mDevice); 7619 effect->setMode(mAudioFlinger->getMode()); 7620 } 7621 // create effect handle and connect it to effect module 7622 handle = new EffectHandle(effect, client, effectClient, priority); 7623 lStatus = effect->addHandle(handle.get()); 7624 if (enabled != NULL) { 7625 *enabled = (int)effect->isEnabled(); 7626 } 7627 } 7628 7629Exit: 7630 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7631 Mutex::Autolock _l(mLock); 7632 if (effectCreated) { 7633 chain->removeEffect_l(effect); 7634 } 7635 if (effectRegistered) { 7636 AudioSystem::unregisterEffect(effect->id()); 7637 } 7638 if (chainCreated) { 7639 removeEffectChain_l(chain); 7640 } 7641 handle.clear(); 7642 } 7643 7644 if (status != NULL) { 7645 *status = lStatus; 7646 } 7647 return handle; 7648} 7649 7650sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7651{ 7652 Mutex::Autolock _l(mLock); 7653 return getEffect_l(sessionId, effectId); 7654} 7655 7656sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7657{ 7658 sp<EffectChain> chain = getEffectChain_l(sessionId); 7659 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7660} 7661 7662// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7663// PlaybackThread::mLock held 7664status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7665{ 7666 // check for existing effect chain with the requested audio session 7667 int sessionId = effect->sessionId(); 7668 sp<EffectChain> chain = getEffectChain_l(sessionId); 7669 bool chainCreated = false; 7670 7671 if (chain == 0) { 7672 // create a new chain for this session 7673 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7674 chain = new EffectChain(this, sessionId); 7675 addEffectChain_l(chain); 7676 chain->setStrategy(getStrategyForSession_l(sessionId)); 7677 chainCreated = true; 7678 } 7679 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7680 7681 if (chain->getEffectFromId_l(effect->id()) != 0) { 7682 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7683 this, effect->desc().name, chain.get()); 7684 return BAD_VALUE; 7685 } 7686 7687 status_t status = chain->addEffect_l(effect); 7688 if (status != NO_ERROR) { 7689 if (chainCreated) { 7690 removeEffectChain_l(chain); 7691 } 7692 return status; 7693 } 7694 7695 effect->setDevice(mDevice); 7696 effect->setMode(mAudioFlinger->getMode()); 7697 return NO_ERROR; 7698} 7699 7700void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7701 7702 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7703 effect_descriptor_t desc = effect->desc(); 7704 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7705 detachAuxEffect_l(effect->id()); 7706 } 7707 7708 sp<EffectChain> chain = effect->chain().promote(); 7709 if (chain != 0) { 7710 // remove effect chain if removing last effect 7711 if (chain->removeEffect_l(effect) == 0) { 7712 removeEffectChain_l(chain); 7713 } 7714 } else { 7715 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7716 } 7717} 7718 7719void AudioFlinger::ThreadBase::lockEffectChains_l( 7720 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7721{ 7722 effectChains = mEffectChains; 7723 for (size_t i = 0; i < mEffectChains.size(); i++) { 7724 mEffectChains[i]->lock(); 7725 } 7726} 7727 7728void AudioFlinger::ThreadBase::unlockEffectChains( 7729 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7730{ 7731 for (size_t i = 0; i < effectChains.size(); i++) { 7732 effectChains[i]->unlock(); 7733 } 7734} 7735 7736sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7737{ 7738 Mutex::Autolock _l(mLock); 7739 return getEffectChain_l(sessionId); 7740} 7741 7742sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7743{ 7744 size_t size = mEffectChains.size(); 7745 for (size_t i = 0; i < size; i++) { 7746 if (mEffectChains[i]->sessionId() == sessionId) { 7747 return mEffectChains[i]; 7748 } 7749 } 7750 return 0; 7751} 7752 7753void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7754{ 7755 Mutex::Autolock _l(mLock); 7756 size_t size = mEffectChains.size(); 7757 for (size_t i = 0; i < size; i++) { 7758 mEffectChains[i]->setMode_l(mode); 7759 } 7760} 7761 7762void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7763 EffectHandle *handle, 7764 bool unpinIfLast) { 7765 7766 Mutex::Autolock _l(mLock); 7767 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7768 // delete the effect module if removing last handle on it 7769 if (effect->removeHandle(handle) == 0) { 7770 if (!effect->isPinned() || unpinIfLast) { 7771 removeEffect_l(effect); 7772 AudioSystem::unregisterEffect(effect->id()); 7773 } 7774 } 7775} 7776 7777status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7778{ 7779 int session = chain->sessionId(); 7780 int16_t *buffer = mMixBuffer; 7781 bool ownsBuffer = false; 7782 7783 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7784 if (session > 0) { 7785 // Only one effect chain can be present in direct output thread and it uses 7786 // the mix buffer as input 7787 if (mType != DIRECT) { 7788 size_t numSamples = mNormalFrameCount * mChannelCount; 7789 buffer = new int16_t[numSamples]; 7790 memset(buffer, 0, numSamples * sizeof(int16_t)); 7791 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7792 ownsBuffer = true; 7793 } 7794 7795 // Attach all tracks with same session ID to this chain. 7796 for (size_t i = 0; i < mTracks.size(); ++i) { 7797 sp<Track> track = mTracks[i]; 7798 if (session == track->sessionId()) { 7799 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7800 track->setMainBuffer(buffer); 7801 chain->incTrackCnt(); 7802 } 7803 } 7804 7805 // indicate all active tracks in the chain 7806 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7807 sp<Track> track = mActiveTracks[i].promote(); 7808 if (track == 0) continue; 7809 if (session == track->sessionId()) { 7810 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7811 chain->incActiveTrackCnt(); 7812 } 7813 } 7814 } 7815 7816 chain->setInBuffer(buffer, ownsBuffer); 7817 chain->setOutBuffer(mMixBuffer); 7818 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7819 // chains list in order to be processed last as it contains output stage effects 7820 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7821 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7822 // after track specific effects and before output stage 7823 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7824 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7825 // Effect chain for other sessions are inserted at beginning of effect 7826 // chains list to be processed before output mix effects. Relative order between other 7827 // sessions is not important 7828 size_t size = mEffectChains.size(); 7829 size_t i = 0; 7830 for (i = 0; i < size; i++) { 7831 if (mEffectChains[i]->sessionId() < session) break; 7832 } 7833 mEffectChains.insertAt(chain, i); 7834 checkSuspendOnAddEffectChain_l(chain); 7835 7836 return NO_ERROR; 7837} 7838 7839size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7840{ 7841 int session = chain->sessionId(); 7842 7843 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7844 7845 for (size_t i = 0; i < mEffectChains.size(); i++) { 7846 if (chain == mEffectChains[i]) { 7847 mEffectChains.removeAt(i); 7848 // detach all active tracks from the chain 7849 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7850 sp<Track> track = mActiveTracks[i].promote(); 7851 if (track == 0) continue; 7852 if (session == track->sessionId()) { 7853 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7854 chain.get(), session); 7855 chain->decActiveTrackCnt(); 7856 } 7857 } 7858 7859 // detach all tracks with same session ID from this chain 7860 for (size_t i = 0; i < mTracks.size(); ++i) { 7861 sp<Track> track = mTracks[i]; 7862 if (session == track->sessionId()) { 7863 track->setMainBuffer(mMixBuffer); 7864 chain->decTrackCnt(); 7865 } 7866 } 7867 break; 7868 } 7869 } 7870 return mEffectChains.size(); 7871} 7872 7873status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7874 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7875{ 7876 Mutex::Autolock _l(mLock); 7877 return attachAuxEffect_l(track, EffectId); 7878} 7879 7880status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7881 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7882{ 7883 status_t status = NO_ERROR; 7884 7885 if (EffectId == 0) { 7886 track->setAuxBuffer(0, NULL); 7887 } else { 7888 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7889 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7890 if (effect != 0) { 7891 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7892 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7893 } else { 7894 status = INVALID_OPERATION; 7895 } 7896 } else { 7897 status = BAD_VALUE; 7898 } 7899 } 7900 return status; 7901} 7902 7903void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7904{ 7905 for (size_t i = 0; i < mTracks.size(); ++i) { 7906 sp<Track> track = mTracks[i]; 7907 if (track->auxEffectId() == effectId) { 7908 attachAuxEffect_l(track, 0); 7909 } 7910 } 7911} 7912 7913status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7914{ 7915 // only one chain per input thread 7916 if (mEffectChains.size() != 0) { 7917 return INVALID_OPERATION; 7918 } 7919 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7920 7921 chain->setInBuffer(NULL); 7922 chain->setOutBuffer(NULL); 7923 7924 checkSuspendOnAddEffectChain_l(chain); 7925 7926 mEffectChains.add(chain); 7927 7928 return NO_ERROR; 7929} 7930 7931size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7932{ 7933 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7934 ALOGW_IF(mEffectChains.size() != 1, 7935 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7936 chain.get(), mEffectChains.size(), this); 7937 if (mEffectChains.size() == 1) { 7938 mEffectChains.removeAt(0); 7939 } 7940 return 0; 7941} 7942 7943// ---------------------------------------------------------------------------- 7944// EffectModule implementation 7945// ---------------------------------------------------------------------------- 7946 7947#undef LOG_TAG 7948#define LOG_TAG "AudioFlinger::EffectModule" 7949 7950AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7951 const wp<AudioFlinger::EffectChain>& chain, 7952 effect_descriptor_t *desc, 7953 int id, 7954 int sessionId) 7955 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7956 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7957 // mDescriptor is set below 7958 // mConfig is set by configure() and not used before then 7959 mEffectInterface(NULL), 7960 mStatus(NO_INIT), mState(IDLE), 7961 // mMaxDisableWaitCnt is set by configure() and not used before then 7962 // mDisableWaitCnt is set by process() and updateState() and not used before then 7963 mSuspended(false) 7964{ 7965 ALOGV("Constructor %p", this); 7966 int lStatus; 7967 if (thread == NULL) { 7968 return; 7969 } 7970 7971 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7972 7973 // create effect engine from effect factory 7974 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7975 7976 if (mStatus != NO_ERROR) { 7977 return; 7978 } 7979 lStatus = init(); 7980 if (lStatus < 0) { 7981 mStatus = lStatus; 7982 goto Error; 7983 } 7984 7985 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7986 return; 7987Error: 7988 EffectRelease(mEffectInterface); 7989 mEffectInterface = NULL; 7990 ALOGV("Constructor Error %d", mStatus); 7991} 7992 7993AudioFlinger::EffectModule::~EffectModule() 7994{ 7995 ALOGV("Destructor %p", this); 7996 if (mEffectInterface != NULL) { 7997 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7998 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7999 sp<ThreadBase> thread = mThread.promote(); 8000 if (thread != 0) { 8001 audio_stream_t *stream = thread->stream(); 8002 if (stream != NULL) { 8003 stream->remove_audio_effect(stream, mEffectInterface); 8004 } 8005 } 8006 } 8007 // release effect engine 8008 EffectRelease(mEffectInterface); 8009 } 8010} 8011 8012status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8013{ 8014 status_t status; 8015 8016 Mutex::Autolock _l(mLock); 8017 int priority = handle->priority(); 8018 size_t size = mHandles.size(); 8019 EffectHandle *controlHandle = NULL; 8020 size_t i; 8021 for (i = 0; i < size; i++) { 8022 EffectHandle *h = mHandles[i]; 8023 if (h == NULL || h->destroyed_l()) continue; 8024 // first non destroyed handle is considered in control 8025 if (controlHandle == NULL) 8026 controlHandle = h; 8027 if (h->priority() <= priority) break; 8028 } 8029 // if inserted in first place, move effect control from previous owner to this handle 8030 if (i == 0) { 8031 bool enabled = false; 8032 if (controlHandle != NULL) { 8033 enabled = controlHandle->enabled(); 8034 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8035 } 8036 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8037 status = NO_ERROR; 8038 } else { 8039 status = ALREADY_EXISTS; 8040 } 8041 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8042 mHandles.insertAt(handle, i); 8043 return status; 8044} 8045 8046size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8047{ 8048 Mutex::Autolock _l(mLock); 8049 size_t size = mHandles.size(); 8050 size_t i; 8051 for (i = 0; i < size; i++) { 8052 if (mHandles[i] == handle) break; 8053 } 8054 if (i == size) { 8055 return size; 8056 } 8057 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8058 8059 mHandles.removeAt(i); 8060 // if removed from first place, move effect control from this handle to next in line 8061 if (i == 0) { 8062 EffectHandle *h = controlHandle_l(); 8063 if (h != NULL) { 8064 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8065 } 8066 } 8067 8068 // Prevent calls to process() and other functions on effect interface from now on. 8069 // The effect engine will be released by the destructor when the last strong reference on 8070 // this object is released which can happen after next process is called. 8071 if (mHandles.size() == 0 && !mPinned) { 8072 mState = DESTROYED; 8073 } 8074 8075 return size; 8076} 8077 8078// must be called with EffectModule::mLock held 8079AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8080{ 8081 // the first valid handle in the list has control over the module 8082 for (size_t i = 0; i < mHandles.size(); i++) { 8083 EffectHandle *h = mHandles[i]; 8084 if (h != NULL && !h->destroyed_l()) { 8085 return h; 8086 } 8087 } 8088 8089 return NULL; 8090} 8091 8092size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8093{ 8094 ALOGV("disconnect() %p handle %p", this, handle); 8095 // keep a strong reference on this EffectModule to avoid calling the 8096 // destructor before we exit 8097 sp<EffectModule> keep(this); 8098 { 8099 sp<ThreadBase> thread = mThread.promote(); 8100 if (thread != 0) { 8101 thread->disconnectEffect(keep, handle, unpinIfLast); 8102 } 8103 } 8104 return mHandles.size(); 8105} 8106 8107void AudioFlinger::EffectModule::updateState() { 8108 Mutex::Autolock _l(mLock); 8109 8110 switch (mState) { 8111 case RESTART: 8112 reset_l(); 8113 // FALL THROUGH 8114 8115 case STARTING: 8116 // clear auxiliary effect input buffer for next accumulation 8117 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8118 memset(mConfig.inputCfg.buffer.raw, 8119 0, 8120 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8121 } 8122 start_l(); 8123 mState = ACTIVE; 8124 break; 8125 case STOPPING: 8126 stop_l(); 8127 mDisableWaitCnt = mMaxDisableWaitCnt; 8128 mState = STOPPED; 8129 break; 8130 case STOPPED: 8131 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8132 // turn off sequence. 8133 if (--mDisableWaitCnt == 0) { 8134 reset_l(); 8135 mState = IDLE; 8136 } 8137 break; 8138 default: //IDLE , ACTIVE, DESTROYED 8139 break; 8140 } 8141} 8142 8143void AudioFlinger::EffectModule::process() 8144{ 8145 Mutex::Autolock _l(mLock); 8146 8147 if (mState == DESTROYED || mEffectInterface == NULL || 8148 mConfig.inputCfg.buffer.raw == NULL || 8149 mConfig.outputCfg.buffer.raw == NULL) { 8150 return; 8151 } 8152 8153 if (isProcessEnabled()) { 8154 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8155 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8156 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8157 mConfig.inputCfg.buffer.s32, 8158 mConfig.inputCfg.buffer.frameCount/2); 8159 } 8160 8161 // do the actual processing in the effect engine 8162 int ret = (*mEffectInterface)->process(mEffectInterface, 8163 &mConfig.inputCfg.buffer, 8164 &mConfig.outputCfg.buffer); 8165 8166 // force transition to IDLE state when engine is ready 8167 if (mState == STOPPED && ret == -ENODATA) { 8168 mDisableWaitCnt = 1; 8169 } 8170 8171 // clear auxiliary effect input buffer for next accumulation 8172 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8173 memset(mConfig.inputCfg.buffer.raw, 0, 8174 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8175 } 8176 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8177 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8178 // If an insert effect is idle and input buffer is different from output buffer, 8179 // accumulate input onto output 8180 sp<EffectChain> chain = mChain.promote(); 8181 if (chain != 0 && chain->activeTrackCnt() != 0) { 8182 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8183 int16_t *in = mConfig.inputCfg.buffer.s16; 8184 int16_t *out = mConfig.outputCfg.buffer.s16; 8185 for (size_t i = 0; i < frameCnt; i++) { 8186 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8187 } 8188 } 8189 } 8190} 8191 8192void AudioFlinger::EffectModule::reset_l() 8193{ 8194 if (mEffectInterface == NULL) { 8195 return; 8196 } 8197 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8198} 8199 8200status_t AudioFlinger::EffectModule::configure() 8201{ 8202 uint32_t channels; 8203 if (mEffectInterface == NULL) { 8204 return NO_INIT; 8205 } 8206 8207 sp<ThreadBase> thread = mThread.promote(); 8208 if (thread == 0) { 8209 return DEAD_OBJECT; 8210 } 8211 8212 // TODO: handle configuration of effects replacing track process 8213 if (thread->channelCount() == 1) { 8214 channels = AUDIO_CHANNEL_OUT_MONO; 8215 } else { 8216 channels = AUDIO_CHANNEL_OUT_STEREO; 8217 } 8218 8219 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8220 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8221 } else { 8222 mConfig.inputCfg.channels = channels; 8223 } 8224 mConfig.outputCfg.channels = channels; 8225 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8226 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8227 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8228 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8229 mConfig.inputCfg.bufferProvider.cookie = NULL; 8230 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8231 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8232 mConfig.outputCfg.bufferProvider.cookie = NULL; 8233 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8234 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8235 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8236 // Insert effect: 8237 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8238 // always overwrites output buffer: input buffer == output buffer 8239 // - in other sessions: 8240 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8241 // other effect: overwrites output buffer: input buffer == output buffer 8242 // Auxiliary effect: 8243 // accumulates in output buffer: input buffer != output buffer 8244 // Therefore: accumulate <=> input buffer != output buffer 8245 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8246 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8247 } else { 8248 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8249 } 8250 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8251 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8252 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8253 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8254 8255 ALOGV("configure() %p thread %p buffer %p framecount %d", 8256 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8257 8258 status_t cmdStatus; 8259 uint32_t size = sizeof(int); 8260 status_t status = (*mEffectInterface)->command(mEffectInterface, 8261 EFFECT_CMD_SET_CONFIG, 8262 sizeof(effect_config_t), 8263 &mConfig, 8264 &size, 8265 &cmdStatus); 8266 if (status == 0) { 8267 status = cmdStatus; 8268 } 8269 8270 if (status == 0 && 8271 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8272 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8273 effect_param_t *p = (effect_param_t *)buf32; 8274 8275 p->psize = sizeof(uint32_t); 8276 p->vsize = sizeof(uint32_t); 8277 size = sizeof(int); 8278 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8279 8280 uint32_t latency = 0; 8281 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8282 if (pbt != NULL) { 8283 latency = pbt->latency_l(); 8284 } 8285 8286 *((int32_t *)p->data + 1)= latency; 8287 (*mEffectInterface)->command(mEffectInterface, 8288 EFFECT_CMD_SET_PARAM, 8289 sizeof(effect_param_t) + 8, 8290 &buf32, 8291 &size, 8292 &cmdStatus); 8293 } 8294 8295 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8296 (1000 * mConfig.outputCfg.buffer.frameCount); 8297 8298 return status; 8299} 8300 8301status_t AudioFlinger::EffectModule::init() 8302{ 8303 Mutex::Autolock _l(mLock); 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_INIT, 8311 0, 8312 NULL, 8313 &size, 8314 &cmdStatus); 8315 if (status == 0) { 8316 status = cmdStatus; 8317 } 8318 return status; 8319} 8320 8321status_t AudioFlinger::EffectModule::start() 8322{ 8323 Mutex::Autolock _l(mLock); 8324 return start_l(); 8325} 8326 8327status_t AudioFlinger::EffectModule::start_l() 8328{ 8329 if (mEffectInterface == NULL) { 8330 return NO_INIT; 8331 } 8332 status_t cmdStatus; 8333 uint32_t size = sizeof(status_t); 8334 status_t status = (*mEffectInterface)->command(mEffectInterface, 8335 EFFECT_CMD_ENABLE, 8336 0, 8337 NULL, 8338 &size, 8339 &cmdStatus); 8340 if (status == 0) { 8341 status = cmdStatus; 8342 } 8343 if (status == 0 && 8344 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8345 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8346 sp<ThreadBase> thread = mThread.promote(); 8347 if (thread != 0) { 8348 audio_stream_t *stream = thread->stream(); 8349 if (stream != NULL) { 8350 stream->add_audio_effect(stream, mEffectInterface); 8351 } 8352 } 8353 } 8354 return status; 8355} 8356 8357status_t AudioFlinger::EffectModule::stop() 8358{ 8359 Mutex::Autolock _l(mLock); 8360 return stop_l(); 8361} 8362 8363status_t AudioFlinger::EffectModule::stop_l() 8364{ 8365 if (mEffectInterface == NULL) { 8366 return NO_INIT; 8367 } 8368 status_t cmdStatus; 8369 uint32_t size = sizeof(status_t); 8370 status_t status = (*mEffectInterface)->command(mEffectInterface, 8371 EFFECT_CMD_DISABLE, 8372 0, 8373 NULL, 8374 &size, 8375 &cmdStatus); 8376 if (status == 0) { 8377 status = cmdStatus; 8378 } 8379 if (status == 0 && 8380 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8381 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8382 sp<ThreadBase> thread = mThread.promote(); 8383 if (thread != 0) { 8384 audio_stream_t *stream = thread->stream(); 8385 if (stream != NULL) { 8386 stream->remove_audio_effect(stream, mEffectInterface); 8387 } 8388 } 8389 } 8390 return status; 8391} 8392 8393status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8394 uint32_t cmdSize, 8395 void *pCmdData, 8396 uint32_t *replySize, 8397 void *pReplyData) 8398{ 8399 Mutex::Autolock _l(mLock); 8400// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8401 8402 if (mState == DESTROYED || mEffectInterface == NULL) { 8403 return NO_INIT; 8404 } 8405 status_t status = (*mEffectInterface)->command(mEffectInterface, 8406 cmdCode, 8407 cmdSize, 8408 pCmdData, 8409 replySize, 8410 pReplyData); 8411 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8412 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8413 for (size_t i = 1; i < mHandles.size(); i++) { 8414 EffectHandle *h = mHandles[i]; 8415 if (h != NULL && !h->destroyed_l()) { 8416 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8417 } 8418 } 8419 } 8420 return status; 8421} 8422 8423status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8424{ 8425 Mutex::Autolock _l(mLock); 8426 return setEnabled_l(enabled); 8427} 8428 8429// must be called with EffectModule::mLock held 8430status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8431{ 8432 8433 ALOGV("setEnabled %p enabled %d", this, enabled); 8434 8435 if (enabled != isEnabled()) { 8436 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8437 if (enabled && status != NO_ERROR) { 8438 return status; 8439 } 8440 8441 switch (mState) { 8442 // going from disabled to enabled 8443 case IDLE: 8444 mState = STARTING; 8445 break; 8446 case STOPPED: 8447 mState = RESTART; 8448 break; 8449 case STOPPING: 8450 mState = ACTIVE; 8451 break; 8452 8453 // going from enabled to disabled 8454 case RESTART: 8455 mState = STOPPED; 8456 break; 8457 case STARTING: 8458 mState = IDLE; 8459 break; 8460 case ACTIVE: 8461 mState = STOPPING; 8462 break; 8463 case DESTROYED: 8464 return NO_ERROR; // simply ignore as we are being destroyed 8465 } 8466 for (size_t i = 1; i < mHandles.size(); i++) { 8467 EffectHandle *h = mHandles[i]; 8468 if (h != NULL && !h->destroyed_l()) { 8469 h->setEnabled(enabled); 8470 } 8471 } 8472 } 8473 return NO_ERROR; 8474} 8475 8476bool AudioFlinger::EffectModule::isEnabled() const 8477{ 8478 switch (mState) { 8479 case RESTART: 8480 case STARTING: 8481 case ACTIVE: 8482 return true; 8483 case IDLE: 8484 case STOPPING: 8485 case STOPPED: 8486 case DESTROYED: 8487 default: 8488 return false; 8489 } 8490} 8491 8492bool AudioFlinger::EffectModule::isProcessEnabled() const 8493{ 8494 switch (mState) { 8495 case RESTART: 8496 case ACTIVE: 8497 case STOPPING: 8498 case STOPPED: 8499 return true; 8500 case IDLE: 8501 case STARTING: 8502 case DESTROYED: 8503 default: 8504 return false; 8505 } 8506} 8507 8508status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8509{ 8510 Mutex::Autolock _l(mLock); 8511 status_t status = NO_ERROR; 8512 8513 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8514 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8515 if (isProcessEnabled() && 8516 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8517 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8518 status_t cmdStatus; 8519 uint32_t volume[2]; 8520 uint32_t *pVolume = NULL; 8521 uint32_t size = sizeof(volume); 8522 volume[0] = *left; 8523 volume[1] = *right; 8524 if (controller) { 8525 pVolume = volume; 8526 } 8527 status = (*mEffectInterface)->command(mEffectInterface, 8528 EFFECT_CMD_SET_VOLUME, 8529 size, 8530 volume, 8531 &size, 8532 pVolume); 8533 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8534 *left = volume[0]; 8535 *right = volume[1]; 8536 } 8537 } 8538 return status; 8539} 8540 8541status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8542{ 8543 Mutex::Autolock _l(mLock); 8544 status_t status = NO_ERROR; 8545 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8546 // audio pre processing modules on RecordThread can receive both output and 8547 // input device indication in the same call 8548 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8549 if (dev) { 8550 status_t cmdStatus; 8551 uint32_t size = sizeof(status_t); 8552 8553 status = (*mEffectInterface)->command(mEffectInterface, 8554 EFFECT_CMD_SET_DEVICE, 8555 sizeof(uint32_t), 8556 &dev, 8557 &size, 8558 &cmdStatus); 8559 if (status == NO_ERROR) { 8560 status = cmdStatus; 8561 } 8562 } 8563 dev = device & AUDIO_DEVICE_IN_ALL; 8564 if (dev) { 8565 status_t cmdStatus; 8566 uint32_t size = sizeof(status_t); 8567 8568 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8569 EFFECT_CMD_SET_INPUT_DEVICE, 8570 sizeof(uint32_t), 8571 &dev, 8572 &size, 8573 &cmdStatus); 8574 if (status2 == NO_ERROR) { 8575 status2 = cmdStatus; 8576 } 8577 if (status == NO_ERROR) { 8578 status = status2; 8579 } 8580 } 8581 } 8582 return status; 8583} 8584 8585status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8586{ 8587 Mutex::Autolock _l(mLock); 8588 status_t status = NO_ERROR; 8589 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8590 status_t cmdStatus; 8591 uint32_t size = sizeof(status_t); 8592 status = (*mEffectInterface)->command(mEffectInterface, 8593 EFFECT_CMD_SET_AUDIO_MODE, 8594 sizeof(audio_mode_t), 8595 &mode, 8596 &size, 8597 &cmdStatus); 8598 if (status == NO_ERROR) { 8599 status = cmdStatus; 8600 } 8601 } 8602 return status; 8603} 8604 8605void AudioFlinger::EffectModule::setSuspended(bool suspended) 8606{ 8607 Mutex::Autolock _l(mLock); 8608 mSuspended = suspended; 8609} 8610 8611bool AudioFlinger::EffectModule::suspended() const 8612{ 8613 Mutex::Autolock _l(mLock); 8614 return mSuspended; 8615} 8616 8617bool AudioFlinger::EffectModule::purgeHandles() 8618{ 8619 bool enabled = false; 8620 Mutex::Autolock _l(mLock); 8621 for (size_t i = 0; i < mHandles.size(); i++) { 8622 EffectHandle *handle = mHandles[i]; 8623 if (handle != NULL && !handle->destroyed_l()) { 8624 handle->effect().clear(); 8625 if (handle->hasControl()) { 8626 enabled = handle->enabled(); 8627 } 8628 } 8629 } 8630 return enabled; 8631} 8632 8633status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8634{ 8635 const size_t SIZE = 256; 8636 char buffer[SIZE]; 8637 String8 result; 8638 8639 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8640 result.append(buffer); 8641 8642 bool locked = tryLock(mLock); 8643 // failed to lock - AudioFlinger is probably deadlocked 8644 if (!locked) { 8645 result.append("\t\tCould not lock Fx mutex:\n"); 8646 } 8647 8648 result.append("\t\tSession Status State Engine:\n"); 8649 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8650 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8651 result.append(buffer); 8652 8653 result.append("\t\tDescriptor:\n"); 8654 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8655 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8656 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8657 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8658 result.append(buffer); 8659 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8660 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8661 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8662 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8663 result.append(buffer); 8664 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8665 mDescriptor.apiVersion, 8666 mDescriptor.flags); 8667 result.append(buffer); 8668 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8669 mDescriptor.name); 8670 result.append(buffer); 8671 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8672 mDescriptor.implementor); 8673 result.append(buffer); 8674 8675 result.append("\t\t- Input configuration:\n"); 8676 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8677 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8678 (uint32_t)mConfig.inputCfg.buffer.raw, 8679 mConfig.inputCfg.buffer.frameCount, 8680 mConfig.inputCfg.samplingRate, 8681 mConfig.inputCfg.channels, 8682 mConfig.inputCfg.format); 8683 result.append(buffer); 8684 8685 result.append("\t\t- Output configuration:\n"); 8686 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8687 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8688 (uint32_t)mConfig.outputCfg.buffer.raw, 8689 mConfig.outputCfg.buffer.frameCount, 8690 mConfig.outputCfg.samplingRate, 8691 mConfig.outputCfg.channels, 8692 mConfig.outputCfg.format); 8693 result.append(buffer); 8694 8695 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8696 result.append(buffer); 8697 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8698 for (size_t i = 0; i < mHandles.size(); ++i) { 8699 EffectHandle *handle = mHandles[i]; 8700 if (handle != NULL && !handle->destroyed_l()) { 8701 handle->dump(buffer, SIZE); 8702 result.append(buffer); 8703 } 8704 } 8705 8706 result.append("\n"); 8707 8708 write(fd, result.string(), result.length()); 8709 8710 if (locked) { 8711 mLock.unlock(); 8712 } 8713 8714 return NO_ERROR; 8715} 8716 8717// ---------------------------------------------------------------------------- 8718// EffectHandle implementation 8719// ---------------------------------------------------------------------------- 8720 8721#undef LOG_TAG 8722#define LOG_TAG "AudioFlinger::EffectHandle" 8723 8724AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8725 const sp<AudioFlinger::Client>& client, 8726 const sp<IEffectClient>& effectClient, 8727 int32_t priority) 8728 : BnEffect(), 8729 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8730 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8731{ 8732 ALOGV("constructor %p", this); 8733 8734 if (client == 0) { 8735 return; 8736 } 8737 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8738 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8739 if (mCblkMemory != 0) { 8740 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8741 8742 if (mCblk != NULL) { 8743 new(mCblk) effect_param_cblk_t(); 8744 mBuffer = (uint8_t *)mCblk + bufOffset; 8745 } 8746 } else { 8747 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8748 return; 8749 } 8750} 8751 8752AudioFlinger::EffectHandle::~EffectHandle() 8753{ 8754 ALOGV("Destructor %p", this); 8755 8756 if (mEffect == 0) { 8757 mDestroyed = true; 8758 return; 8759 } 8760 mEffect->lock(); 8761 mDestroyed = true; 8762 mEffect->unlock(); 8763 disconnect(false); 8764} 8765 8766status_t AudioFlinger::EffectHandle::enable() 8767{ 8768 ALOGV("enable %p", this); 8769 if (!mHasControl) return INVALID_OPERATION; 8770 if (mEffect == 0) return DEAD_OBJECT; 8771 8772 if (mEnabled) { 8773 return NO_ERROR; 8774 } 8775 8776 mEnabled = true; 8777 8778 sp<ThreadBase> thread = mEffect->thread().promote(); 8779 if (thread != 0) { 8780 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8781 } 8782 8783 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8784 if (mEffect->suspended()) { 8785 return NO_ERROR; 8786 } 8787 8788 status_t status = mEffect->setEnabled(true); 8789 if (status != NO_ERROR) { 8790 if (thread != 0) { 8791 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8792 } 8793 mEnabled = false; 8794 } 8795 return status; 8796} 8797 8798status_t AudioFlinger::EffectHandle::disable() 8799{ 8800 ALOGV("disable %p", this); 8801 if (!mHasControl) return INVALID_OPERATION; 8802 if (mEffect == 0) return DEAD_OBJECT; 8803 8804 if (!mEnabled) { 8805 return NO_ERROR; 8806 } 8807 mEnabled = false; 8808 8809 if (mEffect->suspended()) { 8810 return NO_ERROR; 8811 } 8812 8813 status_t status = mEffect->setEnabled(false); 8814 8815 sp<ThreadBase> thread = mEffect->thread().promote(); 8816 if (thread != 0) { 8817 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8818 } 8819 8820 return status; 8821} 8822 8823void AudioFlinger::EffectHandle::disconnect() 8824{ 8825 disconnect(true); 8826} 8827 8828void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8829{ 8830 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8831 if (mEffect == 0) { 8832 return; 8833 } 8834 // restore suspended effects if the disconnected handle was enabled and the last one. 8835 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8836 sp<ThreadBase> thread = mEffect->thread().promote(); 8837 if (thread != 0) { 8838 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8839 } 8840 } 8841 8842 // release sp on module => module destructor can be called now 8843 mEffect.clear(); 8844 if (mClient != 0) { 8845 if (mCblk != NULL) { 8846 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8847 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8848 } 8849 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8850 // Client destructor must run with AudioFlinger mutex locked 8851 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8852 mClient.clear(); 8853 } 8854} 8855 8856status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8857 uint32_t cmdSize, 8858 void *pCmdData, 8859 uint32_t *replySize, 8860 void *pReplyData) 8861{ 8862// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8863// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8864 8865 // only get parameter command is permitted for applications not controlling the effect 8866 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8867 return INVALID_OPERATION; 8868 } 8869 if (mEffect == 0) return DEAD_OBJECT; 8870 if (mClient == 0) return INVALID_OPERATION; 8871 8872 // handle commands that are not forwarded transparently to effect engine 8873 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8874 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8875 // no risk to block the whole media server process or mixer threads is we are stuck here 8876 Mutex::Autolock _l(mCblk->lock); 8877 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8878 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8879 mCblk->serverIndex = 0; 8880 mCblk->clientIndex = 0; 8881 return BAD_VALUE; 8882 } 8883 status_t status = NO_ERROR; 8884 while (mCblk->serverIndex < mCblk->clientIndex) { 8885 int reply; 8886 uint32_t rsize = sizeof(int); 8887 int *p = (int *)(mBuffer + mCblk->serverIndex); 8888 int size = *p++; 8889 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8890 ALOGW("command(): invalid parameter block size"); 8891 break; 8892 } 8893 effect_param_t *param = (effect_param_t *)p; 8894 if (param->psize == 0 || param->vsize == 0) { 8895 ALOGW("command(): null parameter or value size"); 8896 mCblk->serverIndex += size; 8897 continue; 8898 } 8899 uint32_t psize = sizeof(effect_param_t) + 8900 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8901 param->vsize; 8902 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8903 psize, 8904 p, 8905 &rsize, 8906 &reply); 8907 // stop at first error encountered 8908 if (ret != NO_ERROR) { 8909 status = ret; 8910 *(int *)pReplyData = reply; 8911 break; 8912 } else if (reply != NO_ERROR) { 8913 *(int *)pReplyData = reply; 8914 break; 8915 } 8916 mCblk->serverIndex += size; 8917 } 8918 mCblk->serverIndex = 0; 8919 mCblk->clientIndex = 0; 8920 return status; 8921 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8922 *(int *)pReplyData = NO_ERROR; 8923 return enable(); 8924 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8925 *(int *)pReplyData = NO_ERROR; 8926 return disable(); 8927 } 8928 8929 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8930} 8931 8932void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8933{ 8934 ALOGV("setControl %p control %d", this, hasControl); 8935 8936 mHasControl = hasControl; 8937 mEnabled = enabled; 8938 8939 if (signal && mEffectClient != 0) { 8940 mEffectClient->controlStatusChanged(hasControl); 8941 } 8942} 8943 8944void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8945 uint32_t cmdSize, 8946 void *pCmdData, 8947 uint32_t replySize, 8948 void *pReplyData) 8949{ 8950 if (mEffectClient != 0) { 8951 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8952 } 8953} 8954 8955 8956 8957void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8958{ 8959 if (mEffectClient != 0) { 8960 mEffectClient->enableStatusChanged(enabled); 8961 } 8962} 8963 8964status_t AudioFlinger::EffectHandle::onTransact( 8965 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8966{ 8967 return BnEffect::onTransact(code, data, reply, flags); 8968} 8969 8970 8971void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8972{ 8973 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8974 8975 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8976 (mClient == 0) ? getpid_cached : mClient->pid(), 8977 mPriority, 8978 mHasControl, 8979 !locked, 8980 mCblk ? mCblk->clientIndex : 0, 8981 mCblk ? mCblk->serverIndex : 0 8982 ); 8983 8984 if (locked) { 8985 mCblk->lock.unlock(); 8986 } 8987} 8988 8989#undef LOG_TAG 8990#define LOG_TAG "AudioFlinger::EffectChain" 8991 8992AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8993 int sessionId) 8994 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8995 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8996 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8997{ 8998 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8999 if (thread == NULL) { 9000 return; 9001 } 9002 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9003 thread->frameCount(); 9004} 9005 9006AudioFlinger::EffectChain::~EffectChain() 9007{ 9008 if (mOwnInBuffer) { 9009 delete mInBuffer; 9010 } 9011 9012} 9013 9014// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9015sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9016{ 9017 size_t size = mEffects.size(); 9018 9019 for (size_t i = 0; i < size; i++) { 9020 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9021 return mEffects[i]; 9022 } 9023 } 9024 return 0; 9025} 9026 9027// getEffectFromId_l() must be called with ThreadBase::mLock held 9028sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9029{ 9030 size_t size = mEffects.size(); 9031 9032 for (size_t i = 0; i < size; i++) { 9033 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9034 if (id == 0 || mEffects[i]->id() == id) { 9035 return mEffects[i]; 9036 } 9037 } 9038 return 0; 9039} 9040 9041// getEffectFromType_l() must be called with ThreadBase::mLock held 9042sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9043 const effect_uuid_t *type) 9044{ 9045 size_t size = mEffects.size(); 9046 9047 for (size_t i = 0; i < size; i++) { 9048 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9049 return mEffects[i]; 9050 } 9051 } 9052 return 0; 9053} 9054 9055void AudioFlinger::EffectChain::clearInputBuffer() 9056{ 9057 Mutex::Autolock _l(mLock); 9058 sp<ThreadBase> thread = mThread.promote(); 9059 if (thread == 0) { 9060 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9061 return; 9062 } 9063 clearInputBuffer_l(thread); 9064} 9065 9066// Must be called with EffectChain::mLock locked 9067void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9068{ 9069 size_t numSamples = thread->frameCount() * thread->channelCount(); 9070 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9071 9072} 9073 9074// Must be called with EffectChain::mLock locked 9075void AudioFlinger::EffectChain::process_l() 9076{ 9077 sp<ThreadBase> thread = mThread.promote(); 9078 if (thread == 0) { 9079 ALOGW("process_l(): cannot promote mixer thread"); 9080 return; 9081 } 9082 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9083 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9084 // always process effects unless no more tracks are on the session and the effect tail 9085 // has been rendered 9086 bool doProcess = true; 9087 if (!isGlobalSession) { 9088 bool tracksOnSession = (trackCnt() != 0); 9089 9090 if (!tracksOnSession && mTailBufferCount == 0) { 9091 doProcess = false; 9092 } 9093 9094 if (activeTrackCnt() == 0) { 9095 // if no track is active and the effect tail has not been rendered, 9096 // the input buffer must be cleared here as the mixer process will not do it 9097 if (tracksOnSession || mTailBufferCount > 0) { 9098 clearInputBuffer_l(thread); 9099 if (mTailBufferCount > 0) { 9100 mTailBufferCount--; 9101 } 9102 } 9103 } 9104 } 9105 9106 size_t size = mEffects.size(); 9107 if (doProcess) { 9108 for (size_t i = 0; i < size; i++) { 9109 mEffects[i]->process(); 9110 } 9111 } 9112 for (size_t i = 0; i < size; i++) { 9113 mEffects[i]->updateState(); 9114 } 9115} 9116 9117// addEffect_l() must be called with PlaybackThread::mLock held 9118status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9119{ 9120 effect_descriptor_t desc = effect->desc(); 9121 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9122 9123 Mutex::Autolock _l(mLock); 9124 effect->setChain(this); 9125 sp<ThreadBase> thread = mThread.promote(); 9126 if (thread == 0) { 9127 return NO_INIT; 9128 } 9129 effect->setThread(thread); 9130 9131 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9132 // Auxiliary effects are inserted at the beginning of mEffects vector as 9133 // they are processed first and accumulated in chain input buffer 9134 mEffects.insertAt(effect, 0); 9135 9136 // the input buffer for auxiliary effect contains mono samples in 9137 // 32 bit format. This is to avoid saturation in AudoMixer 9138 // accumulation stage. Saturation is done in EffectModule::process() before 9139 // calling the process in effect engine 9140 size_t numSamples = thread->frameCount(); 9141 int32_t *buffer = new int32_t[numSamples]; 9142 memset(buffer, 0, numSamples * sizeof(int32_t)); 9143 effect->setInBuffer((int16_t *)buffer); 9144 // auxiliary effects output samples to chain input buffer for further processing 9145 // by insert effects 9146 effect->setOutBuffer(mInBuffer); 9147 } else { 9148 // Insert effects are inserted at the end of mEffects vector as they are processed 9149 // after track and auxiliary effects. 9150 // Insert effect order as a function of indicated preference: 9151 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9152 // another effect is present 9153 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9154 // last effect claiming first position 9155 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9156 // first effect claiming last position 9157 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9158 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9159 // already present 9160 9161 size_t size = mEffects.size(); 9162 size_t idx_insert = size; 9163 ssize_t idx_insert_first = -1; 9164 ssize_t idx_insert_last = -1; 9165 9166 for (size_t i = 0; i < size; i++) { 9167 effect_descriptor_t d = mEffects[i]->desc(); 9168 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9169 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9170 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9171 // check invalid effect chaining combinations 9172 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9173 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9174 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9175 return INVALID_OPERATION; 9176 } 9177 // remember position of first insert effect and by default 9178 // select this as insert position for new effect 9179 if (idx_insert == size) { 9180 idx_insert = i; 9181 } 9182 // remember position of last insert effect claiming 9183 // first position 9184 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9185 idx_insert_first = i; 9186 } 9187 // remember position of first insert effect claiming 9188 // last position 9189 if (iPref == EFFECT_FLAG_INSERT_LAST && 9190 idx_insert_last == -1) { 9191 idx_insert_last = i; 9192 } 9193 } 9194 } 9195 9196 // modify idx_insert from first position if needed 9197 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9198 if (idx_insert_last != -1) { 9199 idx_insert = idx_insert_last; 9200 } else { 9201 idx_insert = size; 9202 } 9203 } else { 9204 if (idx_insert_first != -1) { 9205 idx_insert = idx_insert_first + 1; 9206 } 9207 } 9208 9209 // always read samples from chain input buffer 9210 effect->setInBuffer(mInBuffer); 9211 9212 // if last effect in the chain, output samples to chain 9213 // output buffer, otherwise to chain input buffer 9214 if (idx_insert == size) { 9215 if (idx_insert != 0) { 9216 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9217 mEffects[idx_insert-1]->configure(); 9218 } 9219 effect->setOutBuffer(mOutBuffer); 9220 } else { 9221 effect->setOutBuffer(mInBuffer); 9222 } 9223 mEffects.insertAt(effect, idx_insert); 9224 9225 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9226 } 9227 effect->configure(); 9228 return NO_ERROR; 9229} 9230 9231// removeEffect_l() must be called with PlaybackThread::mLock held 9232size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9233{ 9234 Mutex::Autolock _l(mLock); 9235 size_t size = mEffects.size(); 9236 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9237 9238 for (size_t i = 0; i < size; i++) { 9239 if (effect == mEffects[i]) { 9240 // calling stop here will remove pre-processing effect from the audio HAL. 9241 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9242 // the middle of a read from audio HAL 9243 if (mEffects[i]->state() == EffectModule::ACTIVE || 9244 mEffects[i]->state() == EffectModule::STOPPING) { 9245 mEffects[i]->stop(); 9246 } 9247 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9248 delete[] effect->inBuffer(); 9249 } else { 9250 if (i == size - 1 && i != 0) { 9251 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9252 mEffects[i - 1]->configure(); 9253 } 9254 } 9255 mEffects.removeAt(i); 9256 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9257 break; 9258 } 9259 } 9260 9261 return mEffects.size(); 9262} 9263 9264// setDevice_l() must be called with PlaybackThread::mLock held 9265void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9266{ 9267 size_t size = mEffects.size(); 9268 for (size_t i = 0; i < size; i++) { 9269 mEffects[i]->setDevice(device); 9270 } 9271} 9272 9273// setMode_l() must be called with PlaybackThread::mLock held 9274void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9275{ 9276 size_t size = mEffects.size(); 9277 for (size_t i = 0; i < size; i++) { 9278 mEffects[i]->setMode(mode); 9279 } 9280} 9281 9282// setVolume_l() must be called with PlaybackThread::mLock held 9283bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9284{ 9285 uint32_t newLeft = *left; 9286 uint32_t newRight = *right; 9287 bool hasControl = false; 9288 int ctrlIdx = -1; 9289 size_t size = mEffects.size(); 9290 9291 // first update volume controller 9292 for (size_t i = size; i > 0; i--) { 9293 if (mEffects[i - 1]->isProcessEnabled() && 9294 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9295 ctrlIdx = i - 1; 9296 hasControl = true; 9297 break; 9298 } 9299 } 9300 9301 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9302 if (hasControl) { 9303 *left = mNewLeftVolume; 9304 *right = mNewRightVolume; 9305 } 9306 return hasControl; 9307 } 9308 9309 mVolumeCtrlIdx = ctrlIdx; 9310 mLeftVolume = newLeft; 9311 mRightVolume = newRight; 9312 9313 // second get volume update from volume controller 9314 if (ctrlIdx >= 0) { 9315 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9316 mNewLeftVolume = newLeft; 9317 mNewRightVolume = newRight; 9318 } 9319 // then indicate volume to all other effects in chain. 9320 // Pass altered volume to effects before volume controller 9321 // and requested volume to effects after controller 9322 uint32_t lVol = newLeft; 9323 uint32_t rVol = newRight; 9324 9325 for (size_t i = 0; i < size; i++) { 9326 if ((int)i == ctrlIdx) continue; 9327 // this also works for ctrlIdx == -1 when there is no volume controller 9328 if ((int)i > ctrlIdx) { 9329 lVol = *left; 9330 rVol = *right; 9331 } 9332 mEffects[i]->setVolume(&lVol, &rVol, false); 9333 } 9334 *left = newLeft; 9335 *right = newRight; 9336 9337 return hasControl; 9338} 9339 9340status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9341{ 9342 const size_t SIZE = 256; 9343 char buffer[SIZE]; 9344 String8 result; 9345 9346 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9347 result.append(buffer); 9348 9349 bool locked = tryLock(mLock); 9350 // failed to lock - AudioFlinger is probably deadlocked 9351 if (!locked) { 9352 result.append("\tCould not lock mutex:\n"); 9353 } 9354 9355 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9356 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9357 mEffects.size(), 9358 (uint32_t)mInBuffer, 9359 (uint32_t)mOutBuffer, 9360 mActiveTrackCnt); 9361 result.append(buffer); 9362 write(fd, result.string(), result.size()); 9363 9364 for (size_t i = 0; i < mEffects.size(); ++i) { 9365 sp<EffectModule> effect = mEffects[i]; 9366 if (effect != 0) { 9367 effect->dump(fd, args); 9368 } 9369 } 9370 9371 if (locked) { 9372 mLock.unlock(); 9373 } 9374 9375 return NO_ERROR; 9376} 9377 9378// must be called with ThreadBase::mLock held 9379void AudioFlinger::EffectChain::setEffectSuspended_l( 9380 const effect_uuid_t *type, bool suspend) 9381{ 9382 sp<SuspendedEffectDesc> desc; 9383 // use effect type UUID timelow as key as there is no real risk of identical 9384 // timeLow fields among effect type UUIDs. 9385 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9386 if (suspend) { 9387 if (index >= 0) { 9388 desc = mSuspendedEffects.valueAt(index); 9389 } else { 9390 desc = new SuspendedEffectDesc(); 9391 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9392 mSuspendedEffects.add(type->timeLow, desc); 9393 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9394 } 9395 if (desc->mRefCount++ == 0) { 9396 sp<EffectModule> effect = getEffectIfEnabled(type); 9397 if (effect != 0) { 9398 desc->mEffect = effect; 9399 effect->setSuspended(true); 9400 effect->setEnabled(false); 9401 } 9402 } 9403 } else { 9404 if (index < 0) { 9405 return; 9406 } 9407 desc = mSuspendedEffects.valueAt(index); 9408 if (desc->mRefCount <= 0) { 9409 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9410 desc->mRefCount = 1; 9411 } 9412 if (--desc->mRefCount == 0) { 9413 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9414 if (desc->mEffect != 0) { 9415 sp<EffectModule> effect = desc->mEffect.promote(); 9416 if (effect != 0) { 9417 effect->setSuspended(false); 9418 effect->lock(); 9419 EffectHandle *handle = effect->controlHandle_l(); 9420 if (handle != NULL && !handle->destroyed_l()) { 9421 effect->setEnabled_l(handle->enabled()); 9422 } 9423 effect->unlock(); 9424 } 9425 desc->mEffect.clear(); 9426 } 9427 mSuspendedEffects.removeItemsAt(index); 9428 } 9429 } 9430} 9431 9432// must be called with ThreadBase::mLock held 9433void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9434{ 9435 sp<SuspendedEffectDesc> desc; 9436 9437 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9438 if (suspend) { 9439 if (index >= 0) { 9440 desc = mSuspendedEffects.valueAt(index); 9441 } else { 9442 desc = new SuspendedEffectDesc(); 9443 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9444 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9445 } 9446 if (desc->mRefCount++ == 0) { 9447 Vector< sp<EffectModule> > effects; 9448 getSuspendEligibleEffects(effects); 9449 for (size_t i = 0; i < effects.size(); i++) { 9450 setEffectSuspended_l(&effects[i]->desc().type, true); 9451 } 9452 } 9453 } else { 9454 if (index < 0) { 9455 return; 9456 } 9457 desc = mSuspendedEffects.valueAt(index); 9458 if (desc->mRefCount <= 0) { 9459 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9460 desc->mRefCount = 1; 9461 } 9462 if (--desc->mRefCount == 0) { 9463 Vector<const effect_uuid_t *> types; 9464 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9465 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9466 continue; 9467 } 9468 types.add(&mSuspendedEffects.valueAt(i)->mType); 9469 } 9470 for (size_t i = 0; i < types.size(); i++) { 9471 setEffectSuspended_l(types[i], false); 9472 } 9473 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9474 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9475 } 9476 } 9477} 9478 9479 9480// The volume effect is used for automated tests only 9481#ifndef OPENSL_ES_H_ 9482static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9483 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9484const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9485#endif //OPENSL_ES_H_ 9486 9487bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9488{ 9489 // auxiliary effects and visualizer are never suspended on output mix 9490 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9491 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9492 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9493 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9494 return false; 9495 } 9496 return true; 9497} 9498 9499void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9500{ 9501 effects.clear(); 9502 for (size_t i = 0; i < mEffects.size(); i++) { 9503 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9504 effects.add(mEffects[i]); 9505 } 9506 } 9507} 9508 9509sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9510 const effect_uuid_t *type) 9511{ 9512 sp<EffectModule> effect = getEffectFromType_l(type); 9513 return effect != 0 && effect->isEnabled() ? effect : 0; 9514} 9515 9516void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9517 bool enabled) 9518{ 9519 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9520 if (enabled) { 9521 if (index < 0) { 9522 // if the effect is not suspend check if all effects are suspended 9523 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9524 if (index < 0) { 9525 return; 9526 } 9527 if (!isEffectEligibleForSuspend(effect->desc())) { 9528 return; 9529 } 9530 setEffectSuspended_l(&effect->desc().type, enabled); 9531 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9532 if (index < 0) { 9533 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9534 return; 9535 } 9536 } 9537 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9538 effect->desc().type.timeLow); 9539 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9540 // if effect is requested to suspended but was not yet enabled, supend it now. 9541 if (desc->mEffect == 0) { 9542 desc->mEffect = effect; 9543 effect->setEnabled(false); 9544 effect->setSuspended(true); 9545 } 9546 } else { 9547 if (index < 0) { 9548 return; 9549 } 9550 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9551 effect->desc().type.timeLow); 9552 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9553 desc->mEffect.clear(); 9554 effect->setSuspended(false); 9555 } 9556} 9557 9558#undef LOG_TAG 9559#define LOG_TAG "AudioFlinger" 9560 9561// ---------------------------------------------------------------------------- 9562 9563status_t AudioFlinger::onTransact( 9564 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9565{ 9566 return BnAudioFlinger::onTransact(code, data, reply, flags); 9567} 9568 9569}; // namespace android 9570